Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
I am still using Asterisk 1.4 because of the Asterisk GUI. I don't understand why it was ever dropped, it's easy to setup (no SQL databases), quick, works well and in my experiance it gets along with manual config file changes. The only real issue I've encountered with 1.4 is Digium can't seem to properly build RPMs... Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- 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] is res_timing_timerfd module stable in 1.8?
hi: my current system is 1.6.2. I have dahdi hardware card. I must disable res_timing_timerfd module or sometimes phone calls would become silent suddenly. I don't know the situation in 1.8. I heard that timing is still a problem in 1.8. should I keep using res_timing_dahdi or I can use res_timing_timerfd to get some benefit if I upgrade to 1.8? thank a lot for information!! Regards, tbskyd -- _ -- 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] is res_timing_timerfd module stable in 1.8?
New Text at Bottom: --- hi: my current system is 1.6.2. I have dahdi hardware card. I must disable res_timing_timerfd module or sometimes phone calls would become silent suddenly. I don't know the situation in 1.8. I heard that timing is still a problem in 1.8. should I keep using res_timing_dahdi or I can use res_timing_timerfd to get some benefit if I upgrade to 1.8? thank a lot for information!! Regards, tbskyd -- Hi, There are a few issues on Mantis that people think are related to timing and timerfd. I'm not sure what the benefits of timerfd over dahdi are when you have the dahdi hardware (maybe someone else could comment). I'm currently testing 1.8.4 with dahdi timing (no hardware) to see if it solves some of the problems we have been having. Its working so far (touch wood) but its only three days and about 500 calls in. I would say if you have the dahdi hardware timer then use it. If you have been having a timing related issue with 1.6 and/or 1.8 have a look on issues.asterisk.org and see if anyone else has reported similar problems that you can add a bit more info to, if not make your own report. Regards, Nic. -- _ -- 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] Discussion: Are we ready to leave 1.4 behind?
5 maj 2011 kl. 18.30 skrev Ira: At 07:56 AM 5/5/2011, you wrote: So how can we fix this? How can we get more people involded? What makes projects like FedoraTesting[3] and DebianTesting[4] popular? How can the Asterisk project reproduce their success? Well, it's not a lot of people willing to run beta software on their phone system. Phones need to work and for most people they need to work perfectly all the time. I'm one of those oddities that will always run beta software if given the chance but my experience is that quite rare. As I've said before, I'm more then willing to help with answering questions about the testsuite or reviewing code that people want to get merged in. We also have an IRC channel, #asterisk-testing available for people to join, ask question, idle, lurk, etc, or if you want to reply to this thread, feel free. But get involved! :) So I'm the person who has never been able to keep 1.8 alive on my system for more than a minute or two and I've probably tried more than 10 different betas and release versions. I posted a bug report which was closed in minutes, I posted the problem on this list every few tries and zero response. I tried to figure out mIRC. It's installed on my machine but I've never got past that. I just don't get the instructions. I know that all the people involved in the project are Linux heads, but some of us, like me, have a Linux box only because of Asterisk and if you want my help, you need to make being involved accessible and stop assuming we all know what you know. I see the words, jut post a bug report on Mantis posted all the time and I'm sure it means as little to others as it means to me. Maybe there needs to be a web page somewhere, Asterisk beta testing for dummies so that you can point us to so you don't have to answer the stupid questions over and over. I've beta tested enough and had enough beta testers to understand the kinds of things that make it possible to get bugs fixed, but it's usually a very small percentage of users that understand that. Thanks for the feedback, Ira. It makes me very sad to hear what you say and I hope that we can get more resources from the community to assist in the process to make it more friendly. We want to get those bug reports. The one thing I hate to hear when I'm travelling at conferences is that oh, I known that bug for a long time but did not bother to report it. Apologies for your experience with the bug process. Regards, /Olle -- _ -- 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] Discussion: Are we ready to leave 1.4 behind?
Thanks for the feedback, Ira. It makes me very sad to hear what you say and I hope that we can get more resources from the community to assist in the process to make it more friendly. We want to get those bug reports. The one thing I hate to hear when I'm travelling at conferences is that oh, I known that bug for a long time but did not bother to report it. Apologies for your experience with the bug process. Indeed, it seems as though there might be a problem of discoverability of how to report issues. Is it too burdensome to suggest attaching this link (along with a short description) to the footer of list e-mails? http://www.asterisk.org/developers/bug-guidelines That does a fair job (though not perfect, and I think suggestions for improvement are welcome) of detailing the process. It's probably also incumbent upon us all, as a community, to do a better job than just report it on Mantis. I'm quite certain every one of us would like the most stable, bug-free code in Asterisk as is possible, and if it takes an extra minute or two of our time to help get the issues reported in the first place it will be time well-spent. - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy 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
[asterisk-users] TCP Trigger on incoming call request
Hi, is it possible to configure a TCP trigger to a predefined address and port on a incoming call request? Some background: For example Client 1 tries to call Client 2, Client 1 is sending the call request to Asterisk (SIP-Server). Asterisk open a connection to the predefined address and port and send a simple TCP message (trigger) with caller and callee ID and close the connection afterwards. Is this scenario possible without any plugin like the Java API or similar? Thanks, Daniel -- _ -- 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] TCP Trigger on incoming call request
Daniel, Have you thought about using CURL from the Dialplan? Henk Daniel Isenmann schreef: Hi, is it possible to configure a TCP trigger to a predefined address and port on a incoming call request? Some background: For example “Client 1” tries to call “Client 2”, “Client 1” is sending the call request to Asterisk (SIP-Server). Asterisk open a connection to the predefined address and port and send a simple TCP message (trigger) with caller and callee ID and close the connection afterwards. Is this scenario possible without any plugin like the Java API or similar? Thanks, Daniel -- _ -- 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] TCP Trigger on incoming call request
Look at function CURL -Original Message- From: Daniel Isenmann daniel.isenm...@seetec.de Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 6 May 2011 13:04:09 To: asterisk-users@lists.digium.comasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] TCP Trigger on incoming call request -- _ -- 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] Asterisk 1.6.2.18, Cisco 79XX not registering
Hi all, I have a production server running with about 90 Cisco 79[46]1's and SIP release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and upgraded last night after hours. (Seemed low risk to me!) Much to my surprise, not a single one of the Cisco 79XX phones would register. Since it's a production server, I rolled back to 1.6.2.9 and everything was fine. All my Linksys SPA phones and Polycom speaker phones registered just fine. I am now setting up test servers with both 1.6.2.18 and 1.8.3.3 to collect some debug. I am just curious has anyone else had SIP issues with these phones and updating Asterisk broke them? I will post results of my findings after I have time to collect them. Cassius Smitha -- _ -- 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] Missed call when call is answered by other phone
Hi, I use follow me and have several SIP phones answering, works nice but: All phones that did not answer a call have the number in missed call list even if answered by other ext. CDR gets messy too. Difficult to see if call is answered. I was thinking of possible solution: Turn of missed call on phones and instead have a php script make a list on web or maybe if not a smarter way over SIP? Any good idea? Best regards HB -- _ -- 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 1.6.2.18, Cisco 79XX not registering
I seem to recall this issue mentioned on asterisk-dev. Check issues.digium.com and see if there is anything similar to your issue. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Friday, May 06, 2011 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering Hi all, I have a production server running with about 90 Cisco 79[46]1's and SIP release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and upgraded last night after hours. (Seemed low risk to me!) Much to my surprise, not a single one of the Cisco 79XX phones would register. Since it's a production server, I rolled back to 1.6.2.9 and everything was fine. All my Linksys SPA phones and Polycom speaker phones registered just fine. I am now setting up test servers with both 1.6.2.18 and 1.8.3.3 to collect some debug. I am just curious - has anyone else had SIP issues with these phones and updating Asterisk broke them? I will post results of my findings after I have time to collect them. Cassius Smitha -- _ -- 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] Background music during a call
Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the two parties will enjoy/maybe not. I could'nt make it to work. Anyways, I think this feature will require some creative dialplanning as its not supported by default by the software. If anyone can tell me how to create a ghost call then the rest I may be able to figure out myself. If there is another way plz share coz im on a deadline. I am a wee bit of a programmer also, so if your idea needs changes in the code please dont hesitate to share, otherwise you WILL get a call from me with a special background noise crafted just for you :) Meanwhile i'll try my best to come up with a solution. Cheers -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.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] Cannot built kmod-dahdi-linux for PAE kvariant from SRPM
After many moons I have revisited this problem and found a solution that moves the problem further up the stack. I will post my new problem separately but just for completeness here is the solution. Original problem: trying to build kmod-dahdi-linux for out of date PAE kernel. Errors: rpmbuild --bb ~/localrpms/SPECS/dahdi-linux-kmod.spec --target=i686 --define kversion `uname -r` + make KVERS=2.6.18-128.el5xen modules You do not appear to have the sources for the 2.6.18-128.el5xen kernel installed. make: *** [modules] Error 1 error: Bad exit status from /var/tmp/rpm-tmp.78040 (%build) Solution: Specify the kernel variant in the rpm build command rpmbuild --bb ~/localrpms/SPECS/dahdi-linux-kmod.spec --target=i686 \ --define kversion `uname -r` --define kvariants 'PAE' Steve Hindmarch -- _ -- 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 1.6.2.18, Cisco 79XX not registering
On 6 May 2011 16:30, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Friday, May 06, 2011 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering Hi all, I have a production server running with about 90 Cisco 79[46]1's and SIP release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and upgraded last night after hours. (Seemed low risk to me!) Much to my surprise, not a single one of the Cisco 79XX phones would register. Since it's a production server, I rolled back to 1.6.2.9 and everything was fine. All my Linksys SPA phones and Polycom speaker phones registered just fine. I am now setting up test servers with both 1.6.2.18 and 1.8.3.3 to collect some debug. I am just curious - has anyone else had SIP issues with these phones and updating Asterisk broke them? I will post results of my findings after I have time to collect them. Cassius Smitha I seem to recall this issue mentioned on asterisk-dev. Check issues.digium.com and see if there is anything similar to your issue. I also remember this being mentioned - I believe it was fixed in the chan_sip Via: header handling code. The fix is in branches/1.6.2 already, so you should be able to grab the patch without too much trouble. Regards, Steve -- _ -- 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] Cannot install dahdi-linux on (old) PAE kernel.
I am trying to install dahdi-linux from packages onto an OEL5u3 server which has an old kernel (5.2.6.18_128) and is a PAE variant. As there are no kmod packages now available for this kernel I am having to build them from source packages. I have installed the dahdi-firmware-2.0.0-1_centos5 RPM directly. I have downloaded the following SRPMS dahdi-linux-2.4.1.2-1_centos5.src.rpm dahdi-linux-kmod-2.4.1.2-1_centos5.2.6.18_238.9.1.el5.src.rpm I have installed them and built them to create these RPMS dahdi-linux-2.4.1.2-1_centos5.i686.rpm dahdi-linux-devel-2.4.1.2-1_centos5.i686.rpm kmod-dahdi-linux-PAE-2.4.1.2-1_centos5.2.6.18_128.el5.i686.rpm Now if I try to installed the packages I get the following dependency error. $ sudo rpm -ivh kmod-dahdi-linux-PAE-2.4.1.2-1_centos5.2.6.18_128.el5.i686.rpm dahdi-linux-2.4.1.2-1_centos5.i686.rpm error: Failed dependencies: kmod-dahdi-linux is needed by dahdi-linux-2.4.1.2-1_centos5.i686 An inspection of the kmod-dahdi-linux RPM shows that is provides kmod-dahdi-linux-PAE, not kmod-dahdi-linux. How do I get the dahdi-linux package to recognise the kmod PAE package as the right one for this kernel? Steve Hindmarch -- _ -- 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] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer
Hi, On Thu, May 05, 2011 at 05:30:04PM -0400, Paul Belanger wrote: On 11-05-05 05:14 PM, Mark G Thomas wrote: Hi, I think this must be a bug introduced with 1.6.2.17.something. When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or 1.6.2.18, my AEL Dial() commands with the U options fail with the following error: [May 3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non-existent destination for gosub: (Context:screen, Extension:s, Priority:1) You might want to have a look at: https://issues.asterisk.org/view.php?id=18910 Thanks. This is it. If I'm reading this right, it describes the change which broke things for me, but no solution applicable to my Dial() command U flag, which is invoking my AEL GoSub (Macro). Switching the Dials back to the M flag doesn't fix it either. It sure seems to me this change to AEL has had unexpected consequences in terms of breaking things in dialplans. Mark -- Mark G. Thomas (m...@misty.com) Web: http://mgtinternet.com/ Tel: +1-215-512-0112 US: 877-512-0112 -- _ -- 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] QueueCallerAbandon is not triggering after 1.8.3.3...
Has anyone else noticed that QueueCallerAbandon is not showing up in the AMI after the 1.8.3.3? Am I missing something? I'm getting what seems like everything else but QueueCallerAbandon. v/r, Me -- _ -- 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] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer
The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Mark G Thomas Sent: Friday, May 06, 2011 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer Hi, On Thu, May 05, 2011 at 05:30:04PM -0400, Paul Belanger wrote: On 11-05-05 05:14 PM, Mark G Thomas wrote: Hi, I think this must be a bug introduced with 1.6.2.17.something. When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or 1.6.2.18, my AEL Dial() commands with the U options fail with the following error: [May 3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non- existent destination for gosub: (Context:screen, Extension:s, Priority:1) You might want to have a look at: https://issues.asterisk.org/view.php?id=18910 Thanks. This is it. If I'm reading this right, it describes the change which broke things for me, but no solution applicable to my Dial() command U flag, which is invoking my AEL GoSub (Macro). Switching the Dials back to the M flag doesn't fix it either. It sure seems to me this change to AEL has had unexpected consequences in terms of breaking things in dialplans. I was under the impression that this had been fixed, although perhaps it's not yet in a release. Is there a chance you try with the latest 1.6.2 branch from SVN? - Brad -- _ -- 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] Configuring Voicemail in Asterisk 1.8
Hi All; Already in the voicemail.conf file, I added the extension 500 and kindly find below my voicemail configuration: [Internal] 0 = 1234,Gama Operator,opera...@gama.com 500 = 1234,Operator,opera...@gama.com 501 = 1234,Employer Name,employer_em...@gama.com 502 = 1234,Employer Name,employer_em...@gama.com Asterisk version is 1.8 and currently I am getting this warning message: [May 7 19:32:46] WARNING[4328]: app_voicemail.c:5535 leave_voicemail: No entry in voicemail config file for 'u500' So what I might be missing? Regards Bilal -- _ -- 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] Configuring Voicemail in Asterisk 1.8
change u500 in extension.conf because asterisk 1.8 user 500,u like following exten = open-NOANSWER,1,VoiceMail(5800,u) Date: Fri, 6 May 2011 09:49:17 -0700 From: bilmar...@yahoo.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Configuring Voicemail in Asterisk 1.8 Hi All; Already in the voicemail.conf file, I added the extension 500 and kindly find below my voicemail configuration: [Internal] 0 = 1234,Gama Operator,opera...@gama.com 500 = 1234,Operator,opera...@gama.com 501 = 1234,Employer Name,employer_em...@gama.com 502 = 1234,Employer Name,employer_em...@gama.com Asterisk version is 1.8 and currently I am getting this warning message: [May 7 19:32:46] WARNING[4328]: app_voicemail.c:5535 leave_voicemail: No entry in voicemail config file for 'u500' So what I might be missing? Regards Bilal -- _ -- 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] Gateway GSM x Comercio Indevido ?
Boa tarde a todos, Colegas estou a procura de um gateway GSM para ligar ao servidor asterisk de nossa empresa, o objetivo é interligar clientes e parceiros comerciais a nossa central, reduzindo custo das ligações para celular, porem ao ver o regulamento dos planos oferecidos pelas operadoras vi que trata-se de uso indevido a comercialização do serviço bem como a utilização dos chips em equipamentos como GSM Box, Black Box e equipamentos similares. Enfim, minha duvida é: Como então é realizado a venda que vejo em diversos sites de serviços voip de ligação celular que custa até R$ 0,30 centavos (exemplo) ? Qual seria a melhor solução em equipamento, tendo em mente que a idéia seria que o asterisk possa realizar cerca de 40 ligações simultaneamente para celulares da vivo, tim, claro e oi ? Agradeço a quem puder me esclarecer. Cláudio Duarte -- _ -- 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] Cannot install dahdi-linux on (old) PAE kernel.
On Fri, May 6, 2011 at 11:58 AM, stephen.hindma...@bt.com wrote: I am trying to install dahdi-linux from packages onto an OEL5u3 server which has an old kernel (5.2.6.18_128) and is a PAE variant. As there are no kmod packages now available for this kernel I am having to build them from source packages. [snip] kmod-dahdi-linux is needed by dahdi-linux-2.4.1.2-1_centos5.i686 An inspection of the kmod-dahdi-linux RPM shows that is provides “kmod-dahdi-linux-PAE”, not “kmod-dahdi-linux”. How do I get the dahdi-linux package to recognise the kmod PAE package as the right one for this kernel? Hi Steven, Can you put the .spec file from dahdi-linux-kmod package up? I can get the srpm, but I'm stuck on a weak machine at the moment. Maybe I can help you to modify it to also provide non-PAE requirement? - Bob Beers -- _ -- 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] Background music during a call
On Fri, May 6, 2011 at 6:30 PM, Rizwan Hisham rizwanhas...@gmail.com wrote: I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). Let's start with your actual dialplan (without the background music) and we could start from that point. Hint: I am planning to use option G of the Dial application + a meetme room where a ghost call will play the specified MOH class (lovely/cursing). HTH, Ioan. -- _ -- 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] Configuring Voicemail in Asterisk 1.8
You are using an old format for specifying the mailbox. See core show application voicemail for the correct usage. Also read ALL the UPGRADE*.txt files. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Friday, May 06, 2011 12:49 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Configuring Voicemail in Asterisk 1.8 Hi All; Already in the voicemail.conf file, I added the extension 500 and kindly find below my voicemail configuration: [Internal] 0 = 1234,Gama Operator,opera...@gama.com 500 = 1234,Operator,opera...@gama.com 501 = 1234,Employer Name,employer_em...@gama.com 502 = 1234,Employer Name,employer_em...@gama.com Asterisk version is 1.8 and currently I am getting this warning message: [May 7 19:32:46] WARNING[4328]: app_voicemail.c:5535 leave_voicemail: No entry in voicemail config file for 'u500' So what I might be missing? Regards Bilal -- _ -- 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] polycom page custom ring
Currently we have following working page now i want to add custom ring type so
people pay attention. Anybody know about what would be the variable to change
custom ringer
[all-page]
exten = s,1,Set(TIMEOUT(absolute)=15)
exten = s,n,AGI(page.agi)
exten = s,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = s,n,Set(CALLERID(name)=Page)
exten = s,n,Playback(silence/1)
exten = s,n,Page(${PAGE_GROUP})
exten = s,n,Hangup()
-S
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Re: [asterisk-users] Cannot install dahdi-linux on (old) PAE kernel.
On Fri, May 6, 2011 at 1:27 PM, Bob Beers bob.be...@gmail.com wrote: Hi Steven, Can you put the .spec file from dahdi-linux-kmod package up? Nevermind, I got it. :-) Looking at it now. Did you CC [Packager: Jason Parker jpar...@digium.com]? - Bob -- _ -- 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] question on ways to activate voicemail light on polycom
Is there a way in asterisk to Activate/Clear the blinking light on polycom phones indicating VM. Either from an AGI or some way in the dialplan? I want to be able to control this light for from my application. I have an AGI to do something similiar to VM and want to light /clear the light myself. 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] Asterisk 1.6.2.18, Cisco 79XX not registering
It was my problem ;) https://issues.asterisk.org/view.php?id=18951 fixed in svn On 6 May 2011 16:45, Steve Davies davies...@gmail.com wrote: On 6 May 2011 16:30, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Friday, May 06, 2011 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering Hi all, I have a production server running with about 90 Cisco 79[46]1's and SIP release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and upgraded last night after hours. (Seemed low risk to me!) Much to my surprise, not a single one of the Cisco 79XX phones would register. Since it's a production server, I rolled back to 1.6.2.9 and everything was fine. All my Linksys SPA phones and Polycom speaker phones registered just fine. I am now setting up test servers with both 1.6.2.18 and 1.8.3.3 to collect some debug. I am just curious - has anyone else had SIP issues with these phones and updating Asterisk broke them? I will post results of my findings after I have time to collect them. Cassius Smitha I seem to recall this issue mentioned on asterisk-dev. Check issues.digium.com and see if there is anything similar to your issue. I also remember this being mentioned - I believe it was fixed in the chan_sip Via: header handling code. The fix is in branches/1.6.2 already, so you should be able to grab the patch without too much trouble. Regards, Steve -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Cannot install dahdi-linux on (old) PAE kernel.
On Fri, May 6, 2011 at 2:12 PM, Bob Beers bob.be...@gmail.com wrote:
On Fri, May 6, 2011 at 1:27 PM, Bob Beers bob.be...@gmail.com wrote:
Hi Steven,
Can you put the .spec file from dahdi-linux-kmod package up?
Nevermind, I got it. :-)
Looking at it now.
Not sure if this will work, but I'd try adding, before line 86:
#Workaround for PAE
%if %{paevar} == PAE
Provides: kmod-dahdi-linux
%endif
Can't actually test it myself, sorry.
- Bob
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Re: [asterisk-users] question on ways to activate voicemail light on polycom
On Fri, May 6, 2011 at 2:14 PM, Jerry Geis ge...@pagestation.com wrote: Is there a way in asterisk to Activate/Clear the blinking light on polycom phones indicating VM. Either from an AGI or some way in the dialplan? I want to be able to control this light for from my application. I have an AGI to do something similiar to VM and want to light /clear the light myself. Thanks, Jerry I don't think there is a way to do it natively inside of asterisk, but I control it from a shell script. The shell script parses the output of sip show peers, crafts an application/simple-message-summary SIP message and then uses netcat to send it to the appropriate IP address / port. -M -- _ -- 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] Supermicro X7SPE (Atom) as Asterisk server?
Has anyone used this board as an Asterisk server? http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=HIPMI=Y I'm mostly interested about the possible compatibility issues this board may have with the AEX800 card. -- _ -- 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] Supermicro X7SPE (Atom) as Asterisk server?
On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanian va...@arminco.com wrote: Has anyone used this board as an Asterisk server? http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=HIPMI=Y I'm mostly interested about the possible compatibility issues this board may have with the AEX800 card. Yes that is a great system and the built-in IPMI is a livesaver... if you are using a full size harddrive you need to apply some protection to the card in the case (the superserver 1U). They are close but not touching... -- ~~~ Andrew lathama Latham lath...@gmail.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] question on ways to activate voicemail light on polycom
The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Friday, May 06, 2011 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] question on ways to activate voicemail light on polycom Is there a way in asterisk to Activate/Clear the blinking light on polycom phones indicating VM. Either from an AGI or some way in the dialplan? I want to be able to control this light for from my application. I have an AGI to do something similiar to VM and want to light /clear the light myself. Thanks, Jerry Yes, use the MinivmMWI application. - Brad -- _ -- 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] Cannot install dahdi-linux on (old) PAE kernel.
On 05/06/2011 01:30 PM, Bob Beers wrote:
Not sure if this will work, but I'd try adding, before line 86:
#Workaround for PAE
%if %{paevar} == PAE
Provides: kmod-dahdi-linux
%endif
Can't actually test it myself, sorry.
- Bob
You'd probably want to modify the kmodtool that comes with it, to just always
provide kmod-dahdi-linux.
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
At 05:39 AM 5/6/2011, you wrote: Thanks for the feedback, Ira. It makes me very sad to hear what you say and I hope that we can get more resources from the community to assist in the process to make it more friendly. We want to get those bug reports. The one thing I hate to hear when I'm travelling at conferences is that oh, I known that bug for a long time but did not bother to report it. Apologies for your experience with the bug process. No worries, I'm not angry, and I'm weirdly good at finding bugs no one else can so I'm used to being ignored. Ira -- _ -- 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 latest branch safe for production ?
- Original Message - On Thu, 2011-05-05 at 14:13 +, satish patel wrote: Hi All, Just wondering is it safe to use asterisk 1.8 latest branch on production ? http://svn.asterisk.org/svn/asterisk/branches/1.8/ Revision 317100 -S We've been running 1.8.3.2 with the patch to fix the local channel issue (https://issues.asterisk.org/view.php?id=18818) For about a month in our test environment and it's been pretty stable. I would strongly advise that you run and version you wish to migrate to in a test environment for a good while as there are differences between 1.4 and 1.8 that are quite subtle and hard to pick up on (e.g. how to set outbound CLI correctly in CDR). Most of the big issues we found were due to our use of RealTime architecture. I get the impression that RealTime is not that widely used and therefore not that widely tested. To any development people out there, one we get these 1.8 servers into production I may well offer my services for testing with an emphasis on RealTime... Are you not seeing issues with *8 call pick up then ? -- Thanks, Phil -- _ -- 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] Blacklist with *30
Dear, when I dial *30 in order to get instructions to blacklist an extension, Idon't get the menu but I get a new dial tone. What happen please ??? What can I do to solve this ??? Thanks a lot, Alejandro -- _ -- 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] Tricky: Progress, Delay, DTMF / background calling
Hi,
has the following been done before respectively is it possible with
Asterisk? I searched the archives but couldn't locate anything.
1. Call to comes in via SIP.
2. Call is not answered yet but progress continues.
3. At the moment the call comes in something like this gets spawned in the
background:
Dial(SIP/123456@provider,,D(ww${EXTEN})
which should translate to:
Dial(SIP/123456@provider,,D(ww)
But even better would be take the ${EXTEN} and put some w's between them:
Dial(SIP/123456@provider,,D(ww5ww5ww5ww5)
4. After a pretermined amount of time since the call came in respectively
the Dial command was spawned in the background, e.g. 15 seconds,
Asterisk answers the call and the call legs are connected together.
So, with some fantasy commands, something like this:
_X.,1,Progress
_X.,2,DialBackground(SIP/123456@provider,,D(ww${EwwXwwTwwEwwN}),ANSWER-AND-CONNECT-LEGS(15)
I hope my request is not too cryptic. In short: I'd like to receive calls
to arbitrary extensions, but not answer them directly, only after a Dial
command has been spawned and a predetermined amount of time has passed
since the Dial command has been spawned / since the Dial command has
connected to 123456.
Possible?
I'm new to the list, hi! :)
Thank you!
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Re: [asterisk-users] Tricky: Progress, Delay, DTMF / background calling
On a second thought, I don't need the predetermined delay. I can probably
just set that with additional w's in the DialBackground command (which I
made up).
So rather something like:
_X.,1,Progress
_X.,2,DialBackground(SIP/123456@provider,,D(ww${EwwXwwTwwEwwN}ww))
_X.,3,ConnectLegs
Thanks again.
Hi,
has the following been done before respectively is it possible with
Asterisk? I searched the archives but couldn't locate anything.
1. Call to comes in via SIP.
2. Call is not answered yet but progress continues.
3. At the moment the call comes in something like this gets spawned in the
background:
Dial(SIP/123456@provider,,D(ww${EXTEN})
which should translate to:
Dial(SIP/123456@provider,,D(ww)
But even better would be take the ${EXTEN} and put some w's between them:
Dial(SIP/123456@provider,,D(ww5ww5ww5ww5)
4. After a pretermined amount of time since the call came in respectively
the Dial command was spawned in the background, e.g. 15 seconds,
Asterisk answers the call and the call legs are connected together.
So, with some fantasy commands, something like this:
_X.,1,Progress
_X.,2,DialBackground(SIP/123456@provider,,D(ww${EwwXwwTwwEwwN}),ANSWER-AND-CONNECT-LEGS(15)
I hope my request is not too cryptic. In short: I'd like to receive calls
to arbitrary extensions, but not answer them directly, only after a Dial
command has been spawned and a predetermined amount of time has passed
since the Dial command has been spawned / since the Dial command has
connected to 123456.
Possible?
I'm new to the list, hi! :)
Thank you!
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Re: [asterisk-users] Occasional call from asterisk
Hi Brian,
Did you find a solution to your problem? or at least got a working dial-plan
for it? I have the same problem again as well and want to know what to do
with the dial-plan to off-set the effect at least since Telco says it's not
their issue.
Regards,
Bruce
On Thu, Apr 7, 2011 at 5:53 PM, Brian Henning bhenn...@pineinst.com wrote:
Hi,
Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight?
Here's
relevant snippets from my extensions.conf and Master.csv log:
This line shows up in Master.csv:
,,1-NOANSWER,inbound,,DAHDI/1-1,SIP/505-0150,Dial,SIP/5
01SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr,2011-04-07
21:37:05,2011-04-07 21:37:16,2011-04-07
21:37:21,16,5,ANSWERED,DOCUMENTATION,1302212225.444,
Here's [inbound] from extensions.conf:
[inbound]
exten = s,1,Answer
exten = s,n,Ringing
exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
exten = s,n,Goto(1-${DIALSTATUS},1)
exten = 1-ANSWER,1,Hangup
exten =
_1-.,1,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
exten = _1-.,n,Goto(2-${DIALSTATUS},1)
exten = 2-ANSWER,1,Hangup
exten = _2-.,1,Voicemail(499@default,u)
exten = _2-.,2,Hangup
The idea is that first 504 and 506 ring, then if neither of them answer,
everyone rings. Works great most of the time.
I have a hunch that maybe this happens if the inbound caller hangs up while
the first Dial() is ringing, but I would've expected to see the first Dial
(to 504 and 506) show up in the Master.csv log, and it's not there. (The
preceding line of the log is a call from almost an hour earlier). In that
case though I'd expect to see 1-CANCEL in the log instead. Perhaps if
the
caller happens to hang up right between the two Dial() commands?..
As an aside, the Set(CALLERID...) bit doesn't work. The idea was to
prepend
a 9 so that a SIP user could use the redial feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line). Unfortunately the 9 does not get prepended.
Thanks in advance for any and all advice!
~Brian
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USA
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||phone: 919.782.8320
fax: 919.782.8323
email: bhenn...@pineinst.com
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Re: [asterisk-users] Occasional call from asterisk
Telco always says it is not their issue.
This is all over google, did you even check? Did you check your options in
chan_dahdi.conf?
hanguponpolarityswitch=yes
I am not sure if that is your problem but it would be helpful to list the
things you have found, tested, and ruled out.
As for prepending a 9 for redial, I would say doing it in the [outbound]
dial context would be best practice.
For my installations, I have eliminated the need to Dial 9 for an outside
line That goes back to the key systems where 9 got you an outside line.
I have also eliminated the need to dial 1 as well. A good dialplan makes
these legacy, I still leave them there to avoid confusion.
For some clients that use TDM and VoIP, I may make 8 + number go over VoIP
and 9 + whatever go over TDM.
Default without the 8 or 9 is to go out over TDM or whatever the customer
wants, or TDM if they seem lost.
I don't give them too many decisions to make, just educate them on the
options programmed into the system.
Last thing, your dialplan looks too over engineered. How about this and
fixing your callerID syntax?
[inbound]
exten = s,1,Answer
exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
exten = s,n,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
exten = s,n,Voicemail(499@default,u)
exten = s,n,Hangup
Thanks,
Steve Totaro
On Fri, May 6, 2011 at 10:54 PM, Bruce B bruceb...@gmail.com wrote:
Hi Brian,
Did you find a solution to your problem? or at least got a working
dial-plan for it? I have the same problem again as well and want to know
what to do with the dial-plan to off-set the effect at least since Telco
says it's not their issue.
Regards,
Bruce
On Thu, Apr 7, 2011 at 5:53 PM, Brian Henning bhenn...@pineinst.comwrote:
Hi,
Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight?
Here's
relevant snippets from my extensions.conf and Master.csv log:
This line shows up in Master.csv:
,,1-NOANSWER,inbound,,DAHDI/1-1,SIP/505-0150,Dial,SIP/5
01SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr,2011-04-07
21:37:05,2011-04-07 21:37:16,2011-04-07
21:37:21,16,5,ANSWERED,DOCUMENTATION,1302212225.444,
Here's [inbound] from extensions.conf:
[inbound]
exten = s,1,Answer
exten = s,n,Ringing
exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
exten = s,n,Goto(1-${DIALSTATUS},1)
exten = 1-ANSWER,1,Hangup
exten =
_1-.,1,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
exten = _1-.,n,Goto(2-${DIALSTATUS},1)
exten = 2-ANSWER,1,Hangup
exten = _2-.,1,Voicemail(499@default,u)
exten = _2-.,2,Hangup
The idea is that first 504 and 506 ring, then if neither of them answer,
everyone rings. Works great most of the time.
I have a hunch that maybe this happens if the inbound caller hangs up
while
the first Dial() is ringing, but I would've expected to see the first Dial
(to 504 and 506) show up in the Master.csv log, and it's not there. (The
preceding line of the log is a call from almost an hour earlier). In that
case though I'd expect to see 1-CANCEL in the log instead. Perhaps if
the
caller happens to hang up right between the two Dial() commands?..
As an aside, the Set(CALLERID...) bit doesn't work. The idea was to
prepend
a 9 so that a SIP user could use the redial feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line). Unfortunately the 9 does not get prepended.
Thanks in advance for any and all advice!
~Brian
--
Brian Henning, Software Engineer
/\Pine Research Instrumentation
//\\ 5908 Triangle Drive
///\\\ Raleigh, NC 27617
USA
||
||phone: 919.782.8320
fax: 919.782.8323
email: bhenn...@pineinst.com
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