[Asterisk-Users] TO header field bug using asterisk
brian k. west
brian at bkw.org
Sat May 15 10:08:39 MST 2004
you also have
fromuser
fromdomain
use em
bkw
----- Original Message -----
From: "usmankhan" <usmankhan at iphonica.com>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, May 15, 2004 6:10 AM
Subject: [Asterisk-Users] TO header field bug using asterisk
> Hi,
>
> I have setup and configured asterisk server using SIP. I defined
> two users 2000 and 2001 in the sip.conf file. The extension
> defintions from sip.conf are shown below:
>
> [2000]
> type=friend
> username=2000
> secret=hi
> host=dynamic
> context=from-sip
> mailbox=100
>
> [2001]
> type=friend
> username=2001
> secret=hi
> host=dynamic
> context=from-sip
> mailbox=101
>
> I made entries in extensions.conf, shown below, to place calls to
> 2000 and 2001
>
> [from-sip]
> exten=>2000,1,Dial(SIP/2000,20)
> exten=>2000,2,Voicemail(u2000)
> exten=>2000,102,Voicemail(b2000)
> exten=>2000,103,Hangup
>
> exten=>2001,1,Dial(SIP/2001,20)
> exten=>2001,2,Voicemail(u2001)
> exten=>2001,102,Voicemail(b2001)
> exten=>2001,103,Hangup
>
> Initially I tested out the server by registering SJPhone user
> agents and successfully placing calls between them.
>
> Next I replaced the SJPhones with our VOIP gateways. Everytime I dialed
> either extension I always got the unavailable IVR message. I tried looking
> deeper into the problem and took ethereal traces and was able to
> isolate the problem. For some reason asterisk has problems in
> rewriting the TO header field when it forwards the INVITE request to
> the callee. This is what the TO header field looks like when it is
> sent by the caller to asterisk
> (192.168.0.44 is the IP address of asterisk):
>
> To: <sip:2000 at 192.168.0.44:5060>
>
> and this is what it looks like when it is forwarded to the callee by
> asterisk (192.168.0.243 is the IP address of the callee).
>
> To: <sip:192.168.0.243>
>
> Since the URI does not contain the user part the callee replies with
> 404 not found and the call fails. I have thought hard, compared
> signaling traces but cant really make out how to make my gateways
> work, seems like an asterisk bug. Any ideas? I would really appreciate any
> help in this regard.
>
> Regards,
> Danish
>
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