Check if the both endpoints are receiving the packets properly and also, if
your billing system is working properly. Some bad formed packets may result
in call drop.
Fernando Schmitt
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of serusers-request(a)lists.iptel.org
Sent: quarta-feira, 19 de outubro de 2005 08:00
To: serusers(a)lists.iptel.org
Subject: Serusers Digest, Vol 30, Issue 19
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Today's Topics:
1. Re: mangling request URI according to To: and userlocation
value (Francesco Fondelli)
2. call drops after about 30 seconds (maka)
3. 423 message for Register Expires (share phone)
4. SIP Dialer Sending Many BYE Packets (sagar)
5. Re: [OT] MAX TNT as a Media Gateway (Vamsi Pottangi)
6. Can I send Binary Data through SIP IM? (Abhijit A. Mahajani)
7. How many BHCC? (Matteo Piazza)
----------------------------------------------------------------------
Message: 1
Date: Tue, 18 Oct 2005 14:11:11 +0200
From: Francesco Fondelli <francesco.fondelli(a)gmail.com>
Subject: [Serusers] Re: mangling request URI according to To: and
userlocation value
To: serusers(a)lists.iptel.org
Message-ID: <4354E65F.5030601(a)gmail.com>
Content-Type: text/plain; charset=us-ascii; format=flowed
and assume that entry in userloc has a "Server:
Bar 1.0 PBX" value
sorry, here I meant the "User-Agent: 'Bar 1.0 PBX'" entry.
thank you very much
Ciao
FF
------------------------------
Message: 2
Date: Tue, 18 Oct 2005 15:32:30 +0300
From: maka <icokan(a)gmail.com>
Subject: [Serusers] call drops after about 30 seconds
To: SER <serusers(a)lists.iptel.org>
Message-ID:
<826761540510180532v4683e151x2a9a6ba209f1061d(a)mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
hello everyone,
I am using ser-0.9.0 together wit asterisk-1.0.6. I am testing it with a
couple of hardware AT-320 IP phones from Atcom, using a PA168S chip.
They seem to be working fine, with stun behind nat, and they can call other
user-agents (softphones, asterisk adn the pstn through it), but whenever I
make a call between the two phones, everytime the call is dropped after
about 34 seconds, even when calling between phones on Public IP addresses.
I am actually clueless why this happens, I tried changing the NAT ttl value,
bu to no effect, and it is not supposed to be a codec mismatch too since
both phones use absolutely the same codecs, in the same order of preference.
Appreciate the feedback, cheers
--
I'm sick and tired of being sick and tired...