01-27-2017 02:51 AM - edited 03-17-2019 09:19 AM
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
01-31-2017 12:01 AM
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:
01-31-2017 01:07 AM
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.
01-31-2017 08:10 AM
Thanks for posting your solution!
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