Re: [asterisk-users] asterisk-1.8 problem with one-way audio with no nat
Matt Riddell li...@venturevoip.com wrote: On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote: Hi. I have a soft phone -- expresstalk-- on a computer in my network and I use the internal ip address of the asterisk box to register the phone. But using asterisk-1.8 between revisions 281912 and 281982 it breaks -- after a few seconds of the call, I lose audio from the asterisk box to my soft phone, but not the other way around. This looks like one commit, but obviously I would like to know what's going on here? What's in the commit? Its the 282911 commit seems to break audio to the soft phone, but not to my ata -- very strange. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?
Hi Everyone, I have two servers as the following that are trunked with each other via IAX2 trunk: Server A: Asterisk 1.4.21.2 (Elastix Flavor) Server B (IP # 72.72.72.72): Asterisk 1.6.2.0 (Vanilla) Server B can place calls to Server A but when trying to place calls from Server A to Server B this is what I am getting: pbx*CLI originate iax2/mel/14161234567 extension s...@null Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3ms SCall: 16389 DCall: 0 [72.72.72.72:4569] VERSION : 2 CALLED NUMBER : 14161234567 CODEC_PREFS : (gsm) CALLING NUMBER : CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: LANGUAGE: en USERNAME: mel FORMAT : 2 CAPABILITY : 57346 ADSICPE : 2 DATE TIME : 2010-09-02 02:14:50 Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 3ms SCall: 1 DCall: 16389 [72.72.72.72:4569] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 16389 DCall: 1 [72.72.72.72:4569] -- Hungup 'IAX2/mel-16389' As you can see above, Subclass: REJECT comes back wtih no cause code. Usually there is a cause code to debug but in this case there is no Cause code. Trunks on both sides in the context=from-internal so it's not an Inbound Route issue. Any pointers are much appreciated. -Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?
On Thu, Sep 02, 2010 at 02:21:11AM -0400, bruce bruce wrote: Hi Everyone, I have two servers as the following that are trunked with each other via IAX2 trunk: Server A: Asterisk 1.4.21.2 (Elastix Flavor) Any chance you could upgrade that? Elastix has newer versions of Asterisk, for starters. Server B (IP # 72.72.72.72): Asterisk 1.6.2.0 (Vanilla) Is it configured to talk to old IAX2 peers? http://downloads.asterisk.org/pub/security/IAX2-security.html http://downloads.asterisk.org/pub/security/AST-2009-006.html -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?
I'd rather find the problem than upgrade blindly. Upgrading not always solves the problem and has the potential to break other things. Thanks for the offer though. Bug # 16753 applies. call token not required was set in the trunk and problem solved. There is not warning for this in iax2 debug but there is in core set debug. That's a petty. -Bruce On Thu, Sep 2, 2010 at 3:19 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Thu, Sep 02, 2010 at 02:21:11AM -0400, bruce bruce wrote: Hi Everyone, I have two servers as the following that are trunked with each other via IAX2 trunk: Server A: Asterisk 1.4.21.2 (Elastix Flavor) Any chance you could upgrade that? Elastix has newer versions of Asterisk, for starters. Server B (IP # 72.72.72.72): Asterisk 1.6.2.0 (Vanilla) Is it configured to talk to old IAX2 peers? http://downloads.asterisk.org/pub/security/IAX2-security.html http://downloads.asterisk.org/pub/security/AST-2009-006.html -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?
Maybe dvossel can re-open issue # 16753 and fix the warning to show on iax2 debug as well along with core set debug like all other warnings. That way it's straight forward. That ticket shouldn't have been closed without a fix. On Thu, Sep 2, 2010 at 4:11 AM, bruce bruce bruceb...@gmail.com wrote: I'd rather find the problem than upgrade blindly. Upgrading not always solves the problem and has the potential to break other things. Thanks for the offer though. Bug # 16753 applies. call token not required was set in the trunk and problem solved. There is not warning for this in iax2 debug but there is in core set debug. That's a petty. -Bruce On Thu, Sep 2, 2010 at 3:19 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Thu, Sep 02, 2010 at 02:21:11AM -0400, bruce bruce wrote: Hi Everyone, I have two servers as the following that are trunked with each other via IAX2 trunk: Server A: Asterisk 1.4.21.2 (Elastix Flavor) Any chance you could upgrade that? Elastix has newer versions of Asterisk, for starters. Server B (IP # 72.72.72.72): Asterisk 1.6.2.0 (Vanilla) Is it configured to talk to old IAX2 peers? http://downloads.asterisk.org/pub/security/IAX2-security.html http://downloads.asterisk.org/pub/security/AST-2009-006.html -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] agi playback to execute say.conf settings
Hi all, I am using asterisk-1.6.2.10. I changed say.conf script for customized number reading. In the extension.conf: -- [number-to-voice] exten = 8765,1,playback(num:344345,say) exten = 8765,n,hangup It executes corresponding say.conf script and produces good results for me. but when I write it in agi does not working. Here is agi debug output from asterisk. SIP/6000-000aAGI Rx EXEC playback num:333456,say -- AGI Script Executing Application: (playback) Options: (num:333456,say) SIP/6000-000aAGI Tx 200 result=0 Anybody have any ideas to work it out in agi playback ? Thanks, Ashik -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi issue on sangoma A200
Hi max, Please read India-CID.txt file in asterisk documentation directory. Thanks, Ashik On Wed, Aug 11, 2010 at 10:35 AM, Max Alex max.aster...@gmail.com wrote: Hi, Thanks for this information, but it is not working for both the issues, I have tried with the configuration with cidsignalling, cidstart etc.. Can any one provide more help for this. Thanks, Max Alex Voip Developer On Mon, Aug 9, 2010 at 5:31 PM, asteriskguru asteriskguru beaasteriskg...@gmail.com wrote: Hi max, Have look on my blog regarding this. http://ashikalim.blogspot.com/2010/08/config-asterisk-india-pstn-lines_09.html Thanks, Ashik On Sat, Aug 7, 2010 at 11:15 AM, Max Alex max.aster...@gmail.com wrote: Hi All, I have Sangoma A200 Card installed on my system, I have centos 5.5 with 64 bit, Here are the description for asterisk and dahdi. Asterisk 1.6..2.9 Dahdi: 2.3.0.1 I have two issues with dahdi 1) I am not getting full callerid on my phones from sangoma card to asterisk users. if i am connecting analog phone directly then i am getting callerid properly. I am in india and using Airtel Connection, I have set variables in chan_dahdi.conf as well for callerid but the not getting full digits in callerid, it is coming with 8 digits only. 2) Another issue is when I am hanging up the phone from inbound or outbound from the dahdi channel, it takes 5-6 seconds to dropping the call. Here are the confguration file for chan_dahdi.conf - ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2010-07-30 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=default usecallerid=yes callerid=asreceived hanguponpolarityswitch=yes answeronpolarityswitch=yes ;cidstart=ring cidstart=polarity_IN ;cidsignalling=dtmf cidsignalling=dtmf hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no useincomingcalleridondahditransfer=yes ;callerid=asreceived ;Sangoma AFT-A200 [slot:4 bus:2 span:1] wanpipe1 context=from-internal group=1 echocancel=yes callerid=asreceived signalling = fxo_ks channel = 1 context=from-internal group=1 echocancel=yes callerid=asreceived signalling = fxo_ks channel = 2 context=from-zaptel group=0 echocancel=yes callerid=asreceived signalling = fxs_ks channel = 3 context=from-zaptel group=0 echocancel=yes callerid=asreceived signalling = fxs_ks channel = 4 --- Please hemp me for this issues. Thanks, Max Alex Voip Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Recording Questions
Hi, 1) I want to create add *1 call recording and wanted to know whether the file is created during recording or only after? I want to syncronise the recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I've added wW to the Dial() command, and also Set(DYNAMIC_FEATURES=automon) (don't know if the last one is necessary). The line in features.conf says automon = *1 and I restarted asterisk once the changes were made. Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording Questions
1) I want to create add *1 call recording and wanted to know whether the file is created during recording or only after? I want to syncronise the recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I've added wW to the Dial() command, and also Set(DYNAMIC_FEATURES=automon) (don't know if the last one is necessary). The line in features.conf says automon = *1 and I restarted asterisk once the changes were made. Sorry, just re-read my email and realised I didn't ask any questions and it sounded quite rude. Basically, I'm trying to allow one of my clients to record calls and download them onto their PC. I'm thinking of creating a web interface for this, which is where my first question comes in. However, I can't seem to get it working. I think it's something to do with inband and rfc2833 but when I change it, the menu systems seem to stop working. Can anyone assist? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording Questions
1) The file is written in real time. Personally I would add a dialplan entry into the 'h' extension to move the recording into a different directory when the call ends. That will make your syncronisation much easier. Dan Journo wrote: Hi, 1) I want to create add *1 call recording and wanted to know whether the file is created during recording or only after? I want to syncronise the recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I’ve added wW to the Dial() command, and also Set(DYNAMIC_FEATURES=automon) (don’t know if the last one is necessary). The line in features.conf says automon = *1 and I restarted asterisk once the changes were made. Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording Questions
The DTMF mode can cause problems. The main rule is to make sure everything is using the same method. I normally use SIP-Info as the method as it allows to rtp stream to be switch directly between the two end points but asterisk still sees all the dtmf digits. Dan Journo wrote: 1) I want to create add *1 call recording and wanted to know whether the file is created during recording or only after? I want to syncronise the recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I've added wW to the Dial() command, and also Set(DYNAMIC_FEATURES=automon) (don't know if the last one is necessary). The line in features.conf says automon = *1 and I restarted asterisk once the changes were made. Sorry, just re-read my email and realised I didn't ask any questions and it sounded quite rude. Basically, I'm trying to allow one of my clients to record calls and download them onto their PC. I'm thinking of creating a web interface for this, which is where my first question comes in. However, I can't seem to get it working. I think it's something to do with inband and rfc2833 but when I change it, the menu systems seem to stop working. Can anyone assist? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording Questions
1) I use a bash script I wrote to check if call recordings are being written to and if not then move them. I move them to a locally mounted NFS share but this will work with any type of locally mounted share (Samba for Windows). I run the script every minute with cron. It also sorts the recordings in directories based on date. If you just want to sync files rather than move, just change the mv commands to cp commands. Script attached. On 09/02/2010 12:27 PM, Gareth Blades wrote: The DTMF mode can cause problems. The main rule is to make sure everything is using the same method. I normally use SIP-Info as the method as it allows to rtp stream to be switch directly between the two end points but asterisk still sees all the dtmf digits. Dan Journo wrote: 1) I want to create add *1 call recording and wanted to know whether the file is created during recording or only after? I want to syncronise the recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I've added wW to the Dial() command, and also Set(DYNAMIC_FEATURES=automon) (don't know if the last one is necessary). The line in features.conf says automon = *1 and I restarted asterisk once the changes were made. Sorry, just re-read my email and realised I didn't ask any questions and it sounded quite rude. Basically, I'm trying to allow one of my clients to record calls and download them onto their PC. I'm thinking of creating a web interface for this, which is where my first question comes in. However, I can't seem to get it working. I think it's something to do with inband and rfc2833 but when I change it, the menu systems seem to stop working. Can anyone assist? Thanks Dan p style=margin: 0; padding: 0; border-collapse: collapse; font-family: Tahoma, Arial, Sans-Serif; font-size: 10px; color: #33; ---DISCLAIMERbr / The information contained in this message is private and confidential and intended only for the recipient named above. If you are not the intended recipient you are notified that any communication, circulation or copying of the information contained in this message is strictly prohibited. If you have received this message in error please notify us immediately by telephone in order that we are made aware of this fact and the message can be returned to us at our address as indicated above. Activity and use of the Sheffield City Taxis e-mail service is monitored to secure its effective operation and for other lawful business purposes. Sheffield City Taxis Ltd. Registered Office: 912 City Road, Sheffield, S2 1GQ. Registered in England no: 4674148. Sheffield City Taxis Limited uses regularly updated anti-virus software in an attempt to reduce the possibility of infection. However we do not guarantee that any attachments to this e-mail are virus free./p MoveCallRecs.sh Description: Bourne shell script -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording Questions
On Thu, 2010-09-02 at 07:21 -0400, Dan Journo wrote: 1) I want to create add *1 call recording and wanted to know whether the file is created during recording or only after? I want to syncronise the recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I've added wW to the Dial() command, and also Set(DYNAMIC_FEATURES=automon) (don't know if the last one is necessary). The line in features.conf says automon = *1 and I restarted asterisk once the changes were made. Sorry, just re-read my email and realised I didn't ask any questions and it sounded quite rude. Basically, I'm trying to allow one of my clients to record calls and download them onto their PC. I'm thinking of creating a web interface for this, which is where my first question comes in. However, I can't seem to get it working. I think it's something to do with inband and rfc2833 but when I change it, the menu systems seem to stop working. Can anyone assist? Thanks Dan We've mounted a separate storage device onto both the web server and asterisk server. The recorded calls are saved directly onto the storage device and the web server can read off it directly too. This has the added advantage of allowing the web server to create sub directories on the monitor directory if you have more than one client using the same asterisk server -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording Questions
How do you sort out the issue of having 2 wav files per call? Also, when I press *1, asterisk thinks that both the caller and the callee have pressed *1 and therefore it starts recording twice (therefore making 4 wav files). Any idea what's going on there? Heres the CLI output:- -- Called 01615556...@supplier -- SIP/supplier-0055 is making progress passing it to SIP/clientone_201-0054 -- SIP/supplier-0055 answered SIP/clientone_201-0054 -- SIP/kesher_201-0054 Playing 'beep' (language 'en') -- User hit '*1' to record call. filename: wav|auto-1283429941-SIP-clientone_201-0054-01615556607|m -- SIP/supplier-0055 Playing 'beep' (language 'en') -- User hit '*1' to record call. filename: wav|auto-1283429941-01615556607-SIP-clientone_201-0054|m Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail - disable * 0 and #
Hi all Been looking to find a way to stop the dtmf keys * 0 and # managing call flow in the dialplan - I just want VM to stop recording on silence or hangup. I know I can trap the exit and loop back around but just want to ignore the keys totally. Any suggestions P -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail - disable * 0 and #
oops - forgot to say this is voicemail() on Version 1.4.33.1 P _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy Grice Sent: 02 September 2010 13:32 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Voicemail - disable * 0 and # Hi all Been looking to find a way to stop the dtmf keys * 0 and # managing call flow in the dialplan - I just want VM to stop recording on silence or hangup. I know I can trap the exit and loop back around but just want to ignore the keys totally. Any suggestions P -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording Questions
On Thu, 2010-09-02 at 08:20 -0400, Dan Journo wrote:
How do you sort out the issue of having 2 wav files per call?
Also, when I press *1, asterisk thinks that both the caller and the callee
have pressed *1 and therefore it starts recording twice (therefore making 4
wav files). Any idea what's going on there?
Heres the CLI output:-
-- Called 01615556...@supplier
-- SIP/supplier-0055 is making progress passing it to
SIP/clientone_201-0054
-- SIP/supplier-0055 answered SIP/clientone_201-0054
-- SIP/kesher_201-0054 Playing 'beep' (language 'en')
-- User hit '*1' to record call. filename:
wav|auto-1283429941-SIP-clientone_201-0054-01615556607|m
-- SIP/supplier-0055 Playing 'beep' (language 'en')
-- User hit '*1' to record call. filename:
wav|auto-1283429941-01615556607-SIP-clientone_201-0054|m
Thanks
Dan
Sounds like it's using Monitor rather than MixMonitor.
I had a quick look at this:
http://www.voip-info.org/wiki/view/Asterisk+config+features.conf
And it looks like you might be better off creating your own macro for
one touch recording and adding it to the features.conf as shown in this
part of that web page
Examples
One Touch Recording (applicationmap) with WAV to MP3 Conversion Macro.
extensions.conf :
[macro-apprecord]
exten = s,1,GotoIf($[${XAD} = 0 | ${XAD} = ]?startrec:stoprec)
exten = s,n(startrec),Playback(startmonitor)
exten = s,n,Set(XAD=1)
exten = s,n,Set(FILENAME=${TIMESTAMP}-OUT
${CALLERID(number)}-^-${UNIQUEID})
exten = s,n,Set(MONITOR_EXEC_ARGS= nice -n 19 /usr/local/bin/lame -b
96 -t -F -m m --bitwidth 16 --quiet
/var/spool/asterisk/monitor/${FILENAME}.wav
/var/spool/asterisk/monitor/${FILENAME}.mp3 rm -f
/var/spool/asterisk/monitor/${FILENAME}.wav)
exten = s,n,Monitor(wav,${FILENAME},m)
exten = s,n,MacroExit
exten = s,n(stoprec),StopMonitor
exten = s,n,Set(XAD=0)
exten = s,n,Playback(stopmonitor)
exten = s,n,MacroExit
features.conf :
apps = *9,caller,Macro,apprecord
but using MixMonitor rather than monitor.
Let me know how you got on with it as I think I'm going to be asked to
do this in the next month or 2.
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] Voicemail - disable * 0 and #
On Thu, Sep 2, 2010 at 8:31 AM, Paddy Grice pa...@wizaner.com wrote: Any suggestions You have to modify the source code. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 calls getting rejected without aCAUSE CODE. How to debug this?
1.4 will talk to 1.6 but results haven't been as good in my experience as native 1.4-1.4 or 1.6-1.6 communication. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] agi playback to execute say.conf settings
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asteriskguru asteriskguru Subject: [asterisk-users] agi playback to execute say.conf settings Hi all, I am using asterisk-1.6.2.10. I changed say.conf script for customized number reading. snip but when I write it in agi does not working. Here is agi debug output from asterisk. SIP/6000-000aAGI Rx EXEC playback num:333456,say -- AGI Script Executing Application: (playback) Options: (num:333456,say) SIP/6000-000aAGI Tx 200 result=0 Anybody have any ideas to work it out in agi playback ? Replace playback num:334456,say with say number 334456 Refer to http://www.voip-info.org/wiki/view/say+number -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording Questions
On 09/02/2010 01:09 PM, Ishfaq Malik wrote: On Thu, 2010-09-02 at 07:21 -0400, Dan Journo wrote: 1) I want to create add *1 call recording and wanted to know whether the file is created during recording or only after? I want to syncronise the recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I've added wW to the Dial() command, and also Set(DYNAMIC_FEATURES=automon) (don't know if the last one is necessary). The line in features.conf says automon = *1 and I restarted asterisk once the changes were made. Sorry, just re-read my email and realised I didn't ask any questions and it sounded quite rude. Basically, I'm trying to allow one of my clients to record calls and download them onto their PC. I'm thinking of creating a web interface for this, which is where my first question comes in. However, I can't seem to get it working. I think it's something to do with inband and rfc2833 but when I change it, the menu systems seem to stop working. Can anyone assist? Thanks Dan We've mounted a separate storage device onto both the web server and asterisk server. The recorded calls are saved directly onto the storage device and the web server can read off it directly too. This has the added advantage of allowing the web server to create sub directories on the monitor directory if you have more than one client using the same asterisk server Beware that if you have lots of concurrent calls writing all these simultaneously to disk can be heavy on the disk I/O load. I have our recordings written to a solid state drive rather than straight to storage disks then moved to long term storage to avoid this problem. You could use a ram disk if you have enough memory, this is probably cheaper. Also if the share you're writing directly to goes down call recordings will stop being written, where as if you try and copy/move them after they're finished then the cp/mv will just fail but your recordings will still be written locally and stored up until the share is available again. And yes, sounds like its using Monitor() and not MixMonitor(). p style=margin: 0; padding: 0; border-collapse: collapse; font-family: Tahoma, Arial, Sans-Serif; font-size: 10px; color: #33; ---DISCLAIMERbr / The information contained in this message is private and confidential and intended only for the recipient named above. If you are not the intended recipient you are notified that any communication, circulation or copying of the information contained in this message is strictly prohibited. If you have received this message in error please notify us immediately by telephone in order that we are made aware of this fact and the message can be returned to us at our address as indicated above. Activity and use of the Sheffield City Taxis e-mail service is monitored to secure its effective operation and for other lawful business purposes. Sheffield City Taxis Ltd. Registered Office: 912 City Road, Sheffield, S2 1GQ. Registered in England no: 4674148. Sheffield City Taxis Limited uses regularly updated anti-virus software in an attempt to reduce the possibility of infection. However we do not guarantee that any attachments to this e-mail are virus free./p -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording Questions
I have our recordings written to a solid state drive rather than straight to storage disks then moved to long term storage to avoid this problem. Not an option for me at the moment. I'm running Asterisk on a cloud to reduce startup costs. Once I reach around 1,000 extensions, I'll move over the physical servers. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-1.8 problem with one-way audio with no nat
On Thursday 02 September 2010 01:13:35 cov...@ccs.covici.com wrote: Matt Riddell li...@venturevoip.com wrote: On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote: Hi. I have a soft phone -- expresstalk-- on a computer in my network and I use the internal ip address of the asterisk box to register the phone. But using asterisk-1.8 between revisions 281912 and 281982 it breaks -- after a few seconds of the call, I lose audio from the asterisk box to my soft phone, but not the other way around. This looks like one commit, but obviously I would like to know what's going on here? What's in the commit? Its the 282911 commit seems to break audio to the soft phone, but not to my ata -- very strange. That doesn't make any sense. Revision 282911 is a merge to a team branch, nothing related to the 1.8 branch. Maybe 282891 (same change, but to the 1.8 branch)? Or did you fat finger the revision? -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH in the middle of the call
Hi, I have the same problem too, but i can´t provide more information, because i can´t find more information, just in the log show me that´s it´s starting a MOH normally. It´s happen on random way, without nothing similar on each call. Using Elastix 1.6, x64 with ATA LinkSys PAP2. Em 01/09/2010 16:57, Stefan Schmidt escreveu: Danny Nicholas schrieb: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dario Quiroz *Subject:* [asterisk-users] MOH in the middle of the call Hi, I have a very strange problem. In the middle of the call the MOH starts for 30 seconds approximately. After this the call run normally. Anybody have an ideia or has some similar problem? Thanks in advance!! You haven’t provided enough information. Guesses would be that it is a normal thing or that you are getting some kind of perhaps SIP error that is causing a momentary disconnect, triggering MOH until the condition resolves itself. i had some problems like this, but only when a snom phone transfered a call. if you use asterisk 1.6.x this could also be an answer bug which is allready been fixed. this bug cause some strange issues with moh and wrong codec write formats. but without further information its just a random guess ;) best regards steve smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google Voice-like feature.
I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. Asterisk 1.4 -- though I could probably upgrade. Suggestions on how to make this happen? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice-like feature.
Ken D'Ambrosio wrote: I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. Asterisk 1.4 -- though I could probably upgrade. Suggestions on how to make this happen? Thanks! -Ken You can get them to acknowledge by executing a macro when the call is connected using the M parameter in the dial command. However if the mobile was answered and the confirmation not entered you would have to flag that destination as being dead and then jump back to the dial command again and omit that destination for the next attempt. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice-like feature.
On Thursday 02 Sep 2010, Ken D'Ambrosio wrote: I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. The problem is that, if one of the destination phones is diverting to voicemail, you won't know it's voicemail until it's answered -- by which time it's already too late. The best you could hope to do is: park the incoming call; ring all the handsets at once; and when each one answers, play a recorded message giving the number to pick up the parked call. If any of them successfully picks up the parked call, then of course you need to abort the Dial() to the other ones. If no-one picks up the parked call within a reasonable timeframe, it can be sent to Asterisk's own voicemail. -- AJS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice-like feature.
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Subject: [asterisk-users] Google Voice-like feature. I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. Asterisk 1.4 -- though I could probably upgrade. Suggestions on how to make this happen? This might work - Exten = 1234,1,Dial(DAHDI/1/w#1#2#3,30,p) The Privacy mode switch on the dial would make the called party have to press 1 to accept the call. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice-like feature.
Ken D'Ambrosio ken at jots.org writes: I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. Take a look at Followme() and followme.conf. Lonnie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx + Asterisk problem - NEED HELP
PROBLEM: result of dahdi_cfg: DAHDI Tools Version - 2.2.1 How nice. Why don't you use the latest? 2.3.0[.1]? (2.4.0 should be released shortly, but that's not really something users are expected to guess). -- I decided to use the same versions with the tutorial. I did the other way but I had (more) problems. What's the output of lsdahdi ? This suggests a misconfigured /etc/dahdi/chan_dahdi.conf or one of its -- ### Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER) 1 FXOFXSKS (SWEC: MG2) 2 FXOFXSKS (SWEC: MG2) RED 3 FXSFXOKS (SWEC: MG2) 4 FXSFXOKS (SWEC: MG2) Haveing just 'wctdm' in /etc/dahdi/modules would save you a second or two on loading (and a bunch of log messages). -Thanks. Done this. zapata.conf is ignored for asterisk = 1.6.0 . Please tell me you don't use 1.4.x . - AHA. That is my big fault (How Should I do the configuration. use chan_dahdi_additional.conf?) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx + Asterisk problem - NEED HELP
Now it is done... asterisk*CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudodefault default In Service 1from-pstn en default In Service 2from-pstn en default In Service using chan_dahdi.conf.template. from-internal is missing etc. But I will try to configure it right. (More questions are on the way.) On 2 September 2010 18:36, Mehmet Kuzulugil mehmetkuzulu...@gmail.comwrote: PROBLEM: result of dahdi_cfg: DAHDI Tools Version - 2.2.1 How nice. Why don't you use the latest? 2.3.0[.1]? (2.4.0 should be released shortly, but that's not really something users are expected to guess). -- I decided to use the same versions with the tutorial. I did the other way but I had (more) problems. What's the output of lsdahdi ? This suggests a misconfigured /etc/dahdi/chan_dahdi.conf or one of its -- ### Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER) 1 FXOFXSKS (SWEC: MG2) 2 FXOFXSKS (SWEC: MG2) RED 3 FXSFXOKS (SWEC: MG2) 4 FXSFXOKS (SWEC: MG2) Haveing just 'wctdm' in /etc/dahdi/modules would save you a second or two on loading (and a bunch of log messages). -Thanks. Done this. zapata.conf is ignored for asterisk = 1.6.0 . Please tell me you don't use 1.4.x . - AHA. That is my big fault (How Should I do the configuration. use chan_dahdi_additional.conf?) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google Voice-like feature.
Want to thank everyone who mailed; a couple of your ideas got me going down certain paths, and found the answer here: http://www.voip-info.org/wiki/view/Asterisk+tips+findme Again, thanks! -Ken original message - I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. Asterisk 1.4 -- though I could probably upgrade. Suggestions on how to make this happen? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx + Asterisk problem - NEED HELP
One more thing. Can anybody point me to a sample configuration for 2 PSTN lines and 2 internal phones. (May be plus a SIP server) for Asterisk 1.6.x On 1 September 2010 15:41, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Wed, Sep 01, 2010 at 02:58:59PM +0300, Mehmet Kuzulugil wrote: Hello, After installing on Ubuntu 10.04 using the tutorial on http://hmontoliu.blogspot.com/2010/03/installing-asterisk-and-freepbx-on.html I have a running instance of Asterisk. PROBLEM: result of dahdi_cfg: DAHDI Tools Version - 2.2.1 How nice. Why don't you use the latest? 2.3.0[.1]? (2.4.0 should be released shortly, but that's not really something users are expected to guess). DAHDI Version: 2.2.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 4 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 Setting echocan for channel 3 to mg2 Setting echocan for channel 4 to mg2 The problem is about asterisk CLI results: asterisk*CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) asterisk*CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudodefaultdefault In Service What's the output of lsdahdi ? This suggests a misconfigured /etc/dahdi/chan_dahdi.conf or one of its --- INFO: related lspci result: 07:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface related dahdi_hardware result: pci::07:00.0 wctdm+ e159:0001 Wildcard TDM400P REV I One more thing: r...@asterisk:~# lsmod|grep dahdi dahdi_echocan_mg2 5729 4 dahdi_transcode 6836 1 wctc4xxp dahdi_voicebus 41854 2 wctdm24xxp,wcte12xp dahdi 210885 12 dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 1675 2 dahdi,hisax Haveing just 'wctdm' in /etc/dahdi/modules would save you a second or two on loading (and a bunch of log messages). it seems my dahdi/system.conf is ok. # Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER) fxsks=1 echocanceller=mg2,1 fxsks=2 echocanceller=mg2,2 fxoks=3 echocanceller=mg2,3 fxoks=4 echocanceller=mg2,4 # Global data loadzone= tr defaultzone= tr And this is zapata.conf: zapata.conf is ignored for asterisk = 1.6.0 . Please tell me you don't use 1.4.x . [channels] language=en ; include zap extensions defined in AMP #include zapata_additional.conf ; XTDM20B Port #1,2 plugged into PSTN ;AMPLABEL:Channel %c - Button %n context=from-pstn faxdetect=incoming echotraining=800 group=0 busydetect=yes busycount=4 hanguponpolarityswitch relaxdtmf=yes callprogress=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=2.0 txgain=2.0 immediate=yes signalling=fxs_ks channel=1-2 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya - What are their current cost?
I am not interested in open source solutions. I want to know how much the propriety systems cost in terms of licensing. Specially Avaya now a days per extension. Exclusive or Inclusive of the hardware for 10 agents as noted. Thanks On Thu, Sep 2, 2010 at 8:18 AM, Muhammad Shomail Haider msh0...@gmail.comwrote: Hi Bruce, It all depends what exactly you are in need of. A basic call center solution will only cost $500 exclusive of hardware, depending on your need you will have to decide what type of servers you need or weather you would have handsets or softphones, type of headgears you want, kind of workstation you will need. I work for a company that provides open source call center solution. You can visit the website www.crystalconsulting.pk or if you want more detail you can email the detail of the requirements on sa...@crystalconsulting.pkor you can email me and I can revert back to you with detail. Regards, Shomail On Fri, Aug 27, 2010 at 2:03 PM, justmun...@gmail.com wrote: Hi Everyone, Just a quick estimate of what Call Center Software/Hardware providers charge now a days for a 10 seat and 20 seat with upfront costs and monthly licensing cost? Thanks, -- Muhammad Shomail Haider www.shomail.blogspot.com www.facebook.com/shomail www.twitter.com/shomail www.linkedin.com/in/shomail -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya - What are their current cost?
El 02/09/10 11:32, bruce bruce escribió: I am not interested in open source solutions. Then what are your doing here? I want to know how much the propriety systems cost in terms of licensing. Specially Avaya now a days per extension. Exclusive or Inclusive of the hardware for 10 agents as noted. Go to your Avaya daddy... -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya - What are their current cost?
He is looking for competitive information...what are prospects paying for Avaya when they could be saving lots of money with Asterisk systems. Probably a better question for the biz list, but he doesn't deserve the responses he's getting. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina Sent: Thursday, September 02, 2010 12:11 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya - What are their current cost? El 02/09/10 11:32, bruce bruce escribió: I am not interested in open source solutions. Then what are your doing here? I want to know how much the propriety systems cost in terms of licensing. Specially Avaya now a days per extension. Exclusive or Inclusive of the hardware for 10 agents as noted. Go to your Avaya daddy... -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to finish an AGI
Hello community, I need to finish an AGI script when it invokes a macro from dialplan, how can i do that? it's quite confusing...the macro is making a hangup but the script continues Thanks -- Salu2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to finish an AGI
On Thu, 2 Sep 2010, Danny Dias wrote: I need to finish an AGI script when it invokes a macro from dialplan, how can i do that? it's quite confusing...the macro is making a hangup but the script continues I don't understand your question, but I'm guessing it has something to do with: 1) How to continue an AGI if a hangup occurs during execution -- trap HUP. 2) How to execute an AGI after a hangup -- use deadagi() in the h extension 3) The AGI is invoking a macro -- I have no clue with the level of detail provided. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to create a coredump for Asterisk
Hi everybody, sometimes we have an Asterisk-crash, but no clue why this is happening, so I'm trying to make a coredump to analyse it. I compiled Asterisk 1.4.20.1 on CentOS 5.4 i386 with DEBUG_THREADS and DONT_OPTIMIZE, then I start it with: # /bin/bash /usr/sbin/safe_asterisk This should do an ulimit -c unlimited, but I entered it in the terminal again. A # ps -ef | grep asterisk tells me that Asterisk is running as root and with the g-option for writing a coredump: root 21622 1 0 20:15 pts/000:00:00 /bin/bash /usr/sbin/safe_asterisk root 21627 21622 1 20:15 pts/000:00:00 /usr/sbin/asterisk -f -vvvg -c In the asterisk-start-script, the coredump-dir is configured as: DUMPDROP=/tmp Unfortunately, if I kill all asterisk-processes with kill -9 ..., a coredump never is writen to /tmp, I also looked in other dirs. Any idea what is going wrong here? Some links I found, but they do not help me: http://www.asterisk.org/doxygen/trunk/AstDebug.html http://www.voip-info.org/wiki/view/Asterisk+debugging http://man.sourcentral.org/centos5/5+core http://de.w3support.net/index.php?db=soid=17965 Thanks a lot, -- Chau y hasta luego, Thorolf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to create a coredump for Asterisk
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorolf Godawa Subject: [asterisk-users] How to create a coredump for Asterisk snip Just my opinion, but Asterisk probably isn't going to dump when you kill the process; something internal has to trigger it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya- What are their current cost?
Don- He is looking for competitive information...what are prospects paying for Avaya when they could be saving lots of money with Asterisk systems. Probably a better question for the biz list, but he doesn't deserve the responses he's getting. Agree. If it weren't for extreme high cost of telecom/voice software and gear in the late 1990s, then Mark would not have tackled it and there would be no Asterisk. Asterisk has helped create a fiercely competitive situation. Feature and capabilities comparison is a relevant topic on an Asterisk user group. -Jeff -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina Sent: Thursday, September 02, 2010 12:11 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya - What are their current cost? El 02/09/10 11:32, bruce bruce escribió: I am not interested in open source solutions. Then what are your doing here? I want to know how much the propriety systems cost in terms of licensing. Specially Avaya now a days per extension. Exclusive or Inclusive of the hardware for 10 agents as noted. Go to your Avaya daddy... -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya- What are their current cost?
He doesn't deserve the responses, but it seems that boundaries are being pushed in both sides of the response. If he thinks he's on the biz list, that's one thing, but in the purely open discussion, don't be dissing open source either. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.6.2.11 freezes the server
Hi, I have a problem that the machine running asterisk 1.6.2.11 freezes unexpectly time to time. Sometimes it runs for 4 weeks without any problem, sometimes after a free it freezes again in 24 hours. But usually it runs normally for 1 month or so before it freezes again. I could not find any additional info in any file located in /var/log/ including asterisk's messages file. Verbosity was set to 1000 but nothing can be found which would indicate explain/indicate the problem. If we connect a keyboard and a monitor to the server we see that the machine is completely frozen, no login is possible. No network services of the server are reachable but the IP address of the server can be pinged. Mandriva 2010.0 is installed on the machine with kernel version 2.6.31.13-server-1mnb. First we experienced the problem with Mandriva 2010.0 + asterisk 1.6.2.5 + kernel 2.6.31.13-desktop-1mnb on a desktop PC HW. Then we changed this simple PC hardware to a Supermicro server because we thought that it is a HW issue. We changed asterisk to 1.6.2.8 and kernel version to 2.6.31.13-server-1mnb as well. But it did not solve the problem. No wwe installed asterisk 1.6.2.11 but teh problem still exists. There are around 150 SIP devices connected to this server making max 4 simultaneous calls at the same time. Each SIP user has it's own context (outgoing) and there is a common incoming context for the users. We use Realtime: sipusers = mysql,pbx,sip_buddies sippeers = mysql,pbx,sip_buddies voicemail = mysql,pbx,voicemail_users extensions = mysql,pbx,extensions meetme = mysql,pbx,meetme phonenumbers = mysql,pbx,phonenumbers We kept the old HW with the old asterisk environment for warm backup. That one is running now for 76 days without any problem. But there is no traffic on this backup server of course. The active server freezes approx every month at least once. Have you got an idea how we could catch why this asterisk installation freezes time to time? What I can see is that the 2G memory is eaten up quite quickly by asterisk. Could you please advise. Many thanks. Regards, George -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya- Whatare their current cost?
It could be that I'm entirely confused, but I think he asked what people are paying for Avaya solutions--so he'd know what competitive pricing would be for the open source solution he's prepared to offer. When someone replied with open-source suggestions, he pointed out that that was not the information he was looking for. He did not say that he's not interested in providing open source solutions for his clients. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, September 02, 2010 1:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya- Whatare their current cost? He doesn't deserve the responses, but it seems that boundaries are being pushed in both sides of the response. If he thinks he's on the biz list, that's one thing, but in the purely open discussion, don't be dissing open source either. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channel Signalling
There´s a way to get the channel signalling in dialplan?
I have changed the code in channels/chan_dahdi.c and includes:
} else if (!strcasecmp(data, signalling)) {
ast_mutex_lock(p-lock);
snprintf(buf, len, %s, sig2str(p-sig));
ast_mutex_unlock(p-lock);
those lines on dahdi_func_read and works perfect, but wanna know if there is
a way without touch in code.
So in dialplan I can use:
exten = s,n,GotoIf($[${CHANNEL(channeltype)}=DAHDI]?:end)
exten = s,n,GotoIf($[${CHANNEL(signalling)}=MFC/R2]? mfcr2)
exten = s,n,GotoIf($[${CHANNEL(signalling)}=ISDN PRI]? isdnpri)
exten = s,n,GotoIf($[${CHANNEL(signalling)}=FXS Kewlstart]?
fxskewstart)
Regards,
Arnaldo.
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Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya- Whatare their current cost?
Thanks Don for clarification. There are lots of people on this list that hastily decide to answer without even reading a post properly. I am sure they won't even read the follow-ups. They just talk for the sake of talking. Sickens me! Please note the subject line in my original post: To compete with Avaya - What are their current cost? My question is specifically related to Avaya and other propriety call centers because I want to compete with them with Asterisk. If you know recent prices please post back. If not, don't bother. Also, please do not private message me. I will move this post to Biz List as I just noted I posted to wrong listing. Thanks to those who tried meaningful posts. On Thu, Sep 2, 2010 at 3:06 PM, Don Kelly d...@donkelly.biz wrote: It could be that I'm entirely confused, but I think he asked what people are paying for Avaya solutions--so he'd know what competitive pricing would be for the open source solution he's prepared to offer. When someone replied with open-source suggestions, he pointed out that that was not the information he was looking for. He did not say that he's not interested in providing open source solutions for his clients. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, September 02, 2010 1:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya- Whatare their current cost? He doesn't deserve the responses, but it seems that boundaries are being pushed in both sides of the response. If he thinks he's on the biz list, that's one thing, but in the purely open discussion, don't be dissing open source either. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to finish an AGI
Hello Steven...
Sorry for my poor explanation...what i'm trying to do is to invoke a Macro
from my AGI, like this:
$agi-exec(Macro,check-call-limit);
If the Macro checks that the group_name is bigger than a number specified
for every peer with setvar it should Hangup the call (frobidden,1 in the
Gotoif...) but this is not happening, the AGI always continue with is
process and it doesn´t play attention to the Hangup in the macro, the macro
is here:
[macro-check-call-limit]
exten = s,1,Set(group_name=out_calls_user_${SIPCHANINFO(peername)})
exten = s,n,Set(GROUP()=${group_name})
exten = s,n,GotoIf($[${GROUP_COUNT(${group_name})}
${MAX_OUT_CALLS_PER_USER}] forbidden,1)
; EXITO:
exten = s,n,MacroExit
; FRACASO:
exten = forbidden,1,NoOp(*** llamada saliente bloqueada: el usuario
${SIPCHANINFO(peername)} tiene actualmente
${MATH(${GROUP_COUNT(${group_name})})-1,int)} llamadas salientes)
exten = forbidden,n,Hangup(21) ; ISUP 21 = SIP 403 (Forbidden)
What should i do to finish the macro if this macro reachs the Hangup?
Thanks for your help my friend!
2010/9/2 Steve Edwards asterisk@sedwards.com
On Thu, 2 Sep 2010, Danny Dias wrote:
I need to finish an AGI script when it invokes a macro from dialplan,
how can i do that? it's quite confusing...the macro is making a hangup
but the script continues
I don't understand your question, but I'm guessing it has something to do
with:
1) How to continue an AGI if a hangup occurs during execution -- trap HUP.
2) How to execute an AGI after a hangup -- use deadagi() in the h
extension
3) The AGI is invoking a macro -- I have no clue with the level of detail
provided.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] How to finish an AGI
What should i do to finish the macro if this macro reachs the Hangup?
I tried to say: What should i do to finish the *AGI* if this macro reachs
the Hangup?
2010/9/2 Danny Dias ing.diasda...@gmail.com
Hello Steven...
Sorry for my poor explanation...what i'm trying to do is to invoke a Macro
from my AGI, like this:
$agi-exec(Macro,check-call-limit);
If the Macro checks that the group_name is bigger than a number specified
for every peer with setvar it should Hangup the call (frobidden,1 in the
Gotoif...) but this is not happening, the AGI always continue with is
process and it doesn´t play attention to the Hangup in the macro, the macro
is here:
[macro-check-call-limit]
exten = s,1,Set(group_name=out_calls_user_${SIPCHANINFO(peername)})
exten = s,n,Set(GROUP()=${group_name})
exten = s,n,GotoIf($[${GROUP_COUNT(${group_name})}
${MAX_OUT_CALLS_PER_USER}] forbidden,1)
; EXITO:
exten = s,n,MacroExit
; FRACASO:
exten = forbidden,1,NoOp(*** llamada saliente bloqueada: el usuario
${SIPCHANINFO(peername)} tiene actualmente
${MATH(${GROUP_COUNT(${group_name})})-1,int)} llamadas salientes)
exten = forbidden,n,Hangup(21) ; ISUP 21 = SIP 403 (Forbidden)
What should i do to finish the macro if this macro reachs the Hangup?
Thanks for your help my friend!
2010/9/2 Steve Edwards asterisk@sedwards.com
On Thu, 2 Sep 2010, Danny Dias wrote:
I need to finish an AGI script when it invokes a macro from dialplan,
how can i do that? it's quite confusing...the macro is making a hangup
but the script continues
I don't understand your question, but I'm guessing it has something to do
with:
1) How to continue an AGI if a hangup occurs during execution -- trap HUP.
2) How to execute an AGI after a hangup -- use deadagi() in the h
extension
3) The AGI is invoking a macro -- I have no clue with the level of detail
provided.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Salu2
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Re: [asterisk-users] How to finish an AGI
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias Subject: Re: [asterisk-users] How to finish an AGI snip This isn't really a task for AGI since it is by nature single-call specific. As I interpret what I read, you are calling this AGI from within a call and you want it to hang up all calls in a group when the group has exceeded it's group limit. If this is indeed the case, you should make a cron job to poll asterisk and do a soft hangup on the group when call-limit is exceeded. Steve (as usual) will have a better answer, but that's my .02. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to finish an AGI
No nicolas...that's not what i want...by the way sound very complicated :( What i need is to FINISH THE AGI when the MACRO reachs the Hangup...dont worry for the purpose of the macro, if the macro reachs the hangup the Agi should stop working, but it continues with his job... :( 2010/9/2 Danny Nicholas da...@debsinc.com *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Dias *Subject:* Re: [asterisk-users] How to finish an AGI snip This isn’t really a task for AGI since it is by nature single-call specific. As I interpret what I read, you are calling this AGI from within a call and you want it to hang up all calls in a group when the group has exceeded it’s group limit. If this is indeed the case, you should make a cron job to poll asterisk and do a soft hangup on the group when call-limit is exceeded. Steve (as usual) will have a better answer, but that’s my .02. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Salu2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to finish an AGI
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias Subject: Re: [asterisk-users] How to finish an AGI No nicolas...that's not what i want...by the way sound very complicated :( What i need is to FINISH THE AGI when the MACRO reachs the Hangup...dont worry for the purpose of the macro, if the macro reachs the hangup the Agi should stop working, but it continues with his job... :( Does the AGI have a SIG(HUP) = IGNORE (pardon the syntax since I don't know if it's PERL/PHP/whatever)? If so, the AGI is indestructible (will finish or have to be killed) You could have the macro set a variable at hangup and kill the AGI when it returned AGI runs Macro runs Macro gets hangup Set xx=yes Returns to AGI If (xx=yes exit) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to finish an AGI
YES YES...that's what i want ;) so simple but i was so tired :( I will try it and let you know ;) THANKS my friend 2010/9/2 Danny Nicholas da...@debsinc.com *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Dias *Subject:* Re: [asterisk-users] How to finish an AGI No nicolas...that's not what i want...by the way sound very complicated :( What i need is to FINISH THE AGI when the MACRO reachs the Hangup...dont worry for the purpose of the macro, if the macro reachs the hangup the Agi should stop working, but it continues with his job... :( Does the AGI have a SIG(HUP) = IGNORE (pardon the syntax since I don’t know if it’s PERL/PHP/whatever)? If so, the AGI is “indestructible” (will finish or have to be “killed”) You could have the macro set a variable at hangup and kill the AGI when it returned AGI runs Macro runs Macro gets hangup Set xx=yes Returns to AGI If (xx=yes exit) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Salu2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to finish an AGI
On Thu, 2 Sep 2010, Danny Dias wrote:
Sorry for my poor explanation...what i'm trying to do is to invoke a Macro from
my AGI, like this:
$agi-exec(Macro,check-call-limit);
If the Macro checks that the group_name is bigger than a number specified for
every peer with setvar it should Hangup the call (frobidden,1 in the Gotoif...)
but this
is not happening, the AGI always continue with is process and it doesn´t play
attention to the Hangup in the macro, the macro is here:
[macro-check-call-limit]
exten = s,1,Set(group_name=out_calls_user_${SIPCHANINFO(peername)})
exten = s,n,Set(GROUP()=${group_name})
exten = s,n,GotoIf($[${GROUP_COUNT(${group_name})}
${MAX_OUT_CALLS_PER_USER}] forbidden,1)
; EXITO:
exten = s,n,MacroExit
; FRACASO:
exten = forbidden,1,NoOp(*** llamada saliente bloqueada: el usuario
${SIPCHANINFO(peername)} tiene actualmente
${MATH(${GROUP_COUNT(${group_name})})-1,int)} llamadas
salientes)
exten = forbidden,n,Hangup(21) ; ISUP 21 = SIP 403 (Forbidden)
The concept of calling a macro from within an AGI seem convoluted, but may
work. I've never tried it.
Any particular reason you don't want to put the logic of the macro in your
AGI?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
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[asterisk-users] Asterisk processing URI's
How does asterisk process URI's that get sent to it? I am having a issue with a Cisco phone, where 99% works except the call forwarding. The phone issues a X-cisco-serviceuri-cfwdall which can be seen when running a sip debug on the peer directly. However the system tries to lookup the request as a extension, aka X-cisco-serviceuri-cfwdall-extension And of course can't find this extension. Is there any documentation regarding this out there specifically? I am interested in how those references work, not just for the cfwdall, but other URI's as well.. The older cisco phones used to deal with Call forwarding all by sending a SIP 302 message with Temporarily moved tag. The new xml config based ones seem to use the serviceuri xml feature. If anyone has any info on this, it would be great. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to create a coredump for Asterisk
Unfortunately, if I kill all asterisk-processes with kill -9 ..., a coredump never is writen to /tmp, I also looked in other dirs. Try kill -6 (i.e. SIGABRT). That usually triggers a core dump for me. Luki -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk failing when recording calls
We have a server that has been in operation since December of last year. Two days ago we started seeing this messages over and over (maybe a couple thousand in a minute): [Sep 2 17:46:19] DEBUG[7422] audiohook.c: Write factory 0x2aaad40a0038 was pretty quick last time, waiting for them. [Sep 2 17:46:19] DEBUG[7421] chan_dahdi.c: Write returned -1 (Resource temporarily unavailable) on channel 34 For the past two days Asterisk seems to fail at random intervals. It does not crash but it stops processing calls. You need to restart Asterisk to restore service. We are running Asterisk 1.6.2.11 with Freepbx on a CentOS 5.5 server. We started with version 1.6.2.4 and upgraded all the way to the latest just in case it was a bug that has been fixed. The server has a TE210P card with only one port in use and 4 USB100 Sangoma dual FXO modules. We use DAHDI 2.3.0.1 and Wanpipe 3.5.15 for the cards. As far as I can see they problem may happen when a call is being recorded but I have no definitive proof. The USB100 seem to be a little unstable at times. Sometimes when you reboot the server one of them will not even be listed by LSUSB or wanrouter. You need to completely power off the server so they will show up again. Everything seems to start fine but I see this message: BUG: warning at /usr/src/dahdi-linux-2.3.0.1/drivers/dahdi/dahdi-base.c:5866/dahdi_register() (Tainted: G ) Call Trace: [883cafe6] :dahdi:dahdi_register+0x56/0x309 [88515721] :wanpipe:wp_usb_tdmv_remora_software_init+0x5ee/0x916 [8850db5d] :wanpipe:wp_usb_new_if+0x291/0x5ef [800cbbc9] __kzalloc+0x9/0x21 [884a7970] :wanrouter:wan_device_new_if+0x2d0/0x4dd [884a870b] :wanrouter:wanrouter_ioctl+0x3ba/0x936 [8804c1b6] :ext3:ext3_file_write+0x16/0x91 [800182c3] do_sync_write+0xc7/0x104 [80066b88] do_page_fault+0x4fe/0x874 [800a09d8] autoremove_wake_function+0x0/0x2e [80042181] do_ioctl+0x55/0x6b [80030204] vfs_ioctl+0x457/0x4b9 [800b7605] audit_syscall_entry+0x180/0x1b3 [8004c633] sys_ioctl+0x59/0x78 [8005d28d] tracesys+0xd5/0xe0 I do not know if this is normal because Wanpipe patches DAHDI or if this indicates a problem. Any ideas or recommendations? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 670 with Extension Module | Busy Lamp Field | Directed Pickup | Speed Dial | etc
Has anyone successfully made this scenario work in 1.4. I found info at http://www.voip-info.org/wiki/view/Asterisk+presence indicating that this does not work with 1.4 implementations. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording Questions
In asterisk.conf, use these options:- cache_record_files = yes ; Cache recorded sound files to another directory during recording record_cache_dir = /tmp ; Specify cache directory (used in cnjunction with cache_record_files) -- Regards, Prince Singh Drishti-Soft Solutions Pvt Ltd W: http://www.drishti-soft.com B: http://blog.drishti-soft.com On Thu, Sep 2, 2010 at 7:22 PM, Dan Journo d...@keshercommunications.comwrote: I have our recordings written to a solid state drive rather than straight to storage disks then moved to long term storage to avoid this problem. Not an option for me at the moment. I'm running Asterisk on a cloud to reduce startup costs. Once I reach around 1,000 extensions, I'll move over the physical servers. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
