Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
On 12/14/2013 01:29 AM, Martin wrote: If I need to use SIP, from where to get the suitable firmware for these Cisco IP Phones 7942G? Be careful, not all versions of SIP firmware work with asterisk. I do have 8-3-1 (cmterm-7941_7961-sip.8-3-1)here and it works just fine with my 7961. Downloaded somewhere. Version 9.x is broken, SIP only works over TCP. I thought that was fixed in the latest 9.x? Where do u download the SIP firmware usually for your Cisco IP Phones? I have a 7961 and just registered at cisco.com then logged in, did a search and was offered the firmware files for free. Regards, Patrick -- _ -- 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] Why doesn't Asterisk try to prevent transcoding
Let's say I have two devices configured and the follow call scenarios occur. [100] disallow=all allow=g722ulaw Polycom phone with g722,ulaw,alaw,g729 [101] disallow=all allow=ulaw Polycom phone with g722,ulaw,alaw,g729 101 dials 100 - ulaw to ulaw is chosen 100 dials 101 - g722 to ulaw is chosen Ideally when 100 dials 101 ulaw would be chosen since it is the common format. Looking into this deeper Device 100 sends INVITE to Asterisk offering g722,ulaw,alaw,g729 Asterisk sends INVITE to device 101 offering ulaw Device 101 sends 200 OK to Asterisk offering ulaw Asterisk sends 200 OK to device 100 offering g722,ulaw I can prevent transcoding by adding SIP_CODEC_INBOUND=ulaw to the dialplan for extension 101. This causes Asterisk to send 200 OK to device 100 offering ulaw. Am I missing why Asterisk wouldn't just offer the highest priority codec they have in common to prevent transcoding? Ryan -- _ -- 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] Why doesn't Asterisk try to prevent transcoding
On Sat, Dec 14, 2013 at 10:31 PM, Ryan Wagoner rswago...@gmail.com wrote: Let's say I have two devices configured and the follow call scenarios occur. [100] disallow=all allow=g722ulaw Polycom phone with g722,ulaw,alaw,g729 [101] disallow=all allow=ulaw Polycom phone with g722,ulaw,alaw,g729 101 dials 100 - ulaw to ulaw is chosen 100 dials 101 - g722 to ulaw is chosen Ideally when 100 dials 101 ulaw would be chosen since it is the common format. Looking into this deeper Device 100 sends INVITE to Asterisk offering g722,ulaw,alaw,g729 Asterisk sends INVITE to device 101 offering ulaw Device 101 sends 200 OK to Asterisk offering ulaw Asterisk sends 200 OK to device 100 offering g722,ulaw I can prevent transcoding by adding SIP_CODEC_INBOUND=ulaw to the dialplan for extension 101. This causes Asterisk to send 200 OK to device 100 offering ulaw. Am I missing why Asterisk wouldn't just offer the highest priority codec they have in common to prevent transcoding? Ryan I should have mentioned I'm using Asterisk 11.2-cert2. The core debug from the above shows [2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7911 sip_new: *** Our native formats are (g722) [2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7912 sip_new: *** Joint capabilities are (ulaw|g722) [2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7913 sip_new: *** Our capabilities are (ulaw|g722) [2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7914 sip_new: *** AST_CODEC_CHOOSE formats are g722 [2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7911 sip_new: *** Our native formats are (ulaw) [2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7912 sip_new: *** Joint capabilities are (nothing) [2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7913 sip_new: *** Our capabilities are (ulaw) [2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7914 sip_new: *** AST_CODEC_CHOOSE formats are ulaw [2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7916 sip_new: *** Our preferred formats from the incoming channel are (g722) I'm looking at the code now. I am hoping to write a patch, if I can wrap my head around the code, to determine join capabilities between the joint capabilities of each channel. If this exists then set both channels this codec. Ryan -- _ -- 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
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
