[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] Block Specific Number on Inbound
Greetings, Is there a way to block a specific inbound number? I've found code online for blocking all nocallerid and all 800, etc. but nothing for a specific number. My company is wanting me to block a specific number. Is this possible in Asterisk 1.4 and 1.6 or do I need to go through my Service Provider? Thanks, Kevin Oravits Phone Sys 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
Re: [asterisk-users] music on hold
I've noticed on our system the sound files have to be in an exact format for
Asterisk to play them.
Bit Rate: 128kbps
Audio sample size: 16 bit
Channels: 1(mono)
Audio Sample rate: 8kHz
Audio format: PCM
I actually downloaded a program and remixed the audio files to match these
settings. Before that, I couldn't get my Asterisk to play any non-standard
music.
Kevin Oravits
From: Danny Nicholas [mailto:da...@debsinc.com]
Sent: Tuesday, October 04, 2011 11:48 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] music on hold
You have files in /var/lib/asterisk/moh1?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Tuesday, October 04, 2011 12:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] music on hold
i configure new music on hold like below in order to play music for outbond
calls
i want tp play a music until answer form customer
[default1]
mode=files
directory=/var/lib/asterisk/moh1
exten = 0678XX,1,Set(CALLERID(number)=520XX)
exten = 0678XX,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 0678XX,n,Dial(Zap/g1/${EXTEN},,m(default1))
exten = 0678XX,n,Hangup()
when i put the default music i can listen without issue but when i put another
music .wav Or gsm or Mp3
there is no music there is just the ringing
--
_
-- 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] Log for voicemail to email?
I am having a problem with one of my sites where they are not receiving the voicemail to email. I've done a lot of troubleshooting and can't find the issue. It would be helpful if there was a log I could look at so that I could see perhaps where the email is being rejected. Does anyone know of a log that runs on Asterisk that would have this history? I'm running Asterisk 1.6 on CentOS 5.6. The server is not behind a firewall, the Firewall on the box is disabled, SELinux is disabled and I've added the IP to our filters. Oddly, we have the same setup at other sites but this is the only site it is not working at. Any ideas would be great. Thanks, Kevin -- _ -- 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] MOH making calls appear hung up
Greetings, I'm have asterisk servers at about 10 sites, all using Polycom IP 450 phones. With one of my sites, we're having an issue where when a call is transferred, the MOH is not playing and all the caller is hearing is silence. The caller of course thinks they have been hung up on, but the call is actually still in progress and gets successfully transferred if they wait until the person answers. I have researched online and even consulted our 3rd party vendor but no one seems to know how to fix it. Anyone have any advice? Any help would be appreciated. Thanks, Stivaro -- _ -- 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] MOH making calls appear hung up
I noticed the CLI shows that the music on hold actually stops for some reason? Here's the output of my CLI: Connected to Asterisk 1.6.2.19 currently running on localhost (pid = 6363) Verbosity is at least 28 -- Executing [s@ivr-boi-ntc-day:3] Answer(SIP/gw1-05d6, ) in new stack -- Executing [s@ivr-boi-ntc-day:4] Wait(SIP/gw1-05d6, 1) in new stack -- Executing [s@ivr-boi-ntc-day:5] Dial(SIP/gw1-05d6, SIP/1021,20) in new stack == Using SIP RTP CoS mark 5 -- Called 1021 -- SIP/1021-05d7 is ringing -- SIP/1021-05d7 answered SIP/gw1-05d6 -- Packet2Packet bridging SIP/gw1-05d6 and SIP/1021-05d7 -- Started music on hold, class 'default', on SIP/gw1-05d6 == Using SIP RTP CoS mark 5 -- Executing [6937@from-sip:1] Macro(SIP/1021-05d8, stdexten,6937,sip/6937) in new stack -- Executing [s@macro-stdexten:1] Wait(SIP/1021-05d8, 1) in new stack -- Executing [s@macro-stdexten:2] Dial(SIP/1021-05d8, sip/6937,20) in new stack == Using SIP RTP CoS mark 5 -- Called 6937 -- SIP/6937-05d9 is ringing -- Stopped music on hold on SIP/gw1-05d6 == Spawn extension (ivr-boi-ntc-day, s, 5) exited non-zero on 'SIP/1021-05d8ZOMBIE' -- Nobody picked up in 2 ms -- Executing [s@macro-stdexten:3] Goto(SIP/gw1-05d6, s-NOANSWER,1) in new stack -- Goto (macro-stdexten,s-NOANSWER,1) -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail(SIP/gw1-05d6, 6937,u) in new stack -- SIP/gw1-05d6 Playing '/var/spool/asterisk/voicemail/default/6937/unavail.slin' (language 'en') == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'SIP/gw1-05d6' in macro 'stdexten' == Spawn extension (from-sip, 6937, 1) exited non-zero on 'SIP/gw1-05d6' Thanks! Kevin Oravits From: Danny Nicholas [mailto:da...@debsinc.com] Sent: Tuesday, August 30, 2011 11:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] MOH making calls appear hung up It seems a reasonable likelihood that your moh at the offending site does not match the codec of the call (IE your MOH is wav and your call codec is SLIN). Set your verbosity and debug up to 15 and try a call to verify this. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits Sent: Tuesday, August 30, 2011 1:53 PM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] MOH making calls appear hung up Greetings, I'm have asterisk servers at about 10 sites, all using Polycom IP 450 phones. With one of my sites, we're having an issue where when a call is transferred, the MOH is not playing and all the caller is hearing is silence. The caller of course thinks they have been hung up on, but the call is actually still in progress and gets successfully transferred if they wait until the person answers. I have researched online and even consulted our 3rd party vendor but no one seems to know how to fix it. Anyone have any advice? Any help would be appreciated. Thanks, Stivaro -- _ -- 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] MOH making calls appear hung up
Thanks Danny. I tried that but all that did is make it so when I call the site,
I get hold music instead of ringing. Still has no affect on the call transfer
MOH. :/
Interestingly, the music is playing for about 3-5 seconds before stopping
during the transfer.
I've built all of my phone servers the same at my sites and I'm still sorta
green on some of this stuff. When doing a call transfer, is it using the
macro-stdexten or does it go to the IVR dial plan? Because the entry you noted
is in the IVR dialplan, not the macro-stdexten.
Here's my macro-stdexten:
[macro-stdexten]
exten = s,1,wait(1)
exten = s,2,Dial(${ARG2},20)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s,4,Dial(${ARG2},15) ; Ring phone for 15 seconds
exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy
announce
exten = s-BUSY,2,Hangup
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail
exten = s-NOANSWER,2,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten = a,1,VoicemailMain(${CALLERIDNUM})
exten = a,2,Hangup
Here's my IVR Dialplan:
exten = s,1,Set(TIMEOUT(digit)=4)
exten = s,n,Wait(1)
exten = s,n,Answer
exten = s,n,Dial(SIP/1021,20)
exten = s,n,Wait(1)
exten = s,n,Dial(SIP/6909,15)
exten = s,n,Dial(SIP/6904,15)
exten = s,n,Voicemail(1021,u)
exten = s,n,Hangup
Note: I removed the ,m because it was only affecting the new incoming calls.
Here's my CLI output:
== Using SIP RTP CoS mark 5
-- Executing [1021@from-PRI:1] Goto(SIP/gw1-0066, ivr-boi-ntc,s,1)
in new stack
-- Goto (ivr-boi-ntc,s,1)
-- Executing [s@ivr-boi-ntc:1] GotoIfTime(SIP/gw1-0066,
07:00-17:00,mon-fri,*,*?ivr-boi-ntc-day,s,1) in new stack
-- Goto (ivr-boi-ntc-day,s,1)
-- Executing [s@ivr-boi-ntc-day:1] Set(SIP/gw1-0066,
TIMEOUT(digit)=4) in new stack
-- Digit timeout set to 4.000
-- Executing [s@ivr-boi-ntc-day:2] Wait(SIP/gw1-0066, 1) in new
stack
-- Executing [s@ivr-boi-ntc-day:3] Answer(SIP/gw1-0066, ) in new
stack
-- Executing [s@ivr-boi-ntc-day:4] Dial(SIP/gw1-0066,
SIP/1021,20,m) in new stack
== Using SIP RTP CoS mark 5
-- Called 1021
-- Started music on hold, class 'default', on SIP/gw1-0066
-- SIP/1021-0067 is ringing
-- SIP/1021-0067 answered SIP/gw1-0066
-- Stopped music on hold on SIP/gw1-0066
-- Packet2Packet bridging SIP/gw1-0066 and SIP/1021-0067
== Using SIP RTP CoS mark 5
-- Executing [12086597642@from-sip:1] Set(SIP/6908-0068,
CALLERID(num)=2084331021) in new stack
-- Executing [12086597642@from-sip:2] Dial(SIP/6908-0068,
SIP/gw1/12086597642,60) in new stack
== Using SIP RTP CoS mark 5
-- Called gw1/12086597642
-- SIP/gw1-0069 is ringing
-- Started music on hold, class 'default', on SIP/gw1-0066
-- SIP/gw1-0069 is making progress passing it to SIP/6908-0068
== Using SIP RTP CoS mark 5
-- Executing [6911@from-sip:1] Macro(SIP/1021-006a,
stdexten,6911,sip/6911) in new stack
-- Executing [s@macro-stdexten:1] Wait(SIP/1021-006a, 1) in new
stack
-- Executing [s@macro-stdexten:2] Dial(SIP/1021-006a, sip/6911,20)
in new stack
== Using SIP RTP CoS mark 5
-- Called 6911
-- SIP/6911-006b is ringing
-- Stopped music on hold on SIP/gw1-0066
== Spawn extension (ivr-boi-ntc-day, s, 4) exited non-zero on
'SIP/1021-006aZOMBIE'
== Spawn extension (macro-stdexten, s, 2) exited non-zero on
'SIP/gw1-0066' in macro 'stdexten'
== Spawn extension (from-sip, 6911, 1) exited non-zero on 'SIP/gw1-0066'
-- SIP/gw1-0069 answered SIP/6908-0068
-- Packet2Packet bridging SIP/6908-0068 and SIP/gw1-0069
Thanks,
Kevin Oravits
From: Danny Nicholas [mailto:da...@debsinc.com]
Sent: Tuesday, August 30, 2011 12:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up
Your Dial command stops the MOH - if the command were Dial(SIP/1021,20,m) the
music would continue until connected or timed-out.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 2:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up
I noticed the CLI shows that the music on hold actually stops for some reason?
Here's the output of my CLI:
Connected to Asterisk 1.6.2.19 currently running on localhost (pid = 6363)
Verbosity is at least 28
-- Executing [s@ivr-boi-ntc-day:3] Answer(SIP/gw1-05d6, ) in new
stack
-- Executing [s@ivr-boi-ntc-day:4] Wait(SIP/gw1-05d6, 1) in new
stack
-- Executing [s@ivr-boi-ntc-day:5] Dial(SIP/gw1-05d6, SIP/1021,20)
in new stack
== Using SIP RTP CoS mark 5
-- Called 1021
-- SIP/1021-05d7 is ringing
-- SIP/1021-05d7 answered SIP/gw1-05d6
