[asterisk-users] Prevent Agent Login from a second extension
Hi friends, Is there any way to prevent an Agent from logging in from a second extension if he is already logged on from an extension. Right now, the scenario is if he login from a second extension, asterisk will automatically log him off from first extension. What I need is that asterisk should tell him that he is already logged on from an extension and should prevent him from logging in again from another extn. The problem with existing scenario is that, I am not getting CDR record for the automatic log out event. I need this for evaluation purposes. I am using asterisk 1.2.30. I have 1.4 also but that also is having the same behavior. Thanks in advance for any help. Regards Shanavaz. ___ -- 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] Prevent Agent Login from a second extension
I think you have to write your own agent login and logout so that you will not have this problem. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shanavaz E A Sent: Wednesday, 2 September 2009 4:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Prevent Agent Login from a second extension Hi friends, Is there any way to prevent an Agent from logging in from a second extension if he is already logged on from an extension. Right now, the scenario is if he login from a second extension, asterisk will automatically log him off from first extension. What I need is that asterisk should tell him that he is already logged on from an extension and should prevent him from logging in again from another extn. The problem with existing scenario is that, I am not getting CDR record for the automatic log out event. I need this for evaluation purposes. I am using asterisk 1.2.30. I have 1.4 also but that also is having the same behavior. Thanks in advance for any help. Regards Shanavaz. ___ -- 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] queue issue
It depends on what you want to do to people who are queued; if you want them to be queued, you create a queue with only one member, and have agents log on and log off as necessary; if you don't want callers to be queued, likely I would not use a queue but woul dial the agent straight. l. PS. this is quite an unusual requirement, what is it for? 2009/9/1 Paul Hales pdha...@optusnet.com.au I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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 selective install
On Tue, Sep 01, 2009 at 10:53:00PM -0300, Valter Nogueira wrote: Is there any way to not install all DAHDI drivers? All that I need is the dummy driver for timming purposes. Edit drivers/dahdi/Kbuild and rem-out all drivers besides dahdi/dahdi-base and dahdi-dummy . -- 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
Re: [asterisk-users] how does wrapuptime work in queue.conf
Aht i would do is prepare a music on hold that has embedded the advertisements ( like one every 20 or 30 seconds) so that the caller hears more advertisements as the call progresses; and they are queued immediately, so no time is wasted. l. 2009/8/27 Andy Kuo aku...@gmail.com Hi Barry, Thank you for the hint, but I forgot to mention that we have a few advertisements, and we want the callers to listen to only one at a time, and in a round robin or random order. Using Playback() doesn't seem to serve that purpose. Is there any better way to achieve that? Thanks. Andy On Thu, Aug 27, 2009 at 11:56 AM, Barry L. Klineblkl...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Andy Kuo wrote: Hi list, I'd like to have the callers to listen to the advertisement (music on hold) before the agents answer them. So, I have wrapuptime=10 in queue.conf, but the call still goes straight to the agents without delay. Andy -- wrapuptime is the number of seconds that Asterisk waits between the time a agent hangs up with a caller and the next time that Asterisk sends a call to the newly-available agent. Wrap up time gives the agent a few moments to complete his last call and prepare for the next. What you need to do is use Playback() for your advertisement, then Queue() the call. Otherwise it acts just as you said, provided an agent is available. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKltbjCFu3bIiwtTARAjE0AKCGFEchqYoGWyaeHqlIH+iNyzBKygCgqibn X/gSnE7W7EHnwiUpRC1FLRs= =pdMh -END PGP SIGNATURE- ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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] queue issue
A situation where staff want a mobile and their SIP handset to share an extension - but to make sure the mobile or SIP handset do not ring if they are speaking on the other one... PaulH Lenz Emilitri wrote: It depends on what you want to do to people who are queued; if you want them to be queued, you create a queue with only one member, and have agents log on and log off as necessary; if you don't want callers to be queued, likely I would not use a queue but woul dial the agent straight. l. PS. this is quite an unusual requirement, what is it for? 2009/9/1 Paul Hales pdha...@optusnet.com.au mailto:pdha...@optusnet.com.au I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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
[asterisk-users] Skype for Asterisk callfile question
Hi list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[originator@]destination) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228 MaxRetries: 0 Dial(Skype/asterisk...@somebodyonskype) Priority: 1 Callerid: Somebodyonskype somebodyonskype --- SIP/228 is my desk phone, i purposely did not include a context (is it necessary for Skype calls? i guess not because i also did not create a 'dial plan' for it, should be just dump and go i guess) But i guess something must be wrong on my dial string : [Sep 2 09:39:10] NOTICE[8834]: pbx_spool.c:255 apply_outgoing: Syntax error at line 3 of /var/spool/asterisk/outgoing/REMCO.CALL [Sep 2 09:39:10] WARNING[8834]: pbx_spool.c:260 apply_outgoing: At least one of app or extension must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/REMCO.CALL [Sep 2 09:39:10] WARNING[8834]: pbx_spool.c:427 scan_service: Invalid file contents in /var/spool/asterisk/outgoing/REMCO.CALL, deleting [Sep 2 09:39:10] WARNING[8834]: pbx_spool.c:482 scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/REMCO.CALL' Where am i going wrong? Thanks!! ___ -- 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] Skype for Asterisk callfile question
On 2/09/09 7:45 PM, Remco Barendse wrote: So i create a callfile that looks like this: --- Channel: SIP/228 MaxRetries: 0 Dial(Skype/asterisk...@somebodyonskype) Priority: 1 Callerid: Somebodyonskypesomebodyonskype You're combining technologies there :) You can do: Channel Context Extension Priority Or Channel Application Data Looks like you want Channel: SIP/228 Application: Dial Data: Skype/asterisk...@somebodyonskype -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] web meetme PHP undefined variable
I am hoping maybe some of you have come across these before in your experience with web meetme. Below are the messages im receiveing when I load the web meetme home page. Notice: Undefined variable: s in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 9 Notice: Undefined variable: logoff_section in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 12 Notice: Undefined variable: logoff_section in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 19 Notice: Undefined index: auth in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 29 Notice: Undefined variable: AUTH_USER in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 39 Notice: Undefined index: auth in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 45 Notice: Undefined index: auth in /usr/local/apache2/htdocs/web-meetme/lib/header.inc on line 28 Notice: Undefined variable: logoff_sel in /usr/local/apache2/htdocs/web-meetme/lib/header.inc on line 35 ___ -- 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] web meetme PHP undefined variable
On 2/09/09 8:14 PM, Glen Ganderton wrote: I am hoping maybe some of you have come across these before in your experience with web meetme. Below are the messages im receiveing when I load the web meetme home page. I'd say it's just a warning. If you edit: /etc/php/apache2/php.ini and look for display_errors=E_ALL and change it to something a bit less intense :) Note that the php.ini file may be somewhere else :) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Problem with Cisco 7911G and ABE 2.1.2C - randomly cannot DIAL
Guys, I assure you this is probably the most interesting and weird problem you have encountered (or definitely up there). I'm using ABE 2.1.2C and roughly 500 or so Cisco 7911G Phones. The following is what happens: When trying to dial a number from the cisco 7911G phone it may randomly get stuck on 'Dialing'. The SIP history on the asterisk end goes like this: 1. Cisco - INVITE - Asterisk 2. Asterisk - Proxy Authorization Required - Cisco 3. Cisco - gone to sleep. If I login to the cisco phone, (using log/log) i can see that the phone is trying constantly to reach the asterisk server. It retransmits the packet 10 times and then gives up and the call terminates. This behavior depends primarily on the length of the number dialed, though sometimes the 'mood' of the Cisco also comes into play. It may be impossible to dial 11 digit numbers at one point but 2 hours later it may magically work. All other phones on this network work perfectly. This includes Linksys, polycom and Grandstream. The most expensive phone causes issues - go figure! :) The following alleviates the problem: 1. If I mess around with the length of the 'realm' parameter in sip.conf it can randomly make the phones happy or unhappy. The smaller the better, however some phones may still get stuck on dialing. 2. Removing secrets from the sip.conf friend entry fixes this issue. Its the proxy authorization message that really pisses the cisco off. 4. I have tried randomly to use insecure, fromdomain, etc but nothing works. Also- the same phones work perfectly with our Asterisk 1.4.26 office server. Help is most appreciated. We are at a loss here. -- Faraz R Khan CEO Emergen Consulting Pvt Ltd www.emergen.biz +92.21.529.0381 (3 lines) x200 ___ -- 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] internet connection lagged - * lagged ...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, In a local network, an asterisk with 30 phones. For external call, there is a few ITSP. When internet connection lagged (ping as 1800 ms) the internal phones also lagged. ITSP and phones are then UNREACHABLE. If it restart asterisk (always with the internet connection lagged), the ITSP does no register (normal) but internal phones does no register also ... During asterisk start, module chan_sip is sufficiently long to load. Do you have an idea ? Best regards, Antoine Patte -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkqeLmYACgkQBnIOcv+j7+yPKgCgi3myJplD6HZl3bSK7Lu8dPGW j/EAoN0X6LOG8qmFi+iYnZIx8QidWrL6 =eF7l -END PGP SIGNATURE- ___ -- 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] Skype for Asterisk callfile question
On Wed, 2 Sep 2009, Matt Riddell wrote: On 2/09/09 7:45 PM, Remco Barendse wrote: So i create a callfile that looks like this: --- Channel: SIP/228 MaxRetries: 0 Dial(Skype/asterisk...@somebodyonskype) Priority: 1 Callerid: Somebodyonskypesomebodyonskype You're combining technologies there :) Not hindered by any knowledge i was trying to get things working :) Thanks, it seems to make sense to Asterisk now :) The first time i configured SFA to allow incoming calls only, reloading the module does not allow outbound calls still (direction is not mentioed as a directive for which asterisk needs to be restarted but it doesn't really matter). Thanks again for your 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] SIP and other phones other then local network
Thanks Matt ! I found the configuration of SIP phones little bit more complex as compare to IAX ... So howz about using IAX2 Any other that will require less or zero configuration other than Asterisk server On Wed, Sep 2, 2009 at 12:28 AM, Matt Riddell li...@venturevoip.com wrote: On 2/09/09 2:28 AM, Pascal Bruno wrote: For example if it was Alex to reply to that msg, i would feel bad for this guy, because Alex would make him feel like if he cannot do this by himself or use google to find that answer by himself, he does not belong to that list. He would never give him a chance and try to help him. :) That's what I'm here for :) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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 -- Best Regards Shakeel Abbas ___ -- 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] internet connection lagged - * lagged ...
On Wed, 2 Sep 2009, Antoine Patte wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, In a local network, an asterisk with 30 phones. For external call, there is a few ITSP. When internet connection lagged (ping as 1800 ms) the internal phones also lagged. ITSP and phones are then UNREACHABLE. If it restart asterisk (always with the internet connection lagged), the ITSP does no register (normal) but internal phones does no register also ... During asterisk start, module chan_sip is sufficiently long to load. Do you have an idea ? DNS. Run a caching DNS server on your Asterisk box, or a suitable device on your network. (eg. the DHCP server) Gordon ___ -- 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] SIP and other phones other then local network
On 2/09/09 9:10 PM, ABBAS SHAKEEL wrote: Thanks Matt ! I found the configuration of SIP phones little bit more complex as compare to IAX ... So howz about using IAX2 Any other that will require less or zero configuration other than Asterisk server IAX2 is a touchy subject with some people. I personally use it as much as possible (mainly for the trunking capabilities), but I'm also the VoIP provider, so it's a bit easier when you can configure both ends. I've seen situations a few times where IAX2 peers seem to disappear, and even restarting Asterisk doesn't bring them back - the only fix is to change bindport to 45691 for about 30 seconds then change it back. This points to the fact that it's a screwy NAT that causes the problem. On the whole though, I do personally prefer IAX2 - the only major problem being the lack of a load balancer (although there is one I haven't used for a while). A lot of people say that if you have problems with IAX2 then you should just move to SIP, but I really do like IAX2. I even worked on the PA1688/AR1688 based IAX2 phones for a while trying to get them to customers. At the end of the day though, we've found Polycom/Linksys/Cisco phones to be better for the end user, and then get them to register to Asterisk boxes which trunk calls back to our servers via IAX2. If you're doing more than one call (or moving around behind different NAT's) then it really does seem to be an improvement to me. The problem is, there aren't that many people that agree with me :) There are more developers who understand SIP (it's a text based signalling protocol after all, compared to the binary based IAX2). I could probably argue most points in a battle between IAX2 and SIP, but at the end of the day you're best to use what ever suits your needs the most. It's like programming languages - just because I can program in C, doesn't mean I write my scripts in it. You need to use the right tool for the job - be it bash, perl, php, C, Java, C#, IAX2 or SIP. :) So, the choice is really yours. I just personally really like IAX2. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] queue issue
One way to do this would be to use hints and an AGI to control dialing.
Let's say you have extensions 100 and 101 and each staffer also has a cell
(555-1212 and 555-1213). When you dial 100, you want to ring 100 and
555-1212 if both are available and the same with 101 and 555-1213. This
snippet would do it:
- exten = s,1XX,Macro(ring-group,${EXTEN})
- exten = s,1XX,playback(vm-goodbye)
- exten = s,1XX,hangup
- [macro-ring-group]
- exten = s,1,AGI(checkhints.agi,${ARG1})
- exten = s,n,gotoif($[${LINESTAT} = BUSY]?inuse)
- exten = s,n,Dial(SIP/${ARG1}DAHDI/g1/${CELLLINE},60)
- exten = s,n,hangup
- exten = s,n(inuse),playback(line-in-use)
- exten = s,n,hangup
The AGI checks the hint for 100 or 101 and assigns CELLLINE to call the
cell. If either is in use, LINESTAT is set to BUSY, otherwise set to AVAIL.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hales
Sent: Wednesday, September 02, 2009 2:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] queue issue
A situation where staff want a mobile and their SIP handset to share an
extension - but to make sure the mobile or SIP handset do not ring if
they are speaking on the other one...
PaulH
Lenz Emilitri wrote:
It depends on what you want to do to people who are queued; if you
want them to be queued, you create a queue with only one member, and
have agents log on and log off as necessary; if you don't want callers
to be queued, likely I would not use a queue but woul dial the agent
straight.
l.
PS. this is quite an unusual requirement, what is it for?
2009/9/1 Paul Hales pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au
I have a _very_ specific situation where I need queues to work in
a very
specific manner - I need the queue to only accept one call at a time,
even though several phones are attached to it.
My memory tells me that queues might have even worked this way in the
distant past (pre 1.0)...but I am willing to be mistaken.
Is this even remotely possible?
PaulH
--
Loway - home of QueueMetrics - http://queuemetrics.com
___
-- 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
___
-- 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 selective install
Just edit /etc/dahdi/modules and comment out all drivers. Normally you would comment out all except the card you have installed. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Valter Nogueira Sent: Tuesday, September 01, 2009 8:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DAHDI selective install Is there any way to not install all DAHDI drivers? All that I need is the dummy driver for timming purposes. Thanks, Valter ___ -- 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] 1.6.1 + TDM840 FSK MWI problem
- Barry Miller asterisk-us...@notanet.net wrote: Hi, Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? The ability to do line reversal MWI was added into the 1.6.2 branch. Looking through the 1.6.1 code base, I don't see anything other than fsk MWI (with and without Ring Pulse Alert Signalling.) In any case, this is set by defining mwisendtype in chan_dahdi. The default for this is fsk spills. It can be set to nofsk if you want to disable the fsk spills. The line reversal is set by specifying mwisendtype=lrev Regards, Doug Bailey ___ -- 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] SIP and other phones other then local network
On Wed, 2 Sep 2009, Matt Riddell wrote: On 2/09/09 9:10 PM, ABBAS SHAKEEL wrote: So howz about using IAX2 IAX2 is a touchy subject with some people. I personally use it as much as possible... Ditto. IAX just seems to work. I know many have had their issues with IAX, but the number of failure modes seems smaller than SIP. I have an IAXy that I use occasionally when I travel and it also just works regardless of where in the world I plug it in. Supposedly SIP gives better audio quality in some situations. I've seen situations a few times where IAX2 peers seem to disappear, and even restarting Asterisk doesn't bring them back - the only fix is to change bindport to 45691 for about 30 seconds then change it back. This points to the fact that it's a screwy NAT that causes the problem. I've never seen this, but I'm a 1.2 Luddite so maybe this is a more recent feature. The problem is, there aren't that many people that agree with me :) +1. I'd hate to see IAX deprecated in the mindset of the Asterisk users. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] internet connection lagged - * lagged ...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Gordon Henderson wrote: DNS. Run a caching DNS server on your Asterisk box, or a suitable device on your network. (eg. the DHCP server) The network gateway has already a dns cache. Inaddition, the ip of itsp were resolved properly. I also think this issue but has the hostname of the ITSP were determined ... - -- Antoine Patte -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkqekl0ACgkQBnIOcv+j7+yHAwCfZClaEHIDeTKK2HykDhK9rykA P0sAnRTRkYwu3ZJu2AGDh0JzQRAXVpP8 =JLIn -END PGP SIGNATURE- ___ -- 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] Very simple callback application needed
I have need of a very simple callback function - when any call is made to a special SIP DID, the call is not answered but Asterisk then calls a pre-determined number - no need for CallerID to capture the calling number. Does anyone have a simple script to do this? Chris -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] Very simple callback application needed
As I read this, it's not truly a callback; it's more of a notify; you call 555-1212 and want asterisk to call 555-1313? If this is actually the case, you would just do this in your dialplan: - exten = 5551212,1,dial(DAHDI/g1/5551313,60) This would effectively make asterisk do a new call to bridge A to B. If you wanted a non-bridged call, you could set up a call file and do this: - exten = 5551212,1,System(/bin/cp newcall.call /var/spool/asterisk/outgoing) - exten = 5551212,2,hangup Just my .02 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Mason (Lists) Sent: Wednesday, September 02, 2009 10:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Very simple callback application needed I have need of a very simple callback function - when any call is made to a special SIP DID, the call is not answered but Asterisk then calls a pre-determined number - no need for CallerID to capture the calling number. Does anyone have a simple script to do this? Chris -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_dahdi.so fails to load : Inappropriate ioctl for device
h...@cfht.hawaii.edu wrote: Aloha, I'm not sure why I'm getting this error, but I can't seem to get chan_dahdi to load. SIP IAX2 are working fine. Debian 4 w/ 2.6.28 kernel. Asterisk 1.6.1.5, dahdi-linux 2.2.0.2, dahdi-tools-2.2.0 CLI module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. [Sep 1 10:57:51] WARNING[31696]: pbx.c:4550 ast_register_application2: Already have an application 'DAHDISendKeypadFacility' [Sep 1 10:57:51] ERROR[31696]: chan_dahdi.c:8786 mkintf: Unable to get parameters: Inappropriate ioctl for device [Sep 1 10:57:51] ERROR[31696]: chan_dahdi.c:14170 build_channels: Unable to register channel '1' snip Note, I have compiled DAHDI 2.2.0.2 but it still shows 2.1.0.4 in the tool. Version bug? If it should say 2.2.0.2, then that could be my problem. But how do I correct that? # dahdi_cfg -vvv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.1.0.4 Echo Canceller(s): Configuration == I think you are correct and that this is your problem. If you have dahdi-tools 2.2.0 installed, but using and older version of dahdi-linux, you will get these errors since the format of some of the ioctls have changed. (related to https://issues.asterisk.org/view.php?id=14499) How did you install dahdi-linux? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- 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] followme Script
Hello, I am looking for a follow me script, where users can toggle follow me from their extensions and add follow me numbers from their extensions. Thanks -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- 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] Allowing multiple callers to join a public speaking session...?
Hi All, As is obvious by my joining the list, I'm interested in learning more about Asterisk. I have downloaded the PDF manual (for version 1.4) and am beginning to go through it. What I'm looking for in the short- term, however, is a more concise reference for common Asterisk configurations and setups. I currently have a non-profit client to which I am donating work. They are looking to allow callers to listen in to public speaking sessions. They currently have a single phone line with call waiting and are using an archaic one-person switch to then allow folks to call- chain via 3-way calling. What they want is basically a switchboard that allows multiple people (5 to 10) to call in at a time of their choosing and begin listening to the in-progress session. My first question would be: Is Asterisk the proper tool for this job (or is there something else you'd recommend)? A follow-up question would be: What kind of cost is involved in a small setup of this nature? Your input is much appreciated. Best, Michael___ -- 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] Allowing multiple callers to join a public speaking session...?
An Asterisk MeetMe conference sounds like the ideal sort of scenario for you, allowing people to join in or drop off during a session as they please. N. li...@mgreg.com wrote: Hi All, As is obvious by my joining the list, I'm interested in learning more about Asterisk. I have downloaded the PDF manual (for version 1.4) and am beginning to go through it. What I'm looking for in the short-term, however, is a more concise reference for common Asterisk configurations and setups. I currently have a non-profit client to which I am donating work. They are looking to allow callers to listen in to public speaking sessions. They currently have a single phone line with call waiting and are using an archaic one-person switch to then allow folks to call-chain via 3-way calling. What they want is basically a switchboard that allows multiple people (5 to 10) to call in at a time of their choosing and begin listening to the in-progress session. My first question would be: Is Asterisk the proper tool for this job (or is there something else you'd recommend)? A follow-up question would be: What kind of cost is involved in a small setup of this nature? Your input is much appreciated. Best, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing multiple callers to join a publicspeaking session...?
In my opinion, Asterisk would be an acceptable, if not proper tool for this task.If the sessions aren't live, you might be better off offering them as podcasts. But since you posted the question here, the simplest way to offer this would be to connect an asterisk installation to 5-10 SIP DID's and offer the sessions as a conference call. You could probably do this for $200-500 USD per month, possibly less with a hosted asterisk solution. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of li...@mgreg.com Sent: Wednesday, September 02, 2009 12:19 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Allowing multiple callers to join a publicspeaking session...? Hi All, As is obvious by my joining the list, I'm interested in learning more about Asterisk. I have downloaded the PDF manual (for version 1.4) and am beginning to go through it. What I'm looking for in the short-term, however, is a more concise reference for common Asterisk configurations and setups. I currently have a non-profit client to which I am donating work. They are looking to allow callers to listen in to public speaking sessions. They currently have a single phone line with call waiting and are using an archaic one-person switch to then allow folks to call-chain via 3-way calling. What they want is basically a switchboard that allows multiple people (5 to 10) to call in at a time of their choosing and begin listening to the in-progress session. My first question would be: Is Asterisk the proper tool for this job (or is there something else you'd recommend)? A follow-up question would be: What kind of cost is involved in a small setup of this nature? Your input is much appreciated. Best, Michael ___ -- 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] Allowing multiple callers to join a public speaking session...?
On Wed, 2 Sep 2009, li...@mgreg.com wrote: Hi All, As is obvious by my joining the list, I'm interested in learning more about Asterisk. I have downloaded the PDF manual (for version 1.4) and am beginning to go through it. What I'm looking for in the short-term, however, is a more concise reference for common Asterisk configurations and setups. I currently have a non-profit client to which I am donating work. They are looking to allow callers to listen in to public speaking sessions. They currently have a single phone line with call waiting and are using an archaic one-person switch to then allow folks to call-chain via 3-way calling. What they want is basically a switchboard that allows multiple people (5 to 10) to call in at a time of their choosing and begin listening to the in-progress session. My first question would be: Is Asterisk the proper tool for this job (or is there something else you'd recommend)? A follow-up question would be: What kind of cost is involved in a small setup of this nature? Your input is much appreciated. Best, Michael Hi Michael, Yes, I think you are on the right track. A Meetme conference is what you need, and perhaps a service to provide a DID number that would allow multiple people to call in to your conference at the same time (without purchasing POTS hardware, dealing with echo issues, etc.). Checkout www.ipcomms.net. I use them for a number of DID services. Their rates are decent and their support folks know asterisk. Cheers, j ___ -- 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] internet connection lagged - * lagged ...
On Wed, 2 Sep 2009, Antoine Patte wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Gordon Henderson wrote: DNS. Run a caching DNS server on your Asterisk box, or a suitable device on your network. (eg. the DHCP server) The network gateway has already a dns cache. Inaddition, the ip of itsp were resolved properly. I also think this issue but has the hostname of the ITSP were determined ... Try watching the outbound traffic from the box with tcpdump during one of these outages. You will likely see the DNS requests that are timing out and causing the problem. j - -- Antoine Patte -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkqekl0ACgkQBnIOcv+j7+yHAwCfZClaEHIDeTKK2HykDhK9rykA P0sAnRTRkYwu3ZJu2AGDh0JzQRAXVpP8 =JLIn -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing multiple callers to join a public speaking session...?
MeetMe agreed, but depending on how many people you expect to be listening, i think you can do this on a virtual server with minimal bandwidth, you can probably do this very very cheaply, or even find someone that will host it for free since it's non profit, unless of course you're talking about hundreds of people listening... but for ~20 i don't think it will cost too much at all, and i'm talking not much in the non profit sense, so i don't think you'd need hundreds... one of the servers i manage costs £40/month (1and1) which is currently handling over 100 calls with no complaints at all, so you should certainly be able to get something much cheaper than that, and i'm sure i've seen ISPs doing free services for non profit organisations in the past. Cheers Geraint 2009/9/2 li...@mgreg.com li...@mgreg.com Hi All, As is obvious by my joining the list, I'm interested in learning more about Asterisk. I have downloaded the PDF manual (for version 1.4) and am beginning to go through it. What I'm looking for in the short-term, however, is a more concise reference for common Asterisk configurations and setups. I currently have a non-profit client to which I am donating work. They are looking to allow callers to listen in to public speaking sessions. They currently have a single phone line with call waiting and are using an archaic one-person switch to then allow folks to call-chain via 3-way calling. What they want is basically a switchboard that allows multiple people (5 to 10) to call in at a time of their choosing and begin listening to the in-progress session. My first question would be: Is Asterisk the proper tool for this job (or is there something else you'd recommend)? A follow-up question would be: What kind of cost is involved in a small setup of this nature? Your input is much appreciated. Best, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing multiple callers to join a public speaking session...?
On Sep 2, 2009, at 1:33 PM, Jeff LaCoursiere wrote: Hi Michael, Yes, I think you are on the right track. A Meetme conference is what you need, and perhaps a service to provide a DID number that would allow multiple people to call in to your conference at the same time (without purchasing POTS hardware, dealing with echo issues, etc.). Checkout www.ipcomms.net. I use them for a number of DID services. Their rates are decent and their support folks know asterisk. Cheers, j Thanks for the posts thus far! In all honesty I'm looking for a complete in house solution. I don't mind spending up to $500-600 on equipment if necessary. I just want to know that when I'm done there are no residual costs, etc. Is Asterisk capable of this kind of setup/ management? As for labor, I'm willing to donate as much as is necessary. Thanks again, Michael ___ -- 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] weird caller ID addition when no caller id is received for incoming call
Hi, I am using a SPA 3000 as a PSTN gateway. Incoming PSTN calls are connected to Asterisk through SPA 3000 (it has a fxo port) via SIP. Everything is fine with this call scenario, but if the incoming PSTN call has no caller ID, then Asterisk receives the call with contact header and from header as sip:192.168.254.5 When it sends the same call to an internal extension Asterisk adds a caller ID 192168254254 to both from and contact fields. sip:192168254...@192.168.254.5 I checked all configuration files but couldn't find anything similar to this additional caller ID. Is it a default string ? How can I remove it ? If anyone is interested in analyzing the network trace I can send it to a given email address. Thanks ___ -- 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] More Echo
Greetings, I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium TE121B PCI express card with a VPMADT032 echo cancellation module, connected to an ATT 24 channel PRI. When I run dahdi show channel X on an active channel, I see this: Echo Cancellation: 128 taps unless TDM bridged, currently ON So I know the echo cancellation is working, however when I call a local analog land line, I get discernible echo. Here is my chan_dahdi.conf: [channels] ; configuration for T1 card as PRI language = en group = 1 echocancel = yes echotraining = yes signalling = pri_cpe switchtype = 4ess usecallerid = yes context = incoming channel = 1-23 Someone has to have had some experience with these hardware echo cancellers, any ideas? Should I adjust my rx and tx gains? Any advice would be very helpful. Thank you. -- Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com ___ -- 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] Asterisk MWI issue
- Original Message - From: Jeff LaCoursiere j...@jeff.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 02, 2009 1:41 AM Subject: Re: [asterisk-users] Asterisk MWI issue I'm only top posting to keep the flow going. Otherwise this would get messy. ilker - you should consider bottom posting to not raise the ire of others on the list. This may be a silly question, but do you have mailbox= filled in with the extension's number on the SIP extension page? If not asterisk will not generate the INFO. Hi Jeff, I have 9...@default in the mailbox field of the extension. But I already receive the voicemails correctly and I receive MWI when there is new voicemail. The problem is that I don't receive notification to clear MWI when voicemail is deleted. Regards, ilker ___ -- 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] New Languages: Call for contributions
(also posted today on http://blogs.digium.com/2009/09/02/new- languages/ ) Asterisk is being used all over the world, in dozens or even hundreds of nations, in a huge variety of linguistic settings. Until now, the official Asterisk distribution has come in only three language “flavors” – English, French, and Spanish. We are long overdue for getting more languages into the “main” Asterisk distribution, and over the past few weeks there has been quite a bit of work done getting licensing and practical concepts understood to the point where we are comfortable with expanding the number of available languages at the discretion of the community. There has been a document submitted for inclusion with Asterisk which outlines the protocol process, practical requirements, and license criteria for having a new language submitted to Asterisk as part of the official distribution. It should come as no surprise that we’re asking for all contributions to be in the Creative Commons v3.0 Share- Alike/Attribution licensing regime, as this is clearly the best (or only) method for distributing works such as audio recordings with an open-source package such as Asterisk. We’re also insisting that the talent that creates any language files be available for others to hire, so that there does not become a bottleneck with new prompts for others who wish to expand the range of recordings. Lastly among the important notes is that in the rare instances where we have new prompts as part of the “core” package requirements, anyone who has submitted a language package is under a non-binding community commitment to get the new prompts created in their language for addition. (This is a rare event, so hopefully is not overly burdensome to contributors.) This is truly a community participation request – there are far too many languages in the world for this to work without being almost entirely contributed by active Asterisk users and developers. The complexities of adding new languages is significant – there are intricacies in the “say.c” sections of code which determine how numbers and dates are pronounced. There are differences in the way voicemail prompts are created for playback. New languages may not be functionally complete if they require code to handle certain nuances of sentence structure, and the inclusion of new language audio files does not mean that they will be sensible in that particular language even if accepted. However, the first step is to get the language recordings in there, and then others can come in and correct the code once they have half the puzzle in their hands – that’s the spirit of open-source! There are at least 35 language or dialect versions already existing in third-party repositories (http://www.voip-info.org/wiki/view/Asterisk+multi-language ) and of those there are probably a quarter that have more than one voicing in male or female talent formats. I’d love to see the majority of those find their way into Asterisk as selectable language options. If you know the person that has created one of these language sets, please forward them the new language guideline link below! I’ll be trying to contact all of the language contributors, but often there are linguistic barriers or dead-ends for contact data. To read the requirements and to get started on your language contribution to Asterisk, see this document which will soon be part of the Asterisk standard distribution: Asterisk Language Submisson Criteria, part of issue #15771. JT References: https://issues.asterisk.org/file_download.php?file_id=23667type=bug https://issues.asterisk.org/view.php?id=15771 --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- 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] weird caller ID addition when no caller id isreceived for incoming call
You will need to put a fullname entry into users.conf. I'm guessing that Asterisk is generating this because it's not finding an entry there. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ilker Aktuna Sent: Wednesday, September 02, 2009 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] weird caller ID addition when no caller id isreceived for incoming call Hi, I am using a SPA 3000 as a PSTN gateway. Incoming PSTN calls are connected to Asterisk through SPA 3000 (it has a fxo port) via SIP. Everything is fine with this call scenario, but if the incoming PSTN call has no caller ID, then Asterisk receives the call with contact header and from header as sip:192.168.254.5 When it sends the same call to an internal extension Asterisk adds a caller ID 192168254254 to both from and contact fields. sip:192168254...@192.168.254.5 I checked all configuration files but couldn't find anything similar to this additional caller ID. Is it a default string ? How can I remove it ? If anyone is interested in analyzing the network trace I can send it to a given email address. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More Echo
Jason Baker wrote: So I know the echo cancellation is working, however when I call a local analog land line, I get discernible echo. echocancelwhenbridged=yes Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] chan_dahdi.so fails to load : Inappropriate ioctl for device
Shaun Ruffell wrote: I think you are correct and that this is your problem. If you have dahdi-tools 2.2.0 installed, but using and older version of dahdi-linux, you will get these errors since the format of some of the ioctls have changed. (related to https://issues.asterisk.org/view.php?id=14499) How did you install dahdi-linux Hi, Thanks for the reply. I had a hunch that was my problem. I just typed make make install in the dahdi-linux directory. But I did some more digging, and sure enough, modules.dep was not getting updated to take advantage of the most recent modules. Not sure if it's my system or the install script, but it kept linking to the old modules. I had to move the old module directory out of /lib/modules in order for it to link to the new ones. Thanks again for all your help! I appreciate it. Herb ___ -- 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] More Echo
Thank you, I will try that and get back to the mailing list with some info on whether it was successful or not. Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com Doug Lytle wrote: Jason Baker wrote: So I know the echo cancellation is working, however when I call a local analog land line, I get discernible echo. echocancelwhenbridged=yes Doug ___ -- 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] Allowing multiple callers to join a public speaking session...?
Asterisk is perfectly capable of it, your limiting factor will be bandwidth if you want to do it in-house... you'll obviously need enough bandwidth for all of your callers to be able to hear... unless of course you'll be using real phone lines, in which case you'll need to buy the appropriate hardware for your phone lines. Cheers Geraint 2009/9/2 li...@mgreg.com li...@mgreg.com On Sep 2, 2009, at 1:33 PM, Jeff LaCoursiere wrote: Hi Michael, Yes, I think you are on the right track. A Meetme conference is what you need, and perhaps a service to provide a DID number that would allow multiple people to call in to your conference at the same time (without purchasing POTS hardware, dealing with echo issues, etc.). Checkout www.ipcomms.net. I use them for a number of DID services. Their rates are decent and their support folks know asterisk. Cheers, j Thanks for the posts thus far! In all honesty I'm looking for a complete in house solution. I don't mind spending up to $500-600 on equipment if necessary. I just want to know that when I'm done there are no residual costs, etc. Is Asterisk capable of this kind of setup/ management? As for labor, I'm willing to donate as much as is necessary. Thanks again, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More Echo
Jason Baker wrote: language = en group = 1 echocancel = yes echotraining = yes signalling = pri_cpe switchtype = 4ess usecallerid = yes context = incoming channel = 1-23 Just noted that your system is out of Saginaw. The system below is out of Livonia, with an ATT PRI as well. Note the rx/txgain entries, it may be useful as well: switchtype=national context=pri signalling=pri_cpe echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=-1.0 txgain=-4.0 -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Allowing multiple callers to join a public speaking session...?
On another note... have you considered using a simple shoutcast setup instead? There will be a way (many ways probably) to hook this in with asterisk if necessary. You may have better results if it's simply listening the callers need to do, and depending on the audience that will be listening may work out easier and cheaper too. 2009/9/2 li...@mgreg.com li...@mgreg.com On Sep 2, 2009, at 1:33 PM, Jeff LaCoursiere wrote: Hi Michael, Yes, I think you are on the right track. A Meetme conference is what you need, and perhaps a service to provide a DID number that would allow multiple people to call in to your conference at the same time (without purchasing POTS hardware, dealing with echo issues, etc.). Checkout www.ipcomms.net. I use them for a number of DID services. Their rates are decent and their support folks know asterisk. Cheers, j Thanks for the posts thus far! In all honesty I'm looking for a complete in house solution. I don't mind spending up to $500-600 on equipment if necessary. I just want to know that when I'm done there are no residual costs, etc. Is Asterisk capable of this kind of setup/ management? As for labor, I'm willing to donate as much as is necessary. Thanks again, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More Echo
Interesting. I will give that a try. Also, any idea between the difference in switchtype between national and 4ess? All the documentation I read labeled 4ess as ATT, but I didn't try the national to see if it changed anything, like echo or signal quality. Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com Doug Lytle wrote: Jason Baker wrote: language = en group = 1 echocancel = yes echotraining = yes signalling = pri_cpe switchtype = 4ess usecallerid = yes context = incoming channel = 1-23 Just noted that your system is out of Saginaw. The system below is out of Livonia, with an ATT PRI as well. Note the rx/txgain entries, it may be useful as well: switchtype=national context=pri signalling=pri_cpe echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=-1.0 txgain=-4.0 ___ -- 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] weird caller ID addition when no caller idisreceived for incoming call
- Original Message - From: Danny Nicholas da...@debsinc.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, September 02, 2009 9:38 PM Subject: Re: [asterisk-users] weird caller ID addition when no caller idisreceived for incoming call You will need to put a fullname entry into users.conf. I'm guessing that Asterisk is generating this because it's not finding an entry there. For which object should I add this ? The incoming trunk ? or the inbound route ? in fact there is no full name for this user, because it's not a user. It's just a sip trunk that acts as a bridge between Asterisk and PSTN. ___ -- 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] More Echo
Jason Baker wrote: Interesting. I will give that a try. Also, any idea between the difference in switchtype between national and 4ess? All the documentation I read labeled 4ess as ATT, but I didn't try the national to see if it changed anything, like echo or signal quality. Differences? I've not read up on it, so I wouldn't be of use there. All our PRIs have been setup that way, since it's worked on all installs, I've never tried other types. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] weird caller ID addition when no calleridisreceived for incoming call
The trunk is a non-descript user, like a DAHDI line or SIP line. The entry isn't required to make the line function, just for caller-id handling. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ilker Aktuna Sent: Wednesday, September 02, 2009 2:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] weird caller ID addition when no calleridisreceived for incoming call - Original Message - From: Danny Nicholas da...@debsinc.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, September 02, 2009 9:38 PM Subject: Re: [asterisk-users] weird caller ID addition when no caller idisreceived for incoming call You will need to put a fullname entry into users.conf. I'm guessing that Asterisk is generating this because it's not finding an entry there. For which object should I add this ? The incoming trunk ? or the inbound route ? in fact there is no full name for this user, because it's not a user. It's just a sip trunk that acts as a bridge between Asterisk and PSTN. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing multiple callers to join a public speaking session...?
On Wed, 2009-09-02 at 14:03 -0400, li...@mgreg.com wrote: On Sep 2, 2009, at 1:33 PM, Jeff LaCoursiere wrote: Hi Michael, Yes, I think you are on the right track. A Meetme conference is what you need, and perhaps a service to provide a DID number that would allow multiple people to call in to your conference at the same time (without purchasing POTS hardware, dealing with echo issues, etc.). Checkout www.ipcomms.net. I use them for a number of DID services. Their rates are decent and their support folks know asterisk. Cheers, j Thanks for the posts thus far! In all honesty I'm looking for a complete in house solution. I don't mind spending up to $500-600 on equipment if necessary. I just want to know that when I'm done there are no residual costs, etc. Is Asterisk capable of this kind of setup/ management? As for labor, I'm willing to donate as much as is necessary. snip Absolutely. It doesn't sound like you need much firepower. You may even be able to carve off a virtual server for it. We don't do that in order to minimize latency but I'm sure lots of folks swear by such a setup. You will have the typical maintenance - updates, security patches, any client side changes. I would imagine your biggest challenge will be getting people into the system. If they are all internal (I was originally assuming they were not), they can all use soft phones and head sets. Since it is a monologue, you may even be able to dispense with the headsets. If folks are calling in from outside your network, it gets a little trickier. If they all have Internet connections, they can establish direct SIP connections to your PBX. If they are coming in from the PSTN, you will need phone lines. You could talk to a VoIP carrier and see if they can replace your PSTN access and then you would have the best of all worlds. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] [UOL - Manutenões Desktop] Controlling call duration ...
Hello there! The only available way to control call duration is using the RTCC patch (discussed here https://issues.asterisk.org/view.php?id=6335; and mainteined here http://ast.varna.net/;) ? The purpouse is to have a way to monitor (probably on a per-minute basis) and hangup costly calls (and/or multiple calls initiated by same SIP user). Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- 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] weird caller ID addition when nocalleridisreceived for incoming call
- Original Message - From: Danny Nicholas da...@debsinc.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, September 02, 2009 10:32 PM Subject: Re: [asterisk-users] weird caller ID addition when nocalleridisreceived for incoming call The trunk is a non-descript user, like a DAHDI line or SIP line. The entry isn't required to make the line function, just for caller-id handling. Ok; but I don't have any entry for the trunk in the users.conf file. Possibly it's in another file. How can I identify the correct place for that definition ? ___ -- 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] Does L(x:y:z) Dial option work on Asterisk version 1.4 ?
Hello there!
I'm testing Dial call limit option on Asterisk version 1.4.26, but
it's not working.
The issued command is: Dial(SIP/${EXTEN}|20|RtT|L(30:6:2)).
Am I missing something ?
Does it only work with Asterisk version 1.6.X ?
Thanks and best regards,
--
__At.,
_
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br mailto:@tqi.com.br
: www.tqi.com.br http://www.tqi.com.br
( + 55 (34)3291-1700
( + 55 (34)9971-2572
___
-- 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] weird caller ID addition whennocalleridisreceived for incoming call
Outside of my pay grade; maybe Jared Smith will read this and pipe in with an idea. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ilker Aktuna Sent: Wednesday, September 02, 2009 3:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] weird caller ID addition whennocalleridisreceived for incoming call - Original Message - From: Danny Nicholas da...@debsinc.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, September 02, 2009 10:32 PM Subject: Re: [asterisk-users] weird caller ID addition when nocalleridisreceived for incoming call The trunk is a non-descript user, like a DAHDI line or SIP line. The entry isn't required to make the line function, just for caller-id handling. Ok; but I don't have any entry for the trunk in the users.conf file. Possibly it's in another file. How can I identify the correct place for that definition ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does L(x:y:z) Dial option work on Asterisk version 1.4 ?
Mauro Sergio Ferreira Brasil wrote: Am I missing something ? Does it only work with Asterisk version 1.6.X ? core show application dial under my 1.4.21 install shows the option, so I would have to say that it's available in 1.4.x. As for it's proper usage, I don't know. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] problem with agi script not getting variable
I am learning agi scripting using php. I m using phpagi 2.x on asterisk 1.2.
I hve written a simple script that reads out the callerid using flite. My
problem is that I seems the script is not getting the callerID.
Bellow is the script
_
#!/usr/bin/php -q
?php
/**
* @package phpAGI_examples
* @version 2.0
*/
set_time_limit(30);
require('phpagi.php');
error_reporting(E_ALL);
$agi = new AGI();
$agi-answer();
$cid = $agi-parse_callerid();
$agi-exec(Flite,\Hello, {$cid['name']}.\);
$agi-exec(flite,\Goodbye\);
$agi-hangup();
?
___
and below is my agi debug output
-- Launched AGI Script /var/lib/asterisk/agi-bin/hints.php
AGI Tx agi_request: hints.php
AGI Tx agi_channel: SIP/1215-e5b8
AGI Tx agi_language: en
AGI Tx agi_type: SIP
AGI Tx agi_uniqueid: 1251926037.3
AGI Tx agi_callerid: 1215
AGI Tx agi_calleridname: device
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: 1220
AGI Tx agi_rdnis: unknown
AGI Tx agi_context: privileged
AGI Tx agi_extension: 1220
AGI Tx agi_priority: 2
AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode:
AGI Tx AGI Rx EXEC Flite Hello, .
-- AGI Script Executing Application: (Flite) Options: (Hello, .)
-- Playing '/tmp/flite_buf_VTgzTg' (language 'en')
As you can see, the callerID is not palyed out. What could I be doing wrong?
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer relationship management
(CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM
can help you achieve better customer satisfaction and sales
___
-- 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] Payload size of 30ms
Here's the story... Nortel system set to use g711 @ 30ms payload ... Asterisk box would need to communicate to that box @ 30 ms and another end point at 20 ms. I've seen discussions of setting this to a different size, but seems to be limited to the entire codec and not on a per peer basis. Anyone have luck with this? The Asterisk can be 1.4 or 1.6.x... I've a preference for 1.6.0.x but it's not set in stone :) Fred Posner f...@teamforrest.com ___ -- 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] DISA() fails to recognize dtmf where WaitExten() succeeds (DAHDI-PRI)
Is there any known reason that the DISA() routine should behave
differently than WaitExten() as far as recognizing DTMF tones? If
not, I suspect there's a bug here.
Try it yourself--two DID's on our PRI, numbers below let you test each routine:
It is my observation that some setups/phones DO and some DO NOT
express this variance.
--I could not show any variance on a sprint mobile phone
--I could not get T-Mobile to recognize ANY digits using DISA()
--I could only get digits 5,6,7,8,9,0 to be recognized with DISA() on
a Norstar PBX using PRI, and Analog trunks
(I tested four (count 'em) different installations, with only minor
variation)
I can make some setups/phones operate slightly closer to parity
(between the two routines) if I invoke the relaxdtmf parameter in
chan_dahdi.conf, but it does not completely eliminate the variance,
but irrespective of whether ANY specific configuration, network, or
device CAN or CAN'T send DTMF properly, DISA() and WaitExten() should
behave the same.
It has also been my observation that this variance does not express
itself as reliably with sip-terminated calls--perhaps because it's
dependent upon the specific media gateway that terminates the call,
and your ITSP's rate-deck may push you opportunistically to different
media gateways. Yesterday my sip-terminated Polycom IP-650 behaved
exactly like the Norstar systems described above, today it is
perfectly reliable (Telasip.com).
TRY IT FOR YOURSELF (dial-in numbers below):
For BOTH routines (one phone number for each), press a digit, Allison
will say it back to you:
As of this posting, RelaxDTMF is OFF. I will leave this configured
for at least 48 hours.
TEST -- WaitExten()
Call 312-445-5905 to run the [without-disa] routine below:
(lifted from dialplan)
press any DTMF key after hearing the beep
Allison will speak it back to you.
It should work 100% of the time
[without-disa]
exten = s,1,Answer(1000)
exten = s,n,background(beep)
exten = s,n,Waitexten()
exten = s,n,Hangup()
exten = _X,1,Saydigits(${EXTEN})
exten = _X,n,Goto(s,1)
TEST -- DISA()
Call 312-445-5906 to run this application:
(Again, lifted from dialplan)
press any DTMF key after hearing the DISA dialtone
If your dial tone is not interrupted by Allison speaking your digit,
DTMF was not recognized.
Depending on idiosyncratic details, this may work none, all, or only
for some digits.
[with-disa]
exten = s,1,Answer(1000)
exten = s,n,DISA(no-password,with-disa)
exten = _X,1,Saydigits(${EXTEN})
exten = _X,n,Goto(s,1)
Thoughts? YOUR test results?
Thanks
-Karl
Install details:
Asterisk 1.6.0.13, Dahdi 2.0.2, TE-212P HWEC, Centos 2.6.18-128.4.1.el5
___
-- 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] DISA() fails to recognize dtmf where WaitExten() succeeds (DAHDI-PRI)
Karl Fife wrote: TE-212P HWEC Grabbing at straws here, turn off EC and test again. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Allowing multiple callers to join a public speaking session...?
On Sep 2, 2009, at 3:35 PM, John A. Sullivan III wrote: Absolutely. It doesn't sound like you need much firepower. You may even be able to carve off a virtual server for it. We don't do that in order to minimize latency but I'm sure lots of folks swear by such a setup. You will have the typical maintenance - updates, security patches, any client side changes. I would imagine your biggest challenge will be getting people into the system. If they are all internal (I was originally assuming they were not), they can all use soft phones and head sets. Since it is a monologue, you may even be able to dispense with the headsets. If folks are calling in from outside your network, it gets a little trickier. If they all have Internet connections, they can establish direct SIP connections to your PBX. If they are coming in from the PSTN, you will need phone lines. You could talk to a VoIP carrier and see if they can replace your PSTN access and then you would have the best of all worlds. Hope this helps - John I'm sure I will encounter it in the book, but I'm looking to understand what actually needs to occur. Basically their scenario is a small auditorium that is already connected to the existing phone line so that ones may listen in over the 3-way to 3-way to 3-way (ad infinitum) chain. They have a *very* simple setup. There is no internet or internal network. That said, is there any way technologically to branch/bridge a normal phone line using Asterisk (or anything else), or must I have some other number/service coming in? Also, I believe there was a bit of confusion with an earlier post. Although they wish to *host* the entire setup in-house, they will have external callers. I'm certainly not opposed to the various proposed solutions, but given the nature of the project you can understand that I don't want to spend resources on items they don't absolutely need. Best, Michael___ -- 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] outbound calls not ringing still
i have posted this before but was unable to resolve it. i have some new info so i figured i would try again. the trace from bandwidth.com are below. they are telling me that the ip that is bold should be our ip not bandwidth.com. i have changed every setting that i can see and nothing fixes this. Where would i change this at? they cannot tell me. INVITE sip:+185993133...@216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP 216.82.224.202:5060;branch=z9hG4bK3691b08c;rport From:8592192438sip:8592192...@64.191.130.78;tag=as0707d433 To:sip:+185993133...@216.82.224.202 Contact:sip:8592192...@216.82.224.202 Call-ID: 0f3bdcc9171ef53148e7bab413aea...@64.191.130.78 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 02 Sep 2009 21:10:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 412 v=0 o=root 3831 3831 IN IP4 216.82.224.202 s=session c=IN IP4 216.82.224.202 t=0 0 m=audio 17050 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 12426 RTP/AVP 31 34 103 a=rtpmap:31 H261/9 a=rtpmap:34 H263/9 a=rtpmap:103 h263-1998/9 a=sendrecv _ Windows Live: Make it easier for your friends to see what you’re up to on Facebook. http://windowslive.com/Campaign/SocialNetworking?ocid=PID23285::T:WLMTAGL:ON:WL:en-US:SI_SB_facebook:082009___ -- 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] 1.6.1 + TDM840 FSK MWI problem
On Wed, Sep 02, 2009 at 09:44:05AM -0500, Doug Bailey wrote: - Barry Miller asterisk-us...@notanet.net wrote: Hi, Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? The ability to do line reversal MWI was added into the 1.6.2 branch. Looking through the 1.6.1 code base, I don't see anything other than fsk MWI (with and without Ring Pulse Alert Signalling.) In any case, this is set by defining mwisendtype in chan_dahdi. The default for this is fsk spills. It can be set to nofsk if you want to disable the fsk spills. The line reversal is set by specifying mwisendtype=lrev Regards, Doug Bailey Thanks, but that's not the problem. I _want_ FSK. A few ast_debug's in chan_dahdi tell me that after calling vmwi_generate(), it's taking the MWI_SEND_SPILL path through mwi_send_thread(), and happily sending about 9K bytes of spill, 160 bytes at a time. But my phones (and a butt-set) tell me that nothing is being received. I don't understand the DAHDI ioctls very well. Is it possible that the TDM840 is not in the correct state when the spill is transmitted? Thanks again, --Barry ___ -- 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] outbound calls not ringing still
On Wed, 2009-09-02 at 21:31 +, Ott Rose wrote: i have posted this before but was unable to resolve it. i have some new info so i figured i would try again. the trace from bandwidth.com are below. they are telling me that the ip that is bold should be our ip not bandwidth.com. i have changed every setting that i can see and nothing fixes this. Where would i change this at? they cannot tell me. INVITE sip:+185993133...@216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP 216.82.224.202:5060;branch=z9hG4bK3691b08c;rport From:8592192438sip:8592192...@64.191.130.78;tag=as0707d433 To:sip:+185993133...@216.82.224.202 Contact:sip:8592192...@216.82.224.202 Call-ID: 0f3bdcc9171ef53148e7bab413aea...@64.191.130.78 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 02 Sep 2009 21:10:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 412 v=0 o=root 3831 3831 IN IP4 216.82.224.202 s=session c=IN IP4 216.82.224.202 t=0 0 m=audio 17050 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 12426 RTP/AVP 31 34 103 a=rtpmap:31 H261/9 a=rtpmap:34 H263/9 a=rtpmap:103 h263-1998/9 a=sendrecv snip I know very little about how ringing works but are they providing any kind of status information to you? Do you need to furnish the ring if they are not? It seems to me I saw quite a few articles about providing ring tone, what causes it to fail, and how to work around it. I assume you've searched for those already. Just a few thoughts - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] how does wrapuptime work in queue.conf
Hi Barry,
I used a while loop and Playback() like you suggested. It does the
job. Thank you for the suggestion. I just thought there might be
some built-in function or parameters in queue.conf that can do the
trick.
Thanks.
Andy
On Thu, Aug 27, 2009 at 12:32 PM, Barry L. Klineblkl...@attglobal.net wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Andy Kuo wrote:
Hi Barry,
Thank you for the hint, but I forgot to mention that we have a few
advertisements, and we want the callers to listen to only one at a
time, and in a round robin or random order. Using Playback() doesn't
seem to serve that purpose. Is there any better way to achieve that?
Use the RAND function to generate or pick a filename.
exten = Set(advert=advert${RAND(1,10)})
exten = Playback(${advert})
That of course assumes that your advertisements are in files named
advert1.xxx through advert10.xxx (where xxx is wav,sln,etc)
Barry
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)
iD8DBQFKlt9cCFu3bIiwtTARAktAAJ4wFexOIhfN3aCjoIr11MKueZk4swCeK7Xt
RhKepfm4CplaaeCHwtbpzWI=
=6ojM
-END PGP SIGNATURE-
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how does wrapuptime work in queue.conf
Hi Lenz, That's what I was doing, putting the ad in MOH, but the callers only hear it when the agents are busy. When there are available agents, the callers just got connected to the agents without delay and hear no ads. The combination of a while loop and Playback() seem to be the only way to do it so far. Thanks. Andy On Wed, Sep 2, 2009 at 12:09 AM, Lenz Emilitrilenz.lo...@gmail.com wrote: Aht i would do is prepare a music on hold that has embedded the advertisements ( like one every 20 or 30 seconds) so that the caller hears more advertisements as the call progresses; and they are queued immediately, so no time is wasted. l. 2009/8/27 Andy Kuo aku...@gmail.com Hi Barry, Thank you for the hint, but I forgot to mention that we have a few advertisements, and we want the callers to listen to only one at a time, and in a round robin or random order. Using Playback() doesn't seem to serve that purpose. Is there any better way to achieve that? Thanks. Andy On Thu, Aug 27, 2009 at 11:56 AM, Barry L. Klineblkl...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Andy Kuo wrote: Hi list, I'd like to have the callers to listen to the advertisement (music on hold) before the agents answer them. So, I have wrapuptime=10 in queue.conf, but the call still goes straight to the agents without delay. Andy -- wrapuptime is the number of seconds that Asterisk waits between the time a agent hangs up with a caller and the next time that Asterisk sends a call to the newly-available agent. Wrap up time gives the agent a few moments to complete his last call and prepare for the next. What you need to do is use Playback() for your advertisement, then Queue() the call. Otherwise it acts just as you said, provided an agent is available. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKltbjCFu3bIiwtTARAjE0AKCGFEchqYoGWyaeHqlIH+iNyzBKygCgqibn X/gSnE7W7EHnwiUpRC1FLRs= =pdMh -END PGP SIGNATURE- ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing multiple callers to join a public speaking session...?
At 02:11 PM 9/2/2009, you wrote: That said, is there any way technologically to branch/bridge a normal phone line using Asterisk (or anything else), or must I have some other number/service coming in? Also, I believe there was a bit of confusion with an earlier post. Although they wish to *host* the entire setup in-house, they will have external callers. I'm certainly not opposed to the various proposed solutions, but given the nature of the project you can understand that I don't want to spend resources on items they don't absolutely need. I think if you have the upstream bandwidth that you could get a single number from a VOIP provider, I pay $1.50/month or so from Flowroute, and pay about a penny per minute for each active connection. I've found they tend to severely limit the number of concurrent connections, I think I'm only allowed 2 or 4, but I think that's mainly a protection from fraud thing and that a bit of negotiation could get that limit raised to meet your needs. That means for 20 callers you're looking at an internet connection with adequate bandwidth, a asterisk box, I'd guess most any reasonable leftover computer made in the last 4 years would work and then it's 25 cents a minute for however long the conferences last. If you do that, you can get in for essentially 0 hardware cost and it's easy to set up and test for a $30 up front payment for a few thousand minutes. Which makes lots of sense if the number of conference minutes is small, if that number gets high, you might see if a T1 is cheaper as you might get that with free incoming minutes. The T1 card will increase the hardware cost quite a bit. The part I don't know and maybe someone else can help is how to get your conference sound track into Asterisk. Ira ___ -- 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] Voipbuster not ringing, other SIP peers are ringing...
Does anybody else see the same behavior for VoipBuster connections? When I trace one of the other SIP peers, I see it sends this message: -- --- SIP read from 82.101.62.99:5060 --- SIP/2.0 180 Ringing Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl Contact: sip:82.101.62.99:5060 Content-Type: application/sdp CSeq: 103 INVITE From: ** sip:***...@sip.xs4all.nl;tag=as70e84199 Record-Route: sip:82.101.62.115;lr;r2=on;ftag=as70e84199,sip:82.101.63.5;lr;r2=on;ftag=as70e84199 Server: Cirpack/v4.41b (gw_sip) To: sip:0031*...@sip.xs4all.nl;tag=00-08168-044b6f36-245cd72c7 Via: SIP/2.0/UDP ***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92 Content-Length: 182 v=0 o=cp10 125193221174 125193221174 IN IP4 82.101.62.66 s=SIP Call c=IN IP4 194.109.8.2 t=0 0 m=audio 36984 RTP/AVP 8 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=ptime:20 a=sendrecv - --- (12 headers 10 lines) --- Found RTP audio format 8 Peer audio RTP is at port 194.109.8.2:36984 Found audio description format PCMA for ID 8 Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 194.109.8.2:36984 -- SIP/*-089ca9b8 is ringing -- SIP/*-089ca9b8 is making progress passing it to IAX2/2104-2287 Scheduling destruction of SIP dialog '740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 82.101.62.99:5060: CANCEL sip:0031**...@sip.xs4all.nl SIP/2.0 Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK07c2ed92;rport From: ** sip:**...@sip.xs4all.nl;tag=as70e84199 To: sip:0031**...@sip.xs4all.nl Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- However when I dial exactly the same from VoipBuster, I see this instead: -- --- SIP read from 77.72.169.129:5060 --- SIP/2.0 183 Session progress Via: SIP/2.0/UDP 195.164.89.135:5060;branch=z9hG4bK6d7efb43;rport From: * sip:**...@sip.voipbuster.com;tag=as1374705a To: sip:0031**...@sip.voipbuster.com;tag=120113ac4a54a269af9e2c Contact: sip:0031**...@77.72.169.129:5060 Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com CSeq: 103 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 162 v=0 o=* 1251932194 1251932194 IN IP4 194.221.62.33 s=SIP Call c=IN IP4 194.221.62.33 t=0 0 m=audio 8958 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 - --- (11 headers 8 lines) --- Found RTP audio format 0 Peer audio RTP is at port 194.221.62.33:8958 Found audio description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 194.221.62.33:8958 -- SIP/-089dc538 is making progress passing it to IAX2/2104-8077 == Connect attempt from '127.0.0.1' unable to authenticate Scheduling destruction of SIP dialog '1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 77.72.169.129:5060: CANCEL sip:0031**...@sip.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK6d7efb43;rport From: ** sip:***...@sip.voipbuster.com;tag=as1374705a To: sip:0031**...@sip.voipbuster.com Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- As you can see, there are different packets being sent, and in the 2nd case, there is no is ringing message, which is rather irritating... Any suggestions would be appreciated... TIA -- FP ___ -- 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] Help with call scenario
I am new to AGI. I have written my first php agi script that gets the extension dialed and says it back the caller using flite. I am stuck on how to pass the comand asterisk –rx “core show hints to asterisk and get the data back. This isn’t the recommended way, but it does work: Let’s say extension A is 100 and B is 101. Set up hints for 100 and 101. Then do a quick and dirty agi to parse “asterisk –rx “core show hints” “ for InUse. If any of the 4 lines of 100 are in use, hints will report it as inuse, so you can use that to report back to b (101) that 100 is busy. ___ -- 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] Voipbuster not ringing, other SIP peers are ringing...
Francesco Peeters wrote: Does anybody else see the same behavior for VoipBuster connections? When I trace one of the other SIP peers, I see it sends this message: -- --- SIP read from 82.101.62.99:5060 --- SIP/2.0 180 Ringing Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl Contact: sip:82.101.62.99:5060 Content-Type: application/sdp CSeq: 103 INVITE From: ** sip:***...@sip.xs4all.nl;tag=as70e84199 Record-Route: sip:82.101.62.115;lr;r2=on;ftag=as70e84199,sip:82.101.63.5;lr;r2=on;ftag=as70e84199 Server: Cirpack/v4.41b (gw_sip) To: sip:0031*...@sip.xs4all.nl;tag=00-08168-044b6f36-245cd72c7 Via: SIP/2.0/UDP ***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92 Content-Length: 182 v=0 o=cp10 125193221174 125193221174 IN IP4 82.101.62.66 s=SIP Call c=IN IP4 194.109.8.2 t=0 0 m=audio 36984 RTP/AVP 8 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=ptime:20 a=sendrecv - --- (12 headers 10 lines) --- Found RTP audio format 8 Peer audio RTP is at port 194.109.8.2:36984 Found audio description format PCMA for ID 8 Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 194.109.8.2:36984 -- SIP/*-089ca9b8 is ringing -- SIP/*-089ca9b8 is making progress passing it to IAX2/2104-2287 Scheduling destruction of SIP dialog '740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 82.101.62.99:5060: CANCEL sip:0031**...@sip.xs4all.nl SIP/2.0 Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK07c2ed92;rport From: ** sip:**...@sip.xs4all.nl;tag=as70e84199 To: sip:0031**...@sip.xs4all.nl Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- However when I dial exactly the same from VoipBuster, I see this instead: -- --- SIP read from 77.72.169.129:5060 --- SIP/2.0 183 Session progress Via: SIP/2.0/UDP 195.164.89.135:5060;branch=z9hG4bK6d7efb43;rport From: * sip:**...@sip.voipbuster.com;tag=as1374705a To: sip:0031**...@sip.voipbuster.com;tag=120113ac4a54a269af9e2c Contact: sip:0031**...@77.72.169.129:5060 Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com CSeq: 103 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 162 v=0 o=* 1251932194 1251932194 IN IP4 194.221.62.33 s=SIP Call c=IN IP4 194.221.62.33 t=0 0 m=audio 8958 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 - --- (11 headers 8 lines) --- Found RTP audio format 0 Peer audio RTP is at port 194.221.62.33:8958 Found audio description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 194.221.62.33:8958 -- SIP/-089dc538 is making progress passing it to IAX2/2104-8077 == Connect attempt from '127.0.0.1' unable to authenticate Scheduling destruction of SIP dialog '1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 77.72.169.129:5060: CANCEL sip:0031**...@sip.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK6d7efb43;rport From: ** sip:***...@sip.voipbuster.com;tag=as1374705a To: sip:0031**...@sip.voipbuster.com Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- As you can see, there are different packets being sent, and in the 2nd case, there is no is ringing message, which is rather irritating... Any suggestions would be appreciated... TIA BTW: I am talking about the ringtone the caller should hear... The other side is ringing, and calls are established just fine, but it is very irritating to hear nothing until the call either fails or gets picked up... -- FP ___ -- 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:
Re: [asterisk-users] queue issue
On 3/09/09 11:34 AM, Paul Hales wrote: Hmmm.any idea how I can use hints to monitor their mobile phones? Unless the call came in via Asterisk, you can't. Why not just have the desk phone accept one call (i.e. call/group/whatever limit) and then use app_followme? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] queue issue
Matt Riddell wrote: On 3/09/09 11:34 AM, Paul Hales wrote: Hmmm.any idea how I can use hints to monitor their mobile phones? Unless the call came in via Asterisk, you can't. The calls will - so it should be able (at the very least with the asterisk internal DB - which I don't fully trust due to reboots and the odd weird behaviour) Why not just have the desk phone accept one call (i.e. call/group/whatever limit) and then use app_followme? The issue is that both phones have to ring at the same time.And it's easy enough to stop the mobile from ringing if the SIP phone is in use, but the other way around is the challengeIt's doable, but I want to find the right solution. PaulH ___ -- 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] queue issue
They don't want to log in, and they want both to ring if they are free -
this is a very large site, so they need to be contactable at all times.
PaulH
Lenz Emilitri wrote:
I would have them log on with the mobile when they need it, and log
off when they don't. When the mobile is not present you would simply
dial the local extension.
You could have something like:
local/1...@agents
that does something like:
if ( DBSET(has_mobile) ) {
dial( Zap/g0/MYMOBILENUM )
} else {
dial( SIP/123 )
}
and have anothe extension set/reset the has_mobile property in the AstDB.
You could then call Local/1...@gaents directkly or make it a member of
the queue (with known issues on some version of *) :-)
l.
2009/9/2 Paul Hales pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au
A situation where staff want a mobile and their SIP handset to
share an
extension - but to make sure the mobile or SIP handset do not ring if
they are speaking on the other one...
PaulH
Lenz Emilitri wrote:
It depends on what you want to do to people who are queued; if you
want them to be queued, you create a queue with only one member, and
have agents log on and log off as necessary; if you don't want
callers
to be queued, likely I would not use a queue but woul dial the agent
straight.
l.
PS. this is quite an unusual requirement, what is it for?
2009/9/1 Paul Hales pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au mailto:pdha...@optusnet.com.au
I have a _very_ specific situation where I need queues to
work in
a very
specific manner - I need the queue to only accept one call
at a time,
even though several phones are attached to it.
My memory tells me that queues might have even worked this
way in the
distant past (pre 1.0)...but I am willing to be mistaken.
Is this even remotely possible?
PaulH
--
Loway - home of QueueMetrics - http://queuemetrics.com
___
-- 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
--
Loway - home of QueueMetrics - http://queuemetrics.com
___
-- 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] queue issue
Hmmm.any idea how I can use hints to monitor their mobile phones?
PaulH
Danny Nicholas wrote:
One way to do this would be to use hints and an AGI to control dialing.
Let's say you have extensions 100 and 101 and each staffer also has a cell
(555-1212 and 555-1213). When you dial 100, you want to ring 100 and
555-1212 if both are available and the same with 101 and 555-1213. This
snippet would do it:
- exten = s,1XX,Macro(ring-group,${EXTEN})
- exten = s,1XX,playback(vm-goodbye)
- exten = s,1XX,hangup
- [macro-ring-group]
- exten = s,1,AGI(checkhints.agi,${ARG1})
- exten = s,n,gotoif($[${LINESTAT} = BUSY]?inuse)
- exten = s,n,Dial(SIP/${ARG1}DAHDI/g1/${CELLLINE},60)
- exten = s,n,hangup
- exten = s,n(inuse),playback(line-in-use)
- exten = s,n,hangup
The AGI checks the hint for 100 or 101 and assigns CELLLINE to call the
cell. If either is in use, LINESTAT is set to BUSY, otherwise set to AVAIL.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hales
Sent: Wednesday, September 02, 2009 2:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] queue issue
A situation where staff want a mobile and their SIP handset to share an
extension - but to make sure the mobile or SIP handset do not ring if
they are speaking on the other one...
PaulH
Lenz Emilitri wrote:
It depends on what you want to do to people who are queued; if you
want them to be queued, you create a queue with only one member, and
have agents log on and log off as necessary; if you don't want callers
to be queued, likely I would not use a queue but woul dial the agent
straight.
l.
PS. this is quite an unusual requirement, what is it for?
2009/9/1 Paul Hales pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au
I have a _very_ specific situation where I need queues to work in
a very
specific manner - I need the queue to only accept one call at a time,
even though several phones are attached to it.
My memory tells me that queues might have even worked this way in the
distant past (pre 1.0)...but I am willing to be mistaken.
Is this even remotely possible?
PaulH
--
Loway - home of QueueMetrics - http://queuemetrics.com
___
-- 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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue issue
On 3/09/09 12:21 PM, Paul Hales wrote: Matt Riddell wrote: On 3/09/09 11:34 AM, Paul Hales wrote: Hmmm.any idea how I can use hints to monitor their mobile phones? Unless the call came in via Asterisk, you can't. The calls will - so it should be able (at the very least with the asterisk internal DB - which I don't fully trust due to reboots and the odd weird behaviour) Then it's easy :) Use func_devstate - you can set custom device states for things - and btw the Asterisk DB is pretty stable - we're using it in pretty large call centres without (touch wood) ever having any problems. A lot more than I can say for MySQL :) Oh, by the way, func_devstate was only added to 1.4 a few months back - although if you're stuck with a particular version, the backport always applied cleanly for me. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Inquiry:Problem with Call Parking
In any event, the real problem is probably that you forgot to 'include = parkedcalls' in your dialplan. Steve On 9/2/09, Lyle Giese l...@lcrcomputer.net wrote: And now that the whole world of Asterisk has your sip user ids and passwords, you should change all of the passwords that are in that file and yes, change the passwords in all your phones. Lyle Giese LCR Computer Services, Inc. hadi motamedi wrote: Thank you for your reply . Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Regards H.Motamedi On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney) john@compuware.com mailto:john@compuware.com wrote: Just a quick guess - is it because you did not program your Polycom digit plan properly in sip.cfg? From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Tuesday, 1 September 2009 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inquiry:Problem with Call Parking Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my features.conf . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let me know what extra config needs to be done for putting it into operation ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- Sent from my mobile device ___ -- 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] SIP and other phones other then local network
Thanks MATT and steve :) Is there some thing where i dont configuration at nat level ... So that no change on Internet router etc On Wed, Sep 2, 2009 at 8:13 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 2 Sep 2009, Matt Riddell wrote: On 2/09/09 9:10 PM, ABBAS SHAKEEL wrote: So howz about using IAX2 IAX2 is a touchy subject with some people. I personally use it as much as possible... Ditto. IAX just seems to work. I know many have had their issues with IAX, but the number of failure modes seems smaller than SIP. I have an IAXy that I use occasionally when I travel and it also just works regardless of where in the world I plug it in. Supposedly SIP gives better audio quality in some situations. I've seen situations a few times where IAX2 peers seem to disappear, and even restarting Asterisk doesn't bring them back - the only fix is to change bindport to 45691 for about 30 seconds then change it back. This points to the fact that it's a screwy NAT that causes the problem. I've never seen this, but I'm a 1.2 Luddite so maybe this is a more recent feature. The problem is, there aren't that many people that agree with me :) +1. I'd hate to see IAX deprecated in the mindset of the Asterisk users. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- Best Regards Shakeel Abbas ___ -- 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] SIP and other phones other then local network
On 3/09/09 4:36 PM, ABBAS SHAKEEL wrote: Thanks MATT and steve :) :) No problems. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] DISA() fails to recognize dtmf where WaitExten() succeeds (DAHDI-PRI)
- Original Message - From: Doug Lytle supp...@drdos.info To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 02, 2009 3:58 PM Subject: Re: [asterisk-users] DISA() fails to recognize dtmf where WaitExten() succeeds (DAHDI-PRI) Karl Fife wrote: TE-212P HWEC Grabbing at straws here, turn off EC and test again. Doug You were right. Turning off echocan makes DTMF detection via WaitExten() vs DISA() behave the same way. Any theories as to why one routine would behave differently than the other with Echo Cancellation enabled? Should it be considered a defect if EC (often necessary) breaks inband DTMF detection on TDM? -Karl ___ -- 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
