[asterisk-users] Bufferbloat! Friday on VUC @ 12 Noon EST
Hi all, What is Bufferbloat? http://gettys.wordpress.com/bufferbloat-faq/ Maybe this kind of discussion will bring out the John Todds of this world, I can only hope and dream: Bufferbloat: http://www.voipusersconference.org/2011/bufferbloat/ Call in and talk to Jim Gettys, who co-developed X Window System and was a part of HTTP/1.1 - this is someone we'll all be proud to meet, and you can do this by calling via SIP, Skype, widget or, gasp PSTN Friday January 28th at 12 Noon EST: sip:200...@login.zipdx.com (g722 if you got it, otherwise g711) thanks to zipdx.com skype:vuc.me thanks to Tim Panton and PhoneFromHere.com PSTN: (567) 252-2286 thanks to Alex Graham Bell iNum: +883 5100 123 94882 Text on IRC #vuc on Freenode.net - http://vuc.me/irc If in doubt about the time in your zone, look here: http://vuc.me/next Hear you there... /r -- _ -- 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] Recommended Windows client to display CID?
On Thu, 27 Jan 2011 08:46:11 +0100, Magnus Persson mag...@westel.se wrote: If you want someting really light weight there is always the old winpopup protocoll. Thanks for the tip. It's a nice alternative, although I'd like an app that keeps a list of pop-ups, in case the user was away and would like to see who called during their leave. -- _ -- 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] Recommended Windows client to display CID?
On Wed, 26 Jan 2011 14:52:59 +0100, Gilles codecompl...@free.fr wrote: Are there open-source solutions you could recommend? I had another idea: It'd be cool if the application could either just display CID information, or also search Outlook for a matching Contact and open the relevant page so that the user can review/add information for that person. Poor man's CRM :-) -- _ -- 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] Recommended Windows client to display CID?
On Thu, Jan 27, 2011 at 11:32:03AM +0100, Gilles wrote: On Thu, 27 Jan 2011 08:46:11 +0100, Magnus Persson mag...@westel.se wrote: If you want someting really light weight there is always the old winpopup protocoll. Thanks for the tip. It's a nice alternative, although I'd like an app that keeps a list of pop-ups, in case the user was away and would like to see who called during their leave. An instant-messaging client, as suggested before. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended Windows client to display CID?
On Thu, 27 Jan 2011 12:49:06 +0200, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: An instant-messaging client, as suggested before. Right. Just a reply to Magnus' suggestion. -- _ -- 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] Recommended Windows client to display CID?
On Thursday 27 Jan 2011, Gilles wrote: I had another idea: It'd be cool if the application could either just display CID information, or also search Outlook for a matching Contact and open the relevant page so that the user can review/add information for that person. Poor man's CRM :-) . And this is where you hit the proprietary wall. TTBOMK there isn't a published API to do this sort of thing in Outlook. There is almost certainly a secret API that only Microsoft know about; but if and when that hidden API gets leaked, you can bet it will be used for malicious purposes by someone. You would do much better in the long run to look at replacing Outlook with some Open Source alternative -- and sooner, rather than later. -- 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] Recommended Windows client to display CID?
http://code.google.com/p/outcall/ ? -- _ -- 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] Recommended Windows client to display CID?
On Thu, 27 Jan 2011 11:35:05 +, A J Stiles asterisk_l...@earthshod.co.uk wrote: You would do much better in the long run to look at replacing Outlook with some Open Source alternative -- and sooner, rather than later. But then, Outlook is pretty much what every office worker uses. Looks like it's possible to access Outlook through COM/Automation: http://stackoverflow.com/search?q=outlook+look+up+contacts If someone's already connected Asterisk and Outlook in some way, I'm interested in any feedback. -- _ -- 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] Recommended Windows client to display CID?
Not to be redundant, but http://code.google.com/p/outcall/ On Thu, Jan 27, 2011 at 6:23 AM, Gilles codecompl...@free.fr wrote: On Thu, 27 Jan 2011 11:35:05 +, A J Stiles asterisk_l...@earthshod.co.uk wrote: You would do much better in the long run to look at replacing Outlook with some Open Source alternative -- and sooner, rather than later. But then, Outlook is pretty much what every office worker uses. Looks like it's possible to access Outlook through COM/Automation: http://stackoverflow.com/search?q=outlook+look+up+contacts If someone's already connected Asterisk and Outlook in some way, I'm interested in any feedback. -- _ -- 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] Callback when available
Hi All, I would like to implement a call-back option when called user is busy. Consider this scenario: 1. A caller is calling a number which is busy on another call. 2. The system will prompt the caller (press 3 to be called back etc.) to be called back when called user is available. 3. Caller hangs up. problem: how to monitor called user status after calling user has hanged-up? Dialling plan has terminated at this point... 4. The called user terminates his previous call. 5. The system calls the caller and prompts him to wait for connection. 6. The system calls the called user and bridges the call upon pick up. I can use any version of Asterisk as required. Any opinions and ideas would be appreciated. Kind Regards, Harel This electronic message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the named addressee you should not disseminate or distribute a copy of this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. EasyCall Ltd 11 Cornwall's Parade P.O.Box 1488 Gibraltar. Office : +350 20077889 Fax : +350 20076727. www.easycall.gi supp...@easycall.gi -- _ -- 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] Recommended Windows client to display CID?
On Thu, 27 Jan 2011 06:24:53 -0600, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: http://code.google.com/p/outcall/ Thanks a lot. I'll check it out. -- _ -- 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] Callback when available
Look into Call Completion Supplementary Services for Asterisk 1.8 https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+%28CCSS%29 On Thu, Jan 27, 2011 at 6:48 AM, Harel Cohen ha...@easycall.gi wrote: Hi All, I would like to implement a call-back option when called user is busy. Consider this scenario: 1. A caller is calling a number which is busy on another call. 2. The system will prompt the caller (“press 3 to be called back” etc.) to be called back when called user is available. 3. Caller hangs up. problem: how to monitor called user status after calling user has hanged-up? Dialling plan has terminated at this point… 4. The called user terminates his previous call. 5. The system calls the caller and prompts him to wait for connection. 6. The system calls the called user and bridges the call upon pick up. I can use any version of Asterisk as required. Any opinions and ideas would be appreciated. Kind Regards, Harel This electronic message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the named addressee you should not disseminate or distribute a copy of this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. EasyCall Ltd 11 Cornwall’s Parade P.O.Box 1488 Gibraltar. Office : +350 20077889 Fax : +350 20076727. www.easycall.gi supp...@easycall.gi -- _ -- 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] res_fax
Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? Thanks Bryant -- _ -- 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] res_fax
On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? My schedule is pretty full today, but I will take another look over the code and see what might be going on. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Thursday, January 27, 2011 10:31 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? My schedule is pretty full today, but I will take another look over the code and see what might be going on. -- Kevin Thanks I am continuing with other parts of my fax code as well for now. I will test any changes as you are able to make them. Bryant -- _ -- 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] Anybody ever see this before?
[Jan 27 10:59:56] ERROR[3382]: chan_dahdi.c:12405 dahdi_pri_error: Should have only transmitted 0 frames! [Jan 27 10:59:56] ERROR[3382]: chan_dahdi.c:12405 dahdi_pri_error: Should have only transmitted 0 frames! I just saw it fly across my CLI. -- _ -- 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] Recommended Windows client to display CID?
HI , Please give idea for Multi tenant with Trixbox or elastix. Thanks Amardeep Rana -- _ -- 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] Multi-Tenant
HI , Please give idea for Multi tenant with Trixbox or elastix. Thanks Amardeep Rana -- _ -- 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] Queue - agent auto-answer
Hi, Is there any way to have queue member interface answer automatically? Basically when agentA is called, his phone picks up with no intervention from his part? (assuming of course he's available and not on the phone, and not paused). I already manage this with the Page application (using exten = s,n,SIPAddHeader(Alert-Info: Ring Answer)) and Polycom phones. But how do I do this for calls that are handled by the Queue application? Mike -- _ -- 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] Queue - agent auto-answer
I believe all you need to do is to do the same thing just before running the Queue command...checking On Thu, Jan 27, 2011 at 10:45 AM, Mike l...@net-wall.com wrote: Hi, Is there any way to have queue member interface answer automatically? Basically when agentA is called, his phone picks up with no intervention from his part? (assuming of course he’s available and not on the phone, and not paused). I already manage this with the Page application (using exten = s,n,SIPAddHeader(Alert-Info: Ring Answer)) and Polycom phones. But how do I do this for calls that are handled by the Queue application? Mike -- _ -- 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] Queue - agent auto-answer
We do something similar to this by logging a Local channel (eg: Local/1234@AgentContext) into the queue that passes each call through a few lines of dialplan code before going to the SIP extension. Jordan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: 27 January 2011 16:46 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Queue - agent auto-answer Hi, Is there any way to have queue member interface answer automatically? Basically when agentA is called, his phone picks up with no intervention from his part? (assuming of course he's available and not on the phone, and not paused). I already manage this with the Page application (using exten = s,n,SIPAddHeader(Alert-Info: Ring Answer)) and Polycom phones. But how do I do this for calls that are handled by the Queue application? Mike -- _ -- 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] Queue - agent auto-answer
Ah, there we go, what you'll need to do is some magic with Local channelscheck out FreePBX's code, it's a little more than I wish to copy/paste On Thu, Jan 27, 2011 at 10:55 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: I believe all you need to do is to do the same thing just before running the Queue command...checking On Thu, Jan 27, 2011 at 10:45 AM, Mike l...@net-wall.com wrote: Hi, Is there any way to have queue member interface answer automatically? Basically when agentA is called, his phone picks up with no intervention from his part? (assuming of course he’s available and not on the phone, and not paused). I already manage this with the Page application (using exten = s,n,SIPAddHeader(Alert-Info: Ring Answer)) and Polycom phones. But how do I do this for calls that are handled by the Queue application? Mike -- _ -- 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] Multi-Tenant
On 11-01-27 11:41 AM, Amardeep Rana wrote: Please give idea for Multi tenant with Trixbox or elastix. http://astbook.asteriskdocs.org -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) 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
Re: [asterisk-users] Queue - agent auto-answer
Is there any way to have queue member interface answer automatically? Basically when agentA is called, his phone picks up with no intervention from his part? (assuming of course hes available and not on the phone, and not paused). I already manage this with the Page application (using exten = s,n,SIPAddHeader(Alert-Info: Ring Answer)) and Polycom phones. But how do I do this for calls that are handled by the Queue application? I believe all you need to do is to do the same thing just before running the Queue command...checking And that is indeed correct. I had tried that of course, but by calling line1 of my phone with line2 of the same phone. What did I expect... So after reading your email I had a facepalm moment and tried it properly, and it works. Unfortunately it seems that this can be done per queue, but not per agent, but that'll work for my purposes. Mike -- _ -- 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] Queue - agent auto-answer
Yeah, if you want per agent, you'll need to use local channels for the agent interface definition, then in the callagent context, you'll need to parse the agent's extension and determine if that agent is supposed to have autoanswer or not... func_odbc and a little dialplan logic should work nicely :) (Of course, I'm biased, I work almost exclusively with database driven solutions :P ) Cheers mate! On Thu, Jan 27, 2011 at 1:06 PM, Mike l...@net-wall.com wrote: Is there any way to have queue member interface answer automatically? Basically when agentA is called, his phone picks up with no intervention from his part? (assuming of course he’s available and not on the phone, and not paused). I already manage this with the Page application (using exten = s,n,SIPAddHeader(Alert-Info: Ring Answer)) and Polycom phones. But how do I do this for calls that are handled by the Queue application? I believe all you need to do is to do the same thing just before running the Queue command...checking And that is indeed correct. I had tried that of course, but by calling line1 of my phone with line2 of the same phone. What did I expect... So after reading your email I had a facepalm moment and tried it properly, and it works. Unfortunately it seems that this can be done per queue, but not per agent, but that'll work for my purposes. Mike -- _ -- 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] Multi-Tenant
Oh man, I'm sorry, but I laughed so hard at that response, I think I peed a little :P To the original poster, Mr Belanger is most definitely being VERY kind compared to what some people might have responded with A little effort (and showing that you have put in that effort) goes a long way in an OSS users' mailing list On Thu, Jan 27, 2011 at 11:59 AM, Paul Belanger pabelan...@digium.com wrote: On 11-01-27 11:41 AM, Amardeep Rana wrote: Please give idea for Multi tenant with Trixbox or elastix. http://astbook.asteriskdocs.org -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) 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 -- _ -- 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] Queue - agent auto-answer
Yeah, if you want per agent, you'll need to use local channels for the agent interface definition, then in the callagent context, you'll need to parse the agent's extension and determine if that agent is supposed to have autoanswer or not... func_odbc and a little dialplan logic should work nicely :) (Of course, I'm biased, I work almost exclusively with database driven solutions :P ) So do I, but this might be more trouble than it`s worth in this particular case. But I`ll keep this knowledge on hand, might be useful one day. Mike -- _ -- 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] OT: VoIP Users Conf Feb 4 with LifeSize
Hi All, My appologies for the off-topic post, but I thought it would still be of interest to this list. If you've been reading our site at http://vuc.me you've no doubt seen that we have the video call with LifeSize scheduled for Feb 4th. What? You didn't know? Here's t scoop: http://www.voipusersconference.org/2011/hd-video-conferencing-with-lifes ize/ We have a number of Logitech C910 HD webcams to give away this coming week. These should help some people join the video conference. Additionally, there are a few things to be considered: 1. If you want to win a webcam you must register for the call with LifeSize. They have put up a page on their web site specifically for the VUC. That url is: http://www.lifesize.com/voip-register Registration will be open until 5pm CST Friday, Jan 28th. This allows us some time to see that the webcams are distributed before the call the following week. 2. If you want to join the video portion of the call you must use a LifeSize client. Unless you have LifeSize hardware already that means the LifeSize Desktop for Windows 2.0. This software is a free download from their web site. The trial download runs for 30 days before it expires. 3. To participate in the video portion of the call you will need around 1 mbps of available bandwidth in each direction. That will allow for 720p HD video. 4. No. There is no Mac or Linux version of the client software. Randy has run the Windows version on a Mac inside Parallels. It worked surprising well, albeit at VGA resolution. 5. We don't have any access to the LifeSize video bridge prior to the call. That means that we cannot put any effort into experimenting with other soft clients. Hey, it's a LifeSize call...they're committing resources to allow us this exercise. We should be respectful and let them have the stage for the hour. Everyone will be able to watch the web stream in any case. Be certain to register for the call with LifeSize by 5pm CST this Friday! Your odds on winning a camera are pretty good Everyone on the irc channel at the appropriate time during the Feb 4 call will be eligible to win the Logitech Harmony Universal remote control. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- 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] res_fax
On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? Revision 304599 should fix this (and I also changed the option letter from 'n' to 'F' since it really means 'force audio'). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Thursday, January 27, 2011 3:08 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? Revision 304599 should fix this (and I also changed the option letter from 'n' to 'F' since it really means 'force audio'). - Kevin I will rebuild and test in a bit. Thanks Bryant -- _ -- 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] A1200P comments?
Hi all, Does anyone have any good/bad comments on the A1200P 12-port fxo/fxs card from OpenVox? I'll be using one to with 8-12 fxo interfaces. The cards will be plugging into a cable-modem / phone adapter. We weren't able to port the numbers, so we're going to use the existing PSTN connection and replace all of the office phones. With these short distances, will I need to worry about echo? Do these devices have echo cancellation? TIA, -- Take care and have fun, Mike Diehl. -- _ -- 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] chan_sip bug? (Asterisk 1.4)
Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk stop working after the upgrade. Here is the sip debug: --- --- SIP read from 208.65.xxx.xxx:5060 --- INVITE sip:1778xxx@10.11.22.77:5060 SIP/2.0 Via: SIP/2.0/UDP 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-d9175178645e9146-1---d8754z-;rport Via: SIP/2.0/UDP 208.65.xxx.xxx:5061;branch=z9hG4bK-uhhmj2ir4ew6cn4p;rport=5061 Max-Forwards: 69 Record-Route: sip:208.65.xxx.xxx;lr Contact: Anonymoussip:208.65.xxx.xxx:5061 To: sip:1778...@208.65.xxx.xxx:5060 From: sip:604...@208.65.xxx.xxx:5060;tag=ixpa27sbhn3inu5x.o Call-ID: 550d3...@208.72.xxx.xxx~o CSeq: 819 INVITE Expires: 300 Content-Disposition: session Content-Type: application/sdp User-Agent: Sippy cisco-GUID: 2851810672-711266784-2763915291-559912524 h323-conf-id: 2851810672-711266784-2763915291-559912524 Content-Length: 109 v=0 o=Sippy 223452192 0 IN IP4 74.205.216.77 s=- t=0 0 m=audio 33830 RTP/AVP 0 c=IN IP4 74.205.216.777 - --- (17 headers 6 lines) --- Sending to 208.65.xxx.xxx : 5060 (NAT) Using INVITE request as basis request - 550d3...@208.72.xxx.xxx~o Found peer 'FreePhoneLine' Found RTP audio format 0 [2011-01-27 14:35:18] WARNING[2911]: chan_sip.c:5948 process_sdp_c: Unable to lookup RTP Audio host in c= line, 'IN IP4 74.205.216.777' [2011-01-27 14:35:18] WARNING[2911]: chan_sip.c:5741 process_sdp: Insufficient information in SDP (c=)... --- It seems in the SIP INVITE, the IP 74.205.216.77 somehow changed to 74.205.216.777. I am not sure this is a bug of Asterisk or not. Regards, Jian -- _ -- 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] chan_sip bug? (Asterisk 1.4)
On Thu, 27 Jan 2011 14:52:06 -0800 Jian Gao jian@sjgeophysics.com wrote: Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk stop working after the upgrade. Here is the sip debug: --- --- SIP read from 208.65.xxx.xxx:5060 --- That packet is coming from the other end (Sippy). The problem is probably there. However, it could be that the networking routines in Asterisk have added a 7 at the end. You could compare a tcpdump of that packet to what Asterisk sees. If the tcpdump shows .777 then the problem is in Sippy. If it shows .77 then the problem is in Asterisk. INVITE sip:1778xxx@10.11.22.77:5060 SIP/2.0 Via: SIP/2.0/UDP 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-d9175178645e9146-1---d8754z-;rport Via: SIP/2.0/UDP 208.65.xxx.xxx:5061;branch=z9hG4bK-uhhmj2ir4ew6cn4p;rport=5061 Max-Forwards: 69 Record-Route: sip:208.65.xxx.xxx;lr Contact: Anonymoussip:208.65.xxx.xxx:5061 To: sip:1778...@208.65.xxx.xxx:5060 From: sip:604...@208.65.xxx.xxx:5060;tag=ixpa27sbhn3inu5x.o Call-ID: 550d3...@208.72.xxx.xxx~o CSeq: 819 INVITE Expires: 300 Content-Disposition: session Content-Type: application/sdp User-Agent: Sippy cisco-GUID: 2851810672-711266784-2763915291-559912524 h323-conf-id: 2851810672-711266784-2763915291-559912524 Content-Length: 109 v=0 o=Sippy 223452192 0 IN IP4 74.205.216.77 s=- t=0 0 m=audio 33830 RTP/AVP 0 c=IN IP4 74.205.216.777 - --- (17 headers 6 lines) --- Sending to 208.65.xxx.xxx : 5060 (NAT) Using INVITE request as basis request - 550d3...@208.72.xxx.xxx~o Found peer 'FreePhoneLine' Found RTP audio format 0 [2011-01-27 14:35:18] WARNING[2911]: chan_sip.c:5948 process_sdp_c: Unable to lookup RTP Audio host in c= line, 'IN IP4 74.205.216.777' [2011-01-27 14:35:18] WARNING[2911]: chan_sip.c:5741 process_sdp: Insufficient information in SDP (c=)... --- It seems in the SIP INVITE, the IP 74.205.216.77 somehow changed to 74.205.216.777. I am not sure this is a bug of Asterisk or not. -- _ -- 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] RTP keepalive doesn't work
Hey guys, I'm using asterisk 1.6.2.13 and have an endpoint which uses silence suppression which I can't turn off. I've set rtpkeepalive=10 in sip.conf [general], as well as under the peer details for our sip provider but it doesn't seem to do anything. Rtp debug shows that we are receiving RTP from the SIP provider, and forwarding it to the end point, but no RTP packets are sent back to the provider (ie. No keep alives). I did find a bug report of this exact issue, but it was closed with the message to ask the mailing list... Any ideas? -- _ -- 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] A1200P comments?
Hi Mike, I have used the A1200P without hardware echo cancelation and didn't have any major issues. The one problem I had was that caller ID simply would not work on the A1200P, it was fine on the A400P however. This was a year ago though so things may have changed a little. Regards, Ryan. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Friday, 28 January 2011 7:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; mdi...@diehlnet.com Subject: [asterisk-users] A1200P comments? Hi all, Does anyone have any good/bad comments on the A1200P 12-port fxo/fxs card from OpenVox? I'll be using one to with 8-12 fxo interfaces. The cards will be plugging into a cable-modem / phone adapter. We weren't able to port the numbers, so we're going to use the existing PSTN connection and replace all of the office phones. With these short distances, will I need to worry about echo? Do these devices have echo cancellation? TIA, Take care and have fun, Mike Diehl. -- _ -- 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] RTP keepalive doesn't work
So, I've done some more testing and got some more info. I have one endpoint that does silence suppression and one that doesn't. When the silence suppressing endpoint stops sending RTP, asterisk stops sending RTP to the other endpoint. I have disabled directmedia and directrtpsetup and it made no difference. I have even forced one endpoint to use GSM and the other to use ULAW (forcing asterisk to re encode everything) and asterisk STILL stops sending RTP when the endpoint does... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Tucker Sent: Friday, 28 January 2011 11:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion (asterisk-users@lists.digium.com)' Subject: [asterisk-users] RTP keepalive doesn't work Hey guys, I'm using asterisk 1.6.2.13 and have an endpoint which uses silence suppression which I can't turn off. I've set rtpkeepalive=10 in sip.conf [general], as well as under the peer details for our sip provider but it doesn't seem to do anything. Rtp debug shows that we are receiving RTP from the SIP provider, and forwarding it to the end point, but no RTP packets are sent back to the provider (ie. No keep alives). I did find a bug report of this exact issue, but it was closed with the message to ask the mailing list... Any ideas? -- _ -- 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] SendFAX dialplan example
Hi! I am playing with SendFAX but cant really figure out how it is working. I have a “fax” /var/spool/asterisk/tmp/fax.tiff that i would like to send to a “physical” fax at numer 0317998901. Can some1 write me a simple dialplan that just “grab” the file and send it to 0317998901? /Magnus-- _ -- 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
