Re: [asterisk-users] possible bug in Asterisk 16
On 06/11/2019 10.51, Tony Mountifield wrote:
>
> If you did this instead, it should work:
>
> exten => _0./_888[24],1,Verbose(0,EXTERNAL CALL)
> same => n,Macro(record)
> same => n,Set(CALLERID(num)=044111)
> exten => _0./044111,n,DIAL(SIP/044111/${EXTEN})
> same => n,Hangup()
it works ! Thanks. You are a genius.
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Re: [asterisk-users] possible bug in Asterisk 16
In article <4fb12839-c24c-bd27-51f9-2d85a4bff...@gmx.ch>,
Fourhundred Thecat <400the...@gmx.ch> wrote:
> Hello,
>
> I am experiencing weird problem in Asterisk 16.2, possibly a bug. Same
> thing works fine in Asterisk 11. Here is the situation:
>
> I have 2 extensions on 2 phones. 4 extensions in total.
>
> phone 1:
> 8882
> 8382
>
> phone 2:
> 8884
> 8384
>
> And I have 2 SIP trunks for outgoing calls. I want to call via SIP1 when
> called via 8882 or 8884, and SIP2 when called via 8382 or 8384.
>
> And one last detail. SIP1 has caller ID spoofing enabled (used for call
> forwarding), so for outgoing calls, I have to explicitly set caller ID,
> otherwise 8882/8884 would be shown.
>
> Here is my config:
>
> exten => _0./_888[24],1,Verbose(0,EXTERNAL CALL)
> same => n,Macro(record)
> same => n,Set(CALLERID(num)=044111)
> same => n,DIAL(SIP/044111/${EXTEN})
> same => n,Hangup()
>
> exten => _0./_838[24],1,Verbose(0,EXTERNAL CALL)
> same => n,Macro(record)
> same => n,DIAL(SIP/044222/${EXTEN})
> same => n,Hangup()
>
> Now here the problem. The combination of /_888[24] and
> Set(CALLERID(num)=044111) causes problems:
>
> sent to invalid extension but no invalid handler:
> context,exten,priority=fullaccess,079333,4
>
> When I comment the line Set(CALLERID, everything works.
> Or when i change the first line to:
>
> exten => _0.,1,Verbose(0,EXTERNAL CALL)
>
> Everything works as well. Only the combination of both
> /_888[24] and Set(CALLERID(num)=044111) causes problems:
>
> can anybody understand what is happening here ?
Yes, I think so. Let's rewrite your first section as it is stored internally,
without the "same" and "n" shortcuts:
exten => _0./_888[24],1,Verbose(0,EXTERNAL CALL)
exten => _0./_888[24],2,Macro(record)
exten => _0./_888[24],3,Set(CALLERID(num)=044111)
exten => _0./_888[24],4,DIAL(SIP/044111/${EXTEN})
exten => _0./_888[24],5,Hangup()
Notice that all steps match both destination number and source caller-ID.
When you get to step 4, you have just changed the source caller-ID to
044111, which no longer matches _888[24]
If you did this instead, it should work:
exten => _0./_888[24],1,Verbose(0,EXTERNAL CALL)
same => n,Macro(record)
same => n,Set(CALLERID(num)=044111)
exten => _0./044111,n,DIAL(SIP/044111/${EXTEN})
same => n,Hangup()
I've just done the above on my Asterisk 16 box, and the loaded dialplan
looks like this:
hp3*CLI> dialplan show testing
[ Context 'testing' created by 'pbx_config' ]
'_0.' (CID match '044111') => 4. DIAL(SIP/044111/${EXTEN})
[extensions.conf:883]
5. Hangup()
[extensions.conf:884]
'_0.' (CID match '_888[24]') => 1. Verbose(0,EXTERNAL CALL)
[extensions.conf:880]
2. Macro(record)
[extensions.conf:881]
3. Set(CALLERID(num)=044111)
[extensions.conf:882]
-= 2 extensions (5 priorities) in 1 context. =-
Notice that the "n" converted to "4" correctly even though the extension
changed.
I wasn't aware it would until I tried it.
Why your original version works fine in Asterisk 11 I don't know. Maybe the
handling of caller-ID changed.
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org
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[asterisk-users] possible bug in Asterisk 16
Hello,
I am experiencing weird problem in Asterisk 16.2, possibly a bug. Same
thing works fine in Asterisk 11. Here is the situation:
I have 2 extensions on 2 phones. 4 extensions in total.
phone 1:
8882
8382
phone 2:
8884
8384
And I have 2 SIP trunks for outgoing calls. I want to call via SIP1 when
called via 8882 or 8884, and SIP2 when called via 8382 or 8384.
And one last detail. SIP1 has caller ID spoofing enabled (used for call
forwarding), so for outgoing calls, I have to explicitly set caller ID,
otherwise 8882/8884 would be shown.
Here is my config:
exten => _0./_888[24],1,Verbose(0,EXTERNAL CALL)
same => n,Macro(record)
same => n,Set(CALLERID(num)=044111)
same => n,DIAL(SIP/044111/${EXTEN})
same => n,Hangup()
exten => _0./_838[24],1,Verbose(0,EXTERNAL CALL)
same => n,Macro(record)
same => n,DIAL(SIP/044222/${EXTEN})
same => n,Hangup()
Now here the problem. The combination of /_888[24] and
Set(CALLERID(num)=044111) causes problems:
sent to invalid extension but no invalid handler:
context,exten,priority=fullaccess,079333,4
When I comment the line Set(CALLERID, everything works.
Or when i change the first line to:
exten => _0.,1,Verbose(0,EXTERNAL CALL)
Everything works as well. Only the combination of both
/_888[24] and Set(CALLERID(num)=044111) causes problems:
can anybody understand what is happening here ?
thanks,
--
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Re: [asterisk-users] Possible BUG on Asterisk 13.9.1
Saint Michael wrote: I get this message very often: [Jun 26 23:52:49] ERROR[10396]: channel.c:1278 ast_channel_by_name_cb: BUG! Must supply a channel name or partial name to match! I could file a bug if somebody tells me how to obtain a trace for this. I cannot imagine. I use PJSIP It appears as though something is requesting a channel but not giving enough information. Are you using AMI or have something outside of Asterisk that is getting information about things? What you'd need to investigate is the console output and that information. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] Possible BUG on Asterisk 13.9.1
I get this message very often: [Jun 26 23:52:49] ERROR[10396]: channel.c:1278 ast_channel_by_name_cb: BUG! Must supply a channel name or partial name to match! I could file a bug if somebody tells me how to obtain a trace for this. I cannot imagine. I use PJSIP -- _ -- 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] Possible bug - queue doesn't play hold music
On Wed, 2012-12-19 at 11:16 -0600, Richard Mudgett wrote: On Fri, 2012-12-14 at 15:16 +, Ishfaq Malik wrote: Hi Can someone else please check the following: We have installed asterisk 1.8.18.0 onto our development and test servers. They were previously on 1.8.7.0 When an inbound call executes a queue, I can see in the logs that the hold music is supposed to start playing but there is no sound. If the call is answered and the callee puts the caller on hold, I can see the same log message of hold music starting but this time the hold music can be heard. This is happening on both installations of 1.8.18.0. If other people are experiencing the same thing we can raise a bug on it. Log excerpts below with my comments after a # symbol -- Executing [s@ethn-xx-work:4] Queue(SIP/x.x.x.x-0061, test-ish,Tn,,,600) -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: no music heard == Using SIP RTP CoS mark 5 -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 answered SIP/x.x.x.x-0061 -- Stopped music on hold on SIP/x.x.x.x-0061 [2012-12-14 14:44:04] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: music can be heard [2012-12-14 14:44:12] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Stopped music on hold on SIP/x.x.x.x-0061 == Spawn extension (ethn-xx-work, s, 4) exited non-zero on 'SIP/x.x.x.x-0061' Really could do with a second opinion on this issue as it would quite a serious bug if it is one... The incoming call leg does not appear to be answered yet so I would not expect the caller to be able to hear MOH. Richard The queue command in the dialplan does not have the r option, therefore it should play the moh as specified in the queue's specific config rather than a ringing tone. This has been the behaviour in previous 1.8 and in 1.4. Also, why does the log indicate moh starting? -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] Possible bug - queue doesn't play hold music
On Thu, 2012-12-20 at 09:23 +, Ishfaq Malik wrote: On Wed, 2012-12-19 at 11:16 -0600, Richard Mudgett wrote: On Fri, 2012-12-14 at 15:16 +, Ishfaq Malik wrote: Hi Can someone else please check the following: We have installed asterisk 1.8.18.0 onto our development and test servers. They were previously on 1.8.7.0 When an inbound call executes a queue, I can see in the logs that the hold music is supposed to start playing but there is no sound. If the call is answered and the callee puts the caller on hold, I can see the same log message of hold music starting but this time the hold music can be heard. This is happening on both installations of 1.8.18.0. If other people are experiencing the same thing we can raise a bug on it. Log excerpts below with my comments after a # symbol -- Executing [s@ethn-xx-work:4] Queue(SIP/x.x.x.x-0061, test-ish,Tn,,,600) -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: no music heard == Using SIP RTP CoS mark 5 -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 answered SIP/x.x.x.x-0061 -- Stopped music on hold on SIP/x.x.x.x-0061 [2012-12-14 14:44:04] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: music can be heard [2012-12-14 14:44:12] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Stopped music on hold on SIP/x.x.x.x-0061 == Spawn extension (ethn-xx-work, s, 4) exited non-zero on 'SIP/x.x.x.x-0061' Really could do with a second opinion on this issue as it would quite a serious bug if it is one... The incoming call leg does not appear to be answered yet so I would not expect the caller to be able to hear MOH. Richard The queue command in the dialplan does not have the r option, therefore it should play the moh as specified in the queue's specific config rather than a ringing tone. This has been the behaviour in previous 1.8 and in 1.4. Also, why does the log indicate moh starting? My apologies, you are correct. Every other time I've used MoH in a queue it has been afetr a Playback or Background, both of which answer the channel... -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] Possible bug - queue doesn't play hold music
On Fri, 2012-12-14 at 15:16 +, Ishfaq Malik wrote: Hi Can someone else please check the following: We have installed asterisk 1.8.18.0 onto our development and test servers. They were previously on 1.8.7.0 When an inbound call executes a queue, I can see in the logs that the hold music is supposed to start playing but there is no sound. If the call is answered and the callee puts the caller on hold, I can see the same log message of hold music starting but this time the hold music can be heard. This is happening on both installations of 1.8.18.0. If other people are experiencing the same thing we can raise a bug on it. Log excerpts below with my comments after a # symbol -- Executing [s@ethn-xx-work:4] Queue(SIP/x.x.x.x-0061, test-ish,Tn,,,600) -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: no music heard == Using SIP RTP CoS mark 5 -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 answered SIP/x.x.x.x-0061 -- Stopped music on hold on SIP/x.x.x.x-0061 [2012-12-14 14:44:04] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: music can be heard [2012-12-14 14:44:12] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Stopped music on hold on SIP/x.x.x.x-0061 == Spawn extension (ethn-xx-work, s, 4) exited non-zero on 'SIP/x.x.x.x-0061' Really could do with a second opinion on this issue as it would quite a serious bug if it is one... -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] Possible bug - queue doesn't play hold music
On Fri, 2012-12-14 at 15:16 +, Ishfaq Malik wrote: Hi Can someone else please check the following: We have installed asterisk 1.8.18.0 onto our development and test servers. They were previously on 1.8.7.0 When an inbound call executes a queue, I can see in the logs that the hold music is supposed to start playing but there is no sound. If the call is answered and the callee puts the caller on hold, I can see the same log message of hold music starting but this time the hold music can be heard. This is happening on both installations of 1.8.18.0. If other people are experiencing the same thing we can raise a bug on it. Log excerpts below with my comments after a # symbol -- Executing [s@ethn-xx-work:4] Queue(SIP/x.x.x.x-0061, test-ish,Tn,,,600) -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: no music heard == Using SIP RTP CoS mark 5 -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 answered SIP/x.x.x.x-0061 -- Stopped music on hold on SIP/x.x.x.x-0061 [2012-12-14 14:44:04] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: music can be heard [2012-12-14 14:44:12] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Stopped music on hold on SIP/x.x.x.x-0061 == Spawn extension (ethn-xx-work, s, 4) exited non-zero on 'SIP/x.x.x.x-0061' Really could do with a second opinion on this issue as it would quite a serious bug if it is one... The incoming call leg does not appear to be answered yet so I would not expect the caller to be able to hear MOH. Richard -- _ -- 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] Possible bug - queue doesn't play hold music
Hi Can someone else please check the following: We have installed asterisk 1.8.18.0 onto our development and test servers. They were previously on 1.8.7.0 When an inbound call executes a queue, I can see in the logs that the hold music is supposed to start playing but there is no sound. If the call is answered and the callee puts the caller on hold, I can see the same log message of hold music starting but this time the hold music can be heard. This is happening on both installations of 1.8.18.0. If other people are experiencing the same thing we can raise a bug on it. Log excerpts below with my comments after a # symbol -- Executing [s@ethn-xx-work:4] Queue(SIP/x.x.x.x-0061, test-ish,Tn,,,600) -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: no music heard == Using SIP RTP CoS mark 5 -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 answered SIP/x.x.x.x-0061 -- Stopped music on hold on SIP/x.x.x.x-0061 [2012-12-14 14:44:04] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: music can be heard [2012-12-14 14:44:12] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Stopped music on hold on SIP/x.x.x.x-0061 == Spawn extension (ethn-xx-work, s, 4) exited non-zero on 'SIP/x.x.x.x-0061' -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] Possible bug (or feature?) in extension matching and parking feature
I wanted a custom extension to match miss-dialled numbers in my dialplan. I've included the following: exten = _X.,1,Answer() exten = _X.,n,Playback(extension_not_found_please_make_sure_you_dial_nine_in_front_of_external_numbers) exten = _X.,n,Hangup() However, this has the curious side effect of making the parking extensions (located at 700, 701-720 - as per defaults) invisible. For some reason Asterisk doesn't seem to include those extensions on an even footing with extensions in the local context when extensions matching, although they are included with include = parkedcalls. I can't park calls by transferring them to 700, I can't reach calls parked with the one step dialling sequence (which continues to work) any more. My understanding is the * would match the most specific extension first - but it would appear that this doesn't apply to parking extensions. All other extensions work fine when using the custom extension above. I'm running Asterisk 10.1.2 -- _ -- 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] Possible bug (or feature?) in extension matching and parking feature
My understanding regarding the pattern match order is that Asterisk will not search include= contexts unless there is no matching extension in the original context. So, since _X. matches anything, the include=parkedcalls context will never be searched. A better way to accomplish what you want to do is to use exten = i (the Asterisk reserved extension for invalid) to catch mis-dialed calls. That way Asterisk will search for correct extensions in the original context, then search your include= contexts, then fall through to exten = i if nothing matches. And the i DOES need to be lower case. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Arcus Sent: Sunday, February 26, 2012 5:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Possible bug (or feature?) in extension matching and parking feature I wanted a custom extension to match miss-dialled numbers in my dialplan. I've included the following: exten = _X.,1,Answer() exten = _X.,n,Playback(extension_not_found_please_make_sure_you_dial_nine_in_front_of_external_numbers) exten = _X.,n,Hangup() However, this has the curious side effect of making the parking extensions (located at 700, 701-720 - as per defaults) invisible. For some reason Asterisk doesn't seem to include those extensions on an even footing with extensions in the local context when extensions matching, although they are included with include = parkedcalls. I can't park calls by transferring them to 700, I can't reach calls parked with the one step dialling sequence (which continues to work) any more. My understanding is the * would match the most specific extension first - but it would appear that this doesn't apply to parking extensions. All other extensions work fine when using the custom extension above. I'm running Asterisk 10.1.2 -- _ -- 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 The message does not contain any threats AVG for MS Exchange Server (2012.0.1913 - 2114/4833)-- _ -- 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] Possible bug (or feature?) in extension matching and parking feature
Thanks Noah. I wasn't aware that the extension matching only searches the includes if there is no match in the original context. Thanks again On 26/02/12 13:32, Noah Engelberth wrote: My understanding regarding the pattern match order is that Asterisk will not search include= contexts unless there is no matching extension in the original context. So, since _X. matches anything, the include=parkedcalls context will never be searched. A better way to accomplish what you want to do is to use exten = i (the Asterisk reserved extension for invalid) to catch mis-dialed calls. That way Asterisk will search for correct extensions in the original context, then search your include= contexts, then fall through to exten = i if nothing matches. And the i DOES need to be lower case. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Arcus Sent: Sunday, February 26, 2012 5:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Possible bug (or feature?) in extension matching and parking feature I wanted a custom extension to match miss-dialled numbers in my dialplan. I've included the following: exten = _X.,1,Answer() exten = _X.,n,Playback(extension_not_found_please_make_sure_you_dial_nine_in_front_of_external_numbers) exten = _X.,n,Hangup() However, this has the curious side effect of making the parking extensions (located at 700, 701-720 - as per defaults) invisible. For some reason Asterisk doesn't seem to include those extensions on an even footing with extensions in the local context when extensions matching, although they are included with include = parkedcalls. I can't park calls by transferring them to 700, I can't reach calls parked with the one step dialling sequence (which continues to work) any more. My understanding is the * would match the most specific extension first - but it would appear that this doesn't apply to parking extensions. All other extensions work fine when using the custom extension above. I'm running Asterisk 10.1.2 -- _ -- 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 -- Linux vehicle CCTV - www.open-t.co.uk/iroko -- _ -- 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] Possible Bug? .call files executing multiple times
Thanks Danny. Changing the ownership of the .call files seems to have fixed the problem and I can now see that asterisk is adding a StartRetry line to the end of the file after it makes the first call, which it was unable to do before since the file was owned by root: cp test5703.call /tmp/test.call chown asterisk:asterisk /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ Thanks, Brandon On 08/29/2011 05:41 PM, Danny Nicholas wrote: Asterisk has to be able to execute and rewrite the file - the call file is updated in place and when the call is considered successful, removed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 4:28 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times Also I should note that we use the 'noatime' attribute on the /var filesystem, would this cause the problem below? On 08/29/2011 05:22 PM, Brandon Phelps wrote: Here is the contents of the .call file. The file is the same before the move as after (I did a cat on the file after the move, while the phone was ringing a second time): Channel: Local/5703@ext-main Callerid: MyCompany8005551234 Set: TicketNumber=100 Set: CallerID_Num=8005551234 Set: CALLSTATUS=0 Context: ext-autodialer MaxRetries: 0 WaitTime: 45 Extension: s Priority: 1 We have tried using a SIP channel as well (as opposed to Local) with the same results. The s extension of ext-autodialer runs an AGI script which makes use of those Set: variables. I can most easily reproduce the problem by simply not answering the call. After 2 or 3 rings line 2 on the phone lights up indicating another call. If I reject the first call and answer the second call, it's the same script. Also during my most recent test the following happened: 1. I moved file to /var/spool/asterisk/outgoing 2. Phone rang on line 1 3. I let phone continue to ring 4. After 3 rings, line 2 started ringing (another call from the same .call file) 5. I rejected both calls, sending both to voicemail. 6. 6 or 7 seconds after rejecting both calls, the phone rang a 3rd time. 7. I let the phone ring until it was automatically moved to voicemail and finally the .call file was removed. On 08/29/2011 11:00 AM, Danny Nicholas wrote: Can you post the .call file (with called number blacked out) before call and after 1-2 calls? (file 1 should be before you mv to /v/s/a/o, file 2 should be from /v/s/a/o). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 8:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times On 08/19/2011 09:14 AM, Brandon Phelps wrote: Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Sorry to bring this back up but I am still having this issue and haven't had any luck resolving it. It should be noted that the .call files in question are set to MaxRetries: 0, and simply connect the call to the 's' extension in a custom context. From there the context is pretty complicated, running some AGI scripts along with some dealing with user input, basically a simple IVR. Any help would be appreciated. Thanks, Brandon -- _ -- 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
Re: [asterisk-users] Possible Bug? .call files executing multiple times
On 08/19/2011 09:14 AM, Brandon Phelps wrote: Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Sorry to bring this back up but I am still having this issue and haven't had any luck resolving it. It should be noted that the .call files in question are set to MaxRetries: 0, and simply connect the call to the 's' extension in a custom context. From there the context is pretty complicated, running some AGI scripts along with some dealing with user input, basically a simple IVR. Any help would be appreciated. Thanks, Brandon -- _ -- 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] Possible Bug? .call files executing multiple times
Can you post the .call file (with called number blacked out) before call and after 1-2 calls? (file 1 should be before you mv to /v/s/a/o, file 2 should be from /v/s/a/o). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 8:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times On 08/19/2011 09:14 AM, Brandon Phelps wrote: Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Sorry to bring this back up but I am still having this issue and haven't had any luck resolving it. It should be noted that the .call files in question are set to MaxRetries: 0, and simply connect the call to the 's' extension in a custom context. From there the context is pretty complicated, running some AGI scripts along with some dealing with user input, basically a simple IVR. Any help would be appreciated. Thanks, Brandon -- _ -- 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] Possible Bug? .call files executing multiple times
Here is the contents of the .call file. The file is the same before the move as after (I did a cat on the file after the move, while the phone was ringing a second time): Channel: Local/5703@ext-main Callerid: MyCompany 8005551234 Set: TicketNumber=100 Set: CallerID_Num=8005551234 Set: CALLSTATUS=0 Context: ext-autodialer MaxRetries: 0 WaitTime: 45 Extension: s Priority: 1 We have tried using a SIP channel as well (as opposed to Local) with the same results. The s extension of ext-autodialer runs an AGI script which makes use of those Set: variables. I can most easily reproduce the problem by simply not answering the call. After 2 or 3 rings line 2 on the phone lights up indicating another call. If I reject the first call and answer the second call, it's the same script. Also during my most recent test the following happened: 1. I moved file to /var/spool/asterisk/outgoing 2. Phone rang on line 1 3. I let phone continue to ring 4. After 3 rings, line 2 started ringing (another call from the same .call file) 5. I rejected both calls, sending both to voicemail. 6. 6 or 7 seconds after rejecting both calls, the phone rang a 3rd time. 7. I let the phone ring until it was automatically moved to voicemail and finally the .call file was removed. On 08/29/2011 11:00 AM, Danny Nicholas wrote: Can you post the .call file (with called number blacked out) before call and after 1-2 calls? (file 1 should be before you mv to /v/s/a/o, file 2 should be from /v/s/a/o). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 8:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times On 08/19/2011 09:14 AM, Brandon Phelps wrote: Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Sorry to bring this back up but I am still having this issue and haven't had any luck resolving it. It should be noted that the .call files in question are set to MaxRetries: 0, and simply connect the call to the 's' extension in a custom context. From there the context is pretty complicated, running some AGI scripts along with some dealing with user input, basically a simple IVR. Any help would be appreciated. Thanks, Brandon -- _ -- 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
Re: [asterisk-users] Possible Bug? .call files executing multiple times
Also I should note that we use the 'noatime' attribute on the /var filesystem, would this cause the problem below? On 08/29/2011 05:22 PM, Brandon Phelps wrote: Here is the contents of the .call file. The file is the same before the move as after (I did a cat on the file after the move, while the phone was ringing a second time): Channel: Local/5703@ext-main Callerid: MyCompany 8005551234 Set: TicketNumber=100 Set: CallerID_Num=8005551234 Set: CALLSTATUS=0 Context: ext-autodialer MaxRetries: 0 WaitTime: 45 Extension: s Priority: 1 We have tried using a SIP channel as well (as opposed to Local) with the same results. The s extension of ext-autodialer runs an AGI script which makes use of those Set: variables. I can most easily reproduce the problem by simply not answering the call. After 2 or 3 rings line 2 on the phone lights up indicating another call. If I reject the first call and answer the second call, it's the same script. Also during my most recent test the following happened: 1. I moved file to /var/spool/asterisk/outgoing 2. Phone rang on line 1 3. I let phone continue to ring 4. After 3 rings, line 2 started ringing (another call from the same .call file) 5. I rejected both calls, sending both to voicemail. 6. 6 or 7 seconds after rejecting both calls, the phone rang a 3rd time. 7. I let the phone ring until it was automatically moved to voicemail and finally the .call file was removed. On 08/29/2011 11:00 AM, Danny Nicholas wrote: Can you post the .call file (with called number blacked out) before call and after 1-2 calls? (file 1 should be before you mv to /v/s/a/o, file 2 should be from /v/s/a/o). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 8:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times On 08/19/2011 09:14 AM, Brandon Phelps wrote: Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Sorry to bring this back up but I am still having this issue and haven't had any luck resolving it. It should be noted that the .call files in question are set to MaxRetries: 0, and simply connect the call to the 's' extension in a custom context. From there the context is pretty complicated, running some AGI scripts along with some dealing with user input, basically a simple IVR. Any help would be appreciated. Thanks, Brandon -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Possible Bug? .call files executing multiple times
Asterisk has to be able to execute and rewrite the file - the call file is updated in place and when the call is considered successful, removed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 4:28 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times Also I should note that we use the 'noatime' attribute on the /var filesystem, would this cause the problem below? On 08/29/2011 05:22 PM, Brandon Phelps wrote: Here is the contents of the .call file. The file is the same before the move as after (I did a cat on the file after the move, while the phone was ringing a second time): Channel: Local/5703@ext-main Callerid: MyCompany 8005551234 Set: TicketNumber=100 Set: CallerID_Num=8005551234 Set: CALLSTATUS=0 Context: ext-autodialer MaxRetries: 0 WaitTime: 45 Extension: s Priority: 1 We have tried using a SIP channel as well (as opposed to Local) with the same results. The s extension of ext-autodialer runs an AGI script which makes use of those Set: variables. I can most easily reproduce the problem by simply not answering the call. After 2 or 3 rings line 2 on the phone lights up indicating another call. If I reject the first call and answer the second call, it's the same script. Also during my most recent test the following happened: 1. I moved file to /var/spool/asterisk/outgoing 2. Phone rang on line 1 3. I let phone continue to ring 4. After 3 rings, line 2 started ringing (another call from the same .call file) 5. I rejected both calls, sending both to voicemail. 6. 6 or 7 seconds after rejecting both calls, the phone rang a 3rd time. 7. I let the phone ring until it was automatically moved to voicemail and finally the .call file was removed. On 08/29/2011 11:00 AM, Danny Nicholas wrote: Can you post the .call file (with called number blacked out) before call and after 1-2 calls? (file 1 should be before you mv to /v/s/a/o, file 2 should be from /v/s/a/o). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 8:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times On 08/19/2011 09:14 AM, Brandon Phelps wrote: Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Sorry to bring this back up but I am still having this issue and haven't had any luck resolving it. It should be noted that the .call files in question are set to MaxRetries: 0, and simply connect the call to the 's' extension in a custom context. From there the context is pretty complicated, running some AGI scripts along with some dealing with user input, basically a simple IVR. Any help would be appreciated. Thanks, Brandon -- _ -- 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
[asterisk-users] Possible Bug? .call files executing multiple times
Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, -- Brandon -- _ -- 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] Possible Bug? .call files executing multiple times
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Friday, August 19, 2011 8:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Possible Bug? .call files executing multiple times Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Two things 1. Asterisk 1.8.what? 2. once the call file goes to /v/s/a/o, it becomes a self-logger; the file won't go away until Asterisk senses that the call is successfully completed. You can counteract this by putting Maxtries:1 in your call file. -- _ -- 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] Possible Bug? .call files executing multiple times
We are running version 1.8.5.0. I'll try the Maxtries and see what happens. Brandon On 08/19/2011 09:19 AM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Friday, August 19, 2011 8:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Possible Bug? .call files executing multiple times Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Two things 1. Asterisk 1.8.what? 2. once the call file goes to /v/s/a/o, it becomes a self-logger; the file won't go away until Asterisk senses that the call is successfully completed. You can counteract this by putting Maxtries:1 in your call file. -- _ -- 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] Possible Bug? .call files executing multiple times
Danny, Now that I notice, MaxRetries has a default of 0 anyway, so not setting it should have not retried at all anyway. Although I did set it to 0 manually and still get the double calls. Brandon On 08/19/2011 09:31 AM, Brandon Phelps wrote: We are running version 1.8.5.0. I'll try the Maxtries and see what happens. Brandon On 08/19/2011 09:19 AM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Friday, August 19, 2011 8:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Possible Bug? .call files executing multiple times Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Two things 1. Asterisk 1.8.what? 2. once the call file goes to /v/s/a/o, it becomes a self-logger; the file won't go away until Asterisk senses that the call is successfully completed. You can counteract this by putting Maxtries:1 in your call file. -- _ -- 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
Re: [asterisk-users] Possible Bug? .call files executing multiple times
Danny, Now that I notice, MaxRetries has a default of 0 anyway, so not setting it should have not retried at all anyway. Although I did set it to 0 manually and still get the double calls. Brandon On 08/19/2011 09:31 AM, Brandon Phelps wrote: We are running version 1.8.5.0. I'll try the Maxtries and see what happens. Brandon On 08/19/2011 09:19 AM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Friday, August 19, 2011 8:15 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Possible Bug? .call files executing multiple times Hello all, We are setting up an auto-dialer to call customers based on the opening of tickets in our internal ticketing system. Everything is going fine so far except for one snag: To test the system we are implementing I am manually moving .call files into the /var/spool/asterisk/outgoing directory like this: asterisk@dialerdev:~# cp test5703.call /tmp/test.call mv /tmp/test.call /var/spool/asterisk/outgoing/ This works great and the call is immediately started, however more often than not (ie. not all the time, but most of the time) after answering the call or rejecting it (sending it to voicemail), another call is performed using the same file. I notice that when a call is initiated the .call file is not removed immediately. Instead, asterisk waits until the call is completed before removing the call file, so it seems like 5-10 seconds into the call since the .call file still exists another call is placed. Any advice on how we can avoid this situation and ensure that only one call is made per .call file? The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. Thanks, Two things 1. Asterisk 1.8.what? 2. once the call file goes to /v/s/a/o, it becomes a self-logger; the file won't go away until Asterisk senses that the call is successfully completed. You can counteract this by putting Maxtries:1 in your call file. -- _ -- 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] Possible bug in Hangup() (Asterisk 1.4.x)
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,Hangup(${HANGUPCAUSE})
I have noticed that no matter what value we set in the Hangup(cause
code) commands, if the call is not answered by peer01 for any reason,
the actual cause code returned to the calling party is a 503, no matter
what the ${HANGUPCAUSE} is.
Even if we set a fixed value like Hangup(1) (which should give a 404) or
Hangup(17) (which should give a 486), the cause code returned is always
a 503.
Has anyone else noticed this? I went through the issue tracker but I
couldn't find any relevant bug posted in the past. I am certain that in
previous versions I could set the reply message to the desired value, so
I wonder if this is a bug in this particular version (1.4.33.1).
--
Best regards,
Vlasis Hatzistavrou.
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Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
My guess is since the call was never answered you should be looking at
${DIALSTATUS}
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 15, 2011, at 5:02 AM, Vlasis Hatzistavrou wrote:
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,Hangup(${HANGUPCAUSE})
I have noticed that no matter what value we set in the Hangup(cause code)
commands, if the call is not answered by peer01 for any reason, the actual
cause code returned to the calling party is a 503, no matter what the
${HANGUPCAUSE} is.
Even if we set a fixed value like Hangup(1) (which should give a 404) or
Hangup(17) (which should give a 486), the cause code returned is always a 503.
Has anyone else noticed this? I went through the issue tracker but I couldn't
find any relevant bug posted in the past. I am certain that in previous
versions I could set the reply message to the desired value, so I wonder if
this is a bug in this particular version (1.4.33.1).
--
Best regards,
Vlasis Hatzistavrou.
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Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
Hello Jim,
Thank you for the reply.
The problem is not reading the ${HANGUPCAUSE} or the ${DIALSTATUS}. It
is that the Hangup(cause) command seems to ignore its argument and
just sends a 503 cause to the caller for all unanswered calls no matter
what...
Hangup(cause) was working as expected in previous versions and I
wonder if something was broken along the way that went by unnoticed. I
am just asking in the list in case I am missing something too obvious
before posting a bug.
--
Best regards,
Vlasis Hatzistavrou.
On 15/4/2011 4:22 μμ, Jim Dickenson wrote:
My guess is since the call was never answered you should be looking at
${DIALSTATUS}
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Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
If what you showed is your whole dialplan then none of the i or t or h
extensions are going to be executed for a non answered call.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 15, 2011, at 6:46 AM, Vlasis Hatzistavrou wrote:
Hello Jim,
Thank you for the reply.
The problem is not reading the ${HANGUPCAUSE} or the ${DIALSTATUS}. It is
that the Hangup(cause) command seems to ignore its argument and just sends
a 503 cause to the caller for all unanswered calls no matter what...
Hangup(cause) was working as expected in previous versions and I wonder if
something was broken along the way that went by unnoticed. I am just asking
in the list in case I am missing something too obvious before posting a bug.
--
Best regards,
Vlasis Hatzistavrou.
On 15/4/2011 4:22 μμ, Jim Dickenson wrote:
My guess is since the call was never answered you should be looking at
${DIALSTATUS}
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Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
On 15 April 2011 13:02, Vlasis Hatzistavrou vh...@kinetix.gr wrote:
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,Hangup(${HANGUPCAUSE})
I have noticed that no matter what value we set in the Hangup(cause code)
commands, if the call is not answered by peer01 for any reason, the actual
cause code returned to the calling party is a 503, no matter what the
${HANGUPCAUSE} is.
Even if we set a fixed value like Hangup(1) (which should give a 404) or
Hangup(17) (which should give a 486), the cause code returned is always a
503.
Has anyone else noticed this? I went through the issue tracker but I
couldn't find any relevant bug posted in the past. I am certain that in
previous versions I could set the reply message to the desired value, so I
wonder if this is a bug in this particular version (1.4.33.1).
Strictly speaking you can only Hangup (BYE) an answered and fully
established call. In SIP terms, a hangup that occurs before an answer
is a CANCEL, and I believe a CANCEL is always represented by a 503
code in chan_sip.
Regards,
Steve
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Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
The h extension is executed after the remote end peer01 rejects the call
with a 408. I verified it by altering the dialplan as:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,NoOp(Hangup cause is: ${HANGUPCAUSE})
exten = h,n,Hangup(${HANGUPCAUSE})
and I saw in the Asterisk CLI that the correct hangupcause is shown.
--
Best regards,
Vlasis Hatzistavrou.
On 15/4/2011 5:01 μμ, Jim Dickenson wrote:
If what you showed is your whole dialplan then none of the i or t or h
extensions are going to be executed for a non answered call.
--
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Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
Hello Steve, On 15/4/2011 5:07 μμ, Steve Davies wrote: Strictly speaking you can only Hangup (BYE) an answered and fully established call. In SIP terms, a hangup that occurs before an answer is a CANCEL, and I believe a CANCEL is always represented by a 503 code in chan_sip. Regards, Steve I see what you mean, but it is the called end (peer01) that rejects the call with a 408 message, it is not the originator that is canceling the call. The call flow is this: Caller-Asterisk-Peer01 and Asterisk receives a STATUS 408 message from Peer01 instead of an answer. Asterisk then sends a STATUS 503 to the Caller, instead of sending a STATUS 408. The question is how to copy the correct cause code from the terminating end to the originating end. I tried setting Hangup(1) to send a 404 to the called, a Hangup(17) to send a 486 to the caller and pretty much any other value in the Hangup() but Asterisk will keep on sending a 503. I don't believe that my memory fails me, I'm pretty sure I could set a desirable cause in the Hangup() command in previous versions... -- Best regards, Vlasis Hatzistavrou. -- _ -- 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] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Thu, Dec 23, 2010 at 1:39 PM, Tilghman Lesher tilgh...@meg.abyt.es wrote: On Thursday 23 December 2010 09:16:26 Bryant Zimmerman wrote: In the voip-info posting Right here is why you fail. Voip-info is very often wrong. Refer to the documentation that comes with Asterisk for definitive information. In this case, the h extension should be in the calling context, not within the Macro itself. -- Tilghman -- _ -- 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 +1 for the informative answer -- _ -- 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] Possible Bug (Include ${HANGUPCAUSE} in CDR)
Tilghman
This does not make any sense. In the voip-info posting for the h
extension it specifically states that to handle h while in a macro that
the macro needs an h extension. The h extension runs inside the macro
but the CDR data is not being updated correctly. Also the rc-ANSWER entry
in the macro does not update the CDR with the ${HANGUPCAUSE} either after
the far end hangs up. This is diffently inconsistent behavior here. Both
the DIAL and h extension are inside the macro so the behaivior should be
consistent. If I am understanding you correctly the only way we can get a
CDR to update after a dial is to not do any DIAL calls in a MACRO is this
what you are saying? Otherwise your logic may be flawed or we have a very
big logic bug in the Asterisk Macro system.
From: http://www.voip-info.org/wiki/view/Asterisk+h+extension
Be aware: Macros require their own h extension as they do not make use
of the calling context's h extension!
Tilghamn thanks for the feed back the back and forth here is great and
helps a lot it is giving me more ideas to test against.
Bryant
From: Tilghman Lesher tilgh...@meg.abyt.es
Sent: Thursday, December 23, 2010 12:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wednesday 22 December 2010 21:08:56 Bryant Zimmerman wrote:
My h extension is in the same context as my Dial commands. Here is a
cut back version of the code.
The cause_code value is never stored in the mysql databae even when set
in the h extension or the
when set in rc-ANSWER' OR doDialStd
[macro-OBD-DoOutboundDial]
exten = h,1,NoOp(Cause Code = ${HANGUPCAUSE})
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
exten = h,n,Goto(rc-${DIALSTATUS},1)
There's the problem. The h extension should be in whatever context is
calling the Macro, not in the Macro context itself.
--
Tilghman
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Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Thursday 23 December 2010 09:16:26 Bryant Zimmerman wrote: In the voip-info posting Right here is why you fail. Voip-info is very often wrong. Refer to the documentation that comes with Asterisk for definitive information. In this case, the h extension should be in the calling context, not within the Macro itself. -- Tilghman -- _ -- 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] Possible Bug (Include ${HANGUPCAUSE} in CDR)
Ok I can't get my CDR values to set from the h extension in either 1.6.2 or
1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may be
; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Really what I need to be able to do is in the h quickly store some values
to the CDR then.
For the write of the CDR and stopping the billing seconds. Then continue
to process some cleanup funcitons.
How can I work arround asterisk not honoring the endbeforehexten=no.
Is there some other way to achieve this?
Bryant I need the cause code stored.Really what I need to be able to do is
in the h quickly store some values to the CDR then.For the write of the CDR
and stopping the billing seconds. Then continue to process some cleanup
funcitons.How can I work arround asterisk not honoring the
endbeforehexten=no.Is there some other way to achieve this?Bryant
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Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wed, 2010-12-22 at 12:42 -0500, Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either
1.6.2 or 1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may
be
; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Really what I need to be able to do is in the h quickly store some
values to the CDR then.
For the write of the CDR and stopping the billing seconds. Then
continue to process some cleanup funcitons.
How can I work arround asterisk not honoring the endbeforehexten=no.
Is there some other way to achieve this?
Bryant
I need the cause code stored.Really what I need to be able to do is in
the h quickly store some values to the CDR then.For the write of the
CDR and stopping the billing seconds. Then continue to process some
cleanup funcitons.How can I work arround asterisk not honoring the
endbeforehexten=no.Is there some other way to achieve this?Bryant
Is the CDR line your only h line? I ask because if you only have one
priority for h then you MUST have:
exten = h,1,Set(CDR(cause_code)=${HANGUPCAUSE})
This is because the dialplan will not use n for the first priority and
thus will never run.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Wednesday, December 22, 2010 11:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Developers Mailing List
Subject: Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
Ok I can't get my CDR values to set from the h extension in either 1.6.2 or
1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are finished
; executing. By enabling this option, the CDR will be ended before executing
; the h extension so that CDR values such as end and billsec may be
; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Really what I need to be able to do is in the h quickly store some values to
the CDR then.
For the write of the CDR and stopping the billing seconds. Then continue to
process some cleanup funcitons.
How can I work arround asterisk not honoring the endbeforehexten=no.
Is there some other way to achieve this?
Bryant
I need the cause code stored.Really what I need to be able to do is in the h
quickly store some values to the CDR then.For the write of the CDR and
stopping the billing seconds. Then continue to process some cleanup
funcitons.How can I work arround asterisk not honoring the
endbeforehexten=no.Is there some other way to achieve this?Bryant
Okay, this is a mentally challenged solution, but at the h extension you
have ${UNIQUEID} and ${HANGUPCAUSE} available to you. Use DeadAGI to
stuff these values into the CDR or into another file that you can
cross-reference. If you use the two-file method, you would just select the
matching file using uniqueid. That's what I did when I had this problem on
a version of 1.4.
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Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wednesday 22 December 2010 11:42:33 Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either 1.6.2
or 1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may
be ; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Sounds like your h extension is in the wrong context. Try including some
information about where you are putting the h extension and what includes
you're doing.
--
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Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
My h extension is in the same context as my Dial commands. Here is a cut
back version of the code.
The cause_code value is never stored in the mysql databae even when set in
the h extension or the
when set in rc-ANSWER' OR doDialStd
[macro-OBD-DoOutboundDial]
exten = s,1,Macro(${ARG1})
exten = s,n,Set(CALLERID(name)=${siteDefaultCIDName})
exten = s,n,Set(CALLERID(number)=${siteDefaultCIDNumber})
exten = s,n,SipAddHeader(X-interNetGR-linetype:${gbl_ibclinetype})
exten = s,n,SipAddHeader(X-interNetGR-actlineid:${gbl_actlineid})
exten = s,n,Set(GROUP()=${siteGrpLineCount})
exten = s,n,Set(c_DialArg=${ARG2})
exten = s,n,Set(c_DialExten=${MACRO_EXTEN})
exten = s,n,GoSub(DoLineCountCheck,1)
exten = s,n,GotoIf($[${siteOverLineCount}=1]?OverLineCount,1)
exten = s,n,GosubIf($[${c_DialExten}=${siteDirSer}]?OverLineCount,1)
exten = s,n,GosubIf($[${c_DialExten}=411]?nofeature,1)
exten = s,n,GosubIf($[${siteUseE164}=1]?doDialE164,1:doDialStd,1)
exten = s,n,Goto(rc-${DIALSTATUS},1)
exten = s,n,Busy(60)
exten = s,n,Hangup()
exten = h,1,NoOp(Cause Code = ${HANGUPCAUSE})
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
exten = h,n,Goto(rc-${DIALSTATUS},1)
exten = doDialStd,1,NoOp(Calling Using No E164)
exten =
doDialStd,n,Macro(OBD-CheckOutboundNumber,${c_DialArg}${c_DialExten})
exten =
doDialStd,n,Dial(${siteDefaultOutboundTrunk}/${c_DialArg}${c_DialExten},120,
ge${siteDialOptionsPublic})
exten = doDialStd,n,Set(CDR(cause_code)=${HANGUPCAUSE})
exten = doDialStd,n,Return
exten = rc-ANSWER,1,NoOp(Do Return ANSWER)
exten = rc-ANSWER,n,Set(CDR(cause_code)=${HANGUPCAUSE})
exten = rc-ANSWER,n,Hangup()
exten = rc-BUSY,1,NoOp(Do Return BUSY)
exten = rc-BUSY,n,Busy()
exten = rc-BUSY,n,Hangup()
exten = rc-NOANSWER,1,NoOp(Do Return NOANSWER)
exten = rc-NOANSWER,n,NoOp(Cause Code = ${HANGUPCAUSE})
exten = rc-NOANSWER,n,Hangup()
Any more feed back would be appercaited.
Bryant
From: Tilghman Lesher tilgh...@meg.abyt.es
Sent: Wednesday, December 22, 2010 3:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wednesday 22 December 2010 11:42:33 Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either 1.6.2
or 1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may
be ; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Sounds like your h extension is in the wrong context. Try including some
information about where you are putting the h extension and what
includes
you're doing.
--
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Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
No this is just a snip of the much larger code.
The h extension is runing but no values port dial function aer being written.
If I do a Set(CDR(field)=Value) before the dial
The value is stored. See my other response for a larger snip of code.
Bryant
From: Carlos Chavez cur...@telecomabmex.com
Sent: Wednesday, December 22, 2010 2:46 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wed, 2010-12-22 at 12:42 -0500, Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either
1.6.2 or 1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may
be
; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Really what I need to be able to do is in the h quickly store some
values to the CDR then.
For the write of the CDR and stopping the billing seconds. Then
continue to process some cleanup funcitons.
How can I work arround asterisk not honoring the endbeforehexten=no.
Is there some other way to achieve this?
Bryant
I need the cause code stored.Really what I need to be able to do is in
the h quickly store some values to the CDR then.For the write of the
CDR and stopping the billing seconds. Then continue to process some
cleanup funcitons.How can I work arround asterisk not honoring the
endbeforehexten=no.Is there some other way to achieve this?Bryant
Is the CDR line your only h line? I ask because if you only have one
priority for h then you MUST have:
exten = h,1,Set(CDR(cause_code)=${HANGUPCAUSE})
This is because the dialplan will not use n for the first priority and
thus will never run.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wednesday 22 December 2010 21:08:56 Bryant Zimmerman wrote:
My h extension is in the same context as my Dial commands. Here is a
cut back version of the code.
The cause_code value is never stored in the mysql databae even when set
in the h extension or the
when set in rc-ANSWER' OR doDialStd
[macro-OBD-DoOutboundDial]
exten = h,1,NoOp(Cause Code = ${HANGUPCAUSE})
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
exten = h,n,Goto(rc-${DIALSTATUS},1)
There's the problem. The h extension should be in whatever context is
calling the Macro, not in the Macro context itself.
--
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Re: [asterisk-users] Possible bug in chan_sip:add_sdp
This code is really ugly und hard to verify. Please file a bug report at https://issues.asterisk.org/ thanks klaus Am 06.05.2010 23:54, schrieb Richard Kenner: I can confirm that the following fixes my problem: --- chan_sip.c (revision 261450) +++ chan_sip.c (working copy) @@ -10357,12 +10357,22 @@ strlen(connection) + strlen(session_time); if (needaudio) len += m_audio-used + a_audio-used + strlen(hold); + else if (p-offered_media[SDP_AUDIO].offered) + len += strlen(m=audio 0 RTP/AVP \r\n) + strlen(p-offered_media[SDP_AUDIO].text); + if (needvideo) /* only if video response is appropriate */ len += m_video-used + a_video-used + strlen(bandwidth) + strlen(hold); + else if (p-offered_media[SDP_VIDEO].offered) + len += strlen(m=video 0 RTP/AVP \r\n) + strlen(p-offered_media[SDP_VIDEO].text); + if (needtext) /* only if text response is appropriate */ len += m_text-used + a_text-used + strlen(hold); + else if (p-offered_media[SDP_TEXT].offered) + len += strlen(m=text 0 RTP/AVP \r\n) + strlen(p-offered_media[SDP_TEXT].text); if (add_t38) len += m_modem-used + a_modem-used; + else if (p-offered_media[SDP_IMAGE].offered) + len += strlen(m=image 0 udptl t38\r\n); add_header(resp, Content-Type, application/sdp); add_header_contentLength(resp, len); -- _ -- 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] Possible bug in chan_sip:add_sdp
This code is really ugly und hard to verify. Since the computation of the is being done with separate code from the actual output, the code in that part of the module is indeed ugly. But I wanted to make the smallest possible change. However, I do suggest that the full output string be built up and the output as once. Please file a bug report at https://issues.asterisk.org/ Will do. -- _ -- 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] Possible bug in chan_sip:add_sdp
Am I missing something here? I see
if (needvideo) { /* only if video response is appropriate */
add_line(resp, m_video-str);
add_line(resp, a_video-str);
add_line(resp, hold); /* Repeat hold for the video stream */
} else if (p-offered_media[SDP_VIDEO].offered) {
snprintf(dummy_answer, sizeof(dummy_answer), m=video 0 RTP/AVP\
%s\r\n, p-offered_media[SDP_VIDEO].text);
add_line(resp, dummy_answer);
But len, which was used to set Content-Length, isn't updated to onclide
that dummy. Doesn't it need to be?
I think this may be a problem with a connection to my Polycom VSX.
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Re: [asterisk-users] Possible bug in chan_sip:add_sdp
I can confirm that the following fixes my problem: --- chan_sip.c (revision 261450) +++ chan_sip.c (working copy) @@ -10357,12 +10357,22 @@ strlen(connection) + strlen(session_time); if (needaudio) len += m_audio-used + a_audio-used + strlen(hold); + else if (p-offered_media[SDP_AUDIO].offered) + len += strlen(m=audio 0 RTP/AVP \r\n) + strlen(p-offered_media[SDP_AUDIO].text); + if (needvideo) /* only if video response is appropriate */ len += m_video-used + a_video-used + strlen(bandwidth) + strlen(hold); + else if (p-offered_media[SDP_VIDEO].offered) + len += strlen(m=video 0 RTP/AVP \r\n) + strlen(p-offered_media[SDP_VIDEO].text); + if (needtext) /* only if text response is appropriate */ len += m_text-used + a_text-used + strlen(hold); + else if (p-offered_media[SDP_TEXT].offered) + len += strlen(m=text 0 RTP/AVP \r\n) + strlen(p-offered_media[SDP_TEXT].text); if (add_t38) len += m_modem-used + a_modem-used; + else if (p-offered_media[SDP_IMAGE].offered) + len += strlen(m=image 0 udptl t38\r\n); add_header(resp, Content-Type, application/sdp); add_header_contentLength(resp, len); -- _ -- 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] Possible bug in app_meetme.c
Is this patch correct? The doesn't make logical sense to me. I think
it should be || and making this change fixes the problem I have with SIP
phones in MeetMe conferences. If it's correct, is there someplace more
formal that I should submit it to?
*** app_meetme.c.old2009-10-11 17:56:44.0 -0400
--- app_meetme.c2009-10-17 14:22:29.0 -0400
***
*** 2901,2905
to write out all the samples.
*/
! if (user-talking
!(confflags CONFFLAG_OPTIMIZETALKER)) {
careful_write(fd,
f-data.ptr, f-datalen, 0);
}
--- 2901,2905
to write out all the samples.
*/
! if (user-talking ||
!(confflags CONFFLAG_OPTIMIZETALKER)) {
careful_write(fd,
f-data.ptr, f-datalen, 0);
}
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Re: [asterisk-users] Possible bug in app_meetme.c
On Sat, Oct 17, 2009 at 2:26 PM, Richard Kenner ken...@gnat.com wrote: Is this patch correct? The doesn't make logical sense to me. I think it should be || and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? I don't know if it's correct since I don't know what the CONFFLAG_OPTIMIZETALKER flag does nor what your problem is. Giving more background info should help. If you want to submit a patch use issues.asterisk.org and read the guidelines before submitting the bug report. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- 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] Possible bug in app_meetme.c
On Sat, Oct 17, 2009 at 2:26 PM, Richard Kenner ken...@gnat.com wrote:
Is this patch correct? The doesn't make logical sense to me. I think
it should be || and making this change fixes the problem I have with SIP
phones in MeetMe conferences. If it's correct, is there someplace more
formal that I should submit it to?
*** app_meetme.c.old 2009-10-11 17:56:44.0 -0400
--- app_meetme.c 2009-10-17 14:22:29.0 -0400
***
*** 2901,2905
to write out all the
samples.
*/
! if (user-talking
!(confflags CONFFLAG_OPTIMIZETALKER)) {
careful_write(fd,
f-data.ptr, f-datalen, 0);
}
--- 2901,2905
to write out all the
samples.
*/
! if (user-talking ||
!(confflags CONFFLAG_OPTIMIZETALKER)) {
careful_write(fd,
f-data.ptr, f-datalen, 0);
}
I have a limited acquaintance with the app_meetme.c code, having also
modified it for my own purposes in the past.
For some historical perspective, you should know that originally in
1.6, talker optimization was forced on, with no way to turn it off.
Many people reported that it sounded awful, and patches were released
to make optimization be off by default, with a flag to turn it on, the
way it was in 1.4
There are more details, especially as regards versions were various
patches from trunk were merged into the releases, but that's the very
quick version. If you want bug numbers, please check out
http://issues.asterisk.org and search for talker optimization and
meetme.
When I first tried to use MeetMe with 1.6, I had a lot of
voice-cut-out problems, which I was able to verify were caused by
over-aggressive talker optimization, resulting in me going to 1.6.1.*
series as that series at that time had the 'optimization off by
default' feature that I wanted.
From a quick glance at your patch, I would say that it probably tries
to address the audio quality problems I and others were experiencing.
It's possible, and probably likely that somebody else already fixed
this. Please be sure to check through the issues, and make sure you
compare your work against app_meetme.c in trunk, as your work will NOT
make it into asterisk by posting it to asterisk-users. If you didn't
already know, you should know that SVN is often way ahead of the
1.6.0. series for many reasons. For the latest versions for each 1.6
series, you should check through:
http://svnview.digium.com/svn/asterisk/branches/
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Re: [asterisk-users] Possible bug in app_meetme.c
On Sat, Oct 17, 2009 at 2:26 PM, Richard Kenner ken...@gnat.com wrote: Is this patch correct? The doesn't make logical sense to me. I think it should be || and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? So now I actually read through your patch. Please read my last post about getting up to speed with how the history of talker optimization with MeetMe() in 1.6 has gone. Having said that, I think you may be using talker optimization and that this was causing the same voice cut out problem I was having. What version are you running? Does that version support disabling talker optimization? Have you tried disabling talker optimization? I appreciate your work on this, but I fear your issues may have already been solved a rather long time ago. ___ -- 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] Possible bug in app_meetme.c
David Backeberg wrote: From a quick glance at your patch, I would say that it probably tries to address the audio quality problems I and others were experiencing. No, it's fixing a much more serious issue. As I sent to this list twice, when I have a conference between Dahdi ports and SIP phones, the people on the SIP phones were never heard. I had the 'o' option specified. The code, as written, says for non-Dahdi (e.g., SIP phones), send to the channel if talking AND not optimizing talkers. It seems to me that it should send to the challel if talking OR not optimizing talkers. ___ -- 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] Possible bug in app_meetme.c
What version are you running? 1.6.2.0-rc2 Does that version support disabling talker optimization? Yes. Have you tried disabling talker optimization? Yes. That's how I found the bug. I got no audio from the SIP phone into the conference, so I decided I'd try seeing if it did if the SIP phone specified no options. I did. So I turned them on one at time until I saw that it was the talker optimization that was causing the complete lack of audion. Then I looked at the only two uses of that flag alone and found what looked like a logic bug. ___ -- 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] Possible Bug?
Tzafrir Cohen, I'm not a linux expert, i just wanted to share what happened to me... Only after new detection of Sound Board, i got the audio from Asterisk services like MOH, Voicemail busy message ... working fine. Probably there's a better faster way to run kudzu, i tried from command line and it didn't detect my audio board... On 7/6/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Jul 06, 2006 at 02:27:59AM +0100, Marco Mouta wrote: it has happen to me , no sound after removing x100p board , and i found, only because i was not accessing remote the server. i was localy in the server, and in boot i detect that kudzu had removed my sound board from the config... all running well lbut no audio for playing files like voicemail i've just unplugged sound card, then boot,then plug the card, new kudzu detection, and all begun working fine... And you need a local sound card for? I'm also surprised there is no faster way to run kudzu than to (yuck) reboot. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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] Possible Bug?
Tzafrir Cohen wrote: On Thu, Jul 06, 2006 at 02:27:59AM +0100, Marco Mouta wrote: I'm also surprised there is no faster way to run kudzu than to (yuck) reboot. How about just typing /usr/sbin/kudzu as root from a console? W ___ --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] Possible Bug?
On Thu, Jul 06, 2006 at 09:52:57AM +0100, Marco Mouta wrote: Tzafrir Cohen, I'm not a linux expert, i just wanted to share what happened to me... Only after new detection of Sound Board, i got the audio from Asterisk services like MOH, Voicemail busy message ... working fine. Probably there's a better faster way to run kudzu, i tried from command line and it didn't detect my audio board... To check for the problem of zaptel timing, run zttest. I really don't see how this is related to a sound board (Unless you're using a single chan_alsa for announcements and such). -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Possible Bug?
I believe Ive found a possible bug in the Zaptel channel drivers. Ive been able to recreate this on a couple of servers. One with Asterisk 1.2.7.1 and 1.2.9.1,and also Zaptel 1.2.5 and Zaptel 1.2.6. I was testing a server configuration with some T1s we have with ATT. When I disconnected the ATT T1s we were no longer able to check voicemail. I noticed then that no messages from the server could be heard. Playback, voicemail or any other message the server would play to the caller on the phone. It would display on the CLI that it was playing the message but nothing could be heard and it would hang the call until it timed out. It took a lot of trial and error before I figured out that if I unloaded all related files to the Zaptel drivers the messages could then be heard. Reactivated the Zaptel with no T1s attached and again it killed the messages. One server has a Wct4xxp card and the other using dynamic with Redfones from Fonbridge. Both reacted the same way. This would not be good if we had a fiber cut somewhere and then our users could no longer use their voicemail or any other application that played sounds. Can anyone else recreate this? Do any of you think this is a bug? Should I submit a bug report? Ive heard to submit a bug to send it to mantis. What is this and where or how do I do this? 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] Possible Bug?
I believe I've found a possible bug in the Zaptel channel drivers. I've been able to recreate this on a couple of servers. One with Asterisk 1.2.7.1 and 1.2.9.1,and also Zaptel 1.2.5 and Zaptel 1.2.6. I was testing a server configuration with some T1's we have with ATT. When I disconnected the ATT T1's we were no longer able to check voicemail. I noticed then that no messages from the server could be heard. Playback, voicemail or any other message the server would play to the caller on the phone. It would display on the CLI that it was playing the message but nothing could be heard and it would hang the call until it timed out. It took a lot of trial and error before I figured out that if I unloaded all related files to the Zaptel drivers the messages could then be heard. Reactivated the Zaptel with no T1's attached and again it killed the messages. One server has a Wct4xxp card and the other using dynamic with Redfone's from Fonbridge. Both reacted the same way. Sounds more like you have set the T1s as your primary timing source with no secondary. Are the phones connected to this analog off a channel bank by chance? If so if you don't specify something to take over timing, the channel banks won't sync audio properly to the Asterisk server and what you describe is likely to occur. Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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] Possible Bug?
I actually have a server that *may* have exhibited the same problem. I can't say for certain as the system itself was acting a bit wonky, but I can say for certain that after we disconnected the PRI (actually, switched it over to the secondary box in our cluster) that users connected to that system (as opposed to the secondary one) could not hear anything. This was with SIP phones as well, not analog (Polycom 601s to be more precise). Regards, - Brad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin SavoySent: Wednesday, July 05, 2006 4:15 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [asterisk-users] Possible Bug? I believe Ive found a possible bug in the Zaptel channel drivers. Ive been able to recreate this on a couple of servers. One with Asterisk 1.2.7.1 and 1.2.9.1,and also Zaptel 1.2.5 and Zaptel 1.2.6. I was testing a server configuration with some T1s we have with ATT. When I disconnected the ATT T1s we were no longer able to check voicemail. I noticed then that no messages from the server could be heard. Playback, voicemail or any other message the server would play to the caller on the phone. It would display on the CLI that it was playing the message but nothing could be heard and it would hang the call until it timed out. It took a lot of trial and error before I figured out that if I unloaded all related files to the Zaptel drivers the messages could then be heard. Reactivated the Zaptel with no T1s attached and again it killed the messages. One server has a Wct4xxp card and the other using dynamic with Redfones from Fonbridge. Both reacted the same way. This would not be good if we had a fiber cut somewhere and then our users could no longer use their voicemail or any other application that played sounds. Can anyone else recreate this? Do any of you think this is a bug? Should I submit a bug report? Ive heard to submit a bug to send it to mantis. What is this and where or how do I do this? 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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Possible Bug?
it has happen to me , no sound after removing x100p board , and i found, only because i was not accessing remote the server. i was localy in the server, and in boot i detect that kudzu had removed my sound board from the config... all running well lbut no audio for playing files like voicemail i've just unplugged sound card, then boot,then plug the card, new kudzu detection, and all begun working fine... be aware if u r remote acess, you won't notice this kudzu issue on audio board detection... hope it helps! On 7/5/06, Watkins, Bradley [EMAIL PROTECTED] wrote: I actually have a server that *may* have exhibited the same problem. I can't say for certain as the system itself was acting a bit wonky, but I can say for certain that after we disconnected the PRI (actually, switched it over to the secondary box in our cluster) that users connected to that system (as opposed to the secondary one) could not hear anything. This was with SIP phones as well, not analog (Polycom 601s to be more precise). Regards, - Brad From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Kevin SavoySent: Wednesday, July 05, 2006 4:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [asterisk-users] Possible Bug? I believe I've found a possible bug in the Zaptel channel drivers. I've been able to recreate this on a couple of servers. One with Asterisk 1.2.7.1 and 1.2.9.1,and also Zaptel 1.2.5 and Zaptel 1.2.6. I was testing a server configuration with some T1's we have with ATT. When I disconnected the ATT T1's we were no longer able to check voicemail. I noticed then that no messages from the server could be heard. Playback, voicemail or any other message the server would play to the caller on the phone. It would display on the CLI that it was playing the message but nothing could be heard and it would hang the call until it timed out. It took a lot of trial and error before I figured out that if I unloaded all related files to the Zaptel drivers the messages could then be heard. Reactivated the Zaptel with no T1's attached and again it killed the messages. One server has a Wct4xxp card and the other using dynamic with Redfone's from Fonbridge. Both reacted the same way. This would not be good if we had a fiber cut somewhere and then our users could no longer use their voicemail or any other application that played sounds. Can anyone else recreate this? Do any of you think this is a bug? Should I submit a bug report? I've heard to submit a bug to send it to mantis. What is this and where or how do I do this? 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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___ --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] Possible Bug?
On Thu, Jul 06, 2006 at 02:27:59AM +0100, Marco Mouta wrote: it has happen to me , no sound after removing x100p board , and i found, only because i was not accessing remote the server. i was localy in the server, and in boot i detect that kudzu had removed my sound board from the config... all running well lbut no audio for playing files like voicemail i've just unplugged sound card, then boot,then plug the card, new kudzu detection, and all begun working fine... And you need a local sound card for? I'm also surprised there is no faster way to run kudzu than to (yuck) reboot. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Possible Bug in SIP Stack.
I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is APX 8000 - Interaction SIP Proxy 3.0.013 - asterisk server. When I use Asterisk version 10.0.10 everything works perfectly, however when I use 1.2.4 I lose the ability to receive calls from the PSTN. All I get is the following error in my SIP Proxies error logs: SIPSession::proxyResponseImmediately(): Failed to retrieve next Via, don't know where to send responseSIP/2.0 180 Ringing From: MODESITT,CHRIS sip:[EMAIL PROTECTED]:5060;user=phone;tag=4fdc9d0e-1e600f94-ed7e623f To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as4fc8aa8a Call-ID: [EMAIL PROTECTED] CSeq: 5466974 INVITE User-Agent: Asterisk PBX I still can make outbound calls with no-problems, any ideas? Thanks Chris ___ --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] Possible bug with GotoIfTime
Running a fairly recent subversion release of Asterisk, I'm running into a problem using labels (as opposed to priorities) with this application. Here is the dialplan segment: ; isolate gotoiftime bug with labels ;exten = 806,1,GotoIfTime(8:00-20:00,*,*,*?2:4) exten = 806,1,GotoIfTime(8:00-20:00,*,*,*?light:dark) exten = 806,n(light),noop(light) exten = 806,n,hangup exten = 806,n(dark),noop(dark) exten = 806,n,hangup As coded, this is what happens when it executes: -- Executing GotoIfTime(IAX2/hack-2, 8:00-20:00|*|*|*?light:dark) in new stack Jan 7 18:38:09 NOTICE[28137]: pbx.c:1705 pbx_extension_helper: No such label 'light:dark' in extension '806' in context 'default' Jan 7 18:38:09 WARNING[28137]: pbx.c:6312 ast_parseable_goto: Priority 'light:dark' must be a number 0, or valid label == Spawn extension (default, 806, 1) exited non-zero on 'IAX2/hack-2' -- Hungup 'IAX2/hack-2' But if I disable the second exten line instead of the first, it works properly. Beware. ___ --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] Possible bug with GotoIfTime
Bill Michaelson wrote: Running a fairly recent subversion release of Asterisk, I'm running into a problem using labels (as opposed to priorities) with this application. Here is the dialplan segment: ; isolate gotoiftime bug with labels ;exten = 806,1,GotoIfTime(8:00-20:00,*,*,*?2:4) exten = 806,1,GotoIfTime(8:00-20:00,*,*,*?light:dark) exten = 806,n(light),noop(light) exten = 806,n,hangup exten = 806,n(dark),noop(dark) exten = 806,n,hangup As coded, this is what happens when it executes: -- Executing GotoIfTime(IAX2/hack-2, 8:00-20:00|*|*|*?light:dark) in new stack Jan 7 18:38:09 NOTICE[28137]: pbx.c:1705 pbx_extension_helper: No such label 'light:dark' in extension '806' in context 'default' Jan 7 18:38:09 WARNING[28137]: pbx.c:6312 ast_parseable_goto: Priority 'light:dark' must be a number 0, or valid label == Spawn extension (default, 806, 1) exited non-zero on 'IAX2/hack-2' -- Hungup 'IAX2/hack-2' But if I disable the second exten line instead of the first, it works properly. Beware. ___ --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 GotoIfTime(8:00-20:00|*|*|*?light:dark|s|1) -- . -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- . signature.asc Description: OpenPGP digital signature ___ --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] Possible bug in record?
Im trying to get record to append to a file, Im
using this:-
exten = 2,n,record(/tmp/${UNIQUEID}.gsm|5|0|a)
And its creating a new file?
If I check /tmp I can see the same filename being reused
each time, but the file jjust contains the latest recording.
Can anybody else confirm this?
Steve
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Re: [Asterisk-Users] Possible bug in record?
Steve Hanselman wrote:
I’m trying to get record to append to a file, I’m using this:-
exten = 2,n,record(/tmp/${UNIQUEID}.gsm|5|0|a)
And it’s creating a new file?
I don't think record has an option to append. Use a script and cat the two
files together.
Doug
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[Asterisk-Users] Possible Bug in Asterisk 1.2.0 with Queues and MOH
I think I discovered a bug. I have a dual Xeon machine running * 1.2.0 I have a queue defined to play the default music on hold class, which simply plays an mp3 file. When a call comes into the queue (note that there are no agents logged in, but I have joinempty=yes and leavewhenempty=no in queues.conf) * answers, puts the call in the queue and the caller is holding listening to MOH. What's interesting is that there is no MOH playing, or at least nothing is heard. When a second call comes in, the new caller does in fact hear music on hold. When the first caller hangs up while the second caller is still waiting, the MOH the second caller was listening to stops playing until another call comes in and is put into the queue. It seems that the music on hold is only heard when there are two or more callers waiting in the queue. Has anyone else experienced this? Also, while testing this, I was running vmstat 1 on a separate window and it showed me some interesting things about CPU utilization. When there are no calls, CPU is 100% idle. When the first call comes in, CPU utilization drops to 50% idle, which, in my case, I think one of the CPUs is 100% busy and the other is 100% idle. When additional calls come in, it only drops the idle % by 1-2 percent per call. When the calls hang up, CPU idle goes back to 100%. At first, I thought it could be mpg123 taking all the CPU but running top shows a single (among many) asterisk process using 99.9% of one of the CPUs. This may be another bug. Any comments/ideas? Thanks, Waldo ___ --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] Possible bug in agent monitoring
On Tue, 2005-11-15 at 23:11 +, Julian Lyndon-Smith wrote: Done. Done. Done :) http://bugs.digium.com/view.php?id=5762 Julian. In case anyone is interested, this has worked like this for at least 18 months, I had the same problem, so I just changed to using Monitor before dropping the call into the queue. Actually, I just added a monitor just after we answer the call, that was, the entire call is recorded. I simply assumed that it was meant to work that way, since if using the queue, or agent, to record, then the call is actually finished at the point it is transferred. ie, the call is no longer an agent call, just a normal ext call, and it isn't in a queue either PS, using the above method, it can be useful^h^h^h^h^h^hinteresting to listen to what the caller is saying/etc while they are on hold as well :) Regards, Adam ___ --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
Re: [Asterisk-Users] Possible bug in agent monitoring
Bugger. I had hoped that it was a recent bug .. As a matter of interest, how do you know which agents have answered the call ? I liked having the agent number as part of the monitored filename. Julian. Adam Goryachev wrote: On Tue, 2005-11-15 at 23:11 +, Julian Lyndon-Smith wrote: Done. Done. Done :) http://bugs.digium.com/view.php?id=5762 Julian. In case anyone is interested, this has worked like this for at least 18 months, I had the same problem, so I just changed to using Monitor before dropping the call into the queue. Actually, I just added a monitor just after we answer the call, that was, the entire call is recorded. I simply assumed that it was meant to work that way, since if using the queue, or agent, to record, then the call is actually finished at the point it is transferred. ie, the call is no longer an agent call, just a normal ext call, and it isn't in a queue either PS, using the above method, it can be useful^h^h^h^h^h^hinteresting to listen to what the caller is saying/etc while they are on hold as well :) Regards, Adam ___ --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 ___ --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
Re: [Asterisk-Users] Possible bug in agent monitoring
On Wed, 2005-11-16 at 11:26 +, Julian Lyndon-Smith wrote: Bugger. I had hoped that it was a recent bug .. As a matter of interest, how do you know which agents have answered the call ? I liked having the agent number as part of the monitored filename. Julian. Well, in my case, there are only a very small number of people who work here, so simply listening to the recording is enough to recognise the voice. However, CDR records also supplement this for automated searching. I was concerned back at the beginning, but since I switched to using gsm format for the recordings, I haven't bothered looking into how to avoid recording while the caller is waiting in the queue (size of gsm is approx 1/10 th compared to wav). Regards, Adam ___ --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] Possible bug in agent monitoring
using CVS-HEAD (CVS-D2005.10.28.07.54.07-11/13/05-08:33:54) We monitor (record) all inbound calls to our queues, using recordagentcalls=yes and recordformat=gsm in the agents.conf file. If a call comes in to a queue, and is answered by an agent (let's say 6001) then I have a recording for agent-6001-xxx-yyy.gsm. if the agent wants to transfer the call to another agent (an attended xfer), then the recording is terminated at the exact time the inbound call is transferred to the second agent. Anyone seen this ? Julian ___ --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
Re: [Asterisk-Users] Possible bug in agent monitoring
Bad form replying to my own post, but I'm getting my ass chewed here because we need to listen to a call I think has been trashed ;) I've now tried using the Monitor command: 1) Incoming Call Answered By Extension A 2) Conversation between caller and A 3) A decides to transfer call to B 4) A talks to B 5) A transfers to B 6) B has conversation with caller 7) Hangup. This results in the full call being recorded. (up to point 7) 1) Incoming Call Answered By Queue 2) Queue Calls Agent A 3) Agent A confirms ready to talk 4) Conversation between caller and Agent A 5) Agent A decides to transfer call to Extension B 6) Agent A talks to extension B 7) Agent A transfers to extension B 8) Extension B has conversation with caller 9) Hangup. This results in the call being recorded, but only up to point 6 (actuall y to point 5 - between 5 and 6 is only the sound from the caller, i.e. not the conversation between A and B - that's a different call) As you can see, we are now missing the conversation between the caller and Extension B. I really need an answer to this - even if it is a pointer to the code so that I can start looking for the problem. I would do away with the agent recording and record through the dialplan, but I have some queues that use members and not agents. AARRGGHH. ;) Julian Julian Lyndon-Smith wrote: using CVS-HEAD (CVS-D2005.10.28.07.54.07-11/13/05-08:33:54) We monitor (record) all inbound calls to our queues, using recordagentcalls=yes and recordformat=gsm in the agents.conf file. If a call comes in to a queue, and is answered by an agent (let's say 6001) then I have a recording for agent-6001-xxx-yyy.gsm. if the agent wants to transfer the call to another agent (an attended xfer), then the recording is terminated at the exact time the inbound call is transferred to the second agent. Anyone seen this ? Julian ___ --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 ___ --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
Re: [Asterisk-Users] Possible bug in agent monitoring
On 11/15/05, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: Bad form replying to my own post, but I'm getting my ass chewed here because we need to listen to a call I think has been trashed ;) I've now tried using the Monitor command: 1) Incoming Call Answered By Extension A 2) Conversation between caller and A 3) A decides to transfer call to B 4) A talks to B 5) A transfers to B 6) B has conversation with caller 7) Hangup. This results in the full call being recorded. (up to point 7) 1) Incoming Call Answered By Queue 2) Queue Calls Agent A 3) Agent A confirms ready to talk 4) Conversation between caller and Agent A 5) Agent A decides to transfer call to Extension B 6) Agent A talks to extension B 7) Agent A transfers to extension B 8) Extension B has conversation with caller 9) Hangup. This results in the call being recorded, but only up to point 6 (actuall y to point 5 - between 5 and 6 is only the sound from the caller, i.e. not the conversation between A and B - that's a different call) As you can see, we are now missing the conversation between the caller and Extension B. I really need an answer to this - even if it is a pointer to the code so that I can start looking for the problem. I would do away with the agent recording and record through the dialplan, but I have some queues that use members and not agents. AARRGGHH. ;) Julian Julian Lyndon-Smith wrote: using CVS-HEAD (CVS-D2005.10.28.07.54.07-11/13/05-08:33:54) We monitor (record) all inbound calls to our queues, using recordagentcalls=yes and recordformat=gsm in the agents.conf file. If a call comes in to a queue, and is answered by an agent (let's say 6001) then I have a recording for agent-6001-xxx-yyy.gsm. if the agent wants to transfer the call to another agent (an attended xfer), then the recording is terminated at the exact time the inbound call is transferred to the second agent. Anyone seen this ? Please open a bug on http://bugs.digium.com/ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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
Re: [Asterisk-Users] Possible bug in agent monitoring
Done. Done. Done :) http://bugs.digium.com/view.php?id=5762 Julian. BJ Weschke wrote: On 11/15/05, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: Bad form replying to my own post, but I'm getting my ass chewed here because we need to listen to a call I think has been trashed ;) I've now tried using the Monitor command: 1) Incoming Call Answered By Extension A 2) Conversation between caller and A 3) A decides to transfer call to B 4) A talks to B 5) A transfers to B 6) B has conversation with caller 7) Hangup. This results in the full call being recorded. (up to point 7) 1) Incoming Call Answered By Queue 2) Queue Calls Agent A 3) Agent A confirms ready to talk 4) Conversation between caller and Agent A 5) Agent A decides to transfer call to Extension B 6) Agent A talks to extension B 7) Agent A transfers to extension B 8) Extension B has conversation with caller 9) Hangup. This results in the call being recorded, but only up to point 6 (actuall y to point 5 - between 5 and 6 is only the sound from the caller, i.e. not the conversation between A and B - that's a different call) As you can see, we are now missing the conversation between the caller and Extension B. I really need an answer to this - even if it is a pointer to the code so that I can start looking for the problem. I would do away with the agent recording and record through the dialplan, but I have some queues that use members and not agents. AARRGGHH. ;) Julian Julian Lyndon-Smith wrote: using CVS-HEAD (CVS-D2005.10.28.07.54.07-11/13/05-08:33:54) We monitor (record) all inbound calls to our queues, using recordagentcalls=yes and recordformat=gsm in the agents.conf file. If a call comes in to a queue, and is answered by an agent (let's say 6001) then I have a recording for agent-6001-xxx-yyy.gsm. if the agent wants to transfer the call to another agent (an attended xfer), then the recording is terminated at the exact time the inbound call is transferred to the second agent. Anyone seen this ? Please open a bug on http://bugs.digium.com/ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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 ___ --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] possible bug, what do you think?
We recently changed file formats on our server to wav49 from gsm. Several users had saved messages in gsm format. When a user attempts to forward an old message to a user and they prepend the message with a recording, the process seems to be flawed. From what I can tell, the prepend message is recorded to a temporary file, in my case msg-prepend.WAV then after the prepend is finished recording, asterisk attempts to merge the two audio files into one. Since it cannot find a msg.WAV file (the file is msg.gsm) it throws an error. The end result is that the user gets a new message envelope in their INBOX (msg.txt) but there is no associated .WAV file to go along with it. The desired behavior here is to a) notify the user who is attempting to forward this message that the process failed so that the asterisk admin (me) can fix the issue or b) convert the file to the proper format and then merge the two together. What do you all think? Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --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
Re: [Asterisk-Users] possible bug, what do you think?
Andy Goss wrote: error. The end result is that the user gets a new message envelope in their INBOX (msg.txt) but there is no associated .WAV file to go along with it. The desired behavior here is to a) notify the user who is attempting to forward this message that the process failed so that the asterisk admin (me) can fix the issue or b) convert the file to the proper format and then merge the two together. What do you all think? Yes, that is a bug, but there is no simple way to handle it. app_voicemail would have to work quite a bit differently to make this work seamlessly. Basically, what you need to do when you change the formats in your voicemail.conf file is manually go through the spool area and ensure that every message is present in all the selected formats, just as if those formats had been selected initially. Otherwise you can get very bizarre behavior, depending on the codec in use when a user deals with the mailbox. ___ --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] Possible bug in meetme when hangup
When I callin to a meetme (first one) and it prompts for a PIN access if I hangup at that point the meetme is created with 0 parties. I would think if I hangup it should not create the meetme yet? If I go ahead and enter the valid PIN it says I am the only in in the meetme, then I hangup and the meetme is no longer active. Is that a bug?? It seems like hanging up before the pin is entered should also destroy the meetme if there are no parties in the meetme. Just curious. Jerry ___ 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
Re: [Asterisk-Users] possible bug in chan_capi concerning context handling
On Sun, 13 Mar 2005 21:49:52 +0200, Dimitris Kounalakis [EMAIL PROTECTED] wrote: Hello *Martijn, Thank you for your response. *That was my opinion too, it looses the context due to a bug, and can anyone confirm it also? But I have no output from the command Show channels, and it happens so quickly that it is impossible to issue the command before falling to the default context. In the logs, I can see that the channel exists like CAPI[contr1/2810211694]/0 but this is druring call only. Any other way to debug it more (or to solve it)? My /etc/asterisk/capi.conf is: - [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] controller=1 msn=2810111694 incomingmsn=* devices=2 softdtmf=1 callgroup=1 context=isdn On my system I have the devices=2 as the last line this works for me [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=330417 incomingmsn=* controller=1 softdtmf=1 accountcode= context=from-pstn echocancel=yes echotail=64 devices=2 ___ 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
Re: [Asterisk-Users] possible bug in chan_capi concerning context handling
Thank you for your response Marco.
I do. The problem is that all incomings calls from ISDN are handled by the
default s extension in the context [default] and not by an s extension
in the context [isdn] or by the msm numbers as extensions in the context
[isdn].
So, what is the reason for the context directive in /etc/asterisk/capi.conf
to exit? What exactly is it used for?
Dimitris
*Marco Supino wrote:*
-
Do you have an 's' extention in the default context ?
Marco.
Dimitris Kounalakis wrote:
/ Hello,
// I am trying to configure asterisk 1.0.7pre to get incoming calls from an
// ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is
// that the context is not recognised in the /etc/asterisk/capi.conf
// I have in /etc/asterisk/capi.conf 's section [interfaces] the
// following directive
// context=isdn
//
// and the following directive in /etc/asterisk/extensions.conf in the
// context [isdn]
// [isdn]
// exten = s,1,Dial(SIP/${DNID:4},60,tr)
//
//
// Here follows the debug info I get when an incoming call starts:
//
//
// == CONNECT_IND
// (PLCI=0x101,DID=2810111694,CID=2810111694,CIP=0x1,CONTROLLER=0x1)
//-- creating pipe for PLCI=0x101 msn = 2810111694
//sent ALERT_REQ PLCI = 0x101
// == Starting CAPI[contr1/2810111694]/3 at ,2810111694,1 failed so
// falling back to exten 's'
// == Starting CAPI[contr1/2810111694]/3 at ,s,1 still failed so falling
// back to context 'default'
// Mar 13 11:52:41 WARNING[10744]: pbx.c:1893 ast_pbx_run: Channel
// 'CAPI[contr1/2810111694]/3' sent into invalid extension 's' in context
// 'default', but no invalid handler
//-- CAPI Hangingup
// -
//
// When I move the exten = s,1,Dial(${DNID:4},60,tr) in the context
// [default] of the /etc/asterisk/extensions.conf, I get the following
// debug info and the sip phone rings ok:
// --
//
// == CONNECT_IND
// (PLCI=0x101,DID=2810111694,CID=2810111694,CIP=0x1,CONTROLLER=0x1)
//-- creating pipe for PLCI=0x101 msn = 2810111694
//sent ALERT_REQ PLCI = 0x101
// == Starting CAPI[contr1/2810111694]/4 at ,2810111694,1 failed so
// falling back to exten 's'
// == Starting CAPI[contr1/2810111694]/4 at ,s,1 still failed so falling
// back to context 'default'
//-- Executing Dial(CAPI[contr1/2810111694]/4, SIP/111694|60|tr) in
// new stack
//-- Called 111694
// --
//
//
// Is this a bug? It does not handle the context, so, it can not find what
// to do, it works only with the default context.
//
// Thank you in advance,
// Dimitris
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Re: [Asterisk-Users] possible bug in chan_capi concerning context handling
On Sun, Mar 13, 2005 at 06:44:52PM +0200, Dimitris Kounalakis wrote: Thank you for your response Marco. I do. The problem is that all incomings calls from ISDN are handled by the default s extension in the context [default] and not by an s extension in the context [isdn] or by the msm numbers as extensions in the context [isdn]. Looking at the line here: // == Starting CAPI[contr1/2810111694]/3 at ,2810111694,1 failed so // falling back to exten 's' It looks like the context is blank. What does the show command in asterisk show the context as being (paste output please). -- Martijn van Oosterhout Ecomtel Pty Ltd ___ 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
Re: [Asterisk-Users] possible bug in chan_capi concerning context handling
Hello *Martijn, Thank you for your response. *That was my opinion too, it looses the context due to a bug, and can anyone confirm it also? But I have no output from the command Show channels, and it happens so quickly that it is impossible to issue the command before falling to the default context. In the logs, I can see that the channel exists like CAPI[contr1/2810211694]/0 but this is druring call only. Any other way to debug it more (or to solve it)? My /etc/asterisk/capi.conf is: - [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] controller=1 msn=2810111694 incomingmsn=* devices=2 softdtmf=1 callgroup=1 context=isdn -- *Martijn van Oosterhout wrote: *On Sun, Mar 13, 2005 at 06:44:52PM +0200, Dimitris Kounalakis wrote: / Thank you for your response Marco. // // I do. The problem is that all incomings calls from ISDN are handled by the // default s extension in the context [default] and not by an s extension // in the context [isdn] or by the msm numbers as extensions in the context // [isdn]. / Looking at the line here: / // == Starting CAPI[contr1/2810111694]/3 at ,2810111694,1 failed so // // falling back to exten 's' / It looks like the context is blank. What does the show command in asterisk show the context as being (paste output please). -- Martijn van Oosterhout Ecomtel Pty Ltd ___ 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] possible bug in chan_capi concerning context handling
Hello,
I am trying to configure asterisk 1.0.7pre to get incoming calls from an
ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is
that the context is not recognised in the /etc/asterisk/capi.conf
I have in /etc/asterisk/capi.conf 's section [interfaces] the
following directive
context=isdn
and the following directive in /etc/asterisk/extensions.conf in the
context [isdn]
[isdn]
exten = s,1,Dial(SIP/${DNID:4},60,tr)
Here follows the debug info I get when an incoming call starts:
== CONNECT_IND
(PLCI=0x101,DID=2810111694,CID=2810111694,CIP=0x1,CONTROLLER=0x1)
-- creating pipe for PLCI=0x101 msn = 2810111694
sent ALERT_REQ PLCI = 0x101
== Starting CAPI[contr1/2810111694]/3 at ,2810111694,1 failed so
falling back to exten 's'
== Starting CAPI[contr1/2810111694]/3 at ,s,1 still failed so falling
back to context 'default'
Mar 13 11:52:41 WARNING[10744]: pbx.c:1893 ast_pbx_run: Channel
'CAPI[contr1/2810111694]/3' sent into invalid extension 's' in context
'default', but no invalid handler
-- CAPI Hangingup
-
When I move the exten = s,1,Dial(${DNID:4},60,tr) in the context
[default] of the /etc/asterisk/extensions.conf, I get the following
debug info and the sip phone rings ok:
--
== CONNECT_IND
(PLCI=0x101,DID=2810111694,CID=2810111694,CIP=0x1,CONTROLLER=0x1)
-- creating pipe for PLCI=0x101 msn = 2810111694
sent ALERT_REQ PLCI = 0x101
== Starting CAPI[contr1/2810111694]/4 at ,2810111694,1 failed so
falling back to exten 's'
== Starting CAPI[contr1/2810111694]/4 at ,s,1 still failed so falling
back to context 'default'
-- Executing Dial(CAPI[contr1/2810111694]/4, SIP/111694|60|tr)
in new stack
-- Called 111694
--
Is this a bug? It does not handle the context, so, it can not find what
to do, it works only with the default context.
Thank you in advance,
Dimitris
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Re: [Asterisk-Users] possible bug in chan_capi concerning context handling
Do you have an 's' extention in the default context ?
Marco.
Dimitris Kounalakis wrote:
Hello,
I am trying to configure asterisk 1.0.7pre to get incoming calls from an
ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is
that the context is not recognised in the /etc/asterisk/capi.conf
I have in /etc/asterisk/capi.conf 's section [interfaces] the
following directive
context=isdn
and the following directive in /etc/asterisk/extensions.conf in the
context [isdn]
[isdn]
exten = s,1,Dial(SIP/${DNID:4},60,tr)
Here follows the debug info I get when an incoming call starts:
== CONNECT_IND
(PLCI=0x101,DID=2810111694,CID=2810111694,CIP=0x1,CONTROLLER=0x1)
-- creating pipe for PLCI=0x101 msn = 2810111694
sent ALERT_REQ PLCI = 0x101
== Starting CAPI[contr1/2810111694]/3 at ,2810111694,1 failed so
falling back to exten 's'
== Starting CAPI[contr1/2810111694]/3 at ,s,1 still failed so falling
back to context 'default'
Mar 13 11:52:41 WARNING[10744]: pbx.c:1893 ast_pbx_run: Channel
'CAPI[contr1/2810111694]/3' sent into invalid extension 's' in context
'default', but no invalid handler
-- CAPI Hangingup
-
When I move the exten = s,1,Dial(${DNID:4},60,tr) in the context
[default] of the /etc/asterisk/extensions.conf, I get the following
debug info and the sip phone rings ok:
--
== CONNECT_IND
(PLCI=0x101,DID=2810111694,CID=2810111694,CIP=0x1,CONTROLLER=0x1)
-- creating pipe for PLCI=0x101 msn = 2810111694
sent ALERT_REQ PLCI = 0x101
== Starting CAPI[contr1/2810111694]/4 at ,2810111694,1 failed so
falling back to exten 's'
== Starting CAPI[contr1/2810111694]/4 at ,s,1 still failed so falling
back to context 'default'
-- Executing Dial(CAPI[contr1/2810111694]/4, SIP/111694|60|tr) in
new stack
-- Called 111694
--
Is this a bug? It does not handle the context, so, it can not find what
to do, it works only with the default context.
Thank you in advance,
Dimitris
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[Asterisk-Users] Possible Bug: Crash when Parking Calls
Hi, I'm relativle new to *, so I may be wrong. I build up * from cvs today (show version: CVS-01/15/04-16:27:36). In an test I use 2 SIP phones (linphone) to connect to eachother. The phones are called via the extensions 100 (user 'kwe') and 200 (user 'phone'). I can call from one to another and I can park a call and take it again: 1) call from 100 to 200 2) press # and dial 700 on phone '200' 3) hear the voice and get the call back (dial 701 on phone '200') the steps can be repeated. But * crahes if I do the following during the same call: 4) pree # and dial 700 on phone '100' I can reproduce this. Every time I park a call first on one end and lateron on the other end, * crahes. On console I see the following: ---snip--- -- Called phone -- SIP/phone-c9da is ringing -- SIP/phone-c9da answered SIP/kwe-9857 -- Attempting native bridge of SIP/kwe-9857 and SIP/phone-c9da -- Started music on hold, class 'default', on SIP/kwe-9857 -- Playing 'pbx-transfer' (language 'de') -- Stopped music on hold on SIP/kwe-9857 -- Started music on hold, class 'default', on SIP/kwe-9857 == Parked SIP/kwe-9857 on 701 -- Playing 'digits/7' (language 'de') -- Playing 'digits/0' (language 'de') -- Playing 'digits/1' (language 'de') == Spawn extension (default, 200, 1) exited KEEPALIVE on 'SIP/kwe-9857' -- Executing ParkedCall(SIP/phone-b583, 701) in new stack -- Stopped music on hold on SIP/kwe-9857 -- Channel SIP/phone-b583 connected to parked call 701 -- Attempting native bridge of SIP/kwe-9857 and SIP/phone-b583 -- Started music on hold, class 'default', on SIP/phone-b583 -- Playing 'pbx-transfer' (language 'de') -- Stopped music on hold on SIP/phone-b583 -- Started music on hold, class 'default', on SIP/phone-b583 == Parked SIP/phone-b583 on 701 -- Playing 'digits/7' (language 'de') -- Playing 'digits/0' (language 'de') -- Playing 'digits/1' (language 'de') == Spawn extension (default, 701, 1) exited non-zero on 'SIP/phone-b583' -- Stopped music on hold on SIP/phone-b583 pbx*CLI Disconnected from Asterisk server Executing last minute cleanups ---snip--- I would be glad, if someone could give me some information: a) Yes I can reproduce this b) Yes, it's a known bug c) No, I can not reproduce this In case of c): could You please give me some advise, how to locate the error? Thanks in advance Karsten ___ 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
