[asterisk-users] Dropping incompatible voice frame error
Greetings, We have an old analog phone system running Asterisk 1.2.13 (not my choice lol). Everything has been working wonderfully until today. The site is experiencing dropped and missed calls. When I tried calling the site, I did get through however the CLI was flooded with hundreds of copies of the following: Jan 25 15:46:54 NOTICE[2343]: channel.c:1956 ast_read: Dropping incompatible voice frame on Local/4227@from-sip-a3d8,2 of format ulaw since our native format has changed to slin I tried Googling the error but unfortunately, there were lots of reports of the problem and not a single solution or fix. My thinking is that it has to do with the service provider but I want to do my homework before pointing the finger. Any assistance would be greatly appreciated. Thanks! Kevin Oravits Phone Sys Admin/Tech Admin -- _ -- 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] Dropping incompatible voice frame on DAHDI/i1/xxxxxxx of format slin since our native format has changed to 0x4 (ulaw)
Hey, Sometime i am getting following messaged on asterisk CLI console just wondering what these messages are look like some codec related. [May 31 12:26:14] NOTICE[7349]: channel.c:4074 __ast_read: Dropping incompatible voice frame on DAHDI/i1/2031444389-28e of format slin since our native format has changed to 0x4 (ulaw) -- _ -- 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] Dropping incompatible voice frame
Processor: Intel Dual Core Xeon 3.0GHz - Host: CentOS 5.6 (64 bit) -- Virtualbox 4 (64 bit) --- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3 tail -f full shows the below: [May 19 12:00:53] NOTICE[6821] channel.c: Dropping incompatible voice frame on SIP/voxbone.com-0139 of format ulaw since our native format has changed to 0x8 (alaw) [May 19 12:01:05] NOTICE[6827] channel.c: Dropping incompatible voice frame on SIP/4420-013a of format alaw since our native format has changed to 0x4 (ulaw) I am confused... In the first line, it says native format has changed to alaw and next line it says native format has changed to ulaw... Thx Sanjay -- _ -- 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] Dropping incompatible voice frame
For 2 different hosts. SIP/voxbone.com and SIP/4420 From: RSCL Mumbai Sent: Thu 5/19/2011 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropping incompatible voice frame Processor: Intel Dual Core Xeon 3.0GHz - Host: CentOS 5.6 (64 bit) -- Virtualbox 4 (64 bit) --- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3 tail -f full shows the below: [May 19 12:00:53] NOTICE[6821] channel.c: Dropping incompatible voice frame on SIP/voxbone.com-0139 of format ulaw since our native format has changed to 0x8 (alaw) [May 19 12:01:05] NOTICE[6827] channel.c: Dropping incompatible voice frame on SIP/4420-013a of format alaw since our native format has changed to 0x4 (ulaw) I am confused... In the first line, it says native format has changed to alaw and next line it says native format has changed to ulaw... Thx Sanjay -- _ -- 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] Dropping incompatible voice frame
But why does *our *native format keep changing :) Going by layman terms, if native format is alaw and someone speaks to me in uLaw, I will say *format changed*. But if native format is alaw and someone is talking with me in alaw, I should be happy. On Thu, May 19, 2011 at 10:28 PM, Terry Brummell te...@brummell.net wrote: For 2 different hosts. SIP/voxbone.com and SIP/4420 -- *From:* RSCL Mumbai *Sent:* Thu 5/19/2011 12:23 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Dropping incompatible voice frame Processor: Intel Dual Core Xeon 3.0GHz - Host: CentOS 5.6 (64 bit) -- Virtualbox 4 (64 bit) --- Asterisk 1.6.2.13 via 64 bit Elastix 2.0.3 tail -f full shows the below: [May 19 12:00:53] NOTICE[6821] channel.c: Dropping incompatible voice frame on SIP/voxbone.com-0139 of format ulaw since our native format has changed to 0x8 (alaw) [May 19 12:01:05] NOTICE[6827] channel.c: Dropping incompatible voice frame on SIP/4420-013a of format alaw since our native format has changed to 0x4 (ulaw) I am confused... In the first line, it says native format has changed to alaw and next line it says native format has changed to ulaw... Thx Sanjay -- _ -- 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] Dropping incompatible voice frame
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Thursday, May 19, 2011 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dropping incompatible voice frame But why does our native format keep changing :) Going by layman terms, if native format is alaw and someone speaks to me in uLaw, I will say format changed. But if native format is alaw and someone is talking with me in alaw, I should be happy. As far as I can tell this is a bug. I've also experienced similar issues with our 1.8 box, but this is a production box and not easy to gather the needed troubleshooting info. My solution is to make sure no transcoding is going on. -- _ -- 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] Dropping incompatible voice frame
Possibly or possibly not. Most (IMO) calls are placed initially with the choice 2-3 or more codecs. Normally one codec is negotiated and life goes on, but IAX is a little different from a SIP/DAHDI call. The most certain remedy I can think of for this it to just unallow the alaw codec on IAX calls. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Thursday, April 29, 2010 8:52 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dropping incompatible voice frame Hi, What does this message imply? [Apr 29 14:46:30] NOTICE[32175] channel.c: Dropping incompatible voice frame on IAX2/trunk1-9085 of format alaw since our native format has changed to 0x4 (ulaw) If voice frames have been dropped then I suppose that the call quality may be affected? Vieri -- _ -- 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] Dropping incompatible voice frame
Hi, What does this message imply? [Apr 29 14:46:30] NOTICE[32175] channel.c: Dropping incompatible voice frame on IAX2/trunk1-9085 of format alaw since our native format has changed to 0x4 (ulaw) If voice frames have been dropped then I suppose that the call quality may be affected? Vieri -- _ -- 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] Dropping incompatible voice frame error
I have a SIP phone calling an AGI application. It starts out this way: -- Executing [...@macro-call-agi:2] AGI(SIP/151-b414f0c8, computer-temp.sh,darwin,) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/computer-temp.sh Then I get a dozen or so copies of: [Nov 30 22:40:03] NOTICE[28300]: channel.c:2962 __ast_read: Dropping incompatible voice frame on SIP/151-b414f0c8 of format ulaw since our native format has changed to 0x2 (gsm) and then it's fine: -- Playing '/var/lib/asterisk/agi-bin/machine_darwin_prefix' (escape_digits=) (sample_offset 0) -- AGI Script Executing Application: (Playback) Options: (/tmp/say_agi28305.2) -- SIP/151-b414f0c8 Playing '/tmp/say_agi28305.2.ulaw' (language 'en') Searching for that error on the web produces stuff that looks unrelated. Does anybody know what could be going on here? ___ -- 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] Dropping incompatible voice frame error
Richard Kenner wrote: I have a SIP phone calling an AGI application. It starts out this way: -- Executing [...@macro-call-agi:2] AGI(SIP/151-b414f0c8, computer-temp.sh,darwin,) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/computer-temp.sh Then I get a dozen or so copies of: [Nov 30 22:40:03] NOTICE[28300]: channel.c:2962 __ast_read: Dropping incompatible voice frame on SIP/151-b414f0c8 of format ulaw since our native format has changed to 0x2 (gsm) and then it's fine: -- Playing '/var/lib/asterisk/agi-bin/machine_darwin_prefix' (escape_digits=) (sample_offset 0) -- AGI Script Executing Application: (Playback) Options: (/tmp/say_agi28305.2) -- SIP/151-b414f0c8 Playing '/tmp/say_agi28305.2.ulaw' (language 'en') Searching for that error on the web produces stuff that looks unrelated. Does anybody know what could be going on here? What version of Asterisk are you running? This sounds similar to an issue with AGI's I saw a while ago, but I can't quite remember exactly what the issue (or issue number) was. Updating to the latest version from subversion in the 1.4 (or whatever other branch you're using) may help. Leif Madsen. ___ -- 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] Dropping incompatible voice frame error
What version of Asterisk are you running? This sounds similar to an issue with AGI's I saw a while ago, but I can't quite remember exactly what the issue (or issue number) was. 1.6.2.0-rc2 ___ -- 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] Dropping incompatible voice frame
Thanks, placing: Disallow=all Allow=ulaw In the specific iaxy device context fixed it. I had always thought that allowing all possible valid codecs under the general context would work and the devices would sort it out upon handshake. Guess not. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven J. Douglas Sent: Wednesday, January 28, 2009 9:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dropping incompatible voice frame Don't use g729 in the iax.conf for the IAXY device. It doesn't support it. Regards, Steve Adam Robins wrote: I am using a Polycom SIP phone (ext 2042) to call an analog phone connected via an IAXY (ext 2120). The analog phone rings, and when I answer, I can hear the person speaking on the SIP phone, but they cannot hear me. However, if I originate the call from the analog phone to the SIP phone, it works just fine. In SIP.conf: Disallow=all Allow=g729 Allow=ulaw Canreinvite=no In IAX.conf: Disallow=all Allow=ulaw Allow=g729 Transfer=no Codecpriority=host CLI shows: [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- Executing [2...@international:1] Dial(SIP/2042-b7b0cc88, IAX2/2120|12|oWwtT) in new stack [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- Called 2120 [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Call accepted by 192.168.2.61 (format ulaw) [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Format for call is ulaw [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- IAX2/2120-3849 is ringing [Jan 28 16:04:33] VERBOSE[3428] logger.c: [Jan 28 16:04:33] -- IAX2/2120-3849 answered SIP/2042-b7b0cc88 [Jan 28 16:04:33] NOTICE[3428] channel.c: Dropping incompatible voice frame on IAX2/2120-3849 of format g729 since our native format has changed to 0x4 (ulaw) [Jan 28 16:04:41] VERBOSE[3428] logger.c: [Jan 28 16:04:41] -- Hungup 'IAX2/2120-3849' This is Asterisk 1.4.22, but it also happened on 1.2.4. If I call an IAX2/ulaw softphone from the SIP phone, it works fine. Could it be something in the IAXY provisioning? Any ideas are appreciated. Thanks. ___ -- 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 ___ -- 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] Dropping incompatible voice frame
I am using a Polycom SIP phone (ext 2042) to call an analog phone connected via an IAXY (ext 2120). The analog phone rings, and when I answer, I can hear the person speaking on the SIP phone, but they cannot hear me. However, if I originate the call from the analog phone to the SIP phone, it works just fine. In SIP.conf: Disallow=all Allow=g729 Allow=ulaw Canreinvite=no In IAX.conf: Disallow=all Allow=ulaw Allow=g729 Transfer=no Codecpriority=host CLI shows: [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- Executing [2...@international:1] Dial(SIP/2042-b7b0cc88, IAX2/2120|12|oWwtT) in new stack [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- Called 2120 [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Call accepted by 192.168.2.61 (format ulaw) [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Format for call is ulaw [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- IAX2/2120-3849 is ringing [Jan 28 16:04:33] VERBOSE[3428] logger.c: [Jan 28 16:04:33] -- IAX2/2120-3849 answered SIP/2042-b7b0cc88 [Jan 28 16:04:33] NOTICE[3428] channel.c: Dropping incompatible voice frame on IAX2/2120-3849 of format g729 since our native format has changed to 0x4 (ulaw) [Jan 28 16:04:41] VERBOSE[3428] logger.c: [Jan 28 16:04:41] -- Hungup 'IAX2/2120-3849' This is Asterisk 1.4.22, but it also happened on 1.2.4. If I call an IAX2/ulaw softphone from the SIP phone, it works fine. Could it be something in the IAXY provisioning? Any ideas are appreciated. Thanks. ___ -- 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] Dropping incompatible voice frame
IMO it is a bridging problem. The evidence of this is: SIP - Analog - no outgoing audio connection Analog - SIP (actually Analog - * - SIP - everything ok. Try putting an Answer() in front of Dial() in your dialplan (extensions.conf) and see if this goes away. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Wednesday, January 28, 2009 4:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropping incompatible voice frame I am using a Polycom SIP phone (ext 2042) to call an analog phone connected via an IAXY (ext 2120). The analog phone rings, and when I answer, I can hear the person speaking on the SIP phone, but they cannot hear me. However, if I originate the call from the analog phone to the SIP phone, it works just fine. In SIP.conf: Disallow=all Allow=g729 Allow=ulaw Canreinvite=no In IAX.conf: Disallow=all Allow=ulaw Allow=g729 Transfer=no Codecpriority=host CLI shows: [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- Executing [2...@international:1] Dial(SIP/2042-b7b0cc88, IAX2/2120|12|oWwtT) in new stack [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- Called 2120 [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Call accepted by 192.168.2.61 (format ulaw) [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Format for call is ulaw [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- IAX2/2120-3849 is ringing [Jan 28 16:04:33] VERBOSE[3428] logger.c: [Jan 28 16:04:33] -- IAX2/2120-3849 answered SIP/2042-b7b0cc88 [Jan 28 16:04:33] NOTICE[3428] channel.c: Dropping incompatible voice frame on IAX2/2120-3849 of format g729 since our native format has changed to 0x4 (ulaw) [Jan 28 16:04:41] VERBOSE[3428] logger.c: [Jan 28 16:04:41] -- Hungup 'IAX2/2120-3849' This is Asterisk 1.4.22, but it also happened on 1.2.4. If I call an IAX2/ulaw softphone from the SIP phone, it works fine. Could it be something in the IAXY provisioning? Any ideas are appreciated. Thanks. ___ -- 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
Re: [asterisk-users] Dropping incompatible voice frame
Don't use g729 in the iax.conf for the IAXY device. It doesn't support it. Regards, Steve Adam Robins wrote: I am using a Polycom SIP phone (ext 2042) to call an analog phone connected via an IAXY (ext 2120). The analog phone rings, and when I answer, I can hear the person speaking on the SIP phone, but they cannot hear me. However, if I originate the call from the analog phone to the SIP phone, it works just fine. In SIP.conf: Disallow=all Allow=g729 Allow=ulaw Canreinvite=no In IAX.conf: Disallow=all Allow=ulaw Allow=g729 Transfer=no Codecpriority=host CLI shows: [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- Executing [2...@international:1] Dial(SIP/2042-b7b0cc88, IAX2/2120|12|oWwtT) in new stack [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- Called 2120 [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Call accepted by 192.168.2.61 (format ulaw) [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Format for call is ulaw [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- IAX2/2120-3849 is ringing [Jan 28 16:04:33] VERBOSE[3428] logger.c: [Jan 28 16:04:33] -- IAX2/2120-3849 answered SIP/2042-b7b0cc88 [Jan 28 16:04:33] NOTICE[3428] channel.c: Dropping incompatible voice frame on IAX2/2120-3849 of format g729 since our native format has changed to 0x4 (ulaw) [Jan 28 16:04:41] VERBOSE[3428] logger.c: [Jan 28 16:04:41] -- Hungup 'IAX2/2120-3849' This is Asterisk 1.4.22, but it also happened on 1.2.4. If I call an IAX2/ulaw softphone from the SIP phone, it works fine. Could it be something in the IAXY provisioning? Any ideas are appreciated. Thanks. ___ -- 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
[asterisk-users] Dropping Incompatible Voice Frame
Having a problem here that I can't seem to find a fix for. PSTN call comes in, operator answers, transfers call to a phone behind an IAXy. Caller hears no sound after being transferred. IAXy can hear caller, but not vice versa. Client reads: NOTICE[11342]: channel.c:1950 ast_read: Dropping incompatible voice frame on IAX2/jim-7 of format gsm since our native format has changed to ulaw Not sure how to proceed. Please advise. Thanks, Jay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dropping incompatible voice frame
This didnt work for me either. I tried using the patch at the link below and it didnt work either. If I were to guess what was happening here, it would be when the call is forwarded by the phone Asterisk doesnt know which device to send the call to. How does it know to open a Zap channel and dial the command? What tells Asterisk to open Zap channel and dial the number the phone had it its forward? Am I off track here? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail Sent: Wednesday, June 28, 2006 10:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dropping incompatible voice frame This is known issue, we fixed it by putting an answer() in the dial plan before it gets forwarded,thefix transcode_via_sln=no (detailed in the bug tracker)didn't work for me. YMMV. http://bugs.digium.com/view.php?id=4101 On 6/28/06, Kevin Savoy [EMAIL PROTECTED] wrote: Sorry if this has been posted before but I'm having an issue where I get the following on my CLI. ast_read: Dropping incompatible voice frame on Local/XX of format ulaw since our native form has changed to slin A call comes in on our main to toll free number on an ATT T1 line and is sent to phone 4000. This is our secretary's desk. If she leaves the desk she forwards the phone to one of our sister companies so that they would answer the call. This call is sent back out the ATT T1. If she answers the call and then forwards outside the building it works fine but if she forwards her phone outside the building to auto forward the call when she is away from her desk we get the above error. I have recreated this on my own phone (both hers and mine are Polycom 501's) and with a Cisco 7960. I also tried a different toll free number with the same results. I searched the internet and found four people having the same issue but none have gotten responses on how to fix it. Each time it was something similar where the call was redirected. I know the T1's are configured correctly because all other incoming and outgoing calls work fine until this error occurs. Then nothing works. I am using Asterisk 1.2.7.1 with Zaptel 1.2.5 and Libpri 1.2.2 . I have tried using both a digium Wctxxp 4 port and RedFone's Fonebridges and have gotten the same result both ways so the problem is within Asterisk itself. I also tried allow=all in sip.conf as well as specifically listing allow= slin and all other formats to no avail. Also when this happens the channel is no longer usable even though Asterisk thinks it is available. When the next call is placed it times out because that channel has been locked by the above error. The only way out is a complete reboot and reset of all systems. Not good. Any help would be greatly appreciated. If I had hair left I'd be pulling it out about now. Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropping incompatible voice frame
Sorry if this has been posted before but Im having an issue where I get the following on my CLI. ast_read: Dropping incompatible voice frame on Local/XX of format ulaw since our native form has changed to slin A call comes in on our main to toll free number on an ATT T1 line and is sent to phone 4000. This is our secretarys desk. If she leaves the desk she forwards the phone to one of our sister companies so that they would answer the call. This call is sent back out the ATT T1. If she answers the call and then forwards outside the building it works fine but if she forwards her phone outside the building to auto forward the call when she is away from her desk we get the above error. I have recreated this on my own phone (both hers and mine are Polycom 501s) and with a Cisco 7960. I also tried a different toll free number with the same results. I searched the internet and found four people having the same issue but none have gotten responses on how to fix it. Each time it was something similar where the call was redirected. I know the T1s are configured correctly because all other incoming and outgoing calls work fine until this error occurs. Then nothing works. I am using Asterisk 1.2.7.1 with Zaptel 1.2.5 and Libpri 1.2.2. I have tried using both a digium Wctxxp 4 port and RedFones Fonebridges and have gotten the same result both ways so the problem is within Asterisk itself. I also tried allow=all in sip.conf as well as specifically listing allow=slin and all other formats to no avail. Also when this happens the channel is no longer usable even though Asterisk thinks it is available. When the next call is placed it times out because that channel has been locked by the above error. The only way out is a complete reboot and reset of all systems. Not good. Any help would be greatly appreciated. If I had hair left Id be pulling it out about now. Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropping incompatible voice frame
This is known issue, we fixed it by putting an answer() in the dial plan before it gets forwarded,thefix transcode_via_sln=no (detailed in the bug tracker)didn't work for me. YMMV. http://bugs.digium.com/view.php?id=4101 On 6/28/06, Kevin Savoy [EMAIL PROTECTED] wrote: Sorry if this has been posted before but I'm having an issue where I get the following on my CLI. ast_read: Dropping incompatible voice frame on Local/XX of format ulaw since our native form has changed to slin A call comes in on our main to toll free number on an ATT T1 line and is sent to phone 4000. This is our secretary's desk. If she leaves the desk she forwards the phone to one of our sister companies so that they would answer the call. This call is sent back out the ATT T1. If she answers the call and then forwards outside the building it works fine but if she forwards her phone outside the building to auto forward the call when she is away from her desk we get the above error. I have recreated this on my own phone (both hers and mine are Polycom 501's) and with a Cisco 7960. I also tried a different toll free number with the same results. I searched the internet and found four people having the same issue but none have gotten responses on how to fix it. Each time it was something similar where the call was redirected. I know the T1's are configured correctly because all other incoming and outgoing calls work fine until this error occurs. Then nothing works. I am using Asterisk 1.2.7.1 with Zaptel 1.2.5 and Libpri 1.2.2 . I have tried using both a digium Wctxxp 4 port and RedFone's Fonebridges and have gotten the same result both ways so the problem is within Asterisk itself. I also tried allow=all in sip.conf as well as specifically listing allow= slin and all other formats to no avail. Also when this happens the channel is no longer usable even though Asterisk thinks it is available. When the next call is placed it times out because that channel has been locked by the above error. The only way out is a complete reboot and reset of all systems. Not good. Any help would be greatly appreciated. If I had hair left I'd be pulling it out about now. Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropping incompatible voice frame
Unfortunately setting Redirection to never did not change anything. I am still getting the errors, but now they are no longer isolated to one phone, I am getting them on several extensions. Once again here is a copy of the error: Jan 24 17:45:59 NOTICE[5161]: channel.c:1903 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to alaw Jan 24 17:45:59 NOTICE[5161]: channel.c:1903 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to alaw Jan 24 17:45:59 NOTICE[5161]: channel.c:1903 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to alaw Jan 24 17:45:59 NOTICE[5161]: channel.c:1903 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to alaw These errors generally last anywhere from 6 - 10 secs. Sip.conf specifies only g729 as the available codecs. There are plenty of licenses as well. The SNOM 320s are set to use g729 and then alaw. I would appreciate any help. Joe -- End of Forwarded Message ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropping incompatible voice frame
Joe Sorry I did not get back to you on this. Set your redirect event to never. Andrew On 1/19/06, Joseph Rothstein [EMAIL PROTECTED] wrote: I am now getting these messages on a second box running a different version of Asterisk. If anyone has any idea what is causing these, or how to avoid them I would be very grateful. 157 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 158 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 159 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 160 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 161 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 162 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 163 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 164 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw Thanks, Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropping incompatible voice frame
I am now getting these messages on a second box running a different version of Asterisk. If anyone has any idea what is causing these, or how to avoid them I would be very grateful. 157 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 158 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 159 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 160 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 161 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 162 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 163 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 164 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw Thanks, Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropping incompatible voice frame
On Thursday 19 January 2006 13:48, Joseph Rothstein wrote: I am now getting these messages on a second box running a different version of Asterisk. If anyone has any idea what is causing these, or how to avoid them I would be very grateful. 157 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw I have had a similar issue but was saying :of format slin since our native format has changed to ulaw whatever. The problem was: wrong configuration of FXO port dialplan(spa3000). Kind of - simultaneous use of PSTN dialplan and Call Forward Settings on User tab... This is just a guess since your info is not enough. benchev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropping incompatible voice frame
I have been getting the following messages on Asterisk for a couple of my client's SNOM phones: 7881 Jan 12 14:52:22 NOTICE[6538] channel.c: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to ulaw ... 8185 Jan 12 14:52:28 NOTICE[6538] channel.c: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to ulaw Messages received for almost 6 secs, generating more than 300 log messages. The next occurrence was an hour later or so, lasting 16 secs, generating more than 800 messages. 8186 Jan 12 15:58:14 NOTICE[6773] channel.c: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to ulaw 8999 Jan 12 15:58:30 NOTICE[6773] channel.c: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to ulaw Sip.conf is setup to use only g729 for these SNOM phones. The SNOMs are also setup to use first g729. G729 licenses are available no problem. There is no error message or anything out of the ordinary preceeding these messages. If any one has any ideas regarding these messages, and what is causing them I would really appreciate a response. Regards to all, Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropping incompatible voice frame
I keep getting console messages like the following: Nov 17 11:22:33 NOTICE[16767]: channel.c:1320 ast_read: Dropping incompatible voice frame on SIP/sipuser-5240 of format ALAW since our native format has changed to ULAW I think this is causing some incoming queue calls to ring on an agents extension, however when they pick up the phone, no one is there and the caller gets taken off onhold music, however they are still shown in the queue (show queues). Sometimes, after maybe 30 seconds the caller is bridged to the agent, and sometimes the channel is dropped. Anyone know why this might be happening? All SIP phones are registered to use ULAW. Cheers, Ben Merrills Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropping incompatible voice frame
Hi: i have a problem.
Mi extensions.conf:
exten = _N.,1,Setvar(VOICEMAILREQ=${EXTEN})
exten = _N.,2,SetAccount(${customer})
exten = _N.,3,SetCDRUserField(${VOICEMAILREQ:1})
exten = _N.,4,ResponseTimeout(5)
exten = _N.,5,Background(ifyou)
exten = _N.,6,Background(silence/1)
exten = _N.,7,Background(ifyou)
exten = _N.,8,Background(silence/5)
exten = _N.,9,Background(ifyou)
exten = _N.,10,Background(silence/5)
exten = _N.,11,Background(adio)
exten = _N.,12,Wait,1
exten = _N.,13,Hangup
but in step 5:
-- Executing BackGround(Local/[EMAIL PROTECTED],1,
ifyou) in new stack
Sep 3 11:59:22 WARNING[14350]: format_wav.c:123
check_header: Does not say fmt
Sep 3 11:59:22 WARNING[14350]: file.c:406
ast_filehelper: Unable to open fd on /opt/asterisk/var/lib/sounds/ifyou.wav
Sep 3 11:59:22 WARNING[14350]: file.c:761
ast_streamfile: Unable to open ifyou (format SLINR): No such file or directory
Sep 3 11:59:22 WARNING[14350]: pbx.c:4484
pbx_builtin_background: ast_streamfile failed on Local/[EMAIL PROTECTED],1
fro ifyou
-- Executing BackGround(Local/[EMAIL PROTECTED],1,
silence/1) in new stack
-- Playing 'silence/1' (language 'en')
== Spawn extension (callout, x, 2)
exited non-zero on 'Local/[EMAIL PROTECTED],2'
Sep 3 11:59:22 NOTICE[14350]: channel.c:1287
ast_read: Dropping incompatible voice frame on IAX2/voiptalk/1 of format SLINR
since our native format has changed to GSM
Then step 7 is ok.
Any help?
Thanks
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