Re: [asterisk-users] IAX2 bridge failing
No - but this is a new setup so I can't say it worked before...it just isn't working from the start. I've found the call setup works and once bridged there is one way audio (to the ATA, none from the ATA). And the the connection drops after 30 secs approx because something on the path (or endpoint) realizes something is wrong... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Davis [stda...@multiservice.com] Sent: Sunday, December 15, 2013 12:41 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Did you change your network switch recently? Some Digium IAX ATAs do not behave well with Cisco equipment. On Sat, Dec 14, 2013 at 10:26 PM, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: meant to say restart didn't help either.. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis [mdup...@ocg.camailto:mdup...@ocg.ca] Sent: Saturday, December 14, 2013 11:20 PM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Ok just restart -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Friday, December 13, 2013 11:46 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing I tried transfer=no, transfer=yer, and transfer=mediaonly (with a reload inbetween)same result I agree it sounds like something either end is using the wrong IP/port address somewhere in the call (yet signalling works fine). Anything else to suggest? I was hoping for an externalip type setting but not in iax2 (at least not in 1.4.x.x) From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp [jc...@digium.commailto:jc...@digium.com] Sent: Friday, December 13, 2013 11:44 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Michelle Dupuis wrote: Some more details...I noticed that the call is bridged, and audio goes one way. However, the dial command still times out after 35 seconds (approx), and exists non-zero. While the channels are up, I did an core show channel xxx and found Blocking in: ast_waitfor_nandfds Is this a bug? Or something I can fix through config? Hola, Set transfer=no under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like IAX2 native transfers. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.comhttp://www.digium.com www.asterisk.orghttp://www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 -- Steven Davis VoIP Engineer Multi Service +1-913-663-9748 o +1-913-871-5155 m stda...@multiservice.commailto:stda...@multiservice.com [http://www.multiservice.com/assets/images/logos/ms_email_no_tagline.png]http
Re: [asterisk-users] IAX2 bridge failing
Ok just restart -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Friday, December 13, 2013 11:46 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing I tried transfer=no, transfer=yer, and transfer=mediaonly (with a reload inbetween)same result I agree it sounds like something either end is using the wrong IP/port address somewhere in the call (yet signalling works fine). Anything else to suggest? I was hoping for an externalip type setting but not in iax2 (at least not in 1.4.x.x) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp [jc...@digium.com] Sent: Friday, December 13, 2013 11:44 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Michelle Dupuis wrote: Some more details...I noticed that the call is bridged, and audio goes one way. However, the dial command still times out after 35 seconds (approx), and exists non-zero. While the channels are up, I did an core show channel xxx and found Blocking in: ast_waitfor_nandfds Is this a bug? Or something I can fix through config? Hola, Set transfer=no under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like IAX2 native transfers. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] IAX2 bridge failing
meant to say restart didn't help either.. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis [mdup...@ocg.ca] Sent: Saturday, December 14, 2013 11:20 PM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Ok just restart -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Friday, December 13, 2013 11:46 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing I tried transfer=no, transfer=yer, and transfer=mediaonly (with a reload inbetween)same result I agree it sounds like something either end is using the wrong IP/port address somewhere in the call (yet signalling works fine). Anything else to suggest? I was hoping for an externalip type setting but not in iax2 (at least not in 1.4.x.x) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp [jc...@digium.com] Sent: Friday, December 13, 2013 11:44 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Michelle Dupuis wrote: Some more details...I noticed that the call is bridged, and audio goes one way. However, the dial command still times out after 35 seconds (approx), and exists non-zero. While the channels are up, I did an core show channel xxx and found Blocking in: ast_waitfor_nandfds Is this a bug? Or something I can fix through config? Hola, Set transfer=no under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like IAX2 native transfers. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] IAX2 bridge failing
Did you change your network switch recently? Some Digium IAX ATAs do not behave well with Cisco equipment. On Sat, Dec 14, 2013 at 10:26 PM, Michelle Dupuis mdup...@ocg.ca wrote: meant to say restart didn't help either.. From: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis [ mdup...@ocg.ca] Sent: Saturday, December 14, 2013 11:20 PM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Ok just restart -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Friday, December 13, 2013 11:46 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing I tried transfer=no, transfer=yer, and transfer=mediaonly (with a reload inbetween)same result I agree it sounds like something either end is using the wrong IP/port address somewhere in the call (yet signalling works fine). Anything else to suggest? I was hoping for an externalip type setting but not in iax2 (at least not in 1.4.x.x) From: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp [ jc...@digium.com] Sent: Friday, December 13, 2013 11:44 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Michelle Dupuis wrote: Some more details...I noticed that the call is bridged, and audio goes one way. However, the dial command still times out after 35 seconds (approx), and exists non-zero. While the channels are up, I did an core show channel xxx and found Blocking in: ast_waitfor_nandfds Is this a bug? Or something I can fix through config? Hola, Set transfer=no under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like IAX2 native transfers. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 -- *Steven Davis* VoIP Engineer Multi Service +1-913-663-9748 o +1-913-871-5155 m stda...@multiservice.com http://www.multiservice.com/ -- -- This email is intended solely for the use of the addressee and may contain information that is confidential, proprietary, or both. If you receive this email in error please immediately notify the sender and delete the email.. -- -- _ -- 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] IAX2 bridge failing
Michelle Dupuis wrote: Some more details...I noticed that the call is bridged, and audio goes one way. However, the dial command still times out after 35 seconds (approx), and exists non-zero. While the channels are up, I did an core show channel xxx and found Blocking in: ast_waitfor_nandfds Is this a bug? Or something I can fix through config? Hola, Set transfer=no under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like IAX2 native transfers. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 bridge failing
I tried transfer=no, transfer=yer, and transfer=mediaonly (with a reload inbetween)same result I agree it sounds like something either end is using the wrong IP/port address somewhere in the call (yet signalling works fine). Anything else to suggest? I was hoping for an externalip type setting but not in iax2 (at least not in 1.4.x.x) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp [jc...@digium.com] Sent: Friday, December 13, 2013 11:44 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Michelle Dupuis wrote: Some more details...I noticed that the call is bridged, and audio goes one way. However, the dial command still times out after 35 seconds (approx), and exists non-zero. While the channels are up, I did an core show channel xxx and found Blocking in: ast_waitfor_nandfds Is this a bug? Or something I can fix through config? Hola, Set transfer=no under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like IAX2 native transfers. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 bridge failing
Some more details...I noticed that the call is bridged, and audio goes one way. However, the dial command still times out after 35 seconds (approx), and exists non-zero. While the channels are up, I did an core show channel xxx and found Blocking in: ast_waitfor_nandfds Is this a bug? Or something I can fix through config? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis [mdup...@ocg.ca] Sent: Thursday, December 12, 2013 5:08 PM To: Asterisk Users List Subject: [asterisk-users] IAX2 bridge failing I am trying to connect an IAX ATA to an Asterisk 1.4.21.2 system. The Asterisk system has been stable for years, and has no trouble bridge SIP phone sets to IAX trunks. When I initiate a call from the IAX ATA, something goes wrong.One rare occasion it works fine, but usually there is no audio passed. I have a snippet of the console below. Notice no bridging message...not sure if that's a clue? The dialplan seems to execute properly, and I can watch the destination system which answers the call and starts playing media (monkeys) which I don't hear. Any ideas on what is going on? Since this is IAX in and IAX out, NAT should not be an issue (even through there is NAT on both sides). Since media moves on the same UDP port as call setup, also proves should not be a network problem (I think) Can someone point me to a solution? Thanks! (IP's and ISP and phone number disguised) - Executing [s@macro-dialexternal:57] GotoIf(IAX2/S-14468, 1?dialnormal) in new stack -- Goto (macro-dialexternal,s,60) -- Executing [s@macro-dialexternal:60] Dial(IAX2/S-14468, IAX2/ISP123/1234567890|60|W) in new stack -- Called ISP123/1234567890 -- Call accepted by 201.191.37.138 (format ulaw) -- Format for call is ulaw -- IAX2/ISP123-2261 answered IAX2/S-14468 -- Channel 'IAX2/S-14468' ready to transfer -- Channel 'IAX2/ISP123-2261' ready to transfer -- Hungup 'IAX2/ISP123-2261' -- _ -- 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] IAX2 bridge failing
I am trying to connect an IAX ATA to an Asterisk 1.4.21.2 system. The Asterisk system has been stable for years, and has no trouble bridge SIP phone sets to IAX trunks. When I initiate a call from the IAX ATA, something goes wrong.One rare occasion it works fine, but usually there is no audio passed. I have a snippet of the console below. Notice no bridging message...not sure if that's a clue? The dialplan seems to execute properly, and I can watch the destination system which answers the call and starts playing media (monkeys) which I don't hear. Any ideas on what is going on? Since this is IAX in and IAX out, NAT should not be an issue (even through there is NAT on both sides). Since media moves on the same UDP port as call setup, also proves should not be a network problem (I think) Can someone point me to a solution? Thanks! (IP's and ISP and phone number disguised) - Executing [s@macro-dialexternal:57] GotoIf(IAX2/S-14468, 1?dialnormal) in new stack -- Goto (macro-dialexternal,s,60) -- Executing [s@macro-dialexternal:60] Dial(IAX2/S-14468, IAX2/ISP123/1234567890|60|W) in new stack -- Called ISP123/1234567890 -- Call accepted by 201.191.37.138 (format ulaw) -- Format for call is ulaw -- IAX2/ISP123-2261 answered IAX2/S-14468 -- Channel 'IAX2/S-14468' ready to transfer -- Channel 'IAX2/ISP123-2261' ready to transfer -- Hungup 'IAX2/ISP123-2261' -- _ -- 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
