we started with them two days ago .. and we are facing plenty of False Answer cases on several destinations although ppl said they have a policy against FAS.. anyway i don't know i will be looking into another method to send the RTP to another server, thanks for the info
________________________________ > Date: Sat, 10 Apr 2010 18:06:22 -0400 > From: bruceb...@gmail.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Sending RTP media to a different server than > SIP Signaling > > Oh, I see. I haven't done a lot of testing on this new IP since the change of > gateways happened but I did Canada calls and they go fine. However, this > exact provider lies down to their teeth when it comes to problems of call > quality and calls not routing. They never accept faults. They even have > problems sending calls to Canada and USA. They failed to pass calls to India > as well over times. I had a funny issue where they were blocking one specific > area code in USA without even telling us. It was just a regular area code. > They told me it was blocked but I know it was a lie because they wanted to > cover their a$$ as the route was down and it wasn't blocked. > > > I doubt the problem is with sending calls to different media gateway as I > think SIP signals take care of that. Just like canreinvite feature. But I > reserve the right to be wrong. > > -Bruce > > > On Sat, Apr 10, 2010 at 4:45 PM, Tarek Sawah> wrote: > > > > > > you got the name EXACTLY! > > i already am doing what you suggest but facing problems with some > destinations and they claim that the problem is with my Asterisk server not > their routes! > > > > > > > > -- > > AHD Tarek Sawah > > > > Integrated Digital Systems > > > > CCNA, MCSE, RHCE, VoIP > > > > Syria: +963 944 618286 > > > > USA: +1 347 562 2308 > > > > > > > > > > > > > > > > > > ________________________________ > >> Date: Sat, 10 Apr 2010 15:50:52 -0400 > >> From: bruceb...@gmail.com > >> To: asterisk-users@lists.digium.com > >> Subject: Re: [asterisk-users] Sending RTP media to a different server than >> SIP Signaling > >> > >> Just a week ago, I have been in the same situation. Provider was changing >> from Cisco gateways to I think Nextone and hence provided me many IPs. > >> > >> I found out that the media IPs don't matter and just played around with my >> NAT settings and all calls can go through just fine by using simply: > >> > >> > >> host=111.111.111.111 > >> > >> and the 111.111.111.111 is just their SIP signaling IP. Their gateway will >> then transfer asterisk to proper gateways for media. > >> > >> Just give it a try; it should work. But my efforts on finding anything >> regarding this failed on Google as well. > >> > >> > >> P.S. the voip provider name starts with a T and end with A. > >> > >> Regards, > >> Bruce > >> > >> On Sat, Apr 10, 2010 at 11:50 AM, Tarek Sawah> wrote: > >> > >> > >> > >> Greetings list > >> > >> i'm trying to connect with a VoIP provider for termination.. and they have >> offered us three servers to connect with > >> > >> one SIP Signaling server and Two Media servers .. > >> > >> googled for a week and didn't find a way to do this.. so my question. is it >> possible to be done? > >> > >> Asterisk server 1.4.26.3 > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> _________________________________________________________________ > >> > >> The New Busy is not the too busy. Combine all your e-mail accounts with >> Hotmail. > >> > >> http://www.windowslive.com/campaign/thenewbusy?tile=multiaccount&ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 > > >> > >> > >> -- > >> > >> _____________________________________________________________________ > >> > >> -- 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 > >> > >> > > > > _________________________________________________________________ > > The New Busy is not the old busy. Search, chat and e-mail from your inbox. > > http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3 > > > -- > > _____________________________________________________________________ > > -- 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 > > _________________________________________________________________ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_1 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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