Re: [asterisk-users] How to give users the capability to set CDR userfield for some calls
2013/1/17 Kevin Larsen kevin.lar...@pioneerballoon.com
Possibly switch to using subroutines instead of Macros. Macros are being
deprecated in place of subroutines.
Interesting thing to try.
The trouble is I can't find any usable example of calling Gosub routines
from features.conf's application map.
I've found old references explaining that this is not supported but I don't
if it's still valid or not.
Any ex
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From:Olivier oza_4...@yahoo.fr
To:Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date:01/17/2013 10:29 AM
Subject:[asterisk-users] How to give users the capability to set
CDRuserfield for some calls
Sent by:asterisk-users-boun...@lists.digium.com
--
Hello,
To my surprise, with asterisk 1.8 (I've not tried with other versions), it
seems you cannot set CDR's userfield from within a dialplan macro called by
dynamic features.
See :
testfeature = *321,self/callee,Macro,toto
[macro-toto]
exten = s,1,Verbose(0,Into macro-toto with CDR(src) set to ${CDR(src)})
exten = s,n,Set(CDR(userfield)=foobar)
I'm planning to use this feature to let users mark in CDR an ongoing call
as malicious or important or whatever.
Any hint ?
Regards--
_
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Re: [asterisk-users] Asterisk 11- Answer with [m=image 0 udptl t38] and Call Drop
No luck so far, should I consider it a bug in Asterisk 11 as I have tried different version of Asterisk 11 as well. Carrier sends BYE with service not implemented where as asterisk advertise udptl in SDP for Answer. I do not want it to be advertised by Asterisk 11 in Answer() as I am not using it(udptl, fax etc) in any case. On Thu, Jan 17, 2013 at 11:15 AM, Salman Zafar msalman...@gmail.com wrote: Thanks Jordan, for having a look at this matter. Yes, that is what Asterisk 11 is sending. Here are complete sip debugs from Asterisk attached. Please refer to IP mapping from OP to have a better understanding. Is there any way of getting it off from SIP parser on compile time as I am not using this feature and do not intend to use in future. On Wed, Jan 16, 2013 at 7:01 PM, Matthew Jordan mjor...@digium.comwrote: On 01/16/2013 07:28 AM, Salman Zafar wrote: Hello All, I am having a bit peculiar problem with Asterisk 11 for a carrier. This carrier shares quite some information in SDP header, which should not be the problem, however what happen is as follow: Carrier (INVITE) - *SIP Proxy - Asterisk 11 - Answer()* - right after answering call drops... Carrier send a BYE with (cause 79: service or option not implemented). *NOTE: Please refer to complete SIP traces attached. * * * *Also Note:* _Carrier_: 62.61.147.214 _Proxy_: 77.X.X.X:5060 _Asterisk11_: 77.X.X.X:5080 *_Here is Invite SDP from Carrier - Proxy - Asterisk 11_* INVITE sip:69609000@77.X.X.X SIP/2.0 v=0 o=AudiocodesGW 1638819008 1638818710 IN IP4 62.61.147.214 s=Phone-Call c=IN IP4 77.X.X.X t=0 0 m=audio 53372 RTP/AVP 8 118 18 a=rtpmap:8 PCMA/8000 a=rtpmap:118 PCMA/8000 a=gpmd:118 vbd=yes a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv a=rtcp:53373 IN IP4 77.X.X.X m=image 56854 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxMaxBuffer:1024 a=T38FaxMaxDatagram:122 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy /*_SDP:After Answered by Asterisk 11_*/ v=0 o=root 164966782 164966782 IN IP4 77.X.X.X s=Asterisk v11.0.1 c=IN IP4 77.X.X.X t=0 0 m=audio 12636 RTP/AVP 18 8 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv *_m=image 0 udptl t38_* The appropriate way for Asterisk to indicate that it does not support a media stream is to set the port number to 0. We have to inform the offerer that we don't support the media stream; removing it from the SDP completely is not allowed. Per RFC 3264, section 6: An offered stream MAY be rejected in the answer, for any reason. If a stream is rejected, the offerer and answerer MUST NOT generate media (or RTCP packets) for that stream. To reject an offered stream, the port number in the corresponding stream in the answer MUST be set to zero. I have tired by disabling/unloading fax modules as *I am not using* them but no results. Secondly, also tried tweaking of udptl ever-odd nothing worked. You've configured your system to not support fax correctly. Asterisk is rejecting the offered image stream accordingly. The same carrier works for Asterisk 1.6.X and the only difference I have notice so far is the above underlined line in Answered SDP - m=image 0 udptl t38. I think if I some how do not advertise udptl here i would be able to avoid this scenario. I have tried multiple ways to strip off SDP from incoming INVITE at SIP Proxy level but it is not SDP wise enough. I'm not sure what 1.6.x is sending. It's possible that it just completely removed the stream from the SDP answer, which is wrong. Section 6 again: For each m= line in the offer, there MUST be a corresponding m= line in the answer. *Note:* In Asterisk 1.6 = WARNING[32671]: chan_sip.c:8833 process_sdp: Unsupported SDP media type in offer: image 59978 udptl t38 In Asterisk 11 = WARNING[18748][C-002f]: chan_sip.c:10277 process_sdp: Failed to initialize UDPTL, declining image stream An initial glance at this makes me think your carrier is doing something wrong. Just to check, however, is the SDP answer you pasted the entire SDP that Asterisk 11 responds with? Specifically, are there no format attributes for the image stream in the SDP that Asterisk responds with? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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 -- Regards
Re: [asterisk-users] Asterisk 11- Answer with [m=image 0 udptl t38] and Call Drop
On 01/18/2013 05:37 AM, Salman Zafar wrote: No luck so far, should I consider it a bug in Asterisk 11 as I have tried different version of Asterisk 11 as well. Carrier sends BYE with service not implemented where as asterisk advertise udptl in SDP for Answer. I do not want it to be advertised by Asterisk 11 in Answer() as I am not using it(udptl, fax etc) in any case. It is not a bug in Asterisk. Asterisk is following what the RFC says it MUST do. It is, however, a bug with your carrier. I would do the following: 1) Contact your carrier and ask why they are rejecting the 200 OK. 2) Assuming they won't change their behaviour, find out what they want in a response that declines an image media format. Without knowing what your carrier thinks the SDP should look like, any modifications you make to Asterisk will be guesses. 3) When you find out what they want, modify chan_sip so that it answers back with whatever they told you they want. This will occur in chan_sip's add_sdp method - in particular, look for the portion where add_sdp adds *either* the m_modem/a_modem strings to the SDP *or* the pre-formatted decline_m_line. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
Re: [asterisk-users] Asterisk 11- Answer with [m=image 0 udptl t38] and Call Drop
Thanks Matt, that is exactly what I was looking for. On Fri, Jan 18, 2013 at 5:24 PM, Matthew Jordan mjor...@digium.com wrote: On 01/18/2013 05:37 AM, Salman Zafar wrote: No luck so far, should I consider it a bug in Asterisk 11 as I have tried different version of Asterisk 11 as well. Carrier sends BYE with service not implemented where as asterisk advertise udptl in SDP for Answer. I do not want it to be advertised by Asterisk 11 in Answer() as I am not using it(udptl, fax etc) in any case. It is not a bug in Asterisk. Asterisk is following what the RFC says it MUST do. It is, however, a bug with your carrier. I would do the following: 1) Contact your carrier and ask why they are rejecting the 200 OK. 2) Assuming they won't change their behaviour, find out what they want in a response that declines an image media format. Without knowing what your carrier thinks the SDP should look like, any modifications you make to Asterisk will be guesses. 3) When you find out what they want, modify chan_sip so that it answers back with whatever they told you they want. This will occur in chan_sip's add_sdp method - in particular, look for the portion where add_sdp adds *either* the m_modem/a_modem strings to the SDP *or* the pre-formatted decline_m_line. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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 -- Regards ** Muhammad Salman Zafar *** -- _ -- 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] Only silence trying to play streaming MOH
I am having trouble getting streaming MOH to work. As far as I can tell I have everything configured properly but there is only silence. Your help is appreciated. I am running Asterisk 1.8.11-cert10 with mpg123 1.12.1 to play the stream (I have tried madplay, and mpg321, and I compiled streamplayer as well with the same results). I started by finding a working stream and tested this from the shell (and Winamp just to be sure): /usr/bin/mpg123 -q -r 8000 -f 8192 --mono -s http://208.77.21.15:11510 It begins dumping the stream to the screen so I feel pretty confident this is working. In musiconhold.conf I have: [default] mode=files directory=moh sort=random [test] mode=custom ; Note that with mode=custom, a directory is not required, such as when reading ; from a stream. ;directory=/var/lib/asterisk/mohmp3 application=/usr/bin/mpg123 -q -r 8000 -f 8192 --mono -s http://208.77.21.15:11510 CLI moh reload CLI moh show classes Class: default Mode: files Directory: moh Class: test Mode: custom Directory: nodir Application: /usr/bin/mpg123 -q -r 8000 -f 8192 --mono -s http://208.77.21.15:11510 Format: slin After the moh reload I see the mpg123 process running: ps aux | grep mpg myuser 10183 0.0 0.0 14184 1020 ?S07:07 0:00 /usr/bin mpg123 -q -r 8000 -f 8192 --mono -s http://208.77.21.15:11510 Then in extensions.conf I added: exten = 1234,1,NoOp() same = n,Answer() same = n,MusicOnHold(test) same = n,Hangup() CLI dialplan reload Then I dial: == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 -- Executing [1234@features:1] NoOp(SIP/mysip_4405-001f, ) in new stack -- Executing [1234@features:2] Answer(SIP/mysip-001f, ) in new stack -- Executing [1234@features:3] MusicOnHold(SIP/mysip-001f, test) in new stack -- Started music on hold, class 'test', on channel 'SIP/mysip-001f' I hear just dead air. I have tried different settings for buffering the stream, other stream sources, other players, defining a directory with a 0 byte file as some tutorials suggest, streaming to a file, etc but always with the same results. Dead silence. Thank you for your help. Chet Stevens -- _ -- 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] rtptimeout: how to detect it in dialplan?
Hi! I want to forward a call to another destination if the outgoing call leg has an rtptimeout. But as far as I see there is no way to find out if the hangup was due to a rtp timeout or any other reason. I thought that HANGUPCAUSE or DIALSTATUS would be set, but they aren't. Are there any means to detect an rtp timeout in extensions.conf? Thanks Klaus -- _ -- 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] How to give users the capability to set CDR userfield for some calls
Since Gosub is technically an application, you should be able to modify this
snippet in features.conf
testfeature = #9,peer,Playback,tt-monkeys ;Allow both the caller and
callee to play
;;tt-monkeys to the opposite
channel
To this
testfeature = #9,peer,Gosub,play-monkeys,s,1 ;Allow both the caller and
callee to play
;;tt-monkeys to the opposite
channel
And in extensions.conf add
[play-monkeys]
Exten = s,1,playback(tt-monkeys)
Exten = s,n,return()
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Friday, January 18, 2013 3:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to give users the capability to set CDR
userfield for some calls
2013/1/17 Kevin Larsen kevin.lar...@pioneerballoon.com
Possibly switch to using subroutines instead of Macros. Macros are being
deprecated in place of subroutines.
Interesting thing to try.
The trouble is I can't find any usable example of calling Gosub routines
from features.conf's application map.
I've found old references explaining that this is not supported but I don't
if it's still valid or not.
Any ex
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From:Olivier oza_4...@yahoo.fr
To:Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date:01/17/2013 10:29 AM
Subject:[asterisk-users] How to give users the capability to set CDR
userfield for some calls
Sent by:asterisk-users-boun...@lists.digium.com
_
Hello,
To my surprise, with asterisk 1.8 (I've not tried with other versions), it
seems you cannot set CDR's userfield from within a dialplan macro called by
dynamic features.
See :
testfeature = *321,self/callee,Macro,toto
[macro-toto]
exten = s,1,Verbose(0,Into macro-toto with CDR(src) set to ${CDR(src)})
exten = s,n,Set(CDR(userfield)=foobar)
I'm planning to use this feature to let users mark in CDR an ongoing call as
malicious or important or whatever.
Any hint ?
Regards--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com/
http://www.api-digital.com --
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http://www.asterisk.org/hello http://www.asterisk.org/hello
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http://lists.digium.com/mailman/listinfo/asterisk-users
--
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[asterisk-users] Annoying delay after main server goes down
Hello, we have distributed lots of cisco spa303 IP phones and get them work with Asterisk. I have configured proxy and alternate proxy and enabled dual registration features in provisioning files(xml files). All phones are able to subscribe to both of servers. But the problem is, if main server goes down, i am obliged to wait nearly 20 second in order to place a call over second server. How to get alternate proxy work immediately after first server fails ? Thanks in advance. -- _ -- 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] Annoying delay after main server goes down
I think this is a phone problem not an asterisk one. In my experience a SIP (IP) phone takes about 20 seconds to properly negotiate (re)registration (longer for Polycom 501s). The best work-around I could recommend would be to have an intermediate interface like kamailio (sp) that handles the dual registration. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Onur Cem Çelebi Sent: Friday, January 18, 2013 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Annoying delay after main server goes down Hello, we have distributed lots of cisco spa303 IP phones and get them work with Asterisk. I have configured proxy and alternate proxy and enabled dual registration features in provisioning files(xml files). All phones are able to subscribe to both of servers. But the problem is, if main server goes down, i am obliged to wait nearly 20 second in order to place a call over second server. How to get alternate proxy work immediately after first server fails ? Thanks in advance. -- _ -- 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] Annoying delay after main server goes down
On Fri, 2013-01-18 at 18:06 +0200, Onur Cem Çelebi wrote: Hello, we have distributed lots of cisco spa303 IP phones and get them work with Asterisk. I have configured proxy and alternate proxy and enabled dual registration features in provisioning files(xml files). All phones are able to subscribe to both of servers. But the problem is, if main server goes down, i am obliged to wait nearly 20 second in order to place a call over second server. How to get alternate proxy work immediately after first server fails ? Thanks in advance. I think the failover time will be related to the the registration expiry setting in the phones. -- 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] Annoying delay after main server goes down
I would suggest to use linux ha and use same ip, which can failover to second standby server using heartbeat. This activity takes less then 5secs. Mitul On Jan 18, 2013 9:42 PM, Ishfaq Malik i...@pack-net.co.uk wrote: On Fri, 2013-01-18 at 18:06 +0200, Onur Cem Çelebi wrote: Hello, we have distributed lots of cisco spa303 IP phones and get them work with Asterisk. I have configured proxy and alternate proxy and enabled dual registration features in provisioning files(xml files). All phones are able to subscribe to both of servers. But the problem is, if main server goes down, i am obliged to wait nearly 20 second in order to place a call over second server. How to get alternate proxy work immediately after first server fails ? Thanks in advance. I think the failover time will be related to the the registration expiry setting in the phones. -- 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 -- _ -- 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] Voicemail and recordings storage: best practices
Greetings list, I'm currently building a new cluster to replace our ageing Asterisk 1.4 infrastructure - it's easier to start from scratch then migrate users across than it is to upgrade 1.4 to 1.8 in situ. Anyway, it got me thinking about audio recordings in a multi-server environment and whether there was a better way to do it. On our existing 1.4 cluster we NFS mount voicemail and recordings directories from another server (or more accurately a master/backup pair of servers) into each asterisk server. I'd say it's worked 'okay' - but since less than 10% of our users regularly use call recording, it's never really reached a point where I/O throughput has been an issue. So, since I have the opportunity to build up the new cluster from scrach, I thought it was an ideal opportunity to do a quick straw poll of the list and see what approaches others are using to store voicemail and recordings, and to make those available across a multi-server environment. Let the discussions begin. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Any timeframe for the release of the Asterisk 11-Lumenvox connector bridge?
I'm starting to think about migrating from an old Asterisk box to a new one and want to use the Asterisk 11 long term support release, but need Lumenvox integration and I don't see the Asterisk 11 connector bridge for Lumenvox available yet. Lumenvox tech support says this is under Digiums control. Can anyone give an idea of how soon it'll be available? Steve -- _ -- 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] 'Slower but cleaner' G711 option
When you compile asterisk from source there's an option to enable an alternate G711 algorithm which is stated somewhat cryptically to be slower, but cleaner. Does anybody have the authoritative answer as to what the deal is with this? I saw a forum post from somebody who said something about it handling faxes better, and only being marginally slower. If it produces better audio, and isn't much slower why isn't it the default option? -- _ -- 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] rtptimeout: how to detect it in dialplan?
As I read it you can do it like this: From http://www.voip-info.org/wiki/view/Asterisk+sip+rtptimeout Exten = s,1,noop (set rtptimeout so we can have 2 timeouts on a dial) Exten = s,n,Set(rtptimeout=60) Exten = s,n,Dial(SIP/peer1,60) Exten = s,n,Dial(SIP/peer2,60) Haven't tested this. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus Darilion Sent: Friday, January 18, 2013 9:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] rtptimeout: how to detect it in dialplan? Hi! I want to forward a call to another destination if the outgoing call leg has an rtptimeout. But as far as I see there is no way to find out if the hangup was due to a rtp timeout or any other reason. I thought that HANGUPCAUSE or DIALSTATUS would be set, but they aren't. Are there any means to detect an rtp timeout in extensions.conf? Thanks Klaus -- _ -- 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] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start
I cloned this ticket for 11.2 https://issues.asterisk.org/jira/browse/ASTERISK-20962 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Wednesday, January 16, 2013 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start On Wed, Jan 16, 2013 at 1:37 PM, Danny Nicholas da...@debsinc.com wrote: Same issue exists with 11.2 I've created issue 20945 to track this, at least for 1.8.20.0. https://issues.asterisk.org/jira/browse/ASTERISK-20945 -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- 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] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start
On 01/18/2013 03:42 PM, Danny Nicholas wrote: I cloned this ticket for 11.2 https://issues.asterisk.org/jira/browse/ASTERISK-20962 There's actually no need to clone the issue for multiple versions. Bugs are always fixed in the oldest supported release branch (1.8) and merged up stream. Asterisk 11 gets every fix for every bug that originates in the 1.8 branch. See the bug fix section of this page for more information on how the merge process works: https://wiki.asterisk.org/wiki/display/AST/Software+Configuration+Management+Policies Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
Re: [asterisk-users] rtptimeout: how to detect it in dialplan?
On 18 Jan 2013 15:22, Klaus Darilion klaus.mailingli...@pernau.at wrote: Hi! I want to forward a call to another destination if the outgoing call leg has an rtptimeout. But as far as I see there is no way to find out if the hangup was due to a rtp timeout or any other reason. I thought that HANGUPCAUSE or DIALSTATUS would be set, but they aren't. Are there any means to detect an rtp timeout in extensions.conf? Thanks Klaus -- _ -- 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] Any timeframe for the release of the Asterisk 11-Lumenvox connector bridge?
I'm starting to think about migrating from an old Asterisk box to a new one and want to use the Asterisk 11 long term support release, but need Lumenvox integration and I don't see the Asterisk 11 connector bridge for Lumenvox available yet. Lumenvox tech support says this is under Digiums control. Can anyone give an idea of how soon it'll be available? I will need this as well. -- _ -- 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] recrding calls
I would like to outgoing/icoming calls and email the files.
This is what I have:
...
exten = _7.,n,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = _7.,n,Monitor(wav,${CALLFILENAME},m)
...
How do I email these file?
--
Joseph
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Re: [asterisk-users] recrding calls
On 19/1/13 1:25 am, Joseph wrote:
I would like to outgoing/icoming calls and email the files.
This is what I have:
exten = _7.,n,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = _7.,n,Monitor(wav,${CALLFILENAME},m)
How do I email these file?
You probably want to use MixMonitor() instead of Monitor():
http://www.voip-info.org/wiki/view/Asterisk+cmd+Mixmonitor
One of its options allows you to execute a command at the end of
recording, which you can then use to call a script to handle your
recordings however you wish.
Kind regards,
Chris
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Re: [asterisk-users] recrding calls
On 01/19/13 01:34, Chris Bagnall wrote:
On 19/1/13 1:25 am, Joseph wrote:
I would like to outgoing/icoming calls and email the files.
This is what I have:
exten = _7.,n,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = _7.,n,Monitor(wav,${CALLFILENAME},m)
How do I email these file?
You probably want to use MixMonitor() instead of Monitor():
http://www.voip-info.org/wiki/view/Asterisk+cmd+Mixmonitor
One of its options allows you to execute a command at the end of
recording, which you can then use to call a script to handle your
recordings however you wish.
Kind regards,
Chris
I see, so MixMonitor mixes the IN/OUT sound as one file; that is good.
I've modified it but the call is recorded as raw extension and no TIMESTAMP.
exten = 11,n,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = 11,n,MixMonitor(${CALLFILENAME},wav,b)
--
Joseph
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Re: [asterisk-users] recrding calls
On Fri, 18 Jan 2013, Joseph wrote:
I've modified it but the call is recorded as raw extension and no
TIMESTAMP.
exten = 11,n,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = 11,n,MixMonitor(${CALLFILENAME},wav,b)
Please re-read the description for mixmonitor(). The arguments are not the
same as monitor().
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Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] recrding calls
On 01/18/13 18:09, Steve Edwards wrote:
On Fri, 18 Jan 2013, Joseph wrote:
I've modified it but the call is recorded as raw extension and no
TIMESTAMP.
exten = 11,n,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = 11,n,MixMonitor(${CALLFILENAME},wav,b)
Please re-read the description for mixmonitor(). The arguments are not the
same as monitor().
Got it, it should be:
exten = 11,n,Set(CALLFILENAME=${EXTEN}-${TIMESTAMP})
exten = 11,n,MixMonitor(${CALLFILENAME}.wav,b)
However, my argument ${TIMESTAMP} is not taking any effect :-/
--
Joseph
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Re: [asterisk-users] recrding calls
On Fri, Jan 18, 2013 at 6:25 PM, Joseph syscon...@gmail.com wrote:
I would like to outgoing/icoming calls and email the files.
This is what I have:
...
exten = _7.,n,Set(CALLFILENAME=${**EXTEN:1}-${TIMESTAMP})
exten = _7.,n,Monitor(wav,${**CALLFILENAME},m)
...
How do I email these file?
This is how we do it:
exten =
_1NXXNXX,1,Set(recordfilename=/var/spool/asterisk/monitor/${EXTEN}-${TIMESTAMP:0:8}${TIMESTAMP:8}.WAV)
\exten = _1NXXNXX,n,MixMonitor(${recordfilename},b)
exten = _1NXXNXX,n,(dial here or whatever)
exten = h,1,System(/usr/sbin/sendEmail -t u...@domain.com -f
p...@domain.com-u Call recording for ${recordingfilename} -m There
is a new call
recording. -a ${recordfilename})
Google the sendEmail app and download it, very useful for a lot of things.
--
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] recrding calls
On 01/18/13 19:27, Carlos Alvarez wrote:
On Fri, Jan 18, 2013 at 6:25 PM, Joseph [1]syscon...@gmail.com wrote:
I would like to outgoing/icoming calls and email the files.
This is what I have:
...
exten = _7.,n,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = _7.,n,Monitor(wav,${CALLFILENAME},m)
...
How do I email these file?
This is how we do it:
exten =
_1NXXNXX,1,Set(recordfilename=/var/spool/asterisk/monitor/${EXTEN}-
${TIMESTAMP:0:8}${TIMESTAMP:8}.WAV)
\exten = _1NXXNXX,n,MixMonitor(${recordfilename},b)
exten = _1NXXNXX,n,(dial here or whatever)
Thanks Carlos
I'm just concentrating right now on ${TIMESTAMP} variable but is is not working:
I have:
exten =
11,n,Set(recordfilename=/var/spool/asterisk/monitor/${EXTEN}-${TIMESTAMP:0:8}${TIMESTAMP:8}.WAV)
exten = 11,n,MixMonitor(${recordfilename},b)
and the file name I got was:
-11.wav
Why I'm not getting any timestamp?
--
Joseph
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Re: [asterisk-users] recrding calls
Around version 1.4 or 1.6, TIMESTAMP was phased out and replaced with
STRFTIME. See this page for details on how to properly generate a
timestamp:
http://www.voip-info.org/wiki/view/Asterisk+func+strftime
On Fri, Jan 18, 2013 at 8:46 PM, Joseph syscon...@gmail.com wrote:
On 01/18/13 19:27, Carlos Alvarez wrote:
On Fri, Jan 18, 2013 at 6:25 PM, Joseph [1]syscon...@gmail.com wrote:
I would like to outgoing/icoming calls and email the files.
This is what I have:
...
exten = _7.,n,Set(CALLFILENAME=${**EXTEN:1}-${TIMESTAMP})
exten = _7.,n,Monitor(wav,${**CALLFILENAME},m)
...
How do I email these file?
This is how we do it:
exten =
_1NXXNXX,1,Set(**recordfilename=/var/spool/**
asterisk/monitor/${EXTEN}-
${TIMESTAMP:0:8}${TIMESTAMP:8}**.WAV)
\exten = _1NXXNXX,n,MixMonitor(${**recordfilename},b)
exten = _1NXXNXX,n,(dial here or whatever)
Thanks Carlos
I'm just concentrating right now on ${TIMESTAMP} variable but is is not
working:
I have:
exten = 11,n,Set(recordfilename=/var/**spool/asterisk/monitor/${**
EXTEN}-${TIMESTAMP:0:8}${**TIMESTAMP:8}.WAV)
exten = 11,n,MixMonitor(${**recordfilename},b)
and the file name I got was: -11.wav
Why I'm not getting any timestamp?
--
Joseph
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Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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Re: [asterisk-users] recrding calls
Yeah, sorry, that example was from one of our older servers.
On Fri, Jan 18, 2013 at 8:08 PM, Warren Selby wcse...@selbytech.com wrote:
Around version 1.4 or 1.6, TIMESTAMP was phased out and replaced with
STRFTIME. See this page for details on how to properly generate a
timestamp:
http://www.voip-info.org/wiki/view/Asterisk+func+strftime
On Fri, Jan 18, 2013 at 8:46 PM, Joseph syscon...@gmail.com wrote:
On 01/18/13 19:27, Carlos Alvarez wrote:
On Fri, Jan 18, 2013 at 6:25 PM, Joseph [1]syscon...@gmail.com
wrote:
I would like to outgoing/icoming calls and email the files.
This is what I have:
...
exten = _7.,n,Set(CALLFILENAME=${**EXTEN:1}-${TIMESTAMP})
exten = _7.,n,Monitor(wav,${**CALLFILENAME},m)
...
How do I email these file?
This is how we do it:
exten =
_1NXXNXX,1,Set(**recordfilename=/var/spool/**
asterisk/monitor/${EXTEN}-
${TIMESTAMP:0:8}${TIMESTAMP:8}**.WAV)
\exten = _1NXXNXX,n,MixMonitor(${**recordfilename},b)
exten = _1NXXNXX,n,(dial here or whatever)
Thanks Carlos
I'm just concentrating right now on ${TIMESTAMP} variable but is is not
working:
I have:
exten = 11,n,Set(recordfilename=/var/**spool/asterisk/monitor/${**
EXTEN}-${TIMESTAMP:0:8}${**TIMESTAMP:8}.WAV)
exten = 11,n,MixMonitor(${**recordfilename},b)
and the file name I got was: -11.wav
Why I'm not getting any timestamp?
--
Joseph
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Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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TelEvolve
602-889-3003
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Re: [asterisk-users] recrding calls
On 01/18/13 21:08, Warren Selby wrote:
Around version 1.4 or 1.6, TIMESTAMP was phased out and replaced with
STRFTIME. See this page for details on how to properly generate a
timestamp:
[1]http://www.voip-info.org/wiki/view/Asterisk+func+strftime
Thank Warren for the pointer, yes it is working now.
Correct sequence should be:
exten =
11,n,Set(recordfilename=${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
exten = 11,n,MixMonitor(${recordfilename},b)
--
Joseph
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