[asterisk-users] Problems playing audio file over a Page
All;
I have a problem that Ive been working on for a while now, but Im
stuck and cant see what the solution is. I have an Asterisk 1.11 server on
a public IP address and have two phones registered from behind a NAT. I can
send a page to/from each phone without a problem. My problem is that if I
play an audio file over a page, the page disconnects after a few seconds (
seven seconds to be exact ).
Im playing the audio file like so:
exten = s,n,Page(${AVAILCHANS},A(demo-congrats,q)
In the CLI Im seeing this:
[2015-03-27 11:40:26.360] Got RTP packet fromX.X.X.X:2256 (type 00, seq
021523, ts 1374867997, len 000160)
[2015-03-27 11:40:26.362] Sent RTP packet to X.X.X.X:2256 (type 00, seq
050875, ts 050560, len 000160)
[2015-03-27 11:40:26.363] WARNING[11325][C-002d]: pbx.c:6709
__ast_pbx_run: Timeout, but no rule 't' or 'e' in context 'scheduledpages'
Where X.X.X.X is the outside IP address where the phones are coming from.
Im seeing the GotóSent messages several hundred times while the audio is
playing. Like I said, simply paging an extension with a human voice works
just fine. Any insight at all would be greatly appreciated.
Thanks much;
John
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[asterisk-users] Problems with audio
Hello, I'm having some problems with a total SIP Asterisk scenario, some extensions when make internal and outgoing calls can't hear very well the other party, not echo, not packet lostthe problem is that the volume seems to be very low...what could be happening? i'm not sure what to check Thanks! -- Salu2 -- _ -- 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] Problems with audio
Have you checked that the codec order on the phone matched the order set on the server? On Wed, 2010-09-15 at 17:04 +0200, Danny Dias wrote: Hello, I'm having some problems with a total SIP Asterisk scenario, some extensions when make internal and outgoing calls can't hear very well the other party, not echo, not packet lostthe problem is that the volume seems to be very low...what could be happening? i'm not sure what to check Thanks! -- Salu2 -- _ -- 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] Problems with audio
Yes my friend...CONFIRMED!!! G729 on both sides 2010/9/15 Ishfaq Malik i...@pack-net.co.uk Have you checked that the codec order on the phone matched the order set on the server? On Wed, 2010-09-15 at 17:04 +0200, Danny Dias wrote: Hello, I'm having some problems with a total SIP Asterisk scenario, some extensions when make internal and outgoing calls can't hear very well the other party, not echo, not packet lostthe problem is that the volume seems to be very low...what could be happening? i'm not sure what to check Thanks! -- Salu2 -- _ -- 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 -- Salu2 -- _ -- 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] Problems with audio
On Wed, Sep 15, 2010 at 6:08 PM, Danny Dias ing.diasda...@gmail.com wrote: Yes my friend...CONFIRMED!!! G729 on both sides If the problem happen with SIP to SIP calls and with the same codec, the problem is inside the phone. Check if you can pump up the volume inside his configuration. What phones are you using? -- -- Adrià Vidal -- _ -- 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] Problems with audio
Hi, On 09/15/2010 04:04 PM, Danny Dias wrote: Hello, I'm having some problems with a total SIP Asterisk scenario, some extensions when make internal and outgoing calls can't hear very well the other party, not echo, not packet lostthe problem is that the volume seems to be very low...what could be happening? i'm not sure what to check I had this problem with an Asterisk setup few months ago. People outside the company/setup would hear people on the Asterisk side very faintly/low volume. Even after pushing the volume up on the phones to max. In my case, upgrading the firmware of the Grandstream phones we were using solved the problem. I don't know if this is your case as well though. Sebastian -- _ -- 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] Problems with audio
Hello Adriá... We are using Linksys 942, softphones Xlite...it's a macro pbx, with almost 1000 users, we've checked the gain and volume on the phones :( 2010/9/15 Adrià Vidal adriavi...@gmail.com On Wed, Sep 15, 2010 at 6:08 PM, Danny Dias ing.diasda...@gmail.comwrote: Yes my friend...CONFIRMED!!! G729 on both sides If the problem happen with SIP to SIP calls and with the same codec, the problem is inside the phone. Check if you can pump up the volume inside his configuration. What phones are you using? -- -- Adrià Vidal -- _ -- 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 -- Salu2 -- _ -- 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] Problems with audio
Thanks Sebastian, It's the same firmware version for all our linksys phones...and we have hundreds of pbx's runnning this firmwares versions without any problem 2010/9/15 Sebastian s...@open-t.co.uk Hi, On 09/15/2010 04:04 PM, Danny Dias wrote: Hello, I'm having some problems with a total SIP Asterisk scenario, some extensions when make internal and outgoing calls can't hear very well the other party, not echo, not packet lostthe problem is that the volume seems to be very low...what could be happening? i'm not sure what to check I had this problem with an Asterisk setup few months ago. People outside the company/setup would hear people on the Asterisk side very faintly/low volume. Even after pushing the volume up on the phones to max. In my case, upgrading the firmware of the Grandstream phones we were using solved the problem. I don't know if this is your case as well though. Sebastian -- _ -- 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 -- Salu2 -- _ -- 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] Problems mixing audio in queues and playing queue positions
Hi folks, Over the weekend I finally decided to upgrade one of our Asterisk systems from 1.0.9 to 1.2.4 I had no significant problems and all is well in general - as usual Asterisk rules! However, I did run into two small issues. Can anyone help me solve them please? The first one involves queue position announcements, and the second one is regarding monitor-join. A) In 1.0.9, as soon as a caller enters a queue they are played the position announcement (which is what I want) and then it is replayed every X seconds depending on what I have for announce-frequency in queues.conf This is not the case in 1.2.4 though. Effectively the queue position is not played until after the sum of times set for timeout and retry. e.g. from queues.conf: [myqueue] timeout = 10 retry = 5 wrapuptime=5 maxlen = 0 musiconhold = default strategy = ringall announce-frequency = 60 announce-holdtime = yes announce-round-seconds = 0 monitor-format = wav49 monitor-join = yes member = sip/phone1 member = sip/phone2 member = sip/phone3 With this queues.conf configuration, in 1.2.4 the caller won't get their queue position played until after they have been in the queue for 15 seconds, while in 1.0.9 they got it immediately. Any suggestions? I really think it makes more sense for it to be played immediately when the caller joins the queue rather than waiting for the first timeout, which for many configurations might be much longer than the 15 seconds in mine if timeout and retry are set to higher values. B) My second issue is that monitor-join = yes in queue.conf does not seem to work for me - I still get individual -in and -out files for calls in the queue. Admittedly I had this problem in 1.0.9 too, but not in 1.0.7 I don't think. A very significant bit of information here is that using the m option in Monitor() in extensions.conf does not work for me either (I still get individual -in and -out files). The correct soxmix command gets executed (at least it appears on the console) but does not actually have any effect on the files. Manually running the exact same command on the command line does work, and joins the files correctly, so sox and soxmix are there, and are in the path, and work correctly in theory. Any suggestions would be appreciated! Faris. ___ --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
