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 <[email protected]> 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 > <[email protected]>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
