[asterisk-users] ISP trunk session ID?

[Asghar Mohammad]

you can find in [general]  section.
useragent=asterisk        ; Allows you to change the user agent string
                                ; The default user agent string also
contains the Asterisk
                                ; version. If you don't want to expose
this, change the
                                ; useragent string.
sdpsession=asterisk        ; Allows you to change the SDP session name
string, (s=)
                                ; Like the useragent parameter, the default
user agent string
                                ; also contains the Asterisk version.
;sdpowner=root                  ; Allows you to change the username field
in the SDP owner string, (o=)


On Sat, May 11, 2013 at 5:16 AM, Nick Khamis <symack at gmail.com> wrote:

> 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
> >>
> >
>
> --
> _____________________________________________________________________
> -- 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130511/7766b401/attachment.htm>