Re: [asterisk-users] Call abandoned from queue not showing in CDR (possible bug)
On Wed, 2013-01-16 at 08:06 -0600, Matthew Jordan wrote: On 01/16/2013 05:31 AM, Ishfaq Malik wrote: On Thu, 2012-01-12 at 11:51 +, Ishfaq Malik wrote: Hi Everyone This issue has reared it's ugly head again for us. If a call comes into a queue and the caller abandons the call, the call does not show in the CDR. This is also the case for asterisk version 1.8.18 Does anyone have any ideas, or try to replicate it? Thanks in advance Ish Do you have unanswered=yes set in cdr.conf? CDRs in Queues can depend heavily on your dialplan, whether or not the call is Answered prior to it going into the Queue, etc. What is the state of the inbound channel when it goes into the Queue? unanswered=yes in the cdr.conf would have too many side effects for us (i.e. a single cdr entry for each channel rung). To me this behaviour seems inconsistent with that of Dial. If I use dial to call 3 peers and the caller abandons the call I will get a single CDR entry with disposition NO ANSWER. Now if I use Queue to call the same 3 peers that are members of that queue and abandon the call, I get no cdr entry at all. This to me seems wrong. Regards Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] round-robin in asterisk 1.4
I am installing asterisk 1.4 with 2 ISP and i have one card Diguim TE210 with 2 port E1. now i bought another card Diguim TE410 and I want to add it the current configuration : connection (WIMAX) from the first ISP and connection (fiber optic) from the secend ISP. the desired configuration : connection (WIMAX) and connection (radio beam) from the first ISP.from the second ISP no change (still have the fibre optic) my question how to active the round-robin in asterisk 1.4 in order to active the 3 technology (WIMAX-radio beam and fibre optic) any help please -- _ -- 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] round-robin in asterisk 1.4
The simplest way is to use the Random function and to pickup one number from 1 to 3 and use that line. Leandro I am typing from my mobile phone... Il giorno 29/gen/2013 11:35, Salaheddine Elharit salah.elharit...@gmail.com ha scritto: I am installing asterisk 1.4 with 2 ISP and i have one card Diguim TE210 with 2 port E1. now i bought another card Diguim TE410 and I want to add it the current configuration : connection (WIMAX) from the first ISP and connection (fiber optic) from the secend ISP. the desired configuration : connection (WIMAX) and connection (radio beam) from the first ISP.from the second ISP no change (still have the fibre optic) my question how to active the round-robin in asterisk 1.4 in order to active the 3 technology (WIMAX-radio beam and fibre optic) any help please -- _ -- 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] round-robin in asterisk 1.4
thanks leandro how can i use that line in extensions.conf ? 2013/1/29 Leandro Dardini ldard...@gmail.com The simplest way is to use the Random function and to pickup one number from 1 to 3 and use that line. Leandro I am typing from my mobile phone... Il giorno 29/gen/2013 11:35, Salaheddine Elharit salah.elharit...@gmail.com ha scritto: I am installing asterisk 1.4 with 2 ISP and i have one card Diguim TE210 with 2 port E1. now i bought another card Diguim TE410 and I want to add it the current configuration : connection (WIMAX) from the first ISP and connection (fiber optic) from the secend ISP. the desired configuration : connection (WIMAX) and connection (radio beam) from the first ISP.from the second ISP no change (still have the fibre optic) my question how to active the round-robin in asterisk 1.4 in order to active the 3 technology (WIMAX-radio beam and fibre optic) any help please -- _ -- 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] Configuration Required for Remove Queue Member
Sounds like the autopause option? l. 2013/1/28 Ahmed Munir ahmedmunir...@gmail.com I would like to know, is there a method in which we can define the timeout value for a member who already login to the queue but after quite a while if he didn't answer the 3-4 calls (not going to member pause queue) but automatically remove the member from the queue? Please advise. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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] Fast AGI library/support for C C++
Dear All, Is there anyone who is having FastAGI support for C C++? We do have FastAGI working for the JAVA and rest of the language / script. But I am unable to find FastAGI for C/C++. Please let us know how to write FastAGI using C/C++. Thanks in Advance, Kashyap -- _ -- 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] Fast AGI library/support for C C++
On Tuesday 29 January 2013, Kashyap Darji wrote: Dear All, Is there anyone who is having FastAGI support for C C++? We do have FastAGI working for the JAVA and rest of the language / script. But I am unable to find FastAGI for C/C++. Please let us know how to write FastAGI using C/C++. You don't need it! C, being a compiled language, doesn't suffer from interpreter overheads and therefore doesn't require such bodgery. -- AJS Answers come *after* questions. -- _ -- 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 abandoned from queue not showing in CDR (possible bug)
On 01/29/2013 02:52 AM, Ishfaq Malik wrote: On Wed, 2013-01-16 at 08:06 -0600, Matthew Jordan wrote: On 01/16/2013 05:31 AM, Ishfaq Malik wrote: On Thu, 2012-01-12 at 11:51 +, Ishfaq Malik wrote: Hi Everyone This issue has reared it's ugly head again for us. If a call comes into a queue and the caller abandons the call, the call does not show in the CDR. This is also the case for asterisk version 1.8.18 Does anyone have any ideas, or try to replicate it? Thanks in advance Ish Do you have unanswered=yes set in cdr.conf? CDRs in Queues can depend heavily on your dialplan, whether or not the call is Answered prior to it going into the Queue, etc. What is the state of the inbound channel when it goes into the Queue? unanswered=yes in the cdr.conf would have too many side effects for us (i.e. a single cdr entry for each channel rung). To me this behaviour seems inconsistent with that of Dial. If I use dial to call 3 peers and the caller abandons the call I will get a single CDR entry with disposition NO ANSWER. Now if I use Queue to call the same 3 peers that are members of that queue and abandon the call, I get no cdr entry at all. This to me seems wrong. Regards Ish Hi Ish - The behaviour of CDRs in Queue can be interesting at times, and doesn't always match the behaviour of what occurs through Dial. In this particular case, because Queue doesn't Answer a call automatically for you, a lack of an Answer prior to going into Queue means the 'unanswered' logic kicks in for the CDRs. Hence, if a caller abandons a call attempt and no agent ever answered it, Queue/CDR code treats the call as never having been answered and, if you don't have unanswered=yes in your cdr.conf, will not log an entry. Note that there are a few other quirks with CDRs in queues in this and related scenarios, particularly when some of the members are busy (see ASTERISK-17776). We discussed making changes to this behaviour in release branches (see https://reviewboard.asterisk.org/r/2064/), but decided against it due to the ripple effect changes in CDRs have on users. If you're running into similar behaviour, you may want to backport those changes to your version. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Modify from header for anonymous call
Hello, Our supplier requires the From header of a SIP INVITE to contain certain data so the call is placed with a private caller id. It needs to be like this: From: sip:anonymous@anonymous.invalid;user=phone;tag=123455667 How do I configure Asterisk to dial anonymously? Regards, Grant -- _ -- 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 abandoned from queue not showing in CDR (possible bug)
On Tue, 2013-01-29 at 08:32 -0600, Matthew Jordan wrote: On 01/29/2013 02:52 AM, Ishfaq Malik wrote: On Wed, 2013-01-16 at 08:06 -0600, Matthew Jordan wrote: On 01/16/2013 05:31 AM, Ishfaq Malik wrote: On Thu, 2012-01-12 at 11:51 +, Ishfaq Malik wrote: Hi Everyone This issue has reared it's ugly head again for us. If a call comes into a queue and the caller abandons the call, the call does not show in the CDR. This is also the case for asterisk version 1.8.18 Does anyone have any ideas, or try to replicate it? Thanks in advance Ish Do you have unanswered=yes set in cdr.conf? CDRs in Queues can depend heavily on your dialplan, whether or not the call is Answered prior to it going into the Queue, etc. What is the state of the inbound channel when it goes into the Queue? unanswered=yes in the cdr.conf would have too many side effects for us (i.e. a single cdr entry for each channel rung). To me this behaviour seems inconsistent with that of Dial. If I use dial to call 3 peers and the caller abandons the call I will get a single CDR entry with disposition NO ANSWER. Now if I use Queue to call the same 3 peers that are members of that queue and abandon the call, I get no cdr entry at all. This to me seems wrong. Regards Ish Hi Ish - The behaviour of CDRs in Queue can be interesting at times, and doesn't always match the behaviour of what occurs through Dial. In this particular case, because Queue doesn't Answer a call automatically for you, a lack of an Answer prior to going into Queue means the 'unanswered' logic kicks in for the CDRs. Hence, if a caller abandons a call attempt and no agent ever answered it, Queue/CDR code treats the call as never having been answered and, if you don't have unanswered=yes in your cdr.conf, will not log an entry. Note that there are a few other quirks with CDRs in queues in this and related scenarios, particularly when some of the members are busy (see ASTERISK-17776). We discussed making changes to this behaviour in release branches (see https://reviewboard.asterisk.org/r/2064/), but decided against it due to the ripple effect changes in CDRs have on users. If you're running into similar behaviour, you may want to backport those changes to your version. Matt Hi Matt Thanks for the comprehensive response. I think I had better get tinkering but at least I'm now better informed. Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto Provisioning
I would like to auto provision the SPA504G phones we have in the office. What is the best method for this task? Felix This electronic message contains information from BOSH Global Services which may be company sensitive, proprietary, privileged or otherwise protected from disclosure. The information is intended to be used solely by the recipient(s) named above. If you are not an intended recipient, be aware that any review, disclosure, copying, distribution or use of this transmission or its contents is prohibited. If you have received this transmission in error, please notify the sender immediately. -- _ -- 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] Auto Provisioning
On Tue, 29 Jan 2013 19:44:04 + Felix Vazquez felix.vazq...@theboshgroup.com wrote: I would like to auto provision the SPA504G phones we have in the office. What is the best method for this task? The response to this form post looks like a good start: https://supportforums.cisco.com/thread/2022067 -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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] Fast AGI library/support for C C++
AJS == A J Stiles asterisk_l...@earthshod.co.uk writes: Please let us know how to write FastAGI using C/C++. AJS You don't need it! C, being a compiled language, doesn't suffer from AJS interpreter overheads and therefore doesn't require such bodgery. You can also use inetd(8) or xinetd(8) to handle the tcp side of things; it can call your AGI app whenever asterisk makes the tcp connection, and keep it open for future calls. Then, just use stdin and stdout as you would for a normal AGI app. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- 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] Fast AGI library/support for C C++
Dear JimC, Thanks for your comment, but if possible then could you please give some reference link or example of the using the same what you have suggested. Thanks in Advance, Kashyap On Wed, Jan 30, 2013 at 4:12 AM, James Cloos cl...@jhcloos.com wrote: AJS == A J Stiles asterisk_l...@earthshod.co.uk writes: Please let us know how to write FastAGI using C/C++. AJS You don't need it! C, being a compiled language, doesn't suffer from AJS interpreter overheads and therefore doesn't require such bodgery. You can also use inetd(8) or xinetd(8) to handle the tcp side of things; it can call your AGI app whenever asterisk makes the tcp connection, and keep it open for future calls. Then, just use stdin and stdout as you would for a normal AGI app. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- 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] #!/usr/bin/php -q unknown command
Hi, I used elastix with asterisk 1.8 when I run my AGI code, cli give me theses errors: SIP/147-0098AGI Tx agi_callingpres: 0 SIP/147-0098AGI Tx agi_callingani2: 0 SIP/147-0098AGI Tx agi_callington: 0 SIP/147-0098AGI Tx agi_callingtns: 0 SIP/147-0098AGI Tx agi_dnid: unknown SIP/147-0098AGI Tx agi_rdnis: unknown SIP/147-0098AGI Tx agi_context: from-internal SIP/147-0098AGI Tx agi_extension: 90 SIP/147-0098AGI Tx agi_priority: 2 SIP/147-0098AGI Tx agi_enhanced: 0.0 SIP/147-0098AGI Tx agi_accountcode: SIP/147-0098AGI Tx agi_threadid: -1226703984 SIP/147-0098AGI Tx agi_arg_1: SIP/147-0098AGI Tx SIP/147-0098AGI Rx Usage: php [options] [-f] file [--] [args...] SIP/147-0098AGI Tx 510 Invalid or unknown command SIP/147-0098AGI Rx php [options] -r code [--] [args...] SIP/147-0098AGI Tx 510 Invalid or unknown command SIP/147-0098AGI Rx php [options] [-B begin_code] -R code [-E end_code] [--] [args...] . . . . -- SIP/147-0098AGI Script myAGI.php completed, returning 0 I run my php script in bash linux, it seems it is not work with -q parameter what is problem? -- _ -- 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] #!/usr/bin/php -q unknown command
from ssh(console) run which php it should give you path where it is installed. Peter On 30/01/2013 08:28, Muhammad wrote: Hi, I used elastix with asterisk 1.8 when I run my AGI code, cli give me theses errors: SIP/147-0098AGI Tx agi_callingpres: 0 SIP/147-0098AGI Tx agi_callingani2: 0 SIP/147-0098AGI Tx agi_callington: 0 SIP/147-0098AGI Tx agi_callingtns: 0 SIP/147-0098AGI Tx agi_dnid: unknown SIP/147-0098AGI Tx agi_rdnis: unknown SIP/147-0098AGI Tx agi_context: from-internal SIP/147-0098AGI Tx agi_extension: 90 SIP/147-0098AGI Tx agi_priority: 2 SIP/147-0098AGI Tx agi_enhanced: 0.0 SIP/147-0098AGI Tx agi_accountcode: SIP/147-0098AGI Tx agi_threadid: -1226703984 SIP/147-0098AGI Tx agi_arg_1: SIP/147-0098AGI Tx SIP/147-0098AGI Rx Usage: php [options] [-f] file [--] [args...] SIP/147-0098AGI Tx 510 Invalid or unknown command SIP/147-0098AGI Rx php [options] -r code [--] [args...] SIP/147-0098AGI Tx 510 Invalid or unknown command SIP/147-0098AGI Rx php [options] [-B begin_code] -R code [-E end_code] [--] [args...] . . . . -- SIP/147-0098AGI Script myAGI.php completed, returning 0 I run my php script in bash linux, it seems it is not work with -q parameter what is problem? -- _ -- 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] #!/usr/bin/php -q unknown command
*which php* /usr/bin/php the path is ok, php -h gives me some parameter except -q On Wed, Jan 30, 2013 at 10:20 AM, Zyumbilev, Peter pe...@aboutsupport.comwrote: from ssh(console) run which php it should give you path where it is installed. Peter On 30/01/2013 08:28, Muhammad wrote: Hi, I used elastix with asterisk 1.8 when I run my AGI code, cli give me theses errors: SIP/147-0098AGI Tx agi_callingpres: 0 SIP/147-0098AGI Tx agi_callingani2: 0 SIP/147-0098AGI Tx agi_callington: 0 SIP/147-0098AGI Tx agi_callingtns: 0 SIP/147-0098AGI Tx agi_dnid: unknown SIP/147-0098AGI Tx agi_rdnis: unknown SIP/147-0098AGI Tx agi_context: from-internal SIP/147-0098AGI Tx agi_extension: 90 SIP/147-0098AGI Tx agi_priority: 2 SIP/147-0098AGI Tx agi_enhanced: 0.0 SIP/147-0098AGI Tx agi_accountcode: SIP/147-0098AGI Tx agi_threadid: -1226703984 SIP/147-0098AGI Tx agi_arg_1: SIP/147-0098AGI Tx SIP/147-0098AGI Rx Usage: php [options] [-f] file [--] [args...] SIP/147-0098AGI Tx 510 Invalid or unknown command SIP/147-0098AGI Rx php [options] -r code [--] [args...] SIP/147-0098AGI Tx 510 Invalid or unknown command SIP/147-0098AGI Rx php [options] [-B begin_code] -R code [-E end_code] [--] [args...] . . . . -- SIP/147-0098AGI Script myAGI.php completed, returning 0 I run my php script in bash linux, it seems it is not work with -q parameter what is problem? -- _ -- 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
