Re: [asterisk-users] One way audio on outgoing calls
Hi Carlos Le 07/08/2020 à 06:33, Carlos Chavez a écrit : I am having a strange problem with a new provider. We already have a couple SIP trunks working fine. We are trying a new provider but we are having one way audio problems with outgoing calls. Incoming calls do have two way audio, only outgoing calls have this problem. I do not see anything odd with a packet capture and using PJSIP history to check. The provider says that on outgoing calls the get random characters instead of the media port for RTP. We are using Asterisk 16.12.0 with PJSIP. The server is behind NAT so we have external_media_address and external_signaling_address set to the public IP and all relevant ports are forwarded to the Asterisk server. The other SIP trunks work fine, only this new provider has a problem and only for outgoing calls. An rtp set debug on shows only outgoing packets to the media address but no incoming packets. Why would there be a difference that makes it work on incoming calls but not on outgoing? We faced this problem and it was a firewall issue on our side. But if you say that your provider doesn't get the RTP, I understand that they can't return anything. RTP ports ? Cheers -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way audio on outgoing calls
I am having a strange problem with a new provider. We already have a couple SIP trunks working fine. We are trying a new provider but we are having one way audio problems with outgoing calls. Incoming calls do have two way audio, only outgoing calls have this problem. I do not see anything odd with a packet capture and using PJSIP history to check. The provider says that on outgoing calls the get random characters instead of the media port for RTP. We are using Asterisk 16.12.0 with PJSIP. The server is behind NAT so we have external_media_address and external_signaling_address set to the public IP and all relevant ports are forwarded to the Asterisk server. The other SIP trunks work fine, only this new provider has a problem and only for outgoing calls. An rtp set debug on shows only outgoing packets to the media address but no incoming packets. Why would there be a difference that makes it work on incoming calls but not on outgoing? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio on new build
On Mon, Feb 24, 2020 at 10:59 PM Ira wrote: > Hello Asterisk, > > I've been running a CENTOS 5 box with Asterisk 14 and am trying to > move to Asterisk 17.2 on a new Fedora Server 31 box. I built Asterisk > from Source as I've always done and copied all the configuration files > and other stuff from the old box. Everything comes up as expected and > it all seems to work except I have one way audio. I'm still using SIP, > not pjsip. As soon as I put the old box back the one way audio problem > is gone. Any suggestions where I should look? > Is a firewall running on the new system? Have you examined the SIP traffic to make sure the right IP addresses are present (sip set debug on)? Which direction is there no audio? -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way audio on new build
Hello Asterisk, I've been running a CENTOS 5 box with Asterisk 14 and am trying to move to Asterisk 17.2 on a new Fedora Server 31 box. I built Asterisk from Source as I've always done and copied all the configuration files and other stuff from the old box. Everything comes up as expected and it all seems to work except I have one way audio. I'm still using SIP, not pjsip. As soon as I put the old box back the one way audio problem is gone. Any suggestions where I should look? Thanks, Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio - doesn't seem to be NAT issue - SOLVED!
On Sat, 15 Aug 2015 12:42:38 -0300 Joshua Colp jc...@digium.com wrote: I am not sure why this hasn't bit anyone else. Perhaps most Asterisk systems are in one of two classes, connecting to all NAT phones or connecting to all public phones, and I am in a minority situation where I am talking to a mix of setups. Most people run without direct media unless they know the network topology will allow it 100%. Perhaps but the default is to run it. Perhaps the default should be no to prevent these problems. On the other hand, the documentation seemed to suggest that the default should have worked anyway. One leg was public, the other behind a NAT. It should recognize the latter and not try to put then in direct contact. It's almost like it saw the public one and didn't bother checking the other. Or, it checked both with an OR instead of an AND as I said. That seems more likely since it didn't matter who started the call. I don't really care at this point. If 1% of the calls go through the server when they didn't really need to it's no big deal. Cheers. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- 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 - doesn't seem to be NAT issue
Not 100% ure, but maybe play with the canreinvite or directmedia settings. On Wed, Aug 12, 2015 at 3:10 AM, D'Arcy J.M. Cain da...@vex.net wrote: I have been banging my head against the wall for weeks now on this one. I have a switch running NetBSD and Asterisk 11.19.0 although I have had this problem on older versions as well. I, and my users, can call out, we can receive calls, quality is excellent but I cannot talk with one user. The different elements are as follows: The switch as described above which is in a server room on the Internet backbone with a public IP address. My home system which is behind a bridged modem through a Linksys WRT54GS with priority given to my ATA. The ATA is a Cisco SPA112. I also have an actual SIP phone. The problem happens with both. Obviously I am using NAT but both devices work just fine if I am going to the PSTN. My user who is also going through a bridged modem to a Linksys SPA-2102 which is doing the PPPOE so it has a public IP address and no NAT involved although it serves NAT for the connected computer. So here is the problem. While both of us have no problems externally, when we call each other we get one way audio and it is always from me to him no matter who initiates the call. A further test, I can call from the SIP phone to the ATA connected phone and vice versa just fine. That involves two devices behind the same NAT but since they still need to use the server as an intermediary I can't see how that would matter. Given that both of us can make and accept calls and the server is simply connecting two separate channels I can't see where the problem might lie. Can anyone suggest a possible setup issue? I have tried so many things but I am willing to try them again. Feel free to make any suggestion no matter how silly. I really need to fix this. Cheers. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- 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 -- Michael Dupree jr. p: +1-248-935-4147 f: +1-866-671-6867 Skype: MichaelDupreeJr PGP Pub Key: http://www.michaeldupree.net/?page_id=53 This is a private message. This e-mail message, and any attachments thereto, is for the sole use of the intended recipient(s) and may contain legally privileged and/or confidential information. Any unauthorized review, use, disclosure or distribution is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and permanently delete all copies of the original message. --- -- _ -- 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 - doesn't seem to be NAT issue - SOLVED!
On Sat, 15 Aug 2015 16:30:39 +0800 Michael Dupree mich...@easybitllc.com wrote: Not 100% ure, but maybe play with the canreinvite or directmedia settings. Yes! That was it. Just for future searches here is what I did. I added directmedia = no in sip.conf. This fixed the issue. I believe that Asterisk was getting confused when one leg was inside NAT and the other was outside. Perhaps there was an OR where there should be an AND. It makes sense because the other user was the one outside NAT and he could hear me and I could not hear him no matter who initiated the call. He could make outside calls because both he and my provider were on public IPs. I am not sure why this hasn't bit anyone else. Perhaps most Asterisk systems are in one of two classes, connecting to all NAT phones or connecting to all public phones, and I am in a minority situation where I am talking to a mix of setups. Thanks for that. I was going nuts trying to figure this out. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- 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 - doesn't seem to be NAT issue - SOLVED!
On Sat, Aug 15, 2015, at 12:08 PM, D'Arcy J.M. Cain wrote: On Sat, 15 Aug 2015 16:30:39 +0800 Michael Dupree mich...@easybitllc.com wrote: Not 100% ure, but maybe play with the canreinvite or directmedia settings. Yes! That was it. Just for future searches here is what I did. I added directmedia = no in sip.conf. This fixed the issue. I believe that Asterisk was getting confused when one leg was inside NAT and the other was outside. Perhaps there was an OR where there should be an AND. It makes sense because the other user was the one outside NAT and he could hear me and I could not hear him no matter who initiated the call. He could make outside calls because both he and my provider were on public IPs. I am not sure why this hasn't bit anyone else. Perhaps most Asterisk systems are in one of two classes, connecting to all NAT phones or connecting to all public phones, and I am in a minority situation where I am talking to a mix of setups. Most people run without direct media unless they know the network topology will allow it 100%. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
Re: [asterisk-users] One way audio - doesn't seem to be NAT issue
Hi D'Arcy that the server IP for RTP as specified in the initial SIP is correct? Both the server and client are outside of NAT so I don't know what this might mean. They both have public IPs. This was a problem we had when the RTP server negotiated in SIP with our VOIP ITSP on one side of the connection, differed from the IP we were expecting on that side of the connection and was blocked in our firewall. Once we perused the SIP traffic we noted this and added the extra IP to the firewall for RTP traffic. We had slightly different parameters, e. g. that we would have no RTP at all, but a call that did connect to total silence, dialed from either side. Was NAT involved? Yes, NAT was being done at both ends, but it turned out that NATing was not the problem. Also check what RTP port ranges are being used - I have had this one-directional problem where the port range in /etc/asterisk/rtp.conf was too broad, and the firewall on my server was only allowing a smaller subset of RTP ports. rtpstart=1 rtpend=2 which is exactly what my packet filter allows through. I assume you have tried turning your packet filter or firewall off completely (just for a moment) to see if it helped? -- _ -- 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 - doesn't seem to be NAT issue
Hi D'arcy Have you checked your RTP port ranges (I'm sure you have), and also that the server IP for RTP as specified in the initial SIP is correct? Not sure how this will relate to your setup, but we had something similar here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP service provider in the middle. We had slightly different parameters, e. g. that we would have no RTP at all, but a call that did connect to total silence, dialed from either side. We subscribe to two trunk numbers provided by the VOIP service provider at each site in Asterisk. It turned out after carefully looking at the SIP flowing back and forth that the service provider was providing an RTP server IP that specified not the same IP as the SIP server (which is their standard practice) but a -different- RTP server IP. Due to the routing we have, neither system on either side of the SIP negotiated call could send packets to this new RTP server IP. We therefore added a route that specifically allowed that new RTP server IP to be reached by both machines on both sides of the VOIP service provider link. So can you carefully check that the SIP-negotiated RTP streams are going to IPs that are reachable in BOTH directions? Also check what RTP port ranges are being used - I have had this one-directional problem where the port range in /etc/asterisk/rtp.conf was too broad, and the firewall on my server was only allowing a smaller subset of RTP ports. E. g. /etc/asterisk/rtp.conf specified 1 - 5 as allowable RTP ports, but my firewalld firewall under Centos was only allowing 1 - 2 - so I'd regularly get that my SECOND call to test the server would have audio in one direction - because Asterisk was allocating an RTP port on one side of the SIP call that was outside the range my firewalld was allowing. It might require some careful tracing of SIP messages, maybe you can try this? Specifically try to determine what RTP port number is being negotiated when you have your zero-audio back from the remote party - what RTP port and RTP server IP is he using at that moment on his side? Is that port allowed through all the PPP / network segments between you? Is the IP / IPs between you used to transfer RTP reachable from his side? Message: 1 Date: Tue, 11 Aug 2015 15:10:44 -0400 From: D'Arcy J.M. Cain da...@vex.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] One way audio - doesn't seem to be NAT issue Message-ID: 20150811151044.79872ce9@imp Content-Type: text/plain; charset=US-ASCII Given that both of us can make and accept calls and the server is simply connecting two separate channels I can't see where the problem might lie. Can anyone suggest a possible setup issue? I have tried so many things but I am willing to try them again. Feel free to make any suggestion no matter how silly. I really need to fix this. Cheers. -- _ -- 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 - doesn't seem to be NAT issue
On Thu, 13 Aug 2015 10:41:31 +0200 Stefan Viljoen viljo...@verishare.co.za wrote: Have you checked your RTP port ranges (I'm sure you have), and also Yes. The ATA is using a range well within the range open on the server. that the server IP for RTP as specified in the initial SIP is correct? Both the server and client are outside of NAT so I don't know what this might mean. They both have public IPs. Not sure how this will relate to your setup, but we had something similar here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP service provider in the middle. This is an Asterisk server talking to an ATA. We had slightly different parameters, e. g. that we would have no RTP at all, but a call that did connect to total silence, dialed from either side. Was NAT involved? Also check what RTP port ranges are being used - I have had this one-directional problem where the port range in /etc/asterisk/rtp.conf was too broad, and the firewall on my server was only allowing a smaller subset of RTP ports. rtpstart=1 rtpend=2 which is exactly what my packet filter allows through. It might require some careful tracing of SIP messages, maybe you can try this? Specifically try to determine what RTP port number is being negotiated when you have your zero-audio back from the remote party - what RTP port and RTP server IP is he using at that moment on his side? I will check that. Thanks for your suggestions. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- 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 - doesn't seem to be NAT issue
On Tue, Aug 11, 2015, at 04:10 PM, D'Arcy J.M. Cain wrote: I have been banging my head against the wall for weeks now on this one. I have a switch running NetBSD and Asterisk 11.19.0 although I have had this problem on older versions as well. I, and my users, can call out, we can receive calls, quality is excellent but I cannot talk with one user. The different elements are as follows: snip I'd suggest getting a packet capture to see the RTP traffic to see the actual path of things, not just thinking of what it should be. Media doesn't just get lost. It's told to go somewhere ultimately and either that is incorrect for some reason or something is blocking it. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
[asterisk-users] One way audio - doesn't seem to be NAT issue
I have been banging my head against the wall for weeks now on this one. I have a switch running NetBSD and Asterisk 11.19.0 although I have had this problem on older versions as well. I, and my users, can call out, we can receive calls, quality is excellent but I cannot talk with one user. The different elements are as follows: The switch as described above which is in a server room on the Internet backbone with a public IP address. My home system which is behind a bridged modem through a Linksys WRT54GS with priority given to my ATA. The ATA is a Cisco SPA112. I also have an actual SIP phone. The problem happens with both. Obviously I am using NAT but both devices work just fine if I am going to the PSTN. My user who is also going through a bridged modem to a Linksys SPA-2102 which is doing the PPPOE so it has a public IP address and no NAT involved although it serves NAT for the connected computer. So here is the problem. While both of us have no problems externally, when we call each other we get one way audio and it is always from me to him no matter who initiates the call. A further test, I can call from the SIP phone to the ATA connected phone and vice versa just fine. That involves two devices behind the same NAT but since they still need to use the server as an intermediary I can't see how that would matter. Given that both of us can make and accept calls and the server is simply connecting two separate channels I can't see where the problem might lie. Can anyone suggest a possible setup issue? I have tried so many things but I am willing to try them again. Feel free to make any suggestion no matter how silly. I really need to fix this. Cheers. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- 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 internal
Hi All We have a strange issue with our hosted asterisk server running on Debian Internal calls btween extensions using g729 give one way audio As soon as we change the codec to ALAW the issues goes away. Any ideas how to fix this? Outbound calls via a trunk work fine with g729 Kind Regards Andrew Colin Converged Data (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Direct: +27 (0)10 591 4607 Mobile: +27 (0)82 310 3007 Switchboard: +27 (0)10 591 4600 Email: mailto:and...@convergedgroup.net and...@convergedgroup.net Web: http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. -- _ -- 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 internal
On Friday 21 Nov 2014, Andrew Colin wrote: Hi All We have a strange issue with our hosted asterisk server running on Debian Internal calls btween extensions using g729 give one way audio As soon as we change the codec to ALAW the issues goes away. Any ideas how to fix this? Outbound calls via a trunk work fine with g729 Unless you have serious bandwidth issues, just forget about g.729 and change to a-law throughout. A-law is what the PSTN (in civilised countries) uses anyway, so you won't need to transcode (which chews up processor resources and risks compromising quality) for calls to and from the outside world. If you really need to use g.729 and are outside the USA (therefore, beyond the reach of software patents), there is a free version that you can use -- and this one, better than Digium's offering, comes with the Source Code so you can be sure it isn't doing anything nasty behind the scenes. But to be honest, you probably are better off just sticking with a-law. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 internal
You probably do not have enough g729 channels license. On 21-Nov-2014 4:17 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Friday 21 Nov 2014, Andrew Colin wrote: Hi All We have a strange issue with our hosted asterisk server running on Debian Internal calls btween extensions using g729 give one way audio As soon as we change the codec to ALAW the issues goes away. Any ideas how to fix this? Outbound calls via a trunk work fine with g729 Unless you have serious bandwidth issues, just forget about g.729 and change to a-law throughout. A-law is what the PSTN (in civilised countries) uses anyway, so you won't need to transcode (which chews up processor resources and risks compromising quality) for calls to and from the outside world. If you really need to use g.729 and are outside the USA (therefore, beyond the reach of software patents), there is a free version that you can use -- and this one, better than Digium's offering, comes with the Source Code so you can be sure it isn't doing anything nasty behind the scenes. But to be honest, you probably are better off just sticking with a-law. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 internal
I am using the free g729 Kind Regards Andrew Colin Converged Data (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Direct: +27 (0)10 591 4607 Mobile: +27 (0)82 310 3007 Switchboard: +27 (0)10 591 4600 Email: and...@convergedgroup.net Web: http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. From: Mitul Limbani [mailto:mi...@enterux.in] Sent: Friday, November 21, 2014 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Andrew Colin Subject: Re: [asterisk-users] One way audio internal You probably do not have enough g729 channels license. On 21-Nov-2014 4:17 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Friday 21 Nov 2014, Andrew Colin wrote: Hi All We have a strange issue with our hosted asterisk server running on Debian Internal calls btween extensions using g729 give one way audio As soon as we change the codec to ALAW the issues goes away. Any ideas how to fix this? Outbound calls via a trunk work fine with g729 Unless you have serious bandwidth issues, just forget about g.729 and change to a-law throughout. A-law is what the PSTN (in civilised countries) uses anyway, so you won't need to transcode (which chews up processor resources and risks compromising quality) for calls to and from the outside world. If you really need to use g.729 and are outside the USA (therefore, beyond the reach of software patents), there is a free version that you can use -- and this one, better than Digium's offering, comes with the Source Code so you can be sure it isn't doing anything nasty behind the scenes. But to be honest, you probably are better off just sticking with a-law. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 internal
Then something to do with your codec selection priority. On 21-Nov-2014 4:26 PM, Andrew Colin and...@convergedgroup.net wrote: I am using the free g729 Kind Regards Andrew Colin *Converged Data (Pty) Ltd.* *Licensed Telecoms Operator :* (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Direct: +27 (0)10 591 4607 Mobile: +27 (0)82 310 3007 Switchboard: +27 (0)10 591 4600 Email: and...@convergedgroup.net Web: http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. *From:* Mitul Limbani [mailto:mi...@enterux.in] *Sent:* Friday, November 21, 2014 12:51 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* Andrew Colin *Subject:* Re: [asterisk-users] One way audio internal You probably do not have enough g729 channels license. On 21-Nov-2014 4:17 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Friday 21 Nov 2014, Andrew Colin wrote: Hi All We have a strange issue with our hosted asterisk server running on Debian Internal calls btween extensions using g729 give one way audio As soon as we change the codec to ALAW the issues goes away. Any ideas how to fix this? Outbound calls via a trunk work fine with g729 Unless you have serious bandwidth issues, just forget about g.729 and change to a-law throughout. A-law is what the PSTN (in civilised countries) uses anyway, so you won't need to transcode (which chews up processor resources and risks compromising quality) for calls to and from the outside world. If you really need to use g.729 and are outside the USA (therefore, beyond the reach of software patents), there is a free version that you can use -- and this one, better than Digium's offering, comes with the Source Code so you can be sure it isn't doing anything nasty behind the scenes. But to be honest, you probably are better off just sticking with a-law. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 internal
I currently am running on a Intel(R) Xeon(R) CPU E5-2670 v2 @ 2.50GHz Codec im using is codec_g729-ast18-icc-glibc-x86_64-core2.so -- _ -- 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 internal
On Friday 21 Nov 2014, Andrew Colin wrote: I am using the free g729 OK, so there shouldn't be any licencing problems (unless for some reason your Asterisk is wanting to use the paid-for g.729 aot the Free one. Look at the CLI output very, very carefully to see if this might be happening). Did it ever work properly? If your kernel, C library or some other fundamental system component has been updated since you installed g.729, then it might have been broken by the upgrade. Navigating to the folder with the Source Code and re-running `make` followed by `make install` ought to fix it. But why are you using g.729 anyway? What special reason have you for doing it differently than the rest of the world? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 internal
I currently am running on a Intel(R) Xeon(R) CPU E5-2670 v2 @ 2.50GHz Codec im using is codec_g729-ast18-icc-glibc-x86_64-core2.so Kind Regards Andrew Colin Converged Data (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Direct: +27 (0)10 591 4607 Mobile: +27 (0)82 310 3007 Switchboard: +27 (0)10 591 4600 Email: and...@convergedgroup.net Web: http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. From: Mitul Limbani [mailto:mi...@enterux.in] Sent: Friday, November 21, 2014 1:04 PM To: Andrew Colin Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] One way audio internal Then something to do with your codec selection priority. On 21-Nov-2014 4:26 PM, Andrew Colin and...@convergedgroup.net wrote: I am using the free g729 Kind Regards Andrew Colin Converged Data (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Direct: +27 (0)10 591 4607 tel:%2B27%20%280%2910%20591%204607 Mobile: +27 (0)82 310 3007 tel:%2B27%20%280%2982%20310%203007 Switchboard: +27 (0)10 591 4600 tel:%2B27%20%280%2910%20591%204600 Email: and...@convergedgroup.net Web: http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. From: Mitul Limbani [mailto:mi...@enterux.in] Sent: Friday, November 21, 2014 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Andrew Colin Subject: Re: [asterisk-users] One way audio internal You probably do not have enough g729 channels license. On 21-Nov-2014 4:17 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Friday 21 Nov 2014, Andrew Colin wrote: Hi All We have a strange issue with our hosted asterisk server running on Debian Internal calls btween extensions using g729 give one way audio As soon as we change the codec to ALAW the issues goes away. Any ideas how to fix this? Outbound calls via a trunk work fine with g729 Unless you have serious bandwidth issues, just forget about g.729 and change to a-law throughout. A-law is what the PSTN (in civilised countries) uses anyway, so you won't need to transcode (which chews up processor resources and risks compromising quality) for calls to and from the outside world. If you really need to use g.729 and are outside the USA (therefore, beyond the reach of software patents), there is a free version that you can use -- and this one, better than Digium's offering, comes with the Source Code so you can be sure it isn't doing anything nasty behind the scenes. But to be honest, you probably are better off just sticking with a-law. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 with WebRTC (with external asterisk)
Hi Amit, My rtp.conf has the stunaddr listed and icesupport set to yes. It looks like the issue is that the media isn't being sent from 192.168.3.150 to 192.168.3.131 (chrome browser to asteriskrtc.local). When using asteriskrtc.local to originate the call (make a call directly from sipml client to another number on asteriskrtc.local or to a number on another asterisk server) audio flows both ways with no issue, it's just when asteriskgary.local is originating the call that there is no audio flowing from chrome to asteriskrtc.local. I should probably rephrase the above though to say that on tshark I can actually see the packets flowing (tshark host 192.168.3.150): 2.384874 192.168.3.150 - 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik 2.384925 192.168.3.150 - 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik 2.385060 192.168.3.131 - 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:60175 2.385256 192.168.3.131 - 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:65021 2.394891 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 2.415195 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 2.434063 192.168.3.150 - 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik 2.434121 192.168.3.150 - 192.168.3.131 STUN 174 Binding Request user: 15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik 2.434296 192.168.3.131 - 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:60175 2.434462 192.168.3.131 - 192.168.3.150 STUN 122 Binding Success Response XOR-MAPPED-ADDRESS: 192.168.3.150:65021 2.435083 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 2.455310 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 2.475009 192.168.3.131 - 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 Thanks again for your time! Kind Regards, Gary Shergill - Original Message - From: Amit Patkar a...@avhan.com To: asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2014 4:55:57 PM Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk) Please check rtp.conf Look for stunaddr setting. You can try with google STUN server stunaddr = stun.l.google.com:19302 Thanks Regards, Amit Patkar On 5/21/2014 9:13 PM, Gary Shergill wrote: Hi again, Just noticed this is being sent to the wrong thread... first time using a mailing list and I just replied to the mail sent by the mailing list for Amit's reply. Hope this time it works... Anyway, I have audio from 1000 to 6901 working, that was a mistake on my side (I tested using the SIPml demo site and it worked, then realised I was missing a setting). However, the issue still remains where 1000 can not always hear 6901. As mentioned before, this works only SOMETIMES, and when it does work asteriskgary.local sees RTP packets coming FROM 192.168.3.131 (asteriskrtc.local). Unsure what would be causing this, because it does work sometimes and doesn't at others, with no obvious reason either way. Thanks again. Kind Regards, Gary Shergill - Original Message - From: Gary Shergill gsherg...@gltd.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2014 3:36:54 PM Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk) Hi Amit, ICE/STUN is configured correctly. The extension for the webrtc user is defined in sip.conf on the asteriskrtc.local server. The other user is defined in Elastix. I have directmedia=no set for the user on asteriskrtc.local. My exact setup/scenario is below: - asteriskgary.local has a route to dial extensions on my Elastix server. - asteriskgary.local has a route to dial extensions on asteriskrtc.local server. - The call is being originated from asteriskgary.local. The first party is an extension on asteriskgary.local, the destination party is an extension on my Elastix server. What's happening is as follows (this is a reverse of the previous case as 6901 is now dialling 1000): - User on asteriskgary.local places a call to 1000, his number is 6901 - 6901 answers on the web browser and begins to dial 1000 - 1000 answers and the call is established correctly - SOMETIMES 1000 can hear 6901. Other times he can not (seems to be random...) - 6901 can NEVER hear 1000 key: 192.168.3.127 - asteriskgary.local 192.168.3.131 - asteriskrtc.local 192.168.3.150 - machine running chrome browser where 6901 is logged on 192.168.3.100 - phone where 1000 is logged on (1000 can hear 6901) RTP TRACE ON asteriskrtc.local Got RTP packet from192.168.3.150:55148 (type 00, seq 014308, ts 2304496631, len 000160
[asterisk-users] One Way Audio with WebRTC (with external asterisk)
Hi, I've run into a slight issue when using WebRTC and two Asterisk boxes. I am using SIPml as the test WebRTC client. My two asterisk boxes, one of them is configured for WebRTC with websockets, etc (asteriskrtc.local) and the other is just a standard asterisk server (asteriskgary.local). Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to log in to the SIPml webpage and make a call from a SIP Phone to that WebRTC user, and vice versa, and all the media flows. When I try making a call from the other asterisk server (asteriskgary.local) to asteriskrtc.local (all routes are set up) I am seeing the following behaviour: - asteriskgary.local user, 1000, dials asteriskrtc.local number, 6901 - 6901 sees the call and has the option to answer - 6901 answers the call - 6901 can hear 1000 talking - 1000 can not hear 6901 The weird thing is, sometimes it works, sometimes it doesn't... I think it has something to do with the port destination changing when the call is answered but I'm not sure (wireshark suggests that, as it says Port Unreachable). Has anyone tried this before and seen this issue? Or knows why it is and how to debug it? I can provide any logs required, I have some logs from when it works and doesn't. Thank you for your help. Kind Regards, Gary Shergill -- _ -- 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 with WebRTC (with external asterisk)
Hi Gary You need to check if ICE / STUN is configured. How are these extensions configured? If you are in private network, you might have to disable DirectMedia / reInvite for calls going between 2 asterisk boxes. I hope this helps to resolve your issue. *Thanks Regards,* Amit Patkar On 5/21/2014 2:26 PM, Gary Shergill wrote: Hi, I've run into a slight issue when using WebRTC and two Asterisk boxes. I am using SIPml as the test WebRTC client. My two asterisk boxes, one of them is configured for WebRTC with websockets, etc (asteriskrtc.local) and the other is just a standard asterisk server (asteriskgary.local). Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to log in to the SIPml webpage and make a call from a SIP Phone to that WebRTC user, and vice versa, and all the media flows. When I try making a call from the other asterisk server (asteriskgary.local) to asteriskrtc.local (all routes are set up) I am seeing the following behaviour: - asteriskgary.local user, 1000, dials asteriskrtc.local number, 6901 - 6901 sees the call and has the option to answer - 6901 answers the call - 6901 can hear 1000 talking - 1000 can not hear 6901 The weird thing is, sometimes it works, sometimes it doesn't... I think it has something to do with the port destination changing when the call is answered but I'm not sure (wireshark suggests that, as it says Port Unreachable). Has anyone tried this before and seen this issue? Or knows why it is and how to debug it? I can provide any logs required, I have some logs from when it works and doesn't. Thank you for your help. Kind Regards, Gary Shergill -- _ -- 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 with WebRTC (with external asterisk)
=webrtc hasiax = no hassip = yes encryption = yes avpf = yes icesupport = yes videosupport=no directmedia=no canreinvite=no You can see from the trace packets that sometimes asteriskgary.local sees no packets from asteriskrtc.local, and at the same time the packets on asteriskrtc.local show half the number of records (there is no Probation passed - setting RTP source address to 192.168.3.127:15942 which causes twice the number of packets, no idea if this is relevant though). Please ask if you need anything else. I'm totally stumped with this issue... Note that on asteriskgary.local ICE is not configured, I wouldn't have though it would need it as it isn't talking with the webrtc client itself, it is just talking to an Asterisk server (and that asterisk server is the one which talks to the webrtc client). Thank you. Kind Regards, Gary Shergill - Original Message - From: Amit Patkar a...@avhan.com To: asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2014 04:41:50 AM Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk) Hi Gary You need to check if ICE / STUN is configured. How are these extensions configured? If you are in private network, you might have to disable DirectMedia / reInvite for calls going between 2 asterisk boxes. I hope this helps to resolve your issue. *Thanks Regards,* Amit Patkar On 5/21/2014 2:26 PM, Gary Shergill wrote: Hi, I've run into a slight issue when using WebRTC and two Asterisk boxes. I am using SIPml as the test WebRTC client. My two asterisk boxes, one of them is configured for WebRTC with websockets, etc (asteriskrtc.local) and the other is just a standard asterisk server (asteriskgary.local). Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to log in to the SIPml webpage and make a call from a SIP Phone to that WebRTC user, and vice versa, and all the media flows. When I try making a call from the other asterisk server (asteriskgary.local) to asteriskrtc.local (all routes are set up) I am seeing the following behaviour: - asteriskgary.local user, 1000, dials asteriskrtc.local number, 6901 - 6901 sees the call and has the option to answer - 6901 answers the call - 6901 can hear 1000 talking - 1000 can not hear 6901 The weird thing is, sometimes it works, sometimes it doesn't... I think it has something to do with the port destination changing when the call is answered but I'm not sure (wireshark suggests that, as it says Port Unreachable). Has anyone tried this before and seen this issue? Or knows why it is and how to debug it? I can provide any logs required, I have some logs from when it works and doesn't. Thank you for your help. Kind Regards, Gary Shergill -- _ -- 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 with WebRTC (with external asterisk)
Hi again, Just noticed this is being sent to the wrong thread... first time using a mailing list and I just replied to the mail sent by the mailing list for Amit's reply. Hope this time it works... Anyway, I have audio from 1000 to 6901 working, that was a mistake on my side (I tested using the SIPml demo site and it worked, then realised I was missing a setting). However, the issue still remains where 1000 can not always hear 6901. As mentioned before, this works only SOMETIMES, and when it does work asteriskgary.local sees RTP packets coming FROM 192.168.3.131 (asteriskrtc.local). Unsure what would be causing this, because it does work sometimes and doesn't at others, with no obvious reason either way. Thanks again. Kind Regards, Gary Shergill - Original Message - From: Gary Shergill gsherg...@gltd.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2014 3:36:54 PM Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk) Hi Amit, ICE/STUN is configured correctly. The extension for the webrtc user is defined in sip.conf on the asteriskrtc.local server. The other user is defined in Elastix. I have directmedia=no set for the user on asteriskrtc.local. My exact setup/scenario is below: - asteriskgary.local has a route to dial extensions on my Elastix server. - asteriskgary.local has a route to dial extensions on asteriskrtc.local server. - The call is being originated from asteriskgary.local. The first party is an extension on asteriskgary.local, the destination party is an extension on my Elastix server. What's happening is as follows (this is a reverse of the previous case as 6901 is now dialling 1000): - User on asteriskgary.local places a call to 1000, his number is 6901 - 6901 answers on the web browser and begins to dial 1000 - 1000 answers and the call is established correctly - SOMETIMES 1000 can hear 6901. Other times he can not (seems to be random...) - 6901 can NEVER hear 1000 key: 192.168.3.127 - asteriskgary.local 192.168.3.131 - asteriskrtc.local 192.168.3.150 - machine running chrome browser where 6901 is logged on 192.168.3.100 - phone where 1000 is logged on (1000 can hear 6901) RTP TRACE ON asteriskrtc.local Got RTP packet from192.168.3.150:55148 (type 00, seq 014308, ts 2304496631, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054709, ts 2304496624, len 000160) 0x7fe73c021740 -- Probation passed - setting RTP source address to 192.168.3.127:15942 Got RTP packet from192.168.3.127:15942 (type 00, seq 003763, ts 000160, len 000160) Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047008, ts 000160, len 4294967284) Got RTP packet from192.168.3.150:55148 (type 00, seq 014309, ts 2304496791, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054710, ts 2304496784, len 000160) Got RTP packet from192.168.3.150:55148 (type 00, seq 014310, ts 2304496951, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054711, ts 2304496944, len 000160) Got RTP packet from192.168.3.127:15942 (type 00, seq 003764, ts 000320, len 000160) Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047009, ts 000320, len 4294967284) Got RTP packet from192.168.3.150:55148 (type 00, seq 014311, ts 2304497111, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054712, ts 2304497104, len 000160) Got RTP packet from192.168.3.127:15942 (type 00, seq 003765, ts 000480, len 000160) Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047010, ts 000480, len 4294967284) (1000 can hear 6901) RTP TRACE ON asteriskgary.local ... Got RTP packet from192.168.3.131:17836 (type 00, seq 055375, ts 2304603184, len 000160) Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004428, ts 106560, len 000160) Got RTP packet from192.168.3.131:17836 (type 00, seq 055376, ts 2304603344, len 000160) Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004429, ts 106720, len 000160) Got RTP packet from192.168.3.131:17836 (type 00, seq 055377, ts 2304603504, len 000160) Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004430, ts 106880, len 000160) Got RTP packet from192.168.3.131:17836 (type 00, seq 055378, ts 2304603664, len 000160) Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004431, ts 107040, len 000160) ... (no audio) RTP TRACE ON asteriskrtc.local Got RTP packet from192.168.3.127:17796 (type 00, seq 035016, ts 000640, len 000160) Sent RTP packet to 192.168.3.150:53684 (via ICE) (type 00, seq 060981, ts 000640, len 4294967284) Got RTP packet from192.168.3.127:17796 (type 00, seq 035017, ts 000800, len 000160) Sent RTP packet to 192.168.3.150:53684 (via ICE) (type 00, seq 060982, ts 000800, len 4294967284) Got RTP packet from
Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)
Please check rtp.conf Look for stunaddr setting. You can try with google STUN server stunaddr = stun.l.google.com:19302 *Thanks Regards,* Amit Patkar On 5/21/2014 9:13 PM, Gary Shergill wrote: Hi again, Just noticed this is being sent to the wrong thread... first time using a mailing list and I just replied to the mail sent by the mailing list for Amit's reply. Hope this time it works... Anyway, I have audio from 1000 to 6901 working, that was a mistake on my side (I tested using the SIPml demo site and it worked, then realised I was missing a setting). However, the issue still remains where 1000 can not always hear 6901. As mentioned before, this works only SOMETIMES, and when it does work asteriskgary.local sees RTP packets coming FROM 192.168.3.131 (asteriskrtc.local). Unsure what would be causing this, because it does work sometimes and doesn't at others, with no obvious reason either way. Thanks again. Kind Regards, Gary Shergill - Original Message - From: Gary Shergill gsherg...@gltd.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2014 3:36:54 PM Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk) Hi Amit, ICE/STUN is configured correctly. The extension for the webrtc user is defined in sip.conf on the asteriskrtc.local server. The other user is defined in Elastix. I have directmedia=no set for the user on asteriskrtc.local. My exact setup/scenario is below: - asteriskgary.local has a route to dial extensions on my Elastix server. - asteriskgary.local has a route to dial extensions on asteriskrtc.local server. - The call is being originated from asteriskgary.local. The first party is an extension on asteriskgary.local, the destination party is an extension on my Elastix server. What's happening is as follows (this is a reverse of the previous case as 6901 is now dialling 1000): - User on asteriskgary.local places a call to 1000, his number is 6901 - 6901 answers on the web browser and begins to dial 1000 - 1000 answers and the call is established correctly - SOMETIMES 1000 can hear 6901. Other times he can not (seems to be random...) - 6901 can NEVER hear 1000 key: 192.168.3.127 - asteriskgary.local 192.168.3.131 - asteriskrtc.local 192.168.3.150 - machine running chrome browser where 6901 is logged on 192.168.3.100 - phone where 1000 is logged on (1000 can hear 6901) RTP TRACE ON asteriskrtc.local Got RTP packet from192.168.3.150:55148 (type 00, seq 014308, ts 2304496631, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054709, ts 2304496624, len 000160) 0x7fe73c021740 -- Probation passed - setting RTP source address to 192.168.3.127:15942 Got RTP packet from192.168.3.127:15942 (type 00, seq 003763, ts 000160, len 000160) Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047008, ts 000160, len 4294967284) Got RTP packet from192.168.3.150:55148 (type 00, seq 014309, ts 2304496791, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054710, ts 2304496784, len 000160) Got RTP packet from192.168.3.150:55148 (type 00, seq 014310, ts 2304496951, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054711, ts 2304496944, len 000160) Got RTP packet from192.168.3.127:15942 (type 00, seq 003764, ts 000320, len 000160) Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047009, ts 000320, len 4294967284) Got RTP packet from192.168.3.150:55148 (type 00, seq 014311, ts 2304497111, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054712, ts 2304497104, len 000160) Got RTP packet from192.168.3.127:15942 (type 00, seq 003765, ts 000480, len 000160) Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047010, ts 000480, len 4294967284) (1000 can hear 6901) RTP TRACE ON asteriskgary.local ... Got RTP packet from192.168.3.131:17836 (type 00, seq 055375, ts 2304603184, len 000160) Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004428, ts 106560, len 000160) Got RTP packet from192.168.3.131:17836 (type 00, seq 055376, ts 2304603344, len 000160) Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004429, ts 106720, len 000160) Got RTP packet from192.168.3.131:17836 (type 00, seq 055377, ts 2304603504, len 000160) Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004430, ts 106880, len 000160) Got RTP packet from192.168.3.131:17836 (type 00, seq 055378, ts 2304603664, len 000160) Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004431, ts 107040, len 000160) ... (no audio) RTP TRACE ON asteriskrtc.local Got RTP packet from192.168.3.127:17796 (type 00, seq 035016, ts 000640, len 000160) Sent RTP packet to 192.168.3.150:53684 (via ICE) (type 00, seq 060981, ts 000640, len 4294967284) Got RTP packet from
Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)
Hi, I am also trying to integrate sipml5 demo.For that i made some configuration. Call works fine using chrome browser but facing One way audio issue. And firefox browser not able to originate call. Here is the my configuration: http://pastebin.com/EtVzK2T2 let me know if i miss something. On Wed, May 21, 2014 at 9:25 PM, Amit Patkar a...@avhan.com wrote: Please check rtp.conf Look for stunaddr setting. You can try with google STUN server stunaddr = stun.l.google.com:19302 *Thanks Regards,* Amit Patkar On 5/21/2014 9:13 PM, Gary Shergill wrote: Hi again, Just noticed this is being sent to the wrong thread... first time using a mailing list and I just replied to the mail sent by the mailing list for Amit's reply. Hope this time it works... Anyway, I have audio from 1000 to 6901 working, that was a mistake on my side (I tested using the SIPml demo site and it worked, then realised I was missing a setting). However, the issue still remains where 1000 can not always hear 6901. As mentioned before, this works only SOMETIMES, and when it does work asteriskgary.local sees RTP packets coming FROM 192.168.3.131 (asteriskrtc.local). Unsure what would be causing this, because it does work sometimes and doesn't at others, with no obvious reason either way. Thanks again. Kind Regards, Gary Shergill - Original Message - From: Gary Shergill gsherg...@gltd.net gsherg...@gltd.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2014 3:36:54 PM Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk) Hi Amit, ICE/STUN is configured correctly. The extension for the webrtc user is defined in sip.conf on the asteriskrtc.local server. The other user is defined in Elastix. I have directmedia=no set for the user on asteriskrtc.local. My exact setup/scenario is below: - asteriskgary.local has a route to dial extensions on my Elastix server. - asteriskgary.local has a route to dial extensions on asteriskrtc.local server. - The call is being originated from asteriskgary.local. The first party is an extension on asteriskgary.local, the destination party is an extension on my Elastix server. What's happening is as follows (this is a reverse of the previous case as 6901 is now dialling 1000): - User on asteriskgary.local places a call to 1000, his number is 6901 - 6901 answers on the web browser and begins to dial 1000 - 1000 answers and the call is established correctly - SOMETIMES 1000 can hear 6901. Other times he can not (seems to be random...) - 6901 can NEVER hear 1000 key: 192.168.3.127 - asteriskgary.local 192.168.3.131 - asteriskrtc.local 192.168.3.150 - machine running chrome browser where 6901 is logged on 192.168.3.100 - phone where 1000 is logged on (1000 can hear 6901) RTP TRACE ON asteriskrtc.local Got RTP packet from192.168.3.150:55148 (type 00, seq 014308, ts 2304496631, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054709, ts 2304496624, len 000160) 0x7fe73c021740 -- Probation passed - setting RTP source address to 192.168.3.127:15942 Got RTP packet from192.168.3.127:15942 (type 00, seq 003763, ts 000160, len 000160) Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047008, ts 000160, len 4294967284) Got RTP packet from192.168.3.150:55148 (type 00, seq 014309, ts 2304496791, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054710, ts 2304496784, len 000160) Got RTP packet from192.168.3.150:55148 (type 00, seq 014310, ts 2304496951, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054711, ts 2304496944, len 000160) Got RTP packet from192.168.3.127:15942 (type 00, seq 003764, ts 000320, len 000160) Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047009, ts 000320, len 4294967284) Got RTP packet from192.168.3.150:55148 (type 00, seq 014311, ts 2304497111, len 000160) Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054712, ts 2304497104, len 000160) Got RTP packet from192.168.3.127:15942 (type 00, seq 003765, ts 000480, len 000160) Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047010, ts 000480, len 4294967284) (1000 can hear 6901) RTP TRACE ON asteriskgary.local ... Got RTP packet from192.168.3.131:17836 (type 00, seq 055375, ts 2304603184, len 000160) Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004428, ts 106560, len 000160) Got RTP packet from192.168.3.131:17836 (type 00, seq 055376, ts 2304603344, len 000160) Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004429, ts 106720, len 000160) Got RTP packet from192.168.3.131:17836 (type 00, seq 055377, ts 2304603504, len 000160) Sent RTP packet
[asterisk-users] One-way audio with media_address
I'm migrating from Asterisk 1.6.2 to 10.7.0. In 1.6.2, I made a small patch to allow specifying an address for RTP media. That worked. In 10.7.0, this appears to be built in with media_address, but it doesn't work for me. My Asterisk server has multiple addresses, all global address on two different /24's with different routing policies via BGP. I'm connecting to a phone that's over NAT. I have nat=yes in the general section of sip.conf. Everything works fine with the default. But if I specify media_address to be the Asterisk server's address on the other /24, I get one-way audio. I can see with sip debug that the proper address is being given in the SDP data. Audio from the phone is fine. Audio *to* the phone starts out with maybe 1-2 seconds of very garbled audio, then goes quiet. Running traceroute shows that data comes from the phone *to* Asterisk on the desired /24, but goes out with a source address from the other /24 (the default address). I'm not sure if this is the problem or not, but in any event, I think the source address for RTP should be the one in media_address and want it that way for my purposes anyway. Is there a way to configure this to happen? If not, where should I look to make a patch? And is this likely the reason for the one-way audio or is something else the likely cause? -- _ -- 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 calling multiple SIP
Hi, On one of our locations, I am having issues with one-way audio when I call several phones with SIP/Phone_ASIP/Phone_BSIP/Phone_C. When I call the phones individually, they work fine, so it's not a volume setting on the phone. Also this setup has worked at other locations. Any idea's what to look for? Thanks in advance. Kind regards, Roland. -- _ -- 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 using originate...
Dear in normal mode, .call files make a call between the system and who you named remote person, I don't know where are you? in natmode=yes, set qualify=yes. check the negotiated codecs also. Best On Sat, Aug 13, 2011 at 1:29 AM, Carlos Chavez cur...@telecomabmex.comwrote: We are having a problem when trying to use originate or AMI to make a call. We have an Asterisk 1.8.5.0 server which uses a SIP provider to call the PSTN. When dialing from IP phones everything works fine. When you try making the call with originate, AMI or a call file then the remote person can hear you but you cannot hear them. Why would it behave differently when dialing from a phone? The server is behind NAT and uses externaddr to set the external IP (static). Anyone had any experience with this? Here is my (edited) sip.conf entry: [libre-8793] defaultuser=123456789 secret=X fromuser=123456789 trustrpid=yes sendrpid=yes type=peer fromdomain=i2next.com.mx host=i2next.com.mx nat=yes qualify=no insecure=port,invite directmedia=no disallow=all allow=g729 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 -- Pezhman Lali -- _ -- 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 using originate...
We are having a problem when trying to use originate or AMI to make a call. We have an Asterisk 1.8.5.0 server which uses a SIP provider to call the PSTN. When dialing from IP phones everything works fine. When you try making the call with originate, AMI or a call file then the remote person can hear you but you cannot hear them. Why would it behave differently when dialing from a phone? The server is behind NAT and uses externaddr to set the external IP (static). Anyone had any experience with this? Here is my (edited) sip.conf entry: [libre-8793] defaultuser=123456789 secret=X fromuser=123456789 trustrpid=yes sendrpid=yes type=peer fromdomain=i2next.com.mx host=i2next.com.mx nat=yes qualify=no insecure=port,invite directmedia=no disallow=all allow=g729 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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
I've been having a similar (well exactly the same) problem this last week and have been bashing my head trying to fix it. Just one question, are you using RealTime? Ish On Wed, 2011-03-09 at 17:40 -0500, Tim King wrote: I am having trouble with no return audio on inbound calls. I have been working on this for 18 hours and even built a fresh server and moved everything over and I am getting the same results. I need someone that can help get this resolved tonight. I can provide access to all machines involved. Please email me at tim.compnetw...@gmail.com if you can help. -- _ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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
Just fixed our problem with directmedia=no but this only applies if your extensions are behind a nat Ish On Thu, 2011-03-10 at 09:40 +, Ishfaq Malik wrote: I've been having a similar (well exactly the same) problem this last week and have been bashing my head trying to fix it. Just one question, are you using RealTime? Ish On Wed, 2011-03-09 at 17:40 -0500, Tim King wrote: I am having trouble with no return audio on inbound calls. I have been working on this for 18 hours and even built a fresh server and moved everything over and I am getting the same results. I need someone that can help get this resolved tonight. I can provide access to all machines involved. Please email me at tim.compnetw...@gmail.com if you can help. -- _ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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
On 10 March 2011 11:17, Ishfaq Malik i...@pack-net.co.uk wrote: Just fixed our problem with directmedia=no but this only applies if your extensions are behind a nat Ish There are several reasons why directmedia=no might be the correct configuration. 1) NAT - probably the most common reason 2) Routing - Sometimes devices cannot route to each other directly 3) ITSP calls. Many SIP providers will not accept a redirect and I am sure there are many more... Cheers, Steve -- _ -- 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
My message with the configuration attached is awaiting moderator approval. I
will try to paste relevant data here.
*sip.conf*
[general]
context=inbound ;
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
dtmfmode = rfc2833
directmedia=no
[castlewire]
type=user
host=74.204.4.206
context=outb2
dtmfmode=rfc2833
username=castlewire
secret=1234
quallify=yes
canreinvite=no
[equity]
type=friend
host=dynamic
context=outb2
dtmfmode=rfc2833
username=equity
secret=1234
quallify=yes
canreinvite=no
[3000]
type=friend
host=dynamic
nat=yes
context=inbound
dtmfmode=rfc2833
username=3000
secret=1234
quallify=yes
canreinvite=no
[6168182996]
type=friend
host=dynamic
nat=yes
context=outb2
dtmfmode=rfc2833
username=6168182996
secret=1234
quallify=yes
canreinvite=no
[VITELITY]
type=friend
host=64.2.142.93
port=5060
dtmfmode=auto
context=inbound
[QWEST_OUT]
type=friend
host=67.135.79.80
port=5060
dtmfmode=inband
[QWEST8XX_IN]
type=friend
host=67.135.79.199
port=5060
context=qwest800
[DIDX1]
type=peer
host=67.15.128.14
context=inbound
canreinvite=no
[DIDX2]
type=peer
host=67.15.128.18
context=inbound
canreinvite=no
[DIDX3]
type=peer
host=208.44.220.237
context=inbound
canreinvite=no
[DIDX4]
type=peer
host=208.44.220.234
context=inbound
canreinvite=no
[DIDX5]
type=peer
host=209.62.66.242
context=inbound
canreinvite=no
[DIDX6]
type=peer
host=64.246.22.119
context=inbound
canreinvite=no
[DIDX7]
type=peer
host=70.84.58.18
context=inbound
canreinvite=no
[DIDX8]
type=peer
host=174.133.195.194
context=inbound
canreinvite=no
*iax.conf*
[general]
bandwidth=low
disallow=all
allow=ulaw
allow=alaw
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
register=equity_out:1234@74.204.4.166
;register = IAX2/castlewire_trix:1234@74.204.4.206
[CASTLEWIRE]
type=friend
;host=74.204.4.206
host=dynamic
trunk=yes
auth=md5,plaintext,rsa
secret=1234
username=CASTLEWIRE
qualify=yes
context=outb2
[castlewire_trix]
type=friend
;host=74.204.4.206
host=dynamic
trunk=yes
auth=md5,plaintext,rsa
secret=1234
username=castlewire_trix
qualify=yes
context=outb2
requirecalltoken=no
[equity]
type=friend
host=dynamic
context=equity-fix
secret=1234
username=default
channels=10
trunk=yes
timezone=America/Detroit
qualify=yes
requirecalltoken=no
[equity_out]
type=friend
host=dynamic
context=outb2
secret=1234
username=equity_out
channels=10
trunk=yes
timezone=America/Detroit
qualify=yes
requirecalltoken=no
*extensions.conf*
[inbound]
;Equity Logistics
;exten = 6168182400,1,Dial(IAX2/equity/${EXTEN})
;exten = 6168182400,n,Hangup()
;exten = 8182400,1,Dial(IAX2/equity/${EXTEN})
;exten = 8182400,n,Hangup()
exten = 6168182400,1,Dial(SIP/equity/${EXTEN})
exten = 6168182400,n,Hangup()
exten = 6168182996,1,Dial(SIP/${EXTEN})
exten = 6168182996,n,Hangup()
;exten = 6168182996,1,Answer()
;exten = 6168182996,n,Milliwatt()
exten = 3000,1,Dial(SIP/${EXTEN})
exten = 3000,n,Hangup()
;CASTELWIRE NUMBERS
exten = 6168182000,1,Dial(IAX2/castlewire_trix/${EXTEN})
exten = 6168182000,n,Hangup()
;exten = 6168182000,1,Dial(SIP/4403712250@12.194.10.18)
;exten = 6168182000,n,Hangup()
exten = 6168182999,1,Set(portnum=${CALLERID(rdnis)})
exten = 6168182999,n,Set(cutNum=${CUT(portnum|\-|6)})
exten = 6168182999,n,Dial(SIP/${cutNum})
exten = 6168182999,n,Hangup()
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] One Way Audio
Also it could be the routing issue as well. -- Sent from my iPhone On Mar 9, 2011, at 7:43 PM, Duncan Turnbull dun...@e-simple.co.nz wrote: So that suggests audio is flowing as follows in a unidirectional manner 199.173.66.22.53102 74.204.4.5.11732 IP 74.204.4.5.11732 209.216.2.203.60362 Somewhere near the end this pops up which is slightly different, I am guessing 74.204.4.5 is your asterisk box 19:18:36.389548 IP 74.204.4.5.11732 174.133.195.194.18364: UDP, length 172 I am not sure why this is happening or if its still part of the same conversation Overall it looks a bit like the asterisk box thinks it needs to send rtp to a different location than perhaps its meant to i.e. its asymmetric - you can check the sdp in the sip invite to see where media is expected to be sent to There is no rtp coming back from 209.216.2.203 so possibly this is device that isn't meant to be part of the conversation and either doesn't exist or is not expecting anything and thus not responding What are the addresses of the devices in this conversation? so that you can match the traffic to device Cheers Duncan On 10/03/2011, at 1:20 PM, Tim King wrote: It looks like this: 19:18:34.782016 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.789527 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.802064 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.809757 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.821855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.829598 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.842015 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.849764 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.861902 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.869568 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.881882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.889739 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.901882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.909612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.921984 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.929664 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.941855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.949589 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.962003 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.969592 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.981851 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.989543 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.002006 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.009973 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.022008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.029539 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.042071 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.049561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.062008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.069612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.081986 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.089519 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.101918 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.109722 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.122021 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.129590 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.141878 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.149709 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.161886 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.169561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.181879 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.189710 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.201965 IP 199.173.66.22.53103 74.204.4.5.11733: UDP, length 60 19:18:35.201974 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.209552 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.221898 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.229625 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.241894 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.249566 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.261999 IP 199.173.66.22.53102
Re: [asterisk-users] One Way Audio
It looks like the issue was my provider enforcing a codec translation that was not working. On Thu, Mar 10, 2011 at 9:21 AM, Satish Patel satish...@hotmail.com wrote: Also it could be the routing issue as well. -- Sent from my iPhone On Mar 9, 2011, at 7:43 PM, Duncan Turnbull dun...@e-simple.co.nz wrote: So that suggests audio is flowing as follows in a unidirectional manner 199.173.66.22.53102 74.204.4.5.11732 IP 74.204.4.5.11732 209.216.2.203.60362 Somewhere near the end this pops up which is slightly different, I am guessing 74.204.4.5 is your asterisk box 19:18:36.389548 IP 74.204.4.5.11732 174.133.195.194.18364: UDP, length 172 I am not sure why this is happening or if its still part of the same conversation Overall it looks a bit like the asterisk box thinks it needs to send rtp to a different location than perhaps its meant to i.e. its asymmetric - you can check the sdp in the sip invite to see where media is expected to be sent to There is no rtp coming back from 209.216.2.203 so possibly this is device that isn't meant to be part of the conversation and either doesn't exist or is not expecting anything and thus not responding What are the addresses of the devices in this conversation? so that you can match the traffic to device Cheers Duncan On 10/03/2011, at 1:20 PM, Tim King wrote: It looks like this: 19:18:34.782016 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.789527 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.802064 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.809757 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.821855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.829598 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.842015 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.849764 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.861902 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.869568 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.881882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.889739 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.901882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.909612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.921984 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.929664 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.941855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.949589 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.962003 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.969592 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.981851 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.989543 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.002006 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.009973 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.022008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.029539 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.042071 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.049561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.062008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.069612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.081986 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.089519 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.101918 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.109722 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.122021 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.129590 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.141878 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.149709 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.161886 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.169561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.181879 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.189710 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.201965 IP 199.173.66.22.53103 74.204.4.5.11733: UDP, length 60 19:18:35.201974 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.209552 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.221898 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.229625 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.241894 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.249566 IP
Re: [asterisk-users] One Way Audio
Still not working now that audio is restored jitter makes it inaudible? I am ready to move this to commercial if someone can tell me how I need to pay for support, Thanks Tim On Thu, Mar 10, 2011 at 10:19 AM, Tim King t...@compnetwork.net wrote: It looks like the issue was my provider enforcing a codec translation that was not working. On Thu, Mar 10, 2011 at 9:21 AM, Satish Patel satish...@hotmail.comwrote: Also it could be the routing issue as well. -- Sent from my iPhone On Mar 9, 2011, at 7:43 PM, Duncan Turnbull dun...@e-simple.co.nz wrote: So that suggests audio is flowing as follows in a unidirectional manner 199.173.66.22.53102 74.204.4.5.11732 IP 74.204.4.5.11732 209.216.2.203.60362 Somewhere near the end this pops up which is slightly different, I am guessing 74.204.4.5 is your asterisk box 19:18:36.389548 IP 74.204.4.5.11732 174.133.195.194.18364: UDP, length 172 I am not sure why this is happening or if its still part of the same conversation Overall it looks a bit like the asterisk box thinks it needs to send rtp to a different location than perhaps its meant to i.e. its asymmetric - you can check the sdp in the sip invite to see where media is expected to be sent to There is no rtp coming back from 209.216.2.203 so possibly this is device that isn't meant to be part of the conversation and either doesn't exist or is not expecting anything and thus not responding What are the addresses of the devices in this conversation? so that you can match the traffic to device Cheers Duncan On 10/03/2011, at 1:20 PM, Tim King wrote: It looks like this: 19:18:34.782016 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.789527 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.802064 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.809757 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.821855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.829598 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.842015 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.849764 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.861902 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.869568 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.881882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.889739 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.901882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.909612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.921984 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.929664 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.941855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.949589 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.962003 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.969592 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.981851 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.989543 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.002006 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.009973 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.022008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.029539 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.042071 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.049561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.062008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.069612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.081986 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.089519 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.101918 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.109722 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.122021 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.129590 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.141878 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.149709 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.161886 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.169561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.181879 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.189710 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.201965 IP 199.173.66.22.53103 74.204.4.5.11733: UDP, length 60 19:18:35.201974 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.209552 IP 74.204.4.5.11732 209.216.2.203.60362: UDP,
[asterisk-users] One Way Audio
I am having trouble with no return audio on inbound calls. I have been working on this for 18 hours and even built a fresh server and moved everything over and I am getting the same results. I need someone that can help get this resolved tonight. I can provide access to all machines involved. Please email me at tim.compnetw...@gmail.com if you can help. -- _ -- 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
So that suggests audio is flowing as follows in a unidirectional manner 199.173.66.22.53102 74.204.4.5.11732 IP 74.204.4.5.11732 209.216.2.203.60362 Somewhere near the end this pops up which is slightly different, I am guessing 74.204.4.5 is your asterisk box 19:18:36.389548 IP 74.204.4.5.11732 174.133.195.194.18364: UDP, length 172 I am not sure why this is happening or if its still part of the same conversation Overall it looks a bit like the asterisk box thinks it needs to send rtp to a different location than perhaps its meant to i.e. its asymmetric - you can check the sdp in the sip invite to see where media is expected to be sent to There is no rtp coming back from 209.216.2.203 so possibly this is device that isn't meant to be part of the conversation and either doesn't exist or is not expecting anything and thus not responding What are the addresses of the devices in this conversation? so that you can match the traffic to device Cheers Duncan On 10/03/2011, at 1:20 PM, Tim King wrote: It looks like this: 19:18:34.782016 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.789527 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.802064 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.809757 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.821855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.829598 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.842015 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.849764 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.861902 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.869568 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.881882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.889739 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.901882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.909612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.921984 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.929664 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.941855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.949589 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.962003 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.969592 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.981851 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.989543 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.002006 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.009973 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.022008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.029539 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.042071 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.049561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.062008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.069612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.081986 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.089519 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.101918 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.109722 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.122021 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.129590 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.141878 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.149709 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.161886 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.169561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.181879 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.189710 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.201965 IP 199.173.66.22.53103 74.204.4.5.11733: UDP, length 60 19:18:35.201974 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.209552 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.221898 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.229625 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.241894 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.249566 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.261999 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.269701 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.281873 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.289521 IP 74.204.4.5.11732
Re: [asterisk-users] One Way Audio
209.216.2.203 is sip signaling server and 199.173.66.22 is media servers. BTW Did you try config_1 option. Please send us your configuration and we will help you configure it properly. Either you can post them here or you can send them directly to contact-supp...@didforsale.com Jai www.didforsale.com. On Wed, Mar 9, 2011 at 4:43 PM, Duncan Turnbull dun...@e-simple.co.nzwrote: So that suggests audio is flowing as follows in a unidirectional manner 199.173.66.22.53102 74.204.4.5.11732 IP 74.204.4.5.11732 209.216.2.203.60362 Somewhere near the end this pops up which is slightly different, I am guessing 74.204.4.5 is your asterisk box 19:18:36.389548 IP 74.204.4.5.11732 174.133.195.194.18364: UDP, length 172 I am not sure why this is happening or if its still part of the same conversation Overall it looks a bit like the asterisk box thinks it needs to send rtp to a different location than perhaps its meant to i.e. its asymmetric - you can check the sdp in the sip invite to see where media is expected to be sent to There is no rtp coming back from 209.216.2.203 so possibly this is device that isn't meant to be part of the conversation and either doesn't exist or is not expecting anything and thus not responding What are the addresses of the devices in this conversation? so that you can match the traffic to device Cheers Duncan On 10/03/2011, at 1:20 PM, Tim King wrote: It looks like this: 19:18:34.782016 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.789527 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.802064 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.809757 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.821855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.829598 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.842015 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.849764 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.861902 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.869568 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.881882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.889739 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.901882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.909612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.921984 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.929664 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.941855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.949589 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.962003 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.969592 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.981851 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.989543 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.002006 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.009973 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.022008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.029539 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.042071 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.049561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.062008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.069612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.081986 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.089519 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.101918 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.109722 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.122021 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.129590 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.141878 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.149709 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.161886 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.169561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.181879 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.189710 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.201965 IP 199.173.66.22.53103 74.204.4.5.11733: UDP, length 60 19:18:35.201974 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.209552 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.221898 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.229625 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.241894 IP
Re: [asterisk-users] One Way Audio
You can use this link too. http://www.didforsale.com/blog/how-to-setup-your-asterisk-server-with-didforsale Keep the context as context=from-trunk. -Jai On Wed, Mar 9, 2011 at 5:01 PM, Jai Rangi jpra...@didforsale.com wrote: 209.216.2.203 is sip signaling server and 199.173.66.22 is media servers. BTW Did you try config_1 option. Please send us your configuration and we will help you configure it properly. Either you can post them here or you can send them directly to contact-supp...@didforsale.com Jai www.didforsale.com. On Wed, Mar 9, 2011 at 4:43 PM, Duncan Turnbull dun...@e-simple.co.nzwrote: So that suggests audio is flowing as follows in a unidirectional manner 199.173.66.22.53102 74.204.4.5.11732 IP 74.204.4.5.11732 209.216.2.203.60362 Somewhere near the end this pops up which is slightly different, I am guessing 74.204.4.5 is your asterisk box 19:18:36.389548 IP 74.204.4.5.11732 174.133.195.194.18364: UDP, length 172 I am not sure why this is happening or if its still part of the same conversation Overall it looks a bit like the asterisk box thinks it needs to send rtp to a different location than perhaps its meant to i.e. its asymmetric - you can check the sdp in the sip invite to see where media is expected to be sent to There is no rtp coming back from 209.216.2.203209.216.2.203 so possibly this is device that isn't meant to be part of the conversation and either doesn't exist or is not expecting anything and thus not responding What are the addresses of the devices in this conversation? so that you can match the traffic to device Cheers Duncan On 10/03/2011, at 1:20 PM, Tim King wrote: It looks like this: 19:18:34.782016 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.789527 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.802064 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.809757 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.821855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.829598 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.842015 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.849764 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.861902 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.869568 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.881882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.889739 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.901882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.909612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.921984 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.929664 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.941855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.949589 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.962003 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.969592 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.981851 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.989543 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.002006 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.009973 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.022008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.029539 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.042071 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.049561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.062008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.069612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.081986 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.089519 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.101918 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.109722 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.122021 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.129590 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.141878 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.149709 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.161886 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.169561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.181879 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.189710 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.201965 IP 199.173.66.22.53103 74.204.4.5.11733: UDP, length 60 19:18:35.201974 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172
[asterisk-users] One way audio problem
Hi, Asterisk is making a call to a peer. In 200 ok, peer is sending its application server ip in contact field, so asterisk sends ACK to that IP. RTP starts flowing between endpoints and peer plays an IVR and asks for destination number. After entering destination number peer's application server sends INVITE again with different media IP and asterisk accepts with 200 ok. RTP from peer comes from new media IP but asterisk keep sending to their old media IP that came in their 200 ok before and don't send to new one. Hence, I can hear their voice but they can't. Does anyone know how to make asterisk send RTP to new media IP that came in new INVITE within the call? Thanks Deepika -- _ -- 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 for xlite clients behind NAT
I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip[1001];Work type=peer dtmfmode=rfc2833 context=sip insecure=very host=sip.domain.com nat=no[1000];IPKall type=peer dtmfmode=rfc2833 context=sip insecure=very host=voiper.ipkall.com nat=no I pasted the log here - http://pastie.org/1163238 I have tried connecting both of the clients to another sip service(sip2sip.info) and did not have the same problems. Any suggestions would be great. Thanks, Tom -- _ -- 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 for xlite clients behind NAT
On 09/16/2010 06:58 PM, Thomas Johnson wrote: I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip [1001];Work type=peer dtmfmode=rfc2833 context=sip insecure=very host=sip.domain.com http://sip.domain.com nat=no [1000];IPKall type=peer dtmfmode=rfc2833 context=sip insecure=very host=voiper.ipkall.com http://voiper.ipkall.com nat=no You seem to be using nat=no shouldn't that be nat=yes? I pasted the log here - http://pastie.org/1163238 I have tried connecting both of the clients to another sip service(sip2sip.info http://sip2sip.info) and did not have the same problems. Any suggestions would be great. Thanks, Tom -- _ -- 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 for xlite clients behind NAT
the client that is behind nat is [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip do I have to enable nat on all of them? On Thu, Sep 16, 2010 at 1:36 PM, Sebastian s...@open-t.co.uk wrote: On 09/16/2010 06:58 PM, Thomas Johnson wrote: I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip [1001];Work type=peer dtmfmode=rfc2833 context=sip insecure=very host=sip.domain.com http://sip.domain.com nat=no [1000];IPKall type=peer dtmfmode=rfc2833 context=sip insecure=very host=voiper.ipkall.com http://voiper.ipkall.com nat=no You seem to be using nat=no shouldn't that be nat=yes? I pasted the log here - http://pastie.org/1163238 I have tried connecting both of the clients to another sip service( sip2sip.info http://sip2sip.info) and did not have the same problems. Any suggestions would be great. Thanks, Tom -- _ -- 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 for xlite clients behind NAT
On 09/16/2010 07:59 PM, Thomas Johnson wrote: the client that is behind nat is [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip do I have to enable nat on all of them? I don't think so. It's just that you didn't specify which client is which. My next guess is that it is a codec problem. I've had similar problems - and upon checking Asterisk logs - I discovered that the client and Asterisk weren't agreeing correctly on codecs. Can you double-check your X-lite configuration - and maybe try to ulaw or alaw as the only codec at both ends? Sebastian On Thu, Sep 16, 2010 at 1:36 PM, Sebastian s...@open-t.co.uk mailto:s...@open-t.co.uk wrote: On 09/16/2010 06:58 PM, Thomas Johnson wrote: I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip [1001];Work type=peer dtmfmode=rfc2833 context=sip insecure=very host=sip.domain.com http://sip.domain.com http://sip.domain.com nat=no [1000];IPKall type=peer dtmfmode=rfc2833 context=sip insecure=very host=voiper.ipkall.com http://voiper.ipkall.com http://voiper.ipkall.com nat=no You seem to be using nat=no shouldn't that be nat=yes? I pasted the log here - http://pastie.org/1163238 I have tried connecting both of the clients to another sip service(sip2sip.info http://sip2sip.info http://sip2sip.info) and did not have the same problems. Any suggestions would be great. Thanks, Tom -- _ -- 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 for xlite clients behind NAT
I have tried doing that with just ulaw and alaw, respectively, and nothing changed Also, if I disable the firewall in my router I lose incoming audio and outgoing audio works. On Thu, Sep 16, 2010 at 2:50 PM, Sebastian s...@open-t.co.uk wrote: On 09/16/2010 07:59 PM, Thomas Johnson wrote: the client that is behind nat is [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip do I have to enable nat on all of them? I don't think so. It's just that you didn't specify which client is which. My next guess is that it is a codec problem. I've had similar problems - and upon checking Asterisk logs - I discovered that the client and Asterisk weren't agreeing correctly on codecs. Can you double-check your X-lite configuration - and maybe try to ulaw or alaw as the only codec at both ends? Sebastian On Thu, Sep 16, 2010 at 1:36 PM, Sebastian s...@open-t.co.uk mailto:s...@open-t.co.uk wrote: On 09/16/2010 06:58 PM, Thomas Johnson wrote: I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip [1001];Work type=peer dtmfmode=rfc2833 context=sip insecure=very host=sip.domain.com http://sip.domain.com http://sip.domain.com nat=no [1000];IPKall type=peer dtmfmode=rfc2833 context=sip insecure=very host=voiper.ipkall.com http://voiper.ipkall.com http://voiper.ipkall.com nat=no You seem to be using nat=no shouldn't that be nat=yes? I pasted the log here - http://pastie.org/1163238 I have tried connecting both of the clients to another sip service(sip2sip.info http://sip2sip.info http://sip2sip.info) and did not have the same problems. Any suggestions would be great. Thanks, Tom -- _ -- 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 -- _ -- 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 for xlite clients behind NAT
On Thu, Sep 16, 2010 at 5:50 PM, Thomas Johnson tomfma...@gmail.com wrote: Also, if I disable the firewall in my router I lose incoming audio and outgoing audio works. http://www.aocomputing.net/?p=3 -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 for xlite clients behind NAT
I already have that covered [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip The server is not behind NAT only the client above is On Thu, Sep 16, 2010 at 4:59 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Thu, Sep 16, 2010 at 5:50 PM, Thomas Johnson tomfma...@gmail.com wrote: Also, if I disable the firewall in my router I lose incoming audio and outgoing audio works. http://www.aocomputing.net/?p=3 -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 for xlite clients behind NAT
On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson tomfma...@gmail.com wrote: The server is not behind NAT only the client above is Sounds like a phone (not asterisk) issue then, make sure you have setup your NAT and port forwarding properly on the client side. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 for xlite clients behind NAT
If you are using linux firewall, try this, it was very usefull to me: iptables -t nat -A PREROUTING -i ethx -p tcp --dport 5060 -j DNAT --to ip_phoneiptables -t nat -A PREROUTING -i ethx -p udp --dport 5060 -j DNAT --to iip_phoneiptables -A FORWARD -p TCP --dport 5060 -j ACCEPTiptables -A FORWARD -p UDP --dport 5060 -j ACCEPT Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Thu, 16 Sep 2010 18:45:38 -0400 From: paul.belan...@polybeacon.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] one way audio for xlite clients behind NAT On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson tomfma...@gmail.com wrote: The server is not behind NAT only the client above is Sounds like a phone (not asterisk) issue then, make sure you have setup your NAT and port forwarding properly on the client side. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 overlapdial is set to yes
Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local --- hipath 300 - isdn pri asterisk -- isdn pri - telco-- dest. Here is my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes overlapdial=yes autofalltrought=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 ; Span 2: WPE1/1 wanpipe2 card 1 HDB3/CCS/CRC4 RED group=1,12 context=from-internal switchtype = euroisdn ;overlapdial = outgoing priindication = inband signalling = pri_net channel = 32-46,48-62 context = default group = 63 Span 4: WPE1/3 wanpipe4 card 3 HDB3/CCS/CRC4 group=4,14 context=outrt-001-PSTN_E1 switchtype=qsig signalling=pri_cpe ;facilityenable=yes ;callprogress=yes pridialplan=unknown prilocaldialplan=unknown ;priindication = outofband ;overlapdial = incoming ;priexclusive = yes ;pritimer = t200,1000 ;pritimer = t313,4000 ;immediate=yes channel = 94-108,110-124 context = default group = 63 Any suggestion will be much appreciated. Regards, Mac -- _ -- 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 overlapdial is set to yes
Hi Group, I was able to resolve the problem by disabling the echo cancellation in a104d and using the same dahdi config. Thanks... - Original Message From: leonimar cape leo_mac...@yahoo.com To: asterisk-users@lists.digium.com Sent: Wednesday, September 15, 2010 6:12:35 PM Subject: [asterisk-users] One way audio when overlapdial is set to yes Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local --- hipath 300 - isdn pri asterisk -- isdn pri - telco-- dest. Here is my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes overlapdial=yes autofalltrought=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 ; Span 2: WPE1/1 wanpipe2 card 1 HDB3/CCS/CRC4 RED group=1,12 context=from-internal switchtype = euroisdn ;overlapdial = outgoing priindication = inband signalling = pri_net channel = 32-46,48-62 context = default group = 63 Span 4: WPE1/3 wanpipe4 card 3 HDB3/CCS/CRC4 group=4,14 context=outrt-001-PSTN_E1 switchtype=qsig signalling=pri_cpe ;facilityenable=yes ;callprogress=yes pridialplan=unknown prilocaldialplan=unknown ;priindication = outofband ;overlapdial = incoming ;priexclusive = yes ;pritimer = t200,1000 ;pritimer = t313,4000 ;immediate=yes channel = 94-108,110-124 context = default group = 63 Any suggestion will be much appreciated. Regards, Mac -- _ -- 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 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
[asterisk-users] One way audio problem, a=sendonly and a re-invite
Hello all, I have a problem where problem with one way audio, and I think it's related to a=sendonly and a re-invite. Can anyone please assist? The scenario is as follows - We send an INVITE to a peer, and it replies with a 100 Trying, and then a 183 Session Progress message containing a=sendonly. - Asterisk plays the caller music on hold, which I believe is correct if we have an a=sendonly. - Then the peer sends a 200 OK which also has a=sendonly, and then sends a re-invite which I've copied and pasted below. - We have canreinvite=no set in sip.conf, but I'm not sure if we should be rejecting this re-invite or not because it does contain a=sendrecv. If it should be rejected what error should Asterisk return, and how can we establish two way audio? - After this re-invite Asterisk replies with a 100 Trying and then a 200 OK which contains a=recvonly. - Call is established but called party cannot hear caller. Here's the re-invite message - note that Asterisk is on port 5070: U 2010/05/05 12:47:38.139701 (peer):5060 - (asterisk):5070 INVITE sip:(called number)@(asterisk):5070 SIP/2.0. Via: SIP/2.0/UDP (peer):5060;branch=z9hG4bK2sansay7330954rdb6594. To: User sip:(called number)@(asterisk):5070;tag=as3ddcc528. From: sip:(called number)@(peer):5060;tag=sansay7330954rdb6594. Call-ID: 58eb52aa414c5e465c3c1a15603093fb@(asterisk). CSeq: 2 INVITE. Contact: sip:(called number)@(peer):5060. Max-Forwards: 69. Content-Type: application/sdp. Content-Length: 297. . v=0. o=Sansay-VSXi 188 1 IN IP4 (peer). s=Session Controller. c=IN IP4 (other unknown IP, maybe of called number?). t=0 0. m=audio 6932 RTP/AVP 18 0 8 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=ptime:20. Any help would be much appreciated! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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 after Hold
I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet, and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet parameters are all set correctly in sip.conf. An inbound call from Sipphone works great until the local channel places the call on hold. During hold, the Sipphone user cannot hear music, only silence. The silence continues after the hold, though the local phone can hear the Sipphone user. Every possible combination of nat=yes, no, maybe, possibly or never gives the same result. Further, canreinvite=yes/no/nonat has no result. I suspect a possible reinvite issue with Asterisk being out of the RTP stream, so I have tried all the usual variables in the DialI() command as well to no avail. Any thoughts on how to fix one-way-audio after a hold? --Brent -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One-Way Audio after Hold
Brent Torrenga wrote: I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet, and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet parameters are all set correctly in sip.conf. An inbound call from Sipphone works great until the local channel places the call on hold. During hold, the Sipphone user cannot hear music, only silence. The silence continues after the hold, though the local phone can hear the Sipphone user. Every possible combination of nat=yes, no, maybe, possibly or never gives the same result. Further, canreinvite=yes/no/nonat has no result. I suspect a possible reinvite issue with Asterisk being out of the RTP stream, so I have tried all the usual variables in the DialI() command as well to no avail. Any thoughts on how to fix one-way-audio after a hold? I have the same problem, only my customers report that it only happens occasionally. Most of the time, they can transfer calls just fine. They can also put calls on hold and retrieve them as expected. However, sometimes, about once a day, they try to recover a call and the caller can't hear them, but they can hear the caller. I've seen this happen once, but I've been unable to reproduce it reliably. Any ideas? Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way audio with Grandstream HT503
Hello list ! I'm having one way audio on incoming and outgoing calls. Outgoing audio works fine, incoming audio is not working. My setup is the following : incoming calls : PSTN -- FXOport -- HT503 -- WANport -- Asterisk -- WANport -- HT503 (the same) -- FXSport -- DECTphone outgoing calls : DECTphone -- FXSport -- HT503 -- WAN-port -- Asterisk -- internet (VoIPprovider) I've done a TCPdump on the Asterisk-server : 18:20:21.189504 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.193065 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.210111 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.213065 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.229848 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.233064 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.250013 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.253049 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.269737 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.273058 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.289918 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.293048 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.310080 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.313058 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.329819 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.333047 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.349985 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.353054 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.370164 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.373046 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.389886 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.393031 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.410053 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.413042 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.430218 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.433033 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.449957 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.453039 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.469694 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.473039 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.489857 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.493059 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.509593 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.513028 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.530190 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.533039 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.549930 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.553025 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.573037 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.593024 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.613011 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.630162 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.630208 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.633022 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.653012 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.670492 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.673019 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.689799 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.689844 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.693028 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.713036 IP my.asterisk.server.be.20056 vds.server.net.11574: UDP, length 172 18:20:21.730131 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.730176 IP vds.server.net.11574 my.asterisk.server.be.20056: UDP, length 172 18:20:21.733147
[asterisk-users] One Way Audio from External Sip Soft Hard Phone
I have a problem with one way audio on Sip and I guess it may be a NAT issue, in the example below 204 is rung by 208 (xlite external) I dial perfectly but when I get to the answering of the Asterisk, I can hear audio from the Asterisk but cannot get audio to the Asterisk, ie If I ring the voice mail , Asterisk answers and then cannot hear my password... I have put the Ports Forward etc...5004-5080 1-2 Any ideas - even what to test next would be good... -- Executing [...@macro-stdexten:13] Dial(SIP/208-00a10004, SIP/204) in new stack -- Called 204 -- SIP/204-00a11584 is ringing -- SIP/204-00a11584 answered SIP/208-00a10004 [Jul 7 16:15:08] WARNING[834]: chan_sip.c:1993 retrans_pkt: Maximum retries exceeded on transmission NjQ1NTA3Mzg3YzI5ZTNlYzI3MWU2NzE4YWE3ZTI2MDE. for seqno 2 (Critical Response) [Jul 7 16:15:08] WARNING[834]: chan_sip.c:2017 retrans_pkt: Hanging up call NjQ1NTA3Mzg3YzI5ZTNlYzI3MWU2NzE4YWE3ZTI2MDE. - no reply to our critical packet. == Spawn extension (macro-stdexten, s, 13) exited non-zero on 'SIP/208-00a10004' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 13) exited non-zero on 'SIP/208-00a10004' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio from External Sip Soft Hard Phone
On Tue, Jul 7, 2009 at 9:55 PM, Paul Edgarp...@tabs.co.nz wrote: I have a problem with one way audio on Sip and I guess it may be a NAT issue, in the example below 204 is rung by 208 (xlite external) I dial perfectly but when I get to the answering of the Asterisk, I can hear audio from the Asterisk but cannot get audio to the Asterisk, ie If I ring the voice mail , Asterisk answers and then cannot hear my password… I have put the Ports Forward etc…5004-5080 1-2 Any ideas – even what to test next would be good… -- Executing [...@macro-stdexten:13] Dial(SIP/208-00a10004, SIP/204) in new stack -- Called 204 -- SIP/204-00a11584 is ringing -- SIP/204-00a11584 answered SIP/208-00a10004 [Jul 7 16:15:08] WARNING[834]: chan_sip.c:1993 retrans_pkt: Maximum retries exceeded on transmission NjQ1NTA3Mzg3YzI5ZTNlYzI3MWU2NzE4YWE3ZTI2MDE. for seqno 2 (Critical Response) [Jul 7 16:15:08] WARNING[834]: chan_sip.c:2017 retrans_pkt: Hanging up call NjQ1NTA3Mzg3YzI5ZTNlYzI3MWU2NzE4YWE3ZTI2MDE. - no reply to our critical packet. == Spawn extension (macro-stdexten, s, 13) exited non-zero on 'SIP/208-00a10004' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 13) exited non-zero on 'SIP/208-00a10004' Where is the NAT or is it on both sides? Answer that and turn on SIP debugging and post the output and I am sure someone can help you. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way AUDIO
Here's my .02 - local lan is probably behind a firewall meaning that the 5060 gets out ok to send your audio, but the 1-2 range that the other side comes in on is blocked. You don't have the problem with static WAN because it is not behind the firewall or has more ports open. Do a netstat -an during each call and see what is different. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC Sent: Monday, April 06, 2009 6:04 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] One way AUDIO Few Running figures !! On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote: I have a server with 2 Lan Cards. Now, when I am trying to make calls using Local Lan, its One way Audio which means customer cant hear me but if I use Static IP with Wan Connection, it works perfectly. I changed the network from loc1 to loc2 but its same. I tried changing Ethernet Card but no use. What could be the Issue ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way AUDIO
I have a server with 2 Lan Cards. Now, when I am trying to make calls using Local Lan, its One way Audio which means customer cant hear me but if I use Static IP with Wan Connection, it works perfectly. I changed the network from loc1 to loc2 but its same. I tried changing Ethernet Card but no use. What could be the Issue ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way AUDIO
Can it be that any Port got blocked ? On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote: I have a server with 2 Lan Cards. Now, when I am trying to make calls using Local Lan, its One way Audio which means customer cant hear me but if I use Static IP with Wan Connection, it works perfectly. I changed the network from loc1 to loc2 but its same. I tried changing Ethernet Card but no use. What could be the Issue ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way AUDIO
How tcpdump on interface show?? 2009/4/6 David @ULC ucoms2...@gmail.com: Can it be that any Port got blocked ? On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote: I have a server with 2 Lan Cards. Now, when I am trying to make calls using Local Lan, its One way Audio which means customer cant hear me but if I use Static IP with Wan Connection, it works perfectly. I changed the network from loc1 to loc2 but its same. I tried changing Ethernet Card but no use. What could be the Issue ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giancarlo Rubio ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way AUDIO
Few Running figures !! On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote: I have a server with 2 Lan Cards. Now, when I am trying to make calls using Local Lan, its One way Audio which means customer cant hear me but if I use Static IP with Wan Connection, it works perfectly. I changed the network from loc1 to loc2 but its same. I tried changing Ethernet Card but no use. What could be the Issue ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way audio after IVR tree
Hi, I have a couple of users who are having a peculiar problem. On some outbound numbers where there is a deep IVR tree (3+ selections), and then a live person picks up, the live person will be unable to hear them on the phone, but they can hear the live person. I've done packet traces and it appears as though audio is being passed both ways, but the audio from the caller is severely muted before it gets to asterisk. Has anyone seen this before? It's almost like the phone thinks its still sending DTMF or something and mutes the audio. I've seen this happen on both linksys 942 and 962 phones. Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
On Thu, 16 Oct 2008, GNUbie wrote: Hello, On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: A packet trace will probably show exactly what is happening. Try: tcpdump -nlXs 8192 -i eth0 port 5060 You should be able to see the SIP information going back and forth and will probably show you that your NAT rules are applying when they shouldn't. I agree with first turning off your firewall and testing... but if that actually solves the problem you need to know why. This should tell why. Why eth0 when in fact it is not being used AFAIK? My Asterisk box is connected to the LAN via its eth1 interface and the SIP phone is calling from the LAN to the analog telephone via FXO/POTS. Again, below is the call scenario diagram: [SNOM] ==LAN== eth1 [ASTERISK] fxo ==POTS== [ANALOG_TELEPHONE] eth0 || INTERNET You should try on both interfaces. If you see packets on eth0 then your NAT rules are leaking! Try on eth1 to see the SIP headers and tell if your NAT rules are doing what you expect. This is always my first attack... j Please advice. Thank you in advance. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *** Handled by Will's new toy *** ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
GNUbie wrote: What particular configs are you looking for? Below is my current setup and scenario: [snom] ==LAN== [asterisk] ==FXO/POTS == [analog_telephone/mobile_phone] SNOM is using the 192.168.101.102 IP address Asterisk is using 192.168.101.1 IP address for its eth1 interface FXO port is connected to the POTS SNOM doesn't need to go out to the Internet in this scenario, AFAIK. Below is my current NAT rules: # iptables -L -v -t nat Chain PREROUTING (policy ACCEPT 63795 packets, 7162K bytes) pkts bytes target prot opt in out source destination 11460 760K RETURN 0-- anyany 192.168.101.0/24 !192.168.101.0/24 Chain POSTROUTING (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination 11408 757K MASQUERADE 0-- anyeth0192.168.101.0/24 anywhere Chain OUTPUT (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination Please advice if you need more information from me. Regards, GNUbie Having had many years of experience working with iptables I can tell you that when IP Forwarding is enabled on a Linux machine things can get a bit tricky. In my experience using a Masquerade rule can cause some major weirdness. Try doing this: Instead of the Masquerade rule use: iptables -t nat -A POSTROUTING -i eth1 -o eth0 -j SNAT --to-source public ip of eth0 Also, in the general section of your sip.conf make sure you have: bindaddr=192.168.101.1 to make sure asterisk is not sending sip packets using the public IP then effectively trying to communicate with the phone by Masquerading the packets coming in over the eth1 to eth0. This is more than likely what is happening. (It's normlly bindaddr=0.0.0.0) Good luck, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
On Thu, Oct 16, 2008 at 09:22:01AM +0800, GNUbie wrote: Hello Daniel, On Tue, Oct 14, 2008 at 12:12 AM, Daniel Hazelbaker [EMAIL PROTECTED] wrote: Might be a stretch, but does the Asterisk log show that the call was answered? I had this problem when interfacing * with an NEC system to do call parking pickup. The NEC would never give a dialtone (nor did it give answer supervision) so * never knew the call got picked up so audio only worked one way. I ended up rigging * to force the line to be considered answered with a patch. Yes, the call has been answered as per Asterisk logs. The CALLER (SNOM SIP Phone) can hear clearly the voice of the target CALLEE (POTS analog telephone) but it is the CALLEE that cannot hear the CALLER's voice. And yet in the output that you showed us, the channels were not in a state of Up. That is: not in a state of finished dialing and stuff and now part of a call. Could you plese double check that? What is the output of 'core show channels' at the time of a call? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hi, Am Donnerstag, den 16.10.2008, 09:37 +0800 schrieb GNUbie: Hello Karsten, On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote: Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? The SNOM 300 is using the NET interface beside the DC 5V port to connect to the LAN. The Asterisk box is using the eth1 to connect to the LAN. As per your instruction, below is my /etc/asterisk/sip.conf : - - - s n i p - - - [general] realm=pbx.domain.com bindport=5060 bindaddr=0.0.0.0 rtptimeout=60 disallow=all allow=ulaw allow=alaw allow=gsm externip=pbx.domain.com localnet=192.168.101.0/255.255.255.0 jbforce=yes allowtransfers=yes maxexpiry=3600 minexpiry=1800 videosupport=no [internal-phones](!) type=friend host=dynamic context=family dtmfmode=rfc2833 insecure=port,invite canreinvite=no nat=no qualify=yes port=5060 [102](internal-phones) username=102 secret=102 callerid=GNUbie102 [EMAIL PROTECTED] - - - s n i p - - - Thanks for the information. In an earlier post You told us, that the local phones talk to asterisk on eth1 using 192.168.101.0 network. Could You please double check, that the phone did not try to register on another IP? The asterisk is IIRC on a dual homed machine. Is Your phone using a DNS lookup to find the asterisk? To which address is that lookup resolved? Another hint: Is Your SNOM using some sort of STUN to lookup an public address? (Just to eliminate some things). HTH, Karsten ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello, On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: A packet trace will probably show exactly what is happening. Try: tcpdump -nlXs 8192 -i eth0 port 5060 You should be able to see the SIP information going back and forth and will probably show you that your NAT rules are applying when they shouldn't. I agree with first turning off your firewall and testing... but if that actually solves the problem you need to know why. This should tell why. Why eth0 when in fact it is not being used AFAIK? My Asterisk box is connected to the LAN via its eth1 interface and the SIP phone is calling from the LAN to the analog telephone via FXO/POTS. Again, below is the call scenario diagram: [SNOM] ==LAN== eth1 [ASTERISK] fxo ==POTS== [ANALOG_TELEPHONE] eth0 || INTERNET Please advice. Thank you in advance. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Daniel, On Tue, Oct 14, 2008 at 12:12 AM, Daniel Hazelbaker [EMAIL PROTECTED] wrote: Might be a stretch, but does the Asterisk log show that the call was answered? I had this problem when interfacing * with an NEC system to do call parking pickup. The NEC would never give a dialtone (nor did it give answer supervision) so * never knew the call got picked up so audio only worked one way. I ended up rigging * to force the line to be considered answered with a patch. Yes, the call has been answered as per Asterisk logs. The CALLER (SNOM SIP Phone) can hear clearly the voice of the target CALLEE (POTS analog telephone) but it is the CALLEE that cannot hear the CALLER's voice. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Karsten, On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote: Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? The SNOM 300 is using the NET interface beside the DC 5V port to connect to the LAN. The Asterisk box is using the eth1 to connect to the LAN. As per your instruction, below is my /etc/asterisk/sip.conf : - - - s n i p - - - [general] realm=pbx.domain.com bindport=5060 bindaddr=0.0.0.0 rtptimeout=60 disallow=all allow=ulaw allow=alaw allow=gsm externip=pbx.domain.com localnet=192.168.101.0/255.255.255.0 jbforce=yes allowtransfers=yes maxexpiry=3600 minexpiry=1800 videosupport=no [internal-phones](!) type=friend host=dynamic context=family dtmfmode=rfc2833 insecure=port,invite canreinvite=no nat=no qualify=yes port=5060 [102](internal-phones) username=102 secret=102 callerid=GNUbie102 [EMAIL PROTECTED] - - - s n i p - - - Thank you in advance. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Did you try it the magic number of times, three? On Sun, Oct 12, 2008 at 9:57 PM, GNUbie [EMAIL PROTECTED] wrote: Hello Tzafrir, On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: This means Zaptel gets silence from Asterisk. What codecs are used? What do you see on 'sip show channels'? I am using the following codecs: # asterisk -rx 'sip show settings' | grep Codecs Codecs: 0xe (gsm|ulaw|alaw) Below is the CLI output: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-081d11d0, Zap/4/1234567) in new stack -- Called 4/1234567 *CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 192.168.101.102 102 3c27a6824ba 00101/2 0x4 (ulaw) No Rx: INVITE 1 active SIP channel *CLI core show channels Channel Location State Application(Data) Zap/4-1 [EMAIL PROTECTED] Dialing AppDial((Outgoing Line)) SIP/102-081d11d0 [EMAIL PROTECTED]:1 RingDial(Zap/4/1234567) 2 active channels 1 active call Can you call from the FXO to Asterisk? (e.g.: to echo test) There is no problem with an inbound calls. I just tried to call the echo test extension number from my mobile phone via FXO/POTS and it works fine. I can hear my own voice. Thank you. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Change all canreinvites to no. On Wed, Oct 15, 2008 at 9:37 PM, GNUbie [EMAIL PROTECTED] wrote: Hello Karsten, On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote: Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? The SNOM 300 is using the NET interface beside the DC 5V port to connect to the LAN. The Asterisk box is using the eth1 to connect to the LAN. As per your instruction, below is my /etc/asterisk/sip.conf : - - - s n i p - - - [general] realm=pbx.domain.com bindport=5060 bindaddr=0.0.0.0 rtptimeout=60 disallow=all allow=ulaw allow=alaw allow=gsm externip=pbx.domain.com localnet=192.168.101.0/255.255.255.0 jbforce=yes allowtransfers=yes maxexpiry=3600 minexpiry=1800 videosupport=no [internal-phones](!) type=friend host=dynamic context=family dtmfmode=rfc2833 insecure=port,invite canreinvite=no nat=no qualify=yes port=5060 [102](internal-phones) username=102 secret=102 callerid=GNUbie102 [EMAIL PROTECTED] - - - s n i p - - - Thank you in advance. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
canreinvite defaults to yes, whether specified or not. http://www.voip-info.org/wiki/view/tips If you follow these directions adapting to your particular circumstances and it doesn't work, post your whole sip.conf Start asterisk with verbose set to 3 or so and turn on sip debugging. I get somewhere in the debug, you will see local NAT IPs that don't belong there, or it will just work. Thanks, Steve Totaro On Thu, Oct 16, 2008 at 12:12 AM, Steve Totaro [EMAIL PROTECTED] wrote: Change all canreinvites to no. On Wed, Oct 15, 2008 at 9:37 PM, GNUbie [EMAIL PROTECTED] wrote: Hello Karsten, On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote: Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? The SNOM 300 is using the NET interface beside the DC 5V port to connect to the LAN. The Asterisk box is using the eth1 to connect to the LAN. As per your instruction, below is my /etc/asterisk/sip.conf : - - - s n i p - - - [general] realm=pbx.domain.com bindport=5060 bindaddr=0.0.0.0 rtptimeout=60 disallow=all allow=ulaw allow=alaw allow=gsm externip=pbx.domain.com localnet=192.168.101.0/255.255.255.0 jbforce=yes allowtransfers=yes maxexpiry=3600 minexpiry=1800 videosupport=no [internal-phones](!) type=friend host=dynamic context=family dtmfmode=rfc2833 insecure=port,invite canreinvite=no nat=no qualify=yes port=5060 [102](internal-phones) username=102 secret=102 callerid=GNUbie102 [EMAIL PROTECTED] - - - s n i p - - - Thank you in advance. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Steve, On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro [EMAIL PROTECTED] wrote: Did you try it the magic number of times, three? I'm sorry. What do you mean? Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Maybe I have my threads confused but I thought you got one way audio when three calls were made, you only mentioned one call. On Thu, Oct 16, 2008 at 12:44 AM, GNUbie [EMAIL PROTECTED] wrote: Hello Steve, On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro [EMAIL PROTECTED] wrote: Did you try it the magic number of times, three? I'm sorry. What do you mean? Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Sorry, wrong thread, time for bed. I thought this was the thread where the guy was having issues with one way audio on his third call, and his Asterisk server was behind NAT. Good night everyone and have pleasant dreams of 700 point drops in the DOW! OT, did you know if the government took the $700+ billion dollars and did not bail out the greedy banks, we could have immediate relief since for the most part, we could suspend Federal Income tax for everyone. A $300 rebate check, give me a break, how about some real stimulus, a rebate (or lack of theft because there is no law that we as individuals have to pay Federal Income tax, and I dare anyone to point it out, a real law, not something the IRS made up, I don't think they are part of the Legislative branch) weekly or bi-weekly depending on how you get paid. It would be immediate and give more money to the people who need it. All your Fed Income tax pays for anyways is the national debt, the clock just maxed out at $10 trillion. Rather than paying it down below the max and keeping it that way, they are building another one with additional digits. Sorry for a TOTALLY OFF topic post. I screwed up so I thought I might as well rant a little. Apologies in sheer exhaustion, Steve Totaro Thanks, Steve Totaro On Thu, Oct 16, 2008 at 12:46 AM, Steve Totaro [EMAIL PROTECTED] wrote: Maybe I have my threads confused but I thought you got one way audio when three calls were made, you only mentioned one call. On Thu, Oct 16, 2008 at 12:44 AM, GNUbie [EMAIL PROTECTED] wrote: Hello Steve, On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro [EMAIL PROTECTED] wrote: Did you try it the magic number of times, three? I'm sorry. What do you mean? Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Steve, On Thu, Oct 16, 2008 at 12:42 PM, Steve Totaro [EMAIL PROTECTED] wrote: canreinvite defaults to yes, whether specified or not. http://www.voip-info.org/wiki/view/tips If you follow these directions adapting to your particular circumstances and it doesn't work, post your whole sip.conf Start asterisk with verbose set to 3 or so and turn on sip debugging. I get somewhere in the debug, you will see local NAT IPs that don't belong there, or it will just work. My /etc/asterisk/sip.conf is at http://lists.digium.com/pipermail/asterisk-users/2008-October/220256.html and my SIP phone is located within the LAN where the Asterisk box is also part of it. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
On Mon, Oct 13, 2008 at 09:57:33AM +0800, GNUbie wrote: Hello Tzafrir, On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: This means Zaptel gets silence from Asterisk. What codecs are used? What do you see on 'sip show channels'? I am using the following codecs: # asterisk -rx 'sip show settings' | grep Codecs Codecs: 0xe (gsm|ulaw|alaw) Below is the CLI output: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-081d11d0, Zap/4/1234567) in new stack -- Called 4/1234567 *CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 192.168.101.102 102 3c27a6824ba 00101/2 0x4 (ulaw) No Rx: INVITE 1 active SIP channel *CLI core show channels Channel Location State Application(Data) Zap/4-1 [EMAIL PROTECTED] Dialing AppDial((Outgoing Line)) SIP/102-081d11d0 [EMAIL PROTECTED]:1 RingDial(Zap/4/1234567) 2 active channels 1 active call So the call is not established yet, right? This is not a temporary state? Can you call from the FXO to Asterisk? (e.g.: to echo test) There is no problem with an inbound calls. I just tried to call the echo test extension number from my mobile phone via FXO/POTS and it works fine. I can hear my own voice. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Tzafrir, On Mon, Oct 13, 2008 at 3:33 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: So the call is not established yet, right? It is already. The CALLER hears the CALLEE's voice but the CALLEE cannot hear the CALLER's voices. This is not a temporary state? What do you mean? Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
On Mon, Oct 13, 2008 at 9:04 AM, GNUbie [EMAIL PROTECTED] wrote: Hello Tzafrir, On Mon, Oct 13, 2008 at 3:33 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: So the call is not established yet, right? It is already. The CALLER hears the CALLEE's voice but the CALLEE cannot hear the CALLER's voices. This is not a temporary state? What do you mean? Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you are going to dismiss (the most probable) problem (NAT) without posting configs, I am not sure how much help you will get, you will probably be dismissed as well. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Steve, On Mon, Oct 13, 2008 at 9:52 PM, Steve Totaro [EMAIL PROTECTED] wrote: If you are going to dismiss (the most probable) problem (NAT) without posting configs, I am not sure how much help you will get, you will probably be dismissed as well. What particular configs are you looking for? Below is my current setup and scenario: [snom] ==LAN== [asterisk] ==FXO/POTS == [analog_telephone/mobile_phone] SNOM is using the 192.168.101.102 IP address Asterisk is using 192.168.101.1 IP address for its eth1 interface FXO port is connected to the POTS SNOM doesn't need to go out to the Internet in this scenario, AFAIK. Below is my current NAT rules: # iptables -L -v -t nat Chain PREROUTING (policy ACCEPT 63795 packets, 7162K bytes) pkts bytes target prot opt in out source destination 11460 760K RETURN 0-- anyany 192.168.101.0/24 !192.168.101.0/24 Chain POSTROUTING (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination 11408 757K MASQUERADE 0-- anyeth0192.168.101.0/24 anywhere Chain OUTPUT (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination Please advice if you need more information from me. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
On Mon, Oct 13, 2008 at 10:49 AM, GNUbie [EMAIL PROTECTED] wrote: Hello Steve, On Mon, Oct 13, 2008 at 9:52 PM, Steve Totaro [EMAIL PROTECTED] wrote: If you are going to dismiss (the most probable) problem (NAT) without posting configs, I am not sure how much help you will get, you will probably be dismissed as well. What particular configs are you looking for? Below is my current setup and scenario: [snom] ==LAN== [asterisk] ==FXO/POTS == [analog_telephone/mobile_phone] SNOM is using the 192.168.101.102 IP address Asterisk is using 192.168.101.1 IP address for its eth1 interface FXO port is connected to the POTS SNOM doesn't need to go out to the Internet in this scenario, AFAIK. Below is my current NAT rules: # iptables -L -v -t nat Chain PREROUTING (policy ACCEPT 63795 packets, 7162K bytes) pkts bytes target prot opt in out source destination 11460 760K RETURN 0-- anyany 192.168.101.0/24 !192.168.101.0/24 Chain POSTROUTING (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination 11408 757K MASQUERADE 0-- anyeth0192.168.101.0/24 anywhere Chain OUTPUT (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination Please advice if you need more information from me. Regards, GNUbie First, drop firewall/iptables/selinux and try again. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
On Oct 13, 2008, at 9:29 AM, [EMAIL PROTECTED] wrote: IME: One-way audio problems are almost always casued by NAT gateways and/or incorrect NAT settings in sip.conf and/or incorrect IP address or extenal proxy settings in the SIP phone. And reinvite issues in particular. Those have been the only one-way audio problems I've experienced. Setting reinvite=no fixed everything for me. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Norman, On Mon, Oct 13, 2008 at 11:02 PM, Norman Franke [EMAIL PROTECTED] wrote: And reinvite issues in particular. Those have been the only one-way audio problems I've experienced. Setting reinvite=no fixed everything for me. You mean, canreinvite=no? I already have done line on my sip.conf. Thanks. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Steve, On Mon, Oct 13, 2008 at 10:59 PM, Steve Totaro [EMAIL PROTECTED] wrote: First, drop firewall/iptables/selinux and try again. I already turned off the firewall and I don't have SELinux on my system and the problem is still there. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
A packet trace will probably show exactly what is happening. Try: tcpdump -nlXs 8192 -i eth0 port 5060 You should be able to see the SIP information going back and forth and will probably show you that your NAT rules are applying when they shouldn't. I agree with first turning off your firewall and testing... but if that actually solves the problem you need to know why. This should tell why. j On Mon, 13 Oct 2008, GNUbie wrote: Hello Norman, On Mon, Oct 13, 2008 at 11:02 PM, Norman Franke [EMAIL PROTECTED] wrote: And reinvite issues in particular. Those have been the only one-way audio problems I've experienced. Setting reinvite=no fixed everything for me. You mean, canreinvite=no? I already have done line on my sip.conf. Thanks. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *** Handled by Will's new toy *** ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Might be a stretch, but does the Asterisk log show that the call was answered? I had this problem when interfacing * with an NEC system to do call parking pickup. The NEC would never give a dialtone (nor did it give answer supervision) so * never knew the call got picked up so audio only worked one way. I ended up rigging * to force the line to be considered answered with a patch. Daniel On Oct 13, 2008, at 8:57 AM, GNUbie wrote: Hello Steve, On Mon, Oct 13, 2008 at 10:59 PM, Steve Totaro [EMAIL PROTECTED] wrote: First, drop firewall/iptables/selinux and try again. I already turned off the firewall and I don't have SELinux on my system and the problem is still there. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
