CUCM 4.2 integ with SIP Provider

I have terminated a SIP trunk on the voice gateway and configured it as CUBE. What configuration is required at Call manger 4.2 ?
Regards

I've created a route pattern in CUCM 4.2 with wildcard 7!. This is pointing to a route list with H323 gateway.
On the gateway i've created a dial peer :
dial-peer voice 40 voip
description **Outgoing Call to SIP Trunk
destination-pattern 7.T
session protocol sipv2
session target ipv4: 1.1.1.1
dtmf-relay rtp-nte sip-notify h245-alphanumeric
codec g711ulaw
no vad
----------- Is it enough to match the sip voip dial peer on gateway.

Similar Messages

RT31P2 with SIP provider

I used this device for Nikotel and Vyke SIP accounts but it never got registered as ISP is blocking ports.
I then configured one PC with VPN to UK and shared that connection. Gave PC LAN card as gateway in RT31P2. It worked. SIP calls went fine.
My question is - how to configure VPN connection in RT31P2 to bypass ISP port blocking or how to give alternate port settings for SIP provider.

What is STUN server settings. Can this be used in anyway to bypass the ISP proxy and connect to the SIP server.
If ports are blocked is there any other way to use the device for any SIP provider. I had used the device once to make calls using Nikotel (SIP) network but the voice quality was not good. Then I found a version upgrade on the LinkSys site and upgraded the firmware. After this I was not able to get registered to Nikotel. I asked support@ Linksys for the old firmware but they could not send me as they never had in their archive.
I am not sure if the new firmware created a problem or the ISP changed anything.
I would appreciate if anyone could send me the old firmware 1.02 so that I could try it.

Cisco 877 router - Cisco IP phone won't register with SIP provider

Hi all,
I'm having a problem with a Cisco SPA504G phone not registering with the SIP carrier over the Internet. We've recently rolled out a Cisco 877 router onto a new NBN business connection and can't get the pre-configured IP phone to register.
When we tested the phone with the NBN-provided Netgear router, it worked fine, as it did with the previous Cisco 1841 router we were using on a different link.
The way it's setup is using VLANs to define the internal subnets, which are then assigned to the physical interfaces (since the 887 doesn't allow IP assignments to the interfaces directly).
VLAN 100 is the internal network and has a SBS2011 server – assigned to F0 – IP range is 192.168.1.0
VLAN 200 is the guest network and has Internet access only – assigned to F1 – IP range is 10.1.1.0
VLAN 500 is the WAN network and connects to the NBN upstream box – assigned to F3 – external IP address assigned by DHCP
I've been playing around with access lists, nat rules, basically everything in my limited Cisco knowledge to try and figure this out, but to no avail. I have even configured what I believe is unrestricted access to IP, UDP and TCP outbound and inbound to all VLANs and still can't get it to register.
Tried isolating the issue by creating a new VLAN and assigning it to the spare interface and basically allowing everything in and out, but still no luck.
The problem has to be something on the router – probably some small line of config I haven’t removed or added.
I am going to pull my hair out soon, so would really appreciate some assistance from the Cisco gurus out there.
My client has just purchased about 10 of these handsets from their provider so I need to fix this ASAP. The guy who provided them wasn't very helpful, and basically said I'm on my own once we tested using the NBN-provided Netgear router.
Happy to post my config as well.
Please help!!!!

Current configuration : 4912 bytes
version 15.1
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router1
boot-start-marker
boot-end-marker
no aaa new-model
memory-size iomem 10
crypto pki token default removal timeout 0
no ip source-route
ip dhcp excluded-address 10.1.1.1
ip dhcp pool GUEST
network 10.1.1.0 255.255.255.0
dns-server 10.1.1.1 203.50.2.71 139.130.4.4
default-router 10.1.1.1
ip cef
no ip domain lookup
ip domain name network.local
ip name-server 192.168.1.123
ip name-server 203.23.53.12
ip name-server 197.12.32.86
ip name-server 8.8.8.8
no ipv6 cef
license udi pid CISCO887VA-K9 sn FGL171220XY
username admin privilege 15 secret 5 $1$aNsm$N1BCQYkoi8gnURyvloYEX/
controller VDSL 0
interface Ethernet0
no ip address
shutdown
interface ATM0
no ip address
no atm ilmi-keepalive
bridge-group 10
pvc 8/35
interface FastEthernet0
description NAC - Internal network
switchport access vlan 100
no ip address
interface FastEthernet1
description NAC - Guest network
switchport access vlan 200
no ip address
interface FastEthernet2
no ip address
shutdown
interface FastEthernet3
description **** WAN Port ****
switchport access vlan 500
no ip address
interface Vlan1
no ip address
bridge-group 10
hold-queue 100 out
interface Vlan100
description NAC - Internal Vlan
ip address 192.168.1.1 255.255.255.0
ip access-group IN-100 in
ip access-group OUT-100 out
ip nat inside
ip virtual-reassembly in
interface Vlan200
description NAC - Guest Vlan
ip address 10.1.1.1 255.255.255.0
ip access-group IN-200 in
ip access-group OUT-200 out
ip nat inside
ip virtual-reassembly in
interface Vlan500
description **** WAN Vlan ****
ip address dhcp
ip nat outside
no ip virtual-reassembly in
no ip forward-protocol nd
ip http server
ip http access-class 23
ip http secure-server
ip dns server
ip nat inside source list NAT-100 interface Vlan500 overload
ip nat inside source list NAT-200 interface Vlan500 overload
ip nat inside source static tcp 192.168.1.123 25 interface Vlan500 25
ip nat inside source static tcp 192.168.1.123 443 interface Vlan500 443
ip nat inside source static tcp 192.168.1.123 3389 interface Vlan500 3399
ip nat inside source static tcp 192.168.1.123 80 interface Vlan500 80
ip nat inside source static tcp 192.168.1.123 4125 interface Vlan500 4125
ip nat inside source static tcp 192.168.1.124 3389 interface Vlan500 3390
ip nat inside source static tcp 192.168.1.123 987 interface Vlan500 987
ip nat inside source static tcp 192.168.1.123 1723 interface Vlan500 1723
ip route 0.0.0.0 0.0.0.0 55.234.52.43
ip access-list extended IN-100
permit udp any any range bootps bootpc
deny ip 10.1.1.0 0.0.0.255 any
permit ip 192.168.1.0 0.0.0.255 any
ip access-list extended IN-200
permit udp any any range bootps bootpc
permit ip 10.1.1.0 0.0.0.255 any
ip access-list extended NAT-100
deny ip 192.168.0.0 0.0.255.255 192.168.0.0 0.0.255.255
permit ip 192.168.1.0 0.0.0.255 any
ip access-list extended NAT-200
deny ip 10.1.0.0 0.0.255.255 10.1.0.0 0.0.255.255
permit ip 10.1.1.0 0.0.0.255 any
ip access-list extended OUT-100
permit udp any range bootps bootpc any
deny ip 10.1.1.0 0.0.0.255 any
permit ip any 192.168.1.0 0.0.0.255
ip access-list extended OUT-200
permit udp any range bootps bootpc any
deny ip 10.1.1.0 0.0.0.255 192.168.1.0 0.0.0.255
permit ip any 10.1.1.0 0.0.0.255
access-list 23 permit 59.23.164.52
access-list 23 permit 192.168.1.0 0.0.0.255
access-list 23 permit 10.1.1.0 0.0.0.255
access-list 23 permit 120.146.0.0 0.0.255.255
access-list 23 permit 149.185.12.0 0.0.0.255
access-list 23 permit 110.44.28.0 0.0.0.255
access-list 23 permit 110.44.26.0 0.0.0.255
access-list 23 permit 103.25.212.0 0.0.0.255
access-list 23 permit any
bridge 10 protocol ieee
banner motd ^C
* Authorized personnel only! *
^C
line con 0
login local
no modem enable
line aux 0
line vty 0 4
password password01
login local
transport input all
end

Forwarding with SIP Trunking and Retaining CID

We are using SIP trunking to get to the PSTN and CUCM 7.1. The SIP provider only permits calls originating from their own DIDs.
So how can we allow our users to forward all calls to say their cell phone. AND when a call comes in we want them to be able to see the original caller ID. Is it possible? What is the mojo? Thanks!

Hi,
for the SIP trunks there are no limits from the system. Check this out: https://supportforums.cisco.com/message/3795863#3795863
If you have different voice codecs for your phones and the trunk you need DSP ressources for transcoding.
As I know the UC520 have a PVDM2-64 with 4 DSP chips. You can use the DSP calculator from cisco to find out how many DSPs you need. But keep in mind that conferencing and transcoding can't share a DSP processor.
For example 1 DSP for conferencing and 3 for transcoding.
best regards
Christian

Cisco 7942 + SIP Provider

Hello!
Can the Cisco 7942 with SIP Firmware used as standalone SIP device?
I mean can it works with SIP provider through NAT, like it can Cisco SPA-303?

There has been a discussion on this before.
https://supportforums.cisco.com/discussion/11955621/register-cisco-phone-7942-external-voip-provider
However, there was no conclusion to it.
This discussion here talked about registering 7942 with Asterisk.
http://www.experts-exchange.com/Networking/Telecommunications/IP_Telephony/Q_26895490.html
Since Asterisk is a 3rd party PBX, this shows that the phone CAN register with SIP firmware with a Provider. However, you will have to work extensively with the provider to get this done.
For instance, you need to create a custom cnf.xml file for the phone to download. To do this you'll need to copy the configuration from the CUCM, and then modify it as per your needs. Apart from this, the firmware files should also be located on the TFTP server that you're pointing to on the phone.
Also, you need to make sure that the provider doesn't have any mechanism on their side to block messages going out from the phone to their end. Packet captures would help you here.
There isn't a guarantee that this would work, but you can definitely try it.
Thanks

Changing external Caller ID over a SIP Trunk to SIP Provider

I am working with a client and when they place calls out to any external user they have the wrong name showing on the external caller ID.
I have spoken with the SIP provider and apparently they want us to pass the CNAM, or rather they have it setup for us to do this.
I opened a case with Cisco and the TAC engineer said the provider has to do this because it cannot be done from CUCM or the gateway.
For example, it says right now "location A" for external calls and I want to change this to say "location B" .
Is this even possible?

what is the call flow? did you check the caller name in SIP trunk configuration?

Connect Cisco CallManager to external SIP provider

I need to connect my CUCM 5.1 with sip proxy on telco side.IP phones
will connect to CUCM.
The SIP server provide 90 lines with real numbers
Following is the scenario.
Cisco IP phones----------CUCM-------WAN connection to
telco---------SIP proxy server.
Can anybody explain me how this will work, what will be the
configurations and if CUCM has the capability to control the calls
between IP phones and SIP server.

Hi Asim
It is recommended to use CUBE (IP-IP gateway)
Look following url for configuration.
http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a00808ead0f.shtml
Regards..
Mahesh Dawar
www.cisco.com/go/pdihelpdesk

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.

Multiple registration with sip-ua

Hi,
someone know a way to do multiple registration with a single 2811 using sip-ua configuration with multiple accounts??
thnx
s.

Hi, I have got to authenticate more than one account in the SIP provider with the hidden command "credentials" the problem that I have now is how to route all the calls done to the second account to the extension 101.
I want that incoming calls from 964812530 goes to extension 100 and incoming calls from 965072519 goes to extension 101
How can I do it?
I have tried this but it's not running:
sip-ua
credentials username 965072510 password 115849534F43415C557B7967 realm beta.awa
voz.com
authentication username 964812539 password 13544744535D4E7A7A757A70
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 5
timers register 250
registrar ipv4:213.162.201.146 expires 60
sip-server ipv4:213.162.201.146
voice service voip
sip
voice translation-rule 1
rule 1 /1../ /964812539/
voice translation-rule 3
rule 1 /964812539/ /100/
voice translation-rule 4
rule 1 /965072510/ /101/
voice translation-profile SIPout
translate calling 1
voice translation-profile incoming
translate called 3
voice translation-profile incoming2
translate called 4
voip-incoming translation-profile incoming
All the incoming calls are going to extension 100
regards

Phonepower as a SIP provider for UC540

Hello,
has anyone had any experience stting up a SIP trunk on the UC540 with phonepower as the provider? I am having some issues with basic setup. I can't seem to get the UC540 to register with the provider. thanks.
Wayne

Hi Wayne,
Have they given you configuration information to assist you with the setup process?
If they have can you post it without all the sensitive information just to be sure it is everything you need to get it working.
Also as per our private inboxing, can you advise if your UC is WAN facing or if you have a router in front of it?
Cheers,
David Trad.

Cisco Phone 7960 and SIP provider

Hi,
i have an account with a Sip provider.
I have all information for make a connection with xlite sip client but if i try to configure a Cisco Phone with SIP Firmware (7.5), phone not work.
My provider is messagenet.it.
Can you help me?
Thanks

Hello,
have a look at the configuration guide "Getting Started with Your Cisco SIP IP Phone" at
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_guide_chapter09186a0080080edf.html
This should pretty much answer your questions and allow you to succeed with your task.
Hope this helps! Please rate all posts.
Regards, Martin

Controlling which CUCM server communicate over a SIP trunk

We have 3 CUCM servers, two sub and one pub at two different physical locations.
There are two SIP trunk servers (non Cisco), we wish to have the CUCM at the same physical location to communicate with the SIP trunk device at its location, instead of going over the WAN to communicate with the other one.
Communications are initiated from the CUCM side through a route pattern that points to a RL/RG that contains the SIP trunk.
The SIP trunk uses a DP that uses a CUCM group that only contains the local CUCM server.
Can this be setup to reliably control which CUCM server talks to the SIP trunk server at its location?
Do we need to configure something differently?

Hi,
If the CUCM's are located in different physical location then the communication should be over the IP-WAN and if it is about different CUCM Cluster then we can use H323 ICT Trunk between CUCM server for communication which is again WANLink.
Still not clear with your query, However as you want to control the Gwy/Trunks using the Route Pattern. You can create two separate route pattern for internal & external.
Wherein in Internal Route Pattern you can specify the ICT/SipTrunk/GWY which is connected within the cluster under Route List/RouteGroup to route the call.
And for External Route Pattern specify the Gwy/SipTrunk which connects you to the outside world.
Regards,
Venkatesh

Lync 2013 with SIP trunk with panasonic kx-tde200

Hi
My company has installed a panasonic ip-pbx kx-tde for multiline with 100 number range for telephone service.
Now my company is going to replace multiline by sip trunk . It will still work with Panasonic pbx box just need to reprogramme to be able to connected to the sip proxy which is managed by internet service provider.
For this scenario , would Lync 2013 voice work if I just add PSTN gateway which is the ip of panasonic pbx address to the frontend in topology ? Or I may need a mediation server as a must requirement to make lync voice work?
Thanks
WenFei

Media bypass allow a call to basically skip the mediation server once it's established and go directly from gateway (in this case the PBX) and the endpoint (the telephone handset or Lync client) More information here: http://technet.microsoft.com/en-us/library/gg398719.aspx
By having this (if your PBX supports it) you reduce the load on the mediation servers. Before you go too far down the road also make sure that your PBX supports SIP trunks that are SIP over TCP (as Lync doesn't work with SIP over UDP)
Sort of, the easiest way is to add the .com as an additional SIP domain in Topology builder, you will need to create DNS records for it (both internal and external) and you will need to reissue the certs with additional SANs to support the second domain.
YOu will also need to update all the users to use the new suffix of xxx.com. So it's not a small task.
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@imlynced

Prefixing a 9 and 91 to incoming calls from SIP provider for callback

I am wondering what would be the best options for prefixing a 9 or 91 to incoming calls over a sip connection to allow callback from missed calls and recieved calls. The setup is
callmanager 7.1.5 >>>>sip trunk>>>>>>>>>>>>CUBE>>>>>>>>>>>sip to ITSP
I am thinking voice translation rules is the only option for this? any configuration examples for this would be greatly appreciated.
would this work?
voice-translation rule 1
rule 1 // /9/
voice-translation profile prefix_9
translate calling 1
dial-peer voice 101 voip
destination-pattern ???????...$
voice-class codec 1
session protocol sipv2
session target ipv4: to callmanager
incoming called-number .
dtmf-relay rtp-nte
dial-peer voice 1001 voip
translation profile incoming prefix_9
destination-pattern T
session protocol sipv2
session target ipv4: to sip provider
incoming called-number ???????...$
dtmf-relay rtp-nte

Your config should work fine, except your profile is only applied to one dial-peer, make sure you apply it to the one that is used to redirect the call to CUCM.
Also, you did not mention what country you are in, but if this is US you may want to prefix 91 to national calls as carriers don't provide 9 as part of the CLID delivery, also what about your international calls, you may would be more explicit in your first rule to match for national digit string and then have another rule for international.
HTH,
Chris

Cisco CME and Calls through SIP provider

Hello, friends.
There are Cisco (C2801-ADVENTERPRISEK9_IVS-M), Version 15.1 (4) M7.
Telephones connected to SCCP, registered SIP from the provider.
When I try to call to test number 4444 through sip in debug I see:
*Feb 10 01:51:25.317: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP XXXXXXXXXXX:5060;branch=z9hG4bK100D02077;rport=5060
From: "TEST" <sip:[email protected]>;tag=131CC60C-1D40
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvf1OFL39Awnou/oMiaFQrf9jyybhFmf", qop="auth"
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 01:51:25.325: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP XXXXXXXXXX:5060;branch=z9hG4bK100D02077
From: "TEST" <sip:[email protected]>;tag=131CC60C-1D40
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7
Date: Sun, 09 Feb 2014 21:51:25 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Cisco при этом зарегана у провайдера SIP
DC#show sip-ua register status
Line peer expires(sec) registered P-Associ-URI
Configuration:
voice service voip
ip address trusted list
ipv4 178.16.26.122 255.255.255.255
ipv4 144.76.42.108 255.255.255.255
ipv4 176.9.145.115 255.255.255.255
ipv4 5.9.108.25 255.255.255.255
ipv4 78.46.95.118 255.255.255.255
ipv4 89.249.23.194 255.255.255.255
ipv4 178.16.26.124 255.255.255.255
ipv4 176.9.85.133 255.255.255.255
ipv4 46.4.53.86 255.255.255.255
ipv4 5.9.84.165 255.255.255.255
ipv4 78.16.26.122 255.255.255.255
ipv4 77.235.62.222 255.255.255.255
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g711alaw
voice register global
max-dn 10
max-pool 10
voice register dn 1
number 150
voice register dn 2
number 151
voice translation-rule 9
rule 1 /^95/ //
voice translation-rule 1020
rule 1 /^.$/ /40232/
voice translation-profile outgoing
translate calling 1020
translate called 9
mgcp fax t38 ecm
mgcp profile default
dial-peer voice 2 voip
translation-profile outgoing outgoing
destination-pattern 95....
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte
no vad
sip-ua
credentials username 40232 password 7 XXXXXXXXXX realm sip.zadarma.com
authentication username 40232 password 7 XXXXXXXXXXXX realm sip.zadarma.com
registrar dns:sip.zadarma.com:5060 expires 3600
sip-server dns:sip.zadarma.com:5060
connection-reuse
host-registrar
DC#show sip-ua register status
Line                             peer       expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
150                              40001      12           no
40232                            -1         550          yes
SIP provider says cisco trying to call with the internal call number, and it is necessary in order that have an SIP provider:
Wrong Remote-Party-ID: "Vankuver" <sip:61@<my ip>>;party=calling;
Should be so sip:40232@<my ip>
Please help me!

Yes, I behind nat.
*Feb 10 18:11:53.425: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: "954444"
From: "150";tag=7b409f06
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 314
v=0
o=- 2 2 IN IP4 192.168.11.14
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.11.14
t=0 0
m=audio 5724 RTP/AVP 107 0 8 101
a=alt:1 2 : gNONJ/Dj BaLJhmb/ 10.200.16.55 5724
a=alt:2 1 : DQ3e8qud c1qVrWui 192.168.11.14 5724
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
*Feb 10 18:11:53.477: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038E7FF
From: "" >;tag=169E6BC4-1E16
To: [email protected]>
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0541864002-2442400227-2618163141-2285537806
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1392041513
Contact: outside ip cisco cme:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18534 RTP/AVP 0 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
*Feb 10 18:11:53.481: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
From: "150";tag=7b409f06
To: "954444"
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Feb 10 18:11:53.625: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP outside ip cisco cme:5060;branch=z9hG4bK1038E7FF;rport=5060
From: "" ;tag=169E6BC4-1E16
To: [email protected]>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn", qop="auth"
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 18:11:53.633: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038E7FF
From: "150" [email protected]>;tag=169E6BC4-1E16
To: [email protected]>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
*Feb 10 18:11:53.637: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038F25FC
From: "" ;tag=169E6BC4-1E16
To: [email protected]>
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0541864002-2442400227-2618163141-2285537806
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1392041513
Contact: :5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18534 RTP/AVP 0 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
*Feb 10 18:11:53.981: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038F25FC;rport=5060
From: "" ;tag=169E6BC4-1E16
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 18:11:54.385: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.63.X:5060;rport=5060;branch=z9hG4bK1038F25FC
Record-Route:
From: "k40232" ;tag=169E6BC4-1E16
To: [email protected]>;tag=as7e8de8e5
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Zadarma Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 1942395501 1942395501 IN IP4 178.16.26.124
s=Asterisk PBX
c=IN IP4 178.16.26.124
t=0 0
m=audio 12164 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
*Feb 10 18:11:54.409: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.xxxx.xxxx:5060;branch=z9hG4bK10390E63
From: "150" [email protected]>;tag=169E6BC4-1E16
To: [email protected]>;tag=as7e8de8e5
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 102 ACK
Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001
Allow-Events: telephone-event
Content-Length: 0
*Feb 10 18:11:54.429: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
From: "150";tag=7b409f06
To: "954444";tag=169E6F78-88E
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: :5060;transport=tcp>
Supported: replaces
Server: Cisco-SIPGateway/IOS-12.x
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 193
v=0
o=CiscoSystemsSIP-GW-UserAgent 149 3396 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 17190 RTP/AVP 8
c=IN IP4 92.63.108.115
a=rtpmap:8 PCMA/8000
a=ptime:20
*Feb 10 18:11:54.653: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 91.231.141.230:42294;branch=z9hG4bK-d8754z-95374017c126c928-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: "954444";tag=169E6F78-88E
From: "150";tag=7b409f06
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 ACK
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

CUCM 4.2 integ with SIP Provider

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