[asterisk-users] How to kick/mute using ConfBridge application
Hi All, We are currently evaluating the confbridge application while we prepare to upgrade our environment to asterisk v1.6.2.x. We have run in to two issues using it to kick/mute participants in a bridge and would like to ask for the experience of others running the application for any work-arounds. Firstly for kicking participants, would it be possible to use the softhangup application on a channel to effectively kick a participant from a bridge? Secondly, is it possible to mute a participant in the bridge using the AMI or a CLI. Any tips/suggestions would be greatly appreciated. Thanks Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial with MOH
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{\*\htmltag84 span style='font-size:10.0pt;font-family:Verdana,sans-serif;\par color:black'}\htmlrtf {\htmlrtf0 when dialing with m option the MOH will play until the B user answers,
{\*\htmltag4 \par }\htmlrtf \htmlrtf0 but in case of voicemail the user will not hear the prompt until received 200
{\*\htmltag4 \par }\htmlrtf \htmlrtf0 OK.
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{\*\htmltag4 \par }which file i can edit or what should i change to let the MOH stops when it received
{\*\htmltag4 \par }\htmlrtf \htmlrtf0 183 from trunk .
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{\*\htmltag84 span style='font-size:10.0pt;font-family:Verdana,sans-serif;\par color:black'}\htmlrtf {\htmlrtf0 Regards
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Re: [asterisk-users] own Caller ID
On Wed, 2010-06-09 at 19:43 +, Edwin Quijada wrote: Just is PRI line you can do it.. No, not so. I both have some PRI and BRI lines. All of them have a main-number, and some additional numbers Depending on what contract you have with your ISDN-provider the amount of those number can vary. On my BRI lines i have 10 (1 + 9 additional) numbers and on my PRI lines i have 100 numbers (1+99) But it could be more r less (even just 1) With our provider (KPN) you can set your caller is to anything that is assigned to you. If you ommit it, or go beyond your limits, you automagically get the main number assigned to it. hw Date: Tue, 8 Jun 2010 12:44:07 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] own Caller ID On Tue, 8 Jun 2010, taimur hasan wrote: I want to use my own caller id, instead of the caller id of PSTN line, for the outbound calls through DAHDI channel. Is there any way ?? It depends on your technology (POTS, PRI, etc) and your provider. Tell your provider you want to set the outgoing caller ID and see what their response is. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 __ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 registration
Hi all, I think i understand the problem, actually I have two asterisk server. In the extension.conf file on one server I have added exten = 3923903,1,GOTO(s,1,3923903.conf) which reads the corresponding conf file when ever the extension no. through PSTN is called and learns the location of inbound.php which contains the IVR script to be executed. Now what i want is that through this inbound.php , i should be able to call another asterisk server, where I have also configured twinkle as a softphone. The problems: --I am not able to register this softphone on the previous asterisk server as user 2001, though i modified the server's extension and sip file to include the user 2001 under [phones] context. ---cli chan_sip.c:15839 handle_request_register: Registration from 'user1 sip:2...@172.26.48.208 sip%3a2...@172.26.48.208' failed for '172.26.48.62' - No matching peer found shows this error upon registration.. --at my server it shows 3 unmonitored peers, but the previous server doesn't show any peers on sip show peers..though i have added all three users in sip file, and yes reloaded the dial plan. WARNING[9041]: chan_sip.c:2984 create_addr: No such host: 2001 [Jun 10 12:26:46] WARNING[9041]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) is the error when i do not give ip..assuming 2001 to be registered at the server. when i give the ip of my server.. chan_sip.c:20039 handle_request_invite: Call from '' to extension '2001' rejected because extension not found. is the error..call actually lands up on asterisk server but it shows the above error and ofcourse can not be recieved with softphone. Please help me out in this regard. Though above details may be confusing..I have tried to briefly write in case any more explanation needed, please mail me.I am stuck in this so please help. Thanks in advance Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- 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] Out of Office
Hi, It isn't a problem with the list. And it is not 'mine'. It is a problem with your software. I am just one of the thousands of people it is annoying! Perhaps your IT staff could help fix it? CCing the list so everyone is aware of your wonderful customer service. ;) S On 10 Jun 2010, at 11:27, Mary wrote: He is away with no cell phone or e-mail so either be helpful and tell me how to change (step by step) this to take him off your list or write a progrm for your list to fix this so it doesnt happen! Mary Shubert Accessgate.net, Inc. Suite 106 8600 Commodity Circle Orlando, FL 32819 m...@accessgate.net Office Toll Free: (888) 227-9337 Fax: (407) 352-2717 From: Steve Howes steve-li...@geekinter.net Sent: Thursday, June 10, 2010 4:26 AM To: d...@accessgate.net, m...@accessgate.net, jsny...@accessgate.ne Subject: Re: [asterisk-users] Out of Office On 10 Jun 2010, at 06:20, d...@accessgate.net wrote: I will be out of the office starting Wed June 9th and returning Wed June 16th. Please contact Mary at m...@accessgate.net cell 407-267-1463 or Jonathan at jsny...@accessgate.net cell 407-267-0056 or call our main number 888-227-9337. Several thousand people DO NOT need spamming with this daily because you can't configure your mail client/server to reply to a mailing list. Please FIX THIS. S -- _ -- 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] Loud Noise when trying to call through PSTN.
Hi, I am using Asterisknow 1.5. And TDM400P card for interfacing with PSTN line. This setup was working without any problem. But now it is showing issues. When I try to call through PSTN, there is a continuous large noise is hearing from the SIP phone. And can't make the call. When I try to call the PSTN number from mobile there is only engaged tone is hearing. And also the Asterisk server is hanging frequently with lighting all the LEDs in the TDM400p cards. The SIP to SIP calls are working fine. Is this a hardware issue? The TDM400P is under warranty. Any help would be highly appreciated. Thanks, Arun S -- _ -- 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] Group call limit
Hello list, is it possible to group some peers and limit their overall call limit? Ex: 4 peers can make max 2 concurrent calls. Thanks in advance, Alex -- _ -- 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] Out of Office
Shouldn't a moderator block emails from this email address, maybe temporarily? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-10 6:41 AM, Steve Howes steve-li...@geekinter.net wrote: Hi, It isn't a problem with the list. And it is not 'mine'. It is a problem with your software. I am just one of the thousands of people it is annoying! Perhaps your IT staff could help fix it? CCing the list so everyone is aware of your wonderful customer service. ;) S On 10 Jun 2010, at 11:27, Mary wrote: He is away with no cell phone or e-mail so either be helpful and tell me how to change (step by step) this to take him off your list or write a progrm for your list to fix this so it doesnt happen! Mary Shubert Accessgate.net, Inc. Suite 106 8600 Commodity Circle Orlando, FL 32819 m...@accessgate.net Office Toll Free: (888) 227-9337 Fax: (407) 352-2717 From: Steve Howes steve-li...@geekinter.net Sent: Thursday, June 10, 2010 4:26 AM To: d...@accessgate.net, m...@accessgate.net, jsny...@accessgate.ne Subject: Re: [asterisk-users] Out of Office On 10 Jun 2010, at 06:20, d...@accessgate.net wrote: I will be out of the office star... Several thousand people DO NOT need spamming with this daily because you can't configure your mail client/server to reply to a mailing list. Please FIX THIS. S -- _ -- Bandwidth and ... -- _ -- 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] warning : sip_xmit
I'm getting a lot of these on the CLI : [Jun 10 13:41:36] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:37] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:38] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:39] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:40] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:50] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b393130 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:51] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b393130 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:52] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b393130 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:53] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b393130 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:54] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b393130 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:42:04] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b33dad0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:42:05] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b33dad0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:42:06] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b33dad0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:42:07] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b33dad0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted What can I do to stop this ?? What I usually do is restart Asterisk. After 5 to 8 restarts, it goes away... This can not be good practise. Using Asterisk 1.4.30 and sip realtime. Jonas. -- _ -- 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] Group call limit
On 10/06/10 11:56, Alexandru Oniciuc wrote: Hello list, is it possible to group some peers and limit their overall call limit? Ex: 4 peers can make max 2 concurrent calls. Thanks in advance, Alex Hi you can use call-limit in the sip.conf at a peer level Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ring + Music on Hold in the same call
Hi list, is there a way to achieve in asterisk (version 1.4.x) the behavior described below? * a caller place a call to an extension, and I want the caller hears the extension ringing for some seconds, and then hears the music on hold (or a courtesy message) _in the same call;_ * the called extension must continue to ring until answered. With the m(...) option in the Dial command (like the example below) asterisk provides only music on hold while the phone rings. exten = s,n,Dial(SIP/,30,m(default)) I can not use queues because the requirements is to have 1 call and not a lot of calls. Thanks in advance, Matteo -- _ -- 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] Ring + Music on Hold in the same call
Here is one way to do it (works in 1.4.22-1.4.30 at least) exten = s,n,Dial(SIP/,10) exten = s,n,Dial(SIP/,90,m(default)) This snippet will ring for 10 seconds with Ringing, then ring for 90 seconds or until answered with MOH. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana Sent: Thursday, June 10, 2010 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ring + Music on Hold in the same call Hi list, is there a way to achieve in asterisk (version 1.4.x) the behavior described below? * a caller place a call to an extension, and I want the caller hears the extension ringing for some seconds, and then hears the music on hold (or a courtesy message) in the same call; * the called extension must continue to ring until answered. With the m(...) option in the Dial command (like the example below) asterisk provides only music on hold while the phone rings. exten = s,n,Dial(SIP/,30,m(default)) I can not use queues because the requirements is to have 1 call and not a lot of calls. Thanks in advance, Matteo -- _ -- 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] Ring + Music on Hold in the same call
Ok Danny but with this example I have 2 calls in the called phone, and this is what I have to avoid! Regards, Matteo Il 10/06/2010 15.16, Danny Nicholas ha scritto: Here is one way to do it (works in 1.4.22-1.4.30 at least) exten = s,n,Dial(SIP/,10) exten = s,n,Dial(SIP/,90,m(default)) This snippet will ring for 10 seconds with Ringing, then ring for 90 seconds or until answered with MOH. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana Sent: Thursday, June 10, 2010 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ring + Music on Hold in the same call Hi list, is there a way to achieve in asterisk (version 1.4.x) the behavior described below? a caller place a call to an extension, and I want the caller hears the extension ringing for some seconds, and then hears the music on hold (or a courtesy message) in the same call; the called extension must continue to ring until answered. With the m(...) option in the Dial command (like the example below) asterisk provides only music on hold while the phone rings. exten = s,n,Dial(SIP/,30,m(default)) I can not use queues because the requirements is to have 1 call and not a lot of calls. Thanks in advance, Matteo -- Ing. Matteo Campana - System Engineer Mobile: +39 320 4258536 Office: +39 059 821672 Fax: +39 059 821492 Web: www.klarya.it This e-mail transmission may contain legally privileged and/or confidential information. Please do not read it if you are not the intended recipient(s). Any use, distribution, reproduction or disclosure by any other person is strictly prohibited. If you have received this e-mail in error, please notify the sender and destroy the original transmission. -- _ -- 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] understand which asterisk thread is consuming CPU
Dear all using top -H i can see that some asterisk thread are consuming many CPU (sometimes more than 50%) Is there a way to understand what is doing the process with pid 9429 ? i've tried the core show thread command, but it doesn't seem to print any PID information. Thanks to all in advance PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 9429 root 20 0 662m 93m 5596 S 23 3.1 29:28.91 asterisk 13261 root 20 0 662m 93m 5596 S 10 3.1 0:04.54 asterisk 15646 root 20 0 662m 93m 5596 S4 3.1 0:00.82 asterisk 15648 root 20 0 662m 93m 5596 S3 3.1 0:00.88 asterisk 9413 root 20 0 662m 93m 5596 S3 3.1 1:25.85 asterisk 13987 root 20 0 662m 93m 5596 S3 3.1 0:03.22 asterisk 15743 root 20 0 662m 93m 5596 S2 3.1 0:00.82 asterisk 9432 root 20 0 662m 93m 5596 S1 3.1 13:06.55 asterisk 13778 root 20 0 662m 93m 5596 S1 3.1 0:04.82 asterisk 9412 root 20 0 662m 93m 5596 S1 3.1 0:34.84 asterisk 9465 root 20 0 662m 93m 5596 S1 3.1 0:39.63 asterisk 13351 root 20 0 662m 93m 5596 S1 3.1 0:03.02 asterisk 13654 root 20 0 662m 93m 5596 S1 3.1 0:02.64 asterisk 14758 root 20 0 662m 93m 5596 S1 3.1 0:02.22 asterisk 14911 root 20 0 662m 93m 5596 S1 3.1 0:03.28 asterisk 15004 root 20 0 662m 93m 5596 S1 3.1 0:02.04 asterisk 15006 root 20 0 662m 93m 5596 S1 3.1 0:02.68 asterisk 15126 root 20 0 662m 93m 5596 S1 3.1 0:02.50 asterisk 15127 root 20 0 662m 93m 5596 S1 3.1 0:02.82 asterisk 15711 root 20 0 662m 93m 5596 S1 3.1 0:00.76 asterisk 15892 root 20 0 662m 93m 5596 S1 3.1 0:00.68 asterisk 15956 root 20 0 662m 93m 5596 S1 3.1 0:00.68 asterisk -- /*/ nik600 http://www.kumbe.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Out of Office
On Thu, 10 Jun 2010, Zeeshan Zakaria wrote: Shouldn't a moderator block emails from this email address, maybe temporarily? There is no moderator. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] understand which asterisk thread is consuming CPU
On Thu, Jun 10, 2010 at 03:46:31PM +0200, nik600 wrote: Dear all using top -H i can see that some asterisk thread are consuming many CPU (sometimes more than 50%) Is there a way to understand what is doing the process with pid 9429 ? strace -p 9429 This would help if the thread actually does some system calls and not not constantly in userland. -- 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] Ring + Music on Hold in the same call
Not sure how this would work, but you could create a special MOH file that was 10 seconds of ringing followed by the normal MOH - I know this CAN be done, just takes a bit of trial and error. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana Sent: Thursday, June 10, 2010 8:41 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Ring + Music on Hold in the same call Ok Danny but with this example I have 2 calls in the called phone, and this is what I have to avoid! Regards, Matteo Il 10/06/2010 15.16, Danny Nicholas ha scritto: Here is one way to do it (works in 1.4.22-1.4.30 at least) exten = s,n,Dial(SIP/,10) exten = s,n,Dial(SIP/,90,m(default)) This snippet will ring for 10 seconds with Ringing, then ring for 90 seconds or until answered with MOH. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana Sent: Thursday, June 10, 2010 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ring + Music on Hold in the same call Hi list, is there a way to achieve in asterisk (version 1.4.x) the behavior described below? * a caller place a call to an extension, and I want the caller hears the extension ringing for some seconds, and then hears the music on hold (or a courtesy message) in the same call; * the called extension must continue to ring until answered. With the m(...) option in the Dial command (like the example below) asterisk provides only music on hold while the phone rings. exten = s,n,Dial(SIP/,30,m(default)) I can not use queues because the requirements is to have 1 call and not a lot of calls. Thanks in advance, Matteo -- Ing. Matteo Campana - System Engineer Mobile: +39 320 4258536 Office: +39 059 821672 Fax: +39 059 821492 Web: http://www.klarya.it/ www.klarya.it This e-mail transmission may contain legally privileged and/or confidential information. Please do not read it if you are not the intended recipient(s). Any use, distribution, reproduction or disclosure by any other person is strictly prohibited. If you have received this e-mail in error, please notify the sender and destroy the original transmission. image002.jpgimage001.jpg-- _ -- 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] Out of Office
If Doug generates enough spam with his unfortunate rule selection, he will probably get zapped next month; We just have to live with it until the 14th. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, June 10, 2010 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Out of Office On Thu, 10 Jun 2010, Zeeshan Zakaria wrote: Shouldn't a moderator block emails from this email address, maybe temporarily? There is no moderator. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] early media issue from phone co.
Edwin, In your outbound context, you need to have the dialplan evaluate the hangupcause variable and send an appropriate message to your callers. Check out the following URL for some samples that you may adapt for your circumstance. http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause If you need more specific assistance, let me know. Sincerely, Trevor Hammonds -Original Message- From: Edwin Lam Sent: Tuesday, June 08, 2010 4:11 PM Subject: [asterisk-users] early media issue from phone co. hi folks. i have the following puzzle: when i call certain cell phone# using a regular phone POTS. the called cell phone co. usually return a message such as phone travel out of range or phone is busy etc. if the phone is unreachable. now when i have the following setup: sip phone - asterisk - PRI - phone co. i call the same cell# and if it's unavailable. the PRI return cause code 31 and hangup, asterisk will then send a SIP BYE to the sip phone and the channel will simply hangup. how do i get the message on the sip phone? -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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] Out of Office
Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-10 10:19 AM, Danny Nicholas da...@debsinc.com wrote: If Doug generates enough spam with his unfortunate rule selection, he will probably get zapped next month; We just have to live with it until the 14th. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-bou... On Thu, 10 Jun 2010, Zeeshan Zakaria wrote: Shouldn't a moderator block emails from this email ad... -- _ -- 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] Out of Office
I remember at least once, may be two years ago, similar out of office replies were flooding this mailing list almost once every hour or two, and that email address was blocked, with a confirmation to the list that the address was blocked. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-09 3:59 AM, d...@accessgate.net wrote: I will be out of the office starting Wed June 9th and returning Wed June 16th. Please contact Mary at m...@accessgate.net cell 407-267-1463 or Jonathan at jsny...@accessgate.net cell 407-267-0056 or call our main number 888-227-9337. -- _ -- 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] Am I having problems with codecs? or am I not receiving an invite at all from my DID provider?
Hi Guys, I have Spikko setup as provider of DID and outbound routes and I can make calls out but no inbound calls via DID can be made. I did a sip debug which is reported below. I never receive the call though, I have a catch all in my inbound routes and it doesn't hit my context at all or not sip invite comes in: FreePBX: Trunk Name: *Spikko* Peer Detail *username=MyUsername* *type=friend* *secret=MyPassword* *host=sip.spikko.com* *nat=no* *port=5090* *fromuser=MyUsername* *disallow=all* *allow=g729gsmulawalaw* Register String: *MyUsername:mypassw...@sip.spikko.com:5090/MyUsername* Inbound Router: *Send Any DID and ANY CID to Music on Hold* Sip debug: *Really destroying SIP dialog ' 417b3c8f3a97a82d4629343a53b2f...@177.177.177.177' Method: REGISTER* *tel*CLI* *--- SIP read from UDP:82.80.252.29:5090 ---* *INVITE sip:myusern...@177.177.177.177 sip%3amyusern...@177.177.177.177SIP/2.0 * *Via: SIP/2.0/UDP 82.80.252.234:5090;branch=z9hG4bK07b38a0c;rport* *From: Unknown sip:unkn...@82.80.252.234:5090;tag=as24089849* *To: sip:myusern...@177.177.177.177 sip%3amyusern...@177.177.177.177* *Contact: sip:unkn...@82.80.252.234:5090* *Call-ID: 55a4cf1f4e5575e97f8b3b23495f0...@82.80.252.234* *CSeq: 102 INVITE* *User-Agent: AG1* *Max-Forwards: 70* *Date: Thu, 10 Jun 2010 14:58:09 GMT* *Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY* *Supported: replaces* *Content-Type: application/sdp* *Content-Length: 331* * * *v=0* *o=root 6129 6129 IN IP4 82.80.252.234* *s=session* *c=IN IP4 82.80.252.234* *t=0 0* *m=audio 10172 RTP/AVP 18 3 97 101* *a=rtpmap:18 G729/8000* *a=fmtp:18 annexb=no* *a=rtpmap:3 GSM/8000* *a=rtpmap:97 iLBC/8000* *a=fmtp:97 mode=30* *a=rtpmap:101 telephone-event/8000* *a=fmtp:101 0-16* *a=silenceSupp:off - - - -* *a=ptime:20* *a=sendrecv* * * *-* *--- (14 headers 16 lines) ---* *Using INVITE request as basis request - 55a4cf1f4e5575e97f8b3b23495f0...@82.80.252.234* *Found peer 'Spikko' for 'Unknown' from 82.80.252.29:5090* I also sometimes get this even though trunk shows registered and can make calls out: *--- Transmitting (no NAT) to 82.80.252.29:5090 ---* *SIP/2.0 489 Bad event* *Via: SIP/2.0/UDP 82.80.252.234:5090 ;branch=z9hG4bK463b703d;received=82.80.252.29;rport=5090* *From: asterisk sip:aster...@82.80.252.234:5090;tag=as4af8cf81* *To: sip:saarsha...@173.203.29.102 sip%3asaarsha...@173.203.29.102 ;tag=as64c0ba34* *Call-ID: 497197a679122f5d448d324f571f3...@82.80.252.234* *CSeq: 102 NOTIFY* *Server: Asterisk PBX 1.6.2.7* *Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO* *Supported: replaces, timer* *Content-Length: 0* Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Out of Office
I called Mary and chatted about how to suspend Doug's subscription to the list. She's doing her best to take care of it, so let's cut her a little slack :) --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Thursday, June 10, 2010 5:33 AM To: m...@accessgate.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Out of Office Hi, It isn't a problem with the list. And it is not 'mine'. It is a problem with your software. I am just one of the thousands of people it is annoying! Perhaps your IT staff could help fix it? CCing the list so everyone is aware of your wonderful customer service. ;) S On 10 Jun 2010, at 11:27, Mary wrote: He is away with no cell phone or e-mail so either be helpful and tell me how to change (step by step) this to take him off your list or write a progrm for your list to fix this so it doesnt happen! Mary Shubert Accessgate.net, Inc. Suite 106 8600 Commodity Circle Orlando, FL 32819 m...@accessgate.net Office Toll Free: (888) 227-9337 Fax: (407) 352-2717 From: Steve Howes steve-li...@geekinter.net Sent: Thursday, June 10, 2010 4:26 AM To: d...@accessgate.net, m...@accessgate.net, jsny...@accessgate.ne Subject: Re: [asterisk-users] Out of Office On 10 Jun 2010, at 06:20, d...@accessgate.net wrote: I will be out of the office starting Wed June 9th and returning Wed June 16th. Please contact Mary at m...@accessgate.net cell 407-267-1463 or Jonathan at jsny...@accessgate.net cell 407-267-0056 or call our main number 888-227-9337. Several thousand people DO NOT need spamming with this daily because you can't configure your mail client/server to reply to a mailing list. Please FIX THIS. S -- _ -- 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] OT - Astmanproxy download broken ?
2010/6/9 Olivier oza_4...@yahoo.fr Hi, Is Astmanproxy still downloadable ? At the moment, I can't download anything. I'm usually using this http://github.com/davetroy/astmanproxy/tarball/master URL I can use a previous tar file but I would be pleased to know if I should do something around this issue or not. Regards It is now working Ok. -- _ -- 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] [compat] section in asterisk.conf : compatibility with pipe delimiter
On Wednesday 09 June 2010 16:28:44 nik600 wrote: Reading the the upgrade file it seems that the pbx_realtime should affect also the extension.conf settings... where am i wrong? You're just wrong. Extensions.conf is not affected at all by that setting. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Am I having problems with codecs? or am I not receiving an invite at all from my DID provider?
FreePBX questions should be asked at FreePBX forums. As for the asterisk part, where are you defining the context to receive incoming calls? Probably in the trunk settings (Peer Details) you need to add context=from-trunk if FreePBX still uses it as the default context for incoming calls. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-10 11:24 AM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, I have Spikko setup as provider of DID and outbound routes and I can make calls out but no inbound calls via DID can be made. I did a sip debug which is reported below. I never receive the call though, I have a catch all in my inbound routes and it doesn't hit my context at all or not sip invite comes in: FreePBX: Trunk Name: *Spikko* Peer Detail *username=MyUsername* *type=friend* *secret=MyPassword* *host=sip.spikko.com* *nat=no* *port=5090* *fromuser=MyUsername* *disallow=all* *allow=g729gsmulawalaw* Register String: *MyUsername:mypassw...@sip.spikko.com:5090/MyUsername* Inbound Router: *Send Any DID and ANY CID to Music on Hold* Sip debug: *Really destroying SIP dialog ' 417b3c8f3a97a82d4629343a53b2f...@177.177.177.177' Method: REGISTER* *tel*CLI* *--- SIP read from UDP:82.80.252.29:5090 ---* *INVITE sip:myusern...@177.177.177.177 sip%3amyusern...@177.177.177.177SIP/2.0 * *Via: SIP/2.0/UDP 82.80.252.234:5090;branch=z9hG4bK07b38a0c;rport* *From: Unknown sip:unkn...@82.80.252.234:5090;tag=as24089849* *To: sip:myusern...@177.177.177.177 sip%3amyusern...@177.177.177.177* *Contact: sip:unkn...@82.80.252.234:5090* *Call-ID: 55a4cf1f4e5575e97f8b3b23495f0...@82.80.252.234* *CSeq: 102 INVITE* *User-Agent: AG1* *Max-Forwards: 70* *Date: Thu, 10 Jun 2010 14:58:09 GMT* *Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY* *Supported: replaces* *Content-Type: application/sdp* *Content-Length: 331* * * *v=0* *o=root 6129 6129 IN IP4 82.80.252.234* *s=session* *c=IN IP4 82.80.252.234* *t=0 0* *m=audio 10172 RTP/AVP 18 3 97 101* *a=rtpmap:18 G729/8000* *a=fmtp:18 annexb=no* *a=rtpmap:3 GSM/8000* *a=rtpmap:97 iLBC/8000* *a=fmtp:97 mode=30* *a=rtpmap:101 telephone-event/8000* *a=fmtp:101 0-16* *a=silenceSupp:off - - - -* *a=ptime:20* *a=sendrecv* * * *-* *--- (14 headers 16 lines) ---* *Using INVITE request as basis request - 55a4cf1f4e5575e97f8b3b23495f0...@82.80.252.234* *Found peer 'Spikko' for 'Unknown' from 82.80.252.29:5090* I also sometimes get this even though trunk shows registered and can make calls out: *--- Transmitting (no NAT) to 82.80.252.29:5090 ---* *SIP/2.0 489 Bad event* *Via: SIP/2.0/UDP 82.80.252.234:5090 ;branch=z9hG4bK463b703d;received=82.80.252.29;rport=5090* *From: asterisk sip:aster...@82.80.252.234:5090;tag=as4af8cf81* *To: sip:saarsha...@173.203.29.102 sip%3asaarsha...@173.203.29.102 ;tag=as64c0ba34* *Call-ID: 497197a679122f5d448d324f571f3...@82.80.252.234* *CSeq: 102 NOTIFY* *Server: Asterisk PBX 1.6.2.7* *Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO* *Supported: replaces, timer* *Content-Length: 0* Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Out of Office
Thanks Don - Doug isn't going to be a happy camper when Mary gets done with him... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Thursday, June 10, 2010 10:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: m...@accessgate.net Subject: Re: [asterisk-users] Out of Office I called Mary and chatted about how to suspend Doug's subscription to the list. She's doing her best to take care of it, so let's cut her a little slack :) --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Thursday, June 10, 2010 5:33 AM To: m...@accessgate.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Out of Office Hi, It isn't a problem with the list. And it is not 'mine'. It is a problem with your software. I am just one of the thousands of people it is annoying! Perhaps your IT staff could help fix it? CCing the list so everyone is aware of your wonderful customer service. ;) S On 10 Jun 2010, at 11:27, Mary wrote: He is away with no cell phone or e-mail so either be helpful and tell me how to change (step by step) this to take him off your list or write a progrm for your list to fix this so it doesnt happen! Mary Shubert Accessgate.net, Inc. Suite 106 8600 Commodity Circle Orlando, FL 32819 m...@accessgate.net Office Toll Free: (888) 227-9337 Fax: (407) 352-2717 From: Steve Howes steve-li...@geekinter.net Sent: Thursday, June 10, 2010 4:26 AM To: d...@accessgate.net, m...@accessgate.net, jsny...@accessgate.ne Subject: Re: [asterisk-users] Out of Office On 10 Jun 2010, at 06:20, d...@accessgate.net wrote: I will be out of the office starting Wed June 9th and returning Wed June 16th. Please contact Mary at m...@accessgate.net cell 407-267-1463 or Jonathan at jsny...@accessgate.net cell 407-267-0056 or call our main number 888-227-9337. Several thousand people DO NOT need spamming with this daily because you can't configure your mail client/server to reply to a mailing list. Please FIX THIS. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Out of Office
Doug will have it easy. I pity the next member of this group that forgets to take care of business before going on vacation! --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, June 10, 2010 10:34 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Out of Office Thanks Don - Doug isn't going to be a happy camper when Mary gets done with him... -- _ -- 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] tuning software echo cancellation
We have been distributing asterisk servers for several years now, and early on decided that hardware echo can was the way to go. Our first few boxes without it had horrid echo problems, and attempts at tuning in 2006 didn't make any difference. We installed a new server yesterday at a client's location with a Rhino 4 port FXO card (HW EC included), and when an inbound call was answered the oddest shrieking sound was heard by the caller, and the internal SIP phone heard nothing at all. On a call with Rhino support they disabled the echo cancellation module and all was well, though of course we have a horrible echo problem now. We are going through an RMA process with Rhino, which is fine (kudos for them to cross ship - really good support team there). But the client is of course chomping at the bit to get the system live. We are totally out of touch on the subject of software echo cancellation in asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand that when Dahdi detects no HWEC, it enables SWEC by default. Is there anything I can do to tweak the settings to try and make this liveable for the client until we get the card? The server is in the Caribbean, so it may actually be a bit before the card arrives. We would love to get them running before then, but it is so bad right now that we cannot. Thanks for any links to info... Cheers, j -- _ -- 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] tuning software echo cancellation
At 09:49 AM 6/10/2010, you wrote: The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand that when Dahdi detects no HWEC, it enables SWEC by default. Is there anything I can do to tweak the settings to try and make this liveable for the client until we get the card? The server is in the Caribbean, so it may actually be a bit before the card arrives. We would love to get them running before then, but it is so bad right now that we cannot. Thanks for any links to info... If it was me, I'd sure risk the $40 for 4 lines worth of HPEC. I have a Digium card so it was free, but I put up with a year of messing with the other software echo cans before HPEC was released and the day I got it working was the last day I ever heard echo and the last day my wife ever complained about it. OSLEC is also supposed to be good, but HPEC is easy and it works. Might work good enough you can stop buying hardware echo solutions for small installations. 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] tuning software echo cancellation
On Thu, 10 Jun 2010, Jeff LaCoursiere wrote: We are totally out of touch on the subject of software echo cancellation in asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand that when Dahdi detects no HWEC, it enables SWEC by default. Is there anything I can do to tweak the settings to try and make this liveable for the client until we get the card? The server is in the Caribbean, so it may actually be a bit before the card arrives. We would love to get them running before then, but it is so bad right now that we cannot. I've been using OSLEC and TDM400 type cards for a while now (openvox). It just works Gordon -- _ -- 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] tuning software echo cancellation
- Gordon Henderson gordon+aster...@drogon.net escreveu: On Thu, 10 Jun 2010, Jeff LaCoursiere wrote: We are totally out of touch on the subject of software echo cancellation in asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand that when Dahdi detects no HWEC, it enables SWEC by default. Is there anything I can do to tweak the settings to try and make this liveable for the client until we get the card? The server is in the Caribbean, so it may actually be a bit before the card arrives. We would love to get them running before then, but it is so bad right now that we cannot. I've been using OSLEC and TDM400 type cards for a while now (openvox). It just works Gordon I second that, OSLEC is awesome. -- _ -- 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] tuning software echo cancellation
On Thu, 10 Jun 2010, Gordon Henderson wrote: On Thu, 10 Jun 2010, Jeff LaCoursiere wrote: We are totally out of touch on the subject of software echo cancellation in asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand that when Dahdi detects no HWEC, it enables SWEC by default. Is there anything I can do to tweak the settings to try and make this liveable for the client until we get the card? The server is in the Caribbean, so it may actually be a bit before the card arrives. We would love to get them running before then, but it is so bad right now that we cannot. I've been using OSLEC and TDM400 type cards for a while now (openvox). It just works Gordon Isn't OSLEC on by default? Or is this something I must turn on specifically? If it is on it isn't doing much in our case :) Cheers, j -- _ -- 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] tuning software echo cancellation
On Thu, 10 Jun 2010, Jeff LaCoursiere wrote: On Thu, 10 Jun 2010, Gordon Henderson wrote: On Thu, 10 Jun 2010, Jeff LaCoursiere wrote: We are totally out of touch on the subject of software echo cancellation in asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand that when Dahdi detects no HWEC, it enables SWEC by default. Is there anything I can do to tweak the settings to try and make this liveable for the client until we get the card? The server is in the Caribbean, so it may actually be a bit before the card arrives. We would love to get them running before then, but it is so bad right now that we cannot. I've been using OSLEC and TDM400 type cards for a while now (openvox). It just works Isn't OSLEC on by default? Or is this something I must turn on specifically? If it is on it isn't doing much in our case :) I compile up stuff from scratch, so a lot might depend on your distribution.. You need the module dahdi_echocan_oslec loaded, and in /etc/dahdi/system.conf, I have: echocanceller=oslec,1-4 Gordon -- _ -- 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] ISDN - SIP
i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on CentOS 5.5. The only thing, i want to do is a call-redirection from an isdn-call to my mobile via sip-account. My extension conf is: general] static=yes writeprotect=no [globals] OUT_PORT=1 [ISDN] exten = 12345,1,Dial(SIP/012346737...@sipprovider.local) If i call to the msn 12345, the SIP-call is going out, but after a second the call is stopped. What is wrong, with my configuration? Kernel show Jun 10 20:48:58 wolf kernel: hdlc_down unknown prim(280) Jun 10 20:49:04 wolf kernel: MDL_ERROR|REQ (tei_l2) Asterisk shows: P[ 1] MGMT: SSTATUS: L1_ACTIVATED P[ 1] handle_frm: frm-addr:42000103 frm-prim:3f082 P[ 1] channel with stid:0 not in use! P[ 1] handle_frm: frm-addr:42000103 frm-prim:30582 P[ 1] set_channel: bc-channel:0 channel:1 P[ 1] I IND :NEW_CHANNEL oad:xxx dad:12345 pid:2 state:none P[ 1] -- channel:1 mode:TE cause:16 ocause:16 rad: cad: P[ 1] -- info_dad: onumplan:2 dnumplan:4 rnumplan: cpnnumplan:0 P[ 1] -- caps:Speech pi:0 keypad: sending_complete:1 P[ 1] -- screen:0 -- pres:0 P[ 1] -- addr:0 l3id:20007 b_stid:0 layer_id:0 P[ 1] -- facility:Fac_None out_facility:Fac_None P[ 1] -- bc_state:BCHAN_CLEANED P[ 1] Chan not existing at the moment bc-l3id:20007 bc:0x8721e9c event:NEW_CHANNEL port:1 channel:1 P[ 1] NO USERUESRINFO P[ 1] -- found chan (preselected): 1 P[ 1] set_chan_in_stack: 1 P[ 1] setup_bc: with dsp P[ 1] -- Channel is 1 P[ 1] -- TRANSPARENT Mode P[ 1] I IND :SETUP oad:xxx dad:12345 pid:2 state:none P[ 1] -- channel:1 mode:TE cause:16 ocause:16 rad: cad: P[ 1] -- info_dad: onumplan:2 dnumplan:4 rnumplan: cpnnumplan:0 P[ 1] -- caps:Speech pi:0 keypad: sending_complete:1 P[ 1] -- screen:0 -- pres:0 P[ 1] -- addr:50010102 l3id:20007 b_stid:10010100 layer_id:50010180 P[ 1] -- facility:Fac_None out_facility:Fac_None P[ 1] -- bc_state:BCHAN_ACTIVATED P[ 1] -- Bearer: Speech P[ 1] -- Codec: Alaw P[ 0] -- * NEW CHANNEL dad:12345 oad:xxx P[ 1] read_config: Getting Config P[ 1] -- CTON: Unknown P[ 1] -- EXPORT_PID: pid:2 P[ 1] -- PRES: Allowed (0) P[ 1] -- SCREEN: Unscreened (0) P[ 1] * Queuing chan 0x89e5410 P[ 1] I SEND:RELEASE oad:xxx dad:12345 pid:2 P[ 1] -- bc_state:BCHAN_ACTIVATED P[ 1] -- channel:1 mode:TE cause:16 ocause:1 rad: cad: P[ 1] -- info_dad: onumplan:2 dnumplan:4 rnumplan: cpnnumplan:0 P[ 1] -- caps:Speech pi:0 keypad: sending_complete:1 P[ 1] -- screen:0 -- pres:0 P[ 1] -- addr:50010102 l3id:20007 b_stid:10010100 layer_id:50010180 P[ 1] -- facility:Fac_None out_facility:Fac_None P[ 1] GOT SETUP OK P[ 1] Sending msg, prim:34d80 addr:41000104 dinfo:20007 P[ 1] BCHAN: bchan ACT Confirm pid:2 P[ 1] handle_frm: frm-addr:42000103 frm-prim:3f182 P[ 1] -- lib: RELEASE_CR Ind with l3id:20007 P[ 1] -- lib: CLEANING UP l3id: 20007 P[ 1] -- hangup P[ 1] * IND : HANGUPpid:2 ctx:ISDN dad:12345 oad: State:EXTCANTMATCH P[ 1] -- l3id:20007 P[ 1] -- cause:16 P[ 1] -- out_cause:16 P[ 1] -- Channel: mISDN/1-u0 hungup new state:CLEANING P[ 1] $$$ CLEANUP CALLED pid:2 P[ 1] $$$ Cleaning up bc with stid :10010100 pid:2 P[ 1] -- ec_disable P[ 1] Sending Control ECHOCAN_OFF P[ 1] ph_control: c1:2319 c2:0 P[ 1] empty_chan_in_stack: 1 P[ 0] handle_bchan: BC not found for prim:f2481 with addr:55010180 dinfo:0 P[ 0] received 1k Unhandled Bchannel Messages: prim f2481 len 0 from addr 55010180, dinfo 0 on this port. P[ 1] MGMT: SSTATUS: L1_DEACTIVATED -- _ -- 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] Priority between calls in different queues
I may not have made myself clear. I'm not actually trying to change anything. I'm just trying to figure out what is happening. (I'm trying to analyze log files as part of evaluating our call center's performance.) I do know we don't have weights set for our queues. On 6/10/10, Mike l...@virtutel.ca wrote: Hi, Isnt there a parameter called weight for each queue that defines exactly that? I never tried it, but it appeared to do exactly what you want (according to the invaluable but often out-dated wiki). Regards, Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Hapworth Slim Sent: Thursday, June 10, 2010 15:25 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Priority between calls in different queues I'm trying to figure this out. I have agents answer calls from two different queues. We have things set up so that these agents only see one call at a time. Let's say an agent picks up a call while there are calls waiting in both queues. Clearly the head of one of the queues will now start ringing through to the other agents. But which one? Is that something that can be configured, perhaps by saying one queue has priority, or the older call has priority, or something different? Or is it something non-deterministic? -- _ -- 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] tuning software echo cancellation
On Thu, Jun 10, 2010 at 07:25:43PM +0100, Gordon Henderson wrote: On Thu, 10 Jun 2010, Jeff LaCoursiere wrote: Isn't OSLEC on by default? Sadly it's not included in the default DAHDI. Several distros include it. Those also set it as the default EC for system.conf generated by dahdi_genconf. E.g. http://svn.debian.org/viewsvn/pkg-voip/dahdi-tools/trunk/debian/patches/echocan_oslec?view=markup I compile up stuff from scratch, so a lot might depend on your distribution.. You need the module dahdi_echocan_oslec loaded, and in /etc/dahdi/system.conf, I have: echocanceller=oslec,1-4 Actually, dahdi will modprobe the module 'dahdi_echocan_foo' if you have the line 'echocanceller=foo,channels' . So you need the module avaialble, rather than loaded. -- 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] Ring + Music on Hold in the same call
Danny Nicholas wrote: Not sure how this would work, but you could create a special MOH file that was 10 seconds of ringing followed by the normal MOH – I know this CAN be done, just takes a bit of trial and error. That's what I would suggest as well. You could use Monitor() initially to call an extension that you let ring to get the ringing sound, then you could use any of a multiple of tools to combine the ringing onto the start of MoH. Leif. -- _ -- 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] ISDN - SIP
On Fri, Jun 11, 2010 at 12:19:37AM +0200, Gergo Csibra wrote: Thursday, June 10, 2010, 11:19:16 PM, Philipp wrote: i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on CentOS 5.5. The only thing, i want to do is a call-redirection from an isdn-call to my mobile via sip-account. Unless you are using mISDN v2: Do yourself a favour and switch to CAPI with chan_capi and fcpci. mISDN v1 is a guarantee for headaches (and unstable systems). Okay. There's some problems with mISDN v2: I'm unable to compile zaphfc, because there's no source for it. mISDN v2 works with hfcpci too? Certainly there is. It's also part of the standard dahdi-extra patch. See http://git.tzafrir.org.il/?p=dahdi-extra.git;a=tree http://svn.debian.org/viewsvn/pkg-voip/dahdi-linux/trunk/debian/patches/dahdi_linux_extra -- 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
[asterisk-users] Priority between calls in different queues
I'm trying to figure this out. I have agents answer calls from two different queues. We have things set up so that these agents only see one call at a time. Let's say an agent picks up a call while there are calls waiting in both queues. Clearly the head of one of the queues will now start ringing through to the other agents. But which one? Is that something that can be configured, perhaps by saying one queue has priority, or the older call has priority, or something different? Or is it something non-deterministic? -- _ -- 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] tuning software echo cancellation
On Thu, 10 Jun 2010, Gordon Henderson wrote: On Thu, 10 Jun 2010, Jeff LaCoursiere wrote: On Thu, 10 Jun 2010, Gordon Henderson wrote: On Thu, 10 Jun 2010, Jeff LaCoursiere wrote: We are totally out of touch on the subject of software echo cancellation in asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand that when Dahdi detects no HWEC, it enables SWEC by default. Is there anything I can do to tweak the settings to try and make this liveable for the client until we get the card? The server is in the Caribbean, so it may actually be a bit before the card arrives. We would love to get them running before then, but it is so bad right now that we cannot. I've been using OSLEC and TDM400 type cards for a while now (openvox). It just works Isn't OSLEC on by default? Or is this something I must turn on specifically? If it is on it isn't doing much in our case :) I compile up stuff from scratch, so a lot might depend on your distribution.. You need the module dahdi_echocan_oslec loaded, and in /etc/dahdi/system.conf, I have: echocanceller=oslec,1-4 Ahh. I see that the MG2 canceller is installed by default, and I see by Google that it is not very much liked. SVN'ing the latest OSLEC now. Thanks for the advice! Cheers, j -- _ -- 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] warning : sip_xmit
I'm having similar errors too. Customers are also complaining that their connection dropped. Could this be related? (That's the only error that occured when that connection dropped) (using asterisk 1.6.2.8) Kenny -- tel: +32 476 780 692 On 10 Jun 2010, at 13:52, Jonas Kellens wrote: I'm getting a lot of these on the CLI : [Jun 10 13:41:36] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:37] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:38] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:39] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:40] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:50] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b393130 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:51] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b393130 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:52] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b393130 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:53] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b393130 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:41:54] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b393130 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:42:04] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b33dad0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:42:05] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b33dad0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:42:06] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b33dad0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted [Jun 10 13:42:07] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x1b33dad0 (len 554) to 192.168.12.120:2056 returned -1: Operation not permitted What can I do to stop this ?? What I usually do is restart Asterisk. After 5 to 8 restarts, it goes away... This can not be good practise. Using Asterisk 1.4.30 and sip realtime. Jonas. -- _ -- 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] Eyebeam hangs when you dial an unavailable number
I am having problems with Eyebeam when the user dials a number that is not available. This problem exists with both Asterisk 1.4 and 1.6 using Eyebeam or Xlite. The problem seems to be that when the soft phone receives the 503 Unavailable response it will not be able to dial another number for a few minutes. Anything you dial will say that the number is unavailable and it will not even send the number to Asterisk (nothing on the CLI). The soft phone can receive calls, just not send any. I have tested with other soft phones and they do not have this problem, only Counterpath products. Any idea what the problem could be? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] Priority between calls in different queues
Hi, Isnt there a parameter called weight for each queue that defines exactly that? I never tried it, but it appeared to do exactly what you want (according to the invaluable but often out-dated wiki). Regards, Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Hapworth Slim Sent: Thursday, June 10, 2010 15:25 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Priority between calls in different queues I'm trying to figure this out. I have agents answer calls from two different queues. We have things set up so that these agents only see one call at a time. Let's say an agent picks up a call while there are calls waiting in both queues. Clearly the head of one of the queues will now start ringing through to the other agents. But which one? Is that something that can be configured, perhaps by saying one queue has priority, or the older call has priority, or something different? Or is it something non-deterministic? -- _ -- 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] ISDN - SIP
Hi! i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on CentOS 5.5. The only thing, i want to do is a call-redirection from an isdn-call to my mobile via sip-account. Unless you are using mISDN v2: Do yourself a favour and switch to CAPI with chan_capi and fcpci. mISDN v1 is a guarantee for headaches (and unstable systems). Philipp -- _ -- 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] ISDN - SIP
Thursday, June 10, 2010, 11:19:16 PM, Philipp wrote: i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on CentOS 5.5. The only thing, i want to do is a call-redirection from an isdn-call to my mobile via sip-account. Unless you are using mISDN v2: Do yourself a favour and switch to CAPI with chan_capi and fcpci. mISDN v1 is a guarantee for headaches (and unstable systems). Okay. There's some problems with mISDN v2: I'm unable to compile zaphfc, because there's no source for it. mISDN v2 works with hfcpci too? -- Best regards, Gergomailto:csi...@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
[asterisk-users] Dual Atom mobo - call capacity
I'm looking for a small formfactor mobo for an install that needs to handle 25 phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone know what kinds of call volume that will handle? MD -- _ -- 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] Dual Atom mobo - call capacity
Based on comments from Ward Mundy during a recent VUC call I'd expect even a single CPU Atom system to handle that many phones in an office application. Perhaps there may be merit in dual CPU in more of a call center application. http://www.voipusersconference.org/2010/nerd-vittles-incredible-pbx/ Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgra...@mstvp.onsip.com skype mjgraves Original Message Subject: [asterisk-users] Dual Atom mobo - call capacity From: Michelle Dupuis mdup...@ocg.ca Date: Thu, June 10, 2010 7:19 pm To: Asterisk Users List asterisk-users@lists.digium.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] Dual Atom mobo - call capacity
I don't know how it calculates it, but FreePBX shows a bar for total calls that looks like it maxes out at six. We haven't hit that on any installs of this device yet, but that seems pretty low for sure. I know with four calls in progress, all VoIP, transcoding G711u to G.729, the load of the machine is still around .3 . j On Thu, 10 Jun 2010, Michelle Dupuis wrote: I'm looking for a small formfactor mobo for an install that needs to handle 25 phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone know what kinds of call volume that will handle? MD -- _ -- 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] Dual Atom mobo - call capacity
On Thu, 10 Jun 2010, Michelle Dupuis wrote: I'm looking for a small formfactor mobo for an install that needs to handle 25 phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone know what kinds of call volume that will handle? On Thu, 10 Jun 2010, mgra...@mstvp.com wrote: Based on comments from Ward Mundy during a recent VUC call I'd expect even a single CPU Atom system to handle that many phones in an office application. Perhaps there may be merit in dual CPU in more of a call center application. Assuming you're talking about something like the Atom 330... My guess is you will have plenty of horsepower for 25 phone sets -- probably even 25 simultaneous calls. The 330 is dual-core and hyper-threaded so it shows up as 4 CPUs in top. Asterisk is multi-threaded and should distribute the workload. Another advantage is that if you have something CPU heavy like bzip2'ing your database dump or compiling Asterisk from source, there are still several CPUs available for Asterisk. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
