[asterisk-users] Missing audio on playback in 16.0
Hi, I am currently evaluating asterisk 16. I have noticed an issue using application playback. The beginning and the end of the audio file are missing. If I use answer and wait(1) before playback, the beginning is correct. I am using chan_sip, if this is of interest. Best regards Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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] Missing audio
I have a FreePBX system with PRI trunks that's doing a number of things very nicely, but frustrating me in one area. I am using a Grandstream GXW-4008 in an off-premises location to provide POTS service on four ports (this device worked fine in an early application using a hardware VPN to the Asterisk server). The Grandstream has a public static IP port, as does the Asterisk server. Extensions 1021, 1022, 1023, and 1024 register just fine. A ring group, 1020, distributes calls to these extensions and they handle incoming calls in hunt as I'd expect. Calls on the first port are consistently fine. Calls on the other ports are fine for a day or more, then they lose audio or have one-way audio. One mystery for me is that the first port always continues to work. I've assumed that this is some sort of UDP port problem, but I've Googled and studied stuff on-line and haven't figured out what I should be doing to fix it. I'd really appreciate some help. --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax -- _ -- 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] Missing audio from Zaptel channels - SOLVED!?
For those who are interested, the problem appears to NOT exist in 1.2Beta2. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Rod Bacon wrote: I have cross-posted this all over the place, and sent a copy directly to digium support, in the hope of getting to the bottom of a problem that has me pulling my hair out. I currently have 2 production PSTN gateway servers, running asterisk 1.2beta and TE406P cards (upgraded 405 cards, with hardware echo cancelers that we recently purchased on recommendation). We went to the beta version after installing the cancelers, as 1.0.9 kept segfaulting with the cancelers installed. Our PRIs terminate on a DMS100, at the same premises where our servers are co-located. Also in my farm, I have a dedicated IVR server, a VOIP gateway (SIP/IAX/H.323) and clustered MySQL servers running as FastAGI servers, to remove processor load from the PSTN servers. All servers are connected via gigabit Ethernet, and use IAX trunking for inter-server communications. I have been through _everything_ possible to be sure that I don't have any zaptel timing/irq problems (framebuffer, apic, acpi, smp irq affinity, irq latency, etc. etc) and have good zttest results with no frame slips, pops or clicks. After my PSTN gateway servers have been running for a few hours, I notice that some missing audio creeps into the start of each call (makes no difference if the call is ZAP-ZAP native bridge or ZAP-IAX). At best, you miss the first syllable of the first word. At worst, you can miss the first 3 or 4 seconds of audio. Further investigation shows that asterisk is lagging after the second leg of the call is answered (i.e. the time taken to bridge the channels gets longer). If the resultant call is a Zaptel native bridge, then the remaining audio is fine. If the resultant call is not zaptel natively bridged (eg. call is routed via another server, or asterisk remains in the media stream for another reason) then significant delay exists from one end of the call to another (simply put, asterisk seems to slow down). If I restart asterisk (even without removing and reloading zaptel drivers), calls are OK again for a period (typically around 12 hours). A workaround is to simply to install a cron job that periodically restarts asterisk when it's idle, but this is a less than ideal solution from my perspective. Something is definitely changing over time. A memory leak? Runaway process? I really need help in trying to troubleshoot this, as I've run completely out of both patience and ideas. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Missing audio from Zaptel channels
I have cross-posted this all over the place, and sent a copy directly to digium support, in the hope of getting to the bottom of a problem that has me pulling my hair out. I currently have 2 production PSTN gateway servers, running asterisk 1.2beta and TE406P cards (upgraded 405 cards, with hardware echo cancelers that we recently purchased on recommendation). We went to the beta version after installing the cancelers, as 1.0.9 kept segfaulting with the cancelers installed. Our PRIs terminate on a DMS100, at the same premises where our servers are co-located. Also in my farm, I have a dedicated IVR server, a VOIP gateway (SIP/IAX/H.323) and clustered MySQL servers running as FastAGI servers, to remove processor load from the PSTN servers. All servers are connected via gigabit Ethernet, and use IAX trunking for inter-server communications. I have been through _everything_ possible to be sure that I don't have any zaptel timing/irq problems (framebuffer, apic, acpi, smp irq affinity, irq latency, etc. etc) and have good zttest results with no frame slips, pops or clicks. After my PSTN gateway servers have been running for a few hours, I notice that some missing audio creeps into the start of each call (makes no difference if the call is ZAP-ZAP native bridge or ZAP-IAX). At best, you miss the first syllable of the first word. At worst, you can miss the first 3 or 4 seconds of audio. Further investigation shows that asterisk is lagging after the second leg of the call is answered (i.e. the time taken to bridge the channels gets longer). If the resultant call is a Zaptel native bridge, then the remaining audio is fine. If the resultant call is not zaptel natively bridged (eg. call is routed via another server, or asterisk remains in the media stream for another reason) then significant delay exists from one end of the call to another (simply put, asterisk seems to slow down). If I restart asterisk (even without removing and reloading zaptel drivers), calls are OK again for a period (typically around 12 hours). A workaround is to simply to install a cron job that periodically restarts asterisk when it's idle, but this is a less than ideal solution from my perspective. Something is definitely changing over time. A memory leak? Runaway process? I really need help in trying to troubleshoot this, as I've run completely out of both patience and ideas. -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
