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CUCM - Asterisk - CCM express - VoIP provider

Phones are registered at CUCM. Asterisk is used for trunks. 

New branch is added with cisco 2811 (c2800nm-adventerprisek9-mz.151-4.M12a.bin)

I guess problem is in Asterisk - CCM express communication. I try to make outbound call to 994125555555, called number is then translated to 5555555. 

cisco 2811 configuration (internal interface 10.10.100.50, external 1.1.1.2)
Asterisk - 10.10.10.10
VoIP provider - 1.1.1.1

voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 h323
 sip

dial-peer voice 1 voip
 translation-profile outgoing TO-PSTN
 destination-pattern 994.T
 session protocol sipv2
 session target ipv4:1.1.1.1:4546
 session transport udp
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

!
dial-peer voice 102 voip
 description -= LOCAL GROUP =-
 answer-address 121234567
 destination-pattern 121234567
 session protocol sipv2
 session target ipv4:10.10.10.10:5060
 session transport udp
 incoming called-number 121234567
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

Asterisk (FreePBX) trunk configuration

Peer details
type=peer
qualify=yes
port=5060
nat=no
insecure=very
host=10.10.100.50
context=from-internal
canreinvite=yes
allow=alaw&ulaw

User details
type=user
qualify=yes
port=5060
nat=no
insecure=very
host=10.10.100.50
context=from-internal
canreinvite=yes
allow=alaw&ulaw

Logs are in attechments

Everyone's tags (1)
3 REPLIES 3
Rising star

Hi There,The Asterisk server

Hi There,

The Asterisk server is originally sending an INVITE message to the Cisco VGW with a lot of different codecs being offered (including and additional m line for video).

*Jan 27 10:54:32.380: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:994125555555@10.10.100.50:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK00e67cff
Max-Forwards: 70
From: <sip:121234567@10.10.10.10>;tag=as2dc40b5a
To: <sip:994125555555@10.10.100.50:5060>
Contact: <sip:121234567@10.10.10.10:5060>
Call-ID: 1b6627780b08c7df73fc9d66001ff970@10.10.10.10:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 27 Jan 2017 10:09:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "121234567" <sip:121234567@10.10.10.10>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 516

v=0
o=root 957168965 957168965 IN IP4 10.10.10.10
s=Asterisk PBX 11.17.1
c=IN IP4 10.10.10.10
b=CT:10240
t=0 0
m=audio 59792 RTP/AVP 0 9 3 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 49946 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=sendrecv




The Cisco gateway then sends a "500 Internal Server Error" message with the cause code of 127.

*Jan 27 10:54:32.384: //267264/D162593A861C/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK00e67cff
From: <sip:121234567@10.10.10.10>;tag=as2dc40b5a
To: <sip:994125555555@10.10.100.50:5060>;tag=332D0BD8-173D
Date: Fri, 27 Jan 2017 10:54:32 GMT
Call-ID: 1b6627780b08c7df73fc9d66001ff970@10.10.10.10:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=127
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

The cause code of 127 indicates that there is some sort of inter-working issue that the gateway can not solve.


Would it be possible for you to try and disabling all of those extra codecs which are being offered in the initial INVITE message from Asterisk to the Cisco VGW?

Other next steps would be:

  1. Make another test call with the following debugs enabled:
    • debug ccsip messages
    • debug voip ccapi inout



It was interface bind related

It was interface bind related problem.

Asterisk is in private network and VoIP provider in public. So solution was to use global bind interface to interface in private network and dial-peer based bind interface to public interface.

Rising star

Thanks for posting your

Thanks for posting your solution!

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