[asterisk-users] One way audio when dialing multiple registrations
Hi again today when i was doing my research on this issue i found that even dialing a sip user by it's IP also raises this problem. here is what i did, First I dialed my registered user in normal way like this, Dial(SIP/XYZ,30,rtT) and during conversation audio was OK in both ways. Then I dialed the registered user via it's ip and port to which it was registered. like this, Dial(SIP/x...@:5062,30,rtT) during conversation audio was one way just like before (calling party can hear called party but called party can not hear calling). after taking debug trace of both methods what I found was that a SIP HEADER parameter rinstance was missing in to and INVITEt header fields when dialing via IP:PORT. I think this parameter is assigned automatically by asterisk. *NORMAL DIAL * INVITE sip:x...@:28614;rinstance=0266b8b94f488588 SIP/2.0 To: sip:x...@:28614;rinstance=0266b8b94f488588 Contact: sip:1334225...@xxx:5060 *IP DIAL* INVITE sip:x...@xxx:28614 SIP/2.0 To: sip:x...@:28614 Contact: sip:1334225...@xxx:5060 hope this research will ease a bit the quest to find a solution. now question is 1) how can we assign this parameter when dialing via IP:PORT? 2) what else options do we have if we want to dial using IP:PORT mechanism. waiting for your kind resopnse. Nasir Javaid. --- --- sorry for the typo mistake. the actual dial string that I used is like this Dial(SIP/x...@:5062-096afee8,30,rtT) Dial(SIP/x...@:64290-0966ab80,30,rtT) it is not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) it was just a typing mistake that may have diverted all of you. hope this clears what i am trying to do. regards, Nasir Javaid --- I am sure you can't achieve what you are trying to achieve here. Simply use two different extensions instead of one. Considering how SIP communication works, I believe SIP doesn't allow multiple registrations like this. Maybe somebody can correct me here if I am wrong. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-19 12:28 PM, Nasir Javaid nasirjavaidna...@x wrote: thanks a lot zishan and philipp, probably that is the problem that is occurring. I am gonna take some wireshark or etherial trace to further investigate the problem. i don't wanna stuck into port forwarding issue as it will waste lot of time and also normal calling is working on my current port forwarding. what i am currently trying to grab the channel name along with it's unique id and dial it directly like simple Dial(SIP/xyz ) dialing for example Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) ^ | | | but problem is that asterisk assigns random unique-id for every call. and also it is available only when dialing... what are my options? your help will be highly appreciated. regards, Naisr Javaid - Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fashion I have had this problem once and I never tried registering same extension from two different places after that. Try Phillip's suggestion, maybe it'll work for you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 11:42 AM, Philipp von Klitzing klitz...@xx wrote: Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works f... You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp --- Hi Zeeshan, I saw many of your posts on forum. i also put my problem on forum but did not get any satisfying answer. I wish if you could help me out. below is my post. ==
[asterisk-users] One way audio when dialing multiple registrations
sorry for the typo mistake. the actual dial string that I used is like this Dial(SIP/x...@192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/x...@192.168.0.12:64290-0966ab80,30,rtT) it is not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) it was just a typing mistake that may have diverted all of you. hope this clears what i am trying to do. regards, Nasir Javaid --- I am sure you can't achieve what you are trying to achieve here. Simply use two different extensions instead of one. Considering how SIP communication works, I believe SIP doesn't allow multiple registrations like this. Maybe somebody can correct me here if I am wrong. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-19 12:28 PM, Nasir Javaid nasirjavaidna...@x wrote: thanks a lot zishan and philipp, probably that is the problem that is occurring. I am gonna take some wireshark or etherial trace to further investigate the problem. i don't wanna stuck into port forwarding issue as it will waste lot of time and also normal calling is working on my current port forwarding. what i am currently trying to grab the channel name along with it's unique id and dial it directly like simple Dial(SIP/xyz ) dialing for example Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) ^ | | | but problem is that asterisk assigns random unique-id for every call. and also it is available only when dialing... what are my options? your help will be highly appreciated. regards, Naisr Javaid - Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fashion I have had this problem once and I never tried registering same extension from two different places after that. Try Phillip's suggestion, maybe it'll work for you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 11:42 AM, Philipp von Klitzing klitz...@xx wrote: Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works f... You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp --- Hi Zeeshan, I saw many of your posts on forum. i also put my problem on forum but did not get any satisfying answer. I wish if you could help me out. below is my post. == == Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/x...@xxx:5060 SIP/x...@:5678 i dial using following dial string Dial( SIP/x...@xxx:5060 SIP/x...@:5678,30,tTog) both destinations ring at the same time and one that is answered starts conversations. but audio is one sided as i mentioned above. But simply dialing single registration of XYZ like Dial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends. have any idea what is going wrong?? any help will be highly appreciated regards, Nasir Javaid == thanks in advance ... -- _ -- 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] One way audio when dialing multiple registrations
Hi Nasir, Please don't send me direct emails, unless you want to secure my paid consultancy services or want to do some other business. For setting up the RTP, you need to do it on your firewall, which is receiving RTP traffic from these particular IP address. I can't guess how to do it on your router/firewall. And it may still not solve your problem. I would suggest using separate extensions for separate IP addresses. For wireshark sniffing, my following blog might be helpful: http://ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/ Zeeshan -- www.ilovetovoip.com www.trashinternetexplorer.com On Fri, Jul 16, 2010 at 12:21 PM, Zeeshan Zakaria zisha...@gmail.comwrote: Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fashion I have had this problem once and I never tried registering same extension from two different places after that. Try Phillip's suggestion, maybe it'll work for you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 11:42 AM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works f... You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp -- _ -- 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 -- Zeeshan A Zakaria -- _ -- 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] One way audio when dialing multiple registrations
thanks a lot zishan and philipp, probably that is the problem that is occurring. I am gonna take some wireshark or etherial trace to further investigate the problem. i don't wanna stuck into port forwarding issue as it will waste lot of time and also normal calling is working on my current port forwarding. what i am currently trying to grab the channel name along with it's unique id and dial it directly like simple Dial(SIP/xyz ) dialing for example Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) ^ | | | but problem is that asterisk assigns random unique-id for every call. and also it is available only when dialing... what are my options? your help will be highly appreciated. regards, Naisr Javaid - Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fashion I have had this problem once and I never tried registering same extension from two different places after that. Try Phillip's suggestion, maybe it'll work for you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 11:42 AM, Philipp von Klitzing klitz...@xx wrote: Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works f... You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp --- Hi Zeeshan, I saw many of your posts on forum. i also put my problem on forum but did not get any satisfying answer. I wish if you could help me out. below is my post. == == Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/x...@119.68.0.90:5060 SIP/x...@202.16.34.10:5678 i dial using following dial string Dial( SIP/x...@119.68.0.90:5060 SIP/x...@202.16.34.10:5678,30,tTog) both destinations ring at the same time and one that is answered starts conversations. but audio is one sided as i mentioned above. But simply dialing single registration of XYZ like Dial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends. have any idea what is going wrong?? any help will be highly appreciated regards, Nasir Javaid == thanks in advance ... -- _ -- 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] One way audio when dialing multiple registrations
I am sure you can't achieve what you are trying to achieve here. Simply use two different extensions instead of one. Considering how SIP communication works, I believe SIP doesn't allow multiple registrations like this. Maybe somebody can correct me here if I am wrong. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-19 12:28 PM, Nasir Javaid nasirjavaidna...@gmail.com wrote: thanks a lot zishan and philipp, probably that is the problem that is occurring. I am gonna take some wireshark or etherial trace to further investigate the problem. i don't wanna stuck into port forwarding issue as it will waste lot of time and also normal calling is working on my current port forwarding. what i am currently trying to grab the channel name along with it's unique id and dial it directly like simple Dial(SIP/xyz ) dialing for example Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) ^ | | | but problem is that asterisk assigns random unique-id for every call. and also it is available only when dialing... what are my options? your help will be highly appreciated. regards, Naisr Javaid - Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fashion I have had this problem once and I never tried registering same extension from two different places after that. Try Phillip's suggestion, maybe it'll work for you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 11:42 AM, Philipp von Klitzing klitz...@xx wrote: Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works f... You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp --- Hi Zeeshan, I saw many of your posts on forum. i also put my problem on forum but did not get any satisfying answer. I wish if you could help me out. below is my post. == == Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/x...@119.68.0.90:5060 SIP/x...@202.16.34.10:5678 i dial using following dial string Dial( SIP/x...@119.68.0.90:5060 SIP/x...@202.16.34.10:5678,30,tTog) both destinations ring at the same time and one that is answered starts conversations. but audio is one sided as i mentioned above. But simply dialing single registration of XYZ like Dial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends. have any idea what is going wrong?? any help will be highly appreciated regards, Nasir Javaid == thanks in advance ... -- _ -- 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] One way audio when dialing multiple registrations
Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fashion I have had this problem once and I never tried registering same extension from two different places after that. Try Phillip's suggestion, maybe it'll work for you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 11:42 AM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works f... You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp -- _ -- 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] One way audio when dialing multiple registrations
Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/x...@192.168.0.20:5060 SIP/x...@192.168.0.10:5678 i dial using following dial string Dial(SIP/x...@192.168.0.20:5060SIP/x...@192.168.0.10:5678,30,tTog) both destinations ring at the same time and one that is answered starts conversations. but audio is one sided as i mentioned above. But simply dialing single registration of XYZ likeDial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends. have any idea what is going wrong?? any help will be highly appreciated regards, Nasir Javaid -- _ -- 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] One way audio when dialing multiple registrations
One-way audio is mostly firewall problem. Are you behind firewall ? You can check the audio-ports that are being used in the SDP-message by doing a /sip debug/. Maybe you do not have enough UDP-ports open for the audio ? Jonas. On 07/15/2010 04:38 PM, Nasir Javaid wrote: Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/x...@192.168.0.20:5060 http://x...@192.168.0.20:5060 SIP/x...@192.168.0.10:5678 http://x...@192.168.0.10:5678 i dial using following dial string Dial(SIP/x...@192.168.0.20:5060SIP/x...@192.168.0.10:5678 http://x...@192.168.0.10:5678,30,tTog) both destinations ring at the same time and one that is answered starts conversations. but audio is one sided as i mentioned above. But simply dialing single registration of XYZ like Dial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends. have any idea what is going wrong?? any help will be highly appreciated regards, Nasir Javaid -- _ -- 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] One way audio when dialing multiple registrations
Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp -- _ -- 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
