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ERROR:core:forward_reply: no 2nd via found in reply
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rowellOffline



Joined: Aug 07, 2009
Posts: 5

Status: Offline
Posted: Feb 22, 2010 - 05:01 AM Reply with quote Back to top
Hi can somebody help me to fix my issue? I am redirecting my voicemail to Asterisk from Openser but I can't hear anything saying "Please leave your message". When I checked the logs on OpenSER it says this error---- ERROR:core:forward_reply: no 2nd via found in reply

Also it keeps on sending SIP/2.0 487 Request terminated

here is a sip trace

s=Asterisk PBX 1.6.2.4.
c=IN IP4 216.115.67.83.
t=0 0.
m=audio 11258 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.
a=sendrecv.


U 2010/02/21 21:34:02.037277 216.115.67.80:5060 -> 216.115.67.83:5060
ACK sip:8664871222@216.115.67.83 SIP/2.0.
Via: SIP/2.0/UDP ser.gowireless.net:5060;branch=z9hG4bKaae6.c04a08c1.2.
Via: SIP/2.0/UDP 192.168.2.108:13040;received=58.69.139.140;branch=z9hG4bK-d8754z-806b223c922b112c-1---d8754z-;rport=29735.
Max-Forwards: 69.
Contact: <sip:92001@58.69.139.140:29735>.
To: "8664871222"<sip:8664871222@ser.gowireless.net>;tag=as6fe400f1.
From: "92001"<sip:92001@ser.gowireless.net>;tag=b451c80b.
Call-ID: Mzk2ZmYzZDExNWMwM2E1ZWMzN2RhZDRiNWUyNWQ4MzE..
CSeq: 2 ACK.
Proxy-Authorization: Digest username="92001",realm="ser.gowireless.net",nonce="4b8217fd00000117cc5d5370ede6751e9b7b33550ef9a9a4",uri="sip:8664871222@ser.gowireless.net",response="8c3db7b28bce5e7f28fae622461885e8",algorithm=MD5.
User-Agent: X-Lite release 1104o stamp 56125.
Content-Length: 0.
.


U 2010/02/21 21:34:15.959384 216.115.67.83:5060 -> 216.115.67.80:5060
INVITE sip:92001@216.115.67.80 SIP/2.0.
Via: SIP/2.0/UDP 216.115.67.83:5060;branch=z9hG4bK7da8126f;rport.
Max-Forwards: 70.
From: "92001" <sip:92001@216.115.67.83>;tag=as08c64457.
To: <sip:92001@216.115.67.80>.
Contact: <sip:92001@216.115.67.83>.
Call-ID: 4f260c6035e73bbc3734f8ec58f2889f [!at] 216.115.67.83 (replace the [!at] with a @).
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 1.6.2.4.
Date: Mon, 22 Feb 2010 05:34:15 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 265.
.
v=0.
o=root 1578025016 1578025016 IN IP4 216.115.67.83.
s=Asterisk PBX 1.6.2.4.
c=IN IP4 216.115.67.83.
t=0 0.
m=audio 16666 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.
a=sendrecv.


U 2010/02/21 21:34:15.960444 216.115.67.80:5060 -> 216.115.67.83:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 216.115.67.83:5060;branch=z9hG4bK7da8126f;rport=5060;received=216.115.67.83.
From: "92001" <sip:92001@216.115.67.83>;tag=as08c64457.
To: <sip:92001@216.115.67.80>.
Call-ID: 4f260c6035e73bbc3734f8ec58f2889f [!at] 216.115.67.83 (replace the [!at] with a @).
CSeq: 102 INVITE.
Server: Kamailio (1.5.4-notls (i386/linux)).
Content-Length: 0.
.


U 2010/02/21 21:34:16.828784 216.115.67.80:5060 -> 216.115.67.83:5060
INVITE sip:vm92001@pbx.gowireless.net:5060;rinstance=f608ac4e288088e8 SIP/2.0.
Record-Route: <sip:216.115.67.80;lr=on;nat=yes>.
Via: SIP/2.0/UDP ser.gowireless.net:5060;branch=z9hG4bK8ed1.7a995d67.1.
Via: SIP/2.0/UDP 216.115.67.83:5060;received=216.115.67.83;branch=z9hG4bK7da8126f;rport=5060.
Max-Forwards: 69.
From: "92001" <sip:92001@216.115.67.83>;tag=as08c64457.
To: <sip:92001@216.115.67.80>.
Contact: <sip:92001@216.115.67.83>.
Call-ID: 4f260c6035e73bbc3734f8ec58f2889f [!at] 216.115.67.83 (replace the [!at] with a @).
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 1.6.2.4.
Date: Mon, 22 Feb 2010 05:34:15 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 283.
.
v=0.
o=root 1578025016 1578025016 IN IP4 216.115.67.83.
s=Asterisk PBX 1.6.2.4.
c=IN IP4 216.115.67.80.
t=0 0.
m=audio 59876 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.
a=sendrecv.
a=nortpproxy:yes.


U 2010/02/21 21:34:16.829020 216.115.67.83:5060 -> 216.115.67.80:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP ser.gowireless.net:5060;branch=z9hG4bK8ed1.7a995d67.1;received=216.115.67.80.
Via: SIP/2.0/UDP 216.115.67.83:5060;received=216.115.67.83;branch=z9hG4bK7da8126f;rport=5060.
Record-Route: <sip:216.115.67.80;lr=on;nat=yes>.
From: "92001" <sip:92001@216.115.67.83>;tag=as08c64457.
To: <sip:92001@216.115.67.80>.
Call-ID: 4f260c6035e73bbc3734f8ec58f2889f [!at] 216.115.67.83 (replace the [!at] with a @).
CSeq: 102 INVITE.
Server: Asterisk PBX 1.6.2.4.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Contact: <sip:92001@216.115.67.83>.
Content-Length: 0.
.


U 2010/02/21 21:34:16.829531 216.115.67.83:5060 -> 216.115.67.80:5060
CANCEL sip:92001@216.115.67.80 SIP/2.0.
Via: SIP/2.0/UDP 216.115.67.83:5060;branch=z9hG4bK7da8126f;rport.
Max-Forwards: 70.
From: "92001" <sip:92001@216.115.67.83>;tag=as08c64457.
To: <sip:92001@216.115.67.80>.
Call-ID: 4f260c6035e73bbc3734f8ec58f2889f [!at] 216.115.67.83 (replace the [!at] with a @).
CSeq: 102 CANCEL.
User-Agent: Asterisk PBX 1.6.2.4.
Content-Length: 0.
.


U 2010/02/21 21:34:16.829943 216.115.67.80:5060 -> 216.115.67.83:5060
SIP/2.0 200 canceling.
Via: SIP/2.0/UDP 216.115.67.83:5060;branch=z9hG4bK7da8126f;rport=5060;received=216.115.67.83.
From: "92001" <sip:92001@216.115.67.83>;tag=as08c64457.
To: <sip:92001@216.115.67.80>;tag=bbd8f31c58bc6a3266ec5509a254d55b-120f.
Call-ID: 4f260c6035e73bbc3734f8ec58f2889f [!at] 216.115.67.83 (replace the [!at] with a @).
CSeq: 102 CANCEL.
Server: Kamailio (1.5.4-notls (i386/linux)).
Content-Length: 0.
.


U 2010/02/21 21:34:16.829951 216.115.67.80:5060 -> 216.115.67.83:5060
CANCEL sip:vm92001@pbx.gowireless.net:5060;rinstance=f608ac4e288088e8 SIP/2.0.
Via: SIP/2.0/UDP ser.gowireless.net:5060;branch=z9hG4bK8ed1.7a995d67.1.
From: "92001" <sip:92001@216.115.67.83>;tag=as08c64457.
Call-ID: 4f260c6035e73bbc3734f8ec58f2889f [!at] 216.115.67.83 (replace the [!at] with a @).
To: <sip:92001@216.115.67.80>.
CSeq: 102 CANCEL.
Max-Forwards: 70.
User-Agent: Kamailio (1.5.4-notls (i386/linux)).
Content-Length: 0.
.


U 2010/02/21 21:34:16.830138 216.115.67.83:5060 -> 216.115.67.80:5060
SIP/2.0 487 Request Terminated.
Via: SIP/2.0/UDP 216.115.67.83:5060;branch=z9hG4bK7da8126f;received=216.115.67.80;rport=5060.
From: "92001" <sip:92001@216.115.67.83>;tag=as08c64457.
To: <sip:92001@216.115.67.80>;tag=bbd8f31c58bc6a3266ec5509a254d55b-120f.
Call-ID: 4f260c6035e73bbc3734f8ec58f2889f [!at] 216.115.67.83 (replace the [!at] with a @).
CSeq: 102 INVITE.
Server: Asterisk PBX 1.6.2.4.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Length: 0.
.


U 2010/02/21 21:34:16.830154 216.115.67.83:5060 -> 216.115.67.80:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP ser.gowireless.net:5060;branch=z9hG4bK8ed1.7a995d67.1;received=216.115.67.80.
From: "92001" <sip:92001@216.115.67.83>;tag=as08c64457.
To: <sip:92001@216.115.67.80>;tag=bbd8f31c58bc6a3266ec5509a254d55b-120f.
Call-ID: 4f260c6035e73bbc3734f8ec58f2889f [!at] 216.115.67.83 (replace the [!at] with a @).
CSeq: 102 CANCEL.
Server: Asterisk PBX 1.6.2.4.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Length: 0.
.


U 2010/02/21 21:34:17.830718 216.115.67.83:5060 -> 216.115.67.80:5060
SIP/2.0 487 Request Terminated.
Via: SIP/2.0/UDP 216.115.67.83:5060;branch=z9hG4bK7da8126f;received=216.115.67.80;rport=5060.
From: "92001" <sip:92001@216.115.67.83>;tag=as08c64457.
To: <sip:92001@216.115.67.80>;tag=bbd8f31c58bc6a3266ec5509a254d55b-120f.
Call-ID: 4f260c6035e73bbc3734f8ec58f2889f [!at] 216.115.67.83 (replace the [!at] with a @).
CSeq: 102 INVITE.
Server: Asterisk PBX 1.6.2.4.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Length: 0.
.


U 2010/02/21 21:34:18.829549 216.115.67.83:5060 -> 216.115.67.80:5060
SIP/2.0 487 Request Terminated.
Via: SIP/2.0/UDP 216.115.67.83:5060;branch=z9hG4bK7da8126f;received=216.115.67.80;rport=5060.
From: "92001" <sip:92001@216.115.67.83>;tag=as08c64457.
To: <sip:92001@216.115.67.80>;tag=bbd8f31c58bc6a3266ec5509a254d55b-120f.
Call-ID: 4f260c6035e73bbc3734f8ec58f2889f [!at] 216.115.67.83 (replace the [!at] with a @).
CSeq: 102 INVITE.
Server: Asterisk PBX 1.6.2.4.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Length: 0.
.


U 2010/02/21 21:34:20.829266 216.115.67.83:5060 -> 216.115.67.80:5060
SIP/2.0 487 Request Terminated.
Via: SIP/2.0/UDP 216.115.67.83:5060;branch=z9hG4bK7da8126f;received=216.115.67.80;rport=5060.
From: "92001" <sip:92001@216.115.67.83>;tag=as08c64457.
To: <sip:92001@216.115.67.80>;tag=bbd8f31c58bc6a3266ec5509a254d55b-120f.
Call-ID: 4f260c6035e73bbc3734f8ec58f2889f [!at] 216.115.67.83 (replace the [!at] with a @).
CSeq: 102 INVITE.
Server: Asterisk PBX 1.6.2.4.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Length: 0.
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micondaOffline



Joined: Feb 02, 2007
Posts: 357
Location: Germany
Status: Offline
Posted: Feb 25, 2010 - 10:23 AM Reply with quote Back to top
The error might not be related. It states that sip server received a sip reply with only one Via header, therefore it does not know where to send the reply.

Probably it is a problem with nat traversal logic in your config.

I suggest you use kamailio (openser) 3.0 with default config file, enable nat traversal logic via the #!define directive and give it a try to see if you get audio.
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