Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
There you go. This confirms that SIP signaling determines where the calls should go. I would take their word with a grain of salt specially with their whole support center our of India. No disrespect, but it is bad service overall. -Bruce On Sat, Apr 10, 2010 at 6:32 PM, Joshua Colp jc...@digium.com wrote: - Tarek Sawah tareksa...@hotmail.com wrote: 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, The IP address (and port) of where to send audio is negotiated when the call is setup. You can't change it or specify an IP address to use. Even if you did change the IP address you would be sending it to the port associated with the session on the other media gateway. That would just not work. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
out* of india. On Sun, Apr 11, 2010 at 2:26 AM, bruce bruce bruceb...@gmail.com wrote: There you go. This confirms that SIP signaling determines where the calls should go. I would take their word with a grain of salt specially with their whole support center our of India. No disrespect, but it is bad service overall. -Bruce On Sat, Apr 10, 2010 at 6:32 PM, Joshua Colp jc...@digium.com wrote: - Tarek Sawah tareksa...@hotmail.com wrote: 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, The IP address (and port) of where to send audio is negotiated when the call is setup. You can't change it or specify an IP address to use. Even if you did change the IP address you would be sending it to the port associated with the session on the other media gateway. That would just not work. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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
[asterisk-users] Sending RTP media to a different server than SIP Signaling
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=multiaccountocid=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
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
On Sat, Apr 10, 2010 at 8:50 AM, Tarek Sawah tareksa...@hotmail.com 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 I don't believe this can be done in asterisk by itself, but you may be able to use the Linux conntrack stuff (http://netfilter.org/) to rewrite the SDP host information... However, if you want to dive into the world of OpenSIPS, I know you can do this with an OpenSIPS/MediaProxy setup between your asterisk box and your provider. -- James _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=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 -- _ -- 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
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 tareksa...@hotmail.comwrote: 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=multiaccountocid=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 -- _ -- 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
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
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=multiaccountocid=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
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 tareksa...@hotmail.com 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=multiaccountocid=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 -- _ -- 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
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
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=multiaccountocid=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? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
- Tarek Sawah tareksa...@hotmail.com wrote: 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, The IP address (and port) of where to send audio is negotiated when the call is setup. You can't change it or specify an IP address to use. Even if you did change the IP address you would be sending it to the port associated with the session on the other media gateway. That would just not work. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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
