Hello, is it possible to rout ALL RTP Data over Asterisk, like SIP1 <---RTP---> Asterisk <---RTP---> SIP2 I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;) Thanx, Kalle
Try setting canreinvite = no in sip.conf or the database (where you have sipuser setting). Regards, Sanjay Rajdev ----- Original Message ----- From: "kalle odenthal" <kalle.odenthal@genion.de> To: asterisk-users@lists.digium.com Sent: Friday, March 30, 2007 5:52:47 AM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] SIP RTP Tunnel Hello, is it possible to rout ALL RTP Data over Asterisk, like SIP1 <---RTP---> Asterisk <---RTP---> SIP2 I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;) Thanx, Kalle _______________________________________________ --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
Hola Sanjay, this works pretty well in one direction. The Sip User who is registered at the Asterisk. But the Sip user who calls from sipXYZ.com still sends it data diretly to sip user 1. Any idea? Thanx!! -----Original Message----- From: Sanjay Rajdev [mailto:sanjay.rajdev@featherstoneinformatics.com] Sent: Donnerstag, 29. M?rz 2007 18:27 To: kalle odenthal; Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP RTP Tunnel Try setting canreinvite = no in sip.conf or the database (where you have sipuser setting). Regards, Sanjay Rajdev ----- Original Message ----- From: "kalle odenthal" <kalle.odenthal@genion.de> To: asterisk-users@lists.digium.com Sent: Friday, March 30, 2007 5:52:47 AM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] SIP RTP Tunnel Hello, is it possible to rout ALL RTP Data over Asterisk, like SIP1 <---RTP---> Asterisk <---RTP---> SIP2 I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;) Thanx, Kalle _______________________________________________ --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
Also set canreinvite=no between Asterisk and the provider. kalle.odenthal@genion.de wrote:> Hola Sanjay, > > this works pretty well in one direction. The Sip User who is registered at the Asterisk. But the Sip user who calls from sipXYZ.com still sends it data diretly to sip user 1. > > Any idea? > > Thanx!! > > -----Original Message----- > From: Sanjay Rajdev [mailto:sanjay.rajdev@featherstoneinformatics.com] > Sent: Donnerstag, 29. M?rz 2007 18:27 > To: kalle odenthal; Asterisk Users Mailing List - Non-Commercial Discussion > Cc: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] SIP RTP Tunnel > > Try setting canreinvite = no in sip.conf or the database (where you have sipuser setting). > > Regards, > Sanjay Rajdev > > ----- Original Message ----- > From: "kalle odenthal" <kalle.odenthal@genion.de> > To: asterisk-users@lists.digium.com > Sent: Friday, March 30, 2007 5:52:47 AM (GMT+0530) Asia/Calcutta > Subject: [asterisk-users] SIP RTP Tunnel > > Hello, > > is it possible to rout ALL RTP Data over Asterisk, like > > SIP1 <---RTP---> Asterisk <---RTP---> SIP2 > > I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;) > > Thanx, > > Kalle > > > > > _______________________________________________ > --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 > > > > > > _______________________________________________ > --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 >
You should get a packet capture and look at the SDP that is agreed to by
both parties. It sounds like someone is not honoring it.
--------------------------------------------------
Salvatore Giudice
Salvatore.Giudice@VoIPSecurityTraining.com
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
kalle.odenthal@genion.de
Sent: Thursday, March 29, 2007 9:35 PM
To: asterisk-users@lists.digium.com
Subject: RE: RE: [asterisk-users] SIP RTP Tunnel
Hola Sanjay,
this works pretty well in one direction. The Sip User who is registered at
the Asterisk. But the Sip user who calls from sipXYZ.com still sends it data
diretly to sip user 1.
Any idea?
Thanx!!
-----Original Message-----
From: Sanjay Rajdev [mailto:sanjay.rajdev@featherstoneinformatics.com]
Sent: Donnerstag, 29. M?rz 2007 18:27
To: kalle odenthal; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP RTP Tunnel
Try setting canreinvite = no in sip.conf or the database (where you have
sipuser setting).
Regards,
Sanjay Rajdev
----- Original Message -----
From: "kalle odenthal" <kalle.odenthal@genion.de>
To: asterisk-users@lists.digium.com
Sent: Friday, March 30, 2007 5:52:47 AM (GMT+0530) Asia/Calcutta
Subject: [asterisk-users] SIP RTP Tunnel
Hello,
is it possible to rout ALL RTP Data over Asterisk, like
SIP1 <---RTP---> Asterisk <---RTP---> SIP2
I know it seems quite useless. But I want to simulate a IAX -> SIP
connection and have no Phonecard installed on my computer ;)
Thanx,
Kalle
_______________________________________________
--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
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
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Thank you Salvatore,
I have Wireshark ;) But my problem is not to SEE the packets.
I'm trying to do a Handover between VoIP and GSM using an Asterisk Server.
But to avoid the effort to implement the channel: Asterisk->public switched
network->GSM->...radio...->GSM modem-> SmartPhone, I want to
substitute the "GSM" Channel with SIP. This is valid for my work. The
problem is if Asterisk DO tunnel all Packets through its bones when I switch
between SIP and GSM, it connects two SIP participants direcly.
You see my problem?
Saludos,
Kalle
You should get a packet capture and look at the SDP that is agreed to by both
parties. It sounds like someone is not honoring it.
--------------------------------------------------
Salvatore Giudice
Salvatore.Giudice@VoIPSecurityTraining.com
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
kalle.odenthal@genion.de
Sent: Thursday, March 29, 2007 9:35 PM
To: asterisk-users@lists.digium.com
Subject: RE: RE: [asterisk-users] SIP RTP Tunnel
Hola Sanjay,
this works pretty well in one direction. The Sip User who is registered at the
Asterisk. But the Sip user who calls from sipXYZ.com still sends it data diretly
to sip user 1.
Any idea?
Thanx!!
-----Original Message-----
From: Sanjay Rajdev [mailto:sanjay.rajdev@featherstoneinformatics.com]
Sent: Donnerstag, 29. M?rz 2007 18:27
To: kalle odenthal; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP RTP Tunnel
Try setting canreinvite = no in sip.conf or the database (where you have sipuser
setting).
Regards,
Sanjay Rajdev
----- Original Message -----
From: "kalle odenthal" <kalle.odenthal@genion.de>
To: asterisk-users@lists.digium.com
Sent: Friday, March 30, 2007 5:52:47 AM (GMT+0530) Asia/Calcutta
Subject: [asterisk-users] SIP RTP Tunnel
Hello,
is it possible to rout ALL RTP Data over Asterisk, like
SIP1 <---RTP---> Asterisk <---RTP---> SIP2
I know it seems quite useless. But I want to simulate a IAX -> SIP connection
and have no Phonecard installed on my computer ;)
Thanx,
Kalle
_______________________________________________
--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
_______________________________________________
--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