[asterisk-users] Assistance sending mass sms to cellphones
Hello. I would like to know if is possible to send mass sms with an php agi script through asterisk? For example: I have about 50 cellphone numbers I would like to text whenever theres a meeting, I should load the numbers from a database and send a message via web with php and have asterisk send it. I've been googling about it but, I get a lot of providers that already do this but, I would like to learn how to do it myself since my budget is very minimum. Thanks in advanced for your help and time. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Assistance sending mass sms to cellphones
Hello. I would like to know if is possible to send mass sms with an php agi script through asterisk? For example: I have about 50 cellphone numbers I would like to text whenever theres a meeting, I should load the numbers from a database and send a message via web with php and have asterisk send it. I've been googling about it but, I get a lot of providers that already do this but, I would like to learn how to do it myself since my budget is very minimum. Thanks in advanced for your help and time. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Assistance sending mass sms to cellphones
Robert. Thanks for replying. --- On Fri, 8/5/11, Robert Huddleston rhuddles...@gmail.com wrote: From: Robert Huddleston rhuddles...@gmail.com Subject: Re: [asterisk-users] Assistance sending mass sms to cellphones To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Friday, August 5, 2011, 11:50 AM This is off topic... Asterisk will not provide you with the ability to SMS random cell phones. We actually have a group of people belonging to a rotary club and we wanted to automate the sms process... is not random cell phones. Being able to transport the SMS yourself is a grewling process.. Look at software like Kamel... Basically you have three options: ( a ) cheat and use the email method - i.e. determine everyone's carrier and use the email address equivalent ( b ) utilize a 3rd party to transmit the sms for you (cost) and they might Looks like this is the easiest option but, very expensive for what we really want to do. end up doing ( a ) above without you knowing ( c ) spend lots of money and headaches transporting sms yourself. Either way it's off-topic and not related to Asterisk. Sorry, didn't think this wasnt an asterisk related question. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Friday, August 05, 2011 11:42 AM To: asterisk Subject: [asterisk-users] Assistance sending mass sms to cellphones Hello. I would like to know if is possible to send mass sms with an php agi script through asterisk? For example: I have about 50 cellphone numbers I would like to text whenever theres a meeting, I should load the numbers from a database and send a message via web with php and have asterisk send it. I've been googling about it but, I get a lot of providers that already do this but, I would like to learn how to do it myself since my budget is very minimum. Thanks in advanced for your help and time. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chinaroby fxo card - never heard of them
Hello. I'm looking to buy a FXO card to do some testing with two phone lines I have at home and was looking in ebay some and found some cheap ones but, the I've never heard of the brand or manufacturer: chinaroby. They run for about $99 plus shipping. Have any one used these? or please recommend one... Money IS an issue. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift.c:338 engine: Failed to set voice
Hello. I'm trying to set TTS with Cepstral and Swift but can't get it to work. I get this error when testing it: -- SIP/101- Playing 'welcome.gsm' (language 'es') -- Executing [...@local-calls:3] Swift(SIP/101-, Hello this is ceptral) in new stack [Jul 28 18:29:16] NOTICE[5191]: app_swift.c:304 engine: Text to Speak : Hello this is ceptral [Jul 28 18:29:16] ERROR[5191]: app_swift.c:338 engine: Failed to set voice. I'm using: asterisk*CLI core show version Asterisk 1.6.1.18 built by root @ optimum-asterisk on a i686 running Linux on 2010-04-10 01:42:25 UTC I googled around but, there isnt a real solution I could find. Any suggestions? Thanks in advanced for your help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_swift.c:338 engine: Failed to set voice
Do you have cepstral installed and have the voice(s) registered ? try: swift --voices asterisk:~# swift --voices Swift command-line synthesis program Version 5.1.0 of July 2008 Copyright (c) 2000-2006, Cepstral LLC. Voice | Version | Lic? | Gender | Age | Language | Sample Rate ---|-|--||-|--| Marta | 5.1.0 | No | female | 30 | Americas Spanish | 16000 Hz assuming swift is installed an a valid voice is registered, what happens when you type: swift Test Message -o /tmp/file.wav is /tmp/file.wav created ? does it play ? This creates the file and if I download it to my machine I can listen to it. what is the output of: grep ^[a-z] /etc/asterisk/swift.conf asterisk:/home/landysaccount# grep ^[a-z] /etc/asterisk/swift.conf buffer_size=65535 goto_exten=no voice=Marta-8kHz|David-8kHz somewhere should say voice=X. Is that voice installed as per the above swift --voices command ? also, if you're going to be dialing digits with swift, you'll probably run into detection issues unless you use my patch at http://jeremy.kister.net/code/app_swift-1.6.2.patch I had to patch that file in order for me to be able to install swift. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_swift.c:338 engine: Failed to set voice
Jeremy, Thanks a lot that helped and solved the problem. I had it as: voice=Marta-8kHz before and that didn't work and now changed it to voice=Marta. Thanks. I apreciate it. --- On Wed, 7/28/10, Jeremy Kister asterisk...@jeremykister.com wrote: From: Jeremy Kister asterisk...@jeremykister.com Subject: Re: [asterisk-users] app_swift.c:338 engine: Failed to set voice To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, July 28, 2010, 9:08 PM On 7/28/2010 8:33 PM, Landy Landy wrote: asterisk:/home/landysaccount# grep ^[a-z] /etc/asterisk/swift.conf buffer_size=65535 goto_exten=no voice=Marta-8kHz|David-8kHz afaik, the voice parameter is simply the default voice when not specified via the swift binary or the Swift asterisk command. even if it's not, you don't have David registered. try making that: voice=Marta (or possibly: voice=Marta-8kHz) then restart asterisk and give it another shot. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing for residential voip usage
I reinstalled a2billing, now 1.7. Created a trunk, call plan, rate card, added rate, and added rate to call plan. After creating a new customer (CC) now I was able to place a call through a2billing only for the new customers. In voip settings I added a SIP Config with the same information as in my current extensions since I would like to re-use these extension numbers to monitor them. Also changed the context for these to a2billing. When I try to call from these extension I get Enter your pin prompt. Now I'm stuck here. Other than inserting the record into the mysql table how can I espcify the account number and/or cc number and password for a new customer? Thanks. --- On Thu, 6/17/10, Vahan Yerkanian va...@arminco.com wrote: From: Vahan Yerkanian va...@arminco.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: asterisk-users@lists.digium.com Date: Thursday, June 17, 2010, 1:47 AM On 6/17/10 12:49 AM, Steve Edwards wrote: On Wed, 16 Jun 2010, Landy Landy wrote: I'm unable to place any calls through a2billing. I followed instructions here: http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/F.A.Q to DISABLE PIN number request Prompt for some users but, I'm not able to place any calls. I created a trunk with the same name as in my sip.conf and I'm not able to make any calls. I don't know what I'm missing. This is the output when trying to call: == Using SIP RTP CoS mark 5 -- Executing [812022418...@a2billing:1] Answer(SIP/1433631307-0015, ) in new stack -- Executing [812022418...@a2billing:2] Wait(SIP/1433631307-0015, 2) in new stack -- Executing [812022418...@a2billing:3] AGI(SIP/1433631307-0015, a2billing.php,3) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php --SIP/1433631307-0015AGI Script a2billing.php completed, returning -1 I can't debug it or anything I'm stuck please help. If you have CLI version of PHP installed, you can also try running /var/lib/asterisk/agi-bin/a2billing.php directly from the shell, and keep feeding it CR/LF, you'll see step-by-step variable assignment and hopefully the error message that stops it from working. You'll need display_errors on in php.ini for this as well. Most probably you're missing a PHP module or your SQL connection is failing. HTH, Vahan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing for residential voip usage
I'm unable to place any calls through a2billing. I followed instructions here: http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/F.A.Q to DISABLE PIN number request Prompt for some users but, I'm not able to place any calls. I created a trunk with the same name as in my sip.conf and I'm not able to make any calls. I don't know what I'm missing. This is the output when trying to call: == Using SIP RTP CoS mark 5 -- Executing [812022418...@a2billing:1] Answer(SIP/1433631307-0015, ) in new stack -- Executing [812022418...@a2billing:2] Wait(SIP/1433631307-0015, 2) in new stack -- Executing [812022418...@a2billing:3] AGI(SIP/1433631307-0015, a2billing.php,3) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- SIP/1433631307-0015AGI Script a2billing.php completed, returning -1 I can't debug it or anything I'm stuck please help. --- On Tue, 6/15/10, Faisal Hanif fai...@vopium.com wrote: From: Faisal Hanif fai...@vopium.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:26 PM You need to copy or soft link a2billing.conf to /etc/ folder as by default latest version search for it in /etc/ Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Tuesday, June 15, 2010 9:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] a2billing for residential voip usage I copied the config to the a2billing.conf in /etc/asterisk folder. I'm still not able to place any calls yet. Looks like I have to read more on how to configure trunks and providers whick got me confused. I'll learn though. --- On Tue, 6/15/10, Vardan Harutyunyan hvarda...@gmail.com wrote: From: Vardan Harutyunyan hvarda...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 8:03 AM look manual, but in any case the a2billing.conf is in /etc/asterisk/ on can say, where you have place your asterisk configuration files -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com Jimmy Godbout wrote: Hi, Maybe you can just use a reporting tool that will look at the CDR and tell you who's using the phone the most. Some of them will use a DB to store the CDR. If you want, you can even use Excel to look at the csv file created by default and make your own report. http://www.voip-info.org/wiki/view/Asterisk+billing http://www.voip-info.org/wiki/view/Asterisk+GUI (in Billing Call Detail Reporting) http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI Jimmy -Original Message- From: landysacco...@yahoo.com Sent: Tue, 15 Jun 2010 00:11:51 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] a2billing for residential voip usage Ram. Thanks for replying. I have searched / googled about it but can't find a solution to monitor the 4 extensions I have at home. A2billing asks for the number I want to dial but, I don't need that. I would like the extensions to dial out normally and a2billing just record the time and talked time for later review. Thanks. --- On Tue, 6/15/10, ramtalk2...@gmail.com wrote: From: ramtalk2...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:05 AM you see lot of documentation on wiki Google them many success case you see Ram On Tue, Jun 15, 2010 at 7:01 AM, Landy Landylandysacco...@yahoo.com wrote: Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for •VoIP residential services? if yes, how? if no, please guide me to another application I can use along side asterisk. Thanks in advanced for your time. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http
Re: [asterisk-users] a2billing for residential voip usage
Ram. Thanks for replying. I have searched / googled about it but can't find a solution to monitor the 4 extensions I have at home. A2billing asks for the number I want to dial but, I don't need that. I would like the extensions to dial out normally and a2billing just record the time and talked time for later review. Thanks. --- On Tue, 6/15/10, ram talk2...@gmail.com wrote: From: ram talk2...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:05 AM you see lot of documentation on wiki Google them many success case you see Ram On Tue, Jun 15, 2010 at 7:01 AM, Landy Landy landysacco...@yahoo.com wrote: Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for •VoIP residential services? if yes, how? if no, please guide me to another application I can use along side asterisk. Thanks in advanced for your time. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing for residential voip usage
I copied the config to the a2billing.conf in /etc/asterisk folder. I'm still not able to place any calls yet. Looks like I have to read more on how to configure trunks and providers whick got me confused. I'll learn though. --- On Tue, 6/15/10, Vardan Harutyunyan hvarda...@gmail.com wrote: From: Vardan Harutyunyan hvarda...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 8:03 AM look manual, but in any case the a2billing.conf is in /etc/asterisk/ on can say, where you have place your asterisk configuration files -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com Jimmy Godbout wrote: Hi, Maybe you can just use a reporting tool that will look at the CDR and tell you who's using the phone the most. Some of them will use a DB to store the CDR. If you want, you can even use Excel to look at the csv file created by default and make your own report. http://www.voip-info.org/wiki/view/Asterisk+billing http://www.voip-info.org/wiki/view/Asterisk+GUI (in Billing Call Detail Reporting) http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI Jimmy -Original Message- From: landysacco...@yahoo.com Sent: Tue, 15 Jun 2010 00:11:51 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] a2billing for residential voip usage Ram. Thanks for replying. I have searched / googled about it but can't find a solution to monitor the 4 extensions I have at home. A2billing asks for the number I want to dial but, I don't need that. I would like the extensions to dial out normally and a2billing just record the time and talked time for later review. Thanks. --- On Tue, 6/15/10, ramtalk2...@gmail.com wrote: From: ramtalk2...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:05 AM you see lot of documentation on wiki Google them many success case you see Ram On Tue, Jun 15, 2010 at 7:01 AM, Landy Landylandysacco...@yahoo.com wrote: Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for •VoIP residential services? if yes, how? if no, please guide me to another application I can use along side asterisk. Thanks in advanced for your time. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Share photos screenshots in seconds... TRY FREE IM TOOLPACK at http://www.imtoolpack.com/default.aspx?rc=if1 Works in all emails, instant messengers, blogs, forums and social networks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing for residential voip usage
It was already done. My problem now is that I cant' place any calls through a2billing. --- On Tue, 6/15/10, Faisal Hanif fai...@vopium.com wrote: From: Faisal Hanif fai...@vopium.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:26 PM You need to copy or soft link a2billing.conf to /etc/ folder as by default latest version search for it in /etc/ Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Tuesday, June 15, 2010 9:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] a2billing for residential voip usage I copied the config to the a2billing.conf in /etc/asterisk folder. I'm still not able to place any calls yet. Looks like I have to read more on how to configure trunks and providers whick got me confused. I'll learn though. --- On Tue, 6/15/10, Vardan Harutyunyan hvarda...@gmail.com wrote: From: Vardan Harutyunyan hvarda...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 8:03 AM look manual, but in any case the a2billing.conf is in /etc/asterisk/ on can say, where you have place your asterisk configuration files -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com Jimmy Godbout wrote: Hi, Maybe you can just use a reporting tool that will look at the CDR and tell you who's using the phone the most. Some of them will use a DB to store the CDR. If you want, you can even use Excel to look at the csv file created by default and make your own report. http://www.voip-info.org/wiki/view/Asterisk+billing http://www.voip-info.org/wiki/view/Asterisk+GUI (in Billing Call Detail Reporting) http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI Jimmy -Original Message- From: landysacco...@yahoo.com Sent: Tue, 15 Jun 2010 00:11:51 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] a2billing for residential voip usage Ram. Thanks for replying. I have searched / googled about it but can't find a solution to monitor the 4 extensions I have at home. A2billing asks for the number I want to dial but, I don't need that. I would like the extensions to dial out normally and a2billing just record the time and talked time for later review. Thanks. --- On Tue, 6/15/10, ramtalk2...@gmail.com wrote: From: ramtalk2...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:05 AM you see lot of documentation on wiki Google them many success case you see Ram On Tue, Jun 15, 2010 at 7:01 AM, Landy Landylandysacco...@yahoo.com wrote: Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for •VoIP residential services? if yes, how? if no, please guide me to another application I can use along side asterisk. Thanks in advanced for your time. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Share photos screenshots in seconds... TRY FREE IM TOOLPACK at http://www.imtoolpack.com/default.aspx?rc=if1 Works
[asterisk-users] a2billing for residential voip usage
Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for •VoIP residential services? if yes, how? if no, please guide me to another application I can use along side asterisk. Thanks in advanced for your time. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no voicemail on pstn line
Hello List. I am having problems retreiving voicemails on my system. I noticed when someone leaves a message through the pstn line I can't hear anything. I tested leaving a message from one of the extensions and that can be heard. I don't know if is the type of card I'm using for analog ( cheap X100p modem ) calls but, can't hear any message coming in from that line. Any suggestions? Thanks in advanced. Here's voicmail.conf: [general] ; Choose a format to save voicemails as format=gsm volgain=1.1 skipms=3000 maxsilence=10 sayduration=no saycid=no sendvoicemail=no review=yes nextaftercmd=yes listen-control-forward-key=# [default] 100 = 1234,testing 101 = 1234,testing2 102 = 1234,testing3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Important security alert: update your dialplans now!
I have this:
[menu]
exten = _X.,1,answer()
exten = _X.,2,wait(1)
exten = _X.,n,GoTo(ivr,s,1)
[default]
include = record
include = incoming
include = menu
[local-dial]
exten = _1XX,1,Verbose(. In local-dial context, dialing exten: ${EXTEN}
.
exten = _1XX,2,Dial(SIP/${EXTEN},20,tTmkKhHWw)
exten = _1XX,n,voicemail(${EXTEN},u)
exten = _1XX,n,Hangup()
include = agents
include = queue
include = local-iax
include = voicemail
include = timeofday
include = parkedcalls
include = pickup
include = to_client
include = test-agi
include = menu
that goes to an ivr. Can this be a security bridge?
--- On Mon, 2/15/10, Tony Mountifield t...@softins.clara.co.uk wrote:
From: Tony Mountifield t...@softins.clara.co.uk
Subject: Re: [asterisk-users] Important security alert: update your dialplans
now!
To: asterisk-users@lists.digium.com
Date: Monday, February 15, 2010, 11:58 AM
In article 699ee941002150033t7c6e1be5xdba76cb0f68d5...@mail.gmail.com,
Lenz Emilitri lenz.lo...@gmail.com
wrote:
-=-=-=-=-=-
-=-=-=-=-=-
Or one could simply rewrite to:
[incoming-from-voip]
exten =
XXX,1,Dial(${ext...@incoming-from-voip-old)
exten =
,1,Dial(${ext...@incoming-from-voip-old)
exten =
X,1,Dial(${ext...@incoming-from-voip-old)
exten =
XX,1,Dial(${ext...@incoming-from-voip-old)
[incoming-from-voip-old]
exten = _X., 1, dial(SIP/${EXTEN})
To avoid extensive rewriting and fix the current
issue.
l.
Don't forget you still need the underscore to make X
magic:
exten =
_XXX,1,Dial(${ext...@incoming-from-voip-old)
etc.
Tony
--
Tony Mountifield
Work: t...@softins.co.uk
- http://www.softins.co.uk
Play: t...@mountifield.org
- http://tony.mountifield.org
--
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[asterisk-users] Asterisk listens on all NICs
Hello List. I am puzzled and how asterisk listens to calls or connections from clients. When I do a netstat -nat I don't see asterisk listening on port 5060. Now, I'm testing a server with three network interfaces: two to the internet doing load balancing and the other to our LAN. I would like asterisk to only accept connections coming from our LAN but, can't find where to configure this. I know I can do it with iptables and block incoming connections to ports 5060-5070 from the internet but, wondering if it can be confiruged in asterisk. Thanks. -- _ -- 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 listens on all NICs
See http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf Search for bindaddr. Or udpbindaddr for 1.6.2+...also, tcpbindaddr, tlsbindaddr if you plan on adding TCP/TLS SIP support to asterisk. Thanks to everyone who replied for clarifying. -- _ -- 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] How can I get codec info on active calls
Hello All. I would like to know what codec is being used during a call. For example if I have 3 channels on 3 active calls how can I find what codec is beeing used by each client? Thanks in advanced. -- _ -- 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] Help getting info from caller
Hello. Happy New Year to everyone.
I have a small WISP and would like to have customers to call our number to
check their balance. I am planning on writing an AGI with php so it can get the
customer info from the customer database. I don't know how to interact with the
caller while in the agi script so this is what I have in mind:
[test-agi]
exten = 33,1,Answer()
exten = 33,n,Wait(0.5)
exten = 33,n,BackGround(please-enter)
exten = 33,n,BackGround(customer-account)
exten = 33,n, I would like to set a variable here but don't know how -
exten = 33,n,BackGround(enter-password)
exten = 33,n, I would like to set a variable here but don't know how -
exten = 33,n,AGI(testphp.agi,${ACCOUNT},${PASSWORD})
receive the balance here from agi
exten = 33,n,Verbose( This is agi status ...${AGISTATUS}...)
exten = 33,n,hangup()
I've never worked with agi but, I'm reading some documents I found online but,
need more help trying to get this working.
Thanks in advanced for your help.
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Re: [asterisk-users] Help getting info from caller
--- On Sat, 1/2/10, Landy Landy landysacco...@yahoo.com wrote:
From: Landy Landy landysacco...@yahoo.com
Subject: [asterisk-users] Help getting info from caller
To: asterisk-users@lists.digium.com
Date: Saturday, January 2, 2010, 9:01 AM
Hello. Happy New Year to everyone.
I have a small WISP and would like to have customers to
call our number to check their balance. I am planning on
writing an AGI with php so it can get the customer info from
the customer database. I don't know how to interact with the
caller while in the agi script so this is what I have in
mind:
[test-agi]
exten = 33,1,Answer()
exten = 33,n,Wait(0.5)
exten = 33,n,BackGround(please-enter)
exten = 33,n,BackGround(customer-account)
exten = 33,n, I would like to set a variable here
but don't know how -
exten = 33,n,BackGround(enter-password)
exten = 33,n, I would like to set a variable here
but don't know how -
exten = 33,n,AGI(testphp.agi,${ACCOUNT},${PASSWORD})
receive the balance here from agi
exten = 33,n,Verbose( This is agi status
...${AGISTATUS}...)
exten = 33,n,hangup()
I've never worked with agi but, I'm reading some documents
I found online but, need more help trying to get this
working.
Thanks in advanced for your help.
Can I use:
exten = 33,n,Set(ACCOUNT=waitexten()) ???
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Re: [asterisk-users] Help getting info from caller
I was able to test the script, here is what I have:
[CODE]
#!/usr/bin/php -q
?php
//ini_set(include_path,
.:../:./includes:../include:/var/lib/asterisk/agi-bin/includes);
//include( ./includes/optimum_config.php );
$CONF['host'] = 'server';
$CONF['user'] = '';
$CONF['password'] = '';
$CONF['database'] = 'testasterisk';
//include( ./includes/constants.php );
// Defining constants variables for mysql connect database:
define( HOST, $CONF['host'] );
define( USER, $CONF['user'] );
define( PASSWORD, $CONF['password'] );
define( DATABASE, $CONF['database'] );
//include( ./includes/functions.php );
function dba_connect( $query, $connect = 1 ){
if($connect)
$link = mysql_connect( HOST, USER, PASSWORD );
//echo link: . $link . query: . $query;
mysql_select_db( DATABASE ) or
die(Cannot select a Database from the server.);
if( $result = mysql_query( $query ))
return $result;
else
return 0;
}
// don't let this script run for more than 60 seconds
set_time_limit(60);
// turn off output buffering
ob_implicit_flush(false);
/* turn off error reporting, as it will most likely interfere with
the AGI interface
*/
error_reporting(0);
// create file handles if needed
if (!defined('STDIN'))
{
define('STDIN', fopen('php://stdin', 'r'));
}
if (!defined('STDOUT'))
{
define('STDOUT', fopen('php://stdout', 'w'));
}
if (!defined('STDERR'))
{
define('STDERR', fopen('php://stderr', 'w'));
}
$query = select * from balance where bal_cust_id = . $argv[1];
$result = dba_connect( $query, 1 ) or
die( Query: '$query', failed with error message: -- .
mysql_error() . -- );
$record = mysql_fetch_array( $result );
$bal = $record['bal_amount'];
echo SET VARIABLE BALANCE $bal #\n;
echo exec BackGround 'tt-monkeys' #\n;
fclose( STDIN );
fclose( STDOUT );
fclose( STDERR );
exit(0);
?
[/CODE]
extensions.conf:
[test-agi]
exten = 33,1,Answer()
exten = 33,n,Wait(0.5)
exten = 33,n,BackGround(please-enter)
exten = 33,n,BackGround(customer-accounts)
exten = 33,n,Read(ACCOUNT,,4)
;exten = 33,n,BackGround(enter-password)
;exten = 33,n,Read(PASSWORD,,4)
exten = 33,n,AGI(testphp.agi,${ACCOUNT},${PASSWORD})
;exten = 33,n,BackGround(your)
exten = 33,n,BackGround(account-balance-is)
exten = 33,n,SayNumber(${BALANCE})
exten = 33,n,BackGround(dollars)
exten = 33,n,Verbose( This is agi status ...${AGISTATUS}...)
exten = 33,n,hangup()
I was able to get the balance from the db table and have asterisk tell the
caller.
I tried to include some files in agi but kept getting an error that the file
didn't exist.
I would like to thank you for helping me out with this. Is a good starting
point.
Now, I have another thing in mind:
Is asterisk or any other program that can work along side * able to say a name
or any word? For example:
Lets say I have a table with name and last name I would like asterisk to say
balance for john doe is 100 dollars... Is this possible?
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Re: [asterisk-users] Best way ro run 2 or more asterisk servers?
--- On Wed, 12/16/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] Best way ro run 2 or more asterisk servers? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, December 16, 2009, 7:28 AM peering useful: http://astrecipes.net/index.php?n=204Thanksl. I followed exactly what' on that tutorial and can't get it to work. Now, I tried: example Server1 [server2] type=peer context=from_client host=server2-ip Server2 [server1] type=peer context=from_client host=server1-ip without the username and secret and now works but, why isn't working with the usernames? and is this way as secured as using username and secret? Can someone please clarify this. I'm confused, I thought a server needed to be secured with it's username and password. ___ -- 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] Best way ro run 2 or more asterisk servers?
peering useful: http://astrecipes.net/index.php?n=204Thanksl. I followed exactly what' on that tutorial and can't get it to work. Now, I tried: example Server1 [server2] type=peer context=from_client host=server2-ip Server2 [server1] type=peer context=from_client host=server1-ip without the username and secret and now works but, why isn't working with the usernames? and is this way as secured as using username and secret? ___ -- 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] Best way ro run 2 or more asterisk servers?
I'm trying to get two server communicate with each other and call from one to
the other but, I'm having a lot of problems.
I tried to create a iax trunk between the two:
At the server:
[client]
type=friend
username=asterisk2
authuser=asterisk2
fromuser=asterisk2
secret=sss
auth=md5
context=from_client
;peercontext=from_asterisk
host=172.16.0.11
trunk=yes
qualify=yes
iax2 show peers
Name/UsernameHost Mask Port Status
client/asterisk 172.16.0.11 (S) 255.255.255.255 4569 (T) (E) OK (3 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
extensions.conf
[to_client]
exten = _3XX,1,Verbose(. To Asterisk2 Server .)
exten = _3XX,n,Dial(IAX2/${ext...@client)
exten = _3XX,n,Hangup()
[from_client]
include = local-dial
At the client:
[server]
type=friend
host=172.16.0.3
username=asterisk
authuser=asterisk
fromuser=asterisk
secret=xxx
context=from_server
trunk=yes
auth=md5
qualify=yes
iax2 show peers
Name/UsernameHost Mask Port Status
server/asterisk 172.16.0.3 (S) 255.255.255.255 4569 (T) (E) OK (3 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
extensions.conf
[from_server]
include = local-dial
[to_server]
exten = _5XXX,1,Verbose(. Trying to contact ${EXTEN:1} @ asterisk .)
exten = _5XXX,n,Dial(IAX2/${ext...@server)
exten = _5XXX,n,Hangup
According to some reading, I do NOT need to register neither one.
When I try to call from one end to the other I get:
[Dec 15 03:06:04] NOTICE[4265]: chan_iax2.c:10338 socket_process: Host
172.16.0.3 failed to authenticate as 300
Please help.
Thanks.
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Re: [asterisk-users] Best way ro run 2 or more asterisk servers?
Date: Wednesday, December 16, 2009, 1:26 AM trust both the side giving IP address in the sip.conf I did this in the iax.conf file [client] type=friend username=asterisk2 authuser=asterisk2 fromuser=asterisk2 secret=sss auth=md5 context=from_client host=172.16.0.11 trunk=yes qualify=yes for both the client and server do I need it also in the sip.conf? ___ -- 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] Best way ro run 2 or more asterisk servers?
Hello List. I have a question regarding connecting two asterisk servers. I'm trying to learn how asterisk comunicates from server to server. I already have a server running smoothly now, I'm installing another one to test it along side the actual one. I would like to run different scenarios: 1. Have one of the boxes at a different location outside the LAN and have them communicate. 2. Have both boxes on the same physical location with different extensions, for ei. have box 1 serve exts 100 - 200 and box 2 serve exts 300 - 600 and iax2. Box 1 would be connected to a pstn line and box 2 connected to a voip provider. Now, do I need to configure dundi or just have the register option on both boxes? Thanks in advanced 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
Re: [asterisk-users] Unable to open file...
Removing the spaces did it. I works now. I used the space for clarity but, Asterisk didn't like it. Thanks for your time. --- On Sat, 12/12/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] Unable to open file... To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Saturday, December 12, 2009, 10:38 PM Take the whitespace out of your ()'s. It's: exten = 80,n,BackGround(es/good) not exten = 80,n,BackGround( es/good ) Thanks, --Warren Selby On Dec 12, 2009, at 9:16 PM, Landy Landy landysacco...@yahoo.com wrote: Same thing: == Using SIP RTP CoS mark 5 -- Executing [...@outbound:1] Answer(SIP/102-096a48c8, ) in new stack -- Executing [...@outbound:2] Verbose(SIP/102-096a48c8, In timeofday ) in new stack In timeofday -- Executing [...@outbound:3] GotoIfTime(SIP/102-096a48c8, 00:00-12:00,*,*,*?day) in new stack -- Executing [...@outbound:4] GotoIfTime(SIP/102-096a48c8, 12:01-17:59,*,*,*?afternoon) in new stack -- Executing [...@outbound:5] GotoIfTime(SIP/102-096a48c8, 18:00-11:59,*,*,*?night) in new stack -- Goto (outbound,80,16) -- Executing [...@outbound:16] Verbose(SIP/102-096a48c8, Night..) in new stack Night.. -- Executing [...@outbound:17] BackGround(SIP/102-096a48c8, es/ good ) in new stack [Dec 12 23:24:07] WARNING[6343]: file.c:650 ast_openstream_full: File es/good does not exist in any format [Dec 12 23:24:07] WARNING[6343]: file.c:933 ast_streamfile: Unable to open es/good (format 0x8 (alaw)): No such f ile or directory [Dec 12 23:24:07] WARNING[6343]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-096a48c8 for es/good -- Executing [...@outbound:18] BackGround(SIP/102-096a48c8, es/ evening ) in new stack [Dec 12 23:24:07] WARNING[6343]: file.c:650 ast_openstream_full: File es/evening does not exist in any format [Dec 12 23:24:07] WARNING[6343]: file.c:933 ast_streamfile: Unable to open es/evening (format 0x8 (alaw)): No suc h file or directory [Dec 12 23:24:07] WARNING[6343]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-096a48c8 for es/evening -- Executing [...@outbound:19] Hangup(SIP/102-096a48c8, ) in new stack == Spawn extension (outbound, 80, 19) exited non-zero on 'SIP/ 102-096a48c8' This is what the context looks like: [timeofday] exten = 80,1,Answer() exten = 80,n,Verbose( In timeofday ) exten = 80,n,GotoIfTime( 00:00-12:00,*,*,*?day) exten = 80,n,GotoIfTime( 12:01-17:59,*,*,*?afternoon) exten = 80,n,GotoIfTime( 18:00-11:59,*,*,*?night) exten = 80,n(day),Verbose(It's Day..) exten = 80,n,BackGround( es/good ) exten = 80,n,Verbose(Day..) exten = 80,n,BackGround( es/morning ) exten = 80,n,hangup() exten = 80,n(afternoon),Verbose(It's Afternoon..) exten = 80,n,BackGround( es/good ) exten = 80,n,Verbose(afternoon..) exten = 80,n,BackGround( es/afternoon ) exten = 80,n,hangup() exten = 80,n(night),Verbose(Night..) exten = 80,n,BackGround( es/good ) exten = 80,n,BackGround( es/evening ) exten = 80,n,hangup() ___ -- 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] how to randomly use provider?
Hello List. I would like to know how I can use two or more service providers with asterisk to be used randomly for ei, if an user tries to make a call I would like to randomly use a provider. It doesn't matter where the call is destined to. Thanks. ___ -- 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] Unable to open file...
Hi List. Don't know if I already posted about this problem but, if I have I apologize for the double post. I am trying to test a time of day extension dialing 80, all I'm trying to test is if is morning I would like asterisk to say Good Morning but, when I run the test I get the following error message saying that the file doesn't exist and it does: Night.. -- Executing [...@outbound:17] BackGround(SIP/100-096ce078, good ) in new stack [Dec 12 22:53:31] WARNING[6300]: file.c:650 ast_openstream_full: File good does not exist in any format [Dec 12 22:53:31] WARNING[6300]: file.c:933 ast_streamfile: Unable to open good (format 0x4 (ulaw)): No such file or directory [Dec 12 22:53:31] WARNING[6300]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/100-096ce078 for good -- Executing [...@outbound:18] BackGround(SIP/100-096ce078, evening ) in new stack [Dec 12 22:53:31] WARNING[6300]: file.c:650 ast_openstream_full: File evening does not exist in any format [Dec 12 22:53:31] WARNING[6300]: file.c:933 ast_streamfile: Unable to open evening (format 0x4 (ulaw)): No such file or directory [Dec 12 22:53:31] WARNING[6300]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/100-096ce078 for evening asterisk-server:/var/lib/asterisk/sounds/es# ls /var/lib/asterisk/sounds/es/evening.ulaw /var/lib/asterisk/sounds/es/evening.ulaw Is this a bug or am I missing something? Thanks in advanced for your time. ___ -- 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] Unable to open file...
Same thing: == Using SIP RTP CoS mark 5 -- Executing [...@outbound:1] Answer(SIP/102-096a48c8, ) in new stack -- Executing [...@outbound:2] Verbose(SIP/102-096a48c8, In timeofday ) in new stack In timeofday -- Executing [...@outbound:3] GotoIfTime(SIP/102-096a48c8, 00:00-12:00,*,*,*?day) in new stack -- Executing [...@outbound:4] GotoIfTime(SIP/102-096a48c8, 12:01-17:59,*,*,*?afternoon) in new stack -- Executing [...@outbound:5] GotoIfTime(SIP/102-096a48c8, 18:00-11:59,*,*,*?night) in new stack -- Goto (outbound,80,16) -- Executing [...@outbound:16] Verbose(SIP/102-096a48c8, Night..) in new stack Night.. -- Executing [...@outbound:17] BackGround(SIP/102-096a48c8, es/good ) in new stack [Dec 12 23:24:07] WARNING[6343]: file.c:650 ast_openstream_full: File es/good does not exist in any format [Dec 12 23:24:07] WARNING[6343]: file.c:933 ast_streamfile: Unable to open es/good (format 0x8 (alaw)): No such f ile or directory [Dec 12 23:24:07] WARNING[6343]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-096a48c8 for es/good -- Executing [...@outbound:18] BackGround(SIP/102-096a48c8, es/evening ) in new stack [Dec 12 23:24:07] WARNING[6343]: file.c:650 ast_openstream_full: File es/evening does not exist in any format [Dec 12 23:24:07] WARNING[6343]: file.c:933 ast_streamfile: Unable to open es/evening (format 0x8 (alaw)): No suc h file or directory [Dec 12 23:24:07] WARNING[6343]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-096a48c8 for es/evening -- Executing [...@outbound:19] Hangup(SIP/102-096a48c8, ) in new stack == Spawn extension (outbound, 80, 19) exited non-zero on 'SIP/102-096a48c8' This is what the context looks like: [timeofday] exten = 80,1,Answer() exten = 80,n,Verbose( In timeofday ) exten = 80,n,GotoIfTime( 00:00-12:00,*,*,*?day) exten = 80,n,GotoIfTime( 12:01-17:59,*,*,*?afternoon) exten = 80,n,GotoIfTime( 18:00-11:59,*,*,*?night) exten = 80,n(day),Verbose(It's Day..) exten = 80,n,BackGround( es/good ) exten = 80,n,Verbose(Day..) exten = 80,n,BackGround( es/morning ) exten = 80,n,hangup() exten = 80,n(afternoon),Verbose(It's Afternoon..) exten = 80,n,BackGround( es/good ) exten = 80,n,Verbose(afternoon..) exten = 80,n,BackGround( es/afternoon ) exten = 80,n,hangup() exten = 80,n(night),Verbose(Night..) exten = 80,n,BackGround( es/good ) exten = 80,n,BackGround( es/evening ) exten = 80,n,hangup() ___ -- 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] Question about g729
Hello. I am currently testing an asterisk server using the default codecs, I have allow=all, and noticed everytime I test it in a wireless lan the latency rockets off the roof to over 1000ms. I would like to test g729 since it uses less bandwidth but, read somewhere I have to buy a license per every channel I have. Does this means if I have my server connected with 10 sip clients I need to buy a license for 10 or more? ___ -- 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] Question about g729
You only need to purchase 10 licenses, if all 10 clients will be making calls at the same time. Ok. Does this apply only for outbound calls using a voip provider and/or applies to calls within the lan? ___ -- 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] Unable to open sound file error
List. How can I resolve this problem? I've searched on the web but, can't really find a solution. Please help. --- On Wed, 11/25/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: [asterisk-users] Unable to open sound file error To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 25, 2009, 7:45 PM Hello. I have a question regarind sound files in asterisk 1.6. I have a sound package in ulaw format and I would like to know if I have a sip extension with allow=alaw would asterisk convert that file to the codec the user is allowed to? I am having a problem playing a file that exist in /var/lib/asterisk/sounds/es/good.ulaw but asterisk is telling me it doesn't. Here's what I get when I try to dial the extension for test: [Nov 25 20:44:41] WARNING[4334]: file.c:650 ast_openstream_full: File good does not exist in any format [Nov 25 20:44:41] WARNING[4334]: file.c:933 ast_streamfile: Unable to open good (format 0x8 (alaw)): No such file or directory [Nov 25 20:44:41] WARNING[4334]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-09b52260 for good -- Executing [...@default:12] BackGround(SIP/102-09b52260, evening ) in new stack [Nov 25 20:44:41] WARNING[4334]: file.c:650 ast_openstream_full: File evening does not exist in any format [Nov 25 20:44:41] WARNING[4334]: file.c:933 ast_streamfile: Unable to open evening (format 0x8 (alaw)): No such file or directory [Nov 25 20:44:41] WARNING[4334]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-09b52260 for evening -- Executing [...@default:13] Hangup(SIP/102-09b52260, ) in new stack Any suggestions? Thanks in advanced 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] can't call through voip provider
Erik. I already solved this problem and posted it. I was reloading all the setting but, it wasn't changing the provider's ip info. After doing a restart now everything worked. Thanks any ways for your help. --- On Fri, 11/27/09, meetmecall i...@meetmecall.nl wrote: From: meetmecall i...@meetmecall.nl Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, November 27, 2009, 9:51 AM It is not that easy to give the answer. There are lots of itsp typical ways of registration and you haven't provide the info needed to help you out. You need a register line in the general part of sip.conf. It should look something like (mine looks like this register = DID:SECRET:username@ipness.net:6060 And you need a sip entry in sip.conf. For me it looks something like [DID] type=friend host=ipness.net fromuser=DID fromdomain=ipness.net username=username secret=secret insecure=very context=inbound port=6060 qualify=2000 canreinvite=no disallow=all ;allow=ulaw allow=alaw But your provider might need other settings. So ask your provider. If you are on public IP and not behind NAT you should use nat=no From the sip message I make up that the You didn't provide debug info but copied and paste a sip message. If you would like people to help you, you have to provide proper info. CLI output, sip.conf (without passwords and IP adress info) and the sip messages will be helpful. Are you aware of the fact that you need to open UDP ports and not TCP. Your provider should be able to tell you how to configure such an account on an asterisk box, or at least help you to figure it out. A serious ITSP must have customers using Asterisk. If you have no idea what you are doing my advice is to start reading Asterisk: The future of telephony, freely available on http://www.asteriskdocs.org/ . VERY SERIOUS WARNING: Don't put the credentials of a sip account in a mail to a mailing list. People might use your account to call satelite lines for EUR 7,50 per minute. This kind of mistakes might bankcrupt you :-( I hope this helps. Erik On 19 nov 2009, at 22:36, Landy Landy wrote: Can someone please share with me a sip configuration to connect an asterisk server to a voip provider since my configuration isn't working for me. thanks. --- On Thu, 11/19/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 7:51 AM Ok. I do NOT have ports 1-2 opened in. I guess I I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I will keep you posted. I ran this test and there was no difference. I still can't get through. --- Retransmitting #5 (NAT) to 190.80.153.193:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.153.193:5060;branch=z9hG4bK727987ef Max-Forwards: 70 From: 102 sip:77...@190.80.153.193;tag=as23e02274 To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.153.193 Call-ID: 034bf0572cffb96f621211a8439aa...@190.80.153.193 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 12:50:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 752676658 752676658 IN IP4 190.80.153.193 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.153.193 t=0 0 m=audio 10026 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I don't know why I don't see my provider's ip address. Isn't supposed to show in this debug? Here's my sip.conf file again maybe you can catch an error or something I'm missing. [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=route insucure=port,invite allow=all careinvite=yes Please helppp. ___ -- 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
[asterisk-users] Unable to open sound file error
Hello. I have a question regarind sound files in asterisk 1.6. I have a sound package in ulaw format and I would like to know if I have a sip extension with allow=alaw would asterisk convert that file to the codec the user is allowed to? I am having a problem playing a file that exist in /var/lib/asterisk/sounds/es/good.ulaw but asterisk is telling me it doesn't. Here's what I get when I try to dial the extension for test: [Nov 25 20:44:41] WARNING[4334]: file.c:650 ast_openstream_full: File good does not exist in any format [Nov 25 20:44:41] WARNING[4334]: file.c:933 ast_streamfile: Unable to open good (format 0x8 (alaw)): No such file or directory [Nov 25 20:44:41] WARNING[4334]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-09b52260 for good -- Executing [...@default:12] BackGround(SIP/102-09b52260, evening ) in new stack [Nov 25 20:44:41] WARNING[4334]: file.c:650 ast_openstream_full: File evening does not exist in any format [Nov 25 20:44:41] WARNING[4334]: file.c:933 ast_streamfile: Unable to open evening (format 0x8 (alaw)): No such file or directory [Nov 25 20:44:41] WARNING[4334]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-09b52260 for evening -- Executing [...@default:13] Hangup(SIP/102-09b52260, ) in new stack Any suggestions? Thanks in advanced 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
Re: [asterisk-users] can't get pap2 to register from outside the LAN.
How about adding: insecure=invite,port --- On Mon, 11/23/09, Tim Uckun timuc...@gmail.com wrote: From: Tim Uckun timuc...@gmail.com Subject: [asterisk-users] can't get pap2 to register from outside the LAN. To: asterisk-users@lists.digium.com Date: Monday, November 23, 2009, 8:25 PM I am having a hell of a problem trying to get a linksys pap2t to register with my asterisk from outside the LAN. I have tried every combination of NAT, outbound proxy, stun, specify external IP address etc and it just won't work. Here are the relevant details. In asterisk I have set the following. externip=my.ip.address localnet=192.168.0.0/255.255.0.0 nat=yes bindport=5060 here is the sip user deny=0.0.0.0/0.0.0.0 type=friend secret=blah qualify=yes port=5060 pickupgroup= permit=0.0.0.0/0.0.0.0 nat=yes mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/372 context=from-internal canreinvite=no callgroup= callerid=device 372 accountcode= call-limit=50 I have tried nat = no, nat=never, nat=route, and leaving out the nat no difference. On the linksys end I have tried everything I can think of. Nat, no nat, stun, hard coded external IP address etc. I have read dozens of web sites and have tried every suggestion given but no joy. I know other people have had the same problem but none of the links I ran into had a solution that worked for me. This device connects perfectly when inside the lan, take it out and it won't connect no matter what I do. Here is the sip debug trace. What truly puzzles me is the 401 not authorized packets. The password is correct, it connects fine inside the lan but the same username and password fails outside the LAN. [Nov 24 14:18:41] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108;tag=as1f31845b Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0dc307da Content-Length: 0 [Nov 24 14:18:41] Scheduling destruction of SIP dialog 'f4e6d9bc-59a7c...@192.168.50.183' in 32000 ms (Method: REGISTER) [Nov 24 14:18:42] ip --- SIP read from 218.101.6.157:5060 --- REGISTER sip:203.109.148.108 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER Max-Forwards: 70 Contact: 372 sip:3...@192.168.50.183:5060;expires=3600 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces - [Nov 24 14:18:42] --- (12 headers 0 lines) --- [Nov 24 14:18:42] Using latest REGISTER request as basis request [Nov 24 14:18:42] Sending to 218.101.6.157 : 5060 (NAT) [Nov 24 14:18:42] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:3...@203.109.148.108 Content-Length: 0 [Nov 24 14:18:42] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108;tag=as1f31845b Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0dc307da Content-Length: 0 [Nov 24 14:18:42] Scheduling destruction of SIP dialog 'f4e6d9bc-59a7c...@192.168.50.183' in 32000 ms (Method: REGISTER) [Nov 24 14:18:44] ip --- SIP read from 218.101.6.157:5060 --- REGISTER sip:203.109.148.108 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER Max-Forwards: 70 Contact: 372 sip:3...@192.168.50.183:5060;expires=3600 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces
Re: [asterisk-users] can't call through voip provider
Hello. I have my server running for about 30 days. Every time I did some changes to my sip.conf file I did reload in the cli. I thought this would change the new values. Somehow it wasn't. I decided to do a restart now and that used my new settings. The same settings I've been posting here the past week and weren't working. After restarting asterisk I'm able to use my provider via asterisk to make calls. I would like to thank those who helped me. --- On Fri, 11/20/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, November 20, 2009, 8:53 AM Sorry to bother you again with my problem but, is that I can't figure out what's going on with my setup. I have no idea of why my asterisk server is not communicating with my provider's. I've searched, googled, and can't find my solution. I've followed many tutorials but can't get anywhere. --- On Thu, 11/19/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 5:53 PM Nothing. I don't know what in the world is going on with my setup. Here's my FORWARD rules: eth0 = external nic, eth1 = lan 0 0 ACCEPT udp -- eth0 eth10.0.0.0/0 0.0.0.0/0 udp dpts:5060:5070 0 0 ACCEPT udp -- eth0 eth10.0.0.0/0 0.0.0.0/0 udp dpts:1:10100 162 ACCEPT udp -- eth1 eth00.0.0.0/0 0.0.0.0/0 udp dpts:5060:5070 36 2372 ACCEPT udp -- eth1 eth00.0.0.0/0 0.0.0.0/0 udp dpts:1:10100 0 0 ACCEPT tcp -- eth0 eth10.0.0.0/0 0.0.0.0/0 tcp dpts:5060:5070 0 0 ACCEPT tcp -- eth0 eth10.0.0.0/0 0.0.0.0/0 tcp dpts:1:10100 0 0 ACCEPT tcp -- eth1 eth00.0.0.0/0 0.0.0.0/0 tcp dpts:5060:5070 3 144 ACCEPT tcp -- eth1 eth00.0.0.0/0 0.0.0.0/0 tcp dpts:1:10100 and now the debug: etransmitting #5 (NAT) to 190.80.152.200:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.152.200:5060;branch=z9hG4bK794de7aa;rport Max-Forwards: 70 From: 102 sip:77...@190.80.152.200;tag=as5084570c To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.152.200 Call-ID: 22569d3b767276276c6c65c84b314...@190.80.152.200 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 22:53:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 135722140 135722140 IN IP4 190.80.152.200 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.200 t=0 0 m=audio 10074 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I'm already frustrated with this. --- On Thu, 11/19/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 5:11 PM On Thu, Nov 19, 2009 at 3:36 PM, Landy Landy landysacco...@yahoo.com wrote: Can someone please share with me a sip configuration to connect an asterisk server to a voip provider since my configuration isn't working for me. thanks. Who is your voipprovider? Did they give you the settings you're using in your sip.conf? Also, you've got some typos in your sip config (insucure = insecure, careinvite = canreinvite). You could try something like this: [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=yes canreinvite=yes insecure=very disallow=all allow=ulaw allow=alaw -- Thanks, --Warren Selby http://www.selbytech.com -Inline Attachment Follows
Re: [asterisk-users] can't call through voip provider
Sorry to bother you again with my problem but, is that I can't figure out what's going on with my setup. I have no idea of why my asterisk server is not communicating with my provider's. I've searched, googled, and can't find my solution. I've followed many tutorials but can't get anywhere. --- On Thu, 11/19/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 5:53 PM Nothing. I don't know what in the world is going on with my setup. Here's my FORWARD rules: eth0 = external nic, eth1 = lan 0 0 ACCEPT udp -- eth0 eth1 0.0.0.0/0 0.0.0.0/0 udp dpts:5060:5070 0 0 ACCEPT udp -- eth0 eth1 0.0.0.0/0 0.0.0.0/0 udp dpts:1:10100 1 62 ACCEPT udp -- eth1 eth0 0.0.0.0/0 0.0.0.0/0 udp dpts:5060:5070 36 2372 ACCEPT udp -- eth1 eth0 0.0.0.0/0 0.0.0.0/0 udp dpts:1:10100 0 0 ACCEPT tcp -- eth0 eth1 0.0.0.0/0 0.0.0.0/0 tcp dpts:5060:5070 0 0 ACCEPT tcp -- eth0 eth1 0.0.0.0/0 0.0.0.0/0 tcp dpts:1:10100 0 0 ACCEPT tcp -- eth1 eth0 0.0.0.0/0 0.0.0.0/0 tcp dpts:5060:5070 3 144 ACCEPT tcp -- eth1 eth0 0.0.0.0/0 0.0.0.0/0 tcp dpts:1:10100 and now the debug: etransmitting #5 (NAT) to 190.80.152.200:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.152.200:5060;branch=z9hG4bK794de7aa;rport Max-Forwards: 70 From: 102 sip:77...@190.80.152.200;tag=as5084570c To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.152.200 Call-ID: 22569d3b767276276c6c65c84b314...@190.80.152.200 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 22:53:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 135722140 135722140 IN IP4 190.80.152.200 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.200 t=0 0 m=audio 10074 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I'm already frustrated with this. --- On Thu, 11/19/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 5:11 PM On Thu, Nov 19, 2009 at 3:36 PM, Landy Landy landysacco...@yahoo.com wrote: Can someone please share with me a sip configuration to connect an asterisk server to a voip provider since my configuration isn't working for me. thanks. Who is your voipprovider? Did they give you the settings you're using in your sip.conf? Also, you've got some typos in your sip config (insucure = insecure, careinvite = canreinvite). You could try something like this: [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=yes canreinvite=yes insecure=very disallow=all allow=ulaw allow=alaw -- Thanks, --Warren Selby http://www.selbytech.com -Inline Attachment Follows- ___ -- 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] can't call through voip provider
Ok. I do NOT have ports 1-2 opened in. I guess I should try that and see if it works. I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I will keep you posted. I ran this test and there was no difference. I still can't get through. --- Retransmitting #5 (NAT) to 190.80.153.193:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.153.193:5060;branch=z9hG4bK727987ef Max-Forwards: 70 From: 102 sip:77...@190.80.153.193;tag=as23e02274 To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.153.193 Call-ID: 034bf0572cffb96f621211a8439aa...@190.80.153.193 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 12:50:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 752676658 752676658 IN IP4 190.80.153.193 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.153.193 t=0 0 m=audio 10026 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I don't know why I don't see my provider's ip address. Isn't supposed to show in this debug? Here's my sip.conf file again maybe you can catch an error or something I'm missing. [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=route insucure=port,invite allow=all careinvite=yes Please helppp. ___ -- 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] can't call through voip provider
Can someone please share with me a sip configuration to connect an asterisk server to a voip provider since my configuration isn't working for me. thanks. --- On Thu, 11/19/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 7:51 AM Ok. I do NOT have ports 1-2 opened in. I guess I should try that and see if it works. I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I will keep you posted. I ran this test and there was no difference. I still can't get through. --- Retransmitting #5 (NAT) to 190.80.153.193:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.153.193:5060;branch=z9hG4bK727987ef Max-Forwards: 70 From: 102 sip:77...@190.80.153.193;tag=as23e02274 To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.153.193 Call-ID: 034bf0572cffb96f621211a8439aa...@190.80.153.193 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 12:50:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 752676658 752676658 IN IP4 190.80.153.193 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.153.193 t=0 0 m=audio 10026 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I don't know why I don't see my provider's ip address. Isn't supposed to show in this debug? Here's my sip.conf file again maybe you can catch an error or something I'm missing. [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=route insucure=port,invite allow=all careinvite=yes Please helppp. ___ -- 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] can't call through voip provider
Nothing. I don't know what in the world is going on with my setup. Here's my FORWARD rules: eth0 = external nic, eth1 = lan 0 0 ACCEPT udp -- eth0 eth10.0.0.0/00.0.0.0/0 udp dpts:5060:5070 0 0 ACCEPT udp -- eth0 eth10.0.0.0/00.0.0.0/0 udp dpts:1:10100 162 ACCEPT udp -- eth1 eth00.0.0.0/00.0.0.0/0 udp dpts:5060:5070 36 2372 ACCEPT udp -- eth1 eth00.0.0.0/00.0.0.0/0 udp dpts:1:10100 0 0 ACCEPT tcp -- eth0 eth10.0.0.0/00.0.0.0/0 tcp dpts:5060:5070 0 0 ACCEPT tcp -- eth0 eth10.0.0.0/00.0.0.0/0 tcp dpts:1:10100 0 0 ACCEPT tcp -- eth1 eth00.0.0.0/00.0.0.0/0 tcp dpts:5060:5070 3 144 ACCEPT tcp -- eth1 eth00.0.0.0/00.0.0.0/0 tcp dpts:1:10100 and now the debug: etransmitting #5 (NAT) to 190.80.152.200:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.152.200:5060;branch=z9hG4bK794de7aa;rport Max-Forwards: 70 From: 102 sip:77...@190.80.152.200;tag=as5084570c To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.152.200 Call-ID: 22569d3b767276276c6c65c84b314...@190.80.152.200 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 22:53:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 135722140 135722140 IN IP4 190.80.152.200 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.200 t=0 0 m=audio 10074 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I'm already frustrated with this. --- On Thu, 11/19/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 5:11 PM On Thu, Nov 19, 2009 at 3:36 PM, Landy Landy landysacco...@yahoo.com wrote: Can someone please share with me a sip configuration to connect an asterisk server to a voip provider since my configuration isn't working for me. thanks. Who is your voipprovider? Did they give you the settings you're using in your sip.conf? Also, you've got some typos in your sip config (insucure = insecure, careinvite = canreinvite). You could try something like this: [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=yes canreinvite=yes insecure=very disallow=all allow=ulaw allow=alaw -- Thanks, --Warren Selby http://www.selbytech.com -Inline Attachment Follows- ___ -- 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] can't call through voip provider
I have the conf provided in last post.
exten = _9.,1,Dial(SIP/voipprovider/${EXTEN:1})
Yes, I have that in the dialplan.
Does sip show registry show that it's registered
successfully?
*CLI sip show registry
Host dnsmgr Username Refresh State
Reg.Time
0 SIP registrations.
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Re: [asterisk-users] can't call through voip provider
Hello. Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks. --- On Tue, 11/17/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, November 17, 2009, 7:33 AM Thanks for replying. Here is the output of sip set debug peer voipprovider: -- Called 1829257x...@voipprovider Retransmitting #1 (NAT) to myextip:5060: INVITE sip:18292574...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: 102 sip:usern...@myextip;tag=as78863882 To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77632...@190.80.152.7 Call-ID: 2908dd00500059761cc66bd81553e...@190.80.152.7 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 190.80.152.7 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.7 t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (NAT) to myextip:5060: INVITE sip:1829257x...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: 102 sip:usern...@myextip;tag=as78863882 To: sip:1829257x...@myextip Contact: sip:usern...@myextip Call-ID: 2908dd00500059761cc66bd81553e...@myextip CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 myextip s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.7 t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #3 (NAT) to myextip:5060: INVITE sip:1829257x...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: 102 sip:usern...@myextip;tag=as78863882 To: sip:1829257x...@myextip Contact: sip:usern...@myextip Call-ID: 2908dd00500059761cc66bd81553e...@myextip CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 myextip s=Asterisk PBX 1.6.1.5 c=IN IP4 myextip t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Scheduling destruction of SIP dialog '2908dd00500059761cc66bd81553e...@myextip' in 32000 ms (Method: INVITE) By looking at this trace I dont see my provider's ip address anywhere. I guess I'm doing something wrong in my conf. --- On Mon, 11/16/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, November 16, 2009, 9:51 PM On Mon, Nov 16, 2009 at 2:40 PM, Landy Landy landysacco...@yahoo.com wrote: snip I don't know what else to try. When I try to call I get this at the cli: == Using SIP RTP CoS mark 5 -- Executing [91xxx763x...@default:1] Dial(SIP/102-b6a06a40, SIP/1xxx763x...@voipprovider) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx763x...@voipprovider snip We could really use a little more of the CLI output of a failed call. Maybe increase your verbosity to at least 10. Also, what does the SIP debug of a call to the VOIP provider look like (from the cli, type sip set debug peer voipprovider)? -- Thanks, --Warren Selby http://www.selbytech.com
Re: [asterisk-users] can't call through voip provider
Thanks for replying. But how come I'm able to use a softphone to place calls from withing the lan? I really dont get it. What ports should I enable in the INPUT chain? --- On Wed, 11/18/09, Jared Smith jsm...@digium.com wrote: From: Jared Smith jsm...@digium.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 9:28 AM On Wed, 2009-11-18 at 06:01 -0800, Landy Landy wrote: Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks. It appears that Asterisk keeps sending an SIP INVITE message to your provider, but not getting any kind of response. After a number of attempts at re-transmitting the message, it's giving up. You need to check your network configuration and find out why responses from the provider aren't getting back to your Asterisk system. This is typically a problem with firewalls, either on the Asterisk system itself or between Asterisk and your VoIP provider. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] can't call through voip provider
According to the provider he says he doesn't see anything coming in on their side. I've had all ports FORWARD out to ACCEPT but, blocking incoming new connections. I thought when asterisk starts a communication with a remote server using an unprivate port to port 5060 theres already an ESTABLISHED communication. I don't know if I'm having problems with my firewall script or what but, since there isn't any new connections coming form outside I think I'm ok to accept only ESTABLISHED,RELATED coming in. I don't know but, I'm stuck with this problem and don't know what else to do. --- On Wed, 11/18/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 5:03 PM What does your provider see when you attempt to call them? Thanks, --Warren Selby On Nov 18, 2009, at 3:38 PM, Landy Landy landysacco...@yahoo.com wrote: Thanks for replying. But how come I'm able to use a softphone to place calls from withing the lan? I really dont get it. What ports should I enable in the INPUT chain? --- On Wed, 11/18/09, Jared Smith jsm...@digium.com wrote: From: Jared Smith jsm...@digium.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 9:28 AM On Wed, 2009-11-18 at 06:01 -0800, Landy Landy wrote: Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks. It appears that Asterisk keeps sending an SIP INVITE message to your provider, but not getting any kind of response. After a number of attempts at re-transmitting the message, it's giving up. You need to check your network configuration and find out why responses from the provider aren't getting back to your Asterisk system. This is typically a problem with firewalls, either on the Asterisk system itself or between Asterisk and your VoIP provider. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] can't call through voip provider
Ok. I do NOT have ports 1-2 opened in. I guess I should try that and see if it works. I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I will keep you posted. Thanks. --- On Wed, 11/18/09, Danny Nicholas da...@debsinc.com wrote: From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] can't call through voip provider To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 5:18 PM According to what I know, you have to have 5060 open out and 1-2 open in (you can cut this to as small as 1-10004). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Wednesday, November 18, 2009 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't call through voip provider According to the provider he says he doesn't see anything coming in on their side. I've had all ports FORWARD out to ACCEPT but, blocking incoming new connections. I thought when asterisk starts a communication with a remote server using an unprivate port to port 5060 theres already an ESTABLISHED communication. I don't know if I'm having problems with my firewall script or what but, since there isn't any new connections coming form outside I think I'm ok to accept only ESTABLISHED,RELATED coming in. I don't know but, I'm stuck with this problem and don't know what else to do. --- On Wed, 11/18/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 5:03 PM What does your provider see when you attempt to call them? Thanks, --Warren Selby On Nov 18, 2009, at 3:38 PM, Landy Landy landysacco...@yahoo.com wrote: Thanks for replying. But how come I'm able to use a softphone to place calls from withing the lan? I really dont get it. What ports should I enable in the INPUT chain? --- On Wed, 11/18/09, Jared Smith jsm...@digium.com wrote: From: Jared Smith jsm...@digium.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 9:28 AM On Wed, 2009-11-18 at 06:01 -0800, Landy Landy wrote: Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks. It appears that Asterisk keeps sending an SIP INVITE message to your provider, but not getting any kind of response. After a number of attempts at re-transmitting the message, it's giving up. You need to check your network configuration and find out why responses from the provider aren't getting back to your Asterisk system. This is typically a problem with firewalls, either on the Asterisk system itself or between Asterisk and your VoIP provider. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] can't call through voip provider
Thanks for replying. Here is the output of sip set debug peer voipprovider: -- Called 1829257x...@voipprovider Retransmitting #1 (NAT) to myextip:5060: INVITE sip:18292574...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: 102 sip:usern...@myextip;tag=as78863882 To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77632...@190.80.152.7 Call-ID: 2908dd00500059761cc66bd81553e...@190.80.152.7 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 190.80.152.7 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.7 t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (NAT) to myextip:5060: INVITE sip:1829257x...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: 102 sip:usern...@myextip;tag=as78863882 To: sip:1829257x...@myextip Contact: sip:usern...@myextip Call-ID: 2908dd00500059761cc66bd81553e...@myextip CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 myextip s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.7 t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #3 (NAT) to myextip:5060: INVITE sip:1829257x...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: 102 sip:usern...@myextip;tag=as78863882 To: sip:1829257x...@myextip Contact: sip:usern...@myextip Call-ID: 2908dd00500059761cc66bd81553e...@myextip CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 myextip s=Asterisk PBX 1.6.1.5 c=IN IP4 myextip t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Scheduling destruction of SIP dialog '2908dd00500059761cc66bd81553e...@myextip' in 32000 ms (Method: INVITE) By looking at this trace I dont see my provider's ip address anywhere. I guess I'm doing something wrong in my conf. --- On Mon, 11/16/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, November 16, 2009, 9:51 PM On Mon, Nov 16, 2009 at 2:40 PM, Landy Landy landysacco...@yahoo.com wrote: snip I don't know what else to try. When I try to call I get this at the cli: == Using SIP RTP CoS mark 5 -- Executing [91xxx763x...@default:1] Dial(SIP/102-b6a06a40, SIP/1xxx763x...@voipprovider) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx763x...@voipprovider snip We could really use a little more of the CLI output of a failed call. Maybe increase your verbosity to at least 10. Also, what does the SIP debug of a call to the VOIP provider look like (from the cli, type sip set debug peer voipprovider)? -- Thanks, --Warren Selby http://www.selbytech.com -Inline Attachment Follows- ___ -- 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] can't call through voip provider
Hello. Sorry to repost this message but, I don't have the original message in my inbox nor in my sent box. Well, last week I posted a problem I am having trying to use an asterisk server use a voip provider and a pstn. Pstn works fine but, I cant even connect to my provider's server. I don't know what I'm doing wrong. I tried using a soft phone and I'm able to register and make calls with it but, when it comes to rerouting the call through asterisk I not able to establish a call. This is my setup: modem -- router/firewall LAN The asterisk server is on the lan side. I have the modem in bridge mode which assings my router/firewall the external ip address. I have FORWARD to ACCEPT in the router and I still cant establish a connection. My sip.conf file looks like this: [general] externhost=optimumwireless.com localnet=172.16.0.0/16 register = username:sec...@my.service_provider.tld language=es ;allow=gsm allow=all [voipprovider] type=friend host=208.78.163.3 username=username fromuser=username secret=password port=5060 dtmfmode=rfc2833 nat=yes insucure=port,invite allow=all careinvite=yes I don't know what else to try. When I try to call I get this at the cli: == Using SIP RTP CoS mark 5 -- Executing [91xxx763x...@default:1] Dial(SIP/102-b6a06a40, SIP/1xxx763x...@voipprovider) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx763x...@voipprovider Please help me with this I'm running out of options. Thanks in advanced 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
Re: [asterisk-users] Can't connect to voip provider over NAT
According to my provider they´re not receiving any request from us but, now everytime I try to place a call through them I´m getting: *CLI sip show peers Name/username HostDyn Nat ACL Port Status 100(Unspecified)D 5060 Unmonitored 101(Unspecified)D 5060 Unmonitored 102/102172.16.0.15 D 5060 Unmonitored 103/103(Unspecified)D 5060 Unmonitored 104(Unspecified)D 5060 Unmonitored 105(Unspecified)D 5060 Unmonitored 106(Unspecified)D 5060 Unmonitored 107(Unspecified)D 5060 Unmonitored voipprovider/1800890999 MYEXTERNALIP N 5060 Unmonitored 9 sip peers [Monitored: 0 online, 0 offline Unmonitored: 9 online, 0 offline] == Using SIP RTP CoS mark 5 -- Executing [18008909...@default:1] Dial(SIP/102-b6a05db0, SIP/18292574...@voipprovider) in new stack == Using SIP RTP CoS mark 5 -- Called 18008909...@voipprovider It just hangs here and nothing happens.. Here´s my sip.conf file: [general] externhost=myexternalip localnet=172.16.0.0/16 register = username:passw...@sip-gw.advancedvoip.com.do allow=all [voipprovider] type=peer host=sip-gw.advancedvoip.com.do username=username fromuser=username secret=password port=5060 canreinvite=YES dtmfmode=rfc2833 nat=yes What I´m I doing wrong? ___ -- 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] Can't connect to voip provider over NAT
I have iptables FORWARD to ACCEPT by default: iptables -P FORWARD ACCEPT and still have the same problems. Now, the dsl modem is also opened. not blocking any ports as well. --- On Sat, 11/14/09, Michelle Dupuis supp...@ocg.ca wrote: From: Michelle Dupuis supp...@ocg.ca Subject: Re: [asterisk-users] Can't connect to voip provider over NAT To: 'Asterisk Users List' asterisk-users@lists.digium.com Date: Saturday, November 14, 2009, 1:03 PM I'll start with a guess - your asterisk box or firewall is blocking SIP ports. Diagnose that first (stop iptables/check iptables if unsafe) and try again... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Saturday, November 14, 2009 10:15 AM To: Asterisk Users List Subject: Re: [asterisk-users] Can't connect to voip provider over NAT According to my provider they´re not receiving any request from us but, now everytime I try to place a call through them I´m getting: *CLI sip show peers Name/username Host Dyn Nat ACL Port Status 100 (Unspecified) D 5060 Unmonitored 101 (Unspecified) D 5060 Unmonitored 102/102 172.16.0.15 D 5060 Unmonitored 103/103 (Unspecified) D 5060 Unmonitored 104 (Unspecified) D 5060 Unmonitored 105 (Unspecified) D 5060 Unmonitored 106 (Unspecified) D 5060 Unmonitored 107 (Unspecified) D 5060 Unmonitored voipprovider/1800890999 MYEXTERNALIP N 5060 Unmonitored 9 sip peers [Monitored: 0 online, 0 offline Unmonitored: 9 online, 0 offline] == Using SIP RTP CoS mark 5 -- Executing [18008909...@default:1] Dial(SIP/102-b6a05db0, SIP/18292574...@voipprovider) in new stack == Using SIP RTP CoS mark 5 -- Called 18008909...@voipprovider It just hangs here and nothing happens.. Here´s my sip.conf file: [general] externhost=myexternalip localnet=172.16.0.0/16 register = username:passw...@sip-gw.advancedvoip.com.do allow=all [voipprovider] type=peer host=sip-gw.advancedvoip.com.do username=username fromuser=username secret=password port=5060 canreinvite=YES dtmfmode=rfc2833 nat=yes What I´m I doing wrong? ___ -- 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] FW: hi Dan
Pre-judging people doesn't work on mailing lists given the inherent language barriers, etc. I believe language barriers can cause many problems when trying to communicate. I might say something in another language trying to translate a phrase or something, that might not have the same meaning I´m trying to get accross. I´m billingual myself, english is my second language but, I carefully try to choose the correct words when asking for help or even talking to anybody so I don´t offend that person. Let´s have compassion with this guy and let´s give him a break. Looks like his having a lot of problems trying to resolve his issues and frustrations have started to get on him. I put myself on his shoes and know how frustrating things can get from time to time. Also, we need to understand not all everyone has the same understanding capabilities. Some of us are ¨dumber¨ than others. What´s easy for you may not be easy for me and viceversa. ___ -- 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] Can't connect to voip provider over NAT
Have you tried nat=yes in the definition in sip.conf? Yes, I have that definition in sip.conf. Now, I'm getting the following error -- SIP/voipprovider-094132d8 is making progress passing it to SIP/102-09423d58 -- Got SIP response 603 Declined back from 208.xx.xx.xx -- SIP/voipprovider-094132d8 is busy == Everyone is busy/congested at this time (1:1/0/0) and I get a This account number is not valid on the headset. I've called my provider and they've said that everything is fine at their end. I don't know why I'm getting the message saying the account is not valid. ___ -- 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] Can't connect to voip provider over NAT
Hello. I'm trying to test an Asterisk server by using a VOIP provider for international calls but, I'm having problems trying to get my server communicate with theirs. I don't know if I'm having all these issues becuase I'm behind NAT or what. I have the following in my server's sip.conf: [provider] type=peer host=theprovider's server username=username secret=password port=5060 canreinvite=YES dtmfmode=rfc2833 I've tried opening all ports to test this but, still doesn't work. Now, I need to know which especific ports to open in order to allow sip flow correctly. Also enabled/opened ports 5060 - 5070 and the rtp: rtpstart=1 rtpend=2 Don't know what else to try. Please help. Thanks in advanced 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
Re: [asterisk-users] ivr menu not hanging up call
exted != exten Ok. That was the actual error, I guess I needed some sleep. Thanks. ___ -- 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] ivr menu not hanging up call
I am testing an ivr but I'm having problems. The call keeps looping and it
doesn't hangup the call after passing three times through the menu. Here's my
conf:
exten = s,n,NoOp(Here's Count)
exten = s,n,NoOp(${COUNT})
;123,n,Set(COUNT=$[${COUNT} - 1])
exten = s,n,GotoIf($[${COUNT} = 4]?33,1:44,1 )
exten = 1,1,goto(tech-support,s,1)
exten = 2,1,goto(sales,s,1)
exten = 3,1,goto(cust-service,s,1)
exten = 100,1,goto(wilson,s,1)
exten = 102,1,goto(sales,s,1)
exten = i,1,Playback(invalid)
exten = i,n,Playback(please-try-again)
exten = i,n,goto(ivr,s,5)
exten = i,n,Playback(goodbye)
exten = i,n,Hangup
exten = 33,1,PlayBack(please-try-again-later)
exten = 33,n,PlayBack(call-terminated)
exten = 33,n,PlayBack(goodbye)
exted = 33,n,HangUp()
exten = 44,1,goto(ivr,s,5)
exten = t,1,goto(ivr,s,2)
exten = h,1,Hangup
When it enters extension 33 it should hangup the call but, if the caller stays
on the line the exten = t,1,goto(ivr,s,2) takes over and the menu keeps
repeating. Should I just remove that t extension?
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Re: [asterisk-users] No sound on voicemail from analog line
Do you mean that incoming calls on your PSTN line works as they should, but not when they reach the voicemail? or that incomming calls on PSTN are always mute? Incoming calls on PSTN line work as they should but, when someone leaves a voicemail message the messege is mute. When I try to retrieve the messeges I get the prompt that says how many messeages are there. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No sound on voicemail from analog line
--- On Thu, 10/8/09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] No sound on voicemail from analog line To: asterisk-users@lists.digium.com Date: Thursday, October 8, 2009, 4:11 PM On Thu, Oct 08, 2009 at 12:43:00PM -0700, Landy Landy wrote: Hello. I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound. What do you mean by voicemail from PSTN? Asterisk's voicemail or the provider's ? The cards is FXS? FXO? T1? E1? Well, what I mean is on calls coming in from outside on the analog line. The card is one of those old modems X100p, I guess is a clone card. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No sound on voicemail from analog line
Hello. I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound. What can cause that problem? Thanks in advanced for you help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI congestion problem
I have a similar problem with DAHDI. If my server gets rebooted, I can't make any calls until the a call come in from outside. From there I can answer the call and DAHDI works fine afterwards. --- On Mon, 9/28/09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] DAHDI congestion problem To: asterisk-users@lists.digium.com Date: Monday, September 28, 2009, 2:25 AM Just to answer your side issue: On Sun, Sep 27, 2009 at 04:05:30PM -0500, Andy Howell wrote: The only Warning or Error I see is when asterisk first starts a new call. logger.c: -- Starting simple switch on 'DAHDI/1-1' [Sep 27 15:55:50] WARNING[4199] chan_dahdi.c: Unable to enable echo cancellation on channel 1 (No such device) On my TDM400P card, channel 1 is my analog phone, 2 my fax, and 4 the POTS line. More config files etc below. Any ideas? Thanks, Andy /etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Wed Jun 10 22:20:05 2009 -- do not hand edit # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) fxols=1 #echocanceller=mg2,1 fxols=2 #echocanceller=mg2,2 # channel 3, WCTDM/4/2, no module. fxsks=4 echocanceller=mg2,4 You get the ENODEV (No such device) error when trying to create an echo canceller on channel 1 simply because there isn't any echo canceller on channel one. Enable the above echocanceller lines, or use a single one for all of them. But that's not your real issue. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI congestion problem
In your case: is the problem reset by restarting asterisk? 'dahdi resstart'? The problem does not reset by restarting asterisk. I've noticed that I can call other sip phones but, when trying to call out, I get the same (Busy/Congested/Not-Available) congested messege. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI channel congested busy
I also found this weird, I thought my equipment was the problem. Good to know about this issue so, Digium takes care of the problem. I'm running: asterisk-1.6.1.5 dahdi-linux-2.2.0.2 libpri-1.4.10.1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
