[asterisk-users] ISP trunk session ID?
Nick Khamis
symack at gmail.com
Fri May 10 22:16:39 CDT 2013
Sorry to chime in here, is it possible to change the "Server: Asterisk
", "s=Asterisk", and "o=" within sip.conf? What are the directives
exactly please?
Thanks in Advance,
Nick.
On 5/10/13, Asghar Mohammad <asghar144 at gmail.com> wrote:
> hi,
> you can try to change sip user agent and sdp session s , owner in sip
> config same as your phone,s (modem).
> asterisk by default send user agent = asterisk version , s= asterisk , o=
> asterisk.
> some providers are not happy if they see "asterisk" word :)
>
>
>
> On Sat, May 11, 2013 at 12:27 AM, Sergej Petrovsky
> <sergej5561 at yandex.com>wrote:
>
>> Hi folks,
>>
>> What I trying to do here is exactly this:
>> http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt599.html
>>
>> My provider given me a Huawei modem which have 2 phone jacks on it, but
>> instead of using it I rather redirect my POTS number to my PBX. I ran
>> into
>> couple of bumps on the road but now it's "half-working". I extracted the
>> SIP user, pass, server info from the modem and even managed to put my PBX
>> into the same VLAN they use, on the exact same IP address like the modem
>> but there is 1 problem:
>> It seems this modem also sends some session ID to the ISP's sip server,
>> something what Asterisk doesn't by default. So if I do this:
>>
>> 1, Let the modem register at the sip service (the phone number can be
>> called and ringing out)
>> 2, Disconnect the modem
>> 3, Let the PBX connect to the SIP server
>> 4, PBX accepts the calls
>> 5, About 5-10 minutes later it stops doing it, when I call the number it
>> shows busy (beep, beep, beep), no matter if I restart Asterisk or not it
>> won't work anymore just if I do the same trick again
>>
>> I'm sure the remote SIP server breaks the voip channel or something, it
>> does NOT drop me out tho, my PBX can register any time without problem
>> but
>> no packets will ever come forward me anymore. It's kind of hard to solve
>> this from 1 side.
>>
>> There must be some solution for this.
>>
>> Please help!
>>
>> Thank You,
>> Sergej
>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
More information about the asterisk-users
mailing list