WARNING[8914]: chan_sip.c:18487 handle_response_invite

Asterisk Asterisk Support
1

Hello,
I would like to know what this error means :

WARNING[8914]: chan_sip.c:18487 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog ‘789d4b6c04d1e745408577d843890907@192.168.0.27’. Giving up.
> Channel SIP/commpeak-00000027 was never answered

How could i solve this issue since most of the calls are dropped because of this reason.
and there are high call failure rate because of this.

Please help me out of this.

2

A full sip trace could help.

3 


<--- SIP read from UDP:50.112.137.243:5060 --->
SIP/2.0 100 Trying...
Via: SIP/2.0/UDP 192.168.0.27:8745;received=192.168.0.27;branch=z9hG4bK5a419f31;rport=8745
From: "XXXXXXXXXXXX" <sip:XXXXXXXXXXXX@192.168.0.27:8745>;tag=as27b881fd
To: <sip:XXXXXXXXXXXX@sip.commpeak.com>
Call-ID: 10132f412da84f456b00e212472ea7d2@192.168.0.27
CSeq: 103 INVITE
Server: CommPeak SIP Proxy
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:50.112.137.243:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.27:8745;received=192.168.0.27;branch=z9hG4bK0306caf0;rport=8745
Record-Route: <sip:50.112.137.243;lr;did=231.b7033a42>
From: "XXXXXXXXXXXX" <sip:XXXXXXXXXXXX@192.168.0.27:8745>;tag=as4c9e7fd1
To: <sip:XXXXXXXXXXXX@sip.commpeak.com>;tag=UN82F8B6Bpjye
Call-ID: 5c45f7cb6219531562b8cf78293c8314@192.168.0.27
CSeq: 103 INVITE
Contact: <sip:XXXXXXXXXXXX@54.245.38.213:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc3~n20121014T180359Z-1~precise+1+git~20121014T160541Z~762cf88183
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
Remote-Party-ID: "XXXXXXXXXXXX" <sip:XXXXXXXXXXXX@sip.commpeak.com>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1359059782 1359059783 IN IP4 54.245.38.213
s=FreeSWITCH
c=IN IP4 54.245.38.213
t=0 0
m=audio 21672 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

<------------->
--- (17 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 54.245.38.213:21672

<--- SIP read from UDP:50.112.137.243:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.27:8745;received=192.168.0.27;branch=z9hG4bK5a419f31;rport=8745
Record-Route: <sip:50.112.137.243;lr;did=42f.bc3b3d81>
From: "XXXXXXXXXXXX" <sip:XXXXXXXXXXXX@192.168.0.27:8745>;tag=as27b881fd
To: <sip:XXXXXXXXXXXX@sip.commpeak.com>;tag=9vNX310XH443m
Call-ID: 10132f412da84f456b00e212472ea7d2@192.168.0.27
CSeq: 103 INVITE
Contact: <sip:XXXXXXXXXXXX@50.112.54.155:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc3~n20121014T180359Z-1~precise+1+git~20121014T160541Z~762cf88183
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
Remote-Party-ID: "XXXXXXXXXXXX" <sip:XXXXXXXXXXXX@sip.commpeak.com>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1359050701 1359050702 IN IP4 50.112.54.155
s=FreeSWITCH
c=IN IP4 50.112.54.155
t=0 0
m=audio 30760 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

<------------->
--- (17 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 50.112.54.155:30760

<--- SIP read from UDP:50.112.137.243:5060 --->
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.0.27:8745;received=192.168.0.27;branch=z9hG4bK0306caf0;rport=8745
Max-Forwards: 69
From: "XXXXXXXXXXXX" <sip:XXXXXXXXXXXX@192.168.0.27:8745>;tag=as4c9e7fd1
To: <sip:XXXXXXXXXXXX@sip.commpeak.com>;tag=UN82F8B6Bpjye
Call-ID: 5c45f7cb6219531562b8cf78293c8314@192.168.0.27
CSeq: 103 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc3~n20121014T180359Z-1~precise+1+git~20121014T160541Z~762cf88183
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
Remote-Party-ID: "XXXXXXXXXXXX" <sip:XXXXXXXXXXXX@sip.commpeak.com>;party=calling;privacy=off;screen=no



<------------->
--- (14 headers 0 lines) ---
[Jan 25 08:12:44] WARNING[8914]: chan_sip.c:18487 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '5c45f7cb6219531562b8cf78293c8314@192.168.0.27'. Giving up.
Transmitting (no NAT) to 50.112.137.243:5060:
ACK sip:XXXXXXXXXXXX@sip.commpeak.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.27:8745;branch=z9hG4bK0306caf0;rport
Max-Forwards: 70
From: "XXXXXXXXXXXX" <sip:XXXXXXXXXXXX@192.168.0.27:8745>;tag=as4c9e7fd1
To: <sip:XXXXXXXXXXXX@sip.commpeak.com>;tag=UN82F8B6Bpjye
Contact: <sip:XXXXXXXXXXXX@192.168.0.27:8745>
Call-ID: 5c45f7cb6219531562b8cf78293c8314@192.168.0.27
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.24
Content-Length: 0


---
Scheduling destruction of SIP dialog '5c45f7cb6219531562b8cf78293c8314@192.168.0.27' in 32000 ms (Method: INVITE)
       > Channel SIP/commpeak-00000028 was never answered.
Scheduling destruction of SIP dialog '5c45f7cb6219531562b8cf78293c8314@192.168.0.27' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:50.112.137.243:5060 --->
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.0.27:8745;received=192.168.0.27;branch=z9hG4bK5a419f31;rport=8745
From: "XXXXXXXXXXXX" <sip:XXXXXXXXXXXX@192.168.0.27:8745>;tag=as27b881fd
To: <sip:XXXXXXXXXXXX@sip.commpeak.com>;tag=155c340f586c28d0300cf5a6ccf90d99-acbd
Call-ID: 10132f412da84f456b00e212472ea7d2@192.168.0.27
CSeq: 103 INVITE
Server: CommPeak SIP Proxy
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
[Jan 25 08:12:50] WARNING[8914]: chan_sip.c:18487 handle_response_invite: Re-inv
Transmitting (no NAT) to 50.112.137.243:5060:
ACK sip:XXXXXXXXXXXX@sip.commpeak.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.27:8745;branch=z9hG4bK5a419f31;rport
Max-Forwards: 70
From: "XXXXXXXXXXXX" <sip:XXXXXXXXXXXX@192.168.0.27:8745>;tag=as27b881fd
To: <sip:XXXXXXXXXXXX@sip.commpeak.com>;tag=155c340f586c28d0300cf5a6ccf90d99-acb
Contact: <sip:XXXXXXXXXXXX@192.168.0.27:8745>
Call-ID: 10132f412da84f456b00e212472ea7d2@192.168.0.27
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.24
Content-Length: 0


---
Scheduling destruction of SIP dialog '10132f412da84f456b00e212472ea7d2@192.168.0.27
       > Channel SIP/commpeak-00000029 was never answered.
Scheduling destruction of SIP dialog '10132f412da84f456b00e212472ea7d2@192.168.0.27
Really destroying SIP dialog '5c45f7cb6219531562b8cf78293c8314@192.168.0.27' Method
Really destroying SIP dialog '10132f412da84f456b00e212472ea7d2@192.168.0.27' Method

I have provided the sip trace and i have hidded the contact number for privacy reason. i have actually tried for two number and facing same problem most of the time

Please let me know how can i solve this.

4

This isn’t a complete trace. It is missing the INVITE. In particular it is missing the first line of the INVITE, and it is probably the SIP URI in that which the proxy doesn’t like. You will not be able

Although the Asterisk diagnostic is slightly misleading in this case, it is actually a timeout being reported, for both this and the more literal interpretation of this message, the problem lies with the destination system. At best it is producing an inappropriate response to an unacceptable URI. At worst it is generally broken.

(It could simply be an inappropriate response code for a no answer condition.)

Both inbound and outbound Via header are broken, as they include a private address when the packet is actually coming from a public address. With respect to the outbound ones you probably haven’t set the NAT options (externip, externaddr or stunaddr).