SIP outgoing call getting disconnected as it is answered

[Vijay Anand]

Hi All, 

I currently working on the SIP configuration with ITSP, incoming calls are working fine but outgoing is getting dropped when it is answered.

phone(ext7900)-------->sip trunk to cube------>cube------------------> ITSP-------------->callednumber

I have taken the logs, when call 3499579646 from ext 7900, call is getting dropped. Attached log and cube config and incoming call working log attached.

[Abhay Singh Reyal]

Hello Vijay 

Can you try using early offer for the SIP trunk pointing towards the cube and see if that makes any difference ?

Regards

Abhay Reyal

[Vijay Anand]

Yes, i tried that. but result is same. :(

[Nadeem Ahmed]

you have already EO enforced here however issue seems to be with codec mismatch. incoming calls working on G729 ie 18 and outgoing was trying to nego over G711 . You need to add voice-class codec on the outgoing dial-peer which is missing out here. correct the config it should work. also here we are matching dial-peer voice 11 which has no voice class codec.

Incoming call SDP in INVITE

v=0
o=Sippy 3394102149902295958 0 IN IP4 80.78.66.70
s=-
t=0 0
m=audio 43102 RTP/AVP 8 0 18 4 101
c=IN IP4 80.78.66.70
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

200 OK

v=0
o=CiscoSystemsSIP-GW-UserAgent 7151 4756 IN IP4 31.171.153.246
s=SIP Call
c=IN IP4 31.171.153.246
t=0 0
m=audio 17274 RTP/AVP 18 101
c=IN IP4 31.171.153.246
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

In Outgoing

Sent:
INVITE sip:00393499579646@80.78.66.70:5060 SIP/2.0
Via: SIP/2.0/UDP 31.171.153.246:5060;branch=z9hG4bK58149A
From: <sip:35544547900@80.78.66.70>;tag=4F7932B4-770
To: <sip:00393499579646@80.78.66.70>
Date: Tue, 29 Mar 2016 14:47:45 GMT
Call-ID: 891386C-F4F411E5-991EF795-1A3DC70E@31.171.153.246
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0237327360-0000065536-0000831183-0536111370
User-Agent: Cisco-SIPGateway/IOS-15.4.3.M1
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1459262865
Contact: <sip:35544547900@31.171.153.246:5060>
Call-Info: <sip:31.171.153.246:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
P-Preferred-Identity: <sip:35544547900@31.171.153.246>
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 324

v=0
o=CiscoSystemsSIP-GW-UserAgent 9990 7886 IN IP4 31.171.153.246
s=SIP Call
c=IN IP4 31.171.153.246
t=0 0
m=audio 17748 RTP/AVP 18 8 0 101
c=IN IP4 31.171.153.246
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 31.171.153.246:5060;branch=z9hG4bK58149A;rport=65277
Record-Route: <sip:80.78.66.70:5060;transport=udp;lr>
Contact: Anonymous <sip:80.78.66.70:5071>
To: <sip:00393499579646@80.78.66.70>;tag=b6yo5jg3zoxbf3lo.i
From: <sip:35544547900@80.78.66.70>;tag=4F7932B4-770
Call-ID: 891386C-F4F411E5-991EF795-1A3DC70E@31.171.153.246
CSeq: 101 INVITE
Content-Type: application/sdp
Server: Sippy
Portasip-3264-action: answer 1
Content-Length: 191

v=0
o=Sippy 609468768746379133 1 IN IP4 80.78.66.70
s=-
t=0 0
m=audio 58596 RTP/AVP 8 101
c=IN IP4 80.78.66.70
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

add the voice class codec here or create new for outgoing call

dial-peer voice 11 voip
 description *** pstn outgoing calls via SIP Trunk***
 translation-profile outgoing OUTGOING-SIP
 max-conn 30
 destination-pattern 0T
 progress_ind setup enable 3
 session protocol sipv2
 session target ipv4:80.78.66.70
 voice-class codec 1
 voice-class sip asserted-id ppi
 dtmf-relay sip-notify rtp-nte sip-kpml
 fax protocol pass-through g711alaw
 no vad

Br,

Nadeem

[Vijay Anand]

Hi 

I removed and reconfigured the dial-peer but result is same call drops as it is connected for outgoing.

dial-peer voice 11 voip
description *** pstn outgoing calls via SIP Trunk***
translation-profile outgoing OUTGOING-SIP
max-conn 30
destination-pattern 0T
progress_ind setup enable 3
session protocol sipv2
session target ipv4:80.78.66.70:5060
voice-class codec 1
voice-class sip asserted-id ppi
dtmf-relay sip-notify rtp-nte sip-kpml
fax protocol pass-through g711alaw
no vad
!

[Nadeem Ahmed]

Where is phone registered too? also did  you verified are we hitting this dial-peer 11 for outgoing call?

Can you please remove this command under the this dial-peer and make one test call along with debug

fax protocol pass-through g711alaw> remove this.

[Vijay Anand]

Hey Nadeem, 

I resolved the issue..... Thanks for your explanation above which helped to figure out the cause. I re-read the sip messages and found that call manager was sending the bye message as it was not able to negotiate the codec it was receiving the message 200ok. Everything was perfect. It was hardware mtp i configured on the cube was shutdown. 

giving no shut resolved for the issue. Once again thanks, without your explanation i would have missed looking into mtp configuration.

Regards

Vijay

[Nadeem Ahmed]

wonderful Anand. eventually you figured out by yourself.

Br,

Nadeem Ahmed