[asterisk-users] canreinvite=yes problem

Thu Dec 4 09:00:44 CST 2008

I still have:
Client 1
---------Asterisk1--Asterisk2
Client 2

When client1 do a call, asterisk1 forward to asterisk2, asterisk2 forward to
Asterisk1
At this moment, asterisk1 say : 404Not found
But I have insecure=very

This is the sip debug at that moment:

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

<--- SIP read from UDP://192.168.1.151:5060 --->
INVITE sip:33170725012 at 192.168.1.153 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK2b4a242e;rport
Max-Forwards: 70
From: "103" <sip:103 at 192.168.1.151>;tag=as636875d3
To: <sip:33170725012 at 192.168.1.153>
Contact: <sip:103 at 192.168.1.151>
Call-ID: 4bdd7c785c834d662f9523ce5460bf44 at 192.168.1.151
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Thu, 04 Dec 2008 14:55:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 1545198644 1545198644 IN IP4 192.168.1.151
s=Asterisk PBX 1.6.0.1
c=IN IP4 192.168.1.151
t=0 0
m=audio 12272 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (14 headers 13 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 192.168.1.151 : 5060 (NAT)
Using INVITE request as basis request -
4bdd7c785c834d662f9523ce5460bf44 at 192.168.1.151
No user '103' in SIP users list
Found peer 'media' for '103' from 192.168.1.151:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.151:12272
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.151:12272
Looking for 33170725012 in media (domain 192.168.1.153)

<--- Reliably Transmitting (no NAT) to 192.168.1.151:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.1.151:5060;branch=z9hG4bK2b4a242e;received=192.168.1.151;rport=5060
From: "103" <sip:103 at 192.168.1.151>;tag=as636875d3
To: <sip:33170725012 at 192.168.1.153>;tag=as242de969
Call-ID: 4bdd7c785c834d662f9523ce5460bf44 at 192.168.1.151
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

Have you an idea why ?

-----Message d'origine-----
De : asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] De la part de BERGANZ
François
Envoyé : jeudi 4 décembre 2008 09:15
À : 'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet : Re: [asterisk-users] canreinvite=yes problem

Now, I have :

Client 1
---------Asterisk1--Asterisk2
Client 2

I need that sip sign go to Asterisk2
But RTP go to Asterisk1 and no more.

Where have I to insert canreinvite ?

Thank you

-----Message d'origine-----
De : asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] De la part de Eric
"ManxPower" Wieling
Envoyé : mercredi 3 décembre 2008 19:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] canreinvite=yes problem

canreinvite=yes should work as long as 1) there is no NAT involved 
anywhere in the call path, 2) All legs of the call are using the same 
codec, 3) you do not have the t/T/w/W (and maybe a few other) options to 
the Dial line.

Remember the only way you can really tell if a reinvite happens is by 
looking at the RTP audio.  The SIP signaling will not and has never had 
a "reinvite" feature for signaling.

Why did you post the same message at :23, :28, and :35 mins past the 
hour?  If you need immediate support you should contact Digium support 
and pay for a service contract.

BERGANZ François wrote:
> I need to test canreinvite=yes with 2softphones and 1 asterisk.
> I want to have that :
>
http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb
> ridge.png
> But I have that http://www.zimagez.com/zimage/canreinvite.php
> Canreinvite=yes work for all phones or just asterisk?...

-- 
Consulting and design services for LAN, WAN, voice and data.  Based near 
Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs 
echo canceling systems.  Also see http://www.fnords.org/skillslist.html

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