[asterisk-users] The identifier parameter in Dial() command
Hi All, I am new to Asterisk. Now I got one question on the identifier parameter of the Dial() command. I saw as below: exten = 20,1,Dia(Zap/3/5551234). Would you please let me know the meaning of 5551234? Thanks, Songtao___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The identifier parameter in Dial() command
I would strongly suggest you browse: http://www.asteriskdocs.org/ Kind regards, PaulH Songtao Yu wrote: Hi All, I am new to Asterisk. Now I got one question on the identifier parameter of the Dial() command. I saw as below: exten = 20,1,Dia(Zap/3/5551234). Would you please let me know the meaning of 5551234? Thanks, Songtao ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The identifier parameter in Dial() command
Hello Dial(Zap/3/5551234) here 3 is the channel. 5551234 is PSTN number how ever you will have a better understanding after reading this http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial 2009/9/7 Songtao Yu yustao_2...@hotmail.com Hi All, I am new to Asterisk. Now I got one question on the identifier parameter of the Dial() command. I saw as below: exten = 20,1,Dia(Zap/3/5551234). Would you please let me know the meaning of 5551234? Thanks, Songtao ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SUN and PRI ?
On Mon, 7 Sep 2009 02:43:54 +0100 Ex Vito ex.vitor...@gmail.com wrote: The system specs mention PCIe expansion slots, so your only option is the TE420B. -- exvito Hi Ex Vito, shouldn't the card be low profile ? Thanks and have a nice day. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] one touch recording not working lately in asterisk
Hi. Has anyone else seen the problem I am having where one touch recording causes a seg fault? I filed a bug, however I would like to know if anyone else has seen this and if there is a work around unless no one uses this very nice feature. Thanks much for any ideas on this subject. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Record conversations and place soundfile in user-directory
Hello list, is it possible with the monitor-command to record conversations and place the soundfile in a pre-defined directory per user ?! So when extension 200 presses '*#' to record the conversation, the resulting sound file is written to his home directory on the Samba-server. This way each user has his own directory with its recordings that no one else can access (as default rights to the share are those of the user and nothing for others). Asterisk is running as root and it thus able to write to each home directory. But only the user (with his fixed phone extension) can access the share with his sound files. So user 'jonas' has extension 200. When extension 200 presses '*#' the conversation is recorded and the resulting sound file is written to /var/samba/profiles/jonas/recordings. Samba is running on the Asterisk-server. In features.conf a see 1 rule to define the key-combination for recording conversations. The recordings are then written to a default location /var/spool/asterisk/monitor. Thanks for the feedback. Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Send 200 OK with SDP instead of 183 with SDP when ringing starts
You are right, I think the provider has some problems, and this should be fixed. But is also good to know that I can do this workaround for the worse case scenario. thank you ! Olle E. Johansson wrote: 4 sep 2009 kl. 13.40 skrev Marius Ciorecan: Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through which I connected an external PSTN line. I use it as carrier for VoIP calls. I can make successfully calls, but there's one problem, I receive 200 OK with SDP with delay (sometimes more than 30 seconds). So when I make a call through asterisk I receive intially: - 100 Trying - 183 Session Progress, with SDP when the called number respond, I start receiving RTP with voice, also the called receives voice from me, but only after a while asterisk sends 200 OK with SDP. I'm not sure if the problem is from asterisk or from the telephony provider (I think the provider). Is there a posibility to replace 183 with 200 OK ? I mean from the moment when ringing starts to receive 200 OK with SDP instead of 183 ? You can answer() at any point in the dialplan - and that will generate a 200 OK. Like exten = marius,1,answer() exten = marius,n,dial(sip/mariusphone) This will generate an immediate 200 ok, regardless if mariusphone is busy or gone from the network. It's propably not what you want. Asterisk sends 200 OK on the incoming call as soon as we get a connection reply, a 200 OK or something similar in other protocols on the outbound call. For some reason, this happens very late for you and causes your problem. Could be some issue with the service provider, your ISDN connection or -even worse - your IAX2 trunk... (could not resist) Please start with debugging that and solving the real issue, instead of trying to change the functionality in Asterisk :-) Regards, /O --- o...@edvina.net - http://edvina.net Open Unified Communication - building platforms with SIP and XMPP From PBX to large scale implementations for carriers. Contact us today! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange beep when using VoiceMailMain application
Hi Martin,
Thanks for the answer!
2009/9/7 Martin asteriskl...@callthem.info
that's probably for ADSI phones ... chan_local confuses the VoiceMailMain
app
and you hear it ...
I'm experiencing this with different SIP phones and softphones.
Why do you need to call it via chan_local ? Can't
you do Macro or just
call VoiceMailMain directly ?
That's a good question. The reason is that we were experiencing problems
with some DECT phones using the g729 codec and accessing the voicemail. The
phones stopped playing media when they stopped receiving RTP packets for a
few seconds, and usually this would happen between the locutions of the
VoiceMailMain application. So the solution we thought of was to use the Page
application in order to play some background audio at the same time as the
Voicemail. Something like this:
exten = _X.,1,Page(Local/${ext...@voicemail-page
Local/backgro...@voicemail-page,dq)
[voicemail-page]
exten = _X.,1,VoiceMailMain(${ext...@mydomain.com exten...@mydomain.com)
exten = background,1,MusicOnHold()
This has worked pretty well except for this weird beep at the beginning of
the call. While figuring out what might be the problem I observed this
happened if I tried to call VoiceMailMain via chan local.
What do you think?
Best regards,
Santi
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Re: [asterisk-users] stutter playback
This sounds more like the alarm system putting pulses/tones on the line (maybe the alarm has a dialler/anti-cut-line-detection? So, as the alarm is adding stuff AFTER the asterisk box - I doubt you will see anything on the PC itself. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Samad Sent: 22 August 2009 04:48 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] stutter playback On Fri, Aug 21, 2009 at 08:53:23AM -0400, Steve Totaro wrote: On Fri, Aug 21, 2009 at 8:39 AM, Alex Samad a...@samad.com.au wrote: Hi I had a working system, until recently - its asterisk 1.6.1 from debian - not the lastest as the last doesn't seem to work. but somebody who rang me said my voice mail announcement was all stuttery. so i dialed my voicemail box and its really stuttery... so I have done a reboot and its just as bad, now I am not sure what to check to try and get this working again . Alex I would check cpu, diskpace, memory, I/O, network wasn't that, I have a alarm system on the backup pstn line, seems like there is something wrong there, cause when I remove the alarm system from the equation everything seems okay, so I am guessing it was causing some problem on my tdm410 card. strange thing is i did not see any spikes on io , cpu, network... Alex -- Think of it! With VLSI we can pack 100 ENIACs in 1 sq. cm.! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Freepbx database followme disable/enable value
Hello, I am writing an AGI script to achieve the following - Users can Disable/Enable followme from their extension. They can also change the followme details from their extensions. I have looked at the follow me table for freepbx. I can't see the field for the values enabling/disable followme. Is this value stored in the database? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent Agent Login from a second extension
The only way around the 'auto-logout' problem I found was to call a script when agents login. This script checks to see if they are already logged in or not - then, if they are, it does whatever I want (I manually log them off the other phone first - you could play a message instead). HTH Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shanavaz E A Sent: 02 September 2009 07:27 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Prevent Agent Login from a second extension Hi friends, Is there any way to prevent an Agent from logging in from a second extension if he is already logged on from an extension. Right now, the scenario is if he login from a second extension, asterisk will automatically log him off from first extension. What I need is that asterisk should tell him that he is already logged on from an extension and should prevent him from logging in again from another extn. The problem with existing scenario is that, I am not getting CDR record for the automatic log out event. I need this for evaluation purposes. I am using asterisk 1.2.30. I have 1.4 also but that also is having the same behavior. Thanks in advance for any help. Regards Shanavaz. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] onnecting two asterisk using B410p BRI cards
...and did you switch the termination dip switches over (on the NT ports of the B410P card)? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of voip crazy Sent: 17 August 2009 07:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] onnecting two asterisk using B410p BRI cards I just plug the junper in NT mode with no success. VoipCrazy 2009/8/15 Paul Hales pdha...@optusnet.com.au: Use a standard network cable - but you have to activate the 'terminate' jumper on the NT end. - Also, the new BRI stuff in dahdi is much easier to work with than misdn. PaulH voip crazy wrote: Hello all, I'm trying to conect two asterisk servers using two B410p Digium cards. One card on each server. I just setting up the first BRI port on server A as nt_ptp and the first BRI port on server B as te_ptp. I use an ethernet wire to connect the first port of server A (nt_ptp) with the first port on server B (te_ptp) but the port light cotinues blinking on red on both sides once the cable was pluged. Then I use an isdn crossover wire with this king of schema and the lights get blinking red again. Tx+ 3 --+ +- 3 . X Rx+ 4 --+ +- 4 . Tx- 5 --+ +--5 . X Rx- 6 --+ +--6 In both servers when I do in asterisk CLI misdn shos stacks, the port one on each machine shows Server A: BEGIN STACK_LIST: * Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 Server B: BEGIN STACK_LIST: * Port 1 Type NT Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 Which kind of cable should I use? Why both in ports L1Link is failed? How could I solve that? Any clue will be welcomed. Thanks in advance. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Another method to do that is to make the Asterisk monitor output dummy SIP calls rather than sound files. Oreka/Orex can listen to those. Looking for volunteers to test that: http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample This allows recording non-VoIP links, VoIP links where tapping is not convinient, or more selective recording of VoIP calls. Is this similar or the same as the portion of my post that you snipped? Different in many ways, which is why I snipped it. Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. (Actually: recorded calls are sent as RTP streams to the Orex/Oreka server) This records outside of Asterisk. Thus it lacks information available in Asterisk (who really called who). OTOH, it is Asterisk-specific. We actually considered implementing something similar to the Sangoma interface in our driver but realised that doing it in Asterisk would probably be more useful. The overheade seems reasonable. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE420P configuration
Hello I am trying to configure TE420P but i am confused what to give chan_dahdi :( Below is configuration i am using for TDM400P Please help what changes to make in it... Please provide a link as well [trunkgroups] [channels] ;default for channels switchtype=national rxwink=300 hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 busydetect=yes busycount=3 ;busypattern=500,500 ; DEFINING CHANNEL context = incoming_context_for_ptcl signalling=fxs_ks channel= 1 -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help setting IAX variables.
Hi All, I am new to Asterisk and want to perform following on my test project. I have 2 Asterisk servers SA and SB and an IAX trunk from SA to SB. Now I set some variables on SB in the same context where an IAX call lands. My question is , is it possible to access these variables in dialplan of SA? If yes then how? I know about IAXVAR application where variables set in source server of IAX channel can be access from destination server... Any help is greatly appreciated. ---Asterisk User ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Another method to do that is to make the Asterisk monitor output dummy SIP calls rather than sound files. Oreka/Orex can listen to those. Looking for volunteers to test that: http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample This allows recording non-VoIP links, VoIP links where tapping is not convinient, or more selective recording of VoIP calls. Is this similar or the same as the portion of my post that you snipped? Different in many ways, which is why I snipped it. Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. (Actually: recorded calls are sent as RTP streams to the Orex/Oreka server) This records outside of Asterisk. Thus it lacks information available in Asterisk (who really called who). OTOH, it is Asterisk-specific. We actually considered implementing something similar to the Sangoma interface in our driver but realised that doing it in Asterisk would probably be more useful. The overheade seems reasonable. Sorry, I fail to see the difference besides Sangoma implemented it in their Wanpipe drivers and you are attempting copy their idea and do it in Asterisk. Your quote This allows recording non-VoIP links, VoIP links where tapping is not convenient (edited to fix your spelling mistake), or more selective recording of VoIP calls. Isn't that more or less the same thing I said that you snipped, Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. This isn't the biz list, nor the dev list. Snipping out the reference of Sangoma being able to do RTP tap and suggesting people use your experimental dev branch doesn't really help users very much. I really enjoy your use of selective snipping, quoting, and taking things out of context to manipulate threads. You should be a reporter. Too bad it doesn't work on me and I will call you out on it. Please let us users know when your branch gets merged into a Stable Release -- Senior Systems and Network Administrator Triple Canopy, Inc., 2250 Corporate Park Drive, Suite 300 ph. +1.703.673.5191 mob.+1.240.938.1212 FAX.+1.703.673.1279 steve.tot...@triplecanopy.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Feature request: Meetme and invisible users
Hi, Sometimes during a confcall I generate calls into meetme to playback some announcements... If during that time someone joins the conference, the number of participants is announced counting my announcement call... Will it be possible to add an option to the meetme application to mark a participant unvisible and not count it in meetmecount? Basically instead of saying there are 3 other participants in this conference call, asterisk says there are 4 participants in this conference call because my announcement call is considered as a regular participant. Regards, emrah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi/DTMF problem
I posted about this before without ever finding a solution, so I'm trying again. I am upgrading from Fedora 8, asterisk-1.4.21, and zaptel 1.4.11 to Fedora 11, asterisk-1.6.1 and dahdi-2.1 . I have a couple of VOIP phones via SIP and one Wildcard TDM400P REV I with 3 FXS and one FXO port. I am using the same extensions.conf with as few mods as possible. Obviously I have had to convert from a zapata.conf to a chan_dahdi.conf (auto-generated with dahdi_genconf). Almost everything works, with the major exception that incoming calls through the FXO line are dropped as soon as there is a button press. The error logged is: [Aug 23 18:15:39] WARNING[6532] chan_dahdi.c: Cannot handle frames in 2 format [Aug 23 18:15:39] WARNING[6532] file.c: Failed to write frame Which looks like this bug: https://issues.asterisk.org/view.php?id=15129 The bug report hasn't really been updated in almost 4 months. I understand this means that one codec was received while another was expected, but how do I fix it? Where can I look to debug this? This is a show stopper for me because I use a menu to separate calls for my wife, myself, and a local organization into separate mailboxes and distinctive rings. Thanks for any advice, --Greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium hardware support ?
Hello What is your Asterisk problem?, may be I can help you... I had configure a T1 Card TE121 connected with and AVAYA PBX Best regards -Mensaje original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] En nombre de mancyb...@gmail.com Enviado el: Domingo, 06 de Septiembre de 2009 09:33 a.m. Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] Digium hardware support ? Hi All, does Digium provide a service support for a compatibility question about their PRI hardware ? Thanks and have a nice day. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Teleperformance values: Integrity - Respect - Professionalism - Innovation - Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE420P configuration
http://www.voip-info.org/wiki/view/Digium+TE420P http://www.voip-info.org/wiki/view/Digium+TE420PThe best resource is also quite On Mon, Sep 7, 2009 at 3:14 PM, ABBAS SHAKEEL shakeel.abbas@gmail.comwrote: Hello I am trying to configure TE420P but i am confused what to give chan_dahdi :( Below is configuration i am using for TDM400P Please help what changes to make in it... Please provide a link as well [trunkgroups] [channels] ;default for channels switchtype=national rxwink=300 hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 busydetect=yes busycount=3 ;busypattern=500,500 ; DEFINING CHANNEL context = incoming_context_for_ptcl signalling=fxs_ks channel= 1 -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Another method to do that is to make the Asterisk monitor output dummy SIP calls rather than sound files. Oreka/Orex can listen to those. Looking for volunteers to test that: http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample This allows recording non-VoIP links, VoIP links where tapping is not convinient, or more selective recording of VoIP calls. Is this similar or the same as the portion of my post that you snipped? Different in many ways, which is why I snipped it. Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. (Actually: recorded calls are sent as RTP streams to the Orex/Oreka server) This records outside of Asterisk. Thus it lacks information available in Asterisk (who really called who). OTOH, it is Asterisk-specific. We actually considered implementing something similar to the Sangoma interface in our driver but realised that doing it in Asterisk would probably be more useful. The overheade seems reasonable. Sorry, I fail to see the difference besides Sangoma implemented it in their Wanpipe drivers and you are attempting copy their idea and do it in Asterisk. Your quote This allows recording non-VoIP links, VoIP links where tapping is not convenient (edited to fix your spelling mistake), or more selective recording of VoIP calls. Isn't that more or less the same thing I said that you snipped, Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. And what if the call does not go through a TDM card? And ore importantly: how can you tell who is the caller and who is the callee? The rtp-tap interface basically tells you that channel X had a call at time Y. If you control recording through the monitoring interface of Asterisk you can start and stop the recording when you need it. You can also provide better information aobut the call. But again, it means that this is part of Asterisk, and I figure Sangoma has quite a few non-Asterisk customers. This isn't the biz list, nor the dev list. Snipping out the reference of Sangoma being able to do RTP tap and suggesting people use your experimental dev branch doesn't really help users very much. My message was an explicit call for testers, if you haven't noticed :-) I snip content that is not relevant to my reply. Whoever reads this list already read about the Sangoma interface previously. I had nothing to say about it. It was not related to that new branch. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] automatic calls
hi, can anyone knows a way to make automatic calls from a list of numbers stored in a file, one by one, as the calls hangs up. EX: 1º call - hang up - 2º call - hang up - 3º call .. thanks, pn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk extension.conf issue???
Hello All I have the dahdi channels working also I can have a call between the equipments, but when I try to dial a second call I receive the error below: == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/4000-09a44a78' status is 'CHANUNAVAIL' But the problema is that I have a T1 link, it means that I have 22 channels available and 1 in use. Does anyone know how to use the full T1, I am thinking is the extensions.conf file the one is not configured propertly. Please let me know your comments. Best regards Jhon Teleperformance values: Integrity - Respect - Professionalism - Innovation - Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo and Playtones not working on SIP after upgrade
Hello list I had the following echo-test extension on my Asterisk 1.2 setup. exten = 1003,1,Wait(1) exten = 1003,n,Playtones(!1050/1000) exten = 1003,n,Wait(1) exten = 1003,n,StopPlaytones exten = 1003,n,Echo exten = 1003,n,Hangup After migrating my testing server to Asterisk 1.4, and a minor extensions.conf update, everything works just fine. Except for the Playtones and the Echo applications on SIP channels. Over IAX2, both Echo and Playtones works fine on this same extension and system! I googled and tried several things, but nothing seems to work. Basically the log shows it is working, there are no errors or warnings, but there is no sound at all. No beeps, no Echo. Calls, voicemail, moh, and everything else we are using works just fine. We are using Asterisk 1.4.21.2~dfsg-3 (on debian stable), SIP channels with both grandstream and soft phones. Everything on the same network segment. Codec does not seem to affect this behavior (tried them all) Any clues? Thanks! -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help setting IAX variables.
On Monday 07 September 2009 05:55:12 Asterisk User wrote: I am new to Asterisk and want to perform following on my test project. I have 2 Asterisk servers SA and SB and an IAX trunk from SA to SB. Now I set some variables on SB in the same context where an IAX call lands. My question is , is it possible to access these variables in dialplan of SA? If yes then how? I know about IAXVAR application where variables set in source server of IAX channel can be access from destination server... For the variable to be accessed on the destination server, you must explicitly set the IAXVAR variable on the source server. This method does not merely access arbitrary variables on the source server but only variables which have been sent through this mechanism. Just as a version note, you need to be running 1.6.0 or higher to get the IAXVAR mechanism. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo and Playtones not working on SIP after upgrade
On Monday 07 September 2009 09:36:53 Ivan Stepaniuk wrote: I had the following echo-test extension on my Asterisk 1.2 setup. exten = 1003,1,Wait(1) exten = 1003,n,Playtones(!1050/1000) exten = 1003,n,Wait(1) exten = 1003,n,StopPlaytones exten = 1003,n,Echo exten = 1003,n,Hangup After migrating my testing server to Asterisk 1.4, and a minor extensions.conf update, everything works just fine. Except for the Playtones and the Echo applications on SIP channels. Try adding an Answer() in there, before the first Playtones. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SOLVED Re: Echo and Playtones not working on SIP after upgrade
Tilghman Lesher wrote: On Monday 07 September 2009 09:36:53 Ivan Stepaniuk wrote: I had the following echo-test extension on my Asterisk 1.2 setup. exten = 1003,1,Wait(1) exten = 1003,n,Playtones(!1050/1000) exten = 1003,n,Wait(1) exten = 1003,n,StopPlaytones exten = 1003,n,Echo exten = 1003,n,Hangup After migrating my testing server to Asterisk 1.4, and a minor extensions.conf update, everything works just fine. Except for the Playtones and the Echo applications on SIP channels. Try adding an Answer() in there, before the first Playtones. That made the trick, thank you very much. I wonder why does it work on IAX2 channels but not on SIP channels without the Answer command. Anyway, I guess that answering the channel first is the right thing to do. -- Iván Stepaniuk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium hardware support ?
On Mon, 7 Sep 2009 08:48:25 -0500 Juan Cardoza jcard...@tpmex.com wrote: Hello What is your Asterisk problem?, may be I can help you... I had configure a T1 Card TE121 connected with and AVAYA PBX Best regards Hi Juan, thanks for your help. I'm going to choose a 4 ports PRI digium card for this server: http://h10010.www1.hp.com/wwpc/us/en/sm/WF06b/15351-15351-241434-241646-241477-1121586-3638086-3638087.html which specs are here: http://h18004.www1.hp.com/products/quickspecs/12475_na/12475_na.html and I read that the slots are: One 64-bit/133-MHz PCI-X; two 64-bit/100-MHz PCI-X; three x8 PCI Express (x4 speed) so, since the digium PCI-E card is x1, it does not fit in the x8 but, the digium PCI cards, this one: http://www.voipango.com/en/ISDNInterfaces/Digium/BRI-PRI-E1-T1-J1/Digium-TE407PF-T1/E1-PRI-PCI-5-0V-HW-EC.html or this one: http://www.voipango.com/en/ISDNInterfaces/Digium/BRI-PRI-E1-T1-J1/Digium-TE412PF-T1/E1-PRI-PCI-3-3V.html should fit in the PCI-X slot, since PCI-X has backward compatibility toward older PCI. But I still don't understand if the PCI-X slots of the ML350 is 3.3V or 5.0V OR if it has autosense (supports both). Thanks and have a nice day. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium hardware support ?
PCI Express x1 card will work and will fit in the x8 slot PCI-X slots are usually 3.3V Martin On Mon, Sep 7, 2009 at 10:35 AM, mancyb...@gmail.commancyb...@gmail.com wrote: On Mon, 7 Sep 2009 08:48:25 -0500 Juan Cardoza jcard...@tpmex.com wrote: Hello What is your Asterisk problem?, may be I can help you... I had configure a T1 Card TE121 connected with and AVAYA PBX Best regards Hi Juan, thanks for your help. I'm going to choose a 4 ports PRI digium card for this server: http://h10010.www1.hp.com/wwpc/us/en/sm/WF06b/15351-15351-241434-241646-241477-1121586-3638086-3638087.html which specs are here: http://h18004.www1.hp.com/products/quickspecs/12475_na/12475_na.html and I read that the slots are: One 64-bit/133-MHz PCI-X; two 64-bit/100-MHz PCI-X; three x8 PCI Express (x4 speed) so, since the digium PCI-E card is x1, it does not fit in the x8 but, the digium PCI cards, this one: http://www.voipango.com/en/ISDNInterfaces/Digium/BRI-PRI-E1-T1-J1/Digium-TE407PF-T1/E1-PRI-PCI-5-0V-HW-EC.html or this one: http://www.voipango.com/en/ISDNInterfaces/Digium/BRI-PRI-E1-T1-J1/Digium-TE412PF-T1/E1-PRI-PCI-3-3V.html should fit in the PCI-X slot, since PCI-X has backward compatibility toward older PCI. But I still don't understand if the PCI-X slots of the ML350 is 3.3V or 5.0V OR if it has autosense (supports both). Thanks and have a nice day. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] invalid extension
Hello, with Asterisk 1.6.1.6 I try to hangup a call if called extension is not existing. For this purpose I would use the internal i extension but seems not to work. [MyContext] exten = s,1,NoOp(Call is treated as it should) exten = s,n,NoOp(next step) exten = s,n,NoOp(aso ...) exten = _[a-zA-Z].,1,Goto(s,1) ; accept exten LEN 1 alpha exten = _X.,1,Goto(s,1); accept exten LEN 1 numeric exten = i,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = i,n,Hangup ; refused, end of call What I have when calling a one digit extension -in this case h- is: == Using SIP RTP CoS mark 5 [Sep 7 18:51:03] NOTICE[6084]: chan_sip.c:18523 handle_request_invite: Call from '' to extension 'h' rejected because extension not found. == Using SIP RTP CoS mark 5 Should it not go to i extension? If I call the i or s extension it's going well. Am I missing something? -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Aastra phones and Asterisk 1.6.0.14
I am having a strange problem today. When I arrived into the office today two Aastra phones were not working. They can receive calls but not make them. The models are 480i and 480i CT. Other Aastra phones like 55i and 57i work fine, it only seems to affect the older phones. When you try to make a call I can see that Asterisk receives it but nothing else. Even when the phone hangs up I can see the channel is still up: SIP/1003-0a081f48oficina 20011 Down (None) (None)100300:18:21 Juan (None) I cannot get rid of that channel with soft hangup, only restarting Asterisk can do that. Anyone know what may be happening? The phones were working last week. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] All hints say Hold
Today is a strange day. My asterisk server is suddenly saying that all extensions are on hold. All my hints are like this: -= Registered Asterisk Dial Plan Hints =- 4...@hints : SIP/4101 State:HoldWatchers 0 4...@hints : SIP/4100 State:HoldWatchers 0 4...@hints : SIP/4002 State:HoldWatchers 0 4...@hints : SIP/4001 State:HoldWatchers 0 4...@hints : SIP/4000 State:HoldWatchers 0 2...@hints : SIP/2012 State:HoldWatchers 0 2...@hints : SIP/2003 State:HoldWatchers 0 2...@hints : SIP/2002 State:HoldWatchers 0 2...@hints : SIP/2001 State:HoldWatchers 0 1...@hints : SIP/1004 State:HoldWatchers 0 1...@hints : SIP/1003 State:HoldWatchers 0 1...@hints : SIP/1002 State:HoldWatchers 0 I can make and receive calls but BLF on all phones keep blinking like crazy. I was using 1.6.0.14 and upgraded to 1.6.0.15 today but I still have the same problem. Any ideas? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 7, 2009 at 10:09 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Another method to do that is to make the Asterisk monitor output dummy SIP calls rather than sound files. Oreka/Orex can listen to those. Looking for volunteers to test that: http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample This allows recording non-VoIP links, VoIP links where tapping is not convinient, or more selective recording of VoIP calls. Is this similar or the same as the portion of my post that you snipped? Different in many ways, which is why I snipped it. Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. (Actually: recorded calls are sent as RTP streams to the Orex/Oreka server) This records outside of Asterisk. Thus it lacks information available in Asterisk (who really called who). OTOH, it is Asterisk-specific. We actually considered implementing something similar to the Sangoma interface in our driver but realised that doing it in Asterisk would probably be more useful. The overheade seems reasonable. Sorry, I fail to see the difference besides Sangoma implemented it in their Wanpipe drivers and you are attempting copy their idea and do it in Asterisk. Your quote This allows recording non-VoIP links, VoIP links where tapping is not convenient (edited to fix your spelling mistake), or more selective recording of VoIP calls. Isn't that more or less the same thing I said that you snipped, Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. And what if the call does not go through a TDM card? And ore importantly: how can you tell who is the caller and who is the callee? The rtp-tap interface basically tells you that channel X had a call at time Y. I am sure it is pretty trivial to figure out who channel X and Y are based on the channel, time, CID, DID Just a wee bit of code... If it does not go through a TDM card, and is VoIP, then port mirroring works just fine. Sipcallid is a very simple way to match callers to callees. If you control recording through the monitoring interface of Asterisk you can start and stop the recording when you need it. You can also provide better information aobut the call. But again, it means that this is part of Asterisk, and I figure Sangoma has quite a few non-Asterisk customers. Sounds neat, when will it be out of beta? This isn't the biz list, nor the dev list. Snipping out the reference of Sangoma being able to do RTP tap and suggesting people use your experimental dev branch doesn't really help users very much. My message was an explicit call for testers, if you haven't noticed :-) I snip content that is not relevant to my reply. Whoever reads this list already read about the Sangoma interface previously. I had nothing to say about it. It was not related to that new branch. Not everyone who reads the list, reads all the posts, give me a break. It was related to the thread. Your motives and alliances have and always will be for Xorcom and Digium. That is the only reason why you helped me with that BRI install in the US, so you could poke around and try to figure out how Marcin Pycko achieved what you cannot. I may check it out when it is part of a stable backported to 1.4 release, otherwise, I don't run beta in production. Sometimes large sums of money rely on systems, as do much more valuable human lives. -- Senior Systems and Network Administrator Triple Canopy, Inc., 2250 Corporate Park Drive, Suite 300 ph.
Re: [asterisk-users] Aastra phones and Asterisk 1.6.0.14
No # is not used for anything. Asterisk is acting up today so I guess this is not directly related to the phone. All hints say all extensions are on HOLD. On Mon, 2009-09-07 at 13:40 -0400, Michelle Dupuis wrote: Are you using the # symbol in the extension name or to access a feature (eg: outside line)?? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Monday, September 07, 2009 1:12 PM To: Asterisk Users List Subject: [asterisk-users] Aastra phones and Asterisk 1.6.0.14 I am having a strange problem today. When I arrived into the office today two Aastra phones were not working. They can receive calls but not make them. The models are 480i and 480i CT. Other Aastra phones like 55i and 57i work fine, it only seems to affect the older phones. When you try to make a call I can see that Asterisk receives it but nothing else. Even when the phone hangs up I can see the channel is still up: SIP/1003-0a081f48oficina 20011 Down (None) (None)100300:18:21 Juan (None) I cannot get rid of that channel with soft hangup, only restarting Asterisk can do that. Anyone know what may be happening? The phones were working last week. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra phones and Asterisk 1.6.0.14
Are you using the # symbol in the extension name or to access a feature (eg: outside line)?? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Monday, September 07, 2009 1:12 PM To: Asterisk Users List Subject: [asterisk-users] Aastra phones and Asterisk 1.6.0.14 I am having a strange problem today. When I arrived into the office today two Aastra phones were not working. They can receive calls but not make them. The models are 480i and 480i CT. Other Aastra phones like 55i and 57i work fine, it only seems to affect the older phones. When you try to make a call I can see that Asterisk receives it but nothing else. Even when the phone hangs up I can see the channel is still up: SIP/1003-0a081f48oficina 20011 Down (None) (None)100300:18:21 Juan (None) I cannot get rid of that channel with soft hangup, only restarting Asterisk can do that. Anyone know what may be happening? The phones were working last week. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] features.conf : feature map == getting feature to work
Hi there, I need some help with a 'custom' feature. I have following feature defined in features.conf : [applicationmap] opnemencallee = #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m In my dialplan : [from-HostAst] exten = s,1,Set(__DYNAMIC_FEATURES=opnemencallee) exten = s,n,Dial(SIP/grandstream,30) I want the callee to be able to press #3 to be able to record the conversation but when I press these keys on my Grandstream phone, the following is displayed on the CLI : [Sep 7 20:33:49] WARNING[10870]: res_musiconhold.c:665 get_mohbyname: Music on Hold class '/var/samba/profiles/jonaskl/recording' not found Don't know where this comes from... I have tried the same with *3. Same output on the CLI. Yes, I have restarted Asterisk after changes in features.conf. It's not my Grandstream or the DTMF-input because *8 for picking up a ringing phone works well... When I set : opnemencallee = #*3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m and I press #*3, nothing happens... No output on the CLI. There's not much info. I followed the instructions on voip-info.org (which are the same as in features.conf). The module res_features is loaded. Greetingz, Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] invalid extension
Administrator TOOTAI escribió: Hello, with Asterisk 1.6.1.6 I try to hangup a call if called extension is not existing. For this purpose I would use the internal i extension but seems not to work. [MyContext] exten = s,1,NoOp(Call is treated as it should) exten = s,n,NoOp(next step) exten = s,n,NoOp(aso ...) exten = _[a-zA-Z].,1,Goto(s,1) ; accept exten LEN 1 alpha exten = _X.,1,Goto(s,1) ; accept exten LEN 1 numeric exten = i,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = i,n,Hangup; refused, end of call What I have when calling a one digit extension -in this case h- is: == Using SIP RTP CoS mark 5 [Sep 7 18:51:03] NOTICE[6084]: chan_sip.c:18523 handle_request_invite: Call from '' to extension 'h' rejected because extension not found. == Using SIP RTP CoS mark 5 Should it not go to i extension? If I call the i or s extension it's going well. Am I missing something? Hi, The 'i' extension only works in applications like Background(), WaitExten() and everything that uses DTMF to route extensions within a context. As you can see in your call, it won't work directly because asterisk by default will reject a call that doesn't match in the context or included contexts you defined for the user. Because the call is not accepted there's no need for a hangup (in a SIP environment). If you want to explicitly hangup calls using the dialplan, for your case add a one-digit catch all and leave your good calls with a 2-digit minimum: exten = _X,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = _X,n,Hangup exten = _XX.,1,Goto(s,1) ; accept exten LEN 1 numeric That will be enough to hangup what you want to, adjusting it to your needs. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Breaking news, but what happened? 11.000 channels on one server
Olle E. Johansson o...@edvina.net writes: Imaging my surprise this Monday when I installed a plain old Asterisk 1.4 on a new HP server, a DL380 G6, and could run in circles around the old IBM servers. The G6 series is pure magic for everything I've let it touch network-wise. I have three guesses as to why: 1) Lots and lots of bandwidth between CPU and I/O, plus built-in memory controller so any packet copying runs wicked fast. 2) MSI-X seems to really help, at least when combined with modern ethernet chipsets (the original PRO/1000 is looking a bit dated now, but more modern PRO/1000 should still be a good choice). 3) Multi-queue NIC. This should REALLY help when you have lots of cores and CPU threads. Depends on fairly new kernels. I'm not sure which is the answer though. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Breaking news, but what happened? 11.000 channels on one server
On Mon, Sep 7, 2009 at 4:19 PM, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: Olle E. Johansson o...@edvina.net writes: Imaging my surprise this Monday when I installed a plain old Asterisk 1.4 on a new HP server, a DL380 G6, and could run in circles around the old IBM servers. The G6 series is pure magic for everything I've let it touch network-wise. I have three guesses as to why: 1) Lots and lots of bandwidth between CPU and I/O, plus built-in memory controller so any packet copying runs wicked fast. 2) MSI-X seems to really help, at least when combined with modern ethernet chipsets (the original PRO/1000 is looking a bit dated now, but more modern PRO/1000 should still be a good choice). 3) Multi-queue NIC. This should REALLY help when you have lots of cores and CPU threads. Depends on fairly new kernels. I'm not sure which is the answer though. /Benny Packets per second is going to be the eventual bottleneck no matter if it is Asterisk, FreeSwitch, or whatever. Using multiple switches will help but a backplane or interface. can only take so many PPS. http://en.wikipedia.org/wiki/Throughput ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] invalid extension
Miguel Molina a écrit : [...] The 'i' extension only works in applications like Background(), WaitExten() and everything that uses DTMF to route extensions within a context. Well, from reading voip.org it's not really clear than ... [...] Because the call is not accepted there's no need for a hangup (in a SIP environment). Well, I like when logs are clear ;-) and not have to guess :-) If you want to explicitly hangup calls using the dialplan, for your case add a one-digit catch all and leave your good calls with a 2-digit minimum: exten = _X,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = _X,n,Hangup Did it but get 2 hangup! First calling 2...@domain.local == Using SIP RTP CoS mark 5 -- Executing [...@from-guest:1] Goto(SIP/sip.tootai.net-084b1dc8, h,1) in new stack -- Goto (from-guest,h,1) -- Executing [...@from-guest:1] Hangup(SIP/sip.tootai.net-084b1dc8, ) in new stack == Spawn extension (from-guest, h, 1) exited non-zero on 'SIP/sip.tootai.net-084b1dc8' -- Executing [...@from-guest:1] Hangup(SIP/sip.tootai.net-084b1dc8, ) in new stack == Spawn extension (from-guest, h, 1) exited non-zero on 'SIP/sip.tootai.net-084b1dc8' Second calling h...@domain.local == Using SIP RTP CoS mark 5 -- Executing [...@from-guest:1] Hangup(SIP/sip.tootai.net-084c97b8, ) in new stack == Spawn extension (from-guest, h, 1) exited non-zero on 'SIP/sip.tootai.net-084c97b8' -- Executing [...@from-guest:1] Hangup(SIP/sip.tootai.net-084c97b8, ) in new stack == Spawn extension (from-guest, h, 1) exited non-zero on 'SIP/sip.tootai.net-084c97b8' exten = _XX.,1,Goto(s,1) ; accept exten LEN 1 numeric Here your calling a three or more digits ;-) That will be enough to hangup what you want to, adjusting it to your needs. I will leave with this :-) Many thanks for the informations. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf : feature map == getting feature to work
On Monday 07 September 2009 13:40:16 jonas kellens wrote: [applicationmap] opnemencallee = #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m FeatureName = DTMF_sequence,ActivateOn[/ActivatedBy],Application[,AppArguments[,MOH_Class]] it looks like /var/samba/profiles/jonaskl/recording is in the spot for [,MOH_Class] -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Older Aastra phones and Asterisk 1.6
It seems that older Aastra phones (9112i, 9133i, 480i, 480i CT) have a problem with the new SIP implementation in Asterisk 1.6.X that makes them unable to dial. They can receive calls but when you attempt to dial the phone remains silent. You can see in core show channels that the first channel is active and it is impossible to kill it without restarting Asterisk. The solution I found for this is to set session-timers=refuse in sip.conf and now I am able to send calls. I suppose this is a problem with the firmware of those phones as newer versions of Aastra phones (5Xi) work without the modification. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Another method to do that is to make the Asterisk monitor output dummy SIP calls rather than sound files. Oreka/Orex can listen to those. Looking for volunteers to test that: http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample This allows recording non-VoIP links, VoIP links where tapping is not convinient, or more selective recording of VoIP calls. Is this similar or the same as the portion of my post that you snipped? Different in many ways, which is why I snipped it. Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. (Actually: recorded calls are sent as RTP streams to the Orex/Oreka server) This records outside of Asterisk. Thus it lacks information available in Asterisk (who really called who). OTOH, it is Asterisk-specific. We actually considered implementing something similar to the Sangoma interface in our driver but realised that doing it in Asterisk would probably be more useful. The overheade seems reasonable. Sorry, I fail to see the difference besides Sangoma implemented it in their Wanpipe drivers and you are attempting copy their idea and do it in Asterisk. Your quote This allows recording non-VoIP links, VoIP links where tapping is not convenient (edited to fix your spelling mistake), or more selective recording of VoIP calls. Isn't that more or less the same thing I said that you snipped, Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. And what if the call does not go through a TDM card? And ore importantly: how can you tell who is the caller and who is the callee? The rtp-tap interface basically tells you that channel X had a call at time Y. If you control recording through the monitoring interface of Asterisk you can start and stop the recording when you need it. You can also provide better information aobut the call. But again, it means that this is part of Asterisk, and I figure Sangoma has quite a few non-Asterisk customers. This isn't the biz list, nor the dev list. Snipping out the reference of Sangoma being able to do RTP tap and suggesting people use your experimental dev branch doesn't really help users very much. My message was an explicit call for testers, if you haven't noticed :-) I snip content that is not relevant to my reply. Whoever reads this list already read about the Sangoma interface previously. I had nothing to say about it. It was not related to that new branch. -- 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 I imagine this setup will need those two communicating entities to be part of the pabx. But support extension 100 of PABX A (legacy) calls 101 on the same platform. I want asterisk connected to PABX A via E1/T1 to know about that call and start recording (tap) without bridging or being part of that conversation ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is not yet available ODBC support for queue_log in asterisk 1.6?
Hi list, I hope someone could help me. I've started using Asterisk 1.6.0.14 to get queue logs in real time with odbc (our databases are all PostgreSQL) but it's not working. However, cdr odbc is working well. When asterisk starts next message appears: WARNING[4217] config.c: Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available My configuration follows: /etc/asterisk/ extconfig.conf [settings] queue_log = odbc,postgres,asterisk.queue_log /etc/asterisk/res_odbc.conf [qlogrt] enabled = yes dsn = qlogrt ... /etc/odbc.ini [qlogrt] Driver = PostgreSQL Database = postgres Servername = 192.168.X.X UserName = user Password = password Port = 5432 ... /etc/asterisk/cdr_odbc.conf [global] dsn=qlogrt table=asterisk.ast_cdr ... Regards, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
I imagine this setup will need those two communicating entities to be part of the pabx. But support extension 100 of PABX A (legacy) calls 101 on the same platform. I want asterisk connected to PABX A via E1/T1 to know about that call and start recording (tap) without bridging or being part of that conversation Hi, Asterisk won't work as a recording server if the call doesn't go through it. In the IP world it means that both media (RTP) and signalling must pass through asterisk, and in the E1/T1 digital or analog world it means that the call must be bridged through asterisk. A simple dialplan would explain it: exten = s,1,Answer() ;Asterisk receives the call, from the lecagy PBX or from the external link (this should be two different contexts) exten = s,n,MixMonitor(blah) ; Records the conversation, exten = s,n,Dial(Tech/peer/number...,30,rtTwhatever) ; and sends the call back to the legacy PBX or to an external link If you want to record 100% calls, you would have to route every call through asterisk, even internal PBX calls. Even if you want to tap your legacy PBX to a non-asterisk recording server like the ones suggested before in this thread, the calls must go through a link to make tapping possible and you should seek an alternate solution to the internal calls within your legacy PBX. The beauty of asterisk and open source IP-PBXs relies on the native recording capabilities which makes things really easy. When you see that asterisk works and that can do the recordings and much more, you would start thinking on making asterisk your main PBX solution and leaving that legacy PBX for minimal uses. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE420P configuration
How we will the channels will be handled. I want to test it for loop back . On Mon, Sep 7, 2009 at 6:57 PM, ABBAS SHAKEEL shakeel.abbas@gmail.comwrote: http://www.voip-info.org/wiki/view/Digium+TE420P http://www.voip-info.org/wiki/view/Digium+TE420PThe best resource is also quite On Mon, Sep 7, 2009 at 3:14 PM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello I am trying to configure TE420P but i am confused what to give chan_dahdi :( Below is configuration i am using for TDM400P Please help what changes to make in it... Please provide a link as well [trunkgroups] [channels] ;default for channels switchtype=national rxwink=300 hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 busydetect=yes busycount=3 ;busypattern=500,500 ; DEFINING CHANNEL context = incoming_context_for_ptcl signalling=fxs_ks channel= 1 -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi/DTMF problem
On Mon, 2009-09-07 at 07:20 -0600, Greg Woods wrote: incoming calls through the FXO line are dropped as soon as there is a button press. The error logged is: [Aug 23 18:15:39] WARNING[6532] chan_dahdi.c: Cannot handle frames in 2 format [Aug 23 18:15:39] WARNING[6532] file.c: Failed to write frame Which looks like this bug: https://issues.asterisk.org/view.php?id=15129 I didn't solve this, but I worked around it. I eventually gave up and installed the asterisk14 1.4.26 packages from ATrpms. This version I was able to get working with Dahdi. I'll keep my eye on the bug report to see if this ever gets fixed, then I might try to upgrade to 1.6. But I have no urgent need to do so, so I am happy to wait a while and at least I can finally retire the old system. --Greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk extension.conf issue???
On 8/09/09 2:21 AM, Juan Cardoza wrote: Hello All I have the dahdi channels working also I can have a call between the equipments, but when I try to dial a second call I receive the error below: == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/4000-09a44a78' status is 'CHANUNAVAIL' I'd say you're trying to do Dial(DAHDI/1/12345) instead of Dial(DAHDI/g1/12345) I.E. you're trying to use an individual channel rather than a group. Without more information it's just a guess :) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All hints say Hold
On 8/09/09 5:35 AM, Carlos Chavez wrote: Today is a strange day. My asterisk server is suddenly saying that all extensions are on hold. All my hints are like this: -= Registered Asterisk Dial Plan Hints =- 4...@hints : SIP/4101 State:HoldWatchers 0 4...@hints : SIP/4100 State:HoldWatchers 0 4...@hints : SIP/4002 State:HoldWatchers 0 4...@hints : SIP/4001 State:HoldWatchers 0 4...@hints : SIP/4000 State:HoldWatchers 0 2...@hints : SIP/2012 State:HoldWatchers 0 2...@hints : SIP/2003 State:HoldWatchers 0 2...@hints : SIP/2002 State:HoldWatchers 0 2...@hints : SIP/2001 State:HoldWatchers 0 1...@hints : SIP/1004 State:HoldWatchers 0 1...@hints : SIP/1003 State:HoldWatchers 0 1...@hints : SIP/1002 State:HoldWatchers 0 Reload SIP/Restart Phones/Make a change to SIP/Restart Asterisk :) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 7, 2009 at 12:29 AM, Steve Totarostot...@first-notification.com wrote: Did you push it past 300 on two year old hardware and software? old hardware yes. old software no. The servers are more than 3 years old Core 2 Duo Dell Dimension desktop as proof of concept? are core 2 duo's really two years old already? I guess so. I don't really follow the latest hardware news. I have my lab on server-class gear. Port mirroring is basic on almost any newer switch. Login, enable port monitoring, write mem, done. Port mirroring is basic on quality networking gear. I know perfectly well how it works. My point was that replicating ALL traffic on a LAN port seemed a bit like hauling out all the corn plants from the corn field when what you really wanted was just the corn kernels from the ears. That's what I mean by heavy-handed. I've never used the software you've proposed. I realize that replicating all traffic for a port, or in my case, all traffic for a bonded interface is not difficult logically, and is quick to configure. I think it is aesthetically displeasing compared to grabbing the recordings at the place where the calls are already taking place. Personal taste. You're allowed your opinion too, which you've clearly stated. I build robust and redundant systems, separate server for DB, recording, gateways, in an all HA configuration. Me too. Again, taste. Again, how many calls were you able record using RAMdisk? Anywhere 300? As I stated before, this is going to be dependent on how you're manipulating the calls and the gear you're running on. The nice thing about your 'just broadcast the entire LAN to the recording solution' is that the recording service just gets to throw away everything that's not an audio channel, and it doesn't have to do squat to the call. If it COULDN'T do a lot of recordings under these circumstances it wouldn't be worth any money. I don't think I've pushed my solution past 90 simultaneous recordings of MeetMe() mixing, with more than 100 AGI channels running, with assorted ChanSpy() jobs. Bookmark my post, so when you reach your RAMDisk limit, you can join the big league. Anything I do as a scaling solution will be price versus performance. So since we're talking about a commercial solution to replace something that asterisk does, I'll have to find out what your commercial solution costs per channel, and compare that against the cost of cloning out an identical server. My solution scales to parallel servers just fine. Is OrecX really $199 per recorded channel? So that 300 channels you're talking about costs $60,000? So I can buy six $10,000 servers, each of which can run circles around my current solution, and still break even. I like my solution better. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is not yet available ODBC support for queue_log in asterisk 1.6?
On Monday 07 September 2009 17:16:12 Daniel - Asterisk wrote: WARNING[4217] config.c: Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available My configuration follows: /etc/asterisk/ extconfig.conf [settings] queue_log = odbc,postgres,asterisk.queue_log I don't see [postgres] anywhere in your res_odbc.conf. /etc/asterisk/res_odbc.conf [qlogrt] enabled = yes dsn = qlogrt ... -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
Sorry , I checked on my Asterisk pbx and there is no sip.cfg file on it . Can you please let me know how can I make my Asterisk Call Parking as functional ? On Tue, Sep 1, 2009 at 6:23 AM, Lee, John (Sydney) john@compuware.comwrote: Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Yes, I am referring to the Polycom sip.cfg and not the sip.cfg in Asterisk. Somethere down in sip.cfg, there is a line that looks like this: digitmap dialplan.digitmap=#700| ... Basically, Polycom will scan your input to see when it will pass all the keystrokes to Asterisk. In above, if it detects that you have entered #700, it will automatically send it to Asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 line simulation for Asterisk
Hello I have the loop back connector and TE420P card but i dont know how to configure that. Please let me know of any help. I am facing the problem in configuration of channels. i have make changes in chan_dahdi [r...@te420 etc]# dahdi_hardware pci::04:08.0 wct4xxp+ d161:0420 Wildcard TE420 (4th Gen) shows this. this means card is configured. Now i have to do configuration in chan_dahdi.conf or some other files . Please some one shed some light on it. I have asked this question in a different topic as well -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help setting IAX variables.
Thanks Tilghman for your quick reply. I know that we should set variables through IAXVAR on source server to access them on Destination server. I just wanted to know the reverse case, where IAX channel variables set on destination server are accessible on Source server or not. Thanks again for your inputs. --- Asterisk user ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Asterisk sound files
Dear All Can you please do me favor and let me know why my converted sound files are not being played and heared on my Asterisk ? Please find attached my sound files . Actually , I had them recorded as *.wav files and I tried to convert them to *.gsm as the followings : #sox FR3.wav FR3.gsm Then I put my special announcements under /var/lib/asterisk/sounds but these converted announcement files cannot be heared on my Asterisk . Can you please let me know what is the problem ? Regards H.Motamedi FR1.gsm Description: Binary data FR3.gsm Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] confBridge in Asterisk 1.6.2.0-rc1 doesn't stable
confBridge in Asterisk 1.6.2.0-rc1 doesn't stable. It causes segment fault very often and results in asterisk crash. Ian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk sound files
Do you have an error message? On Tue, Sep 8, 2009 at 1:29 PM, hadi motamedi motamed...@gmail.com wrote: Dear All Can you please do me favor and let me know why my converted sound files are not being played and heared on my Asterisk ? Please find attached my sound files . Actually , I had them recorded as *.wav files and I tried to convert them to *.gsm as the followings : #sox FR3.wav FR3.gsm Then I put my special announcements under /var/lib/asterisk/sounds but these converted announcement files cannot be heared on my Asterisk . Can you please let me know what is the problem ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
