Re: [asterisk-users] Assistance sending mass sms to cellphones
Robert. Thanks for replying. --- On Fri, 8/5/11, Robert Huddleston wrote: > From: Robert Huddleston > Subject: Re: [asterisk-users] Assistance sending mass sms to cellphones > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > > 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] 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
[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
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 wrote: > From: Jeremy Kister > Subject: Re: [asterisk-users] app_swift.c:338 engine: Failed to set voice > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > 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] 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
[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:
-- 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.
--
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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 wrote:
> From: Vahan Yerkanian
> 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
> >>
> --AGI
> 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
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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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
-- AGI 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 wrote:
> From: Faisal Hanif
> Subject: Re: [asterisk-users] a2billing for residential voip usage
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>
> 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
> wrote:
>
> > From: Vardan Harutyunyan
> > 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, ram
>
> > wrote:
> > >>
> > >> From: ram
> > >> Subject: Re: [asterisk-users] a2billing for
> > residential voip usage
> > >> To: "Asterisk Users Mailing List -
> Non-Commercial
> > Di
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 wrote: > From: Faisal Hanif > Subject: Re: [asterisk-users] a2billing for residential voip usage > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > > 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 > wrote: > > > From: Vardan Harutyunyan > > 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, ram > > > wrote: > > >> > > >> From: ram > > >> Subject: Re: [asterisk-users] a2billing for > > residential voip usage > > >> To: "Asterisk Users Mailing List - > Non-Commercial > > Discussion" > > >> > > >> 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 > > >> 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, > > >&g
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 wrote: > From: Vardan Harutyunyan > 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, ram > wrote: > >> > >> From: ram > >> Subject: Re: [asterisk-users] a2billing for > residential voip usage > >> To: "Asterisk Users Mailing List - Non-Commercial > Discussion" > >> > >> 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 > >> 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& sc
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 wrote: From: ram Subject: Re: [asterisk-users] a2billing for residential voip usage To: "Asterisk Users Mailing List - Non-Commercial Discussion" 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 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
[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] 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] 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] 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(${EXTEN},u)
exten => _1XX,n,Hangup()
include => agents
include => queue
include => local-iax
include => voicemail
include => timeofday
include => parkedcalls
include => pickup
include => to_client
include => test-agi
include => menu
that goes to an ivr. Can this be a security bridge?
--- On Mon, 2/15/10, Tony Mountifield wrote:
> From: Tony Mountifield
> Subject: Re: [asterisk-users] Important security alert: update your dialplans
> now!
> To: asterisk-users@lists.digium.com
> Date: Monday, February 15, 2010, 11:58 AM
> In article <699ee941002150033t7c6e1be5xdba76cb0f68d5...@mail.gmail.com>,
> Lenz Emilitri
> wrote:
> > -=-=-=-=-=-
> > -=-=-=-=-=-
> >
> > Or one could simply rewrite to:
> >
> > [incoming-from-voip]
> > exten =>
> XXX,1,Dial(${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(${ext...@incoming-from-voip-old)
>
> etc.
>
> Tony
> --
> Tony Mountifield
> Work: t...@softins.co.uk
> - http://www.softins.co.uk
> Play: t...@mountifield.org
> - http://tony.mountifield.org
>
> --
> _
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] How can I get codec info on active calls
Hello All. I would like to know what codec is being used during a call. For example if I have 3 channels on 3 active calls how can I find what codec is beeing used by each client? Thanks in advanced. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help getting info from caller
I was able to test the script, here is what I have:
[CODE]
#!/usr/bin/php -q
[/CODE]
extensions.conf:
[test-agi]
exten => 33,1,Answer()
exten => 33,n,Wait(0.5)
exten => 33,n,BackGround(please-enter)
exten => 33,n,BackGround(customer-accounts)
exten => 33,n,Read(ACCOUNT,,4)
;exten => 33,n,BackGround(enter-password)
;exten => 33,n,Read(PASSWORD,,4)
exten => 33,n,AGI(testphp.agi,${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 ...${AGISTATUS}...")
exten => 33,n,hangup()
I was able to get the balance from the db table and have asterisk tell the
caller.
I tried to include some files in agi but kept getting an error that the file
didn't exist.
I would like to thank you for helping me out with this. Is a good starting
point.
Now, I have another thing in mind:
Is asterisk or any other program that can work along side * able to say a name
or any word? For example:
Lets say I have a table with name and last name I would like asterisk to say
"balance for john doe is 100 dollars"... Is this possible?
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Re: [asterisk-users] Help getting info from caller
--- On Sat, 1/2/10, Landy Landy wrote:
> From: Landy Landy
> Subject: [asterisk-users] Help getting info from caller
> To: asterisk-users@lists.digium.com
> Date: Saturday, January 2, 2010, 9:01 AM
> Hello. Happy New Year to everyone.
>
> I have a small WISP and would like to have customers to
> call our number to check their balance. I am planning on
> writing an AGI with php so it can get the customer info from
> the customer database. I don't know how to interact with the
> caller while in the agi script so this is what I have in
> mind:
>
>
> [test-agi]
> exten => 33,1,Answer()
> exten => 33,n,Wait(0.5)
> exten => 33,n,BackGround(please-enter)
> exten => 33,n,BackGround(customer-account)
> exten => 33,n, I would like to set a variable here
> but don't know how -
> exten => 33,n,BackGround(enter-password)
> exten => 33,n, I would like to set a variable here
> but don't know how -
> exten => 33,n,AGI(testphp.agi,${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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[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] Best way ro run 2 or more asterisk servers?
--- On Wed, 12/16/09, Landy Landy wrote: > From: Landy Landy > Subject: Re: [asterisk-users] Best way ro run 2 or more asterisk servers? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > 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?
> 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
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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[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 wrote:
> From: Warren Selby
> Subject: Re: [asterisk-users] Unable to open file...
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> 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
>
> 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()
> >
> >
> >
> >
> >
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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()
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[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.
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[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
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
[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] 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 wrote: > From: meetmecall > Subject: Re: [asterisk-users] can't call through voip provider > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > 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 => > ::@ipness.net:6060 > > > And you need a sip entry in sip.conf. For me it looks > something like > > [] > type=friend > host=ipness.net > fromuser= > fromdomain=ipness.net > username= > 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 > wrote: > > > >> From: Landy Landy > >> Subject: Re: [asterisk-users] can't call through > voip provider > >> To: "Asterisk Users Mailing List - Non-Commercial > Discussion" > >> > > >> 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" > >> ;tag=as23e02274 > >> To: > >> Contact: > >> 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=rt
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 wrote:
> From: Landy Landy
> Subject: [asterisk-users] Unable to open sound file error
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> 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.
>
>
>
>
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[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.
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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 wrote: > From: Tim Uckun > 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 > ;tag=e25fccc07a79cd65o0 > To: 372 ;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 > ;tag=e25fccc07a79cd65o0 > To: 372 > Call-ID: f4e6d9bc-59a7c...@192.168.50.183 > CSeq: 26779 REGISTER > Max-Forwards: 70 > Contact: 372 > ;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 > ;tag=e25fccc07a79cd65o0 > To: 372 > 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: > 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 > ;tag=e25fccc07a79cd65o0 > To: 372 ;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 > ;tag=e25fccc07a79cd65o0 > To: 372 > Call-ID: f4e6d9bc-59a7c...@192.168.50.183 > CSeq: 26779 REGISTER > Max-Forwards: 70 > Contact: 372 > ;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 14:18:44] --- (12 headers 0 lines) --- > [Nov 24 14:18:44] Using latest REGISTER request as basis > request > [Nov 24 14:18:4
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 wrote: > From: Landy Landy > Subject: Re: [asterisk-users] can't call through voip provider > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > 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 > wrote: > > > From: Landy Landy > > Subject: Re: [asterisk-users] can't call through voip > provider > > To: "Asterisk Users Mailing List - Non-Commercial > Discussion" > > 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" > > ;tag=as5084570c > > To: > > Contact: > > 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 > > wrote: > > > > > From: Warren Selby > > > Subject: Re: [asterisk-users] can't call through > voip > > provider > > > To: "Asterisk Users Mailing List - > Non-Commercial > > Discussion" > > > Date: Thursday, November 19, 2009, 5:11 PM > > > On Thu, Nov 19, > > > 2009 at 3:36 PM, Landy Landy > > > wrote: > > > > > > Can someone please share with me a
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 wrote: > From: Landy Landy > Subject: Re: [asterisk-users] can't call through voip provider > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > 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" > ;tag=as5084570c > To: > Contact: > 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 > wrote: > > > From: Warren Selby > > Subject: Re: [asterisk-users] can't call through voip > provider > > To: "Asterisk Users Mailing List - Non-Commercial > Discussion" > > Date: Thursday, November 19, 2009, 5:11 PM > > On Thu, Nov 19, > > 2009 at 3:36 PM, 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. > > > > > > > > > > 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
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
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" ;tag=as5084570c To: Contact: 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 wrote: > From: Warren Selby > Subject: Re: [asterisk-users] can't call through voip provider > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Date: Thursday, November 19, 2009, 5:11 PM > On Thu, Nov 19, > 2009 at 3:36 PM, 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. > > > > > 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
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 wrote: > From: Landy Landy > Subject: Re: [asterisk-users] can't call through voip provider > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > 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" > ;tag=as23e02274 > To: > Contact: > 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
> > 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" ;tag=as23e02274 To: Contact: 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
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 wrote: > From: Danny Nicholas > Subject: Re: [asterisk-users] can't call through voip provider > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > > 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 > wrote: > > > From: Warren Selby > > Subject: Re: [asterisk-users] can't call through voip > provider > > To: "Asterisk Users Mailing List - Non-Commercial > Discussion" > > > 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 > > > > 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 > > wrote: > > > > > >> From: Jared Smith > > >> Subject: Re: [asterisk-users] can't call > through > > voip provider > > >> To: "Asterisk Users Mailing List - > Non-Commercial > > Discussion" > > > >> > > > >> 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-
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 wrote: > From: Warren Selby > Subject: Re: [asterisk-users] can't call through voip provider > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > 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 > > 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 > wrote: > > > >> From: Jared Smith > >> Subject: Re: [asterisk-users] can't call through > voip provider > >> To: "Asterisk Users Mailing List - Non-Commercial > Discussion" > >> > > >> 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
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 wrote: > From: Jared Smith > Subject: Re: [asterisk-users] can't call through voip provider > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > 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
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 wrote:
> From: Landy Landy
> Subject: Re: [asterisk-users] can't call through voip provider
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> 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" ;tag=as78863882
> To:
> Contact:
> 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" ;tag=as78863882
> To:
> Contact:
> 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" ;tag=as78863882
> To:
> Contact:
> 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
> wrote:
>
> > From: Warren Selby
> > Subject: Re: [asterisk-users] can't call through voip
> provider
> > To: "Asterisk Users Mailing List - Non-Commercial
> Discussion"
> > Date: Monday, November 16, 2009, 9:51 PM
> > On Mon, Nov 16,
> > 2009 at 2:40 PM, Landy Landy
> > wrote:
> >
> >
> >
> > 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",
> > "
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" ;tag=as78863882
To:
Contact:
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" ;tag=as78863882
To:
Contact:
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" ;tag=as78863882
To:
Contact:
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 wrote:
> From: Warren Selby
> Subject: Re: [asterisk-users] can't call through voip provider
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Date: Monday, November 16, 2009, 9:51 PM
> On Mon, Nov 16,
> 2009 at 2:40 PM, Landy Landy
> wrote:
>
>
>
> 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
>
>
>
> 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
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[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.
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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
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 wrote:
> From: Michelle Dupuis
> Subject: Re: [asterisk-users] Can't connect to voip provider over NAT
> To: "'Asterisk Users List'"
> 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?
>
>
>
>
> ___
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> asterisk-users mailing list
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>
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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?
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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= username= secret= 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 wrote: > From: Tzafrir Cohen > 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 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
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 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 wrote: > From: Tzafrir Cohen > 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
