[asterisk-users] Voip Users Conference moves up to 12:00 EST
The VOIP Users Conference (aka asterisk users conference, asterisk being a registered trademark of Digium) has become a kind of international users group as well, with a new Ning network at * http://food4wine.ning.com The Ning site allows posting of files, images, videos and text of possible interest to voip and asterisk users. Anyone with interest in either is welcome to join. Also, the Friday event will now be at 12 Noon ET, 9 AM Pacific, 17:00 GMT in order to cover the widest area possible. Lots of interest from India where it's around 11 PM. Asia, Australia and New Zealand are unfortunately out of range but we're willing to do a second conference for them if there is enough interest. That could be around 0800 UTC (early evening down under, right?). Reply to this off list if you're in that area and interested with your time preference. the VOIP Users Conference. http://www.VoipUsersConference.org IRC on freenode.net #voip-users-conference In a nutshell At 12:00 PM EST, * PSTN in the US, Call (724) 444-7444 * SIP sip:[EMAIL PROTECTED] After the call connects, enter the conference id: 22622# PIN# if your PIN == your CallerID, enter 2# instead of PIN. if you have no PIN, enter 1# The PIN just helps us see who you are to call on you when you want to talk. No personal info is kept. http://groups.google.com/group/Voip-Users-Conference ___ --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] How to bridge two connected calls
Nick Seraphin wrote: ... Once the incoming caller is in the dialplan, issue a Dial() command using both the m option and the M() option, in addition to any other options you would normally be using for Dial(). The m option will play music on hold while the Dial() command does it's thing. ... It works like a charm. Thanks a lot for the precious hint. Alberto. ___ --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] Digium and Asterisk
Michael J. Liberatore wrote: There are many reasons to buy digium cards, mainly digiums owner creating asterisk and all. so when i asked myself your question when starting with * i bought them. well, i myself have had bad luck with their products,2 failed out of warranty, and the others have bad echo and random weird problems. i myself switched to sangoma and have had much better success. they are even more than digium cards but work great. oh and dont even waste your time and money, get echo cancellation on any fxo cards, its the only way to make sure you get good sound quality. -mike I just want to add - for the poor amongst us, that if you use the OSLEC echo canceller with cheap x100p and (from what others have said) other analogue cards, you get excellent echo cancellation. On my cheap card, echo was terrible with the standard EC in the zaptel package. Using OSLEC instead, the echo disappeared. Completely. Al -- The way out is open! http://www.theopensourcerer.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
[asterisk-users] dial in group
I have a TDM400 with all FXO module in it. Only one channel (say channel 3) is plugged to PSTN. In my understand, a dial command Dial(zap/g1/12345677) should search an available channel, which is 3, in group 1 to make a call. However, I found that it will still use channel 1 to make call even it hasn't plugged to the PSTN. Below are the conf files. --zapata.conf-- group=1 signalling=fxs_ks context=incoming channel = 1-8 ___ --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+HylaFAX+SpanDSP+IAXmodem tutorial.
On Saturday 24 November 2007 00:16:11 Steve Totaro wrote: Alex Balashov wrote: Asterisk 1.4 does have this ability natively. However, it is somewhat limited in its flexibility / in terms of what I can do with it, and I have gotten reports that HylaFAX works better. I haven't actually done a comparison between the two. Being someone who hates 1.2, I was strongly tempted to go this route, though. Why would anyone hate the most stable version of Asterisk? What is ABE using these days? If it is not 1.4, I wonder why? Maybe so all the free developers and eager and silly early adopters can iron out the bugs, submit patches and sign away their rights. I am sure if they are not using 1.4 it probably has something to do with reliability and the costs of supporting that release. Any other theories? Yeah, that version C is currently in beta and is very close to release. ABE has to be put through its paces before release and that takes time. I'm sorry if that seems like evidence that Digium isn't supporting 1.4, but it simply isn't true. -- Tilghman ___ --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] OT Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.
Tilghman Lesher wrote: On Saturday 24 November 2007 00:16:11 Steve Totaro wrote: Alex Balashov wrote: Asterisk 1.4 does have this ability natively. However, it is somewhat limited in its flexibility / in terms of what I can do with it, and I have gotten reports that HylaFAX works better. I haven't actually done a comparison between the two. Being someone who hates 1.2, I was strongly tempted to go this route, though. Why would anyone hate the most stable version of Asterisk? What is ABE using these days? If it is not 1.4, I wonder why? Maybe so all the free developers and eager and silly early adopters can iron out the bugs, submit patches and sign away their rights. I am sure if they are not using 1.4 it probably has something to do with reliability and the costs of supporting that release. Any other theories? Yeah, that version C is currently in beta and is very close to release. ABE has to be put through its paces before release and that takes time. I'm sorry if that seems like evidence that Digium isn't supporting 1.4, but it simply isn't true. I am not implying that they do not support 1.4 but you did prove my point that 1.2 is more stable and 1.4 has not been Put through its paces. I would not recommend running a high volume call center on it. Sure, if your PBX takes 50 or so calls a day, it's probably wonderful. If you have a 15,000 average call volume a day and 400+ agents, hm, I think 1.2 might be a little wiser choice. Just personal notes from the trenches, not from the media machines or talking heads. Thanks, 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] TDM808B 8 port FXO setting problem
Hi all, I have seen, in the past, some engineers using a common wire for two pairs (2 subscribers = 3 wires only, not 4 !) to win some subscribers more than the reality can. And like this, you obtain crosstalk, of course. Are you using any kind of echocanceller in front of your TDM808B ? Best Regards, Francois BERGERET F6HQZ France -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] la part de Gustavo Cordeiro Envoyé : vendredi 23 novembre 2007 12:20 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] TDM808B 8 port FXO setting problem Ask for your telco to enable polarity reversal for these lines. Then enable hanguponpolarityswitch in your zapata.conf. About crosstalk I don't have any idea. Maybe a telco or cabling problem... Sds, Gustavo Date: Fri, 23 Nov 2007 02:33:34 -0800 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM808B 8 port FXO setting problem Dear all I have TDM808B 8 port FXO it is configure perfectly but i got some problem of incomming phone Hangup and callerid display problem i am going to explain you the issue i have install asterisk 1.4 and i have 100 of SIP phone now everything is fine but problem is when i incoming call on FXO and dial sip extention SIP phone is rining but when i disconnect my incoming phon from mobile ( i hangup my cell phone ) still my sip phone rining not disconnect notification reached to my sip phone so what is the problem and one more thing some time i got cross talk on phone on Zap channel so is it timeing problme of card or anyconfiguration problem wait for reply PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now. -- Receba as últimas notícias do Brasil e do mundo direto no seu Messenger! É GRÁTIS! Assine já! ___ --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] Annoying PRI Channels Restarting Message
Michael Collins wrote: Is there a reason it resets? Aka does it serve any kind of purpose? Just curious: what protocol variant (i.e. 4/5ESS, DMS, NI2, etc.) are you using? Also, which carrier? Finally, have you turned on PRI debugging to see if it is the telco that is requesting the restart? In some cases the telco will send out a PRI message like 'service' (i.e. service request) to which the CPE will need to respond with a service ack message. Not all telcos behave the same with respect to so-called maintenance messages, so you might want to follow up with the carrier just to be sure nothing is wrong. Probably nothing is wrong but it can't hurt to check. -MC P.S. - the messages might be annoying, but if you've ever had PRI issues then those messages become comforting! It is Asterisk or more specifically Zaptel that causes the resets defined by the resetinterval variable. I have only noticed it on a PRI (5ess and NI2 from what I have personally seen). It has nothing to do with the telco but I wonder what they see on their side? To me it is comforting to see, I have also disabled resetinterval on a box with four Qwest PRIs and had absolutely no problems in the last six or seven months since doing it. Bottom line, I don't really think it is needed and should possibly be defaulted to never. Thanks, Steve Totaro 888.777.1888 ___ --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] e911
*bump* - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett To: asterisk-users@lists.digium.com Sent: Tuesday, November 20, 2007 12:27 PM Subject: [asterisk-users] e911 One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.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___ --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] OT Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.
On Saturday 24 November 2007 09:53:42 Steve Totaro wrote: Tilghman Lesher wrote: On Saturday 24 November 2007 00:16:11 Steve Totaro wrote: Alex Balashov wrote: Asterisk 1.4 does have this ability natively. However, it is somewhat limited in its flexibility / in terms of what I can do with it, and I have gotten reports that HylaFAX works better. I haven't actually done a comparison between the two. Being someone who hates 1.2, I was strongly tempted to go this route, though. Why would anyone hate the most stable version of Asterisk? What is ABE using these days? If it is not 1.4, I wonder why? Maybe so all the free developers and eager and silly early adopters can iron out the bugs, submit patches and sign away their rights. I am sure if they are not using 1.4 it probably has something to do with reliability and the costs of supporting that release. Any other theories? Yeah, that version C is currently in beta and is very close to release. ABE has to be put through its paces before release and that takes time. I'm sorry if that seems like evidence that Digium isn't supporting 1.4, but it simply isn't true. I am not implying that they do not support 1.4 but you did prove my point that 1.2 is more stable and 1.4 has not been Put through its paces. I would not recommend running a high volume call center on it. Sure, if your PBX takes 50 or so calls a day, it's probably wonderful. How exactly did I prove your point? ABE C is about to be released. That says that 1.4 is indeed stable. If you have a 15,000 average call volume a day and 400+ agents, hm, I think 1.2 might be a little wiser choice. Just personal notes from the trenches, not from the media machines or talking heads. I, too, work from the trenches, and I resent your implication. -- Tilghman ___ --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] e911
Mike Hammett wrote on 11/20/07 1:27 PM: One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? I think the easiest answer is going to be to go ahead and put each in their own context. Note that you can include contexts from each other... so say they're all in [downstream-phones] right now (for example)... you can do something like this: [phones-in-account1] include = downstream-phones exten = 911,s,Goto(DialViaAccount1) [phones-in-account2] include = downstream-phones exten = 911,s,Goto(DialViaAccount2) etc. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
[asterisk-users] MSSQL ODBC Connections
Hi all, The asterisk book states the following for using ODBC to connect to an MS database. ‡ The pooling and limit options are quite useful for MS SQL Server and Sybase databases. These permit you to establish multiple connections (up to limit connections) to a database while ensuring that each connection has only one statement executing at once (this is due to a limitation in the protocol used by these database servers). Does anyone know if it is possible to use the same database and single ODBC connection to do both CDR recording with cdr_odbc and dialplan routing based on func_odbc. I have both res_odbc.conf and cdr_odbc.conf pointing to the same DSN in odbc.ini I am starting to think that this limitation in having a single connection would stop this being possible in asterisk - does anyone know otherwise? Thanks Robert McNaught ___ --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 in group
On Sat, 24 Nov 2007, Rilawich Ango wrote: I have a TDM400 with all FXO module in it. Only one channel (say channel 3) is plugged to PSTN. In my understand, a dial command Dial(zap/g1/12345677) should search an available channel, which is 3, in group 1 to make a call. However, I found that it will still use channel 1 to make call even it hasn't plugged to the PSTN. Below are the conf files. --zapata.conf-- group=1 signalling=fxs_ks context=incoming channel = 1-8 You really only want channel = 3 here if it's only channel 3 that's plugged in. Gordon ___ --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] Digium and Asterisk
I got one of this boards and I got it successfully replaced by Avanzada7 (Digium official reseller) immediately. On Nov 24, 2007 6:46 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Actually if you rule out all the clone tormenta cards (nothing wrong.. but very dated design... I wouldnt buy one today) the Digium cards aren't too expensive. Those tormenta cards are the ones you see for $300-400 typically. Some people like Digium others Sangoma. Personally I'm a Sangoma man. Some people report certain main boards and Dell servers aren't compatible with some digium cards. According to a post here on the mailing list someone from Digium implied that they will replace cards with these conflicts with newer model card that does not have these conflicts... your millage may vary I don't believe that forum posting was made in any official capacity but I also doubt that Digium would not do something to correct an issue for an item under warranty. On Nov 22, 2007 8:03 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; Is Digium the best telephony cards to be used with Asterisk? The prices are some how high, any suggestion? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --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+HylaFAX+SpanDSP+IAXmodem tutorial.
On Sat, 24 Nov 2007, Steve Totaro wrote: No functional FreePBX, I just used the ISO for a quick linux install and World Community Grid is a better benchmark than bogomips. Or it could potentially be a hidden CPU hog to leave running to increase your stats in a competitive distributed computing project anticipating that the client is not particularly UNIX savvy and won't find it. But that wouldn't be my preferred theory. Neither of which have any bearing on how I setup Hylafax and Asterisk, otherwise, great job of reverse engineering what I did and documenting it as your own development, ideas, and deployment, lol. For what it's worth, I had to redo the implementation you're describing from scratch as the client had lost some critical backups of configurations required to make it work. It was in the process of figuring out how to do that I picked up what is stated in the article. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] e911
Then I could just make downstream-phones my current outbound context and everything would do what I'm after. I got what you're saying. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Dave Miller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, November 24, 2007 2:25 PM Subject: Re: [asterisk-users] e911 Mike Hammett wrote on 11/20/07 1:27 PM: One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? I think the easiest answer is going to be to go ahead and put each in their own context. Note that you can include contexts from each other... so say they're all in [downstream-phones] right now (for example)... you can do something like this: [phones-in-account1] include = downstream-phones exten = 911,s,Goto(DialViaAccount1) [phones-in-account2] include = downstream-phones exten = 911,s,Goto(DialViaAccount2) etc. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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] MSSQL ODBC Connections
On Saturday 24 November 2007 14:48:02 Robert McNaught wrote: Does anyone know if it is possible to use the same database and single ODBC connection to do both CDR recording with cdr_odbc and dialplan routing based on func_odbc. In 1.4, no. The reason is, cdr_odbc was written prior to res_odbc and therefore does not use its connections. So cdr_odbc attempts to make its own connection to the database. Of course, if you're using MS SQL Server, that's what you want, because concurrency is what disallows the use of a single connection. I have both res_odbc.conf and cdr_odbc.conf pointing to the same DSN in odbc.ini I am starting to think that this limitation in having a single connection would stop this being possible in asterisk - does anyone know otherwise? There is no limitation in 1.4 of having a single connection, as long as you set pooling=yes. So I don't understand your question. Please understand that odbc.ini doesn't set up a connection, only the connection parameters. res_odbc can (and does) create multiple connections based upon those parameters. -- Tilghman ___ --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] Annoying PRI Channels Restarting Message
Well I am glad its normal, I am on a p2p pri so I doubt the telco even notices, but I can see on your end with a pri to the telco they would see the messages maybe. I am considering just changing them from verbose to debug in the next source code rebuild I do so they are there if I want them and hidden from normal usage. Make sense? Any issues with that? Thanks Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, November 24, 2007 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Annoying PRI Channels Restarting Message Michael Collins wrote: Is there a reason it resets? Aka does it serve any kind of purpose? Just curious: what protocol variant (i.e. 4/5ESS, DMS, NI2, etc.) are you using? Also, which carrier? Finally, have you turned on PRI debugging to see if it is the telco that is requesting the restart? In some cases the telco will send out a PRI message like 'service' (i.e. service request) to which the CPE will need to respond with a service ack message. Not all telcos behave the same with respect to so-called maintenance messages, so you might want to follow up with the carrier just to be sure nothing is wrong. Probably nothing is wrong but it can't hurt to check. -MC P.S. - the messages might be annoying, but if you've ever had PRI issues then those messages become comforting! It is Asterisk or more specifically Zaptel that causes the resets defined by the resetinterval variable. I have only noticed it on a PRI (5ess and NI2 from what I have personally seen). It has nothing to do with the telco but I wonder what they see on their side? To me it is comforting to see, I have also disabled resetinterval on a box with four Qwest PRIs and had absolutely no problems in the last six or seven months since doing it. Bottom line, I don't really think it is needed and should possibly be defaulted to never. Thanks, Steve Totaro 888.777.1888 ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Digium and Asterisk
Can you elaborate on OSLEC? I cant say I have heard of it but it sounds very interesting considering it worked for x100p for you which was the worst out of ALL the cards I have ever tried for echo. Thanks Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alan Lord Sent: Saturday, November 24, 2007 6:58 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium and Asterisk Michael J. Liberatore wrote: There are many reasons to buy digium cards, mainly digiums owner creating asterisk and all. so when i asked myself your question when starting with * i bought them. well, i myself have had bad luck with their products,2 failed out of warranty, and the others have bad echo and random weird problems. i myself switched to sangoma and have had much better success. they are even more than digium cards but work great. oh and dont even waste your time and money, get echo cancellation on any fxo cards, its the only way to make sure you get good sound quality. -mike I just want to add - for the poor amongst us, that if you use the OSLEC echo canceller with cheap x100p and (from what others have said) other analogue cards, you get excellent echo cancellation. On my cheap card, echo was terrible with the standard EC in the zaptel package. Using OSLEC instead, the echo disappeared. Completely. Al -- The way out is open! http://www.theopensourcerer.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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] Annoying PRI Channels Restarting Message
I have a p2p t1, I am using national isdn 2, b8zs/esf, one side is pri net one side is pri cpe. The telco is verizon but since it's a point to point link I doubt that matters. I posted recently before I saw your post that I am thinking of changing the code to debug instead of verbose. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Saturday, November 24, 2007 2:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Annoying PRI Channels Restarting Message Is there a reason it resets? Aka does it serve any kind of purpose? Just curious: what protocol variant (i.e. 4/5ESS, DMS, NI2, etc.) are you using? Also, which carrier? Finally, have you turned on PRI debugging to see if it is the telco that is requesting the restart? In some cases the telco will send out a PRI message like 'service' (i.e. service request) to which the CPE will need to respond with a service ack message. Not all telcos behave the same with respect to so-called maintenance messages, so you might want to follow up with the carrier just to be sure nothing is wrong. Probably nothing is wrong but it can't hurt to check. -MC P.S. - the messages might be annoying, but if you've ever had PRI issues then those messages become comforting! ___ --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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+HylaFAX+SpanDSP+IAXmodem tutorial.
LOL. Alex Balashov wrote: On Sat, 24 Nov 2007, Steve Totaro wrote: No functional FreePBX, I just used the ISO for a quick linux install and World Community Grid is a better benchmark than bogomips. Or it could potentially be a hidden CPU hog to leave running to increase your stats in a competitive distributed computing project anticipating that the client is not particularly UNIX savvy and won't find it. But that wouldn't be my preferred theory. Neither of which have any bearing on how I setup Hylafax and Asterisk, otherwise, great job of reverse engineering what I did and documenting it as your own development, ideas, and deployment, lol. For what it's worth, I had to redo the implementation you're describing from scratch as the client had lost some critical backups of configurations required to make it work. It was in the process of figuring out how to do that I picked up what is stated in the article. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] OT Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.
Tilghman Lesher wrote: On Saturday 24 November 2007 09:53:42 Steve Totaro wrote: Tilghman Lesher wrote: On Saturday 24 November 2007 00:16:11 Steve Totaro wrote: Alex Balashov wrote: Asterisk 1.4 does have this ability natively. However, it is somewhat limited in its flexibility / in terms of what I can do with it, and I have gotten reports that HylaFAX works better. I haven't actually done a comparison between the two. Being someone who hates 1.2, I was strongly tempted to go this route, though. Why would anyone hate the most stable version of Asterisk? What is ABE using these days? If it is not 1.4, I wonder why? Maybe so all the free developers and eager and silly early adopters can iron out the bugs, submit patches and sign away their rights. I am sure if they are not using 1.4 it probably has something to do with reliability and the costs of supporting that release. Any other theories? Yeah, that version C is currently in beta and is very close to release. ABE has to be put through its paces before release and that takes time. I'm sorry if that seems like evidence that Digium isn't supporting 1.4, but it simply isn't true. I am not implying that they do not support 1.4 but you did prove my point that 1.2 is more stable and 1.4 has not been Put through its paces. I would not recommend running a high volume call center on it. Sure, if your PBX takes 50 or so calls a day, it's probably wonderful. How exactly did I prove your point? ABE C is about to be released. That says that 1.4 is indeed stable. It proves my point because even 1.2 is not completely stable and that has been put through it's paces. I was present for the big 1.0 release, very stable, not really, more like a media move. We are a real PBX now we have a version 1.0!!!. Windows Millennium was released, I guess according to your logic that it was indeed stable because is was released. Take it a step further, you are saying that any beta software that is about to be released means it's stable. Please think logically about what you just said If you have a 15,000 average call volume a day and 400+ agents, hm, I think 1.2 might be a little wiser choice. Just personal notes from the trenches, not from the media machines or talking heads. I, too, work from the trenches, and I resent your implication I was not aware I was making any implications, just stating my thoughts and observations on large scale implementations (trenches) where twice the average American's annual salary is lost with one hour of downtime. If there is any resentment, it is in your mind and self reflection may be wise as I was not trying to call you a media machine or a talking head, but you may have felt the shoe fit possibly? I don't know, only you do. Thanks, Steve 888.777.1888 ___ --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] H323 registeration and routing the calls
I have not tested it but in theory you should be able to authorize it by setting host= in the peer details. - Original Message - From: bilal ghayyad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, November 09, 2007 11:14 PM Subject: [asterisk-users] H323 registeration and routing the calls Hi All; As I understood that h323 module in asterisk does not support the ability to let the h323 endpoints register at asterisk (this registeration happens at 1719 port), so how asterisk will be able to route the call for the destination IP Phone if it is not registered (so the IP is unknown)? I do not know if current h323 module supports registeration via 1719 port. Any help? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 ___ --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] OT Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.
How exactly did I prove your point? ABE C is about to be released. That says that 1.4 is indeed stable. We have been hearing that for a while. ___ --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] Problem installing Asterisk
- Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 23, 2007 6:43 PM Subject: Re: [asterisk-users] Problem installing Asterisk On Wednesday 21 November 2007 12:13:41 Matt wrote: On Nov 21, 2007 11:45 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 21 November 2007 09:09:13 Matt wrote: I have installed Asterisk with FreeTDS many times before (this same Asterisk and same TDS version)... but today when I did the make it gave me this error: ake[1]: Entering directory `/home/matth/asterisk126/asterisk-1.2.6/cdr' We don't support version 1.2.6 anymore. That is a VERY old version. Sadly, it is one of the only stable versions. Unfortunately, it's also full of identified security issues, some of which are remotely exploitable. Is 1.4 even out of beta yet? I'm not aware that it is, yet it's being forced down people's throats. 1.4 has been out of beta for close to a year. In my eyes it is still Beta till ABE is out there and running smoothly for at least a month or two. My problem is that I do not have time to be a beta tester. Another point when I did enter what I thought were bugs due to my limited knowledge of asterisk (in the beginning) I was told off for not having my facts. I did my research and I thought there was still a bug. I only try to give of my time to those that are willing to be nice about it ;) ___ --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 in group
It works if it specified the port exactly plugged to PSTN. I want to clarify the dial command here. Dial(zap/g1/1234567) It will try channel 1, if it is busy, congested then it will try channel 2 and so on, right? I wonder if I don't plug the PSTN to channel 1, there should not be a dial tone on it. Why it still try channel 1 and make call using it? On Nov 25, 2007 5:00 AM, Gordon Henderson [EMAIL PROTECTED] wrote: On Sat, 24 Nov 2007, Rilawich Ango wrote: I have a TDM400 with all FXO module in it. Only one channel (say channel 3) is plugged to PSTN. In my understand, a dial command Dial(zap/g1/12345677) should search an available channel, which is 3, in group 1 to make a call. However, I found that it will still use channel 1 to make call even it hasn't plugged to the PSTN. Below are the conf files. --zapata.conf-- group=1 signalling=fxs_ks context=incoming channel = 1-8 You really only want channel = 3 here if it's only channel 3 that's plugged in. Gordon ___ --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
