I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060617/8f4449fa/attachment.htm
Il Neofita wrote:> I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, > if I call the traffic still go throw the asterisk. How come?Are you using the same codecs on the SPA3000 and the xlite? If no then there's your reason. -- Linux Home Automation Neil Cherry ncherry@linuxha.com http://www.linuxha.com/ Main site http://linuxha.blogspot.com/ My HA Blog http://home.comcast.net/~ncherry/ Backup site
What does your dial command look like? On 6/17/06, Il Neofita <asteriskmail@gmail.com> wrote:> I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if > I call the traffic still go throw the asterisk. How come? > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > >
This is the dial in extensions exten => _40001,1,Dial(SIP/40001,30) exten => _40002,1,Dial(SIP/40002,30) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060618/e155162e/attachment.htm
cosa vedo a console
-- Executing Dial("SIP/40001-3760", "SIP/40002|30") in
new stack
-- Called 40002
-- SIP/40002-4753 is ringing
-- SIP/40002-4753 answered SIP/40001-3760
-- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753
srvlinux*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last
Message
82.X2.XX3.X3 40002 146b518a4cd 00103/00000 alaw No Tx:
ACK
82.X2.XX3.X3 40001 CBD1DB85-8B 00102/30987 alaw No Tx:
ACK
2 active SIP channels
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How have you confirmed that they did not reinvite? The channels are still
controlled by Asterisk (sip signalling), it is the rtp streams that go direct.
You can do a sip show channel 146b518a4cd on the specific channel to see where
the rtp streams are going. Or ... if this is the only active channel on the box,
just do a rtp debug. If the rtp stream is going through asterisk, it will be
very obvious. If not, you won't see a constant flow of rtp debug messages
going on.
p
From: "Il Neofita" <asteriskmail@gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Date: Sun, 18 Jun 2006 05:01:20 -0400
Subject: Re: [Asterisk-Users] Canreinvite
This is the dial in extensions
exten => _40001,1,Dial(SIP/40001,30)
exten => _40002,1,Dial(SIP/40002,30)
From: "Il Neofita" <asteriskmail@gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Date: Sun, 18 Jun 2006 05:22:35 -0400
Subject: Re: [Asterisk-Users] Canreinvite
cosa vedo a console
-- Executing Dial("SIP/40001-3760", "SIP/40002|30") in
new stack
-- Called 40002
-- SIP/40002-4753 is ringing
-- SIP/40002-4753 answered SIP/40001-3760
-- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753
srvlinux*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last
Message
82.X2.XX3.X3 40002 146b518a4cd 00103/00000 alaw No Tx: ACK
82.X2.XX3.X3 40001 CBD1DB85-8B 00102/30987 alaw No Tx: ACK
2 active SIP channels
---------------------------------
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How can I check if SIP re-invite is really working ? I'm trying it with two grandstream gxp2000. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060728/6b85c10a/attachment.htm