[asterisk-users] DAHDI hangup detection
I am running Asterisk 1.6.1.6 and DAHDI/DAHDI tools 2.2.0.2 compiled from source on a Debian Lenny box, also running FreePBX 2.5.2. I also have an OpenVOX TDM400 card installed with an FXO port on port 1, and an FXS port on port 2. I have a POTS line installed and working on the FXO port. However I've encountered a weird problem that I can't seem to figure out. I have incoming POTS line calls set to ring a SIP extension (a cisco IP phone). It works, but here is the issue: - If an inbound call comes in on the POTS line and hangs up before the call hits voicemail or during the voicemail greeting, Asterisk does not appear to detect this condition. - As a result of this, a voicemail is being left with a dialtone and notifying me As a workaround, I have set busy detection in chan_dahdi.conf and set 12 seconds for a minimum voicemail message in voicemail.conf, but I must be missing something or something is potentially broke. While this workaround works, I don't suspect this behavior I am experiencing is normal and I am having a hard time putting a finger on it. Thanks for any replies. Stephen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI hangup detection
Does linetest.agi exist as part of the Asterisk install? I can't find it
on my box. Also, does this detect dialtone?
Dialtone is being recorded (left as voicemail) without my original
workarounds in place.
On 9/15/09 1:08 PM, Danny Nicholas wrote:
This is how I'd do it with the "old" dialplan (extensions.conf) --
- exten => s,1,dial.
- exten => s,n,Macro(voicemail-test,${EXTEN})
- exten => s,n,hangup
- [macro-voicemail-test]
- Exten => s,1,AGI(linetest.agi,${ARG1})
- Exten => s,n,gotoif($["${VAR}" = "AVAIL"]?hangup)
- Exten => s,n,voicemail...
- Exten => s,n(hangup),hangup
Linetest.agi would run an AMI session to see if DAHDI-X was in use and
return a variable as AVAIL or undef. If the line is in use, record
voicemail, else hangup.
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Stephen Brown Jr
*Sent:* Tuesday, September 15, 2009 11:52 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] DAHDI hangup detection
Ok on the workaround, how would I implement it? I'd like to give that
a shot.
On Tue, Sep 15, 2009 at 12:23, Danny Nicholas <mailto:da...@debsinc.com>> wrote:
The issue is that POTS as a technology does not have Answer/Hangup
Supervision control (This is per the good folks at Digium). Your local
Telco may or may not be of assistance, but a simpler workaround might
be to
hangup on silence or query the line through AMI before transferring to
voicemail.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
<mailto:asterisk-users-boun...@lists.digium.com>
[mailto:asterisk-users-boun...@lists.digium.com
<mailto:asterisk-users-boun...@lists.digium.com>] On Behalf Of Stephen
Brown
Sent: Tuesday, September 15, 2009 11:18 AM
To: asterisk-users@lists.digium.com
<mailto:asterisk-users@lists.digium.com>
Subject: [asterisk-users] DAHDI hangup detection
I am running Asterisk 1.6.1.6 and DAHDI/DAHDI tools 2.2.0.2 compiled
from source on a Debian Lenny box, also running FreePBX 2.5.2. I also
have an OpenVOX TDM400 card installed with an FXO port on port 1, and an
FXS port on port 2. I have a POTS line installed and working on the FXO
port.
However I've encountered a weird problem that I can't seem to figure
out. I have incoming POTS line calls set to ring a SIP extension (a
cisco IP phone). It works, but here is the issue:
- If an inbound call comes in on the POTS line and hangs up before the
call hits voicemail or during the voicemail greeting, Asterisk does not
appear to detect this condition.
- As a result of this, a voicemail is being left with a dialtone and
notifying me
As a workaround, I have set busy detection in chan_dahdi.conf and set 12
seconds for a minimum voicemail message in voicemail.conf, but I must be
missing something or something is potentially broke. While this
workaround works, I don't suspect this behavior I am experiencing is
normal and I am having a hard time putting a finger on it.
Thanks for any replies.
Stephen
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Re: [asterisk-users] DAHDI hangup detection
Following up on this with some testing I have done this afternoon. I have lengthened the default ringtime to 25 seconds from 15, and have tried both kewlstart and loopstart. Kewlstart appears to be the most functional, if I hang up before it hits app_voicemail, it appears to work as intended, and I get this message in the console: == Spawn extension (macro-dial, s, 7) exited non-zero on 'DAHDI/1-1' in macro 'dial' == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'DAHDI/1-1' in macro 'exten-vm' == Spawn extension (from-did-direct, 2001, 1) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' However, if I hangup while the voicemail greeting is playing, then it leaves a 0:34 second message with busy tone and I am notified with the new message. Am I to assume the following? - after 25 seconds, app_voicemail takes over and has no way of knowing the calling party has hung up, therefore recording dial/busy/offhook tone (whatever is present) Is there any conceivable way to fix this? My fear is being bombarded with bogus voice mails if people call the number and decide to hang up while the voice mail greeting is playing. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI disconnect supervision timing
I have, but I wanted to see if I could fix this problem before I started experimenting with that. On 9/25/09 11:24 AM, Danny Nicholas wrote: Have you looked into minimum message length and/or silence parameters? *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Stephen Brown Jr *Sent:* Friday, September 25, 2009 10:20 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] DAHDI disconnect supervision timing Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6 install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1 FXO port and 1 FXS port. I have a POTS line from my phone company attached to the POTS line. I have asked for "disconnect supervision" to be provisioned on my line and they claim to have added it. However, my scenario is as follows: I receive a call, if the caller hangs up before hitting voice mail, the DAHDI channel is released as to be expected (evidenced from console messaging) If the call gets to voicemail and the caller hangs up during the greeting, no hangup condition is ever detected and I am greeted with a useless voicemail moments later. I am using kewlstart signaling etc. I came across this page from Digium: http://kb.digium.com/entry/6/ This suggests adjusting a variable in zaptel.h, as I don't use zaptel, can this same logic be applied to DAHDI somewhere? My theory is that the "disconnect supervision" signal coming from the phone company may be less than 1000ms. Desperately trying to fix this. Thanks, Stephen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI disconnect supervision timing
I've tried the hanguponpolarityswitch parameter as well with no success :( Any clues where in the DAHDI code I might find reference to disconnect supervision timing? On 9/25/09 11:39 AM, Martin wrote: > find the code in dahdi and put printk so you can see in dmesg or > /var/log/messages > if that gets ever detected > > also you may try hanguponpolarityswitch=yes in chan_dahdi.conf > > Martin > > > On Fri, Sep 25, 2009 at 10:19 AM, Stephen Brown Jr > wrote: > >> Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6 >> install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1 >> FXO port and 1 FXS port. I have a POTS line from my phone company attached >> to the POTS line. >> >> I have asked for "disconnect supervision" to be provisioned on my line and >> they claim to have added it. However, my scenario is as follows: >> >> I receive a call, if the caller hangs up before hitting voice mail, the >> DAHDI channel is released as to be expected (evidenced from console >> messaging) >> If the call gets to voicemail and the caller hangs up during the greeting, >> no hangup condition is ever detected and I am greeted with a useless >> voicemail moments later. >> >> I am using kewlstart signaling etc. I came across this page from Digium: >> >> http://kb.digium.com/entry/6/ >> >> This suggests adjusting a variable in zaptel.h, as I don't use zaptel, can >> this same logic be applied to DAHDI somewhere? My theory is that the >> "disconnect supervision" signal coming from the phone company may be less >> than 1000ms. >> >> Desperately trying to fix this. >> >> Thanks, >> Stephen >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> 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 -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI disconnect supervision timing
Have you been able to find a satisfiable config? My biggest headache is the useless voicemails being left if the caller hangs up during the greeting, otherwise it appears to work as intended. Agree on the Zoloft, this is driving me nuts! On 9/25/09 11:45 AM, Danny Nicholas wrote: If you're really going to pursue this, I'd buy stock in Zoloft -- I've got a TDM400 and TDP410 and they both drive me nuts on POTS issues... *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Stephen Brown *Sent:* Friday, September 25, 2009 10:41 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] DAHDI disconnect supervision timing I have, but I wanted to see if I could fix this problem before I started experimenting with that. On 9/25/09 11:24 AM, Danny Nicholas wrote: Have you looked into minimum message length and/or silence parameters? *From:* asterisk-users-boun...@lists.digium.com <mailto:asterisk-users-boun...@lists.digium.com> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Stephen Brown Jr *Sent:* Friday, September 25, 2009 10:20 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] DAHDI disconnect supervision timing Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6 install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1 FXO port and 1 FXS port. I have a POTS line from my phone company attached to the POTS line. I have asked for "disconnect supervision" to be provisioned on my line and they claim to have added it. However, my scenario is as follows: I receive a call, if the caller hangs up before hitting voice mail, the DAHDI channel is released as to be expected (evidenced from console messaging) If the call gets to voicemail and the caller hangs up during the greeting, no hangup condition is ever detected and I am greeted with a useless voicemail moments later. I am using kewlstart signaling etc. I came across this page from Digium: http://kb.digium.com/entry/6/ This suggests adjusting a variable in zaptel.h, as I don't use zaptel, can this same logic be applied to DAHDI somewhere? My theory is that the "disconnect supervision" signal coming from the phone company may be less than 1000ms. Desperately trying to fix this. Thanks, Stephen ___ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- AstriCon 2009 - October 13 - 15Phoenix,Arizona Register Now:http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI disconnect supervision timing
ting [...@macro-vm:5] NoOp("DAHDI/1-1",
"Checking if ext 100 is enabled: ") in new stack
[Sep 25 13:34:56] -- Executing [...@macro-vm:6] GotoIf("DAHDI/1-1",
"1?s-NOANSWER,1") in new stack
[Sep 25 13:34:56] -- Goto (macro-vm,s-NOANSWER,1)
[Sep 25 13:34:56] -- Executing [s-noans...@macro-vm:1]
Macro("DAHDI/1-1", "get-vmcontext,100") in new stack
[Sep 25 13:34:56] -- Executing [...@macro-get-vmcontext:1]
Set("DAHDI/1-1", "VMCONTEXT=default") in new stack
[Sep 25 13:34:56] -- Executing [...@macro-get-vmcontext:2]
GotoIf("DAHDI/1-1", "0?200:300") in new stack
[Sep 25 13:34:56] -- Goto (macro-get-vmcontext,s,300)
[Sep 25 13:34:56] -- Executing [...@macro-get-vmcontext:300]
NoOp("DAHDI/1-1", "") in new stack
[Sep 25 13:34:56] -- Executing [s-noans...@macro-vm:2]
VoiceMail("DAHDI/1-1", "1...@default,u""") in new stack
[Sep 25 13:34:56] -- Playing 'vm-theperson.gsm'
(language 'en')
[Sep 25 13:34:58] -- Playing 'digits/1.gsm' (language 'en')
[Sep 25 13:34:58] -- Playing 'digits/0.gsm' (language 'en')
[Sep 25 13:34:59] -- Playing 'digits/0.gsm' (language 'en')
[Sep 25 13:35:00] -- Playing 'vm-isunavail.gsm'
(language 'en')
[Sep 25 13:35:01] -- Playing 'vm-intro.gsm' (language 'en')
[Sep 25 13:35:07] -- Playing 'beep.gsm' (language 'en')
[Sep 25 13:35:07] -- Recording the message
[Sep 25 13:35:07] -- x=0, open writing:
/var/spool/asterisk/voicemail/default/100/tmp/Tjm8w6 format: wav49,
0x86c7f18
[Sep 25 13:35:07] -- x=1, open writing:
/var/spool/asterisk/voicemail/default/100/tmp/Tjm8w6 format: gsm, 0x865d768
[Sep 25 13:35:07] -- x=2, open writing:
/var/spool/asterisk/voicemail/default/100/tmp/Tjm8w6 format: wav, 0x8683e70
[Sep 25 13:38:05] -- Recording automatically stopped after a silence
of 10 seconds
[Sep 25 13:38:05] -- Playing 'auth-thankyou.gsm'
(language 'en')
[Sep 25 13:38:06] -- Executing [s-noans...@macro-vm:3]
Goto("DAHDI/1-1", "exit-SUCCESS,1") in new stack
[Sep 25 13:38:06] -- Goto (macro-vm,exit-SUCCESS,1)
[Sep 25 13:38:06] -- Executing [exit-succ...@macro-vm:1]
GotoIf("DAHDI/1-1", "0?exit-RETURN,1") in new stack
[Sep 25 13:38:06] -- Executing [exit-succ...@macro-vm:2]
Playback("DAHDI/1-1", "goodbye") in new stack
[Sep 25 13:38:06] -- Playing 'goodbye.ulaw' (language 'en')
[Sep 25 13:38:07] -- Executing [exit-succ...@macro-vm:3]
Hangup("DAHDI/1-1", "") in new stack
[Sep 25 13:38:07] == Spawn extension (macro-vm, exit-SUCCESS, 3)
exited non-zero on 'DAHDI/1-1' in macro 'vm'
[Sep 25 13:38:07] == Spawn extension (macro-exten-vm, s, 18) exited
non-zero on 'DAHDI/1-1' in macro 'exten-vm'
[Sep 25 13:38:07] == Spawn extension (from-did-direct, 100, 1) exited
non-zero on 'DAHDI/1-1'
[Sep 25 13:38:07] -- Hungup 'DAHDI/1-1'
Then I am left with a nice 3 minute long message full of dialtone
followed by the ever famous "howler" tone to indicate an off-hook phone :/
Any clues!!!??? I should also mention I am running FreePBX 2.5.2
Thanks,
Stephen
On 9/25/09 1:28 PM, Tzafrir Cohen wrote:
> On Fri, Sep 25, 2009 at 11:19:39AM -0400, Stephen Brown Jr wrote:
>
>> Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
>> install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
>> FXO port and 1 FXS port. I have a POTS line from my phone company attached
>> to the POTS line.
>>
>> I have asked for "disconnect supervision" to be provisioned on my line and
>> they claim to have added it. However, my scenario is as follows:
>>
>> I receive a call, if the caller hangs up before hitting voice mail, the
>> DAHDI channel is released as to be expected (evidenced from console
>> messaging)
>> If the call gets to voicemail and the caller hangs up during the greeting,
>> no hangup condition is ever detected and I am greeted with a useless
>> voicemail moments later.
>>
> I don't understand this.
>
> Can you enable debug logging and provide log of a call that disconnect
> successfully and a log of a call that fails to disconnect?
>
>
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[asterisk-users] Sending a hook flash to a DAHDI channel
I've got a piece of CPE equipment that has an FXS port that I have tied
to an FXO port on a TDM400 clone card. Normally, if I go off-hook with a
standard telephone connected to it, I get a dialtone. If I dial a digit,
and send a hookflash, the device will provide a dialtone back for the
next available channel on the device.
I'm trying to recreate this same behavior with Asterisk, however, it's
not working. I'm not sure if it's me, the CPE device, Asterisk, my
dialplan code or a combination of all. I am running Asterisk 1.6.0.21,
FreePBX 2.6, and the latest version of DAHDI.
Here's the dialplan logic I am trying to execute, this is in
/etc/asterisk/extensions_custom.conf:
[from-internal-custom]
exten => 1234,1,Dial(DAHDI/2); DAHDI channel 2, FXO
exten => 1234,n,SendDTMF(2) ; I expect to send DTMF digit 2 after
the channel answers?
exten => 1234,n,Wait(1) ; I added a wait statement
exten => 1234,n,Flash() ; Send the hookflash
After the hookflash is when I expect to receive dialtone from the CPE
device, but it does not appear to ever execute the SendDTMF or Flash events:
[Feb 20 15:45:52] -- Starting simple switch on 'DAHDI/4-1'
[Feb 20 15:46:00] -- Executing [1...@from-internal:1]
ResetCDR("DAHDI/4-1", "") in new stack
[Feb 20 15:46:00] -- Executing [1...@from-internal:2]
Dial("DAHDI/4-1", "DAHDI/2") in new stack
[Feb 20 15:46:00] -- Called 2
[Feb 20 15:46:01] WARNING[27502]: chan_dahdi.c:2008 dahdi_enable_ec:
Unable to enable echo cancellation on channel 2 (No such device)
[Feb 20 15:46:01] -- DAHDI/2-1 answered DAHDI/4-1
[Feb 20 15:46:01] -- Native bridging DAHDI/4-1 and DAHDI/2-1
[Feb 20 15:46:06] WARNING[27502]: chan_dahdi.c:2008 dahdi_enable_ec:
Unable to enable echo cancellation on channel 2 (No such device)
[Feb 20 15:46:06] -- Executing [...@from-internal:1]
Macro("DAHDI/4-1", "hangupcall") in new stack
[Feb 20 15:46:06] -- Executing [...@macro-hangupcall:1]
GotoIf("DAHDI/4-1", "1?skiprg") in new stack
[Feb 20 15:46:06] -- Goto (macro-hangupcall,s,4)
[Feb 20 15:46:06] -- Executing [...@macro-hangupcall:4]
GotoIf("DAHDI/4-1", "1?skipblkvm") in new stack
[Feb 20 15:46:06] -- Goto (macro-hangupcall,s,7)
[Feb 20 15:46:06] -- Executing [...@macro-hangupcall:7]
GotoIf("DAHDI/4-1", "1?theend") in new stack
[Feb 20 15:46:06] -- Goto (macro-hangupcall,s,9)
[Feb 20 15:46:06] -- Executing [...@macro-hangupcall:9]
Hangup("DAHDI/4-1", "") in new stack
[Feb 20 15:46:06] == Spawn extension (macro-hangupcall, s, 9) exited
non-zero on 'DAHDI/4-1' in macro 'hangupcall'
[Feb 20 15:46:06] -- Hungup 'DAHDI/2-1'
[Feb 20 15:46:06] == Spawn extension (from-internal, 1234, 2) exited
non-zero on 'DAHDI/4-1'
[Feb 20 15:46:06] -- Hungup 'DAHDI/4-1'
I've also tried it with this variation:
exten => 1234,1,Dial(DAHDI/2/D(2)); DAHDI channel 2, FXO send DTMF
digit 2 before call gets bridged
exten => 1234,n,Wait(1) ; Wait 1 second
exten => 1234,n,Flash() ; Send the hookflash
This yields the results of dialing "2" to the CPE device, again I never
see the Flash event get executed (as evidenced from the console, it's
also possible my verbosity is set too low?)
Any help appreciated
Thanks,
Stephen
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Re: [asterisk-users] Sending a hook flash to a DAHDI channel
> You only get past this point when this call is over. That makes sense now that I think about it. > Can the CO send Flash to the CPE? I am the CO in this case :) I started looking at the "D" option of the dial command, is there any way to send a DTMF digit followed by a hookflash at call origination? For example maybe something like this: exten => 1234,1,Dial(DAHDI/2/D(2,someoptopiontohookflashhere) Or is there an alternative method that I could use for what I am trying to accomplish? -- _ -- 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] Streaming musiconhold via mpg123
At my wits end with this, and can't proceed any further so I'm hoping someone has seen this and can assist. I can not get streaming musiconhold to work with Asterisk. My Asterisk version is 1.8.8.0 and the mpg123 version is 1.9.1, OS is CentOS 5.7. When I call the musiconhold class (default for example) I get nothing but silence. I've exhausted my troubleshooting capabilities at this point, I've tried everything I can think of to include: - a newer version of mpg123, I went with the latest version - verified I could play an MP3 file by itself in Asterisk by using the MP3Player application What does not work, is if I use the mpg123 application for musiconhold to play a standalone file or a streaming source. I seem to be missing something and I just can't quite put a finger on it. -- _ -- 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] Streaming musiconhold via mpg123
On 2/21/2012 3:38 PM, isr...@gmail.com wrote: > There is a bug in up to version 1.8.9 with external moh sources and dahdi > timers Do you have a link to the bug report? I was unable to find anything but it's possible I'm not looking hard enough ;) > Share with us your musiconhold.conf configuration please. Here it is... please excuse the mess, it's been a wild ride so my formatting/commenting has been left in-tact: ; ; Music on hold class definitions ; This is using the new 1.2 config file format, and will not work with 1.0 ; based Asterisk systems ; ; #include musiconhold_custom.conf ; #include musiconhold_additional.conf ;[default] ;mode=custom ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 [test] mode=custom ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000 /var/lib/asterisk/sounds/music/Rolling In The Deep.mp3 I setup a simple 2 digit extension to call the test context and my MP3 file nor my stream will play, and here's something else interesting: If use the MP3Player application to play an MP3, mpg123 spawns and plays it. I came to this conclusion by running ps aux | grep mpg while the song was playing. -- _ -- 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] Streaming musiconhold via mpg123
DAHDI it is are there any known workarounds? I use the FreePBX distro and they are a bit behind, so no telling when they will update. On 2/21/2012 6:45 PM, Israel Gottlieb wrote: > that bug is running since the start of 1.8 and has been fixed in 1.8.9 > > https://issues.asterisk.org/jira/browse/ASTERISK-17474 > > i know it says that after the first time asterisks starts it works but > thats true only if the moh was loaded before the timing > > its a long story but the fix is finally in > > when typing timing test in the cli what timer to get if its dahdi then > thats the probably problem > > > > On Wed, Feb 22, 2012 at 1:34 AM, Stephen Brown > mailto:stephen.brow...@gmail.com>> wrote: > > On 2/21/2012 3:38 PM, isr...@gmail.com <mailto:isr...@gmail.com> > wrote: > > There is a bug in up to version 1.8.9 with external moh sources > and dahdi timers > > Do you have a link to the bug report? I was unable to find > anything but > it's possible I'm not looking hard enough ;) > > > Share with us your musiconhold.conf configuration please. > > Here it is... please excuse the mess, it's been a wild ride so my > formatting/commenting has been left in-tact: > > ; > ; Music on hold class definitions > ; This is using the new 1.2 config file format, and will not work > with 1.0 > ; based Asterisk systems > ; > ; #include musiconhold_custom.conf > ; #include musiconhold_additional.conf > ;[default] > ;mode=custom > ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r > 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 > ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 > http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 > > [test] > mode=custom > ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r > 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 > ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 > http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 > application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000 > /var/lib/asterisk/sounds/music/Rolling In The Deep.mp3 > > I setup a simple 2 digit extension to call the test context and my MP3 > file nor my stream will play, and here's something else > interesting: If > use the MP3Player application to play an MP3, mpg123 spawns and plays > it. I came to this conclusion by running ps aux | grep mpg while the > song was playing. > > -- > _ > -- 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
[asterisk-users] IAX2 over OpenVPN connection.... working but
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 So a friend of mine and I setup a static key based point to point OpenVPN connection from my box to his for the express intent of carrying IAX traffic encrypted. His network on his lan is 172.30.1.0/24 and mine is 10.0.30.0/24. His PBX is located at 172.30.1.48 and mine is at 10.0.30.2. We had an existing working IAX trunk in place prior to the VPN, and after we brought the VPN up we set the host= parameter within Asterisk accordingly on each end to match the local IP's and discovered it did not work. The trunk remained in an UNKNOWN status on each end, even though we could ping each box locally, SSH, and even SIP worked. Here's where I am baffled and I am hoping someone with intricate knowledge of this implementation may be able to explain it to me. What we had to do to get this working was to set the host= parameter to the respective endpoint IP's of the VPN tunnel, 172.10.1.1 in my case, and 172.10.1.2 in his case. Calls flow normally now and we cannot understand how or why. I would have assumed with a destination of either LAN as defined by the routing table it would have left out on the OpenVPN connection by default, and what's even more strange is that IAX is the only protocol that does not appear to function as intended. Any takers? :) -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.17 (MingW32) iEYEARECAAYFAlDE7GcACgkQ3sJXNEncx7is9QCcCciMYFJ7ZXjYxuHC2EYD0PZY waAAniNNx8GuC5To7ajlGR5sYs3yftFK =lcWJ -END PGP SIGNATURE- -- _ -- 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] Installing Asterisk to it's own directory
I'd like to start playing with 1.8, however I don't want to potentially damage anything on my existing 1.6.2 install on my production server. I'd like to test 1.8 against my existing configs leaving my 1.6.2 install untouched. Looking at the output of ./configure --help suggests that it's possible to install Asterisk into another prefix of my choosing, but as this is unfamiliar territory to me I'm not exactly sure how to accomplish this? Ideally, I'd like to just dump the newly compiled 1.8 and all it's dependencies into a standalone directory (say /testing/asterisk or something) and update my init script to point to the new binaries. I also run a Sangoma USB FXO card and DAHDI for a POTS line that I would like to test as well, should it work with the pre-compiled binaries that are already there? (DAHDI, etc) I've never tried this before, and before I potentially break something I'd like to know if it's possible and how to implement it? Thanks, Stephen -- _ -- 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] Installing Asterisk to it's own directory
Thanks... I actually did a ./configure --prefix=/root/asterisk18 and ended up with this: r...@debian-squeeze:~/asterisk18# pwd /root/asterisk18 r...@debian-squeeze:~/asterisk18# ls -al total 32 drwxr-xr-x 8 root root 4096 Nov 19 18:09 . drwx-- 5 root root 4096 Nov 19 18:37 .. drwxr-xr-x 3 root root 4096 Nov 19 18:09 etc drwxr-xr-x 3 root root 4096 Nov 19 18:09 include drwxr-xr-x 3 root root 4096 Nov 19 18:09 lib drwxr-xr-x 2 root root 4096 Nov 19 18:09 sbin drwxr-xr-x 3 root root 4096 Nov 19 18:09 share drwxr-xr-x 6 root root 4096 Nov 19 18:09 var Have I essentially accomplished the same thing by doing it this way? This is in a virtual machine alongside an Asterisk 1.6 install (for testing), I'm still a little gunshy to touch my production box as of yet. but the 1.8 install did work, I was able to make a call to the demo context :) Thanks, Stephen On 11/19/10 10:13 PM, Jose P. Espinal wrote: > Hi Stephen, > > That's what people do when building precompiled packages for certain > distros (along with a few more things). > > I use to do the following when building packages (with a few more options): > > ./configure --prefix=/usr --sysconfdir=/etc > make > make install DESTDIR=/my/destination/directory > > That would create the complete installation structure under > '/my/destination/directory' > > > Regards, > > > > Stephen Brown wrote: >> I'd like to start playing with 1.8, however I don't want to potentially >> damage anything on my existing 1.6.2 install on my production server. >> >> I'd like to test 1.8 against my existing configs leaving my 1.6.2 >> install untouched. Looking at the output of ./configure --help suggests >> that it's possible to install Asterisk into another prefix of my >> choosing, but as this is unfamiliar territory to me I'm not exactly sure >> how to accomplish this? >> >> Ideally, I'd like to just dump the newly compiled 1.8 and all it's >> dependencies into a standalone directory (say /testing/asterisk or >> something) and update my init script to point to the new binaries. I >> also run a Sangoma USB FXO card and DAHDI for a POTS line that I would >> like to test as well, should it work with the pre-compiled binaries that >> are already there? (DAHDI, etc) >> >> I've never tried this before, and before I potentially break something >> I'd like to know if it's possible and how to implement it? >> >> Thanks, >> Stephen >> >> >> -- _ -- 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] Asterisk 1.8 debian packages?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Anyone know of any Asterisk 1.8 deb's available or when they might be included in backports or (hopefully) Squeeze? I can compile from source... but would much rather have a pre-packaged binary if one exists... Thanks, Stephen -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (Darwin) iEYEARECAAYFAkz/1NQACgkQ3sJXNEncx7gWWQCgzq0PDy7czLdujKyZTslsc8Ka /48AoLmEhQwMc1oMBCgqY0qxrhw6UsfE =ge4j -END PGP SIGNATURE- -- _ -- 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] Running SIP on non-standard ports
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I want to have the ability to have anonymous SIP calls hit my server but I want to run it on different ports and create an SRV record for my target domain. My understanding of SIP is limited, but in a nutshell I want to accomplish the following: - - run SIP signaling on port 6200 - - create RTP ports on 6201-62XX Do I really need 10k ports open for RTP!!?? I don't plan on doing more than a 5-10 calls simultaneously, maybe less than that. Does each RTP port represent one channel, or does it take two, one for each end perhaps? Just trying to come up with a number... And my assumption is also that I will only need to create an SRV record for the SIP signaling, will I need to create SRV record(s) for the RTP ports as well? I'm assuming the SIP signaling handles that instead? Thanks in advance... Stephen -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (Darwin) iEYEARECAAYFAkwg8nkACgkQ3sJXNEncx7i1LQCfcgefEgyBb4QC96dBe46dK6DA EYUAoNcVLAt/lr4EzUslnXEzIJMTVt9h =eis2 -END PGP SIGNATURE- -- _ -- 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 with ztdummy
I'm having trouble with ztdummy and I can't seem to figure it out. I am running Zaptel 1.4.12.1 under Debian 4.0 with latest updates applied and I have compiled Zaptel from source along with a new kernel from Debian sources to include 1khz timer support. The modules build fine, yet when I load them I get the following output from dmesg: rtc: lost some interrupts at 1024Hz And then Asterisk fails to load. I don't have any hardware in this box as it will be utilized for a pure VoIP solution so a hardware timing interface is not an option unless I can't get ztdummy to cooperate. What am I missing? I've been fighting this problem for a couple of days now... help appreciated :) Thanks, Stephen ___ -- 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] Problems with ztdummy
> I think I had a board once where I had to specify ACPI=no to the kernel I believe that has taken care of it, it's working now :) I took out all of the ACPI support in the kernel, I don't need it anyhow as this box will be online 24/7 (like any server I suppose) > Any chance you could check this with the etchandahalf (2.6.24) kernel? That was going to be my next step if I couldn't get it resolved Thanks, Stephen On Thu, Dec 18, 2008 at 11:57, Tzafrir Cohen wrote: > On Thu, Dec 18, 2008 at 10:49:03AM -0500, Stephen Brown Jr wrote: >> I'm having trouble with ztdummy and I can't seem to figure it out. I >> am running Zaptel 1.4.12.1 under Debian 4.0 with latest updates >> applied and I have compiled Zaptel from source along with a new kernel >> from Debian sources to include 1khz timer support. >> >> The modules build fine, yet when I load them I get the following >> output from dmesg: >> >> rtc: lost some interrupts at 1024Hz > > Any chance you could check this with the etchandahalf (2.6.24) kernel? > > -- > Tzafrir Cohen > icq#16849755 jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir > > ___ > -- 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] DAHDI hangup detection
Ok on the workaround, how would I implement it? I'd like to give that a shot. On Tue, Sep 15, 2009 at 12:23, Danny Nicholas wrote: > The issue is that POTS as a technology does not have Answer/Hangup > Supervision control (This is per the good folks at Digium). Your local > Telco may or may not be of assistance, but a simpler workaround might be to > hangup on silence or query the line through AMI before transferring to > voicemail. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen > Brown > Sent: Tuesday, September 15, 2009 11:18 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] DAHDI hangup detection > > I am running Asterisk 1.6.1.6 and DAHDI/DAHDI tools 2.2.0.2 compiled > from source on a Debian Lenny box, also running FreePBX 2.5.2. I also > have an OpenVOX TDM400 card installed with an FXO port on port 1, and an > FXS port on port 2. I have a POTS line installed and working on the FXO > port. > > However I've encountered a weird problem that I can't seem to figure > out. I have incoming POTS line calls set to ring a SIP extension (a > cisco IP phone). It works, but here is the issue: > > - If an inbound call comes in on the POTS line and hangs up before the > call hits voicemail or during the voicemail greeting, Asterisk does not > appear to detect this condition. > - As a result of this, a voicemail is being left with a dialtone and > notifying me > > As a workaround, I have set busy detection in chan_dahdi.conf and set 12 > seconds for a minimum voicemail message in voicemail.conf, but I must be > missing something or something is potentially broke. While this > workaround works, I don't suspect this behavior I am experiencing is > normal and I am having a hard time putting a finger on it. > > Thanks for any replies. > > Stephen > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI disconnect supervision timing
Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6 install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1 FXO port and 1 FXS port. I have a POTS line from my phone company attached to the POTS line. I have asked for "disconnect supervision" to be provisioned on my line and they claim to have added it. However, my scenario is as follows: I receive a call, if the caller hangs up before hitting voice mail, the DAHDI channel is released as to be expected (evidenced from console messaging) If the call gets to voicemail and the caller hangs up during the greeting, no hangup condition is ever detected and I am greeted with a useless voicemail moments later. I am using kewlstart signaling etc. I came across this page from Digium: http://kb.digium.com/entry/6/ This suggests adjusting a variable in zaptel.h, as I don't use zaptel, can this same logic be applied to DAHDI somewhere? My theory is that the "disconnect supervision" signal coming from the phone company may be less than 1000ms. Desperately trying to fix this. Thanks, Stephen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dual Atom mobo - call capacity
Ditto I'm running a Supermicro Atom based dual core server and it's rock solid!!! These make excellent servers for Asterisk installation IMHO. On Fri, Jun 11, 2010 at 05:42, --[ UxBoD ]-- wrote: > > - Original Message - > > On Thu, 10 Jun 2010, Michelle Dupuis wrote: > > > > > I'm looking for a small formfactor mobo for an install that needs to > > > handle 25 phone sets (no transcoding). I found a new dual atom > > > 1.66GHz > > > mobo - anyone know what kinds of call volume that will handle? > > > > On Thu, 10 Jun 2010, mgra...@mstvp.com wrote: > > > > > Based on comments from Ward Mundy during a recent VUC call I'd > > > expect > > > even a single CPU Atom system to handle that many phones in an > > > office > > > application. Perhaps there may be merit in dual CPU in more of a > > > call > > > center application. > > > > Assuming you're talking about something like the Atom 330... > > > > My guess is you will have plenty of horsepower for 25 phone sets -- > > probably even 25 simultaneous calls. > > > > The 330 is dual-core and hyper-threaded so it shows up as 4 CPUs in > > "top." > > > > Asterisk is multi-threaded and should distribute the workload. Another > > advantage is that if you have something CPU heavy like bzip2'ing your > > database dump or compiling Asterisk from source, there are still > > several > > "CPUs" available for Asterisk. > > > > I have a single rack server with a Atom 330 and 2GB RAM, six phones > connected and probably a couple of simultaneous calls at one time. This is > how it looks at the moment: > > total used free sharedbuffers cached > Mem: 20498561346480 703376 0 181920 990376 > -/+ buffers/cache: 1741841875672 > Swap: 4095992 04095992 > > top - 10:41:59 up 12 days, 16:03, 1 user, load average: 0.01, 0.00, 0.00 > Tasks: 122 total, 1 running, 121 sleeping, 0 stopped, 0 zombie > Cpu0 : 0.0%us, 0.0%sy, 0.0%ni, 99.9%id, 0.0%wa, 0.0%hi, 0.0%si, > 0.0%st > Cpu1 : 0.0%us, 0.0%sy, 0.0%ni, 98.4%id, 0.0%wa, 1.5%hi, 0.0%si, > 0.0%st > Cpu2 : 0.1%us, 0.0%sy, 0.1%ni, 99.8%id, 0.0%wa, 0.0%hi, 0.0%si, > 0.0%st > Cpu3 : 0.1%us, 0.0%sy, 0.0%ni, 99.8%id, 0.0%wa, 0.0%hi, 0.0%si, > 0.0%st > Mem: 2049856k total, 1346232k used, 703624k free, 181920k buffers > Swap: 4095992k total,0k used, 4095992k free, 990376k cached > > Have a TDM card in the server and also use G729 codec and Skype for > Asterisk. > -- > Thanks, Phil > > -- > _ > -- 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
