Re: [asterisk-users] MixMonitor with b option recording all calls
On Monday 22 Sep 2014, Yuriy Gorlichenko wrote: Hello I have an issue wit MixMonitor. I need to record only answered calls, so I set b option for this but calls still recording even call no answered My asterisk version 12.5.1, at my other servers with older versions of asterisk (11.8 for example) MixMonitor works fine. What technology are you using for your outgoing calls? SIP trunk, IAX trunk, ISDN, mobile or analogue phone lines? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] error receiving a fax ... but with a fax that was received without problems
On 21/09/2014 9:11 PM, Bart Remmerie wrote: Dear Joshua, I don't think this is it: first, this has been working in the past second, why would I get a message like the one below exited non-zero if everything is normal. Joshua is correct, to reliably process the received fax you will need to process it in the 'h' extension. You may want to refer to this thread http://lists.digium.com/pipermail/asterisk-users/2013-June/279625.html, note the afax2email script will e-mail the received tiff fax image if the PDF conversion fails. Larry. -- _ -- 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] Install Asterisk 1.4 and Asterisk 12.4 on the same machine
Hi, We have a server with asterisk 1.4. We are upgrading to asterisk 12.4. Is there a way to install 12.4 on the same machine? At any point we will only run either 1.4 or 12.4. Thank you, Deepak -- _ -- 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] MixMonitor with b option recording all calls
SIP trunks 2014-09-22 12:12 GMT+04:00 A J Stiles asterisk_l...@earthshod.co.uk: On Monday 22 Sep 2014, Yuriy Gorlichenko wrote: Hello I have an issue wit MixMonitor. I need to record only answered calls, so I set b option for this but calls still recording even call no answered My asterisk version 12.5.1, at my other servers with older versions of asterisk (11.8 for example) MixMonitor works fine. What technology are you using for your outgoing calls? SIP trunk, IAX trunk, ISDN, mobile or analogue phone lines? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] MixMonitor with b option recording all calls
THIS IS NOT WHERE YOUR REPLY BELONGS On Monday 22 Sep 2014, Yuriy Gorlichenko wrote: 2014-09-22 12:12 GMT+04:00 A J Stiles asterisk_l...@earthshod.co.uk: On Monday 22 Sep 2014, Yuriy Gorlichenko wrote: Hello I have an issue wit MixMonitor. I need to record only answered calls, so I set b option for this but calls still recording even call no answered My asterisk version 12.5.1, at my other servers with older versions of asterisk (11.8 for example) MixMonitor works fine. What technology are you using for your outgoing calls? SIP trunk, IAX trunk, ISDN, mobile or analogue phone lines? SIP trunks Well, SIP certainly allows for full supervisory information (analogue doesn't, and all calls are deemed answered if the exchange line was available). What have you got in your sip.conf ? And what does your SIP trunk provider have to say on the matter? (It wouldn't be totally unknown for a dodgy telco to provide not-entirely-truthful supe.) -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] SIPAddHeader from a realtime databse
Hi Guys I'm using asterisk 1.8.23.1 When I add a SIP Header from inside the extensions.conf (SIPAddHeader(Alert-Info:http://www.notused.com\;info=alert-internal\;x-line-id=0) ) it works fine. When I try to do the same thing from within a database table, all of the string apart from x-line-id=0 gets ignored. I've tried escaping the semicolon and not escaping it and the result is always the same, just the last part of the full string is expressed. Some of the ways that I have tried to enter the string are below: appdata='Alert-Info:http://www.notused.com \\;info=alert-internal\\;x-line-id=0' appdata='Alert-Info:http://www.notused.com ;info=alert-internal;x-line-id=0' appdata='Alert-Info:http://www.notused.com ;info=alert-internal;x-line-id=0' Does anyone know the correct format to store this in a DB table for it to be expressed correctly? I'm using MySQL. Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI answer too fast
I have a PRI working just fine. There is an automated system that needs to hear at least one ring. Currently when I call into asterisk it answers right away and I do not hear the ring at all. I am 1.4.43 for this system is there a way to set answer on ring or something so the other end at least gets 1 ring. I have dahdi 2.9.1 Thanks jerry -- _ -- 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] PRI answer too fast
is there a way to set answer on ring or something so the other end at least gets 1 ring. First entry in your incoming context should be: exten = s,1,Answer(500) Doug -- _ -- 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] PRI answer too fast
Answer([delay]) If the channel is ringing, answer it, otherwise do nothing. If a delay is specified, Asterisk will wait this number of milliseconds AFTER answering the call. If you want to add a delay prior answering, use Wait. On Mon, Sep 22, 2014 at 9:06 AM, Doug Lytle supp...@drdos.info wrote: is there a way to set answer on ring or something so the other end at least gets 1 ring. First entry in your incoming context should be: exten = s,1,Answer(500) Doug -- _ -- 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 -- Copyright 2014 Derek Andrew (excluding quotations) +1 306 966 4808 Information and Communications Technology University of Saskatchewan Peterson 120; 54 Innovation Boulevard Saskatoon,Saskatchewan,Canada. S7N 2V3 Timezone GMT-6 Typed but not read. -- _ -- 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] Install Asterisk 1.4 and Asterisk 12.4 on the same machine
On 9/22/14, 5:03 AM, Deepak Rawat wrote: Hi, We have a server with asterisk 1.4. We are upgrading to asterisk 12.4. Is there a way to install 12.4 on the same machine? At any point we will only run either 1.4 or 12.4. The answer is that you can but you really shloud not as it complicates lots of things. You can install both versions and run them at the same time as long as you pick different ports for all services like SIP, IAX, AMI, RTP, etc. Not using default ports will mean that you have to modify all configurations for each endpoint you want to connect to the secondary Asterisk. If you only want to try out Asterisk 12 why not use a Virtual Machine for testing? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- 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] Install Asterisk 1.4 and Asterisk 12.4 on the same machine
Thank you for the response Carlos! We have done the testing on a VM. I want to deploy it on a production machine which already has 1.4 on it. I want to keep both so in case there is any issue with 12.4, I can quickly revert to 1.4. I can take care of configuring different ports for different services but will there be any issues due to upgradation of dependencies? Is there a way to install dependencies of asterisk 12.4 at different path so they don't conflict with 1.4 dependencies? On Mon, Sep 22, 2014 at 9:55 PM, Carlos Chavez cur...@telecomabmex.com wrote: On 9/22/14, 5:03 AM, Deepak Rawat wrote: Hi, We have a server with asterisk 1.4. We are upgrading to asterisk 12.4. Is there a way to install 12.4 on the same machine? At any point we will only run either 1.4 or 12.4. The answer is that you can but you really shloud not as it complicates lots of things. You can install both versions and run them at the same time as long as you pick different ports for all services like SIP, IAX, AMI, RTP, etc. Not using default ports will mean that you have to modify all configurations for each endpoint you want to connect to the secondary Asterisk. If you only want to try out Asterisk 12 why not use a Virtual Machine for testing? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- 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] MixMonitor with b option recording all calls
my sip.conf [kamailio_ext1] type=friend host = my.superprovider.com port = 5068 canreinvite = no insecure = invite,port transport=udp trustrpid=yes context = incoming videosupport=no directmedia=no dtlsenable = no tlsenable=no disallow=all allow=alaw allow=opus allow=ulaw connection goes great. SIP session have all packets So astersik send INVITE - to trunk then - TRYING - RINGING - OK -ACK and at the end some of callers send BYE And then Goes OK and ACK full SIP session. Signaling is Ok. At CDR I see ANSWERED Wheb Call Unanswered I see INVITE - to trunk then - TRYING - RINGING some of callers CANCEL OK ACK So There is full session too At CDR I see No Answered session (that is write) And after that I see empty record file 2014-09-22 18:40 GMT+04:00 A J Stiles asterisk_l...@earthshod.co.uk: THIS IS NOT WHERE YOUR REPLY BELONGS On Monday 22 Sep 2014, Yuriy Gorlichenko wrote: 2014-09-22 12:12 GMT+04:00 A J Stiles asterisk_l...@earthshod.co.uk: On Monday 22 Sep 2014, Yuriy Gorlichenko wrote: Hello I have an issue wit MixMonitor. I need to record only answered calls, so I set b option for this but calls still recording even call no answered My asterisk version 12.5.1, at my other servers with older versions of asterisk (11.8 for example) MixMonitor works fine. What technology are you using for your outgoing calls? SIP trunk, IAX trunk, ISDN, mobile or analogue phone lines? SIP trunks Well, SIP certainly allows for full supervisory information (analogue doesn't, and all calls are deemed answered if the exchange line was available). What have you got in your sip.conf ? And what does your SIP trunk provider have to say on the matter? (It wouldn't be totally unknown for a dodgy telco to provide not-entirely-truthful supe.) -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] DAHDI-Linux and DAHDI-Tools 2.10.0.1 Now Available
The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.10.0.1 DAHDI-Tools-v2.10.0.1 dahdi-linux-complete-2.10.0.1+2.10.0.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete * Fixes an issue where ringback tones may be silent https://issues.asterisk.org/jira/browse/DAHLIN-341 * Fixes an issue where updating to kernel 3.16.2 in fedora 20 causes chan_dahdi.c: Read failed with -1: Invalid argument in asterisk logs https://issues.asterisk.org/jira/browse/DAHLIN-340 * Fix an issue with dahdi dynamic spans crashing after some time https://issues.asterisk.org/jira/browse/DAHLIN-328 Shortlog of dahdi-linux changes since v2.10.0: Michael Walton (1): dynamic: Prevent oops due to inverted compile flag Shaun Ruffell (1): dahdi: Fix failure to read / write on kernel 3.16+ Shortlog of dahdi-tools changes since v2.10.0: Oron Peled (1): xpp: revert USB clear_halt change and better overrides. Russ Meyerriecks (1): tonezone: Fix regression in Australian tone patch The diffstat from the dahdi-linux v2.10.0 release: drivers/dahdi/dahdi-base.c| 16 drivers/dahdi/dahdi_dynamic.c | 16 +--- 2 files changed, 25 insertions(+), 7 deletions(-) The diffstat from the dahdi-tools v2.10.0 release: tonezone.c | 7 ++- xpp/xtalk/xusb.c | 8 +++- 2 files changed, 9 insertions(+), 6 deletions(-) For a full list of changes in these releases, please see the shortlog at: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.10.0.1 http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.10.0.1 Issues found in this release can be reported in the DAHDI-Linux [1] and DAHDI-Tools [2] projects at https://issues.asterisk.org/jira [1] https://issues.asterisk.org/jira/browse/DAHLIN [2] https://issues.asterisk.org/jira/browse/DAHTOOL Thank you for your continued support of Asterisk! -- _ -- 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] DAHDI v2.10.0.1 Fixes loadzone=us ringback tones.
In case it wasn't obvious in the DAHDI release announcement. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI v2.10.0.1 Fixes loadzone=us ringback tones.
DAHDI v2.10.0.1 Fixes loadzone=us ringback tones. Thank you!!! Doug -- _ -- 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
