Re: [asterisk-users] Increasing the voice volume from the digium card
Hi there, Have you mentioned to the last email by Matthew? He told you can achieve what you need through setting rxgain and txgain for each channel in zapata.conf (It means you have control each channel seperatly even on a card) Regards. -- M. Shokuie Nia, SENA CO. Date: Thu, 6 Dec 2007 03:52:04 -0800 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com CC: [EMAIL PROTECTED] Subject: Re: [asterisk-users] Increasing the voice volume from the digium card Dears; I have one TMA22B card, I need to increase the volume on the caller side (the voice that come from asterisk to caller, while caller is using his mobile by dialing to asterisk to the fxo port), these ports are the 3rd and 4th ports. Do I have to run: /usr/src/asterisk-1.4/zaptel-1.4/ztmonitor 3 -vv and: /usr/src/asterisk-1.4/zaptel-1.4/ztmonitor 4 -vv Also, what should I write in my zapata.conf file? The fxs_rx: ? Is it to be the Rx reading of the previous commands or more if I need to increase the voice volume? What if I have another TDM22B card, and I need to increase the voice volume at port 3 and 4 of that second card, what the attribute to be bypass for the above commands? Your kindly help is high appreciated. Regards Bilal bilal ghayyad wrote: Hi List; Anyone knows a method (command) to increase the voice volume at diguim card level? Are you trying to do this at some other level than rxgain and txgain settings in zapata.conf? If so, for the analog cards there are some module parameters for doing so. For digital T1/E1 cards, the only way to do it is with the gain options in zapata.conf. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. -- In zapata.conf you can add rxgain and txgain settings and use ztmonitor to get it set. There are some more details on this on voip-info.org. On Nov 29, 2007 1:49 AM, bilal ghayyad wrote: Hi All; I have an IP Trunk established between Asterisk and the VoIP service provider, when call from my mobile to the PBX and then enter the destination number to call via the VoIP, I got a connection but the voice level volume need to be increased, I am trying to find if zaptel (diguim card) can increase the volume (if there is any command can do that)? And if that volume level is possible to be applied only for that IP Trunk and not for others. Any Help? Regards Bilal Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Increasing the voice volume from the diguim cards
Dear Matthew, Would you mind giving somehints about these module parameters to me. I just add two manager commands to chan_zap in order to let the user adjust the gains at runtime but dont know anything about the variables you mentioned and like to know ;) Regards. --- M. Shokuie Nia, SENA Co. Date: Sat, 1 Dec 2007 13:21:28 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Increasing the voice volume from the diguim cards bilal ghayyad wrote: Hi List; Anyone knows a method (command) to increase the voice volume at diguim card level? Are you trying to do this at some other level than rxgain and txgain settings in zapata.conf? If so, for the analog cards there are some module parameters for doing so. For digital T1/E1 cards, the only way to do it is with the gain options in zapata.conf. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_alsa issue
Hi folks, Its the forth day I'm sticking to a problem with chan_alsa, The sound played or captured from the device is choppy time to time. I mean when talking on a console/dsp microphone the other side hear my sound choppy and I'm hearing hers the same but not all the time during a call, sound sometimes are clear. Even when I'm putting the sip side on hold i hear the same choppy music on hold. Any one have any idea how i could get closer to the problem. Any hint would be highly appreciated. -- M. Shokuie Nia. _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is wrong with this mailing list
Hi all, Anyone knows what is wrong with this mailing list its a while all my new posts appear as a reply (branch) for others post, is there any hints i could prevent this issue?? Regards. _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is wrong with this mailing list
HI Erik, thanks for your post, Actually im sending new posts not replying but if you see them correct, how come its wrongly viewed for me. Are you using a speciall software to view mailing lists? Im just using firefox not a special one! Regards. -- M. Shokuie Nia Date: Tue, 13 Nov 2007 23:33:08 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] What is wrong with this mailing list On Nov 13, 2007 11:21 PM, Mohammad Shokuie wrote: Anyone knows what is wrong with this mailing list its a while all my new posts appear as a reply (branch) for others post, is there any hints i could prevent this issue?? I believe your posts are all showing up correctly for me. That said, this sort of thing can happen frequently if, instead of composing a new email to the list, you hit Reply to an existing message and just change the subject line. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is wrong with this mailing list
Hi Erik, By firefox i mean a Hotmail web mail, it means there is no mail client. I dont know if there would be any difference if i subscribe and use other mails like gmail! Regards. -- M. Shokuie Nia Date: Tue, 13 Nov 2007 23:52:03 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] What is wrong with this mailing list On Nov 13, 2007 11:44 PM, Mohammad Shokuie wrote: HI Erik, thanks for your post, Actually im sending new posts not replying but if you see them correct, how come its wrongly viewed for me. Are you using a speciall software to view mailing lists? Im just using firefox not a special one! You're using firefox? How so? I'd recommend either a good email client (Thunderbird) or a good web email interface (gmail). (I'm using gmail's web interface) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk trunk and manager redirect problem
Dear All, Have anyone tested the trunk version and redirect command, it seems the pbx routines changed much and the redirect mechanism doesnt work well with this new changes. When ever i redirect a channel i got the channel hanged up. After a survey in the code i got that when the channel soft hanged up in the async goto the loop in the pbx_run exits and the channel got a real hang up instead of jumping to the begining of the loop in the routine. Regards. -- M. Shokuie Nia _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issue with Asterisk realtime
Dear folks, I'm using * realtime with no problems on most of the systems i've setup but rarely i confront this problem that the asterisk doesn't load from database when the systems rebooted and i have to reload it manually or restart it, but it would work fine afterward, no problem how many times you stop and start the *. It seems, there is a missequence of deamon loading at boot time but i have no clue which deamons! Im using FC5, MySQL5, Asterisk 1.2.18 Any help would be highly appreciated. --- M. Shokuie Nia. _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CAS on Sangoma boards
Dear folks, I would be very thankful if an experienced user can help me out here. I wanna use mfcr2 and unicall library on sangoma boards but so far impossible for me. As Im setting the framing type to CAS (TE_SIGMODE = CAS) on wanpipe I couldnt get the link alarm out (i looped a A102d links) but when setting it to CCS everything works fine and the green lights shine on the back. Can anyone send me a working sample of wanpipe.conf and zaptel.conf for cas signalling? and is it possible that the alarms are because of looping the links (although in ccs mode it works just fine) ? Any help and hint would be highly appreciated. PS. i define the span in zaptel as cas with hdb3 -- M. Shokuie Nia. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + Zap + CAS Signalling
Hi folks, I had a survey online but there i couldnt find a clean sample of CAS signalling on E1 interfaces. I defined a span with CAS framing and HDB3 line coding but dont know which signalling to use for channels. I'd use 3 bit CAS signalling and 20 incoming channels and 10 outgoing ones. Anyone could help me define the signalling for these channels. PS. Im using Sangoma cards. Any help would be highly appreciated. --- M. Shokuie Nia. _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] #Transfer - Timeout is configurable?
Dear Marco, Take a look at featuredigittimeout, that might help :) Regards. --- M. Shokuie Nia From: Marco Mouta [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [asterisk-users] #Transfer - Timeout is configurable? Date: Fri, 20 Oct 2006 15:54:40 +0100 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc7-f8.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Fri, 20 Oct 2006 08:33:38 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id B24CE2FCACB;Fri, 20 Oct 2006 07:54:46 -0700 (MST) Received: from psmtp.com (exprod8mx31.postini.com [64.18.3.131])by lists.digium.com (Postfix) with SMTP id 043422FC908for asterisk-users@lists.digium.com;Fri, 20 Oct 2006 07:54:31 -0700 (MST) Received: from source ([64.233.182.189]) by exprod8mx31.postini.com([64.18.7.10]) with SMTP; Fri, 20 Oct 2006 07:54:49 PDT Received: by nf-out-0910.google.com with SMTP id a25so1647923nfcfor asterisk-users@lists.digium.com;Fri, 20 Oct 2006 07:54:48 -0700 (PDT) Received: by 10.49.80.12 with SMTP id h12mr270922nfl;Fri, 20 Oct 2006 07:54:40 -0700 (PDT) Received: by 10.49.61.4 with HTTP; Fri, 20 Oct 2006 07:54:40 -0700 (PDT) X-Message-Info: txF49lGdW40NbEIHbS9sAQsIwf1mUc3x4qkWMdWhkFg= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com;h=received:message-id:date:from:to:subject:mime-version:content-type:content-transfer-encoding:content-disposition;b=EsdeGoRq2HU+414uKhc1nlvrB+1F3Yul63Gn1RtE0AVBAzJsVuy8H9SAgcOSScDviu4gAzCKkpbOHE3ie+84kH5l4oqdtnSkUoFsS+koelEdh6UgBH27DoYK56dzdZcLfvX6xXaLt5l6HuuYD9K0xB5vK3RpA/3/gCfP4nROYNA= X-pstn-levels: (S:30.80636/99.9 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c X-pstn-addresses: from [EMAIL PROTECTED] [db-null] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 20 Oct 2006 15:33:38.0955 (UTC) FILETIME=[1D6E6DB0:01C6F45D] Hi guys, This should be has an easy answer for you, my users are complaining that when they press # and then ear gorgeous Allison Transfer the timeout is very small, they must enter immediatly the extension to transfer the call. Is it possible to change this? ;transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call This is timeout after pressing the first digit isn't it? -- best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rxfax problem
Hi Steve, As a matter of fact, you've done a greate job in writting this library, no doubts. I really dont know rxgain = 12 makes that much distortion but I'm curios to know if I pass through the incoming fax to an analog fax machine on another fxs line, the machine wouldn't receive the fax too? Anyways, let me take the most benefit as im sure you'd read this post, i have problem with the size of received page which is shrinked, can u give me a hint about this problem too :) Thanks. --- M. Shokuie Nia From: Steve Underwood [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] rxfax problem Date: Fri, 20 Oct 2006 20:20:18 +0800 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc6-f10.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Fri, 20 Oct 2006 05:42:01 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id EFF0B2FC87C;Fri, 20 Oct 2006 05:20:37 -0700 (MST) Received: from psmtp.com (exprod8mx13.postini.com [64.18.3.113])by lists.digium.com (Postfix) with SMTP id B67A62FC82Ffor asterisk-users@lists.digium.com;Fri, 20 Oct 2006 05:20:05 -0700 (MST) Received: from source ([202.14.67.92]) by exprod8mx13.postini.com([64.18.7.10]) with SMTP; Fri, 20 Oct 2006 05:20:20 PDT Received: from [192.168.2.50] (229.166.17.210.dyn.pacific.net.hk[210.17.166.229]) by cwb.pacific.net.hk with ESMTPid k9KCKIfs013165 for asterisk-users@lists.digium.com;Fri, 20 Oct 2006 20:20:19 +0800 X-Message-Info: txF49lGdW43chsCTszkrRosGSMI+inUm7kbzJdpspc0= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com User-Agent: Mozilla Thunderbird 1.0.8-1.1.fc4 (X11/20060501) X-Accept-Language: en-us, en References: [EMAIL PROTECTED] X-pstn-levels: (S:99.9/99.9 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c X-pstn-addresses: from [EMAIL PROTECTED] [db-null] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 20 Oct 2006 12:42:02.0256 (UTC) FILETIME=[241ED900:01C6F445] M. Shokuie Nia wrote: Dear folk, My problem solved after two day research and try and error method ;). It was related to rxgain of the board im using. I've set the rxgain to 12 and it seems made some problem. As far as I got the spandsp is so sensitive about noise on the line and because of that it couldnÂ’t hand shake with other side well. rxfax isn't sensitive to noise at all. At a gain of 12 you've caused overloading and distortion, and the signal cannot be decoded. Many people seem to be nearly deaf. They run systems at massive gain with awful distortion, and seem content until they find something like a modem or DTMF detection doesn't work too well. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rxfax problem (Trainability test failed)
Dear folks, I couldnt receive faxes and get the following debug traces on the console, I appreciate any help or even hints. Using Spandsp-0.2 app_rxfax.c:76 span_message: FLOW Get at 9600bps, modem 1 app_rxfax.c:76 span_message: FLOW Changed from phase 3 to 5 app_rxfax.c:76 span_message: FLOW Non-ECM carrier up app_rxfax.c:76 span_message: FLOW Non-ECM carrier down app_rxfax.c:76 span_message: FLOW Non-ECM carrier up app_rxfax.c:76 span_message: FLOW Non-ECM carrier trained app_rxfax.c:76 span_message: FLOW Non-ECM carrier down app_rxfax.c:76 span_message: FLOW Trainability test failed - longest run of zeros was 3356 app_rxfax.c:76 span_message: FLOW FTT app_rxfax.c:76 span_message: FLOW Non-ECM carrier up app_rxfax.c:76 span_message: FLOW Non-ECM carrier training failed . . app_rxfax.c:76 span_message: FLOW Trainability test failed - longest run of zeros was 2000 app_rxfax.c:76 span_message: FLOW FTT .. app_rxfax.c:76 span_message: FLOW Non-ECM carrier down app_rxfax.c:76 span_message: FLOW Trainability test failed - longest run of zeros was 1696 . chan_zap.c:4351 __zt_exception: Exception on 23, channel 1 chan_zap.c:3539 zt_handle_event: Got event On hook(1) on channel 1 (index 0) app_rxfax.c:329 rxfax_exec: Got hangup Regards. M. Shokuie Nia _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax receive (rx fax) problem
Dear folks, I have problem in fax reception. The astrisk detects the fax tone and jusmps to the fax extension and rxfax application starts and the max machine starts the fax but saddenly stops and seems the rxfax have died. It doesnt returns, not files in the output dir and .. Anyone have any idea or help how could i get whats wrong here. Regards. M. Shokuie Nia _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Quintum tenor configuration with asterisk help
Hi There, We've done this before. We just used TenorAX as a gatwaye for IP-PBX with 160 extensions. There is no big problem just minor tricks in the Tenor and Asterisk configs. Just let me knopw what is your problem exactly. Regards, M. Shokuie Nia. _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How could i get bridged channel partner
Dear folks, In my senario I receive a call on a Zap channel and bridge it to a SIP extension. On the sip client i should get what is the Zap partner of this call. I though i should do it through the manager but i really dont know how. I just couldnt even find what is the SIP channel that the call is using. To make it short i want to let the SIP client to manipulate the Zap parteners Gain dynamically, for this reason i need the name of the Zap channel part of the current bridge. How could i get it? Any help would be appreciated. M. Shokuie Nia. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] txfax tiff file format
Dear folks, I got a problem sending faxes using spandsp. Primerily, when the tiff file made using GIMP 2 with different compresions the fax app break downs whole *. Moreover when i made a tiff file using Microsoft mdi, everything works fine but on the other end of the call, the received fax is shrinked in size. Anyone has any idea whats the right file format and compression type for it? PS. Im using libtiff-3.7.1-2 and spandsp-0.0.2-pre25 Regards. --- M. Shokuie Nia _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ALSA channel (console/dsp) problem
Dear folks, I have a problem with console/dsp using ALSA. I dont know why the output sound is choppy sometimes and also the input one has an awful delay. Is there anyone here with experince about ALSA channels or not? I would be highly appreciated if anyone could help me. Regards. M. Shokuie Nia _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cdr mysql problem
Hi All, Thank you all. As you all mentioned it wasnt so serious and was just a simple authentication problem. Its been solved. Regards. From: Diyanat Ali [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] cdr mysql problem Date: Fri, 16 Dec 2005 07:14:05 -0600 MIME-Version: 1.0 X-Originating-IP: [202.65.140.108] X-Originating-Email: [EMAIL PROTECTED] X-Sender: [EMAIL PROTECTED] Received: from lists.digium.com ([69.16.138.164]) by bay0-mc12-f6.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Fri, 16 Dec 2005 05:20:09 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 3F3FE4419;Fri, 16 Dec 2005 06:14:02 -0700 (MST) Received: from psmtp.com (exprod5mx26.postini.com [64.18.0.181])by lists.digium.com (Postfix) with SMTP id CA1AA4415for asterisk-users@lists.digium.com;Fri, 16 Dec 2005 06:13:57 -0700 (MST) Received: from source ([65.54.162.38]) by exprod5mx26.postini.com([64.18.4.10]) with SMTP; Fri, 16 Dec 2005 08:14:06 EST Received: from mail pickup service by hotmail.com with Microsoft SMTPSVC;Fri, 16 Dec 2005 05:14:06 -0800 Received: from 65.54.162.200 by by108fd.bay108.hotmail.msn.com with HTTP;Fri, 16 Dec 2005 13:14:05 GMT X-Message-Info: LGjzam7y+LuXeCVcGsTIL6sfvJRsVs3A23oEsKN3m/A= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com X-OriginalArrivalTime: 16 Dec 2005 13:14:06.0023 (UTC)FILETIME=[978B6570:01C60242] X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [90/4] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] i am using asterisk 1.2.1 with mysql 5 without any issues, please check your configuration again, make sure you have hostname=localhost too and the dbname, user, password are correct [global] hostname=localhost dbname=databasename user=user password=password port=3306 sock=/var/lib/mysql/mysql.sock Diyanat From: Mohammad Shokuie [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cdr mysql problem Return-Path: [EMAIL PROTECTED] Dear folks, I've just compiled asterisk-addon1.2.1 after installing MySQL and MySQl-devel packages. and adjust my cdr_mysql.conf to use the defined database using username and password. But as soon as starting asterisk i get error messages informing me of error, error message is as follows : cdr_addon_mysql.c : Failed to connect mysql database cdr on localhost and res_config_mysql.c : Failed to connect database server on . Im realy lost and dont know whats wrong. I've checked the connection to MySql in command line using the same user and host and its been connected without any problem. Anyone has any idea whats wrong here. Regards. --- M. Shokuie Nia. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk 1.2.1 and mixmonitor problem
Hi there, Any one confronted a crash in asterisk when using mixmonitor app. When i'm using the mixmonitor app on a briged call as soon as the called party hangs up the call asterisk crashes and the process terminates with following error message : Segmentation fault. Ouch .. error while writing audion data :: broken pipe but when the calling party hangs up, everything is smooth. Anyone has any idea on this issue? TIA. M. Shokuie Nia _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is this possible in Asterisk?
Hi There, I can suggest you to check the dial status variable in dial plan and if its NO_ANSWER guide the caller to voicemail with 'u' option, and if they leave and get back on a fixed time you can take a look for day time night time topic in asterisk documents. HTH, -- M. Shokuie Nia. From: Christian [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Is this possible in Asterisk? Date: Sun, 18 Dec 2005 18:36:14 +0100 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc12-f19.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Sun, 18 Dec 2005 09:48:30 -0800 Received: from arizona.digium.com (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 7E14C4337;Sun, 18 Dec 2005 10:38:02 -0700 (MST) Received: from psmtp.com (unknown [64.18.0.41])by lists.digium.com (Postfix) with SMTP id 85E3E4334for asterisk-users@lists.digium.com;Sun, 18 Dec 2005 10:36:39 -0700 (MST) Received: from source ([64.233.162.192]) by exprod5mx127.postini.com([64.18.4.10]) with SMTP; Sun, 18 Dec 2005 09:36:21 PST Received: by zproxy.gmail.com with SMTP id o1so1103021nzffor asterisk-users@lists.digium.com;Sun, 18 Dec 2005 09:36:21 -0800 (PST) Received: by 10.64.3.11 with SMTP id 11mr451159qbc;Sun, 18 Dec 2005 09:36:20 -0800 (PST) Received: from uwv ( [82.182.183.11])by mx.gmail.com with ESMTP id a29sm1762599qbd.2005.12.18.09.36.19;Sun, 18 Dec 2005 09:36:20 -0800 (PST) X-Message-Info: LGjzam7y+LsP0A1T+4Y1PvJh+HFLKLA3pfw57wIWqLo= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com;h=received:message-id:from:to:references:subject:date:mime-version:content-type:content-transfer-encoding:x-priority:x-msmail-priority:x-mailer:x-mimeole;b=od0eK3XRc1NMxk1DtuLWp8refLZSJl1MqRQ6J8VP/+5XgiSxwZm+ucWQwOiCoNRGoKXQAwYGyCyF5pe+iUfcdGF+0CfCHFEDIBXltT9dL3HZ7b2Os8FCSiQZxEmzKUhydM4WWMpz4qoHrRBXX2m6fXooyczHXAXr8dDlwFL9XwY= References: [EMAIL PROTECTED] [EMAIL PROTECTED] X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2900.2670 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2900.2670 X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [90/4] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 18 Dec 2005 17:48:30.0140 (UTC) FILETIME=[41BFC7C0:01C603FB] Hi Elmar and all others, Will have a look and if I can't get it working I will post here! many thanks! - Original Message - From: Elmar Haneke [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, December 18, 2005 5:17 PM Subject: Re: [Asterisk-Users] Is this possible in Asterisk? Let's say an office has 20 people with 20 extensions and they want to enter a code on their phone when they leave for lunch and a voice will tel lthe caller like: The person you are calling is out of the office and will return at 1 pm. Is this something that is possible? I'm tot shure if there is any documentation regarding this specific topic. For Realisation I would suggest three parts: - Define an Pseudo-Number to be dialed on going to / coming back from lunch - The dialplan for this numbers should be modifiyng the state and playing an appropriate message. - The general dialplan has to read the current stat for the dialled target and act corresponding to this. - To Store the state there are DB-like functions in asterisk - or you can write an AGI. Elmar ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
Re: [Asterisk-Users] SIP and echo cancel
Dear pals, As a matter of fact im serious to know where is the source of echo in a pure VoIP connection, i think the most of echo problems come from hybrid circuits which are not an issue in pure VoIP sessions. Regards. --- M. Shokuie Nia. From: Luki [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] SIP and echo cancel Date: Sat, 17 Dec 2005 21:45:57 -0800 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc11-f10.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Sat, 17 Dec 2005 21:46:37 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id C2F0D2FC360;Sat, 17 Dec 2005 22:46:04 -0700 (MST) Received: from psmtp.com (exprod5mx123.postini.com [64.18.0.37])by lists.digium.com (Postfix) with SMTP id 891FC2FC355for asterisk-users@lists.digium.com;Sat, 17 Dec 2005 22:46:01 -0700 (MST) Received: from source ([64.233.184.204]) by exprod5mx123.postini.com([64.18.4.10]) with SMTP; Sat, 17 Dec 2005 23:46:00 CST Received: by wproxy.gmail.com with SMTP id i13so363732wrafor asterisk-users@lists.digium.com;Sat, 17 Dec 2005 21:45:57 -0800 (PST) Received: by 10.54.65.16 with SMTP id n16mr103252wra;Sat, 17 Dec 2005 21:45:57 -0800 (PST) Received: by 10.54.119.12 with HTTP; Sat, 17 Dec 2005 21:45:57 -0800 (PST) X-Message-Info: LGjzam7y+LspoNBQ/UDNaVvtG42BIjDD5YEU0+10Zno= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com;h=received:message-id:date:from:to:subject:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references;b=gqvH8XlVAQ3n1XKvRfERCjsu4Sw5l6Hs8ypOEpn/YT7wgM+Cu89/2rsLdnLtryXX6cFOSZewO4OJDWzUJ+TPn4iHLCivfP7HgWUtZndd45RaQyh1waRM3xz7TeC8eu4C76YDK1mY1V4lcH7UFNE0KO/a6mS/JqXL+GFHEYPJSBM= References: [EMAIL PROTECTED] X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [90/4] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 18 Dec 2005 05:46:37.0982 (UTC) FILETIME=[69B0F3E0:01C60396] Before I start hacking this into asterisk 1.2.1 I would like to known if others are running into this kind of problem ? Asterisk doesn't do any echo cancellation in the setup you describe; it just passes the audio data, and transcodes if necessary. The endpoints (the 841 phone and the 2002 and 3000 ATAs) are responsible for cancelling echo. The Sipura ATA's generally do a good job cancelling echo. You may want to play with the gain settings in the admin web config for the Sipura ATA. As far as the 841 is concerned, if the handset volume is too loud I noticed you may be getting acoustic echo. Hasn't been a problem for me for PSTN calls or SIP to SIP calls though. If you really want to patch asterisk to apply echo cancellation on the RTP stream on pure VoIP calls, that would be interesting to see how well it works. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and echo cancel
Dear pals, As a matter of fact im serious to know where is the source of echo in a pure VoIP connection, i think the most of echo problems come from hybrid circuits which are not an issue in pure VoIP sessions. Regards. --- M. Shokuie From: Luki [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] SIP and echo cancel Date: Sat, 17 Dec 2005 21:45:57 -0800 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc11-f10.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Sat, 17 Dec 2005 21:46:37 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id C2F0D2FC360;Sat, 17 Dec 2005 22:46:04 -0700 (MST) Received: from psmtp.com (exprod5mx123.postini.com [64.18.0.37])by lists.digium.com (Postfix) with SMTP id 891FC2FC355for asterisk-users@lists.digium.com;Sat, 17 Dec 2005 22:46:01 -0700 (MST) Received: from source ([64.233.184.204]) by exprod5mx123.postini.com([64.18.4.10]) with SMTP; Sat, 17 Dec 2005 23:46:00 CST Received: by wproxy.gmail.com with SMTP id i13so363732wrafor asterisk-users@lists.digium.com;Sat, 17 Dec 2005 21:45:57 -0800 (PST) Received: by 10.54.65.16 with SMTP id n16mr103252wra;Sat, 17 Dec 2005 21:45:57 -0800 (PST) Received: by 10.54.119.12 with HTTP; Sat, 17 Dec 2005 21:45:57 -0800 (PST) X-Message-Info: LGjzam7y+LspoNBQ/UDNaVvtG42BIjDD5YEU0+10Zno= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com;h=received:message-id:date:from:to:subject:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references;b=gqvH8XlVAQ3n1XKvRfERCjsu4Sw5l6Hs8ypOEpn/YT7wgM+Cu89/2rsLdnLtryXX6cFOSZewO4OJDWzUJ+TPn4iHLCivfP7HgWUtZndd45RaQyh1waRM3xz7TeC8eu4C76YDK1mY1V4lcH7UFNE0KO/a6mS/JqXL+GFHEYPJSBM= References: [EMAIL PROTECTED] X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [90/4] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 18 Dec 2005 05:46:37.0982 (UTC) FILETIME=[69B0F3E0:01C60396] Before I start hacking this into asterisk 1.2.1 I would like to known if others are running into this kind of problem ? Asterisk doesn't do any echo cancellation in the setup you describe; it just passes the audio data, and transcodes if necessary. The endpoints (the 841 phone and the 2002 and 3000 ATAs) are responsible for cancelling echo. The Sipura ATA's generally do a good job cancelling echo. You may want to play with the gain settings in the admin web config for the Sipura ATA. As far as the 841 is concerned, if the handset volume is too loud I noticed you may be getting acoustic echo. Hasn't been a problem for me for PSTN calls or SIP to SIP calls though. If you really want to patch asterisk to apply echo cancellation on the RTP stream on pure VoIP calls, that would be interesting to see how well it works. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM01B answering issue
Hi there, As a matter of fact its an awfull issue specially when you are using auto announcement systems. As far as i know its possible to solve this problem on analog boards with tone detection and VAD algorithems but dont think there is anything out there you can use with asterisk and TDM boards, Regards. --- M. Shokuie Nia From: chawki hammoud [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] TDM01B answering issue Date: Sun, 18 Dec 2005 00:19:06 -0800 (PST) MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc12-f3.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Sun, 18 Dec 2005 00:19:39 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id D01952FC481;Sun, 18 Dec 2005 01:19:08 -0700 (MST) Received: from psmtp.com (exprod5mx155.postini.com [64.18.0.224])by lists.digium.com (Postfix) with SMTP id 0C6572FC47Afor asterisk-users@lists.digium.com;Sun, 18 Dec 2005 01:19:06 -0700 (MST) Received: from source ([66.218.94.75]) by exprod5mx155.postini.com([64.18.4.10]) with SMTP; Sun, 18 Dec 2005 02:19:07 CST Received: (qmail 19126 invoked by uid 60001); 18 Dec 2005 08:19:06 - Received: from [66.198.34.52] by web90104.mail.scd.yahoo.com via HTTP;Sun, 18 Dec 2005 00:19:06 PST X-Message-Info: LGjzam7y+LuUSNZFNK+DqnOnhvKURAK0CzR62Nl8W34= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=s1024; d=yahoo.com;h=Message-ID:Received:Date:From:Subject:To:In-Reply-To:MIME-Version:Content-Type:Content-Transfer-Encoding;b=cR/AIO9hQQQijoKTqE2va8Ar4Gk9QGe7ZREcQP0Qzl1Hs4bFEQrAnc4h1x6okO9M2QG8er1/jH4+e328SmtMAqIvUE3czGC8GPiKEeNkdSA8PTqWCpbyoQ7sHh9pLlTNGla1AL4wNjnA6dshViBmvDH53K/6jSpDa+oHqg0ryeE=; X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [90/4] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 18 Dec 2005 08:19:40.0140 (UTC) FILETIME=[CAAF12C0:01C603AB] Hi: I saw a hardware in callshops that attached to analoge line and begin counting from the time call is answered to the time it hangup ,So is there ant hardware or a software added to asterisk to solve this answering issue? --- Steve Underwood [EMAIL PROTECTED] wrote: Andrew Kohlsmith wrote: On Saturday 17 December 2005 22:13, Eric ManxPower Wieling wrote: *sigh* Analog Zap FXO ports consider the call answered as soon as it's finished throwing the DTMF at the telco. This is because a Zap port CAN'T tell when an analog call has been answered. Bah, you're absolutely correct. I keep forgetting about POTS; I think PRI when I think Zap. Its not absolutely correct, but its relatively correct. :-) The above is true for most analogue lines around the world. However, there are some places which provide a positive answer indication on analogue lines. The form varies, but it is typically a reversal of line power, or a short timed break in line power. Similarly, while most of the world's analogue lines no longer provide a positive indication of hangup, some still do. Again, this is usually by reversal or a short timed break. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or
[Asterisk-Users] cdr mysql problem
Dear folks, I've just compiled asterisk-addon1.2.1 after installing MySQL and MySQl-devel packages. and adjust my cdr_mysql.conf to use the defined database using username and password. But as soon as starting asterisk i get error messages informing me of error, error message is as follows : cdr_addon_mysql.c : Failed to connect mysql database cdr on localhost and res_config_mysql.c : Failed to connect database server on . Im realy lost and dont know whats wrong. I've checked the connection to MySql in command line using the same user and host and its been connected without any problem. Anyone has any idea whats wrong here. Regards. --- M. Shokuie Nia. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to configure two Asterisk servers for one callcenter
Hi folks, As a matter of fact we had the same issue but seems there is no way than using two * servers and a SER server at the back end and handling everything there at SER server. I think its nice to take a look at wiki's site under the Asterisk at large topic. We just gave up and left our project coz of mass work needed. Regards. --- M. Shokuie Nia From: Tielin Xu [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to configure two Asterisk servers for one callcenter Date: Fri, 21 Oct 2005 09:39:49 -0700 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by mc11-f34.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Fri, 21 Oct 2005 09:41:46 -0700 Received: from [69.16.138.164] (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id D2E37FC18E;Fri, 21 Oct 2005 11:40:20 -0500 (CDT) Received: from psmtp.com (exprod5mx153.postini.com [64.18.0.222])by lists.digium.com (Postfix) with SMTP id 9D016FC186for asterisk-users@lists.digium.com;Fri, 21 Oct 2005 11:40:16 -0500 (CDT) Received: from source ([205.166.76.16]) by exprod5mx153.postini.com([64.18.4.10]) with SMTP; Fri, 21 Oct 2005 09:40:26 PDT Received: from Gateways-MTA by smtpgw1.nintendo.comwith Novell_GroupWise; Fri, 21 Oct 2005 09:40:04 -0700 Hi All: I have a situation to be resolved. Assume that one location call center with 150 agents. I have two asterisk servers to serve those 150 sip phones. The servers are connected to PSTN as 4 T1/PRI for each. I have a few questions, Can sip phones login to both servers for the call distribution? If yes, saying Asterisk server one sends a call to agent A, when another call comes to Asterisk server two, how does server two know agent A's status since the call connection information is stored in server one? does server one and two sharing the same MySQL database help this issue? Or I have to build a CTI server to control the call traffic for both servers, but it sounds to waste the queue facility on both servers. Please someone give me some ideas to resolve this situation. Thanks in advance. Tielin Xu ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma FXO/FXS cards?
Dear folk, You are right, seems sangoma is going to produce FXO/FXS cards but its still in the lab and not released yet but will do it in near future. Regards,M. Shokuie Nia,CEO,SENA Co. From:"Nathan C. Smith" [EMAIL PROTECTED]Reply-To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo:'Asterisk Users Mailing List - Non-CommercialDiscussion' asterisk-users@lists.digium.comSubject:RE: [Asterisk-Users] Sangoma FXO/FXS cards?Date:Wed, 12 Oct 2005 11:16:53 -0500MIME-Version:1.0Received:from lists.digium.com ([69.16.138.164]) by mc9-f5.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Wed, 12 Oct 2005 09:19:05 -0700Received:from [69.16.138.164] (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 4DD183FD335;Wed, 12 Oct 2005 11:16:54 -0500 (CDT)Received:from psmtp.com (exprod5mx132.postini.com [64.18.0.46])by lists.digium.com (Postfix) with SMTP id 821D23FD32Ffor asterisk-users@lists.digium.com;Wed, 12 Oct 2005 11:16:49 -0500 (CDT)Received:from source ([216.81.229.215]) by exprod5mx132.postini.com([64.18.4.10]) with SMTP; Wed, 12 Oct 2005 11:16:55 CDTReceived:from [10.1.1.2] ([10.1.1.2]:35084 "EHLO dsmexch.ipmvs.com")by mail.ipmvs.com with ESMTP id S53094AbVJLQQz (ORCPTrfc822;asterisk-users@lists.digium.com);Wed, 12 Oct 2005 11:16:55 -0500Received:by DSMEXCH with Internet Mail Service (5.5.2653.19)id RHY6CK8W; Wed, 12 Oct 2005 11:16:55 -0500They will be announced formally soon.-Original Message-From: Paul Dugas [mailto:[EMAIL PROTECTED]Sent: Wednesday, October 12, 2005 10:41 AMTo: Asterisk Mailing ListSubject: [Asterisk-Users] Sangoma FXO/FXS cards?Saw an ad in the latest Linux Journal for Sangoma's FXO/FXS analog cards.it was part of an ad for a reseller.I can't find anything on the resellerssite or Sangoma's site either.Did the ad jump the gun or someting?Isthis for real?Paul--Paul Dugas, Computer Engineer Dugas Enterprises, LLC[EMAIL PROTECTED] phone: 404-932-1355 522 Black Canyon Parkhttp://dugas.cc fax: 866-751-6494 Canton, GA 30114 USA--Onsite at GDOT W.Annex 404-463-2860 x199___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersDon't just search. Find. MSN Search Check out the new MSN Search! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using * and 3rd party GW together
Dear folks, Actually this is my first post here, so sorry for any inconvenience. Im planning for a solution a bit larger in scale than ususal. I'm goin to use * as a PSTN gateway with E1 links and use two other 3rd party Gateways for FXO lines. I should be able to switch from every incoming channel to any outgoing one and also to some SIP softphones. I planned to use SER as a sip server but really dont know were I should enforce my call routing mechanisms. Is SER applicable of doing that or should i write any application on the SER to do so ro is there any need for a softswitch at all? Any help and hints would be highly appreciated, M. Shokuie Nia.Don't just search. Find. MSN Search Check out the new MSN Search! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
