On 06/01/16 11:00, Sergey Basov wrote:
Thank you for patch.
I have applied patches from your commits 526fb4cdc2246284ecbd7de9ed65678da2dbe357 and 284c11ed16bdc5afcc9abf49c9f5c1749ce831e0 to kamailio 4.3.4 it was applied and build successfully.
There is no more errors in kamailio.log.
Thanks for testing and reporting back.
But Packet which is send by siptrace module to homer capture server is send before topoh modifications.
If you can correct this flow it will be great.
I will look into it as I said before.
Cheers, Daniel
Thank you.
you can see this by looking at contact header:
dump from physical interface to upstream
INVITE sip:1111@10.1.23.23;transport=UDP;user=phone SIP/2.0 Record-Route: sip:10.56.42.23;r2=on;lr;ftag=34ff4c7e;did=0ce.621;nat=yes Record-Route: sip:127.0.0.8;line=sr-BRwEk.dED.d0z.z0z.zvB.afyR8vygavebZAeL7LGVeJGVz2epcMQuTfzVGwD.1Szpc01rTfOuYL Via: SIP/2.0/UDP 10.56.42.23;branch=z9hG4bK4fed.265c2242365ddb0a53289d8a3eac05a7.0 Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-s7wTDLa0zUfYZXl5zpl0zpl0z.lRD.zok.sEG.lvBJY.euwReuTfzpl0zpl0z.lRD.zokqZaxuZ9P.c3BJX01R5fO.wCZLZ3WSqMkhBqGgCceVT7G.G.epGwe.wAzh5jz37jDW7ceh52GpZKDpcSBVfSPh7qzh1E Max-Forwards: 69 Contact: sip:127.0.0.8;line=sr-BRwEk.z8zh1LGplEzhdRzsljzU8jzU8Szh10zLMKGplRzhc7BJX0B2n6BbTfYsZT To: sip:1111@10.1.23.23;transport=UDP From: sip:2222@10.1.23.23;transport=UDP;tag=34ff4c7e Call-ID: Y2EyYmU5NGZiNjYxMGY3MDc0M2ZhYTU1MmNkODFjOWQ. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 382 P-Asserted-Identity: sip:635000161@SNSIP.LIFE.COM;user=phone Expires: 180
v=0 o=Z 0 0 IN IP4 10.56.42.23 s=Z c=IN IP4 10.56.42.23 t=0 0 m=audio 18274 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:18275 a=ice-ufrag:MyrLrVTV a=ice-pwd:uoYbKhQTednj4xw3aF2R0ysK6R a=candidate:1zTwRqKsCajOhkZK 1 UDP 2130706431 10.56.42.23 18274 typ host a=candidate:1zTwRqKsCajOhkZK 2 UDP 2130706430 10.56.42.23 18275 typ host
================
dump from homer db
INVITE sip:1111@10.1.23.23;transport=UDP;user=phone SIP/2.0 Record-Route: sip:10.56.42.23;r2=on;lr;ftag=34ff4c7e;did=0ce.621;nat=yes Record-Route: sip:10.1.23.23;r2=on;lr;ftag=34ff4c7e;did=0ce.621;nat=yes Via: SIP/2.0/UDP 10.56.42.23;branch=z9hG4bK4fed.265c2242365ddb0a53289d8a3eac05a7.0 Via: SIP/2.0/UDP 10.10.206.39:5060;received=10.10.206.39;TH=div;branch=z9hG4bK-d8754z-dd463ce3ef9a0812-1---d8754z-;rport=5060 Max-Forwards: 69 Contact: sip:2222@10.10.206.39:5060;transport=UDP To: sip:1111@10.1.23.23;transport=UDP From: sip:2222@10.1.23.23;transport=UDP;tag=34ff4c7e Call-ID: Y2EyYmU5NGZiNjYxMGY3MDc0M2ZhYTU1MmNkODFjOWQ. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 382 TH: dih P-Asserted-Identity: sip:635000161@SNSIP.LIFE.COM;user=phone Expires: 180
v=0 o=Z 0 0 IN IP4 10.56.42.23 s=Z c=IN IP4 10.56.42.23 t=0 0 m=audio 18274 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:18275 a=ice-ufrag:MyrLrVTV a=ice-pwd:uoYbKhQTednj4xw3aF2R0ysK6R a=candidate:1zTwRqKsCajOhkZK 1 UDP 2130706431 10.56.42.23 18274 typ host a=candidate:1zTwRqKsCajOhkZK 2 UDP 2130706430 10.56.42.23 18275 typ host -- Best regards, Sergey Basov e-mail: sergey.v.basov@gmail.com
tel: (+38067) 403-62-54
2016-01-06 11:08 GMT+02:00 Daniel-Constantin Mierla miconda@gmail.com:
On 06/01/16 09:19, Sergey Basov wrote:
Sorry, not only ACK. There is ACK, INVITE, ACK and BYE
OK.
I pushed two patches to master branch that should fix the issue with the error log messages. Can you test with master branch? If all goes fine, then I will consider backporting.
For capturing the packets after topoh, more time is needed to analyze and implement if possible.
Cheers, Daniel
I am using kamailio as statefull proxy.
So in case of normal callflow there is REGISTER and so on..
And with sip_trace enabled in onsend_route errors in kamailio.log still persist.
Jan 6 10:12:01 sip1 /usr/sbin/kamailio[30545]: INFO: <script>: onsend_route entered !!! Jan 6 10:12:01 sip1 /usr/sbin/kamailio[30545]: INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP Jan 6 10:12:01 sip1 /usr/sbin/kamailio[30545]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Jan 6 10:12:01 sip1 /usr/sbin/kamailio[30545]: ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023�#023�#0128*#027#0128*#024Q̌V��>
Is there any possibility do sending message via siptrace module after topoh module?
WBR
Best regards, Sergey Basov e-mail: sergey.v.basov@gmail.com
tel: (+38067) 403-62-54
2016-01-06 10:02 GMT+02:00 Daniel-Constantin Mierla miconda@gmail.com:
No other SIP requests? Only the ACK?
Cheers, Daniel
On 06/01/16 07:55, Sergey Basov wrote:
I just try it. I got only last ACK when call was finished.
I think than siptrace must catch and send duplicate message after topoh module. Because in other case we can see message as it is inside kamailio, not message that will be send to upstream or client...
WBR.
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu