DIAL UP CONFIGURATION ON SOLRIS 10 [INTEL]
I would like to know the steps to configure DIAL UP CONNECTION ON intel base solaris 10.
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http://www.linuxquestions.org/questions/showthread.php?t=419165
Similar Messages
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Site to Site calling issue - Cisco 2911 Dial Peer Configuration
My customer dials from remote site to main site to their main site number, the call by-passes their auto attendant and goes directly to any random available party.
At first fingers were pointing to the their PBX, however we noticed one of their sites that wasn't managed by our company did not have the issue. We cut that site over to our service and the issue started right up. I believe it is possibly due to the way the dial peers are configured and how the calls route into the PBX. Unfortunately I do not understand much about them and curious to know if anyone has any history on a issue similiar to this or any input whatsoever?
Cisco equipment/Dialpeer config below ........
co IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.2(4)M4, RELEASE SOFTWARE (fc2) - Cisco CISCO2911/K9
dial-peer voice 100 voip
description --- VoIP Dial-Peer ---
translation-profile outgoing 7digit
huntstop
preference 1
service session
destination-pattern .T
progress_ind setup enable 3
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 99
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate 14400
fax nsf 000000
ip qos dscp af41 signaling
no vad
dial-peer voice 150 voip
permission none
description 900 block
huntstop
destination-pattern 1900T
session protocol sipv2
session target sip-server
voice-class codec 99
dtmf-relay rtp-nte
ip qos dscp af41 signaling
no vad
dial-peer voice 151 voip
permission none
description 900 block
huntstop
destination-pattern 900T
session protocol sipv2
session target sip-server
voice-class codec 99
dtmf-relay rtp-nte
ip qos dscp af41 signaling
no vad
dial-peer voice 101 pots
description --- INCOMING Calls from PBX ---
incoming called-number .T
direct-inward-dial
dial-peer voice 1001 pots
description --- Calls to the PBX ---
preference 3
destination-pattern .T
port 0/0/1:23
forward-digits 4
Here is some ISDN debug information
BAD CALL
Protocol Profile = Networking Extensions
0xA11C0201420201008014484152545F20484F54454C535F434C4159544F4E
Component = Invoke component
Invoke Id = 66
Operation = CallingName
Name Presentation Allowed Extended
Name = XXXXXXXXXXX
Display i = ''XXXXXXXXXXX''
Calling Party Number i = 0x2180, ''XXXXXXXXXX''
Plan:ISDN, Type:National
Called Party Number i = 0x80, ''6551''
Plan:Unknown, Type:Unknown
Aug 19 16:10:47.242 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB15
Channel ID i = 0xA98381
Exclusive, Channel 1
Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: RX <- CONNECT pd = 8 callref = 0xAB15
Channel ID i = 0xA98381
Exclusive, Channel 1
Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x2B15
GOOD CALL
Protocol Profile = Networking Extensions
0xA116020144020100800E475245454E204D4F554E5441494E
Component = Invoke component
Invoke Id = 68
Operation = CallingName
Name Presentation Allowed Extended
Name = XXXXXXXXXXXXXXXXXX
Display i = ''XXXXXXXXXXX''
Calling Party Number i = 0x2180, ''XXXXXXXXXX''
Plan:ISDN, Type:National
Called Party Number i = 0x80, 'XXXX''
Plan:Unknown, Type:Unknown
Aug 19 16:15:07.999 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB17
Channel ID i = 0xA98381
Exclusive, Channel 1I done the configration via CCA and the running conf i can see two voip dial peer. this is the site where all trunk line roured. Customer from other site2 needs to call outside by taking line from site1.
dial-peer voice 2100 voip
corlist incoming call-internal
description **CCA*INTERSITE inbound call to SITE 1
translation-profile incoming multisiteInbound
incoming called-number 82...
voice-class h323 1
dtmf-relay h245-alphanumeric
fax protocol cisco
no vad
dial-peer voice 2101 voip
corlist incoming call-internal
description **CCA*INTERSITE outbound calls to SITE2
translation-profile outgoing multisiteOutbound
destination-pattern 81...
session target ipv4:192.168.50.1
voice-class h323 1
dtmf-relay h245-alphanumeric
fax protocol cisco
no vad
no dial-peer outbound status-check pots -
DX650 cannot show caller ID name after speed dial is configured
Hi all,
Once I replaced my IP phone from CP-7975 with DX650, I found the incoming caller name display is missing for those caller phone numbers are configured as "one-button speed dial" on the DX650. For other callers calling to DX650, the caller name can be displayed correctly.
Anyone encounters this issue before and how to solve it?? Right now, DX650 is not mapped with the google account and other directory setting.
thanks,
samuelHello,
to resolve your issue go to home screen of DX650 and click Contact , then click serch icon , enter the name of contact and aftar that click copy in my Conctacts Button.
Rollback to phone icon in home screen of DX650 ,click add new speed dial then selec your contact , click save .
and make a test , your probleme we will resolved.
Regards -
Dial Up Configuration - Please Reply...
Hi Friends,
I am absolutely new to Dial Up technologies, I wanted to know basic configuration details without AAA for the same.
My setup is something like:
Cisco 2811 with NM-8AM-V2 this will be dialing into a Cisco 1800 connected to a Multitech modem (Callback feature) via PSTN line.
Kindly requesting to please share some similar configurations for both the Cisco routers.
Thanks
Cheers,
~sultanHi,
Try the following link....
http://www.cisco.com/en/US/tech/tk801/tk36/technologies_configuration_example09186a0080093d2a.shtml
That should help with the basic config.
Hope it helps!
Cheers -
Time based dial plan configuration
Hello experts!
We're trying to maximize security on our VOIP Gateway to avoid being victimized by long distance/international toll fraud. In efforts to address this concern, we're looking to somehow deploy a time based dial plan on our gateway (2821) which based on that it automatically shutdown outbound international dial peer for any calls made during off work hours (i.e 7pm- 7am including weekends):
dial-peer voice 119 pots
destination-pattern 01T
port 0/0/0:23
forward-digits 16
I understand this can be done inside Callmanager (we're running Callmanager 4.1(3)) as well but for extra security pre-caution we'd like to have it on our gateway preferably. Is there any way to accomplish this task on a 2821 VOIP Gateway?
Thanks,You can use kron command.
This is an example:
kron occurrence NIGHT at 20:00 recurring
policy-list SHUT_DIALPEER
kron occurrence DAY at 8:00 recurring
policy-list NOSHUT_DIALPEER
kron policy-list SHUT_DIALPEER
cli dial-peer voice 119 pots
cli shutdown
kron policy-list NOSHUT_DIALPEER
cli dial-peer voice 119 pots
cli no shutdown
Regards. -
At&t business internet services dial-up configuration
Hi
Rarely, but at times when traveling, I require to connect to Internet via dial-up. I use Mac's USB modem, which dialed up after first install no problem, but I could not connect to Web pages via Safari. After some fussing around here's what worked for me (at&t business internet services, basic plan, credit card customer):
System Preferences>Network
Telephone: enter your access number WITHOUT hyphens or spaces, "202xxxxxxx," for example
Account Name: internet.usinet.xxxxx where xxxxx is your actual account user ID
Click "Apply"
Click Connect
Surf the Web.
(Note to at&t, which now offers wi-fi hotspots at Starbucks & McDonald's, but its proprietary dialer software is only supported by Windows' OS--this is not good!; support Mac OS)Great info Flowers. Though your note to AT&T should be directed at them directly.
-
Analog door phone to dial phones configuration
I have a remote office that I'm trying to setup CME. Everything works for now except one thing. There is a analog phone installed on outside of front door for visitors, deliveries, etc. Trying to setup this phone to ring automatically reception desk and also enable for others to answer the line from their phone when there is no one available in the reception area. I was thinking of connecting this door phone to one of the FXS ports on the router and configure PLAR to ring reception desk phone. Any suggestion? how do I setup other phones to be able to answer/pick up when no one at the reception desk? Pickup group? Thanks.
Thanks for your reply kkoeper12. I have configured the following. Did I get this right? I really can't test it at the moment. Just want to check first. Thanks.
voice hunt-group 1 sequential
final 2212
list 2212,2211,2213,2214,2202,2203,2204,2205
timeout 30
pilot 2227
voice-port 0/2/0 ---- this is where I have a analog door phone connected.
connection plar 2227
ephone-dn 26 dual-line
number 2227 -
[W520] Intel AMT missing in BIOS configuration
Hello everyone,
I've been trying to get the Intel Active Management working, but I'm running into some problems trying to enable it.
According to Intel, my CPU supports vPro (http://ark.intel.com/products/50067/Intel-Core-i7-2720QM-Processor-%286M-Cache-2_20-GHz%29) and I can successfully use all the vPro features, except for Intel AMT. Here are my system specifications:
http://i41.tinypic.com/jgs18p.png
My BIOS does have the Intel Management Extensions and I've updated both my BIOS/UEFI and the ME to the latest firmware (1.35 and 7.1.30.1142, respectively). This is the output from MEInfoWin.exe:
http://i41.tinypic.com/2s8s31v.png
The weird thing is that I can access the AMT management interface at http://localhost:16992, but all that is shown is this:
http://i39.tinypic.com/358ny3k.png
The Intel AMT diagnostics tool shows that AMT is supported, but disabled. Here are the results from the tool:
http://ompldr.org/vY2QyYQ/Results.nfo
Does anyone know why this happens? Does my W520 need to be serviced?
Thanks in advance!@christopherlang
Agreed. I ordered with only one hard drive since I already had a spare only to find out that my motherboard doesn't support RAID.
After some searching a little bit, it seems that Lenovo will include AMT only when configured with an Intel wireless card, which I didn't order since my broken W510 has one. They should really make clear which features are included and which aren't. Honestly, I can't believe they would take the time and effort to make 8 different motherboards just to remove features .
Same as you, other than that, I really like this laptop. -
How to configure a virtual dial-peer destination pattern?
There is a virtual dial peer, 22501, that is configured with a destination pattern. When that call comes in the (H323) gateway from the PRI, it, of, course fails. There is a voip dial peer, 301, of 8345.... which it's supposed to hit. But the virtual dial peer is a specific match. How do I take that destination pattern off of dial peer 22501? I can't get in the dial peer like the other, normal, dial peers? I get "invalid command" when trying. Need help getting it out.
301 voip up up 8345.... 0 syst ipv4:10.208.11.251
89900- voip up up 0 syst 000
98765- voip up up 0 syst 4
91919- voip up up 0 syst 191
92929- voip up up 0 syst 292
22501 pots up up 83452342$ 0 50/0/1
22502 pots up down 1 50/0/2
22503 pots up down 0 50/0/3Hi Anthony,
The voice-ports start from 50/0/x created when we configure an ephone-dn on the CME.
Since, i do not see any CME configuration on your gateway, that means you must be using this as SRST.
When IP phones registered on the CUCM loose connectivity, and they register to SRST, these voice ports and dial-peer are dynamically created. And the extension on the IP phone is automatically configured at destination pattern.
Please check if this IP phone(with extension 83452342) is still registered in the SRST mode.
If not, then probably the dynamic configuration has not been washed out completely.
There is no way you can enter in these dial-peer configuration and remove it.
At the moment, you need to reload the gateway to remove this configuration.
Hope this helps.
~Amit -
[SOLVED] Intel DRI does not work after update
Hi,
after update of kernel from 3.5.6 to 3.6.5 and intel driver and deletion of dri2proto and mesa via pacman -Qdt (there are no more dependencies).. acceleration is no longer working
$ export LIBGL_DEBUG=verbose; glxinfo | grep renderer
libGL: OpenDriver: trying /usr/lib/xorg/modules/dri/tls/i965_dri.so
libGL: OpenDriver: trying /usr/lib/xorg/modules/dri/i965_dri.so
libGL error: failed to open drm device: Permission denied
libGL error: failed to load driver: i965
libGL: OpenDriver: trying /usr/lib/xorg/modules/dri/tls/swrast_dri.so
libGL: OpenDriver: trying /usr/lib/xorg/modules/dri/swrast_dri.so
libGL: Can't open configuration file /etc/drirc: No such file or directory.
OpenGL renderer string: Gallium 0.4 on llvmpipe (LLVM 0x301)
Configuration:
SandyBridge I5, Intel HD 3000
Can somebody help?
Last edited by microcz (2012-11-07 06:15:23)OT: A simpler way to check if some packages are installed:
$ pacman -Q intel-dri libgl foobar
intel-dri 9.0-1
libgl 9.0-1
error: package 'foobar' was not found
What's the output of 'groups'? Maybe you need to add yourself to 'video' group?
http://www.linuxquestions.org/questions … nf-794105/
http://www.mail-archive.com/dri-devel@l … 41313.html -
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. -
Dial-Peer matches but fails to call out
Hello,
Am trying to get my CME configured for Callcentric. I have both an inbound and an outbound plan.
With my dial-peers configured for standard 11-digit and 10-digit dialing, calls go to fast busy after all digits except the last two are dialed. Debug shows a dial-peer match initially, then states no match and the call fails. If I change the destination pattern to match my cell phone number exactly, I can dial all the digits but the call still fails. Anyone have a suggestion?
Here are my dial peers:
dial-peer voice 700 voip
description SIP Trunk - Incoming
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target dns:callcentric.com
incoming called-number .%
dial-peer voice 701 voip
description SIP Trunk - Outgoing 3-Digit Calls
translation-profile outgoing SIP_1
preference 1
destination-pattern 9[2-8]11
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
dial-peer voice 702 voip
description SIP Trunk - Outgoing 11-Digit Calls
translation-profile outgoing SIP_1
preference 1
destination-pattern 91[2-9].......
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
dial-peer voice 703 voip
description SIP Trunk - Outgoing 10-Digit Calls
translation-profile outgoing SIP_1
preference 1
destination-pattern 9[2-9].......
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
no vad
And here is the debug associated with a call:
*Dec 26 22:26:27.854: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=7018, Called Number=, Voice-Interface=0x4A4AE7B0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:27.854: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=20009
GMIT-VOICEROUTER01#
*Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9
*Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=91, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=91
*Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Numbe
GMIT-VOICEROUTr=, Called Number=912, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=912
*Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9120, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120
*Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:30.822: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=91207, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:30.822: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=91207
*Dec 26 22:26:30.826: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:30.826: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:31.342: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=912072, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:31.342: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=912072
*Dec 26 22:26:31.346: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:31.346: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:31.542: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9120722, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:31.546: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120722
*Dec 26 22:26:31.546: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:31.546: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=91207227, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=91207227
*Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:32.602: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=912072277, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:32.602: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=912072277
*Dec 26 22:26:32.602: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:32.606: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:33.382: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.382: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.382: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=702
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=702
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=702
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=9120722776, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=702
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=9120722776, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=702
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=702
*Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=702
*Dec 26 22:26:33.574: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=91[2-9]......., Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.578: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
*Dec 26 22:26:36.374: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=7018$, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:36.378: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules AttemptER01#Translation profile:
voice translation-rule 3
rule 1 /^7../ /2072267262/
voice translation-rule 4
rule 1 /^9\(1....\)/ /\1/
rule 2 /^9207\(...\)/ /\1/
rule 3 /^9\(011.*\)/ /\1/
rule 4 /^9\([2-9]11\)/ /\1/
voice translation-profile SIP_1
translate calling 3
translate called 4
Here is debug ccsip messages:
*Dec 27 14:10:16.598: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 207.5.178.214:5060;branch=z9hG4bKd77b793048igqgkfd0g1.1
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
Max-Forwards: 69
Call-ID: SDp6vve01-196593d11e4bf68c71f8a4085d7de7d0-c54gcb0
From: ;tag=SDp6vve01-callagent.gwi.net+1+8bfe3a+d3cf6d84
CSeq: 938331054 OPTIONS
Organization: MetaSwitch
Supported: resource-priority, 100rel
Content-Length: 0
Contact:
To:
*Dec 27 14:10:16.606: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 207.5.178.214:5060;branch=z9hG4bKd77b793048igqgkfd0g1.1
From: ;tag=SDp6vve01-callagent.gwi.net+1+8bfe3a+d3cf6d84
To:
GMIT-VOICEROUT166>;tag=F1B5120-18BD
Date: Fri, 27 Dec 2013 14:10:16 GMT
Call-ID: SDp6vve01-196593d11e4bf68c71f8a4085d7de7d0-c54gcb0
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 938331054 OPTIONS
Supported: 100rel,resource-priority,replaces,sdp-anat
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 172
v=0
o=CiscoSystemsSIP-GW-UserAgent 4484 7548 IN IP4 66.55.220.166
s=SIP Call
c=IN IP4 66.55.220.166
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 66.55.220.166
ER01#
GMIT-VOICEROUTER01#
*Dec 27 14:10:34.834: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5080 SIP/2.0
Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK5795232C
From: "Server Room" [email protected]>;tag=F1B9854-8A5
To: [email protected]>
Date: Fri, 27 Dec 2013 14:10:34 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2066961728-1849102819-2185007278-567139419
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, B
GMIT-VOICEROUTER01#YE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1388153434
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 297
v=0
o=CiscoSystemsSIP-GW-UserAgent 3022 9963 IN IP4 66.55.220.166
s=SIP Call
c=IN IP4 66.55.220.166
t=0 0
m=audio 19258 RTP/AVP 18 101 19
c=IN IP4 66.55.220.166
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20
*Dec 27 14:10:34.906: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK5795232C
f: "Server Room" [email protected]>;tag=F1B9854-8A5
t: [email protected]>
i: [email protected]
CSeq: 1
GMIT-VOICEROUT01 INVITE
Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="3755ae79fd668c2035ebb90cdc12d030", opaque="", stale=TRUE, algorithm=MD5
l: 0
*Dec 27 14:10:34.914: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5080 SIP/2.0
Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK5795232C
From: "Server Room" [email protected]>;tag=F1B9854-8A5
To: [email protected]>
Date: Fri, 27 Dec 2013 14:10:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
*Dec 27 14:10:34.914: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5080 SIP/2.0
Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK579619F9
From: "Server Room" [email protected]>;tag=F1B9854-8A5
To: [email protected]>
Date: Fri, 27 Dec 2013 14:10:34 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2066961728-1849102819-2185007278-567139419
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1388153434
Contact:
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="17772882353",realm="callcentric.com",uri="sip:[email protected]:5080",response="cbac03a76a23b6a35ebbee966c00a577",nonce="3755ae79fd668c2035ebb90cdc12d030",opaque="",algorithm=MD5
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 297
v=0
o=CiscoSystemsSIP-GW-UserAgent 3022 9963 IN IP4 66.55.220.166
s=SIP Call
c=IN IP4 66.55.220.166
t=0 0
m=audio 19258 RTP/AVP 18 101 19
c=IN IP4 66.55.220.166
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20
*Dec 27 14:10:34.990: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Incorrect Authentication
v: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK579619F9
f: "Server Room" [email protected]>;tag=F1B9854-8A5
t: [email protected]>
i: [email protected]
CSeq: 102 INVITE
l: 0
*Dec 27 14:10:35.002: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5080 SIP/2.0
Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK579619F9
From: "Server Room" [email protected]>;tag=F1B9854-8A5
To: [email protected]>
Date: Fri, 27 Dec 2013 14:10:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Here is debug voip ccapi inout:
GMIT-VOICEROUTER01#debug voip ccapi inout
voip ccapi inout debugging is on
GMIT-VOICEROUTER01#
*Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=7018
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-lastrdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=0
cisco-rdnsi=0
cisco-redirectreason=0 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
*Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/cc_api_call_setup_ind_common:
Interface=0x4A4AE7B0, Call Info(
Calling Number=7018,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE,
Incoming Dial-peer=20009, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)
GMIT-VOICEROUT, Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
*Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/ccCheckClipClir:
In: Calling Number=7018(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/ccCheckClipClir:
Out: Calling Number=7018(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Dec 27 14:10:55.326: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:55.326: :cc_get_feature_vsa malloc success
*Dec 27 14:10:55.326: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:55.326: cc_get_feature_vsa count is 1
*Dec 27 14:10:55.326: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:55.326: :FEATURE_VSA attributes are: feature_name:0,feature_time:1282234808,feature_id:151
*Dec 27 14:10:55.330: //12898/8912F77B8243/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=7018(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown))
*Dec 27 14:10:55.330: //12898/8912F77B8243/CCAPI/cc_process_call_setup_ind:
Event=0x49A103B8
*Dec 27 14:10:55.330: //12898/8912F77B8243/CCAPI/ccCallSetContext:
Context=0x4C5A319C
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 12898 with tag 20009 to app "_ManagedAppProcess_Default"
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/ccCallSetupAck:
Call Id=12898
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/cc_api_set_transfer_info:
Transfer Number=, Transfer Reason=0x0
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=TRUE, Tone=Dial Tone,
Tone Direction=Network, Params=0x0, Call Id=12898
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/ccSetDigitTimeouts:
Initial Digit Timeout=-1000(ms), Inter Digit Timeout=-1000(ms)
*Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/ccSetDigitTimeouts:
Call Entry(Inter Digit Timeout=10000(ms), Initial Digit Timeout=10000(ms))
*Dec 27 14:10:55.338: //12898/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x3262, digit_event=0x1, enable=TRUE, consume=FALSE)
*Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=12898
*Dec 27 14:10:55.338: //12898/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x4A4AE7B0, callID=0x3262, disp=0, digit_event=0x1, enable=TRUE, consume=FALSE)
*Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0x0, Interface=0x4A4AE7B0, Call Id=12898
*Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=15000(ms), Inter Digit Timeout=10000(ms))
*Dec 27 14:10:56.650: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=9, DigitBeginFlags=0x0,
Rtp Timestamp=0x9D41D0, Rtp Expiration=0x0
*Dec 27 14:10:56.654: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=9, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:56.654: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:56.970: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=1, DigitBeginFlags=0x0,
Rtp Timestamp=0x9DBED0, Rtp Expiration=0x0
*Dec 27 14:10:56.974: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=1, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:56.974: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:57.290: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=2, DigitBeginFlags=0x0,
Rtp Timestamp=0x9E3BD0, Rtp Expiration=0x0
*Dec 27 14:10:57.294: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=2, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:57.294: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:57.610: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=0, DigitBeginFlags=0x0,
Rtp Timestamp=0x9EB8D0, Rtp Expiration=0x0
*Dec 27 14:10:57.614: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=0, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:57.614: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:57.890: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=7, DigitBeginFlags=0x0,
Rtp Timestamp=0x9F35D0, Rtp Expiration=0x0
*Dec 27 14:10:57.894: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=7, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:57.894: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:58.162: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=2, DigitBeginFlags=0x0,
Rtp Timestamp=0x9FB2D0, Rtp Expiration=0x0
*Dec 27 14:10:58.162: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=2, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:58.162: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:58.314: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=2, DigitBeginFlags=0x0,
Rtp Timestamp=0xA02FD0, Rtp Expiration=0x0
*Dec 27 14:10:58.314: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=2, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:58.314: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:58.582: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=7, DigitBeginFlags=0x0,
Rtp Timestamp=0xA0ACD0, Rtp Expiration=0x0
*Dec 27 14:10:58.582: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=7, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:58.582: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:58.754: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=7, DigitBeginFlags=0x0,
Rtp Timestamp=0xA129D0, Rtp Expiration=0x0
*Dec 27 14:10:58.754: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=7, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:58.754: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:59.022: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=6, DigitBeginFlags=0x0,
Rtp Timestamp=0xA1A6D0, Rtp Expiration=0x0
*Dec 27 14:10:59.022: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=6, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:59.022: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:59.026: //12898/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x3262, digit_event=0x0, enable=FALSE, consume=FALSE)
*Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=12898
*Dec 27 14:10:59.026: //12898/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x4A4AE7B0, callID=0x3262, disp=0, digit_event=0x0, enable=FALSE, consume=FALSE)
*Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0x0, Interface=0x4A4AE7B0, Call Id=12898
*Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=15000(ms), Inter Digit Timeout=10000(ms))
*Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=702, Params=0x4C5A0BDC, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCheckClipClir:
In: Calling Number=20722672628(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCheckClipClir:
Out: Calling Number=20722672628(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCallSetupRequest:
Destination Pattern=91[2-9]......., Called Number=120722776, Digit Strip=FALSE
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCallSetupRequest:
Calling Number=20722672628(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=120722776(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Server Room
Account Number=, Final Destination Flag=FALSE,
Guid=8912F77B-6E37-11E3-8243-90AE21CDDC5B, Outgoing Dial-peer=702
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=20722672628
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=120722776
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-lastrdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=0
cisco-rdnsi=0
cisco-redirectreason=0 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
*Dec 27 14:10:59.034: //12898/8912F77B8243/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x48C27BD0, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=20722672628,(Calling Name=Server Room)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=120722776(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE, Outgoing Dial-peer=702, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
*Dec 27 14:10:59.034: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:59.034: :cc_get_feature_vsa malloc success
*Dec 27 14:10:59.034: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:59.034: cc_get_feature_vsa count is 2
*Dec 27 14:10:59.034: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:59.034: :FEATURE_VSA attributes are: feature_name:0,feature_time:1282234584,feature_id:152
*Dec 27 14:10:59.034: //12899/8912F77B8243/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
*Dec 27 14:10:59.034: //12899/8912F77B8243/CCAPI/ccCallSetContext:
Context=0x4C5A0B8C
*Dec 27 14:10:59.034: //12898/8912F77B8243/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=702
*Dec 27 14:10:59.038: //12899/8912F77B8243/CCAPI/cc_api_call_proceeding:
Interface=0x48C27BD0, Progress Indication=NULL(0)
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/cc_api_call_disconnected:
Cause Value=57, Interface=0x48C27BD0, Call Id=12899
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=57, Retry Count=0)
*Dec 27 14:10:59.270: //12898/xxxxxxxxxxxx/CCAPI/ccCallReleaseResources:
release reserved xcoding resource.
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/ccCallSetAAA_Accounting:
Accounting=0, Call Id=12899
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/ccCallDisconnect:
Cause Value=57, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=57)
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/ccCallDisconnect:
Cause Value=57, Call Entry(Responsed=TRUE, Cause Value=57)
*Dec 27 14:10:59.274: //12899/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x48C27BD0, Tag=0x0, Call Id=12899,
Call Entry(Disconnect Cause=57, Voice Class Cause Code=0, Retry Count=0)
*Dec 27 14:10:59.274: //12899/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
*Dec 27 14:10:59.274: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Dec 27 14:10:59.274: :cc_free_feature_vsa freeing 4C6D58D0
*Dec 27 14:10:59.274: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Dec 27 14:10:59.274: vsacount in free is 1
*Dec 27 14:10:59.278: //12898/8912F77B8243/CCAPI/ccCallDisconnect:
Cause Value=57, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Dec 27 14:10:59.278: //12898/8912F77B8243/CCAPI/ccCallDisconnect:
Cause Value=57, Call Entry(Responsed=TRUE, Cause Value=57)
*Dec 27 14:10:59.278: //12898/8912F77B8243/CCAPI/cc_api_get_transfer_info:
Transfer Number Is Null
*Dec 27 14:11:02.250: //12898/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x4A4AE7B0, Tag=0x0, Call Id=12898,
Call Entry(Disconnect Cause=57, Voice Class Cause Code=0, Retry Count=0)
*Dec 27 14:11:02.250: //12898/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
*Dec 27 14:11:02.250: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Dec 27 14:11:02.250: :cc_free_feature_vsa freeing 4C6D59B0
*Dec 27 14:11:02.250: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Dec 27 14:11:02.250: vsacount in free is 0ER01# -
auto dailer (sip dialer) is dialing the customer number starts with 5XXXXXXX but when customer number is 6XXXXXX or 7XXXXXX is not dialing. Seems to be sip dialer configuration in Voice gateway is having issue. WE have check the PRI cabability to dial all the numbers it is ok. could any one share the sample working configuration of VG, it will hep me to fix this issue.
if its using the same VG for dialing out, please check dial-peer configuration. may be for 5XXXXXXX it would be in place and not for other patterns.
Chintan -
Freetalk-1200 Connect.Me speed dial problem
Just received my long awaited Connect.Me and it's been working just as I expected. The only problem I encountered was that I couldn't assing speed dial numbers to 2 of my contacts. Others work perfectly, when I click them, box of information appears to the middle of the screen, but when I click one of the two with the problem, nothing happens.
I have tried everything like resetting from the button and from the web UI, deleting and adding them again to my contacts but nothing helps. Adding new contacts works fine. I would not rather create a new Skype account for my self - nor ask these two friends to make a new account for themselves - just because of this problem. Any kind of advice is welcome
Cheers from Finland!Skype hasn't supported speed dials in it's software for some time. It's considered a legacy feature. The use of speed dials is configured through the Freetalk Connect Me web interface, not the Skype client. You need to sign into your Connect Me device via your web browser and click on "Contacts and Speed Dials". Once there all your contacts will be listed on the left pane. Select any one of those contacts and click "ADD TO SPEED DIAL". It will assign a speed dial number incrementally on the right pane. You can change that number to a number of your choosing (0-99) at any time. That's all it takes to set up. If you recently add a contact it my not show on that list. If it doesn't sign off the client in which you added the contact or contact changes, then power cycle the Connect Me box and log back in. It should update. If it does not, the only other alternative is to do a factory reset on the unit so that it will purge the list and updated it with the latest copy. Below I attached an image of the screen, though it may take some time until it gets approved and displayed to the general public.
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Hi, how do we configure an imac intel to use a transparent US Http proxy that does not spill our IP address. Can someone help me to do this? Using Safari 3.
ThanksSo, which do I want? FTP proxy, Web proxy, secure web, etc.
How can I tell?
You're the one that knows what protocols you want to proxy.
If you want to proxy HTTP, then enable the HTTP proxy and enter your HTTP proxy server address.
If you want to proxy FTP then enable the FTP proxy and enter your FTP proxy server address...
I can't tell what one(s) you want to do. I had assumed you knew.
Also, do you know a good site to find these free transparent proxies?
Nope, sorry.
Finally, will they slow down my search speeds.
That depends on your choice of proxy.
Any connection you make will be passed to the proxy. It will forward the request for you, retrieve the response from the server and pass it back to you. Therefore you are entirely dependent on the speed of the proxy server itself, and its network connection.
There are two common uses for proxy servers, one is to (attempt to) anonymize your traffic, the other is to improve performance. The latter usually occurs when you have multiple people sharing a network connection (e.g. in an office) and the proxy can cache frequently used data.
In general, unless the proxy server you choose is directly between you and the site you're accessing, you're not likely to get any performance increase.
In addition, since anonymous proxies are sometimes used for... umm... less than honorable reasons, they are sometimes blocked by servers.
In other words, don't expect your web browsing experience to be any better for using an anonymous proxy.
I'm just trying to stream shows from the ABC site.
This will only help if they stream content over a protocol that your proxy supports.
For example, if they stream content over HTTP, you're probably OK, but if they use something like QuickTime's RSTP, RealPlayer, or (heaven forbid) WMV then setting a web proxy will have zero impact - you're not going to use it anyway.
Maybe you are looking for
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How to Use F4 help in Module Pool Programming??????
Hi Friends, This is Jagadeesh, I have an issue Module Pool Programming. Any of you can go through on this and can give an required answer. Issue is as follows, I have an Input/Output field for that i need to give f4 help, based on that field the rela
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I just had a new hard drive installed on my iMac
i just had a new hard drive installed on my iMac (due to the Seagate replacement program) i was able to migrate my back up(i think) but its now acting like a new computer and i can't get past the registration screen. I filled everything out but the "
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GETURL not working in exported movie
I hate to beat a dead horse but I'm at my wit's end. I've tried everything I have read online and it still doesn't work... I have created a keyframe at the end of my flash with... getURL("www.blahblahblah.ca/underconstruction.html"); I export the fla
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Hi Folks I want to measure customer churn by cohort. In the attached example I have two cohorts- 1) customers that were acquired (activated) on the the 23/11/2013 and 2) customers that were acquired on the 23/12/2014. Rather than measuring churn by d
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hi all i 've installed "SAP NETWEAVER 2004S SR2 Developer Workplace Number 51032258" who can tell me the version of this product i will Request license key