Re: [asterisk-users] 1.6.2.1: DTMF trouble with PSTN
On Fri, Feb 05, 2010 at 01:55:03PM -0500, sean darcy wrote: sean darcy wrote: Using 1.6.2.1 with a TDM400, attached to internal analog phones and PSTN. When I dial out to PSTN, I cannot send tones, like push 1 for something stupid. The call itself works, but the DTMF tones fail. -- Starting simple switch on 'DAHDI/1-1' -- Executing [6258...@internal:1] Answer(DAHDI/1-1, ) in new stack -- Executing [6258...@internal:2] Dial(DAHDI/1-1, DAHDI/4/ww2156258013) in new stack -- Called 4/ww2156258013 -- DAHDI/4-1 answered DAHDI/1-1 -- Native bridging DAHDI/1-1 and DAHDI/4-1 -- Hungup 'DAHDI/4-1' Any suggestions? sean This is DAHDI Tools Version - 2.2.1 Do DTMF tones work for others over dahdi? I'd file a bug, but I'd like to make sure it's not just my mistake. Do DTMFs work on a simple call to Asterisk? A simple IVR or VoiceMail. Can you recerd the audio before it gets to Asterisk? use: dahdi_monitor 1 -r rec.raw; sox -r 8000 -c 1 -s -w rec.wav Can you hear the DTMFs in rec.wav? Another thing to try: press a key for a few seconds. Do you hear it continously? -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
My inquiry is to understand how I could configure a system to do it. I have since learned that Asterisk has features in the code to do this (auto dial out, features.conf and .call files.) The 1 example is a bit extreme but it really does not matter what the number is for this. Dialogic has a system that provides notification so I am trying to see how I can build my own. Understanding simultaneous and concurrent call capabilities is important. Karl. Steve. Please don't bother me with you immature insults. On Sat, Feb 6, 2010 at 1:21 AM, Karl Fife karlf...@gmail.com wrote: Nice. :-) Didn't see that, I concede. - Original Message - From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 06, 2010 12:10 AM Subject: Re: [asterisk-users] Dial script On Fri, 5 Feb 2010, Karl Fife wrote: Try this: #rm -rf / Copycat! On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote: Is there any tested script available for this purpose. On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org=20 Sure. Add this to root's crontab: * * * * rm --farce --recursive / -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime
Hi, my Asterisk on debian lenny died after 80 days. server kernel: [7572666.186852] asterisk[3673]: segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l ibpthread-2.7.so[7f3b8e903000+16000] Anything what can be done to find out the reason? best regards Thomas -- _ -- 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] CONNECTEDLINE
Gentlemen, Did tryout CONNECTEDLINE function, was exactly what I have been looking for. But there are at least one thing I cant figure out. Did a very simple and stupid extension 0317998955 and ran a test. My phone (0317998975) dials 955, the display on my phone changes from 955 to Connected Line 955 when my call is answered, shouldn't the display on my phone change to Connected Line 0317998955? exten = 956,1,Goto(0317998956,1) exten = 0317998956,1,Set(CONNECTEDLINE(all)=Connected Line ) exten = 0317998956,n,Answer() exten = 0317998956,n,Wait(2) exten = 0317998956,n,Hangup() -- Executing [...@inputinterior.se:1] Goto(SIP/0317998975-0004, 0317998956,1) in new stack -- Goto (inputinterior.se,0317998956,1) -- Executing [0317998...@inputinterior.se:1] Set(SIP/0317998975-0004, CONNECTEDLINE(all)=Connected Line ) in new stack -- Executing [0317998...@inputinterior.se:2] Answer(SIP/0317998975-0004, ) in new stack -- Executing [0317998...@inputinterior.se:3] Wait(SIP/0317998975-0004, 2) in new stack -- Executing [0317998...@inputinterior.se:4] Hangup(SIP/0317998975-0004, ) in new stack == Spawn extension (inputinterior.se, 0317998956, 4) exited non-zero on 'SIP/0317998975-0004' Asterisk SVN-trunk-r245147M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC /Magnus-- _ -- 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] Dial script
On Sat, Feb 06, 2010 at 05:56:43AM -0500, Thomas Perron wrote: My inquiry is to understand how I could configure a system to do it. I have since learned that Asterisk has features in the code to do this (auto dial out, features.conf and .call files.) The 1 example is a bit extreme but it really does not matter what the number is for this. Dialogic has a system that provides notification so I am trying to see how I can build my own. Understanding simultaneous and concurrent call capabilities is important. Karl. Steve. Please don't bother me with you immature insults. To get notifications you can generally use the manager interface. Steve's first reply in this thread was actually a rather good one. I suggest you follow up there. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN on phones?
On Thu, Feb 04, 2010 at 10:46:49PM -0800, Dave Platt wrote: Anyway - is there someone out there that know the behaviour of OpenVPN in regards of retransmits and such? A VPN that retransmits will at some point hurt you if you transmit media over it, especially if you scale it up. OpenVPN is well-behaved in that way. It uses SSL over TCP for its administrative communications between peers - authentication and key exchange and other protocol negotiations. Nope. It can alternatively use TCP instead of UDP, but there's generally no such seperate administrative TCP communication. At least not in the default settings. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Website Down ?
Karl Fife wrote: Down for me too. It's fixed now, sorry for the disruption. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] Dial script
If Mr. Perron's request were truly academic, he probably would not have sent me an email off-list telling me to go fu¢k myself. - Original Message - From: Thomas Perron thomas.per...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 06, 2010 4:56 AM Subject: Re: [asterisk-users] Dial script My inquiry is to understand how I could configure a system to do it. I have since learned that Asterisk has features in the code to do this (auto dial out, features.conf and .call files.) The 1 example is a bit extreme but it really does not matter what the number is for this. Dialogic has a system that provides notification so I am trying to see how I can build my own. Understanding simultaneous and concurrent call capabilities is important. Karl. Steve. Please don't bother me with you immature insults. On Sat, Feb 6, 2010 at 1:21 AM, Karl Fife karlf...@gmail.com wrote: Nice. :-) Didn't see that, I concede. - Original Message - From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 06, 2010 12:10 AM Subject: Re: [asterisk-users] Dial script On Fri, 5 Feb 2010, Karl Fife wrote: Try this: #rm -rf / Copycat! On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote: Is there any tested script available for this purpose. On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org=20 Sure. Add this to root's crontab: * * * * rm --farce --recursive / -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---BeginMessage--- ---End Message--- -- _ -- 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] Dial script
On Sat, 6 Feb 2010, Thomas Perron wrote: My inquiry is to understand how I could configure a system to do it. I have since learned that Asterisk has features in the code to do this (auto dial out, features.conf and .call files.) The 1 example is a bit extreme but it really does not matter what the number is for this. Dialogic has a system that provides notification so I am trying to see how I can build my own. Understanding simultaneous and concurrent call capabilities is important. Karl. Steve. Please don't bother me with you immature insults. Maybe if you didn't top-post and actually read my replies you would have noticed that my posts were purely informative and directed at helping you. I was neither immature or insulting. Maybe if you had invested a little bit more effort you would have found your answers via Google. Although I already knew the answers, I entered appropriate queries into Google to provide you with links as you requested and to show you how easy it was -- feed a man and teach him to fish. Maybe if you had put enough effort into framing a specific question with sufficient detail the first time you wouldn't have appeared to be what others characterized as a wannabe spammer. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
Karl, You interpretation and assumption about my interest in a technical solution is simply wrong. I wanted positive feedback on the site. Your response was clearly negative and loaded with an insulting tone. I will continue to try and understand the internal protocols and technoligies that make this (and other) solutions work. My advise to you: Drop your ego at the front door. You are wrong. I am trying to learn. Your correct, Google has a lot of information. The purpose of the community is to support one another. Not to instigate like you have done. Later pal. Yes. I did give you an F.. You have no respect for others that are not as smart as you. On Sat, Feb 6, 2010 at 10:54 AM, Karl Fife karlf...@gmail.com wrote: If Mr. Perron's request were truly academic, he probably would not have sent me an email off-list telling me to go fu▎ myself. - Original Message - From: Thomas Perron thomas.per...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 06, 2010 4:56 AM Subject: Re: [asterisk-users] Dial script My inquiry is to understand how I could configure a system to do it. I have since learned that Asterisk has features in the code to do this (auto dial out, features.conf and .call files.) The 1 example is a bit extreme but it really does not matter what the number is for this. Dialogic has a system that provides notification so I am trying to see how I can build my own. Understanding simultaneous and concurrent call capabilities is important. Karl. Steve. Please don't bother me with you immature insults. On Sat, Feb 6, 2010 at 1:21 AM, Karl Fife karlf...@gmail.com wrote: Nice. :-) Didn't see that, I concede. - Original Message - From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 06, 2010 12:10 AM Subject: Re: [asterisk-users] Dial script On Fri, 5 Feb 2010, Karl Fife wrote: Try this: #rm -rf / Copycat! On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote: Is there any tested script available for this purpose. On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org=20 Sure. Add this to root's crontab: * * * * rm --farce --recursive / -- 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 -- 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 -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
On Sat, 6 Feb 2010, Thomas Perron wrote: Karl, [snip] Your correct, Google has a lot of information. You may want to go back and learn how to read replies so you can figure out who said what. Bottom-posting (since it is how most of us humans* read) will help. Not once did Karl mention Google. *) At the risk of showing my linguistic ignorance, are there any languages that write bottom to top? -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Still on spandsp/app_fax and T.38
Hi Vinícius, Comparing the successful log using FFA and the failing one using app_fax, I see the following: In the call using FFA, the far end system sends a V.29/9600bps FAX page, using T.38, which transfers cleanly, and the call ends. In the call using app_fax/spandsp, the SIP has a couple of key difference. Asterisk incorrectly adds T38FaxTranscodingMMR T38FaxTranscodingJBIG to the SDP. I don't know if that might be confusing the far end. The T.38 exchange starts in a very similar way to the FFA call, up to the middle of the TCF data. This should be all zeros, but half way through the far end suddenly starts sending rubbish. Spandsp correctly rejects this. The other system sends bad training data at V.27ter/4800bps. Spandsp rejects it. The other system sends clean training data at V.27ter/2400bps. Spandsp accepts it. The other system sends a page at V.27ter/2400bps. Spandsp accepts it. The only wrongdoing I see is the far end going weird in the training data, yet this doesn't happen with the FFA call. Whatever makes the far end fall over must be something fairly subtle. Since the 2 SDP lines I listed above shouldn't be there, I think they should be removed, and the test tried again. Much of the work in developing a robust T.38 implementation is working around all the crappy implementations out there. Steve On 02/06/2010 01:47 AM, Vinícius Fontes wrote: There you go: http://www.canall.com.br/wireshark_trace_t38_ffa.gz Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Steve Underwoodste...@coppice.org escreveu: Hi Vinícius, Asterisk + Spandsp is working correctly in that call. The other system sends bad training data at V.29/9600bps. Spandsp rejects it. The other system sends bad training data at V.27ter/4800bps. Spandsp rejects it. The other system sends clean training data at V.27ter/2400bps. Spandsp accepts it. The other system sends a page at V.27ter/2400bps. Spandsp accepts it. The bad training data is *really* bad. It should be 1.5s of all zero bits. It starts off with zeros, but after a few hundred milliseconds it changes to complete rubbish. I can't believe the Commetrex engine in Digium's FAX for Asterisk would accept this. Perhaps something subtle means they are sent the correct data. Can you send a wireshark log of a call with FAX for Asterisk? Steve On 02/06/2010 12:43 AM, Vinícius Fontes wrote: No problem, hosted it on my company's website: http://www.canall.com.br/wireshark_trace_t38.gz. Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Steve Underwoodste...@coppice.org escreveu: On 02/06/2010 12:01 AM, Vinícius Fontes wrote: Here's the packet trace I promised: http://www.zshare.net/download/72186098494e6f8c/. As this is a production system, there were a few calls along with the one that interests us. The one you're looking for is that from 5433142...@10.150.65.16 to 5421047...@10.153.66.146. The provider has the address 10.150.65.16 and my box has the address 10.153.66.146. Can you put the file somewhere that actually works. I've downloaded it 5 times now, and it has been cutoff at different points each time. These free file sharing services all seem to do this. Maybe they all run the same broken software. Steve -- _ -- 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] Dial script
Please use your quill and ink pot as well, and remember we can't insert blank paper into the front of a book, only writing on blank pages at the end. Oh wait, the advent of computers has allowed us to conveniently insert the most recent text at the TOP of a message, to prevent people from having to reread the same stuff every time. Just because this list's moderator has chosen bottom posting doesn't make it right, logical, common sense, etc. How about we don't belittle people who don't notice? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Saturday, February 06, 2010 12:19 PM To: Asterisk Users List Subject: Re: [asterisk-users] Dial script On Sat, 6 Feb 2010, Thomas Perron wrote: Karl, [snip] Your correct, Google has a lot of information. You may want to go back and learn how to read replies so you can figure out who said what. Bottom-posting (since it is how most of us humans* read) will help. Not once did Karl mention Google. *) At the risk of showing my linguistic ignorance, are there any languages that write bottom to top? -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Still on spandsp/app_fax and T.38
Steve Underwood wrote: The only wrongdoing I see is the far end going weird in the training data, yet this doesn't happen with the FFA call. Whatever makes the far end fall over must be something fairly subtle. Since the 2 SDP lines I listed above shouldn't be there, I think they should be removed, and the test tried again. In order to test that quickly, you could edit apps/app_fax.c and find the lines that set transcoding_mmr and transcoding_jbig to '1', and change them to '0' (zero) or remove them. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] Dial script
On Sat, 6 Feb 2010, Michelle Dupuis wrote: [snip] Oh wait, the advent of computers has allowed us to conveniently insert the most recent text at the TOP of a message, to prevent people from having to reread the same stuff every time. A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? Just because you can do something with a computer doesn't mean you should. Just because this list's moderator has chosen bottom posting doesn't make it right, logical, common sense, etc. (These lists are not moderated...) The list owner's choice did not make bottom posting right, logical and common sense. It was all of those things already. The list owner just made the right choice. How about we don't belittle people who don't notice? (I don't think I belittled anyone for top-posting.) We all make mistakes, but once we are informed, continuing to make the same mistakes indicates either a lack of consideration or stupidity or both. Simple courtesy would be to only include relevant sections of previous posts and to reply below the quoted text. If you don't like the rules of the playground, find another playground. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
Here's a .sig from the m...@openbsd list, (which I couldn't resist top-posting.) A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing in e-mail? On Sat, Feb 06, 2010 at 10:13:42AM -0800, Steve Edwards wrote: On Sat, 6 Feb 2010, Michelle Dupuis wrote: [snip] Oh wait, the advent of computers has allowed us to conveniently insert the most recent text at the TOP of a message, to prevent people from having to reread the same stuff every time. A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? Just because you can do something with a computer doesn't mean you should. Just because this list's moderator has chosen bottom posting doesn't make it right, logical, common sense, etc. (These lists are not moderated...) The list owner's choice did not make bottom posting right, logical and common sense. It was all of those things already. The list owner just made the right choice. How about we don't belittle people who don't notice? (I don't think I belittled anyone for top-posting.) We all make mistakes, but once we are informed, continuing to make the same mistakes indicates either a lack of consideration or stupidity or both. Simple courtesy would be to only include relevant sections of previous posts and to reply below the quoted text. If you don't like the rules of the playground, find another playground. -- Barry -- _ -- 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] {top|bottom|interleaved} posting, once again
Steve Edwards schrieb: On Sat, 6 Feb 2010, Michelle Dupuis wrote: [snip] Oh wait, the advent of computers has allowed us to conveniently insert the most recent text at the TOP of a message, to prevent people from having to reread the same stuff every time. A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? Just because you can do something with a computer doesn't mean you should. Just because this list's moderator has chosen bottom posting doesn't make it right, logical, common sense, etc. (These lists are not moderated...) The list owner's choice did not make bottom posting right, logical and common sense. It was all of those things already. The list owner just made the right choice. How about we don't belittle people who don't notice? (I don't think I belittled anyone for top-posting.) We all make mistakes, but once we are informed, continuing to make the same mistakes indicates either a lack of consideration or stupidity or both. Simple courtesy would be to only include relevant sections of previous posts and to reply below the quoted text. If you don't like the rules of the playground, find another playground. Actually bottom-posting without trimming anything (SCNR) is about as annoying as top-posting. Interleaved posting is fine, quoting just enough text so everyone can understand the context. But I have almost given up on this endless fight. :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- -- _ -- 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] CONNECTEDLINE
You should take a look and see if any SIP packets are going out that mention Connected Line 0317998955 as either something is or isn't sent out from the asterisk server. On Sat, Feb 6, 2010 at 4:30 AM, Magnus Benngård magnu...@inputinterior.se wrote: Gentlemen, Did tryout CONNECTEDLINE function, was exactly what I have been looking for. But there are at least one thing I cant figure out. Did a very simple and stupid extension 0317998955 and ran a test. My phone (0317998975) dials 955, the display on my phone changes from 955 to Connected Line 955 when my call is answered, shouldn't the display on my phone change to Connected Line 0317998955? exten = 956,1,Goto(0317998956,1) exten = 0317998956,1,Set(CONNECTEDLINE(all)=Connected Line 0317998955) exten = 0317998956,n,Answer() exten = 0317998956,n,Wait(2) exten = 0317998956,n,Hangup() -- Executing [...@inputinterior.se:1] Goto(SIP/0317998975-0004, 0317998956,1) in new stack -- Goto (inputinterior.se,0317998956,1) -- Executing [0317998...@inputinterior.se:1] Set(SIP/0317998975-0004, CONNECTEDLINE(all)=Connected Line 0317998955) in new stack -- Executing [0317998...@inputinterior.se:2] Answer(SIP/0317998975-0004, ) in new stack -- Executing [0317998...@inputinterior.se:3] Wait(SIP/0317998975-0004, 2) in new stack -- Executing [0317998...@inputinterior.se:4] Hangup(SIP/0317998975-0004, ) in new stack == Spawn extension (inputinterior.se, 0317998956, 4) exited non-zero on 'SIP/0317998975-0004' Asterisk SVN-trunk-r245147M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC /Magnus -- _ -- 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 -- _ -- 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] Asterisk going down
May as well check anything that you can. I usually start with memtest86 when i'm curious about system stability. On Fri, Feb 5, 2010 at 2:59 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my friends, My asterisk is going down randomly, following you will find some errors that i could see in the /var/log/asterisk/message at the moment of the crash: [Feb 5 10:32:45] WARNING[6519] chan_sip.c: Maximum retries exceeded on transmission 1850202...@10.4.1.152 for seqno 21 (Critical Response) [Feb 5 10:32:45] WARNING[6519] chan_sip.c: Hanging up call 1850202...@10.4.1.152 - no reply to our critical packet. [Feb 5 10:33:04] NOTICE[6519] chan_sip.c: Call from '346' to extension '3415554' rejected because extension not found. [Feb 5 10:35:31] NOTICE[6519] chan_sip.c: Disconnecting call 'SIP/301-09ad3be8' for lack of RTP activity in 301 seconds [Feb 5 10:36:17] NOTICE[6519] chan_sip.c: Disconnecting call 'SIP/317-b7735220' for lack of RTP activity in 301 seconds [Feb 5 10:38:19] NOTICE[6519] chan_sip.c: Peer '353' is now Reachable. (1ms / 2000ms) [Feb 5 10:42:59] NOTICE[6519] chan_sip.c: Peer '353' is now Reachable. (7ms / 2000ms) [Feb 5 10:51:09] NOTICE[6519] chan_sip.c: Peer '358' is now Reachable. (1ms / 2000ms) [Feb 5 10:53:08] NOTICE[6519] chan_sip.c: Peer '366' is now UNREACHABLE! Last qualify: 108 But later, at 2 pm, Asterisk went down again but with no weird message in /var/log/asterisk/message (just some unreachable messages of some extensions that has always been in the console since i installed Asterisk, but it never crash Asterisk untill last weeks ago): [Feb 5 13:54:11] NOTICE[6536] chan_sip.c: Peer '328' is now Reachable. (1ms / 2000ms) [Feb 5 13:55:18] NOTICE[6536] chan_sip.c: Registration from 'sip:3...@10.4.1.6:5060' failed for '10.4.2.3' - No matching peer found [Feb 5 13:57:40] NOTICE[6536] chan_sip.c: Call from '346' to extension '04265417457' rejected because extension not found. [Feb 5 13:59:15] NOTICE[6536] chan_sip.c: Peer '341' is now Reachable. (2ms / 2000ms) [Feb 5 13:59:25] NOTICE[6536] chan_sip.c: Peer '328' is now Reachable. (1ms / 2000ms) [Feb 5 14:01:43] NOTICE[6536] chan_sip.c: Peer '339' is now UNREACHABLE! Last qualify: 101 [Feb 5 14:04:22] NOTICE[6536] chan_sip.c: Peer '339' is now Reachable. (44ms / 2000ms) [Feb 5 14:04:39] NOTICE[6536] chan_sip.c: Peer '328' is now Reachable. (1ms / 2000ms) [Feb 5 14:09:53] NOTICE[6536] chan_sip.c: Peer '328' is now Reachable. (1ms / 2000ms) I could not make any call, neither internall nor to the pstn, what could be happening here my friends? what should i check in the Asterisk server? is this a network problem? memmory or cpu problems? Thanks in advance for your help -- _ -- 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 -- _ -- 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] {top|bottom|interleaved} posting, once again
At 11:17 AM 2/6/2010, you wrote: Actually bottom-posting without trimming anything (SCNR) is about as annoying as top-posting. Interleaved posting is fine, quoting just enough text so everyone can understand the context. Seems to me if you trim so that only the minimum amount is left it hardly matters. I don't know about everyone else, but I've already read all the prior posts and only need the smallest bit of reference to connect the answer in my mind. I never read bottom posts that are more than 20 lines from the top unless I figure there's a reason I need to. Life is just to short. Ira -- _ -- 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] Dial script
Thomas, Yes you can do this. I actually have done this and run it as a service under the name Meetmecall. I use MSN as the user interface to record the message, create phone lists of the numbers that has to be called and to actually schedule and perform the delivery. It is possible to use it for spam but the customers I have use it to notify, remember, offer or let the callee know about something relevant, but always as part of an already existing relation. With some extra parameters used, you can start a groupcall and use all the other Asterisk magic available like doing a questionarry using a smart IVR etc. etc. I can think about a long list of useful use of this service. I have no idea about the rules and legislation in other countries but in Holland you will end up with serious trouble and extreme high penalties to pay if you actually use it for spamming. I will not send you a copy of the solution but it is based on the use of call files pointing to local channels/extensions where the Asterisk magic is combined in a working (and I think clever) way. The CDR isn't perfect but disable and enable CDR at the proper points in the dial plan and clever use of the USERFIELD variable will result in useable data for billing the users. The CDR shows that most callees, listen to the message until it ends and yes, sometime there are complaints about the use but that is very rare. About the scheduling of the calls to make. It is not Asterisk that limits you. Far before reaching the limits of Asterisk it will be the bandwidth available and the SIP trunk provider that normally doesn't allow an endless number of concurrent calls. When I started this I was working for a Norwegian company offering the dial tone on the internet and I had a server almost directly connected to the backbone of internet with more or less endless bandwidth. I did some stress testing of a call center solution and 80 concurrent calls wasn't a problem and my guess is that you can far beyond 80 calls. It is wise to move the call files one after the other or one batch after the other. Moving large numbers of call files into /var/spool/asterisk/ outgoing might sometimes result in unexpected and not intended results. There are other scenarios but this was my choice. 10.000 calls will take some time but with a 30 seconds message, 20 concurrent calls and 10 seconds average to dial after around 5,5 hours the last phone call will be dialed. If the message is just 15 seconds it will take around 3,5 hours. If you want to deliver in short time, like 10 minutes, you really have to scale up to 420 concurrent calls which doesn't sound doable unless you have real serious budgets. If you put everything in place at your side you will probably run into constraints of the SIP provider and the interconnection to the pstn. btw: Asterisk has the potential to build lots of evil features and lots of standard features can be used in an evil way. Personally I think it is kind of strange that, if a question is asked, one has to explain why the answer is not mend for evil use. We don't have to help someone out and we can refuse because of lots of reasons: no time, not an interesting question, not a single sign of any effort by the one asking the question, not willing to give something away that costs lots of time and energy, the feeling that it will be used in an evil way etc. etc. I think the tone and the content of this discussion harms the Asterisk community as a whole. with friendly regards, Erik de Wild Tripple-o: your asterisk migration partner the Netherlands On 6 feb 2010, at 03:54, Thomas Perron wrote: Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. -- _ -- 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 -- _ -- 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] {top|bottom|interleaved} posting, once again
Funny that a small matter like top | bottom | interleaved posting can lead to a situation that is referred to as a fight . I agree with Ira that keeping all the original lines in place is very annoying when there are more lines then needed to pick up the discussion. I seem to be a top poster by nature, it never crossed my mind that this could be part of a fight or be annoying to others . Sometimes I interleave but I never post at the bottom. Lets take our bits of freedom and consider how to post (not, top, bottom or interleave) as part of our personal style of communicating ;-) erik On 6 feb 2010, at 22:04, Ira wrote: At 11:17 AM 2/6/2010, you wrote: Actually bottom-posting without trimming anything (SCNR) is about as annoying as top-posting. Interleaved posting is fine, quoting just enough text so everyone can understand the context. Seems to me if you trim so that only the minimum amount is left it hardly matters. I don't know about everyone else, but I've already read all the prior posts and only need the smallest bit of reference to connect the answer in my mind. I never read bottom posts that are more than 20 lines from the top unless I figure there's a reason I need to. Life is just to short. Ira -- _ -- 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 -- _ -- 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] 500 Internal Server Error on Cisco 7940 after INVITE
I'm using P0S3-8-12-00 and things are working great with piaf and asterisk 1.4. Drop me a direct line to email, and I can send you my configs and such if that would help diag things for you. On Feb 3, 2010 3:02 PM, i...@comtek.co.uk i...@comtek.co.uk wrote: David Gibbons wrote: snip I have upgraded the phones to the most recent firmware (POS3-08-11-0... Thats straight out of that section. Its the most recent SIP firmware I could find. Application Load ID: 'POS3-08-11-00'. Boot Load ID: PC03A300 DSP Load ID 4.0(5.0)[A0] It seems to mean 8.11. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7960g_7940g/firmware/sip/8_11/english/release/notes/796040sip_811.html#wp1099767 Thanks, Ian -- === Ian Crowther ... -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list T... -- _ -- 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] 2 Asterisk Boxes, Single Voicemail
Can the message-waiting indicator be activated on a SIP phone registered to system B, if the voicemail resides on system A? It was a while ago I wrote this (come to think of it I might even have copied it from somewhere), and I don't remember if its in a working state: - for host in 10.200.3.100; do ###Voicemail MWI### find /var/spool/asterisk/voicemail/ -wholename '/var/spool/asterisk/voicemail/voicemail/*/INBOX/msg.txt' -print| ssh -i /etc/asterisk/syncKey aster...@$host \ 'find /var/spool/asterisk/voicemail -wholename /var/spool/asterisk/voicemail/voicemail/*/INBOX/msg.txt -print0|'\ ' xargs -0 rm -f; '\ 'while read FILE; do '\ ' mkdir -p `dirname $FILE`; '\ ' touch $FILE; '\ 'done' - From memory, the MWI is basically a text file. The above bash script, (called in a cron job on our voicemail server) logs into other servers, deletes all MWI files and then touches files to match the MWI state of the voicemail server. We don't actually have any phones registered with our second server yet, so consider the above untested! Ian -- _ -- 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] Dial script
Thank you for your interesting comments. On Sat, Feb 6, 2010 at 4:14 PM, Erik de Wild: Tripple-o i...@tripple-o.nl wrote: Thomas, Yes you can do this. I actually have done this and run it as a service under the name Meetmecall. I use MSN as the user interface to record the message, create phone lists of the numbers that has to be called and to actually schedule and perform the delivery. It is possible to use it for spam but the customers I have use it to notify, remember, offer or let the callee know about something relevant, but always as part of an already existing relation. With some extra parameters used, you can start a groupcall and use all the other Asterisk magic available like doing a questionarry using a smart IVR etc. etc. I can think about a long list of useful use of this service. I have no idea about the rules and legislation in other countries but in Holland you will end up with serious trouble and extreme high penalties to pay if you actually use it for spamming. I will not send you a copy of the solution but it is based on the use of call files pointing to local channels/extensions where the Asterisk magic is combined in a working (and I think clever) way. The CDR isn't perfect but disable and enable CDR at the proper points in the dial plan and clever use of the USERFIELD variable will result in useable data for billing the users. The CDR shows that most callees, listen to the message until it ends and yes, sometime there are complaints about the use but that is very rare. About the scheduling of the calls to make. It is not Asterisk that limits you. Far before reaching the limits of Asterisk it will be the bandwidth available and the SIP trunk provider that normally doesn't allow an endless number of concurrent calls. When I started this I was working for a Norwegian company offering the dial tone on the internet and I had a server almost directly connected to the backbone of internet with more or less endless bandwidth. I did some stress testing of a call center solution and 80 concurrent calls wasn't a problem and my guess is that you can far beyond 80 calls. It is wise to move the call files one after the other or one batch after the other. Moving large numbers of call files into /var/spool/asterisk/ outgoing might sometimes result in unexpected and not intended results. There are other scenarios but this was my choice. 10.000 calls will take some time but with a 30 seconds message, 20 concurrent calls and 10 seconds average to dial after around 5,5 hours the last phone call will be dialed. If the message is just 15 seconds it will take around 3,5 hours. If you want to deliver in short time, like 10 minutes, you really have to scale up to 420 concurrent calls which doesn't sound doable unless you have real serious budgets. If you put everything in place at your side you will probably run into constraints of the SIP provider and the interconnection to the pstn. btw: Asterisk has the potential to build lots of evil features and lots of standard features can be used in an evil way. Personally I think it is kind of strange that, if a question is asked, one has to explain why the answer is not mend for evil use. We don't have to help someone out and we can refuse because of lots of reasons: no time, not an interesting question, not a single sign of any effort by the one asking the question, not willing to give something away that costs lots of time and energy, the feeling that it will be used in an evil way etc. etc. I think the tone and the content of this discussion harms the Asterisk community as a whole. with friendly regards, Erik de Wild Tripple-o: your asterisk migration partner the Netherlands On 6 feb 2010, at 03:54, Thomas Perron wrote: Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. -- _ -- 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 -- _ -- 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 -- _ -- 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] A2Billing and other prepaid Billing like ASTCC, who is better?
Hi All; I used A2Billing, basically it is nice and fine, but management possibilities is not that rich, so a lot of staff are need to be repeated that let the admin facing a problem of the needed time to do the task. Anyone advise for another open source prepaid billing that is rich by the management features? Also, I hope to find an open source Billing (prepaid and postpaid) that can work with Asterisk and Gnugk at the same time (instead of using one billing for asterisk and one billing for gnugk, specially that gnugk is good for h323 functionalities that are missing in asterisk). Appreciate any help and advise in that direction. Regards Bilal -- _ -- 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] A2Billing and other prepaid Billing like ASTCC, who is better?
Why not pay for missing feature and contribute them to the project. It's a very good product. On 06/02/2010, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I used A2Billing, basically it is nice and fine, but management possibilities is not that rich, so a lot of staff are need to be repeated that let the admin facing a problem of the needed time to do the task. Anyone advise for another open source prepaid billing that is rich by the management features? Also, I hope to find an open source Billing (prepaid and postpaid) that can work with Asterisk and Gnugk at the same time (instead of using one billing for asterisk and one billing for gnugk, specially that gnugk is good for h323 functionalities that are missing in asterisk). Appreciate any help and advise in that direction. Regards Bilal -- _ -- 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 -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com -- _ -- 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] Still on spandsp/app_fax and T.38
Will do. You guys will have my feedback on monday. If everything goes okay with that change, I'll post a patch on Mantis. Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Kevin P. Fleming kpflem...@digium.com escreveu: Steve Underwood wrote: The only wrongdoing I see is the far end going weird in the training data, yet this doesn't happen with the FFA call. Whatever makes the far end fall over must be something fairly subtle. Since the 2 SDP lines I listed above shouldn't be there, I think they should be removed, and the test tried again. In order to test that quickly, you could edit apps/app_fax.c and find the lines that set transcoding_mmr and transcoding_jbig to '1', and change them to '0' (zero) or remove them. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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] Still on spandsp/app_fax and T.38
Vinícius Fontes wrote: Will do. You guys will have my feedback on monday. If everything goes okay with that change, I'll post a patch on Mantis. No need for the patch; it's already on my radar, and if you confirm that it actually solves an interop problem, I'll commit the update to the various branches it belongs in. I'd still like to hear from Steve Underwood if I misinterpreted the MMR/JBIG transcoding function calls in spandsp that led me to enabling these features in the first place... -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] Dial script
On Fri, Feb 5, 2010 at 10:08 PM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 5 Feb 2010, Thomas Perron wrote: Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. Do you mean the dialed numbers are in sequence like 555-555-0001, 555-555-0002* or do you mean dialing the numbers one after the other from a list of customers you have a pre-existing business relationship with? I'm guessing you don't want to sit there from start to finish :) You'll need some sort of database to keep track of which numbers have been called and where to start the next time. You could write a program to create call files or you could write a program to connect to your Asterisk server using AMI and issue originate commands. *) Probably illegal in the United States and any other civilized country. Robo calls for commercial purpose is illegal in the US. So is dialing 10k numbers in sequence, you would have to scrub it first against the DNC list. -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
On Fri, Feb 5, 2010 at 9:54 PM, Thomas Perron thomas.per...@gmail.com wrote: Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. There is a vbs script posted on the Wiki (voip-info.org) that could get you started. -- _ -- 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 -- _ -- 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] Dial script
Sorry Thomas, but I have to agree with Karl on this one. We have been here long enough to smell whats behind a posters motives. There are the guys that make a living out of this one way or another, usually selling a usefull service. Which is quite noticeable on the type of posts those people make and questions they ask. Then there are the ones that decided they have to buy a PBX for their small business and know how to use the Interweb and bumped across this free software and barge in here and bombard the list with questions. Then there are the academic ones, those who just want to learn (some call em white hat hackers/freakers). Then there are the ones that want to use asterisk as a single purpose thingy like calling card platform etc. Those are all legit motives and ALL noticeable from what they post, ask and area of expertise within asterisk. You my friend belong to a group called scammers or spammers, there is no legit reason whatsoever to dial 10,000 numbers in sequence, none whatsoever. If your post would have been really academic then it would have shown some clues that you might know what asterisk is about, but you have shown you know nothing about asterisk and the only interest you have in asterisk is to war dial. Yes and that all from my good sens of smell. On Sat, Feb 6, 2010 at 5:56 AM, Thomas Perron thomas.per...@gmail.com wrote: My inquiry is to understand how I could configure a system to do it. I have since learned that Asterisk has features in the code to do this (auto dial out, features.conf and .call files.) The 1 example is a bit extreme but it really does not matter what the number is for this. Dialogic has a system that provides notification so I am trying to see how I can build my own. Understanding simultaneous and concurrent call capabilities is important. Karl. Steve. Please don't bother me with you immature insults. On Sat, Feb 6, 2010 at 1:21 AM, Karl Fife karlf...@gmail.com wrote: Nice. :-) Didn't see that, I concede. - Original Message - From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 06, 2010 12:10 AM Subject: Re: [asterisk-users] Dial script On Fri, 5 Feb 2010, Karl Fife wrote: Try this: #rm -rf / Copycat! On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote: Is there any tested script available for this purpose. On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org=20 Sure. Add this to root's crontab: * * * * rm --farce --recursive / -- 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 -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
Wow, after my post I decided to use Google: http://www.google.com/search?hl=ensource=hpq=Thomas+Perron+site%3Adigium.comaq=faqi=oq= 3rd result pointed to this: http://lists.digium.com/pipermail/asterisk-users/2009-December/242357.html That was back in December. I guess my nose didn't fool me. While some posts of you showed that you havn't yet figured out how to use asterisk but you do see the potential it has in creating a business, the answers given back then should have pointed you in the right direction if you were really truly an academic person. On Sat, Feb 6, 2010 at 10:30 PM, C F shma...@gmail.com wrote: Sorry Thomas, but I have to agree with Karl on this one. We have been here long enough to smell whats behind a posters motives. There are the guys that make a living out of this one way or another, usually selling a usefull service. Which is quite noticeable on the type of posts those people make and questions they ask. Then there are the ones that decided they have to buy a PBX for their small business and know how to use the Interweb and bumped across this free software and barge in here and bombard the list with questions. Then there are the academic ones, those who just want to learn (some call em white hat hackers/freakers). Then there are the ones that want to use asterisk as a single purpose thingy like calling card platform etc. Those are all legit motives and ALL noticeable from what they post, ask and area of expertise within asterisk. You my friend belong to a group called scammers or spammers, there is no legit reason whatsoever to dial 10,000 numbers in sequence, none whatsoever. If your post would have been really academic then it would have shown some clues that you might know what asterisk is about, but you have shown you know nothing about asterisk and the only interest you have in asterisk is to war dial. Yes and that all from my good sens of smell. On Sat, Feb 6, 2010 at 5:56 AM, Thomas Perron thomas.per...@gmail.com wrote: My inquiry is to understand how I could configure a system to do it. I have since learned that Asterisk has features in the code to do this (auto dial out, features.conf and .call files.) The 1 example is a bit extreme but it really does not matter what the number is for this. Dialogic has a system that provides notification so I am trying to see how I can build my own. Understanding simultaneous and concurrent call capabilities is important. Karl. Steve. Please don't bother me with you immature insults. On Sat, Feb 6, 2010 at 1:21 AM, Karl Fife karlf...@gmail.com wrote: Nice. :-) Didn't see that, I concede. - Original Message - From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 06, 2010 12:10 AM Subject: Re: [asterisk-users] Dial script On Fri, 5 Feb 2010, Karl Fife wrote: Try this: #rm -rf / Copycat! On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote: Is there any tested script available for this purpose. On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org=20 Sure. Add this to root's crontab: * * * * rm --farce --recursive / -- 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 -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] large scale paging
For a case like this I would go with overhead paging. On Fri, Feb 5, 2010 at 4:50 PM, Mark Willis marksli...@markwillis.net wrote: Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500 near-simultaneous INVITEs being sent. Can the SIP channel handle that? Any suggestions or war stories are appreciated. Mark Willis Cartama Consulting LLC 210 698 5097 -- _ -- 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 -- _ -- 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
