DTMF with RTP
Hi all,
I need to implement support for DTMF with RTP (in telephone event payload as stated in RFC 2833/RFC 4733). As I found from searching, there is no API in java that creates DTMF packets in RTP.
Therefore I have planned to generate packets (with the values for 16 bytes composing a DTMF packet) and send them over UDP.
May I know in a case I have not found, does JMF supports for sending DTMF with RTP?
If I try to achieve that as I have mention above, are there any libs or API support in JMF that I can make use of?
Thank you in advance.
regards,
Hasini.
JMF supports adding new payload types, and it's not too difficult of a process. You simply have to define your own code to packetize and depacketize your format.
Here's the sample code for how to do this...
[http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/CustomPayload.html]
Similar Messages
-
Send DTMF with SIP INFO (c2600) configuration question
I have a cisco 2600 with VIC-2FXS port as VOIP Gateway, connecting to SIP Server to receive SIP Incoming calls. I am able to receive call and the vocie has been pass through both way; and I would like the 2600 send DTMF as SIP info but was not able to do so. I have ios 12.3, and from this configuration guide http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter8.html#wp1048824 , it does not require any config for SIP info. I must missing something here, please advice. Thanks.
The config is following -
version 12.3
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
hostname abc
boot-start-marker
boot-end-marker
enable secret 5 xx
enable password xx
memory-size iomem 10
no aaa new-model
ip subnet-zero
no ip routing
no ip cef
interface Ethernet0/0
ip address 192.168.1.15 255.255.255.0
no ip route-cache
full-duplex
no ip http server
ip classless
voice-port 1/0/0
voice-port 1/0/1
voice-port 1/1/0
voice-port 1/1/1
dial-peer voice 1 pots
destination-pattern 1000
port 1/1/0
dial-peer voice 2 pots
destination-pattern 1001
port 1/1/1
dial-peer voice 10 voip
destination-pattern 1.T
session protocol sipv2
session target ipv4:192.168.1.224:5061
session transport udp
codec g711ulaw
dial-peer voice 3 pots
destination-pattern 1100
port 1/0/0
dial-peer voice 4 pots
destination-pattern 1101
port 1/0/1
line con 0
line aux 0
line vty 0 4
login
endI tried to set it, and for IOS 12.3(26) - the latest for 2610 - which dose not have that option. I use dtmf-relay rtp-nte instead; but it did not send RFC2833 event. From ethereal, no OOB events. It seems that the config I have does not have OOB DTMF enable; I compare the config I have with other examples but can not found anything wrong. Any suggestion, and what debug message I should enable, that may help to identify the issue.
Thanks.
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
interface Ethernet0/0
ip address 192.168.1.15 255.255.255.0
full-duplex
voice-port 1/0/0
voice-port 1/0/1
voice-port 1/1/0
voice-port 1/1/1
dial-peer voice 1 pots
destination-pattern 1000
port 1/1/0
dial-peer voice 2 pots
destination-pattern 1001
port 1/1/1
dial-peer voice 10 voip
description Outbound Calls
destination-pattern 1.T
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.1.250
session transport udp
dtmf-relay rtp-nte
no vad
dial-peer voice 3 pots
destination-pattern 1100
port 1/0/0
dial-peer voice 4 pots
destination-pattern 1101
port 1/0/1
dial-peer voice 100 pots
destination-pattern 8...
port 1/1/0
forward-digits 3
dial-peer voice 20 voip
description Incoming calls from PBX
incoming called-number .T
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.1.250
dtmf-relay rtp-nte
no vad -
Hi,
Can someone explain the process how to to G/R with RTP (e.g Pallets) and how can i view RTP stock.
I tried sap help but its confusing.i do't see any tcode to display RTP in stock tab in inventory mangement.
how to add line item for RTP in migo,and how to display RTP stock on your permises
lastly how can i return them to vendor.what would be the process
Thankshi,
You can use any material type like non valuated material (UNBW) important thing is only quantity update and no value update
in material type setting.
for RTP material we never create PO so no need to maintain purchasing and a/c view in MM01 for RTP material)
only maintain basic data and general plant data.
Perform MIGO for the Ordered item and save (post)
*now do the goods receipt only for RTP *
go to MIGO just select Goods receipt -against other --> enter movement type 501 M
and enter the details like material code, quantity , plant .s.loc, partner (vendor) and save
no a/c will be generated. only stock will be updated
now check stock in MMBE select special stock and execute.
just remember always use special stock M for that material for return 502M for store to store transfer 311M etc. -
Setting dimension of RTP packet with 'rtp' jmf
How is it possibile to set the dimension of RTP packet with JMF in the transmitting audio stream with RTP????
thesti wrote:
how JMF deal with RTP packet loss? since my application doesn't handle anything due to RTP packet loss, i believe that JMF has a mechanism to deal with it.It "deals" with it by having a blank spot in the rendering where that packet would have gone... -
Help with RTP: Format of Stream not supported in RTP Session Manager
Hello everyone,
I am quite new to JMF and RTP. So far I've succeeded in capturing audio from the microphone and playing it back. However, I failed when I tried to send the stream over using RTP.
Here's my program, all it does is: get a DataSource from the CaptureDevice, create a Processor with that DataSource, convert the tracks in the Processor to one of the RTP formats, and create an RTP SendStream using the Processor's output DataSource.
I can hear sound by creating a Player for the DataSource; however I get errors when I try to create RTP SendStream for the same output DataSource.
Here's my code:
CaptureDeviceInfo cdinfo;
Format fmt = new AudioFormat(AudioFormat.LINEAR, 8000, 8, 1);
Vector deviceList = CaptureDeviceManager.getDeviceList(fmt);
if (deviceList.size() > 0) {
System.out.println("Device Found.");
cdinfo = (CaptureDeviceInfo) deviceList.firstElement();
} else {
System.out.println("No device!");
return;
DataSource ds = Manager.createDataSource(cdinfo.getLocator());
Processor processor = Manager.createProcessor(ds);
StateHelper sh = new StateHelper(processor);
if (!sh.configure(10000)) {
System.out.println("Could not configure...");
System.exit(-1);
// Get the track control objects
TrackControl track[] = processor.getTrackControls();
System.out.println("Number of tracks:" + track.length);
boolean encodingPossible = false;
// Go through the tracks and try to program one of them to outout some "RTP format"
for (int i = 0; i < track.length; i++) {
try {
track.setFormat(new AudioFormat(AudioFormat.DVI_RTP));
encodingPossible = true;
} catch (Exception e) {
// cannot convert
track[i].setEnabled(false);
if (!encodingPossible) {
System.out.println("Could not encode..");
sh.close();
return;
processor.setContentDescriptor(new ContentDescriptor(ContentDescriptor.RAW));
if (!sh.realize(10000)) {
System.out.println("Could not realize...");
System.exit(-1);
System.out.println("Realized...");
DataSource outSource = processor.getDataOutput();
System.out.println(outSource.getContentType());
processor.start();
player = Manager.createRealizedPlayer(outSource);
player.start();
SessionAddress addr = new SessionAddress(InetAddress.getByName("224.144.251.104"), 8194, 4);
manager.initialize(addr);
//manager.addFormat(new AudioFormat(AudioFormat.GSM_RTP), 1);
System.out.println("RTP Session started...");
stream = manager.createSendStream(processor.getDataOutput(), 0);
I get an error on the last line, the error is: javax.media.format.UnsupportedFormatException: Format of Stream not supported in RTP Session Manager And again, if I try to encode the tracks into *AudioFormat.GSM_RTP* instead of *DVI_RTP*, I get a different error on the same line:Exception in thread "AWT-EventQueue-0" java.lang.NullPointerExceptionWell I don't understand what's happening, is there something I need to do before I can use RTP?
Hope you guys help :)Hi,
seems that you are encoding a track to RTP format but outputting a RAW format.
Your encoding section is also a little bit lazy as you don't check supported formats...
Try this between configured and realized state:
// Get the tracks from the processor
TrackControl [] tracks = processor.getTrackControls();
// Do we have at least one track?
if (tracks == null || tracks.length < 1)
return "Couldn't find tracks in processor";
// Set the output content descriptor to RAW_RTP
// This will limit the supported formats reported from
// Track.getSupportedFormats to only valid RTP formats.
ContentDescriptor cd = new ContentDescriptor(ContentDescriptor.RAW_RTP);
processor.setContentDescriptor(cd);
Format supported[];
Format chosen;
boolean atLeastOneTrack = false;
// Program the tracks.
for (int i = 0; i < tracks.length; i++) {
Format format = tracks.getFormat();
log.info("Input format for RTP conversion: " + format);
if (tracks[i].isEnabled()) {
supported = tracks[i].getSupportedFormats();
// We've set the output content to the RAW_RTP.
// So all the supported formats should work with RTP.
if (supported.length > 0) {
if (supported[i] instanceof VideoFormat) {
tracks[i].setEnabled(false);
continue;
else if (supported[i] instanceof AudioFormat) {
// set audio format for RTP transmission
chosen = new AudioFormat(AudioFormat.DVI_RTP);
tracks[i].setFormat(chosen);
tracks[i].setEnabled(true);
atLeastOneTrack = true;
else
tracks[i].setEnabled(false);
else
tracks[i].setEnabled(false);
else
tracks[i].setEnabled(false);
if (!atLeastOneTrack)
return "Couldn't set any of the tracks to a valid RTP format";
The important thing should be theContentDescriptor cd = new ContentDescriptor(ContentDescriptor.RAW_RTP);part. -
Hello,
I am trying to implement a RTP server and a RTP client, for that purpose I am using JMF, it seems to work good until I tried to transmit MP3 files. When I trasmit MP3, I get the next error message:
Unable to handle format: mpeglayer3, Unknown Sample Rate
Failed to realize: com.sun.media.PlaybackEngine@1700391
Error: Unable to realize com.sun.media.PlaybackEngine@1700391
Error in ControllerErrorEvent: javax.media.ResourceUnavailableEvent[source=com.sun.media.content.unknown.Handler@fa39d7,message=Failed to realize: input media not supported: mpeglayer3 audio]
I have installed the mp3 pluging and I have used the AVTrasmint2 and AVReceive2 class with some modifications to try to solve the problem, but I can not solve it.
I tried to use JMFStudio in both sides (server and client side) and I got the same error.
-->failed to handle a data format change!
I would like to add that I have used two differents computers (one for the server and one for the client) to make all the test.
Could you give me some advise, help or solution for that problem? If someone has a solution(code) to transmit streaming MP3 using RTP and kown that it is running ok, could you show me?
I am doubting if MP3/RTP is supported in JMF, If someone knows another alternative to implement a streaming MP3 audio server and client, could you say me what solution is it?
Thank you very much, I expect that someone could help me.
PD: Sorry for my english, I think that it is not so good.Hi I had the same problem but i manage to solve it.
First of all you have to install the mp3 plug-in and as u mentioned that u already did that.Make sure the mp3plugin.jar file is inside the lib/ext of the jre you are using for your project and one imp thing is first register the plug-in by issuing the command
java com.sun.media.codec.audio.mp3.JavaDecoder
It will modify the jmf.properties binary file.Once your done with this put this updated jmf.properties file in your project folder.After this your code should run perfectally.In case of any problem.Post it.
cheers -
CSCul50370 - C-Series TC does not send DTMF with secondary dial tone or ringback tone
This bug has also been confirmed to affect SX20 Quick Sets.
Hi Jose,
Firstly is this a UC-500 based CME? I am not aware of a 8.5 version for the UC-500 unless this is a pre-production release, as far as I am aware the UC-500 is only supported up to 8.1 at this stage.
It would also be good if you could provide a copy of your running configuration (With sensitive data removed like usernames and or passwords and peoples names and numbers), this will help us in understanding the system configuration.
It also seems that the system has not been configured with CCA as there are no identifying markers in the configuration you posted that would indicated this was a CCA build, which leads me to think that this is a CLI build right??
I am personally not very ofay with the Portugal carrier setup so I best not post up a configuration of what the Voice-Port looks like as it might not be compatible with your local carrier, but I am curious as to why your FXO ports start at 0/0/0 typically on a UC-500 system they would start at 0/1/0 as the first 4 voice ports are normally reserved for FXS (Unless of course this is different for your region), which again makes me wonder if this is a UC-500 system or maybe an ISR (2800 series or above)??
Please post up your configuration it will help out a tremendious amount.
Cheers,
David. -
Hello!! i am transmit two streams (audio and video) with avtransmit3.java (two instances). I need only record in client but when i merge two streams video and audio out of sync. How fix it?
Thanks!!
PD: Play in client is sync ok. Sorry for my english.Sorry. My carrier is Telus. My OS is v5.0.0.832.
-
Re: how can i transmit live video that is captured by a usb webcam with rtp
oh,my oicq number is:282987147
Anyone can leave a message to me.
Thanks a lot !Have you taken a look at
[email protected]
It should show you the basic principles involved
You will probably have to find additional codecs though -
RTP payload(RFC 2833) DTMF handler in JMF
hi all,
anybody tell how I receive RTP payload format vai JMF .I am able to receive DTMF through SIP INFO.
[email protected]Hi Teodor.
Thanks for your answer.
This is my dial-peer 4000:
dial-peer voice 4000 voip
service session
destination-pattern [2-9]T
rtp payload-type nte 98
voice-class codec 55
session protocol sipv2
session target ipv4:65.xxx.xxx.35
dtmf-relay rtp-nte
The voice class codec 55 puts the g729a as the preferred one.
Your answer gave me the idea where to look and found that the calls that doesn't match the dial peer 4000 and go by the default (PeerID= 0) are shown at the show call history voice command as using tx_DtmfRelay=rtp-nte
while the calls that do match the dp 4000 for an unknown reason are shown as using tx_DtmfRelay=inband-voice.
I am looking for a reason but I think it is with the supplier of the DIDs as another supplier using the same dp4000 and also G729a codec looks like using rtp-nte.
If you have any further idea please let me know.
Regards -
Hi
We have noticed that DTMF is not working with UCCX,
Call flow ->
PSTN (SIP)-> CUBE - (SIP)-> CUCM - > UCCX
UCCX DTMF is working with IP Phones which are registered with CUCM, But not working with PSTN incoming calls, also we have Unity Connection, for this DTMF is working with PSTN incoming calls,
Please help to troubleshoot this issue,
Rgds
$What this boils down to is that the CTI Ports do not support RFC2833, only out of band over the CTI QBE protocol. This means that CUCM must "see" the DTMF packets. If CUBE offers KPML in the call SDP, CUCM will automatically request this and translate from the SIP packet into a CTI event. If CUBE is only offering RFC2833, CUCM would need to invoke an MTP on the call to intercept the RTP packets containing the DTMF event.
If this isn't working the most likely reason is that there is no MTP for CUCM to invoke, or the MTP doesn't support RFC2833 DTMF interop. The best way out of this problem is to adjust the CUCM-facing VOIP dial-peer to include both capabilities: 'dtmf-relay rtp-nte sip-kpml' CUBE, assuming it's doing media flow-through, will do the heavy lifting for CUCM. If you set the DTMF Method on the CUCM-side SIP trunk, CUCM will essentially invoke an MTP for *every* call, in most cases needlessly; that's a great way to introduce problems and break things such as faxing.
Please remember to rate helpful responses and identify helpful or correct answers. -
Trouble with my T1's and E1's in the lab - please help.... :-)
I'm working through my CCIE Voice/Collaboration training materials and am just about finished with the physical construction of the lab. At this time I'm just going to install a new T1 card into my BR1 router and I'm trying to get my T1 to HQ (HQ router) and my E1 to BR2 (Branch2 router) up and running. I am enclosing the "show run", "show isdn status" and "show e1/t1 controller" outputs. I am using a 2801 for my HQ router, a 2851 for my PSTN/IP-WAN router, and a 2811 for my BR2 router.
I am using a T1 cable RJ-48C/RJ-48C. I'm embarassed to say it - but I don't have a cable tester at the time. I lended my backup out to a friend and my primary one is not working. I'm also not 100% sure that I'm using the correct cable. I have VWIC2-2MFT-T1/E1 cards in my routers and I have a 2851 (PSTN router) setup to give connectivity via the T1's to HQ and BR1 and E1 connectivity to BR2. I have taken the liberty of attaching my configs, as mentioned I don't think I have cable issues because this is the case with all my cables.
Main issue, in the "show isdn stat" the layer 1 status is "deactivated" and when I do a shut/no shut the status goes to "shutdown" and doesn't come back up despite my efforts to enable the interface. The only way to fix it is to reboot the router. I've got to be missing something - I just want to get my T1's and E1 up for my CCIE Lab. I'm building my lab based on the CCIE Voice specification and have the ability to get it modified eventually to fit the CCIE Collaboration lab.
***PLEASE go easy on me - I'm sure there is a fundamental configuration item or concept I'm not thinking about so I'm preparing to look like a fool - but that's okay....it's part of learning. :-) ***
Any help would be so much appreciated. All configs are pasted below.......
==========================================================
=================START OF BR2 CONFIG=======================
BR2_RTR#show controllers e1
E1 0/0/0 is down.
Applique type is Channelized E1 - balanced
Transmitter is sending remote alarm.
Receiver has loss of signal.
alarm-trigger is not set
Version info Firmware: 20100222, FPGA: 13, spm_count = 0
Framing is CRC4, Line Code is HDB3, Clock Source is Line.
Data in current interval (895 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 895 Unavail Secs
Total Data (last 24 hours)
0 Line Code Violations, 0 Path Code Violations,
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 86400 Unavail Secs
BR2_RTR#show isdn stat
Global ISDN Switchtype = primary-net5
ISDN Serial0/0/0:15 interface
dsl 0, interface ISDN Switchtype = primary-net5
Layer 1 Status:
DEACTIVATED
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0x00000000
Number of L2 Discards = 0, L2 Session ID = 0
Total Allocated ISDN CCBs = 0
BR2_RTR#show inventory
NAME: "2811 chassis", DESCR: "2811 chassis"
PID: CISCO2811 , VID: V06 , SN: FTX1328A0D3
NAME: "VWIC2-1MFT-T1/E1 - 1-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 SubSlot 0", DESCR: "VWIC2-1MFT-T1/E1 - 1-Port RJ-48 Multiflex Trunk - T1/E1"
PID: VWIC2-1MFT-T1/E1 , VID: V01 , SN: FOC11271UAU
NAME: "WAN Interface Card - Serial 2T on Slot 0 SubSlot 1", DESCR: "WAN Interface Card - Serial 2T"
PID: WIC-2T , VID: V01, SN: 35759031
NAME: "PVDMII DSP SIMM with three DSPs on Slot 0 SubSlot 5", DESCR: "PVDMII DSP SIMM with three DSPs"
PID: PVDM2-48 , VID: V01 , SN: FOC12221GJE
NAME: "AIM Service Engine 0", DESCR: "AIM Service Engine"
PID: AIM-CUE , VID: V03 , SN: FOC11505K9D
NAME: "16 Port 10BaseT/100BaseTX EtherSwitch on Slot 1", DESCR: "16 Port 10BaseT/100BaseTX EtherSwitch"
PID: NM-16ESW= , VID: 1.0, SN: FOC09245Q0H
NAME: "Power daughter card for 16 port EtherSwitch NM on Slot 1 SubSlot 0", DESCR: "Power daughter card for 16 port EtherSwitch NM"
PID: , VID: 1.0, SN: FOC09243VGH
NAME: "Gigabit(1000BaseT) module for EtherSwitch NM on Slot 1 SubSlot 1", DESCR: "Gigabit(1000BaseT) module for EtherSwitch NM"
PID: , VID: 1.0, SN: FOC092034R1
BR2_RTR#
BR2_RTR#
BR2_RTR#
BR2_RTR#
BR2_RTR#
BR2_RTR#show run
Building configuration...
Current configuration : 9148 bytes
! No configuration change since last restart
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname BR2_RTR
boot-start-marker
boot-end-marker
card type e1 0 0
enable secret 5 $1$kYuC$TYARPnIw8mjqiVM3CqM15.
no aaa new-model
clock timezone CET 1 0
clock summer-time CET recurring 1 Sun Apr 1:00 last Sun Oct 1:00
network-clock-participate wic 0
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 192.168.30.1 192.168.30.49
ip dhcp excluded-address 192.168.30.70 192.168.30.254
ip dhcp pool PHONES
network 192.168.30.0 255.255.255.0
default-router 192.168.30.1
option 150 ip 3.3.3.3
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-net5
voice service voip
allow-connections sip to sip
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
registrar server expires max 600 min 60
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
voice class h323 1
h225 timeout tcp establish 3
voice register global
mode cme
source-address 3.3.3.3 port 5060
max-dn 20
max-pool 10
load 7960-7940 P0S3-08-6-00
authenticate register
tftp-path flash:
create profile sync 1684632613172238
voice register dn 1
number 3005
name BR2_Phone3
voice register dn 2
number 3006
name BR2_Phone4
voice register template 1
no conference enable
voice register dialplan 1
type 7940-7960-others
pattern 1 3...
pattern 2 999
voice register pool 1
id mac 0008.E31B.7CD4
type 7960
number 1 dn 1
template 1
dtmf-relay sip-notify
username 3005 password cisco
description 3214-3005
codec g711ulaw
voice translation-rule 1
rule 1 /^\(3...$\)/ /3214\1/
voice translation-rule 2
rule 1 /^32143/ /3/
rule 2 /^\+3432143/ /3/
voice translation-rule 3000
rule 1 /^3000/ /1002/
voice translation-profile 3000
translate called 3000
voice translation-profile 4digitDNIS
translate called 2
voice translation-profile 8digitANI
translate calling 1
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2811 sn FTX1328A0D3
redundancy
controller E1 0/0/0
pri-group timeslots 1-3,16
interface Loopback0
ip address 3.3.3.3 255.255.255.255
h323-gateway voip bind srcaddr 3.3.3.3
interface FastEthernet0/0
no ip address
shutdown
duplex auto
speed auto
interface Service-Engine0/0
no ip address
interface FastEthernet0/1
no ip address
duplex auto
speed auto
interface FastEthernet0/1.21
description BR2-PHONES(RTR on a stick)
encapsulation dot1Q 21
ip address 192.168.30.1 255.255.255.0
interface FastEthernet0/1.22
description BR2-DATA(RTR on a stick)
encapsulation dot1Q 22
ip address 192.168.31.1 255.255.255.0
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bchan-number-order ascending
isdn outgoing display-ie
no cdp enable
interface Serial0/1/0
no ip address
shutdown
clock rate 2000000
interface Serial0/1/1
description BR2-RTR_IP-WAN
no ip address
encapsulation frame-relay IETF
no fair-queue
frame-relay lmi-type ansi
interface Serial0/1/1.1 point-to-point
ip address 10.1.1.2 255.255.255.128
frame-relay interface-dlci 301
interface FastEthernet1/0
description BR2-PHONE1
switchport mode trunk
switchport voice vlan 40
no ip address
spanning-tree portfast
interface FastEthernet1/1
description BR2-PHONE2
switchport mode trunk
switchport voice vlan 40
no ip address
spanning-tree portfast
interface FastEthernet1/2
no ip address
interface FastEthernet1/3
no ip address
interface FastEthernet1/4
no ip address
interface FastEthernet1/5
no ip address
interface FastEthernet1/6
no ip address
interface FastEthernet1/7
no ip address
interface FastEthernet1/8
no ip address
interface FastEthernet1/9
no ip address
interface FastEthernet1/10
no ip address
interface FastEthernet1/11
no ip address
interface FastEthernet1/12
no ip address
interface FastEthernet1/13
no ip address
interface FastEthernet1/14
no ip address
interface FastEthernet1/15
no ip address
interface GigabitEthernet1/0
no ip address
interface Vlan1
no ip address
interface Vlan30
description PHONES-VLAN-FOR-LAYER3-SWITCHING
no ip address
shutdown
interface Vlan31
description DATA-VLAN-FOR-LAYER3-SWITCHING
no ip address
shutdown
router ospf 1
network 3.3.3.3 0.0.0.0 area 0
network 10.1.1.0 0.0.0.255 area 0
network 192.168.30.0 0.0.0.255 area 0
network 192.168.31.0 0.0.0.255 area 0
network 192.168.0.0 0.0.255.255 area 0
ip forward-protocol nd
ip http server
no ip http secure-server
ip http path flash:/GUI
ip route 192.168.100.0 255.255.255.0 10.1.1.1
tftp-server flash:Desktops/320x212x12/CampusNight.png
tftp-server flash:Desktops/320x212x12/CiscoFountain.png
tftp-server flash:Desktops/320x212x12/MorroRock.png
tftp-server flash:Desktops/320x212x12/NantucketFlowers.png
tftp-server flash:Desktops/320x212x12/TN-CampusNight.png
tftp-server flash:Desktops/320x212x12/TN-CiscoFountain.png
tftp-server flash:Desktops/320x212x12/TN-Fountain.png
tftp-server flash:Desktops/320x212x12/TN-MorroRock.png
tftp-server flash:Desktops/320x212x12/TN-NantucketFlowers.png
tftp-server flash:Desktops/320x212x12/Fountain.png
tftp-server flash:Desktops/320x212x12/CiscoLogo.png
tftp-server flash:Desktops/320x212x12/TN-CiscoLogo.png
tftp-server flash:Desktops/320x212x12/List.xml
tftp-server flash:Desktops/320x216x16/List.xml
tftp-server flash:Desktops/320x212x16/List.xml
tftp-server flash:ringtones/Analog1.raw
tftp-server flash:ringtones/Analog2.raw
tftp-server flash:ringtones/AreYouThere.raw
tftp-server flash:ringtones/AreYouThereF.raw
tftp-server flash:ringtones/Bass.raw
tftp-server flash:ringtones/CallBack.raw
tftp-server flash:ringtones/Chime.raw
tftp-server flash:ringtones/Classic1.raw
tftp-server flash:ringtones/Classic2.raw
tftp-server flash:ringtones/ClockShop.raw
tftp-server flash:ringtones/DistinctiveRingList.xml
tftp-server flash:ringtones/Drums1.raw
tftp-server flash:ringtones/Drums2.raw
tftp-server flash:ringtones/FilmScore.raw
tftp-server flash:ringtones/HarpSynth.raw
tftp-server flash:ringtones/Jamaica.raw
tftp-server flash:ringtones/KotoEffect.raw
tftp-server flash:ringtones/MusicBox.raw
tftp-server flash:ringtones/Piano1.raw
tftp-server flash:ringtones/Piano2.raw
tftp-server flash:ringtones/Pop.raw
tftp-server flash:ringtones/Pulse1.raw
tftp-server flash:ringtones/Ring1.raw
tftp-server flash:ringtones/Ring2.raw
tftp-server flash:ringtones/Ring3.raw
tftp-server flash:ringtones/Ring4.raw
tftp-server flash:ringtones/Ring5.raw
tftp-server flash:ringtones/Ring6.raw
tftp-server flash:ringtones/Ring7.raw
tftp-server flash:ringtones/RingList.xml
tftp-server flash:ringtones/Sax1.raw
tftp-server flash:ringtones/Sax2.raw
tftp-server flash:ringtones/Vibe.raw
tftp-server flash:PHONE/7940-7960/P0S3-08-6-00.loads alias P0S3-08-6-00.loads
tftp-server flash:PHONE/7940-7960/P0S3-08-6-00.sb2 alias P0S3-08-6-00.sb2
tftp-server flash:PHONE/7940-7960/P0S3-08-6-00.bin alias P0S3-08-6-00.bin
tftp-server flash:PHONE/7940-7960/P0S3-08-6-00.sbn alias P0S3-08-6-00.sbn
control-plane
voice-port 0/0/0:15
translation-profile outgoing 4digitDNIS
mgcp profile default
dial-peer voice 999 pots
translation-profile outgoing 8digitANI
destination-pattern 999
port 0/0/0:15
forward-digits 3
dial-peer voice 1 voip
incoming called-number .
dial-peer voice 901134 pots
destination-pattern 901134T
port 0/0/0:15
dial-peer voice 3000 voip
translation-profile outgoing 3000
destination-pattern 3000
session target ipv4:192.168.15.23
voice-class codec 1
voice-class h323 1
telephony-service
no auto-reg-ephone
max-ephones 10
max-dn 20
ip source-address 3.3.3.3 port 2000
network-locale ES
time-format 24
date-format dd-mm-yy
max-conferences 8 gain -6
web admin system name admin password cisco
dn-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Jan 23 2014 05:43:52
ephone-template 1
softkeys connected Hold Select Trnsfer Endcall HLog Park
ephone-dn 1
number 3001
name BR2_Phone1
ephone-dn 2
number 3002
name BR2_Phone2
ephone 1
device-security-mode none
description 3214-3001
mac-address 0008.A3FD.3A32
ephone-template 1
max-calls-per-button 5
busy-trigger-per-button 3
type 7960
button 1:1
ephone 2
device-security-mode none
description 3214-3002
mac-address 0017.E0C6.E232
ephone-template 1
max-calls-per-button 5
busy-trigger-per-button 3
type 7961
button 1:2
banner motd ^CBR2 ROUTER CUCME/CUE^C
line con 0
password cisco
logging synchronous
login
line aux 0
line 194
no activation-character
no exec
transport preferred none
transport input all
transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
line vty 0 4
password cisco
login
transport input all
line vty 5 15
password cisco
login
transport input all
scheduler allocate 20000 1000
ntp server 172.30.1.2
end
===========END OF BR2 CONFIG=================
===========START OF HQ CONFIG================
HQ-RTR#show inventory
NAME: "chassis", DESCR: "2801 chassis"
PID: CISCO2801 , VID: V02 , SN: FTX1016Y07Z
NAME: "motherboard", DESCR: "C2801 Motherboard with 2 Fast Ethernet"
PID: CISCO2801 , VID: V02 , SN: FOC10140N6M
NAME: "WIC/VIC 2", DESCR: "Two port T1 voice interface daughtercard"
PID: VWIC-2MFT-T1= , VID: 1.0, SN: 32867042
NAME: "WIC/VIC/HWIC 3", DESCR: "WAN Interface Card - Serial 2T"
PID: WIC-2T= , VID: 1.0, SN: 32195023
NAME: "PVDM 0", DESCR: "PVDMII DSP SIMM with three DSPs"
PID: PVDM2-48 , VID: V01 , SN: FOC132935YB
HQ-RTR#
HQ-RTR#show controllers t1
T1 0/2/0 is down.
Applique type is Channelized T1
Cablelength is long gain36 0db
Transmitter is sending remote alarm.
Receiver has loss of signal.
alarm-trigger is not set
Soaking time: 3, Clearance time: 10
AIS State:Clear LOS State:Clear LOF State:Clear
Version info Firmware: 20090113, FPGA: 20, spm_count = 0
Framing is ESF, Line Code is B8ZS, Clock Source is Line.
CRC Threshold is 320. Reported from firmware is 320.
Data in current interval (709 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 709 Unavail Secs
Total Data (last 24 hours)
0 Line Code Violations, 0 Path Code Violations,
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 86400 Unavail Secs
T1 0/2/1 is down.
Applique type is Channelized T1
Cablelength is long gain36 0db
Transmitter is sending remote alarm.
Receiver has loss of signal.
alarm-trigger is not set
Soaking time: 3, Clearance time: 10
AIS State:Clear LOS State:Clear LOF State:Clear
Version info Firmware: 20090113, FPGA: 20, spm_count = 0
Framing is ESF, Line Code is B8ZS, Clock Source is Line.
CRC Threshold is 320. Reported from firmware is 320.
Data in current interval (709 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 709 Unavail Secs
Total Data (last 24 hours)
0 Line Code Violations, 0 Path Code Violations,
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 86400 Unavail Secs
HQ-RTR#show isdn stat
Global ISDN Switchtype = primary-ni
ISDN Serial0/2/0:23 interface
dsl 0, interface ISDN Switchtype = primary-ni
Layer 1 Status:
DEACTIVATED
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0x00000000
Number of L2 Discards = 0, L2 Session ID = 0
Total Allocated ISDN CCBs = 0
HQ-RTR#
HQ-RTR#show run
Building configuration...
Current configuration : 6734 bytes
! Last configuration change at 02:32:03 UTC Tue Feb 4 2014
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname HQ-RTR
boot-start-marker
boot-end-marker
logging buffered 512000 informational
enable secret 5 $1$K8GP$JbYRetpgnaxvy2wnjrPDW/
no aaa new-model
network-clock-participate wic 2
dot11 syslog
ip source-route
ip dhcp excluded-address 192.168.11.1 192.168.11.10
ip dhcp excluded-address 192.168.12.1 192.168.12.10
ip dhcp excluded-address 192.168.13.1 192.168.13.10
ip dhcp excluded-address 192.168.14.1 192.168.14.10
ip dhcp excluded-address 192.168.16.1 192.168.16.10
ip dhcp excluded-address 192.168.17.1 192.168.17.10
ip dhcp pool HQ-BR1-Pool
import all
network 192.168.11.0 255.255.255.0
option 150 ip 10.10.210.10
default-router 192.168.11.1
domain-name proctorlabs.com
dns-server 8.8.4.4 8.8.8.8
lease 8
ip dhcp pool BR2-Pool
import all
network 192.168.12.0 255.255.255.0
option 150 ip 10.10.202.1
default-router 192.168.12.1
domain-name proctorlabs.com
dns-server 8.8.4.4 8.8.8.8
lease 8
ip dhcp pool PSTN-Pool
import all
network 192.168.13.0 255.255.255.0
option 150 ip 10.10.100.2
default-router 192.168.13.1
domain-name proctorlabs.com
dns-server 8.8.4.4 8.8.8.8
lease 8
ip dhcp pool Laptop-Pool
import all
network 192.168.14.0 255.255.255.0
default-router 192.168.14.1
domain-name proctorlabs.com
dns-server 8.8.4.4 8.8.8.8
lease 8
ip dhcp pool WIRELESS-HOME
import all
network 192.168.16.0 255.255.255.0
default-router 192.168.16.1
dns-server 8.8.8.8 4.2.2.2
domain-name proctorlabs.com
lease 8
ip cef
no ip domain lookup
ip domain name proctorlabs.com
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-ni
voice service voip
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2801 sn FTX1016Y07Z
archive
log config
hidekeys
controller T1 0/2/0
pri-group timeslots 1-3,24
controller T1 0/2/1
interface Loopback0
ip address 1.1.1.1 255.255.255.255
interface FastEthernet0/0
description (Outside Public Interface)
ip address dhcp
ip access-group FW-IN in
no ip unreachables
ip mtu 1300
ip nat outside
ip virtual-reassembly in
duplex auto
speed auto
no cdp enable
interface FastEthernet0/1
no ip address
duplex auto
speed auto
interface FastEthernet0/1.11
description (Inside Private Interface)
encapsulation dot1Q 11
ip address 192.168.11.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/1.12
description (Inside Private Interface)
encapsulation dot1Q 12
ip address 192.168.12.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/1.13
description (Inside Private Interface)
encapsulation dot1Q 13
ip address 192.168.13.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/1.14
description (Inside Private Interface)
encapsulation dot1Q 14
ip address 192.168.14.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/1.15
description LAB-SERVERS
encapsulation dot1Q 15
ip address 192.168.15.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/1.16
description WIRELESS-HOME
encapsulation dot1Q 16
ip address 192.168.16.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/1.17
description LAB-HQ-PHONES
encapsulation dot1Q 17
ip address 192.168.17.1 255.255.255.0
ip helper-address 192.168.15.22
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/1.18
description LAB-HQ-DATA
encapsulation dot1Q 18
ip address 192.168.18.1 255.255.255.0
ip helper-address 192.168.15.22
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/1.501
description PSTN-RTR_MGMT-NETWORK
encapsulation dot1Q 501
ip address 172.30.1.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
interface Serial0/2/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn outgoing display-ie
no cdp enable
interface Serial0/3/0
description HQ-RTR_IP-WAN
no ip address
encapsulation frame-relay IETF
no fair-queue
frame-relay lmi-type ansi
interface Serial0/3/0.1 point-to-point
ip address 10.1.1.1 255.255.255.128
ip ospf mtu-ignore
snmp trap link-status
frame-relay interface-dlci 103
interface Serial0/3/0.2 point-to-point
ip address 10.1.1.129 255.255.255.128
ip ospf mtu-ignore
snmp trap link-status
frame-relay interface-dlci 102
interface Serial0/3/1
no ip address
shutdown
clock rate 2000000
router ospf 1
network 1.1.1.1 0.0.0.0 area 0
network 10.1.1.0 0.0.0.255 area 0
network 172.30.1.0 0.0.0.3 area 0
network 192.168.0.0 0.0.255.255 area 0
ip forward-protocol nd
no ip http server
no ip http secure-server
ip nat inside source list 101 interface FastEthernet0/0 overload
ip route 0.0.0.0 0.0.0.0 10.0.0.1 254
ip route 192.168.100.0 255.255.255.0 172.30.1.2
ip route 0.0.0.0 0.0.0.0 dhcp
access-list 101 deny ip 192.168.0.0 0.0.255.255 10.10.0.0 0.0.255.255
access-list 101 permit ip 192.168.0.0 0.0.255.255 any
access-list 102 permit udp any any eq bootps
access-list 102 permit udp any any eq bootpc
access-list 102 permit udp any eq bootpc any
access-list 102 permit udp any eq bootps any
disable-eadi
control-plane
voice-port 0/2/0:23
mgcp fax t38 ecm
mgcp profile default
dial-peer voice 91212 pots
description PSTN-CALLS-TO-NYC-AREA-CODE
destination-pattern 91212T
port 0/2/0:23
forward-digits all
dial-peer voice 1 pots
description INCOMING-DIAL-PEER_PSTN
incoming called-number .
direct-inward-dial
port 0/2/0:23
dial-peer voice 1000 voip
destination-pattern 2123941...
session protocol sipv2
session target ipv4:192.168.15.23
incoming called-number .
voice-class codec 1
dtmf-relay rtp-nte
no vad
dial-peer voice 1001 voip
preference 1
destination-pattern 2123941...
session protocol sipv2
session target ipv4:192.168.15.22
incoming called-number .
voice-class codec 1
dtmf-relay rtp-nte
no vad
sip-ua
retry invite 2
timers trying 300
line con 0
password cisco
logging synchronous
login
line aux 0
line vty 0 4
exec-timeout 30 0
privilege level 15
password cisco
logging synchronous
login
transport input telnet ssh
line vty 5 15
exec-timeout 30 0
privilege level 15
password cisco
logging synchronous
login
transport input telnet ssh
scheduler allocate 20000 1000
end
HQ-RTR#
=============END OF HQ CONFIG=============
=======START OF PSTN-IP-WAN_RTR CONFIG=========
PSTN_IP-WAN_RTR#show inventory
NAME: "2851 chassis", DESCR: "2851 chassis"
PID: CISCO2851 , VID: V01 , SN: FTX0922A1E7
NAME: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 SubSlot 0", DESCR: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1"
PID: VWIC2-2MFT-T1/E1 , VID: V01 , SN: FOC11063UF9
NAME: "WAN Interface Card - Serial 2T on Slot 0 SubSlot 1", DESCR: "WAN Interface Card - Serial 2T"
PID: WIC-2T , VID: V01, SN: 35845606
NAME: "Two port T1 voice interface daughtercard on Slot 0 SubSlot 2", DESCR: "Two port T1 voice interface daughtercard"
PID: VWIC-2MFT-T1= , VID: 1.0, SN: 29803060
NAME: "WAN Interface Card - Serial 2T on Slot 0 SubSlot 3", DESCR: "WAN Interface Card - Serial 2T"
PID: WIC-2T= , VID: 1.0, SN: 23188546
NAME: "PVDMII DSP SIMM with Two DSPs on Slot 0 SubSlot 4", DESCR: "PVDMII DSP SIMM with Two DSPs"
PID: PVDM2-32 , VID: V01 , SN: FOC12045356
PSTN_IP-WAN_RTR#show controllers t1
T1 0/2/0 is down.
Applique type is Channelized T1
Cablelength is long gain36 0db
Description: HQ_T1
Transmitter is sending remote alarm.
Receiver has loss of signal.
alarm-trigger is not set
Soaking time: 3, Clearance time: 10
AIS State:Clear LOS State:Clear LOF State:Clear
Version info Firmware: 20071129, FPGA: 20, spm_count = 0
Framing is ESF, Line Code is B8ZS, Clock Source is Internal.
CRC Threshold is 320. Reported from firmware is 320.
Data in current interval (852 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 852 Unavail Secs
Total Data (last 24 hours)
0 Line Code Violations, 0 Path Code Violations,
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 86400 Unavail Secs
T1 0/2/1 is down.
Applique type is Channelized T1
Cablelength is long gain36 0db
Description: BR1_T1
Transmitter is sending remote alarm.
Receiver has loss of signal.
alarm-trigger is not set
Soaking time: 3, Clearance time: 10
AIS State:Clear LOS State:Clear LOF State:Clear
Version info Firmware: 20071129, FPGA: 20, spm_count = 0
Framing is ESF, Line Code is B8ZS, Clock Source is Internal.
CRC Threshold is 320. Reported from firmware is 320.
Data in current interval (854 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 854 Unavail Secs
Total Data (last 24 hours)
0 Line Code Violations, 0 Path Code Violations,
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 86400 Unavail Secs
PSTN_IP-WAN_RTR#show controllers e1
E1 0/0/0 is down.
Applique type is Channelized E1 - balanced
Cablelength is Unknown
Description: BR2_E1
Transmitter is sending remote alarm.
Receiver has loss of signal.
alarm-trigger is not set
Version info Firmware: 20071011, FPGA: 13, spm_count = 0
Framing is CRC4, Line Code is HDB3, Clock Source is Internal.
Data in current interval (862 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 862 Unavail Secs
Total Data (last 24 hours)
0 Line Code Violations, 0 Path Code Violations,
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 86400 Unavail Secs
E1 0/0/1 is down.
Applique type is Channelized E1 - balanced
Cablelength is Unknown
Transmitter is sending remote alarm.
Receiver has loss of signal.
alarm-trigger is not set
Version info Firmware: 20071011, FPGA: 13, spm_count = 0
Framing is CRC4, Line Code is HDB3, Clock Source is Internal.
Data in current interval (864 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 864 Unavail Secs
Total Data (last 24 hours)
0 Line Code Violations, 0 Path Code Violations,
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 86400 Unavail Secs
PSTN_IP-WAN_RTR#
PSTN_IP-WAN_RTR#
PSTN_IP-WAN_RTR#show isdn status
Global ISDN Switchtype = primary-net5
ISDN Serial0/0/0:15 interface
******* Network side configuration *******
dsl 0, interface ISDN Switchtype = primary-net5
Layer 1 Status:
DEACTIVATED
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0x00000000
Number of L2 Discards = 0, L2 Session ID = 0
ISDN Serial0/0/1:15 interface
******* Network side configuration *******
dsl 1, interface ISDN Switchtype = primary-net5
Layer 1 Status:
DEACTIVATED
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 1 CCBs = 0
The Free Channel Mask: 0x00000000
Number of L2 Discards = 0, L2 Session ID = 0
ISDN Serial0/2/0:23 interface
******* Network side configuration *******
dsl 2, interface ISDN Switchtype = primary-ni
Layer 1 Status:
DEACTIVATED
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 2 CCBs = 0
The Free Channel Mask: 0x00000000
Number of L2 Discards = 0, L2 Session ID = 0
ISDN Serial0/2/1:23 interface
******* Network side configuration *******
dsl 3, interface ISDN Switchtype = primary-ni
Layer 1 Status:
DEACTIVATED
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 3 CCBs = 0
The Free Channel Mask: 0x00000000
Number of L2 Discards = 0, L2 Session ID = 0
Total Allocated ISDN CCBs = 0
PSTN_IP-WAN_RTR#
PSTN_IP-WAN_RTR#show run
Building configuration...
Current configuration : 6518 bytes
! Last configuration change at 23:02:02 CST Tue Feb 4 2014
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname PSTN_IP-WAN_RTR
boot-start-marker
boot-end-marker
card type e1 0 0
logging message-counter syslog
enable secret 5 $1$rLlG$MPPST59p5rs0FfXu8OXp1.
no aaa new-model
clock timezone CST -6
clock summer-time CDT recurring
network-clock-participate wic 0
network-clock-participate wic 2
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 192.168.100.1 192.168.100.10
ip dhcp pool PSTN-PHONE
network 192.168.100.0 255.255.255.0
default-router 192.168.100.1
option 150 ip 192.168.100.1
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
frame-relay switching
isdn switch-type primary-net5
voice translation-rule 1
rule 1 /^011\(.*\)/ /\1/
rule 2 /^1\(.*\)/ /&/
rule 3 /^00\(.*\)/ /\1/
rule 4 /^617\(.*\)/ /1&/
rule 5 /^212\(.*\)/ /1&/
voice translation-rule 2
rule 1 /^617/ /1&/
rule 2 /^212/ /1&/
voice translation-rule 3
rule 1 /^212/ /1&/
rule 2 /^34/ /&/
voice translation-rule 4
rule 1 /^617/ /1&/
rule 2 /^34/ /&/
voice translation-profile BR1-OUT
translate calling 3
voice translation-profile BR2-OUT
translate calling 2
voice translation-profile HQ-OUT
translate calling 4
voice translation-profile PSTN-IN
translate called 1
voice-card 0
crypto pki token default removal timeout 0
archive
log config
hidekeys
controller E1 0/0/0
clock source internal
pri-group timeslots 1-3,16
description BR2_E1
controller E1 0/0/1
clock source internal
pri-group timeslots 1-3,16
controller T1 0/2/0
clock source internal
pri-group timeslots 1-3,24
description HQ_T1
controller T1 0/2/1
clock source internal
pri-group timeslots 1-3,24
description BR1_T1
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.13
description PSTN-PHONE_LAN
encapsulation dot1Q 13
ip address 192.168.100.1 255.255.255.0
interface GigabitEthernet0/1
description MGMT-CONNECTION-via-WIFI
ip address 172.30.1.2 255.255.255.0
duplex auto
speed auto
interface Serial0/0/0:15
description BR2-PSTN-CONNECTION
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
interface Serial0/0/1:15
description BR2-PSTN-CONNECTION
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
interface Serial0/1/0
description FR_to_BR2-RTR
no ip address
encapsulation frame-relay IETF
clock rate 64000
frame-relay lmi-type ansi
frame-relay intf-type dce
frame-relay route 301 interface Serial0/3/0 103
interface Serial0/1/1
no ip address
shutdown
clock rate 2000000
interface Serial0/2/0:23
description HQ-PSTN-CONNECTION
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
interface Serial0/2/1:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
interface Serial0/3/0
description FR_to_HQ-RTR_point-to-point-BR1andBR2
no ip address
encapsulation frame-relay IETF
clock rate 64000
frame-relay lmi-type ansi
frame-relay intf-type dce
frame-relay route 102 interface Serial0/3/1 201
frame-relay route 103 interface Serial0/1/0 301
interface Serial0/3/1
description FR_to_BR1-RTR-to-HQ-RTR
no ip address
encapsulation frame-relay IETF
frame-relay lmi-type ansi
frame-relay intf-type dce
frame-relay route 201 interface Serial0/3/0 102
ip forward-protocol nd
ip route 1.1.1.1 255.255.255.255 172.30.1.1
ip route 2.2.2.2 255.255.255.255 172.30.1.1
ip route 3.3.3.3 255.255.255.255 172.30.1.1
ip route 10.1.1.0 255.255.255.0 172.30.1.1
ip route 192.168.14.0 255.255.255.0 172.30.1.1
ip route 192.168.15.0 255.255.255.0 172.30.1.1
ip route 192.168.16.0 255.255.255.0 172.30.1.1
ip route 192.168.17.0 255.255.255.0 172.30.1.1
ip route 192.168.20.0 255.255.255.0 172.30.1.1
ip route 192.168.21.0 255.255.255.0 172.30.1.1
ip route 192.168.30.0 255.255.255.0 172.30.1.1
ip route 192.168.31.0 255.255.255.0 172.30.1.1
no ip http server
no ip http secure-server
tftp-server flash:P0030801SR02.bin
tftp-server flash:P0030801SR02.loads
tftp-server flash:P0030801SR02.sb2
tftp-server flash:P0030801SR02.sbn
tftp-server P0030801SR02.txt
control-plane
voice-port 0/0/0:15
voice-port 0/2/0:23
voice-port 0/0/1:15
voice-port 0/2/1:23
ccm-manager fax protocol cisco
mgcp fax t38 ecm
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
dial-peer voice 10 pots
description HQ-NATIONAL-CALLS-DIAL-PEER
destination-pattern 2123941...
port 0/2/0:23
forward-digits all
dial-peer voice 20 pots
description BR1-NATIONAL-CALLS-DIAL-PEER
destination-pattern 6178632...
port 0/2/1:23
forward-digits all
dial-peer voice 30 pots
description BR2-NATIONAL-CALLS-DIAL-PEER
destination-pattern 32143...
port 0/0/0:15
forward-digits all
dial-peer voice 31 pots
description BR2-INTL-CALLS-DIAL-PEER
destination-pattern 3432143...
port 0/0/0:15
forward-digits all
telephony-service
em logout 0:0 0:0 0:0
max-ephones 2
max-dn 10
ip source-address 192.168.100.1 port 2000
load 7960-7940 P00303020214
keepalive 10
max-conferences 4 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1
number 12123945001
label +8087812321
description NYC
name NYC-PSTN
ephone-dn 2
number 16178635001
label 911+999
description BOSTON
name BOSTON-PSTN
ephone-dn 3
number 32145001
label 18005551234
description SPAIN
name SPAIN-PSTN
ephone-dn 4
number 3432145002
description SPAIN
name SPAIN-PSTN-INTL
ephone-dn 5
number 5005
label 7812321
description 7812321
ephone-dn 6
number 5006
label x5005
description OFFICE PHONE
ephone 1
device-security-mode none
mac-address 0008.A3FD.39FF
type 7960
button 1:1 2:2 3:3 4:4
button 5:5
banner motd ^CC PSTN-IP-WAN ROUTER ^C
line con 0
password cisco
logging synchronous
login
line aux 0
line vty 0 4
password cisco
login
transport input all
line vty 5 15
password cisco
login
transport input all
scheduler allocate 20000 1000
ntp master
end
PSTN_IP-WAN_RTR#I have went ahead and re-enabled the voice-ports just because I left that out of my original output. See below.....
Do you think I ordered 3 factory made T1 cables from BlackBox and ALL of them came back to me bad? Or perhaps they might not have made them as cross over cables......hmm...any other suggestions?
BR2_RTR(config)#voice-port 0/0/0:15
BR2_RTR(config-voiceport)#no shut
BR2_RTR(config-voiceport)#do sh voice port summ
BR2_RTR(config-voiceport)#do sh voice port summ
IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
=============== == ============ ===== ==== ======== ======== ==
0/0/0:15 01 isdn-voice up down none none y
0/0/0:15 02 isdn-voice up down none none y
0/0/0:15 03 isdn-voice up down none none y
50/0/1 1 efxs up dorm on-hook idle y
50/0/2 1 efxs up dorm on-hook idle y
PWR FAILOVER PORT PSTN FAILOVER PORT
================= ==================
HQ-RTR(config)#voice-port 0/2/0:23
HQ-RTR(config-voiceport)#no shut
HQ-RTR(config-voiceport)#
HQ-RTR(config-voiceport)#
HQ-RTR(config-voiceport)#do sh voice port summ
IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
=============== == ============ ===== ==== ======== ======== ==
0/2/0:23 01 isdn-voice up down none none y
0/2/0:23 02 isdn-voice up down none none y
0/2/0:23 03 isdn-voice up down none none y
PWR FAILOVER PORT PSTN FAILOVER PORT
================= ==================
PSTN_IP-WAN_RTR#conf t
Enter configuration commands, one per line. End with CNTL/Z.
PSTN_IP-WAN_RTR(config)#voice-p
PSTN_IP-WAN_RTR(config)#voice-port 0/0/0:15
PSTN_IP-WAN_RTR(config-voiceport)#no shut
PSTN_IP-WAN_RTR(config-voiceport)#exit
PSTN_IP-WAN_RTR(config)#voice-por
PSTN_IP-WAN_RTR(config)#voice-port 0/2/0:23
PSTN_IP-WAN_RTR(config-voiceport)#no shut
PSTN_IP-WAN_RTR(config-voiceport)#exit
PSTN_IP-WAN_RTR(config)#voice-por
PSTN_IP-WAN_RTR(config)#voice-port 0/0/1:15
PSTN_IP-WAN_RTR(config-voiceport)#no shut
PSTN_IP-WAN_RTR(config-voiceport)#exit
PSTN_IP-WAN_RTR(config)#voice-port 0/2/1:23
PSTN_IP-WAN_RTR(config-voiceport)#no shut
PSTN_IP-WAN_RTR(config-voiceport)#exit
PSTN_IP-WAN_RTR(config)#
PSTN_IP-WAN_RTR(config)#
PSTN_IP-WAN_RTR(config)#
PSTN_IP-WAN_RTR(config)#do sh voice port summ
IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
=============== == ============ ===== ==== ======== ======== ==
0/0/0:15 01 isdn-voice up dorm none none y
0/0/0:15 02 isdn-voice up dorm none none y
0/0/0:15 03 isdn-voice up dorm none none y
0/2/0:23 01 isdn-voice up dorm none none y
0/2/0:23 02 isdn-voice up dorm none none y
0/2/0:23 03 isdn-voice up dorm none none y
0/0/1:15 01 isdn-voice up dorm none none y
0/0/1:15 02 isdn-voice up dorm none none y
0/0/1:15 03 isdn-voice up dorm none none y
0/2/1:23 01 isdn-voice up dorm none none y
0/2/1:23 02 isdn-voice up dorm none none y
0/2/1:23 03 isdn-voice up dorm none none y
50/0/1 1 efxs up dorm on-hook idle y
50/0/2 1 efxs up dorm on-hook idle y
50/0/3 1 efxs up dorm on-hook idle y
50/0/4 1 efxs up dorm on-hook idle y
50/0/5 1 efxs up dorm on-hook idle y
50/0/6 1 efxs up up on-hook idle y
PWR FAILOVER PORT PSTN FAILOVER PORT
================= ==================
PSTN_IP-WAN_RTR(config)# -
Dear All,
I am facing issues with LPCOR configuration on CME 10.5. For International calls the Authentication Prompts triggers some times and some times doen not.
Also when a local call is dialed the Authentication Prompt is triggered some times.Below is the config and debug logs. Need your help to resolve this.
voice lpcor enable
voice lpcor custom
group 10 endusers
group 11 pstn
voice lpcor policy endusers
service fac
accept endusers fac
accept pstn fac
voice lpcor policy pstn
service fac
accept endusers fac
accept pstn fac
application
package auth
param passwd-prompt flash:enter_pin.au
param max-retries 0
param abort-digit *
param term-digit #
param user-prompt flash:enter_account.au
param passwd 12345
param max-digits 32
interface GigabitEthernet0/1.1
encapsulation dot1Q 1 native
ip address 10.25.76.1 255.255.255.0
interface GigabitEthernet0/1.201
encapsulation dot1Q 201
ip address 10.25.77.1 255.255.255.0
voice-port 0/0/0
lpcor outgoing pstn
trunk-group ALL_FXO 1
supervisory disconnect dualtone mid-call
supervisory custom-cptone 2n-gsm
no battery-reversal
input gain -6
output attenuation -3
cptone SA
timeouts call-disconnect 1
timeouts wait-release 1
timing sup-disconnect 50
connection plar 5040
caller-id enable
cable-detect
dial-peer cor custom
name local
name longdistance
name 911
name Internal
name fac-int
name user-fac
dial-peer cor list local
member local
dial-peer cor list call-local
member local
dial-peer cor list call-longdistance
member longdistance
dial-peer cor list user1
member local
member 911
dial-peer cor list user2
member local
member longdistance
member 911
member user-fac
dial-peer cor list user3
member 911
dial-peer cor list call-911
member 911
dial-peer cor list call-internal
member Internal
dial-peer cor list fac-int
member local
member 911
member fac-int
dial-peer cor list user-fac
member user-fac
dial-peer voice 96 pots
trunkgroup ALL_FXO
corlist outgoing call-911
destination-pattern 9[2-6]......
forward-digits 7
dial-peer voice 901 pots
trunkgroup ALL_FXO
corlist outgoing call-911
destination-pattern 901[2-4,6-8].......
forward-digits 10
dial-peer voice 800 pots
trunkgroup ALL_FXO
destination-pattern 9800T
prefix 800
dial-peer voice 900 pots
destination-pattern 9T
port 0/0/3
prefix 9
dial-peer voice 11 pots
destination-pattern 901........
port 0/0/3
forward-digits 10
dial-peer voice 9051 pots
trunkgroup ALL_FXO
corlist outgoing call-local
destination-pattern 905........
forward-digits 10
dial-peer voice 19 pots
trunkgroup ALL_FXO
corlist outgoing fac-int
destination-pattern 900T
translate-outgoing called 1
forward-digits all
dial-peer voice 20 voip
description International calling
service clid_authen_collect
destination-pattern 900T
lpcor outgoing pstn
session target ipv4:10.25.76.1
incoming called-number 9T
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
ephone-dn 1
number 4121
name John
corlist incoming fac-int
ephone 1
lpcor type local
lpcor incoming endusers
mac-address E0D1.730A.21DE
ephone-template 2
type 7942
button 1:1
voice register dn 33
number 4163
call-forward b2bua busy 5000
call-forward b2bua noan 5000 timeout 20
call-forward b2bua unregistered 5000
allow watch
name Joseph
mwi
voice register pool 33
busy-trigger-per-button 4
id mac BC67.1C31.C8AA
type 7821
number 1 dn 33
cor incoming fac-int 1 4163
dtmf-relay rtp-nte
codec g711ulaw
transfer max-length 4
Debug Logs
DAMAC-CME-ANOUD#DEBUg VOIce lpcor all
voip lpcor all debugging is on
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#term
DAMAC-CME-ANOUD#terminal i
DAMAC-CME-ANOUD#terminal i
Apr 12 16:22:39.825: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId F692C420E06611E4BB0CE7FDC5486EA5, SetupTime 16:22:35.615 UTC Sun Apr 12 2015, PeerAddress 4130, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:22:39.825 UTC Sun Apr 12 2015, DisconnectTime 16:22:39.825 UTC Sun Apr 12 2015, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0
Apr 12 16:22:39.825: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:04/12/2015 16:22:35.609,cgn:4130,cdn:,frs:0,fid:2599,fcid:F692C420E06611E4BB0CE7FDC5486EA5,legID:284C,bguid:F692C420E06611E4BB0CE7FDC5486EA5mon
DAMAC-CME-ANOUD#terminal imon
^
% Invalid input detected at '^' marker.
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
Apr 12 16:22:44.089: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor endusers
Apr 12 16:22:44.089: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor endusers index 10
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#show debug
VOIP LPCOR:
debug voip lpcor error call is ON (filter is OFF)
debug voip lpcor error call informational is ON (filter is OFF)
debug voip lpcor error software is ON
debug voip lpcor error software informational is ON
debug voip lpcor detail is ON (filter is OFF)
debug voip lpcor function is ON (filter is OFF)
debug voip lpcor inout is ON (filter is OFF)
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
Apr 12 16:23:22.889: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId FBA1532AE06611E4BB10E7FDC5486EA5, SetupTime 16:22:44.089 UTC Sun Apr 12 2015, PeerAddress 4130, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:23:02.009 UTC Sun Apr 12 2015, DisconnectTime 16:23:22.889 UTC Sun Apr 12 2015, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 1038, ReceiveBytes 166080
Apr 12 16:23:22.889: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:04/12/2015 16:22:44.093,cgn:4130,cdn:,frs:0,fid:2600,fcid:FBA1532AE06611E4BB10E7FDC5486EA5,legID:284D,bguid:FBA1532AE06611E4BB10E7FDC5486EA5
Apr 12 16:23:22.905: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId FBA1532AE06611E4BB10E7FDC5486EA5, SetupTime 16:22:57.795 UTC Sun Apr 12 2015, PeerAddress 0097150107659, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:23:02.015 UTC Sun Apr 12 2015, DisconnectTime 16:23:22.905 UTC Sun Apr 12 2015, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 1038, TransmitBytes 174384, ReceivePackets 1043, ReceiveBytes 166880
Apr 12 16:23:22.905: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:04/12/2015 16:22:57.785,cgn:4130,cdn:0097150107659,frs:0,fid:2601,fcid:FBA1532AE06611E4BB10E7FDC5486EA5,legID:284E,bguid:FBA1532AE06611E4BB10E7FDC5486EA5
Apr 12 16:23:25.317: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor endusers
Apr 12 16:23:25.317: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor endusers index 10
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#un all
All possible debugging has been turned off
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#!ok just send me these logs
DAMAC-CME-ANOUD#!i have to move from here
Apr 12 16:24:02.153: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId 14343755E06711E4BB16E7FDC5486EA5, SetupTime 16:23:25.323 UTC Sun Apr 12 2015, PeerAddress 4130, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:23:43.393 UTC Sun Apr 12 2015, DisconnectTime 16:24:02.153 UTC Sun Apr 12 2015, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 930, ReceiveBytes 148800
Apr 12 16:24:02.153: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:Tnow
DAMAC-CME-ANOUD#\WC,ft:04/12/2015 16:23:25.321,cgn:4130,cdn:,frs:0,fid:2602,fcid:14343755E06711E4BB16E7FDC5486EA5,legID:2850,bguid:14343755E06711E4BB16E7FDC5486EA5
Apr 12 16:24:02.169: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId 14343755E06711E4BB16E7FDC5486EA5, SetupTime 16:23:39.169 UTC Sun Apr 12 2015, PeerAddress 0097150107659, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:23:43.389 UTC Sun Apr 12 2015, DisconnectTime 16:24:02.169 UTC Sun Apr 12 2015, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 930, TransmitBytes 156240, ReceivePackets 937, ReceiveBytes 149920
Apr 12 16:24:02.169: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:04/12/2015 16:23:39.169,cgn:4130,cdn:0097150107659,frs:0,fid:2603,fcid:14343755E06711E4BB16E7FDC5486EA5,legID:2851,bguid:14343755E06711E4BB16E7FDC5486EA5We have come across this issue today in 10.9.5 (so affects 10.9.4 as well) but it was occurring in Sydney as well with a client and for me in Melbourne.
-
Issue with instant ringback when using sip trunk to SP
Hi all,
We use CUCM 8.0.2.
We have a SIP trunk to a SP connected via one of our Cisco 2911 routers configured as a CUBE.
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.0(1)M3, RELEASE SOFTWARE (fc2)
c2900-universalk9-mz.SPA.150-1.M3.bin
Cisco CISCO2911/K9 (revision 1.0)
Technology Package License Information for Module:'c2900'
Technology Technology-package
Current Type
ipbase ipbasek9 Permanent
security securityk9 Permanent
uc uck9 Permanent
data None None
We also have several ISDN lines that run out via various Cisco routers configured as H323 gateways.
We use 7945 and CIPC for our phones.
We're having an issue with calls going via the SIP trunk where we hear ringing instantly after dialling - but before the actual device at the other end starts ringing (considerable difference).
Using the SIP trunk: If I make a call to my mobile phone - I hear ringing instantly - about 3 rings before my mobile phone actually starts ringing - undesireable.
Using H323 gateway: If I make a call to my mobile phone - I hear silence for a bit - then ringing when the mobile starts ringing - desired.
Using SIP trunk: If I make a call to a landline that is ready - it rings instantly for at least 1 ring - before the actual phone I'm calling starts ringing - undesireable.
Using H323 gateway: There is a momentary pause before hearing ringing on my phone and the phone I dialled - desired.
Using SIP trunk: If I make a call to a landline that is off-hook (with no call-waiting/etc.) - it rings once and then returns the busy signal (the worst issue) - undesireable.
Using H323 gateway: There is a momentary pause before hearing busy signal - desired.
Phone to phone internally (same network): Operates as expected (instantly rings locally and on the phone I'm calling). Between phones that utilise the SIP trunk and phones that utilise the H323 gateways within the same network - communication is instant and as expected.
Any ideas why this happens and how to stop it?
I want it to not ring until the situation is known and that it can provide the appropriate feedback (ringing/busy/etc.).
Some possibly relevant config (note that there is a known bug with this IOS that meant I had to declare the codec in each dial-peer as the voice class would not work):
voice service voip
address-hiding
mode border-element
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
header-passing error-passthru
early-offer forced
midcall-signaling passthru
interface GigabitEthernet0/0
ip address x.x.x.x 255.255.255.252
ip access-group acl.SIP-IN in
no ip redirects
no ip unreachables
ip verify unicast reverse-path
ip virtual-reassembly
duplex full
speed 100
no cdp enable
gateway
timer receive-rtp 1200
sip-ua
connection-reuse
gatekeeper
shutdown
dial-peer voice 1 voip
description *** INBOUND CALLS FROM CARRIER ***
translation-profile incoming SIPTRUNK-INCOMING
session protocol sipv2
incoming called-number #blah blah#
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 61 voip
description **** WA, SA AND NT NUMBERS ****
destination-pattern 0[8]........
session protocol sipv2
session target ipv4:<MY SP's SIP SERVER>
incoming called-number 0[8]........
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 81 voip
description **** MOBILE NUMBERS ****
destination-pattern 0[4]........
session protocol sipv2
session target ipv4:<MY SP's SIP SERVER>
incoming called-number 0[4]........
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 500 voip
description *** INBOUND SIP TRUNK TO CUCM PUB ***
translation-profile outgoing SIPTRUNK-CALLING-ADD-0
preference 1
destination-pattern 5[12]..
session protocol sipv2
session target ipv4:<OUR CUCM PUBLISHER IP>
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
Any help or a point in the right direction would be greatly appreciated.
Cheers,
BrettI ended up resolving this issue as follows:
In CUCM, under Device > Device Settings > SIP Profile.
I modifed the profile relevant to my SIP trunk, under the "Trunk Specific Configuration", I set "SIP Rel1XX Options" from "Disabled" to "Send PRACK if 1xx Contains SDP".
Now, I get the expected delay before hearing ringback.
Solved! -
Please help with SIP configuration on 2801 router
Hi All.
Please help me to setup a SIP account. I’m already struggling to do that for a few days, and can’t find out how to finish that. We have 2xISDN lines running, so I need to add a SIP trunk to existing config.
The information from our SIP provider:
We have issued the following DDI range: 018877000 – 99
There is no need to register the DDI’s as these will be offered to your PABX IP address provided to in the completed SIP trunking form.
Configuration details are as follows:
Our Primary Proxy:- 99.234.56.78
Codec supported:- G711Alaw, G729 (G711Alaw is the preferred codec)
Fax Support:- T38 and G711Alaw
DTMF:- RFC2833 and INFO
CLI Method:- Remote-Party-ID
Trunk doesn’t require registration; you just need to send Invite. In cisco this is done through Dial-peer session-target command. We are authenticating your IP address for outgoing calls and incoming calls we then forward to the IP mentioned in the sip form.
This is a SIP configuration on Cisco2801 router (I used outgoing calls only):
translation-rule 10
Rule 0 ^90 0
Rule 1 ^91 1
Rule 2 ^92 2
Rule 3 ^93 3
Rule 4 ^94 4
Rule 5 ^95 5
Rule 6 ^96 6
Rule 7 ^97 7
Rule 8 ^98 8
Rule 9 ^99 9
interface FastEthernet0/0.1
description ***DATA VLAN***
encapsulation dot1Q 1 native
ip address 10.1.1.101 255.255.255.0
interface FastEthernet0/0.2
description ***VOICE VLAN***
encapsulation dot1Q 2
ip address 192.168.22.1 255.255.255.0
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
h323
call start slow
sip
bind control source-interface FastEthernet0/0.2
bind media source-interface FastEthernet0/0.2
registrar server expires max 36000 min 600
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 1 pots
description ### External Dialling via BRI ###
preference 7
destination-pattern 9T
translate-outgoing called 10
direct-inward-dial
port 0/0/0
forward-digits all
dial-peer voice 2 pots
description ### External Dialling via BRI ###
preference 2
destination-pattern 9T
translate-outgoing called 10
direct-inward-dial
port 0/0/1
forward-digits all
dial-peer voice 9000 voip
description ** Outgoing calls to SIP **
preference 1
destination-pattern 9T
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:99.234.56.78:5060
dtmf-relay rtp-nte
codec g711alaw
no vad
sip-ua
timers connect 100
sip-server ipv4:99.234.56.78
I used debugging commands to troubleshoot the calls.
2801(config-dial-peer)#
094509: Jan 24 09:27:06.204: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=211, Called Number=, Voice-Interface=0x65FA35B4,
Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
094510: Jan 24 09:27:06.204: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=20018
094511: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9, Peer Info Type=DIALPEER_INFO_SPEECH
094512: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9
094513: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094514: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094515: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90, Peer Info Type=DIALPEER_INFO_SPEECH
094516: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90
094517: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094518: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094519: Jan 24 09:27:06.912: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=908, Peer Info Type=DIALPEER_INFO_SPEECH
094520: Jan 24 09:27:06.912: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=908
094521: Jan 24 09:27:06.916: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094522: Jan 24 09:27:06.916: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094523: Jan 24 09:27:07.012: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9086, Peer Info Type=DIALPEER_INFO_SPEECH
094524: Jan 24 09:27:07.012: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9086
094525: Jan 24 09:27:07.016: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094526: Jan 24 09:27:07.016: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094527: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862, Peer Info Type=DIALPEER_INFO_SPEECH
094528: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862
094529: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094530: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094531: Jan 24 09:27:07.212: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=908621, Peer Info Type=DIALPEER_INFO_SPEECH
094532: Jan 24 09:27:07.212: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=908621
094533: Jan 24 09:27:07.216: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094534: Jan 24 09:27:07.216: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094535: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9086215, Peer Info Type=DIALPEER_INFO_SPEECH
094536: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9086215
094537: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094538: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094539: Jan 24 09:27:07.412: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862157, Peer Info Type=DIALPEER_INFO_SPEECH
094540: Jan 24 09:27:07.412: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157
094541: Jan 24 09:27:07.416: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094542: Jan 24 09:27:07.416: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094543: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=908621577, Peer Info Type=DIALPEER_INFO_SPEECH
094544: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=908621577
094545: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094546: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094547: Jan 24 09:27:07.612: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9086215777, Peer Info Type=DIALPEER_INFO_SPEECH
094548: Jan 24 09:27:07.612: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9086215777
094549: Jan 24 09:27:07.616: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094550: Jan 24 09:27:07.616: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094551: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094552: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774
094553: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094554: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094555: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH
094556: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774T
094557: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
094558: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=9000
2: Dial-peer Tag=2
3: Dial-peer Tag=1
094559: Jan 24 09:27:10.711: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094560: Jan 24 09:27:10.711: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774
094561: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
094562: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=9000
2: Dial-peer Tag=2
3: Dial-peer Tag=1
094563: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
094564: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
094565: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094566: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774
094567: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
094568: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=9000
2: Dial-peer Tag=2
3: Dial-peer Tag=1
094569: Jan 24 09:27:10.719: fb_get_reject_cause_code: ERROR cause_code NULL
094570: Jan 24 09:27:10.727: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam " <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:10 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397230
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094571: Jan 24 09:27:11.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:11 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397231
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094572: Jan 24 09:27:12.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam " <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:12 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397232
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094573: Jan 24 09:27:14.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam" <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:14 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397234
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
I made some changes in the router configuration.
I removed FA0/0.2 Voice interface from Voice service voip configuration (bind control source-interface FastEthernet0/0.2 and bind media source-interface FastEthernet0/0.2). And now it’s using ip address 10.1.1.101 (data ip).
The debugging is changed now. I can send and receive a respond from SIP server. But It shows an error: SIP/2.0 404 Not Found
Then it moves to ISDN line, and use this line to make a call.
102988: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH
102989: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774T
102990: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
102991: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=9000
2: Dial-peer Tag=2
3: Dial-peer Tag=1
102992: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
102993: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774
102994: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
102995: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=9000
2: Dial-peer Tag=2
3: Dial-peer Tag=1
102996: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
102997: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
102998: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
102999: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774
103000: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
103001: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=9000
2: Dial-peer Tag=2
3: Dial-peer Tag=1
103002: Jan 24 14:45:47.298: fb_get_reject_cause_code: ERROR cause_code NULL
103003: Jan 24 14:45:47.310: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Seam" <sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 14:45:47 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 3989446920-1171263969-2466545983-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327416347
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 2438 9821 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 19412 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
103004: Jan 24 14:45:47.354: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
From: "Sam "<sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142
Call-ID: [email protected]
CSeq: 101 INVITE
Via: SIP/2.0/UDP 10.1.1.101:5060;received=88.99.77.44;branch=z9hG4bK4875CB9
Content-Length: 0
103005: Jan 24 14:45:47.362: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9
From: "Sam " <sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142
Date: Tue, 24 Jan 2012 14:45:47 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
103006: Jan 24 14:45:47.374: %ISDN-6-LAYER2UP: Layer 2 for Interface BR0/0/1, TEI 96 changed to up
103007: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=, Called Number=211, Peer Info Type=DIALPEER_INFO_SPEECH
103008: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=211
103009: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
103010: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20018
103011: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=, Called Number=0862157774, Peer Info Type=DIALPEER_INFO_SPEECH
103012: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=0862157774
103013: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
103014: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
Result=NO_MATCH(-1)
103015: Jan 24 14:46:08.815: %ISDN-6-LAYER2DOWN: Layer 2 for Interface BR0/0/1, TEI 96 changed to down
2801(config-dial-peer)#
Then I removed SIP-UA as I was told there is no registration necessary, only Dial-peer configuration.
But it didn’t affect anything.
Then I add translate-outgoing called 10 command to dial-peer 9000, nothing happened.
Really stuck and don't know where to look at.
Any help will be highly appreciated.
Thanks.Hi Dan.
Yes, I saw that RTP debugging, but what can I change there? Maybe I need to open more ports on ASA for RTP like 19412?
I use Cisco ASDM for ASA to make changes.
There are static NAT rules for: Server source IPs(10.1.1.100) to Outside(translated IPs, 88.99.77.44) for a few ports.
Also I added Security policy access rules for LAN: Any to SIP, and Outside: SIP to any.
For NAT:
I can't add this: for LAN: STATIC ROUTER IP 10.1.1.101 (AS SOURCE) UDP 5060 TO OUTSIDE IP 88.99.77.44
(AS TRANSLATED) UDP 5060
Because there is already translation for the Server.
Debugging looks like that now. There is no Received: SIP/2.0, but I can make an outside call with no audio.
116013: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
116014: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
116015: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
116016: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774
116017: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
116018: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=9000
2: Dial-peer Tag=2
3: Dial-peer Tag=1
116019: Jan 25 15:28:25.588: fb_get_reject_cause_code: ERROR cause_code NULL
116020: Jan 25 15:28:25.600: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:25 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505305
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116021: Jan 25 15:28:26.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:26 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505306
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116022: Jan 25 15:28:27.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:27 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505307
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116026: Jan 25 15:28:57.092: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam" ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:57 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389preference 1771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505337
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
I'll add Incoming dial-peer now.
Not sure what kind of NAT rule should I put into ASA to allow in and out sip traffic.
Appretiate your help.
Thanks a mill.
Maybe you are looking for
-
How do i create gallery with text under thumbnails?
Hi. Im new in Muse and im trying to learn. My next challenge is to make thumbnails with text under like this: And when visitors click a thumbnail it should pop up the original size picture in a lightbox. A mouseover big pic should also work. Like thi
-
FCP Project File & other Editing programs
G'day, I apologise if this has been brought up earlier but this is a question I've had for a while. FCP projects obviously maintain the file extension .fcp for the file projects. My question is can FCP open, use and edit other file projects such as t
-
-encoding type ? Pls, help me !
Could everybody pls answer my question : How many -encoding types used with javac ? (I search in java tool document but don't find the full list of -encoding type) I try to use javac -encoding "UTF16", "unicode" and compile: OK ! When I use javac -en
-
Why does clicking on a tab submit the page? When I hover the mouse over a tab, I see javascriopt:doSubmit('TAB_NAME'). Why is this? What if I want to discard all changes made on my current page and move to the tab target? As a UI concept, arent tabs
-
Hi friends, I have found in 4.6 c profile option in coois t.code.in ecc6.0 i found layout in coois. what is the difference? how to create profiles and how to create layouts.is thery any configuration settings.please help me with