Re: [asterisk-users] BLINDTRANSFER and SIP hardphones
2009/4/24 Kevin P. Fleming kpflem...@digium.com Olivier wrote: When a SIP hardphone is transfering a call while ringing (caller and callee don't speak to each other) using phone's Transfer key, it seems BLINDTRANSFER remains empty. Though I can see a 302 MOVED TEMPORARILY message coming in. If the person performing the transfer has dialed the transferee's number and hears the call ringing, that is not a blind transfer, it is an attended to transfer to a call that hasn't been answered yet. There won't be any variables set for blind transfer, as it isn't one. Here is an extract from SIP debug (7530 is transferring incoming call from 7533 to 7531) : osiris2*CLI --- Transmitting (no NAT) to 192.168.100.122:5060 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.100.122:5060 ;branch=z9hG4bK4697915359658203609-1269236;received=192.168.100.122 From: Alainsip:7...@192.168.100.254:5060;user=phone;tag=c0a80101-135de8 To: sip:7...@192.168.100.254:5060;user=phone;tag=as37f823b2 Call-ID: 364221-c0a80101-...@192.168.100.122 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.0-rc4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:7...@192.168.100.254 sip%3a7...@192.168.100.254 Content-Length: 0 - osiris2*CLI --- SIP read from UDP://192.168.100.123:5060 --- SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK47ed73d6;rport From: Alainsip:7...@192.168.100.254 sip%3a7...@192.168.100.254 ;tag=as2d189259 To: sip:7...@192.168.100.123:5060;user=phone;tag=c0a80101-135999 Call-ID: 1ba5b4c707b15fec0909665f6e9ea...@192.168.100.254 CSeq: 102 INVITE Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: sip:7...@192.168.100.254:5060;user=phone Content-Length: 0 - --- (9 headers 0 lines) --- Transmitting (no NAT) to 192.168.100.123:5060: ACK sip:7...@192.168.100.123:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK47ed73d6;rport Max-Forwards: 70 From: Alain sip:7...@192.168.100.254 sip%3a7...@192.168.100.254 ;tag=as2d189259 To: sip:7...@192.168.100.123:5060;user=phone;tag=c0a80101-135999 Contact: sip:7...@192.168.100.254 sip%3a7...@192.168.100.254 Call-ID: 1ba5b4c707b15fec0909665f6e9ea...@192.168.100.254 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.0-rc4 Content-Length: 0 --- Really destroying SIP dialog ' 1ba5b4c707b15fec0909665f6e9ea...@192.168.100.254' Method: INVITE Audio is at 192.168.100.254 port 13840 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.100.88:29462: INVITE sip:7...@192.168.100.88:29462;rinstance=160ae873c74c4480 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK1f07ec53;rport Max-Forwards: 70 From: Alain sip:7...@192.168.100.254 sip%3a7...@192.168.100.254 ;tag=as0104afde To: sip:7...@192.168.100.88:29462;rinstance=160ae873c74c4480 Contact: sip:7...@192.168.100.254 sip%3a7...@192.168.100.254 Call-ID: 54b4a9fe0fbf10a51da9c2f301061...@192.168.100.254 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.0-rc4 Date: Fri, 24 Apr 2009 05:43:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 271 v=0 o=root 525634823 525634823 IN IP4 192.168.100.254 s=Asterisk PBX 1.6.1.0-rc4 c=IN IP4 192.168.100.254 t=0 0 m=audio 13840 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- osiris2*CLI --- SIP read from UDP://192.168.100.88:29462 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK1f07ec53;rport=5060 To: sip:7...@192.168.100.88:29462;rinstance=160ae873c74c4480 From: Alain sip:7...@192.168.100.254 sip%3a7...@192.168.100.254 ;tag=as0104afde Call-ID: 54b4a9fe0fbf10a51da9c2f301061...@192.168.100.254 CSeq: 102 INVITE Content-Length: 0 So when receiving 302 Moved Temporarily, Asterisk (version 1.6.1-rc4) is issuing a new INVITE and doesn't set any BLINDTRANSFER variable. Thinking back about that, I would say it should have done so. Your opinion ? Would you classify that as an attended transfer ? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
Hi, I am using my own number and not hanging up and audio is also coming please suggest our what might be the problem. Any help is highly appreciated. Thanks. On Thu, Apr 23, 2009 at 9:14 PM, Ruddy Gbaguidi plugwo...@micnes.comwrote: Maybe the customer hangs up during the AMD analysis or you don’t have any audio coming to asterisk through your sip channel. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Sam Hawkin *Sent:* April-23-09 11:00 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] AMD Not Working Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD(SIP/sip-ffe0, ) in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD using the default parameters. -- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] -- AMD: HANGUP any help is highly appreciated. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
Hi, Thanks for your reply I am using my own number and not hanging up. and sip debug is also not showing much information regarding the failure. please suggest our what might be the problem. Any help is highly appreciated. Thanks. On Fri, Apr 24, 2009 at 4:58 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Thu, Apr 23, 2009 at 6:12 PM, Matt Riddell li...@venturevoip.comwrote: On 24/04/2009 3:00 a.m., Sam Hawkin wrote: Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. I'd say that the remote end of the call is hanging up - do a SIP debug so you can see what happens - the best way to test things like this is by calling your own number - that way you can guarantee it doesn't hang up :) -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) You can also run Orecx on the localhost (for very small production or lab systems) or on a different host via mirrored switch port and then listen to all calls (SIP and other VoIP), or RTPTap via Sangoma cards). I have done this many times to catch intermittent problems that are continuously reported by users but cannot be readily reproduced. I just ask that the user log the time of the call and what they experienced, then I can listen to the recording, ascertain all the critical info that users leave off trouble reports, and figure out the commonalities. Obviously, all due notice/permission and/or legal disclosures should be made/given before recording anything. It is great for troubleshooting (and yes, calls do get crossed and all kinds of other strangness in Asterisk, you know, what you write off as user error :-) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
Hi,
Thanks for your reply
We are using the Asterisk 1.2.4.
and below the dialplan path. we are orginating the call to
my number and connection it to context cdtest and extension 1.
[cdtest]
exten = 1,1,NoOp( cb amd issue testing )
exten = 1,2,Macro(Cb-old|/root/business_hours|/root/business_hours)
[macro-Cb]
exten = s,1,NoOp( values in CB arg1 ${ARG1} arg2 ${ARG1} )
exten = s,2,AMD
exten = s,3,GotoIf($[${AMDSTATUS}=AMD_PERSON]?4:7)
exten = s,4,NoOp(Humanplaying--${ARG1})
exten = s,5,Playback(${ARG1})
exten = s,6,Hangup
exten = s,7,GotoIf($[${AMDSTATUS}=AMD_MACHINE]?8:11)
exten = s,8,NoOp(Machine---playing--${ARG2})
exten = s,9,Playback(${ARG2})
exten = s,10,Goto(s|12)
exten = s,11,Playback(${ARG1})
please suggest our what might be the problem.
Any help is highly appreciated.
Thanks.
On Thu, Apr 23, 2009 at 8:36 PM, Matt Florell astma...@gmail.com wrote:
On 4/23/09, Sam Hawkin gvrt...@gmail.com wrote:
Hi All,
I am trying to use the AMD (Answering Machine Detect).
But it is not sending the AMD_Status as either
the Human or Machine, it hangs up in middle.
can any one suggest us, what might be the problem
and possible solution to it.
below is the log
-- Executing AMD(SIP/sip-ffe0, ) in new stack
-- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4)
Apr 23 08:00:26 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD
using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence
[300] totalAnalysisTime [5000] minimumWordLength [120]
betweenWordsSilence
[50] maximumNumberOfWords [5] silenceThreshold [256]
-- AMD: HANGUP
What version of Asterisk are you running this on?
What is the dialplan path that this is running through?
MATT---
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Re: [asterisk-users] Should you use UserEvents for monitoring calls ?
2009/4/22 Olivier oza-4...@myamail.com Hi, I need to monitor call activity from a custom application software. The goal is to display things like who is on call or not, who has forwarded his call to his voicemail, etc ... When using manager's login command with Event parameter set to on, I'm getting tens of events I don't care about but I suppose I won't miss things like transfers, pickups, parking ... Would it be a right move to rely on UserEvents instead ? Then I would specifically have to add those UserEvents in dialplan but I'm afraid to be unable to support things like hangups or transfers, ... What's your opinion about that ? Would you filter system events or add custom uservents ? I tried the UserEvent way and up to now, I can monitor : - simple calls (terminated by caller or callee), - multiple calls (several phones are ringing but one is answering) - attended transfers What I can't monitor at the moment is : - transfers while ringing (callee transfers a call while still ringing) Not tried yet: - monitoring Voicemail, Parking, Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Feature request: manager show events
Hi, To further improve Asterisk documentation, would approve manager show events and manager show event foo commands to be added to CLI ? Today, it is possible to list available manager commands but not to list available events, AFAIK. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] function originate
Hi, Feature originate can be used make call thro' the web. There is a parameter ,Async, in it. I set it to true but there is no effect. Actually, I want to do the following. What I know the function originate is: originate call --- party A party A rings party A answers call party B rings, party A still hear ring party B answers and A B connected. party A will feel weird when she will still hear ring after answering a call until party B answers it. Below is what I want to do: originate call --- party A party A rings party B rings party A answers call A B connected. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday Apr 24 @ 12 Noon: Wideband, or HD Voice as Polycom calls it
Hi, This week (today in fact) Michael Graves talks to Dan Berninger about the future of wideband VoIP and the upcoming conference. Some of you might remember the name from a previous conference about FWD. More about Dan: Daniel Berninger - Washington, DC based independent technolgy analyst. Expert in technical and regulatory aspects of Internet enabled disruptive communications. Active in VoIP since 1995. Daniel worked on the original assessment of VoIP at Bell Laboratories and led early gateway deployments at Verizon , HP , and NASA after joining VocalTec Communications . He won the 1999 VON Pioneer Award as co-founder of the VON Coalition and worked on the founding of ITXC , Vonage , and Free World Dialup . Daniel gets quoted frequently on regulatory, antitrust, and VoIP matters. This should be of interest to all of you as Dan has been an important force in the movement we are all a part of. See you there! http://www.voipUsersConference.org Wideband g722: call 200...@login.zipdx.com (thx to David) g711 ulaw: call 7463#2262...@proxy.ideasip.com (thx to Neil) IRC: #voip-users-conference on irc.freenode.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] timing source problem
hi all, we do have some troubles with zaptel timing source - we have a setup with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk does some handling - calls are leaving on digium card 1 - going to a siemens hipath - there is some call handling - some of the calls then are going from the hipath over a qsig line to a bosch integral PBX - handling the rest of the calls. To be able to get away from the bosch system - we like to put asterisk (1 port free on each card) in the middle of the path siemens - bosch - so that it will be siemens - card 0 asterisk card 1 - bosch. Currently the Siemens hipath is playing the network side - the bosch is cpe. So the siemens hipath does provide the timing source. With asterisk in the middle i can not take the timing source from the siemens link - because i have already the telco line as timing source. But when starting it in this setup - i will get lots of timing source auto card 0! messages. So i think the siemens timing is not in sync. with the telco timing - so mixed up on asterisk with telco line as primary timing will not work when the siemens does try to deliver timing. I have not tried as /etc/zaptel.conf parameter 0 als timing parameter (0 = be master) - but i think it wont work because the siemens wont accept the timing from the asterisk box. Changing configuration of the siemens is not possible. So - here the questions... - is it possible to do what i want to do ? - do you think timing=0 in zaptel.conf will work ? - would it be possible to connect a xorcom 2 PRI channel bank to asterisk to handle the qsig line between the two ? Or will the xorcom then also take the timing from the digum cards - telco lines ? any hints would be nice... many thanks Wolfgang ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Jabber and Presence
2009/4/23 Matt Riddell li...@venturevoip.com: On 18/04/2009 2:28 a.m., Gavin Henry wrote: Hi all, What other open source tools are people using for this? I was looking at Openfire and their asterisk plugin. Is it easy to roll your own with res_jabber.so ?? I used openfire in the past, but have now changed over to using ejabberd. We use PHP classes to send jabber messages from the support system, JabberSend to send messages from the dialplan, and a bot to send messages for live support. Thanks for that Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
Hello,
Well, depending on the version of app_amd that you used when you added
it to Asterisk 1.2, you might need to use HUMAN and MACHINE as the
possible AMDSTATUS instead of AMD_PERSON and AMD_MACHINE. The
AMDSTATUS was changed at some point in the app_amd code, not sure why
they changed it, but that might be your issue.
Also, since you are calling your own number you might want to do an
Answer on the call before running AMD, not sure if that would cause
the hangups you are seeing or not, but it's something to try.
MATT---
On 4/24/09, Sam Hawkin gvrt...@gmail.com wrote:
Hi,
Thanks for your reply
We are using the Asterisk 1.2.4.
and below the dialplan path. we are orginating the call to
my number and connection it to context cdtest and extension 1.
[cdtest]
exten = 1,1,NoOp( cb amd issue testing )
exten =
1,2,Macro(Cb-old|/root/business_hours|/root/business_hours)
[macro-Cb]
exten = s,1,NoOp( values in CB arg1 ${ARG1} arg2 ${ARG1} )
exten = s,2,AMD
exten = s,3,GotoIf($[${AMDSTATUS}=AMD_PERSON]?4:7)
exten =
s,4,NoOp(Humanplaying--${ARG1})
exten = s,5,Playback(${ARG1})
exten = s,6,Hangup
exten = s,7,GotoIf($[${AMDSTATUS}=AMD_MACHINE]?8:11)
exten =
s,8,NoOp(Machine---playing--${ARG2})
exten = s,9,Playback(${ARG2})
exten = s,10,Goto(s|12)
exten = s,11,Playback(${ARG1})
please suggest our what might be the problem.
Any help is highly appreciated.
Thanks.
On Thu, Apr 23, 2009 at 8:36 PM, Matt Florell astma...@gmail.com wrote:
On 4/23/09, Sam Hawkin gvrt...@gmail.com wrote:
Hi All,
I am trying to use the AMD (Answering Machine Detect).
But it is not sending the AMD_Status as either
the Human or Machine, it hangs up in middle.
can any one suggest us, what might be the problem
and possible solution to it.
below is the log
-- Executing AMD(SIP/sip-ffe0, ) in new stack
-- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4)
Apr 23 08:00:26 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD
using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence
[300] totalAnalysisTime [5000] minimumWordLength [120]
betweenWordsSilence
[50] maximumNumberOfWords [5] silenceThreshold [256]
-- AMD: HANGUP
What version of Asterisk are you running this on?
What is the dialplan path that this is running through?
MATT---
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Re: [asterisk-users] timing source problem
Hello, I would suggest that you first methodically try every possible combination of zaptel.conf timing settings(each change follwed by a hard reboot of the Asterisk server) to see if there is a magic combination of settings that will work. I don't know if you have the time for that, or if it takes a while for the timing issues to appear, but that is what I would try. If that still doesn't work, we have solved similar issues with older(2 years ago) Digium quad cards by switching to Sangoma hardware that offers more options for forcing timing in it's wanpipe driver software. Although when I posted about this before in another thread the folks from Digium swear that newer Digium cards(with newer firmware) do not have this problem using the newer Dahdi drivers. What version of Zaptel are you using and how old is your Digium card? MATT--- On 4/24/09, Wolfgang Pichler wpich...@yosd.at wrote: hi all, we do have some troubles with zaptel timing source - we have a setup with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk does some handling - calls are leaving on digium card 1 - going to a siemens hipath - there is some call handling - some of the calls then are going from the hipath over a qsig line to a bosch integral PBX - handling the rest of the calls. To be able to get away from the bosch system - we like to put asterisk (1 port free on each card) in the middle of the path siemens - bosch - so that it will be siemens - card 0 asterisk card 1 - bosch. Currently the Siemens hipath is playing the network side - the bosch is cpe. So the siemens hipath does provide the timing source. With asterisk in the middle i can not take the timing source from the siemens link - because i have already the telco line as timing source. But when starting it in this setup - i will get lots of timing source auto card 0! messages. So i think the siemens timing is not in sync. with the telco timing - so mixed up on asterisk with telco line as primary timing will not work when the siemens does try to deliver timing. I have not tried as /etc/zaptel.conf parameter 0 als timing parameter (0 = be master) - but i think it wont work because the siemens wont accept the timing from the asterisk box. Changing configuration of the siemens is not possible. So - here the questions... - is it possible to do what i want to do ? - do you think timing=0 in zaptel.conf will work ? - would it be possible to connect a xorcom 2 PRI channel bank to asterisk to handle the qsig line between the two ? Or will the xorcom then also take the timing from the digum cards - telco lines ? any hints would be nice... many thanks Wolfgang ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] function originate
You could use 2 originate commands and connect both of them to a meetme room? But surely what you're trying to do is going to confuse the person anyway if they don't hear anyone when they answer? Wouldn't it just be better to play a message after party a answers and then start ringing party b so that party a knows what's going on? 2009/4/24 Rilawich Ango maillist...@gmail.com Hi, Feature originate can be used make call thro' the web. There is a parameter ,Async, in it. I set it to true but there is no effect. Actually, I want to do the following. What I know the function originate is: originate call --- party A party A rings party A answers call party B rings, party A still hear ring party B answers and A B connected. party A will feel weird when she will still hear ring after answering a call until party B answers it. Below is what I want to do: originate call --- party A party A rings party B rings party A answers call A B connected. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] timing source problem
Hello, the issue does occour some seconds after connection the line - but the hard reboot takes some time... The cards are TE420 (4th Gen) - version c01a016a. Zaptel ist 1.4.12.1 The firmware on digium cards can not get flashed - or i am wrong (i have never heard about that) regards, Wolfgang Am Freitag, den 24.04.2009, 06:50 -0400 schrieb Matt Florell: Hello, I would suggest that you first methodically try every possible combination of zaptel.conf timing settings(each change follwed by a hard reboot of the Asterisk server) to see if there is a magic combination of settings that will work. I don't know if you have the time for that, or if it takes a while for the timing issues to appear, but that is what I would try. If that still doesn't work, we have solved similar issues with older(2 years ago) Digium quad cards by switching to Sangoma hardware that offers more options for forcing timing in it's wanpipe driver software. Although when I posted about this before in another thread the folks from Digium swear that newer Digium cards(with newer firmware) do not have this problem using the newer Dahdi drivers. What version of Zaptel are you using and how old is your Digium card? MATT--- On 4/24/09, Wolfgang Pichler wpich...@yosd.at wrote: hi all, we do have some troubles with zaptel timing source - we have a setup with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk does some handling - calls are leaving on digium card 1 - going to a siemens hipath - there is some call handling - some of the calls then are going from the hipath over a qsig line to a bosch integral PBX - handling the rest of the calls. To be able to get away from the bosch system - we like to put asterisk (1 port free on each card) in the middle of the path siemens - bosch - so that it will be siemens - card 0 asterisk card 1 - bosch. Currently the Siemens hipath is playing the network side - the bosch is cpe. So the siemens hipath does provide the timing source. With asterisk in the middle i can not take the timing source from the siemens link - because i have already the telco line as timing source. But when starting it in this setup - i will get lots of timing source auto card 0! messages. So i think the siemens timing is not in sync. with the telco timing - so mixed up on asterisk with telco line as primary timing will not work when the siemens does try to deliver timing. I have not tried as /etc/zaptel.conf parameter 0 als timing parameter (0 = be master) - but i think it wont work because the siemens wont accept the timing from the asterisk box. Changing configuration of the siemens is not possible. So - here the questions... - is it possible to do what i want to do ? - do you think timing=0 in zaptel.conf will work ? - would it be possible to connect a xorcom 2 PRI channel bank to asterisk to handle the qsig line between the two ? Or will the xorcom then also take the timing from the digum cards - telco lines ? any hints would be nice... many thanks Wolfgang ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] listen to prompt before bridging call.
Hi, Can someone please help to resolve the followinng issue: We would like an asterisk user to call a number and when the called party picks up the phone, we play a message (press 1 to accept call, 2 to reject call). Only when the called party presses 1, do we bridge the call and the two parties can communicate. What we would like though is that the person who makes the call be able to listen to the message press 1 to accept call, 2 to reject call) that is played to the called party BUT not be able to communicate with him untill he presses 1. Is this possible in asterisk using php/agi? Any pointers hightly appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FOP and UserEvent()
Hi all, I try to install FOP. It's very nice. In documentation I red that from my dial plan I can launch a popup window with UserEvent() application. I try to follow FOP documentation but I can't popup anything. My structure is: - server 1: Asterisk system - server 2: FOP system - client On client I connect to FOP panel, but I don't see any popup. Someone can help me to configure FOP popups and in the use of UserEvent() application? Thanks all Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feature request: manager show events
Hi, To further improve Asterisk documentation, would approve manager show events and manager show event foo commands to be added to CLI ? Today, it is possible to list available manager commands but not to list available events, AFAIK. Regards The problem is that currently, manager events are not registered, any module is free to launch events and there is no enforcement for the events to have a clearly defined structure. Work has been done lately in trying to make the naming of headers and order to be standard, but there is not programming interface enforcing that behaviour. The available events can be extracted using grep manager_event in the asterisk source code, but I agree it would be nice to see more structure there. Moy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Duplicating existing PBX function
Right now, we have a pbx that auto-answers for extension-to-extension calls, but after the phone has been auto answered, lets the caller press one to cause the phone to start ringing. (for example, the person's not in their office so you want it to ring through to voicemail) I'm able to duplicate the auto answer using the SIP add header function since I have Grandstream phones. I assuming what I need to do is setup a feature that executes a macro when the user presses one. This macro would hangup the callee and then redial the callee without sending the extra SIP header so the phone rings instead of auto-answers. Any suggestions on how to do this? If there's another way, I'm open to that as well. TIA! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feature request: manager show events
Then a suggestion for the next version would be to have a module which has the core set of events that are common to most everything for listing and added too, but still leave it open for the custom events most everyone uses for one thing or another. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error, please notify the sender immediately by reply to sender only message and destroy all electronic and hard copies of the communication, including attachments. Common sense is the collection of prejudices acquired by age eighteen. -- Albert Einstein Once you can accept the universe as matter expanding into nothing that is something,wearing stripes with plaid comes easy. -- Albert Einstein I know a little of everything, but a lot of nothing -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Moises Silva Sent: Friday, April 24, 2009 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Feature request: manager show events Hi, To further improve Asterisk documentation, would approve manager show events and manager show event foo commands to be added to CLI ? Today, it is possible to list available manager commands but not to list available events, AFAIK. Regards The problem is that currently, manager events are not registered, any module is free to launch events and there is no enforcement for the events to have a clearly defined structure. Work has been done lately in trying to make the naming of headers and order to be standard, but there is not programming interface enforcing that behaviour. The available events can be extracted using grep manager_event in the asterisk source code, but I agree it would be nice to see more structure there. Moy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Record in mp3
But have you tried to record directly in mp3, without to covert the file? --- Em qui, 23/4/09, Danny Nicholas da...@debsinc.com escreveu: De: Danny Nicholas da...@debsinc.com Assunto: Re: [asterisk-users] Record in mp3 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Data: Quinta-feira, 23 de Abril de 2009, 17:33 The way I read to do this is to use sox to create a wav file, then use lame to convert the wav to mp3. I did this for some MOH files. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose Enes Mateus Sent: Thursday, April 23, 2009 3:28 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Record in mp3 Somebody knows if I can save files in mp3 with the Record command on Asterisk? I try to recompile sox to suport mp3 but Asterisk return the folowing message when I use the Record command: - Executing [...@liberado15:15] Record(SIP/1201-083453c8, /var/spool/asterisk/alarme/alarme-1201-200905121212:mp3) in new stack -- SIP/1201-083453c8 Playing 'beep' (language 'pt_BR') [Apr 23 17:22:36] ERROR[4494]: format_mp3.c:283 mp3_rewrite: I Can't write MP3 only read them. [Apr 23 17:22:36] WARNING[4494]: file.c:378 fn_wrapper: Unable to rewrite format mp3 [Apr 23 17:22:36] WARNING[4494]: file.c:1092 ast_writefile: Unable to rewrite /var/spool/asterisk/alarme/alarme-1201-200905121212.mp3 [Apr 23 17:22:36] WARNING[4494]: app_record.c:272 record_exec: Could not create file /var/spool/asterisk/alarme/alarme-1201-200905121212 I'am doing something wrong? Thanks Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10 - Celebridades - Música - Esportes -Anexo incorporado- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Record in mp3
you probably don't want to record directly to mp3 as there will be an overhead in converting the audio on the fly and this will probably break your call recordings... you should either record in the codec you are using for phone calls (i think?) or in .wav and then convert afterwards (correct me if i'm wrong someone!). 2009/4/24 Jose Enes Mateus jemat...@yahoo.com.br But have you tried to record directly in mp3, without to covert the file? --- Em *qui, 23/4/09, Danny Nicholas da...@debsinc.com* escreveu: De: Danny Nicholas da...@debsinc.com Assunto: Re: [asterisk-users] Record in mp3 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Data: Quinta-feira, 23 de Abril de 2009, 17:33 The way I read to do this is to use sox to create a wav file, then use lame to convert the wav to mp3. I did this for some MOH files. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jose Enes Mateus *Sent:* Thursday, April 23, 2009 3:28 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Record in mp3 *Somebody knows* if I can save files in mp3 with the Record command on Asterisk? I try to recompile sox to suport mp3 but Asterisk return the folowing message when I use the Record command: - Executing [...@liberado15:15] Record(SIP/1201-083453c8, /var/spool/asterisk/alarme/alarme-1201-200905121212:mp3) in new stack -- SIP/1201-083453c8 Playing 'beep' (language 'pt_BR') [Apr 23 17:22:36] ERROR[4494]: format_mp3.c:283 mp3_rewrite: I Can't write MP3 only read them. [Apr 23 17:22:36] WARNING[4494]: file.c:378 fn_wrapper: Unable to rewrite format mp3 [Apr 23 17:22:36] WARNING[4494]: file.c:1092 ast_writefile: Unable to rewrite /var/spool/asterisk/alarme/alarme-1201-200905121212.mp3 [Apr 23 17:22:36] WARNING[4494]: app_record.c:272 record_exec: Could not create file /var/spool/asterisk/alarme/alarme-1201-200905121212 I'am doing something wrong? Thanks -- Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/- Celebridadeshttp://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/celebridades/- Músicahttp://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/m%C3%BAsica/- Esporteshttp://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/esportes/ -Anexo incorporado- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10http://br.rd.yahoo.com/mail/taglines/mail/*http://br.maisbuscados.yahoo.com/- Celebridadeshttp://br.rd.yahoo.com/mail/taglines/mail/*http://br.maisbuscados.yahoo.com/celebridades/- Músicahttp://br.rd.yahoo.com/mail/taglines/mail/*http://br.maisbuscados.yahoo.com/m%C3%BAsica/- Esporteshttp://br.rd.yahoo.com/mail/taglines/mail/*http://br.maisbuscados.yahoo.com/esportes/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feature request: manager show events
2009/4/24 James A. Shigley j...@answeringserv.com Then a suggestion for the next version would be to have a module which has the core set of events that are common to most everything for listing and added too, but still leave it open for the custom events most everyone uses for one thing or another. Which custom events are you referring to ? UserEvents ? Anyway, I think UserEvent should be listed when typing manager show events, though, of course, its description would be adapted to the fact its content is open So, if I'm not mistaken, at least 1+1+1=3 would welcome such a manager show events command. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2 Beta
Use: console dial Regards, -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian Sent: viernes, 24 de abril de 2009 01:07 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.6.2 Beta Hi all, I have not used Asterisk for some time, but decieed to have a go with it again. I noticed that some commands have been changed, where can one find a list of them except for the help command? I want to simulate a phone like I could do in previous versions of Asterisk so i can type dial and an extension from the Asterisk CLI. Many thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Se certificó que el correo entrante no contiene virus. Comprobada por AVG - www.avg.es Versión: 8.5.287 / Base de datos de virus: 270.12.3/2076 - Fecha de la versión: 04/24/09 07:54:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.2 Beta
Hi all, I have not used Asterisk for some time, but decieed to have a go with it again. I noticed that some commands have been changed, where can one find a list of them except for the help command? I want to simulate a phone like I could do in previous versions of Asterisk so i can type dial and an extension from the Asterisk CLI. Many thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialtones as Progressinband
Hi, exten = 11,1,Playtones(ring) exten = 11,2,Wait(10) exten = 11,102,Busy exten = 11,2,Hangup this plays me the Ringtone what is set in the indications.conf also over an iax2 connection to an other Asterisk with SessionProgress(SIP183). But with this the tone stops: exten = 11,1,Playtones(ring) exten = 11,1,Dial(SIP/11,60,tw) exten = 11,102,Busy exten = 11,2,Hangup I wanna say Asterisk to play the ringtone, what is set on the first Asterisk in the indications.conf, as inband to the SIP-Phones on the second Asterisk. Like the PSTN does. Anybody knows what i have to set? Thx Timm - CPBX Austria by TMS IT-Dienst Hinterstadt 2 4840 Vöcklabruck T: (0720) 50 10 78 (Per ENUM kostenlos erreichbar) M: (0664) 479 79 25 F: (0720) 50 10 78-57 SIP: 2112377 (Terrasip) 0720501078 (Nemox) 0720721226 (PlatinPlus) Meine Mails werden mit Kaspersky AntiVirus überprüft! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Record in mp3
The recording I want is to save some reminders that my users can record. It is not to save a conversation. So I think that there is not an overhead in converting the audio on the fly in this case.But the question is: Asterisk suport generate mp3 files directly? --- Em sex, 24/4/09, Geraint Lee gera...@gmail.com escreveu: De: Geraint Lee gera...@gmail.com Assunto: Re: [asterisk-users] Record in mp3 Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Data: Sexta-feira, 24 de Abril de 2009, 11:50 you probably don't want to record directly to mp3 as there will be an overhead in converting the audio on the fly and this will probably break your call recordings... you should either record in the codec you are using for phone calls (i think?) or in .wav and then convert afterwards (correct me if i'm wrong someone!). 2009/4/24 Jose Enes Mateus jemat...@yahoo.com.br But have you tried to record directly in mp3, without to covert the file? --- Em qui, 23/4/09, Danny Nicholas da...@debsinc.com escreveu: De: Danny Nicholas da...@debsinc.com Assunto: Re: [asterisk-users] Record in mp3 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Data: Quinta-feira, 23 de Abril de 2009, 17:33 The way I read to do this is to use sox to create a wav file, then use lame to convert the wav to mp3. I did this for some MOH files. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose Enes Mateus Sent: Thursday, April 23, 2009 3:28 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Record in mp3 Somebody knows if I can save files in mp3 with the Record command on Asterisk? I try to recompile sox to suport mp3 but Asterisk return the folowing message when I use the Record command: - Executing [...@liberado15:15] Record(SIP/1201-083453c8, /var/spool/asterisk/alarme/alarme-1201-200905121212:mp3) in new stack -- SIP/1201-083453c8 Playing 'beep' (language 'pt_BR') [Apr 23 17:22:36] ERROR[4494]: format_mp3.c:283 mp3_rewrite: I Can't write MP3 only read them. [Apr 23 17:22:36] WARNING[4494]: file.c:378 fn_wrapper: Unable to rewrite format mp3 [Apr 23 17:22:36] WARNING[4494]: file.c:1092 ast_writefile: Unable to rewrite /var/spool/asterisk/alarme/alarme-1201-200905121212.mp3 [Apr 23 17:22:36] WARNING[4494]: app_record.c:272 record_exec: Could not create file /var/spool/asterisk/alarme/alarme-1201-200905121212 I'am doing something wrong? Thanks Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10 - Celebridades - Música - Esportes -Anexo incorporado- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10 - Celebridades - Música - Esportes ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Anexo incorporado- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Record in mp3
Im sure someone will correct this if Im wrong Asterisk cant make direct mp3 records because its not a supported codec. Typically Asterisk records anything as a gsm, ulaw or alaw file, depending on the codec used to run the connection to the phone. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose Enes Mateus Sent: Friday, April 24, 2009 12:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Record in mp3 The recording I want is to save some reminders that my users can record. It is not to save a conversation. So I think that there is not an overhead in converting the audio on the fly in this case.But the question is: Asterisk suport generate mp3 files directly? --- Em sex, 24/4/09, Geraint Lee gera...@gmail.com escreveu: De: Geraint Lee gera...@gmail.com Assunto: Re: [asterisk-users] Record in mp3 Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Data: Sexta-feira, 24 de Abril de 2009, 11:50 you probably don't want to record directly to mp3 as there will be an overhead in converting the audio on the fly and this will probably break your call recordings... you should either record in the codec you are using for phone calls (i think?) or in .wav and then convert afterwards (correct me if i'm wrong someone!). 2009/4/24 Jose Enes Mateus jemat...@yahoo.com.br But have you tried to record directly in mp3, without to covert the file? --- Em qui, 23/4/09, Danny Nicholas da...@debsinc.com escreveu: De: Danny Nicholas da...@debsinc.com Assunto: Re: [asterisk-users] Record in mp3 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Data: Quinta-feira, 23 de Abril de 2009, 17:33 The way I read to do this is to use sox to create a wav file, then use lame to convert the wav to mp3. I did this for some MOH files. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose Enes Mateus Sent: Thursday, April 23, 2009 3:28 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Record in mp3 Somebody knows if I can save files in mp3 with the Record command on Asterisk? I try to recompile sox to suport mp3 but Asterisk return the folowing message when I use the Record command: - Executing [...@liberado15:15] Record(SIP/1201-083453c8, /var/spool/asterisk/alarme/alarme-1201-200905121212:mp3) in new stack -- SIP/1201-083453c8 Playing 'beep' (language 'pt_BR') [Apr 23 17:22:36] ERROR[4494]: format_mp3.c:283 mp3_rewrite: I Can't write MP3 only read them. [Apr 23 17:22:36] WARNING[4494]: file.c:378 fn_wrapper: Unable to rewrite format mp3 [Apr 23 17:22:36] WARNING[4494]: file.c:1092 ast_writefile: Unable to rewrite /var/spool/asterisk/alarme/alarme-1201-200905121212.mp3 [Apr 23 17:22:36] WARNING[4494]: app_record.c:272 record_exec: Could not create file /var/spool/asterisk/alarme/alarme-1201-200905121212 I'am doing something wrong? Thanks _ Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10 http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ - Celebridades http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ celebridades/ - Música http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ m%C3%BAsica/ - Esportes http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ esportes/ -Anexo incorporado- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10 http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ - Celebridades http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ celebridades/ - Música http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ m%C3%BAsica/ - Esportes http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ esportes/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Anexo incorporado- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Veja quais são os assuntos do momento no Yahoo! + Buscados: Top http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ 10 -
[asterisk-users] dahdi_tool reports that dahdi_dummy is UNCONFIGURED
Usually I used real Digium cards in asterisk systems, so I'm running into this for the first time. dahdi_tool reports that dahdi_dummy is in state UNCONFIGURED. This isn't super surprising, as it seems like the configuration files for DAHDI are really intended only for configuring real physical cards. But that begs the question: Is there any legal DAHDI configuration for dahdi_dummy such that dahdi_tool will stop complaining about an UNCONFIGURED dahdi_dummy? I tried making up bs values for span, etc. and of course dahdi_cfg didn't like those. dahdi_scan reports [1] active=yes alarms=UNCONFIGURED description=DAHDI_DUMMY/1 (source: RTC) 1 name=DAHDI_DUMMY/1 manufacturer= devicetype=DAHDI Dummy Timing location= basechan=1 totchans=0 irq=0 dahdi_genconf should in theory auto-generate a legal configuration, which can then be loaded with dahdi_cfg But no such thing happens, I still end up with dahdi_dummy reporting as UNCONFIGURED. In the event that it's relevant, MeetMe rooms are working, and mixing audio. However I'm having SIP audio cut-outs and I thought those may be related. Is it a bug that I don't seem to be able to make a valid configuration for a dahdi_dummy card, or is that expected behavior? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Record in mp3
On Friday 24 April 2009 12:40:45 Danny Nicholas wrote: Im sure someone will correct this if Im wrong Asterisk cant make direct mp3 records because its not a supported codec. Typically Asterisk records anything as a gsm, ulaw or alaw file, depending on the codec used to run the connection to the phone. Well, even for the mp3 support that we do have, it's that we don't have recording support for the format. We do have read support, though, which converts the format on the fly from the compressed mp3 format back into uncompressed signed linear audio. Given the patent protection on the MPEG standard, distributing an MPEG-capable audio compression could raise liability issues. Secondarily, MPEG audio compression takes a lot of CPU. Until the last few years, desktop CPUs weren't even capable of doing realtime MPEG audio compression, which is necessary if you're going to have the recording ready by the time the audio input is terminated. Above and beyond that, even modern CPUs are limited in how many concurrent streams can be MPEG-compressed, which may cause problems if you're encoding multiple channels to MP3 at the same time. Probably the best possible fix would be DSPs capable of audio MPEG-encoding. Not only does this solve concurrency, but since DSPs are generally already licensed for use with a particular codec, liability is addressed, as well. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Step-by-Step Asterisk and Cisco 1760 Help - (Solved)
Jonathan And Dan, Thank you both for the responses. But the problem turned out to be that - I'm an idiot. I was placing both calls to our company number for the voicemail system which I thought was part of a hunt group. As it turns out, only one call can come in at a time on that number, and so I was getting a busy signal because the line really was busy (go figure). I did learn some things, and I post them here for the benefit of all. First of all Dan thanks for showing me how to get at some debug information on the Cisco 1760. I ran the command: #show call history voice brief I got several sections of text that looked like this (but more of them) on the screen: 17BD : 878 3699467830ms.572 +3470 +214330 pid:2212 Originate 2572210 dur 00:03:30 tx:10567/1770108 rx:10542/1686720 10 (normal call clearing (16)) Telephony 0/0 (878) [0/0] tx:210860/210860/0ms g711ulaw noise:-65dBm acom:14dBm long duration call detected:n long dur callduration :n/a timestamp:n/a That first hex number is the call-id (17BD). Seems there were three sections of text for each call. (One for calling and 2 sections for disconnect) After I figured out which ones were associated with my failed call, I ran the second command that Dan suggested: #show call history voice id call-id (show call history voice id 17BD) And it spit out a lot of stuff but I eventually saw an error code: InternalErrorCode=1.1.182.11.26.0 I found this webpage that helped to decode the error. http://docwiki.cisco.com/wiki/Cisco_IOS_Voice_Troubleshooting_and_Monitoring_--_Cisco_VoIP_Internal_Error_Codes The important bits seem to be: - 182 Hardware resources unavailable 26 No application The system could not find an application to take the incoming call. Check your call application and dial peer configurations. Second the configuration changes that Jonathan suggested worked just fine. Thank you for showing me another way to make this work with trunk groups. See below for changes he suggested to my Cisco 1760 configuration: trunk group Outbound description - Outbound calling hunt group hunt-scheme sequential ! voice-port 0/0 trunk-group Outbound 1 ! voice-port 2/0 trunk-group Outbound 2 ! dial-peer voice 2212 pots trunkgroup Outbound description Outbound call hunting destination-pattern .T ! One note here is that it failed the same way, (because the line really was busy) but show call history voice id call-id did not show any error codes in this trunk group configuration. Not sure why. Hope this helps someone and thanks again guys! Jimmy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of Dan Austin Sent: Thursday, April 23, 2009 10:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Step-by-Step Asterisk and Cisco 1760 Help Jimmy wrote: Dan thank you, yes that seems to help. It looks like the bridging is happening now and I see the light come on in the second FXO port, but then I get a busy signal after that and the call still does not complete. If I set the second line as priority 1 it completes the first call on that line and second call gets the busy on the first line. I even tried moving the lines to a different FXO card and the result is the same. Here is my current config for the cisco dial-peers: dial-peer voice 2212 pots preference 2 destination-pattern .T port 2/0 forward-digits all ! dial-peer voice 2211 pots preference 1 destination-pattern .T port 0/0 forward-digits all Thanks again Dan, I think I am much closer now. I think the suggestion by Jonathan will help you finish off your problem, but what you have listed should also have worked. What does your SIP dial-peer look like? After the second call fails, try issuing this command on the cisco: #show call history voice brief Then identify the call id of the failed call and use this: #show call history voice id call-id That will at least tell you why the call failed. I have not worked a lot with the Cisco analog interfaces, but I have setup a healthy number of ISDN ports, with the type of roll-over that you are trying to setup. I can try to help with the Cisco debug logs if you want to take this off list. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Record in mp3
On Fri, Apr 24, 2009 at 01:21:50PM -0500, Tilghman Lesher wrote: On Friday 24 April 2009 12:40:45 Danny Nicholas wrote: Im sure someone will correct this if Im wrong Asterisk cant make direct mp3 records because its not a supported codec. Typically Asterisk records anything as a gsm, ulaw or alaw file, depending on the codec used to run the connection to the phone. Well, even for the mp3 support that we do have, it's that we don't have recording support for the format. We do have read support, though, which converts the format on the fly from the compressed mp3 format back into uncompressed signed linear audio. Given the patent protection on the MPEG standard, distributing an MPEG-capable audio compression could raise liability issues. Use Ogg (well, technically it's Ogg/Vorbis here) How widely is Ogg/Speex supported? I figure it gives smaller files. Alternatively, WAV/GSM (wav49 in Asterisk) is quite widely-used as well. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail number of rings
I'd be really happy if users could use the voicemail menu to change the number of rings until voicemail picks up. It seems like the current model of separate Dial and Voicemail commands would make that difficult, but is there any plan for such a feature in the future? How about a workaround or 3rd party add on? I store the dial timeout for each user in a database, so I know I could make my own little menu for them to set the number of seconds, but people are always a little stupefied by the fact that it's not on the voicemail menu. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail number of rings
Here's a hack to implement this feature
Create global variable RINGS
Use playback and read to let user enter new number
Set RINGS to entered value
Change dial command(s) to include ${RING}
Here's a snippet from my dialplan that changes the default operator
exten = 644,1(start7),Playback(record/nightopext)
exten = 644,n,NoOp(executando - ${extensao} - )
exten = 644,n,BackGround(beep)
exten = 644,n,Read(digito,,3)
exten = 644,n,Gotoif($[ ${LEN(${digito})} != 3]?start7)
exten = 644,n,SayDigits(${digito})
exten = 644,n,Set(GLOBAL(NIGHTOP)=${digito})
exten = 644,n,Set(DB(Nightop/ext)=${digito})
exten = 644,n,BackGround(vm-goodbye)
exten = 644,n,Hangup()
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Moffett
Sent: Friday, April 24, 2009 3:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] voicemail number of rings
I'd be really happy if users could use the voicemail menu to change the
number of rings until voicemail picks up.
It seems like the current model of separate Dial and Voicemail commands
would make that difficult, but is there any plan for such a feature in
the future? How about a workaround or 3rd party add on?
I store the dial timeout for each user in a database, so I know I could
make my own little menu for them to set the number of seconds, but
people are always a little stupefied by the fact that it's not on the
voicemail menu.
___
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Re: [asterisk-users] listen to prompt before bridging call.
but you dont really need both parties listen the same message. You can call the second partie (a new call maybe origiate action) when he answer you play both the message in each channel if he press one you bridge the channels. you first send rings to the caller then you make a new call and play the message and then you bridge the calls using the brige app in asterisk 1.6. David 2009/4/24 Deepak dlal...@gmail.com Hi, Can someone please help to resolve the followinng issue: We would like an asterisk user to call a number and when the called party picks up the phone, we play a message (press 1 to accept call, 2 to reject call). Only when the called party presses 1, do we bridge the call and the two parties can communicate. What we would like though is that the person who makes the call be able to listen to the message press 1 to accept call, 2 to reject call) that is played to the called party BUT not be able to communicate with him untill he presses 1. Is this possible in asterisk using php/agi? Any pointers hightly appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail number of rings
Adam Moffett schrieb: I'd be really happy if users could use the voicemail menu to change the number of rings until voicemail picks up. It seems like the current model of separate Dial and Voicemail commands would make that difficult, but is there any plan for such a feature in the future? How about a workaround or 3rd party add on? I store the dial timeout for each user in a database, so I know I could make my own little menu for them to set the number of seconds, but people are always a little stupefied by the fact that it's not on the voicemail menu. There's my MiniVM by Olle the great. http://www.asterisk.org/doxygen/trunk/Config_minivm_examples.html http://www.voip-info.org/wiki/view/Asterisk+cmd+MiniVM http://www.das-asterisk-buch.de/2.1/minivm.html Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] listen to prompt before bridging call.
Deepak wrote: Hi, Can someone please help to resolve the followinng issue: We would like an asterisk user to call a number and when the called party picks up the phone, we play a message (press 1 to accept call, 2 to reject call). Only when the called party presses 1, do we bridge the call and the two parties can communicate. What we would like though is that the person who makes the call be able to listen to the message press 1 to accept call, 2 to reject call) that is played to the called party BUT not be able to communicate with him untill he presses 1. Is this possible in asterisk using php/agi? Any pointers hightly appreciated. I can come up with an easy way to do part of what you want. If you are using the Dial application, you can use the M option to run a macro on the called channel when he answers. Within that macro, you can play prompts to accept or reject the call. The problem is that this does not allow for the calling party to also hear the prompts. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium Fax for Asterisk
Anyone knows what should be the configuration of the new solution of Digium for fax in order to send and receive faxes from PSTN to a fax machine through an ATA implementing T38 protocol? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] listen to prompt before bridging call.
In order to get 2-way audio without the bridge, you will have to Answer before Dialing. That's going to mess with your CDR, but you could make this dial a custom function that you're not going to worry about in the CDR. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson Sent: Friday, April 24, 2009 4:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] listen to prompt before bridging call. Deepak wrote: Hi, Can someone please help to resolve the followinng issue: We would like an asterisk user to call a number and when the called party picks up the phone, we play a message (press 1 to accept call, 2 to reject call). Only when the called party presses 1, do we bridge the call and the two parties can communicate. What we would like though is that the person who makes the call be able to listen to the message press 1 to accept call, 2 to reject call) that is played to the called party BUT not be able to communicate with him untill he presses 1. Is this possible in asterisk using php/agi? Any pointers hightly appreciated. I can come up with an easy way to do part of what you want. If you are using the Dial application, you can use the M option to run a macro on the called channel when he answers. Within that macro, you can play prompts to accept or reject the call. The problem is that this does not allow for the calling party to also hear the prompts. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Fax for Asterisk
Anthony Cascante wrote: Anyone knows what should be the configuration of the new solution of Digium for fax in order to send and receive faxes from PSTN to a fax machine through an ATA implementing T38 protocol? Fax for Asterisk does not do FAX relay; it terminates or originates FAXes. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] want to set up text based adventure for asterisk
[top posting continued] There was an offer, a long time ago, to have all of the prompts for Zork re-done by Allison in a dramatic reading format, but nobody (coughSIMONcough) ever got a text file with all the strings together, and the time to re-write ZoIP to use audio files of each reply type would be non-trivial. Personally, I think this is a great demo for various voice recognition companies to test their mettle on. Same game, different voice recognition platforms, see how well each VR platform performs. Let people try it out for themselves in something other than a very canned demo environment. JT On Apr 23, 2009, at 1:57 PM, Tim Nelson wrote: A good place to start is here: http://www.venturevoip.com/news.php?rssid=1513 FreePBX includes a module called 'Zoip' which allows you to play Zork via a Text-to-speech engine. Why on Earth someone would want to do so is beyond me but hey... why not. :-) Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Eric Fort eric.f...@gmail.com wrote: Anyone know where I could find a good beginning for using asterisk and the text based game adventure together such that I could play from the nearest phone? Thanks, Eric --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk EC2
Has anyone been able to get asterisk 1.6 running under Xen or Amazon EC2? If yes, can you share your experience please? Is it usable in a production environment? How is the sound quality? Am I likely to suffer from latency issues if the extensions are not located in the US? Any pitfall that I should be aware of? Cheers -- Aryan Ameri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cheap CHEAP ata
Have you checked ebay? Just beware that there are a lot of ATAs on Ebay that are locked to Vonage or similar providers. While they are not impossible to unlock, it requires considerable time and good Linux networking experience, as the process generally involves creating an isolated world (with its own DNS, etc) that mimics the provider and then updates configuration files. If you want a lot of cheap ATAs it might be worth your while to set up such a system, as most of the work would be for the first one and the rest would be relatively easy. On the other hand, if you weren't anticipating this problem, you might get stuck with a bunch of useless paperweights, which wouldn't make the total cost of the solution very cheap. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Record in mp3
Secondarily, MPEG audio compression takes a lot of CPU. Until the last few years, desktop CPUs weren't even capable of doing realtime MPEG audio compression, which is necessary if you're going to have the recording ready by the time the audio input is terminated. Above and beyond that, even modern CPUs are limited in how many concurrent streams can be MPEG-compressed, which may cause problems if you're encoding multiple channels to MP3 at the same time. Well, actually it's lot of CPU for encoding 44kHz stream. I wonder how it would scale to encode 8kHz.. We currently do a daily routine to compress all ulaw files to mp3 at night time, and it takes ~6 hours of processing on 1 CPU (no parallel processing). Regarding legal reasons, can't it be linked with lame within asterisk-addons? Regards, Atis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk EC2
I am running AsteriskNOW as a VM using OpenSUSE 11 and Xen. Download the .iso to the local disk and point the installation source at that. Also, make sure to choose full virtualization NOT paravirtualized for the VM, graphics and the NIC. I also recommend using a DHCP lease and use a custom MAC so that you don't have to assign the static IP using ifconfig or Yast. I think this covers everything you need to get it running. I am using an SPA-3000 for outbound calls over my POTS line and using a combo of SIP hard and soft phones with very good sound quality. Hope this helps! -mikeg On Apr 24, 2009, at 6:30 PM, Aryan Ameri wrote: Has anyone been able to get asterisk 1.6 running under Xen or Amazon EC2? If yes, can you share your experience please? Is it usable in a production environment? How is the sound quality? Am I likely to suffer from latency issues if the extensions are not located in the US? Any pitfall that I should be aware of? Cheers -- Aryan Ameri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cheap CHEAP ata
Thanks for the info!!! 2009/4/24 Wilton Helm wh...@compuserve.com Have you checked ebay? Just beware that there are a lot of ATAs on Ebay that are locked to Vonage or similar providers. While they are not impossible to unlock, it requires considerable time and good Linux networking experience, as the process generally involves creating an isolated world (with its own DNS, etc) that mimics the provider and then updates configuration files. If you want a lot of cheap ATAs it might be worth your while to set up such a system, as most of the work would be for the first one and the rest would be relatively easy. On the other hand, if you weren't anticipating this problem, you might get stuck with a bunch of useless paperweights, which wouldn't make the total cost of the solution very cheap. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't dial out until I dial in once
When I restart or reboot I can not dial out. The dial() incorrectly sees dahdi/1 as busy. I call in once from a cell phone, which is successful then I can call out with out issue. Any ideas would be much appreciated. Sangoma B600de asterisk-1.6.0.9 dahdi-linux-2.1.0.4 linux-2.6.28-gentoo-r5 wanpipe-3.3.16 ###chan_dahdi.conf ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2009-02-21 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=fromPSTN usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=no ;callwaitingcallerid=yes callwaitingcallerid=no threewaycalling=no transfer=yes canpark=yes cancallforward=yes callreturn=no echocancel=no echocancelwhenbridged=no relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;immediate=yes ;Sangoma AFT-B600 [slot:4 bus:5 span:1] wanpipe1 context=fromPSTN group=0 echocancel=no signalling = fxs_ks channel = 1 context=fromPSTN group=0 echocancel=no signalling = fxs_ks channel = 2 context=fromPSTN group=1 echocancel=no signalling = fxs_ks channel = 3 context=fromPSTN group=1 echocancel=no signalling = fxs_ks channel = 4 context=phones group=2 echocancel=no signalling = fxo_ks channel = 5 ~~~ Kind regards, Mike van der Stoop ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cheap CHEAP ata
Google shows one result for low cost ATA: http://www.trixbox.org/forums/vendor-forums-non-certified/linksys-cisco/linksys-pap2-and-rt31p2-low-price Buyer beware! Those are probably counterfeit! On Fri, Apr 24, 2009 at 19:15, Wilton Helm wh...@compuserve.com wrote: Have you checked ebay? Just beware that there are a lot of ATAs on Ebay that are locked to Vonage or similar providers. While they are not impossible to unlock, it requires considerable time and good Linux networking experience, as the process generally involves creating an isolated world (with its own DNS, etc) that mimics the provider and then updates configuration files. If you want a lot of cheap ATAs it might be worth your while to set up such a system, as most of the work would be for the first one and the rest would be relatively easy. On the other hand, if you weren't anticipating this problem, you might get stuck with a bunch of useless paperweights, which wouldn't make the total cost of the solution very cheap. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
Hi,
Thanks for your reply
I have tried the HUMAN as you suggested , but still my problem does not get
solved.
I am answering the call and then running the amd.
Below is the log.
-- AMD: SIP/sip-58ab (null) (null) (Fmt: 4)
Apr 25 00:26:07 NOTICE[27310]: app_amd.c:134 isAnsweringMachine: AMD using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300]
totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50]
maximumNumberOfWords [5] silenceThreshold [256]
-- AMD: HANGUP
vm3*CLI
Any help is highly appreciated.
Thanks.
On Fri, Apr 24, 2009 at 4:03 PM, Matt Florell astma...@gmail.com wrote:
Hello,
Well, depending on the version of app_amd that you used when you added
it to Asterisk 1.2, you might need to use HUMAN and MACHINE as the
possible AMDSTATUS instead of AMD_PERSON and AMD_MACHINE. The
AMDSTATUS was changed at some point in the app_amd code, not sure why
they changed it, but that might be your issue.
Also, since you are calling your own number you might want to do an
Answer on the call before running AMD, not sure if that would cause
the hangups you are seeing or not, but it's something to try.
MATT---
On 4/24/09, Sam Hawkin gvrt...@gmail.com wrote:
Hi,
Thanks for your reply
We are using the Asterisk 1.2.4.
and below the dialplan path. we are orginating the call to
my number and connection it to context cdtest and extension 1.
[cdtest]
exten = 1,1,NoOp( cb amd issue testing )
exten =
1,2,Macro(Cb-old|/root/business_hours|/root/business_hours)
[macro-Cb]
exten = s,1,NoOp( values in CB arg1 ${ARG1} arg2 ${ARG1} )
exten = s,2,AMD
exten = s,3,GotoIf($[${AMDSTATUS}=AMD_PERSON]?4:7)
exten =
s,4,NoOp(Humanplaying--${ARG1})
exten = s,5,Playback(${ARG1})
exten = s,6,Hangup
exten = s,7,GotoIf($[${AMDSTATUS}=AMD_MACHINE]?8:11)
exten =
s,8,NoOp(Machine---playing--${ARG2})
exten = s,9,Playback(${ARG2})
exten = s,10,Goto(s|12)
exten = s,11,Playback(${ARG1})
please suggest our what might be the problem.
Any help is highly appreciated.
Thanks.
On Thu, Apr 23, 2009 at 8:36 PM, Matt Florell astma...@gmail.com
wrote:
On 4/23/09, Sam Hawkin gvrt...@gmail.com wrote:
Hi All,
I am trying to use the AMD (Answering Machine Detect).
But it is not sending the AMD_Status as either
the Human or Machine, it hangs up in middle.
can any one suggest us, what might be the problem
and possible solution to it.
below is the log
-- Executing AMD(SIP/sip-ffe0, ) in new stack
-- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4)
Apr 23 08:00:26 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD
using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500]
afterGreetingSilence
[300] totalAnalysisTime [5000] minimumWordLength [120]
betweenWordsSilence
[50] maximumNumberOfWords [5] silenceThreshold [256]
-- AMD: HANGUP
What version of Asterisk are you running this on?
What is the dialplan path that this is running through?
MATT---
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[asterisk-users] Callweaver/Asterisk 'outgoing' spool
c/c'd to the Asterisk list as this is probably relevant to Asterisk as well. My detailed study of the operation of the 'outgoing' directory reveals that TXFax() does not delete an expired fax batch file (In the 'outgoing' directory) until after the end of the dial plan execution. Is there a way to get it to delete any expired fax batch file before the end of the dial plan execution? Another way to put this, is there any way I can run my System() call after an expired batch file has been deleted? I want to run a script *after* it has deleted any batch files it no longer needs. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
