Re: [asterisk-users] how many UDP ports is required for 1 call
Does anyone know the correct information of my question. All are move round and round . On Tue, Feb 21, 2012 at 7:28 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/21/2012 07:51 AM, Alex Balashov wrote: As many ports as required by the nature of the call, i.e. the protocol(s) used for the bearer. For an IAX2 call, the answer is 'zero' for all of those call types (at least the ones that are supported in IAX2, not all of them are). For protocols that use RTP for media transport, two ports are required for each media stream (one for RTP, one for RTCP). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
On 02/22/2012 06:26 AM, virendra bhati wrote: Does anyone know the correct information of my question. All are move round and round . What does that mean? I answered your question with the correct and complete information. On Tue, Feb 21, 2012 at 7:28 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: On 02/21/2012 07:51 AM, Alex Balashov wrote: As many ports as required by the nature of the call, i.e. the protocol(s) used for the bearer. For an IAX2 call, the answer is 'zero' for all of those call types (at least the ones that are supported in IAX2, not all of them are). For protocols that use RTP for media transport, two ports are required for each media stream (one for RTP, one for RTCP). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.x app_meetme.so
Hello, I can't seem to use MeetMe app in asterisk versions beyond 1.8. Its source file app_meetme.c is present in the apps dir. Also, I can find app_meetme change-logs on the asterisk website. However, the dialplan doesn't have this cmd. I have checked menuselect but it says it has been replaced by app_confbridge. Also, If that *is* the case, does ConfBridge (the newer version of meetme) offer the same options? How do I use them? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
Hi Kevin, http://www.freepbx.org/support/documentation/howtos/howto-setup-a-remote-sip-extension this says 4 active ports for one call Regards, Zohair Raza On Wed, Feb 22, 2012 at 4:38 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/22/2012 06:26 AM, virendra bhati wrote: Does anyone know the correct information of my question. All are move round and round . What does that mean? I answered your question with the correct and complete information. On Tue, Feb 21, 2012 at 7:28 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: On 02/21/2012 07:51 AM, Alex Balashov wrote: As many ports as required by the nature of the call, i.e. the protocol(s) used for the bearer. For an IAX2 call, the answer is 'zero' for all of those call types (at least the ones that are supported in IAX2, not all of them are). For protocols that use RTP for media transport, two ports are required for each media stream (one for RTP, one for RTCP). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Postgresql in Asterisk
Hello, I install asterisk an postgresql 9.1 in gentoo, I already did the configuration in both asterisk and postgresql, in fact If I make a call and asterisk log it to CDR table, my question is: how can I make a function like the ones in func_odbc.conf for postgresql, if I am using res_pgsql.conf instead of res_odbc.conf? I also configure odbc and it connects with echo select 1 | isql -v asterisk-connector with out problems, but when I try an odbc function or restart asterisk it logs: Error SQLConnect=-1 errno=101 [unixODBC]Could not connect to the server; Could not connect to remote socket. and the command CLI odbc show ODBC DSN Settings - Name: asterisk DSN:asterisk-connector Last connection attempt: 2012-02-22 06:45:36 I will appreciate any help. Regards, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
On 02/22/2012 06:41 AM, Zohair Raza wrote: Hi Kevin, http://www.freepbx.org/support/documentation/howtos/howto-setup-a-remote-sip-extension this says 4 active ports for one call It certainly does. Whether it is accurate or not depends on the definition of 'call', as has already been pointed out in this thread. I could also, definitively and accurately, state that the number of UDP ports required for a 'call' is 5 (five). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
On 02/22/2012 07:26 AM, virendra bhati wrote: Does anyone know the correct information of my question. All are move round and round . Well, you know Kevin. Whenever I ask him a question, he just moves round and round... -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Postgresql in Asterisk
On Wed, 2012-02-22 at 06:48 -0600, Sergio Basurto wrote: Hello, I install asterisk an postgresql 9.1 in gentoo, I already did the configuration in both asterisk and postgresql, in fact If I make a call and asterisk log it to CDR table, my question is: I make a typo mistake I mean If I make a call asterisk already log it into CDR table. how can I make a function like the ones in func_odbc.conf for postgresql, if I am using res_pgsql.conf instead of res_odbc.conf? I also configure odbc and it connects with echo select 1 | isql -v asterisk-connector with out problems, but when I try an odbc function or restart asterisk it logs: Error SQLConnect=-1 errno=101 [unixODBC]Could not connect to the server; Could not connect to remote socket. and the command CLI odbc show ODBC DSN Settings - Name: asterisk DSN:asterisk-connector Last connection attempt: 2012-02-22 06:45:36 I will appreciate any help. Regards, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sergio Basurto sbasu...@soft-gator.com Soft Gator S.A de C.V. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
all your base are move round and round fun idea: house music with lyrics composed of nothing but crazy quotes from asterisk-users! yeah, it's time for coffee #1 John Knight Classic City Telco LLC Email: j...@classiccitytelco.com | Main: (706) 995-0200 Direct: (706) 995-0201 | Mobile: (706) 255-9203 On 2/22/2012 7:52 AM, Alex Balashov wrote: On 02/22/2012 07:26 AM, virendra bhati wrote: Does anyone know the correct information of my question. All are move round and round . Well, you know Kevin. Whenever I ask him a question, he just moves round and round... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
On 02/22/2012 07:26 AM, virendra bhati wrote: Does anyone know the correct information of my question. All are move round and round . Why don't you make a call, and watch the network traffic with tcpdump, wireshark or similar? Wireshark in particular has an analysis module that will show you a summary of each network conversation by source and destination IP and port and protocol, as well as a handful of analyzers developed specifically for IP telephony. You can then count for yourself how many ports are used for your particular definition and implementation of call, as well as gain a deeper understanding of what happens on the wire. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
Hi Kevin, I appreciate, that you replyed first and even fast. But you asked for nature of call too. So my question was added with your question . Voice call *How many port of UDP or RTP ?* Video call *How many port of UDP or RTP ?* Fax call*How many port of UDP or RTP ?* T.140 text call* How many port of UDP or RTP ?* As per the reply Voice call is come to 4 ports but rest is not clear. Voice call *4* Video call *??* Fax call*??* T.140 text call* ??* *Will these port of UDP, RPT or Both ?* On Wed, Feb 22, 2012 at 6:08 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/22/2012 06:26 AM, virendra bhati wrote: Does anyone know the correct information of my question. All are move round and round . What does that mean? I answered your question with the correct and complete information. On Tue, Feb 21, 2012 at 7:28 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: On 02/21/2012 07:51 AM, Alex Balashov wrote: As many ports as required by the nature of the call, i.e. the protocol(s) used for the bearer. For an IAX2 call, the answer is 'zero' for all of those call types (at least the ones that are supported in IAX2, not all of them are). For protocols that use RTP for media transport, two ports are required for each media stream (one for RTP, one for RTCP). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.x app_meetme.so
[Digital^Dude] ® wrote: I can't seem to use MeetMe app in asterisk versions beyond 1.8. Its source file app_meetme.c is present in the apps dir It's still available, but marked as depreciated. So, do a make menuselect, Applications, scroll all the way to the bottom and check meetme. 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
On 02/22/2012 07:01 AM, virendra bhati wrote: Hi Kevin, I appreciate, that you replyed first and even fast. But you asked for nature of call too. So my question was added with your question . Voice call *How many port of UDP or RTP ?* Video call *How many port of UDP or RTP ?* Fax call *How many port of UDP or RTP ?* T.140 text call* How many port of UDP or RTP ?* As per the reply Voice call is come to 4 ports but rest is not clear. Voice call *4* Video call *??* Fax call *??* T.140 text call* ??* *Will these port of UDP, RPT or Both ?* It is clear that you are lacking some basic understanding of how SIP and RTP operate; I would suggest that you take Phil Frost's advice and use Wireshark to watch what actually happens when you place calls on your system. Just some points: UDP and RTP are not exclusive; RTP operates over UDP. Voice calls do not require 4 UDP ports (in most cases). FAX calls rarely use RTP. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.x app_meetme.so
I did, and I mentioned it in my earlier email too. Screenshot attached. On Wed, Feb 22, 2012 at 6:03 PM, Doug Lytle supp...@drdos.info wrote: [Digital^Dude] ® wrote: I can't seem to use MeetMe app in asterisk versions beyond 1.8. Its source file app_meetme.c is present in the apps dir It's still available, but marked as depreciated. So, do a make menuselect, Applications, scroll all the way to the bottom and check meetme. 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users attachment: meetme.PNG-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
On 02/22/2012 08:01 AM, virendra bhati wrote: *Will these port of UDP, RPT [assume you mean RTP] or Both ?* It's evident from your response that you do not have a solid understanding of networking fundamentals. The full answer to your question will quickly go out of scope of this list and become an introduction to IP fundamentals. So, I suggest you start by reading these: http://en.wikipedia.org/wiki/OSI_model http://en.wikipedia.org/wiki/Internet_Protocol http://en.wikipedia.org/wiki/User_Datagram_Protocol http://en.wikipedia.org/wiki/Real-time_Transport_Protocol -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.x app_meetme.so
[Digital^Dude] ® wrote: I did, and I mentioned it in my earlier email too. You mentioned that the meetme source was there, I was guessing that the option to compile wasn't checked so the binary wasn't available. I just ran into this myself yesterday when converting a 1.4x box (Still in progress) to a 10.2.0 RC2 and once checked and re-compiled, meetme was available. 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
thanks for suggesting the link. Yes i don't have networking, and good SIP communication knowledge. On Wed, Feb 22, 2012 at 6:41 PM, Phil Frost p...@macprofessionals.comwrote: On 02/22/2012 08:01 AM, virendra bhati wrote: *Will these port of UDP, RPT [assume you mean RTP] or Both ?* It's evident from your response that you do not have a solid understanding of networking fundamentals. The full answer to your question will quickly go out of scope of this list and become an introduction to IP fundamentals. So, I suggest you start by reading these: http://en.wikipedia.org/wiki/**OSI_modelhttp://en.wikipedia.org/wiki/OSI_model http://en.wikipedia.org/wiki/**Internet_Protocolhttp://en.wikipedia.org/wiki/Internet_Protocol http://en.wikipedia.org/wiki/**User_Datagram_Protocolhttp://en.wikipedia.org/wiki/User_Datagram_Protocol http://en.wikipedia.org/wiki/**Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/Real-time_Transport_Protocol -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.x app_meetme.so
- Original Message - From: Doug Lytle supp...@drdos.info To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 22, 2012 7:22:20 AM Subject: Re: [asterisk-users] Asterisk 1.8.x app_meetme.so You mentioned that the meetme source was there, I was guessing that the option to compile wasn't checked so the binary wasn't available. I just ran into this myself yesterday when converting a 1.4x box (Still in progress) to a 10.2.0 RC2 and once checked and re-compiled, meetme was available. Doug Just a few points of clarification: 1. MeetMe is still the preferred conferencing application in Asterisk 1.8. In Asterisk 10, the preferred conferencing application is ConfBridge. Even still, in Asterisk, 10, you can compile and install MeetMe using menuselect. 2. In the screenshot you attached, you cannot choose to compile MeetMe as one of its dependencies is not available, in this case, DAHDI. Note that DAHDI being a dependency for MeetMe was one of the reasons Asterisk 10 moved to using ConfBridge as the default conferencing application. Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How does format_mp3 work?
Hi I was using the Playback application to play an MP3 file after compiling and installing asterisk 1.8.7.0 with format_mp3 and it seems to me that asterisk is transcoding the file to an slin on the fly rather than playing the mp3 itself. Is this what it does? Also, does this mean I might as well change the format of MP3s to WAV seeing as I'm used to doing that anyway? Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.x app_meetme.so
ConfBridge is not much flexible as MeetMe. On Wed, Feb 22, 2012 at 7:19 PM, Matthew Jordan mjor...@digium.com wrote: - Original Message - From: Doug Lytle supp...@drdos.info To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 22, 2012 7:22:20 AM Subject: Re: [asterisk-users] Asterisk 1.8.x app_meetme.so You mentioned that the meetme source was there, I was guessing that the option to compile wasn't checked so the binary wasn't available. I just ran into this myself yesterday when converting a 1.4x box (Still in progress) to a 10.2.0 RC2 and once checked and re-compiled, meetme was available. Doug Just a few points of clarification: 1. MeetMe is still the preferred conferencing application in Asterisk 1.8. In Asterisk 10, the preferred conferencing application is ConfBridge. Even still, in Asterisk, 10, you can compile and install MeetMe using menuselect. 2. In the screenshot you attached, you cannot choose to compile MeetMe as one of its dependencies is not available, in this case, DAHDI. Note that DAHDI being a dependency for MeetMe was one of the reasons Asterisk 10 moved to using ConfBridge as the default conferencing application. Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI: blocking script until playback complete
Greetings list, I've done AGI scripting before, but in the past I've always wanted control to be returned to the dialplan as soon as possible. However, today I have a scenario where I want the script to remain running during the playback of a file so that I can read DTMF at the end of playback. However, doing this: GET DATA en_welcome 5000 6 Results (correctly) in the following in the asterisk console: -- SIP/a.b.c.d-dc027b50 Playing 'en_welcome' (language 'en') But the AGI continues to run on after this point, not waiting for either the sound file to be played, nor for the expected 6 DTMF digits. Adding a simple 10 second sleep/wait to the AGI allows the sound file to be successfully played back. I'm sure I must be missing something very obvious, buy my google-fu is failing me this afternoon. Suggestions gratefully received :-) Thanks in advance. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: blocking script until playback complete
Try passing escape character GET DATA $filename $timeout $max_digits $escape_character Regards, Zohair Raza On Wed, Feb 22, 2012 at 6:40 PM, Chris Bagnall aster...@lists.minotaur.ccwrote: Greetings list, I've done AGI scripting before, but in the past I've always wanted control to be returned to the dialplan as soon as possible. However, today I have a scenario where I want the script to remain running during the playback of a file so that I can read DTMF at the end of playback. However, doing this: GET DATA en_welcome 5000 6 Results (correctly) in the following in the asterisk console: -- SIP/a.b.c.d-dc027b50 Playing 'en_welcome' (language 'en') But the AGI continues to run on after this point, not waiting for either the sound file to be played, nor for the expected 6 DTMF digits. Adding a simple 10 second sleep/wait to the AGI allows the sound file to be successfully played back. I'm sure I must be missing something very obvious, buy my google-fu is failing me this afternoon. Suggestions gratefully received :-) Thanks in advance. Kind regards, Chris -- This email is made from 100% recycled electrons -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does format_mp3 work?
In article 1329920656.2027.37.camel@localhost.localdomain, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I was using the Playback application to play an MP3 file after compiling and installing asterisk 1.8.7.0 with format_mp3 and it seems to me that asterisk is transcoding the file to an slin on the fly rather than playing the mp3 itself. Is this what it does? Of course. You are not going to send mp3-encoded audio over a phone line, are you? That would require an mp3 decoder in every phone. In fact, any mp3 player is transcoding to linear on the fly, since that is part of converting it to audio for the speaker or earphones! What did you think was the difference between trascoding to slin and playing the mp3? Also, does this mean I might as well change the format of MP3s to WAV seeing as I'm used to doing that anyway? If you have the storage space available for the larger WAV files, then yes, this would reduce the CPU consumption of playback. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: blocking script until playback complete
You don't state the Asterisk version you are running, but personal experience tells me you'd better invest in some Rogaine if you're depending on the built-in stuff from AGI for DTMF input. I have personally wasted weeks trying it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zohair Raza Sent: Wednesday, February 22, 2012 8:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AGI: blocking script until playback complete Try passing escape character GET DATA $filename $timeout $max_digits $escape_character Regards, Zohair Raza On Wed, Feb 22, 2012 at 6:40 PM, Chris Bagnall aster...@lists.minotaur.cc wrote: Greetings list, I've done AGI scripting before, but in the past I've always wanted control to be returned to the dialplan as soon as possible. However, today I have a scenario where I want the script to remain running during the playback of a file so that I can read DTMF at the end of playback. However, doing this: GET DATA en_welcome 5000 6 Results (correctly) in the following in the asterisk console: -- SIP/a.b.c.d-dc027b50 Playing 'en_welcome' (language 'en') But the AGI continues to run on after this point, not waiting for either the sound file to be played, nor for the expected 6 DTMF digits. Adding a simple 10 second sleep/wait to the AGI allows the sound file to be successfully played back. I'm sure I must be missing something very obvious, buy my google-fu is failing me this afternoon. Suggestions gratefully received :-) Thanks in advance. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does format_mp3 work?
On Wed, 2012-02-22 at 14:53 +, Tony Mountifield wrote: In article 1329920656.2027.37.camel@localhost.localdomain, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I was using the Playback application to play an MP3 file after compiling and installing asterisk 1.8.7.0 with format_mp3 and it seems to me that asterisk is transcoding the file to an slin on the fly rather than playing the mp3 itself. Is this what it does? Of course. You are not going to send mp3-encoded audio over a phone line, are you? That would require an mp3 decoder in every phone. In fact, any mp3 player is transcoding to linear on the fly, since that is part of converting it to audio for the speaker or earphones! What did you think was the difference between trascoding to slin and playing the mp3? Hmm, when you put it like that, makes me wonder what I was thinking :) Also, does this mean I might as well change the format of MP3s to WAV seeing as I'm used to doing that anyway? If you have the storage space available for the larger WAV files, then yes, this would reduce the CPU consumption of playback. Cheers Tony Thanks for the response Tony -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: blocking script until playback complete
On 22/2/12 2:55 pm, Danny Nicholas wrote: You don't state the Asterisk version you are running, but personal experience tells me you'd better invest in some Rogaine if you're depending on the built-in stuff from AGI for DTMF input. I have personally wasted weeks trying it. Sorry, should have said, latest 1.4 release. Care to elaborate a little on the issues you found when you tried it? Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: blocking script until playback complete
On 22/2/12 2:50 pm, Zohair Raza wrote: Try passing escape character GET DATA $filename $timeout $max_digits $escape_character Not sure I follow - according to the docs, there is no parameter $escape_character The problem seems to be that GET DATA returns control to the script before the audio file has even played, let alone any DTMF tones have been entered. I would have expected script execution to be blocked until the result from GET DATA was available. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: blocking script until playback complete
Are you reading STDIN to initialize your AGI? If not Asterisk may ignore your AGI commands. I recommend using one of the many AGI libs out there. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: Wednesday, February 22, 2012 10:20 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AGI: blocking script until playback complete On 22/2/12 2:50 pm, Zohair Raza wrote: Try passing escape character GET DATA $filename $timeout $max_digits $escape_character Not sure I follow - according to the docs, there is no parameter $escape_character The problem seems to be that GET DATA returns control to the script before the audio file has even played, let alone any DTMF tones have been entered. I would have expected script execution to be blocked until the result from GET DATA was available. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.x app_meetme.so
Doug, I can find the following in asterisk 10 changelogs: The following error will consistently occur when trying to dial into a MeetMe conference when the server does not have DAHDI hardware installed: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) While chan_dahdi is loaded correctly during compilation and install of Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf configuration file in /etc/asterisk is not created by FreePBX if hardware does not exist, causing MeetMe to be unable to open a DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo channel when there is no chan_dahdi.conf file to load. (closes issue ASTERISK-17398) Reported by: Preston Edwards This would mean that meetme should not have dahdi as a compilation dependency. source: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10-current On Wed, Feb 22, 2012 at 7:19 PM, Matthew Jordan mjor...@digium.com wrote: - Original Message - From: Doug Lytle supp...@drdos.info To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 22, 2012 7:22:20 AM Subject: Re: [asterisk-users] Asterisk 1.8.x app_meetme.so You mentioned that the meetme source was there, I was guessing that the option to compile wasn't checked so the binary wasn't available. I just ran into this myself yesterday when converting a 1.4x box (Still in progress) to a 10.2.0 RC2 and once checked and re-compiled, meetme was available. Doug Just a few points of clarification: 1. MeetMe is still the preferred conferencing application in Asterisk 1.8. In Asterisk 10, the preferred conferencing application is ConfBridge. Even still, in Asterisk, 10, you can compile and install MeetMe using menuselect. 2. In the screenshot you attached, you cannot choose to compile MeetMe as one of its dependencies is not available, in this case, DAHDI. Note that DAHDI being a dependency for MeetMe was one of the reasons Asterisk 10 moved to using ConfBridge as the default conferencing application. Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disconnect after 12 seconds w/Cisco 303g Phones
So, I have this customer with a completely bizarr issue. She reports that on either of her shiny new Cisco 303g phone, calls are disconnected at exactly 12 seconds after taking caller off hold. To be clearer: -Answers incoming call, can talk forever no problem. -Places caller on hold and can leave them there forever. -Picks up call and 12 seconds later, call disconnects. I am going to turn on some debug logs on our Asterisk box a bit later and watch the CLI as this happens to see if I can capture anything interesting but wanted to check with the list in the meantime. Other older Linksys/Cisco phone with slightly different model numbers don't have this issue and have worked phone for many moons :-) Never ran into this before, really puzzled. Google, searching the list archives etc, weren't any help. I am thinking this has to be the phones or some settings within. Two phones, same model on same network are doing the same thing. No way to easily get customer a different model phone to test with so I will have to test with softphones to see if removing the phones from the equation resolves the issue. Any help would be appreciated, thanks. --Todd -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: blocking script until playback complete
Chris Bagnall wrote: Greetings list, I've done AGI scripting before, but in the past I've always wanted control to be returned to the dialplan as soon as possible. However, today I have a scenario where I want the script to remain running during the playback of a file so that I can read DTMF at the end of playback. However, doing this: GET DATA en_welcome 5000 6 Results (correctly) in the following in the asterisk console: -- SIP/a.b.c.d-dc027b50 Playing 'en_welcome' (language 'en') But the AGI continues to run on after this point, not waiting for either the sound file to be played, nor for the expected 6 DTMF digits. Adding a simple 10 second sleep/wait to the AGI allows the sound file to be successfully played back. I'm sure I must be missing something very obvious, buy my google-fu is failing me this afternoon. Suggestions gratefully received :-) Thanks in advance. Kind regards, Chris -- Have you tried increasing the timeout value in the command? To me, it appears to be too short. Try setting it to 5. -- Ron Bergin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: blocking script until playback complete
I work with the 1.4 and 10.0 branches and do mainly PERL AGI's. In my testing the GET DATA hasn't worked very reliably. Just for grins, I copied test-agi.agi to get-dig.agi and made these changes on my 10.0.1 box 49,50c49,50 my $result = STDIN; checkresult($result); --- my $result2 = STDIN; checkresult($result2); 52,55c52,56 print STDERR 3. Testing 'sendimage'...; print SEND IMAGE asterisk-image\n; my $result = STDIN; checkresult($result); --- print STDERR 3. Testing 'get digit'...; print GET DATA vm-password 5000 6\n; my $result3 = STDIN; my $result3a=$result3; checkresult($result3); 58,60c59,61 print SAY NUMBER 192837465 \\\n; my $result = STDIN; checkresult($result); --- print SAY NUMBER $result3a \\\n; my $result4 = STDIN; checkresult($result4); 64,65c65,66 my $result = STDIN; checkresult($result); --- my $result5 = STDIN; checkresult($result5); 69,70c70,71 my $result = STDIN; checkresult($result); --- my $result6 = STDIN; checkresult($result6); 72c73 print STDERR 6a. Testing 'record' playback...; --- print STDERR 7. Testing 'record' playback...; 74,75c75,76 my $result = STDIN; checkresult($result); --- my $result7 = STDIN; checkresult($result7); And added this dialplan snippet exten = 1300,1,answer() exten = 1300,n,AGI(get-dig.agi) exten = 1300,n,playback(vm-goodbye,noanswer) exten = 1300,n,hangup() The result - vm-password is played and it waits 5 seconds as you might expect. BUT - whether or not I enter anything, the playback is always 200. So, in my opinion, you would be better off using the dialplan and two agi's if you need DTMF input between. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: Wednesday, February 22, 2012 9:14 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AGI: blocking script until playback complete On 22/2/12 2:55 pm, Danny Nicholas wrote: You don't state the Asterisk version you are running, but personal experience tells me you'd better invest in some Rogaine if you're depending on the built-in stuff from AGI for DTMF input. I have personally wasted weeks trying it. Sorry, should have said, latest 1.4 release. Care to elaborate a little on the issues you found when you tried it? Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: blocking script until playback complete
I gave it from phpagi. It works for me using phpagi's function get_data http://phpagi.sourceforge.net/phpagi22/api-docs/phpAGI/AGI.html Regards, Zohair Raza On Wed, Feb 22, 2012 at 7:20 PM, Chris Bagnall aster...@lists.minotaur.ccwrote: The problem seems to be that GET DATA returns control to the script before the audio file has even played, let alone any DTMF tones have been entered. I would have expected script execution to be blocked until the result from GET DATA was available. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: blocking script until playback complete
On 22/2/12 3:39 pm, Ron Bergin wrote: Have you tried increasing the timeout value in the command? To me, it appears to be too short. Try setting it to 5. Thanks for the suggestion - I tried 5, but no difference, it's not waiting at _all_ for the playback to complete. Just for giggles, I tried exactly the same test on a 1.8 box I have for testing, and the same problem occurs. I'm sure I must be doing something wrong here :-) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.x app_meetme.so
On 02/22/2012 09:23 AM, [Digital^Dude] ® wrote: Doug, I can find the following in asterisk 10 changelogs: The following error will consistently occur when trying to dial into a MeetMe conference when the server does not have DAHDI hardware installed: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) While chan_dahdi is loaded correctly during compilation and install of Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf configuration file in /etc/asterisk is not created by FreePBX if hardware does not exist, causing MeetMe to be unable to open a DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo channel when there is no chan_dahdi.conf file to load. (closes issue ASTERISK-17398) Reported by: Preston Edwards This would mean that meetme should not have dahdi as a compilation dependency. No, this is incorrect. First, you are confusing DAHDI and chan_dahdi. MeetMe absolutely requires, and will always require DAHDI, because DAHDI is used for mixing the audio streams together into conferences. Second, MeetMe also requires chan_dahdi to be loaded, and prior to the patch you listed above, this required a chan_dahdi.conf file to be present. The patch listed above changed changed chan_dahdi to load in a very 'basic' configuration when no chan_dahdi.conf file is present. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.x app_meetme.so
So you mean I can't use dahdi_dummy with meetme? On Wed, Feb 22, 2012 at 9:28 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/22/2012 09:23 AM, [Digital^Dude] ® wrote: Doug, I can find the following in asterisk 10 changelogs: The following error will consistently occur when trying to dial into a MeetMe conference when the server does not have DAHDI hardware installed: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) While chan_dahdi is loaded correctly during compilation and install of Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf configuration file in /etc/asterisk is not created by FreePBX if hardware does not exist, causing MeetMe to be unable to open a DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo channel when there is no chan_dahdi.conf file to load. (closes issue ASTERISK-17398) Reported by: Preston Edwards This would mean that meetme should not have dahdi as a compilation dependency. No, this is incorrect. First, you are confusing DAHDI and chan_dahdi. MeetMe absolutely requires, and will always require DAHDI, because DAHDI is used for mixing the audio streams together into conferences. Second, MeetMe also requires chan_dahdi to be loaded, and prior to the patch you listed above, this required a chan_dahdi.conf file to be present. The patch listed above changed changed chan_dahdi to load in a very 'basic' configuration when no chan_dahdi.conf file is present. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
On Wed, Feb 22, 2012 at 6:23 AM, virendra bhati virbh...@gmail.com wrote: thanks for suggesting the link. Yes i don't have networking, and good SIP communication knowledge. Without understanding how they work, an answer to your question can't be provided. Or more succinctly, you can't formulate a question that gives you a useful answer. Maybe we can start with a more basic question centered around exactly what you are trying to do and why? Is the reason for asking because you want to know how many ports to allocate in a NAT router, for example? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.x app_meetme.so
On Wed, Feb 22, 2012 at 10:30 AM, [Digital^Dude] ® millennium@gmail.com wrote: So you mean I can't use dahdi_dummy with meetme? That's not what he means at all. What it means is, you are required to install the software named DAHDI before you are able to compile and load the asterisk application MeetMe(). It also means you do not need to have a chan_dahdi.conf file in your /etc/asterisk directory. So, to recap, you must install and run DAHDI on the same server as your asterisk box if you want to use MeetMe, but you don't have to use DAHDI anywhere in asterisk itself (for instance, if you don't have any TDM interface cards). The dahdi_dummy virtual device was removed a few versions ago as it was redundant - just installing DAHDI provided the same timing source that dahdi_dummy did. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi and digium debian package
Hi I'm trying to install dahdi. I just need the dahdi timer for conference. I currently using digium debian package for asterisk 1.8.8.1. When i install asterisk-dahdi , i've got several dependencies which came for official debian repository (including the dahdi package) and are outdated. Is it normal than dahdi is not include into digium packages ? Do i have to compil it before install asterisk-dahdi ? Thanks for your help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi and digium debian package
On 12-02-22 12:09 PM, ml asterisk wrote: Hi I'm trying to install dahdi. I just need the dahdi timer for conference. I currently using digium debian package for asterisk 1.8.8.1. When i install asterisk-dahdi , i've got several dependencies which came for official debian repository (including the dahdi package) and are outdated. Is it normal than dahdi is not include into digium packages ? Do i have to compil it before install asterisk-dahdi ? Yes, we do not package DAHDI and rely on the OS version. So it is possible that some of the dependencies will be from Debian. As long as the asterisk-dahdi package comes for the same repository as the asterisk package you should be fine. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi and digium debian package
On Wednesday 22 February 2012, ml asterisk wrote: Hi I'm trying to install dahdi. I just need the dahdi timer for conference. I currently using digium debian package for asterisk 1.8.8.1. When i install asterisk-dahdi , i've got several dependencies which came for official debian repository (including the dahdi package) and are outdated. Is it normal than dahdi is not include into digium packages ? Do i have to compil it before install asterisk-dahdi ? Debian packages are designed for least surprise. By the time a package has made it to Stable it is pretty thoroughly tested, but may be out of date. My best suggestion? Uninstall and purge any Debian packages you installed (backup needed config files first .) and just install Dahdi and Asterisk from the Source Code. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi and digium debian package
On 12-02-22 01:36 PM, A J Stiles wrote: My best suggestion? Uninstall and purge any Debian packages you installed (backup needed config files first .) and just install Dahdi and Asterisk from the Source Code. I don't know how to reply to this. I can only assume you only manage a single box. What if OP is running more then 1 asterisk box, manually compiling asterisk and installing it each time would not be the best solution. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec mismatch on channel
I get this on 1.8.x as well. I assume it is a harmless bug. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Nowrot Sent: Wednesday, February 22, 2012 3:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] codec mismatch on channel Hi I am keep getting this warning message when doing attendant transfer: WARNING[6027] channel.c: Codec mismatch on channel Local/XX@Inside-1f32;1 setting write format to slin from alaw native formats (alaw) What can I do to lose it. I am using asterisk 10.1.2 Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec mismatch on channel
I think it's a warning as opposed to a bug. If the call were happening all in Tecnology (SIP/DAHDI/etc), the warning would be because your channel didn't support the codec (I can't do alaw so I'm gonna talk in slin). The LOCAL channel by definition (AFAIK) doesn't specifically support/deny any codec format. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, February 22, 2012 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] codec mismatch on channel I get this on 1.8.x as well. I assume it is a harmless bug. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Nowrot Sent: Wednesday, February 22, 2012 3:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] codec mismatch on channel Hi I am keep getting this warning message when doing attendant transfer: WARNING[6027] channel.c: Codec mismatch on channel Local/XX@Inside-1f32;1 setting write format to slin from alaw native formats (alaw) What can I do to lose it. I am using asterisk 10.1.2 Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec mismatch on channel
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, February 22, 2012 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] codec mismatch on channel I get this on 1.8.x as well. I assume it is a harmless bug. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Nowrot Sent: Wednesday, February 22, 2012 3:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] codec mismatch on channel Hi I am keep getting this warning message when doing attendant transfer: WARNING[6027] channel.c: Codec mismatch on channel Local/XX@Inside-1f32;1 setting write format to slin from alaw native formats (alaw) What can I do to lose it. I am using asterisk 10.1.2 Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: blocking script until playback complete
On Wed, 22 Feb 2012, Chris Bagnall wrote: However, today I have a scenario where I want the script to remain running during the playback of a file so that I can read DTMF at the end of playback. If by 'remain running' you mean 'do other stuff while the file is playing' you can stream the file in a separate thread while continuing with the 'mainline' code. If you mean your script is returning to the dialplan before the file is finished playing, you're doing something wrong. However, doing this: GET DATA en_welcome 5000 6 Results (correctly) in the following in the asterisk console: -- SIP/a.b.c.d-dc027b50 Playing 'en_welcome' (language 'en') But the AGI continues to run on after this point, not waiting for either the sound file to be played, nor for the expected 6 DTMF digits. Adding a simple 10 second sleep/wait to the AGI allows the sound file to be successfully played back. Sure sounds like you are doing something wrong. Anytime a 'sleep' helps, something is definitely wrong. Most AGI 'weirdness' can be traced to violating the AGI protocol. These are the most common mistakes: 0) Not using an established library. Nobody gets it right the first time :) 1) Not reading the AGI environment from STDIN before issuing any AGI requests. 2) Forgetting that every request must be followed by reading the response. 3) Doing any I/O on STDIN or STDOUT. These channels are connected to Asterisk, not any console. Don't even think about popping in a quick 'debugging printf.' Looking over my own AGI code, I don't see any examples where I used GET DATA. I tend to use STREAM FILE and WAIT FOR DIGIT. I'm a 1.2 Luddite, so any bugs I've been exposed to may have been fixed by now. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: blocking script until playback complete
On Wed, 22 Feb 2012, Zohair Raza wrote: Try passing escape character GET DATA $filename $timeout $max_digits $escape_character Why? Is this 'extra' parameter documented in your version of Asterisk? -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: blocking script until playback complete
On Wed, 22 Feb 2012, Danny Nicholas wrote: So, in my opinion, you would be better off using the dialplan and two agi's if you need DTMF input between. I've written complete 'applications*' as single AGIs so the combination of STREAM FILE and WAIT FOR DIGIT works reliably for me. *) voicemail, chat services, credit card processing, listen to stories, blah, blah, blah. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: blocking script until playback complete
Yes, but it doesn't seem to indicate if the timeout is in seconds of milliseconds. pbx*CLI agi show commands topic get data -= Info about agi 'get data' =- [Syntax] get data file [timeout] [maxdigits] [Description] Stream the given file, and receive DTMF data. Returns the digits received from the channel at the other end. [Synopsis] Prompts for DTMF on a channel [Runs Dead] No [See Also] Not available -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, February 22, 2012 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AGI: blocking script until playback complete On Wed, 22 Feb 2012, Zohair Raza wrote: Try passing escape character GET DATA $filename $timeout $max_digits $escape_character Why? Is this 'extra' parameter documented in your version of Asterisk? -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI: blocking script until playback complete
While it is never actually safe to assume anything about Asterisk, the general setting for seconds is milliseconds. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, February 22, 2012 4:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AGI: blocking script until playback complete Yes, but it doesn't seem to indicate if the timeout is in seconds of milliseconds. pbx*CLI agi show commands topic get data -= Info about agi 'get data' =- [Syntax] get data file [timeout] [maxdigits] [Description] Stream the given file, and receive DTMF data. Returns the digits received from the channel at the other end. [Synopsis] Prompts for DTMF on a channel [Runs Dead] No [See Also] Not available -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, February 22, 2012 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AGI: blocking script until playback complete On Wed, 22 Feb 2012, Zohair Raza wrote: Try passing escape character GET DATA $filename $timeout $max_digits $escape_character Why? Is this 'extra' parameter documented in your version of Asterisk? -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferenced transfers
On Feb 21, 2012, at 17:01 , Phil Frost wrote: On Feb 14, 2012, at 17:13 , isr...@gmail.com wrote: On the snom too Create a conferance and then press the transfer button. That will join the parties and release the receptionist Hmm...You can do that with just hitting the transfer button, or is there more? I'm using a Snom 870 with firmware 8.4.35. I set up the conference, but when I hit transfer, it presents me with a transfer party dialog. It's a bit confusing, because it's not really clear which party is being transferred. Are you using a different phone or firmware? Maybe there's a setting somewhere? Well, I happened to figure it out. On the Snom 870, when you have the conference going, both parties are in the context area. Drag one and drop it on the other and bam, they are connected, and now you can hang up while they continue to talk. Pretty obvious once you figure it out, but rather unconventional, and not documented anywhere best I can tell. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
virbh...@gmail.com wrote: how many UDP ports is required for 1 call. and why . If you mean a voice call, it appears that each host must open three UDP sockets: - One to send/receive SIP commands - Two to receive sound (one for RTP, one for RTCP; The first port is even, the other is odd) This is I think the best answer provided to you so far. The simplest and most relevant I should say. Just to add to that, first port is TCP port and the other two are UDP ports assuming you are using SIP protocol. -Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many UDP ports is required for 1 call
On Wed, Feb 22, 2012 at 5:04 PM, Kevin P. Fleming kpflem...@digium.com wrote: This is the only answer that didn't require the original poster to have to know/learn anything, and thus while it is the 'best' answer, it also doesn't lead to any increased understanding on his part. SIP is most commonly transported over UDP, not TCP. And probably doesn't really answer whatever it is that he really needed to know or do, since how many ports doesn't seem like anything that would be a useful thing to know. Call it SIP trivia. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Park and PARKINGDYNAMIC
I have been trying to get the dynamic parking working.
For some reason when I park a call using this method the console says
it is using the default parking context not the one I am trying to
specidfy. It also is playing the parked extension to the caller. I
am transfering the call to an extension that is doing a goto to the
context below. Any ideas or examples on how to make this work. What
I need to be able to do is have multiple parking lots using the same
extension pools but seperated by a dynamic context of
${account}-Lot. So that each office suite cant cross pickup another
groups parked calls while using the same number pool of 110-120. I
need the dynamic option as all of our calls are database driven and
we can't add a seperate entry per customer to the feautres.conf.
[MSIP-DynPark]
exten = s,1,NoOp(Dynamic Parking)
exten = s,n,NoOp(Return Parked Call)
exten = s,n,GoTo(${CUT(${l_ndeContext}-ndeArgs,~,1)},1)
exten = _XXX,1,Set(PARKINGDYNAMIC=parkinglot_small)
exten = _XXX,n,Set(PARKINGDYNEXTEN=110)
exten = _XXX,n,Set(PARKINGDYNPOS=111-120)
exten = _XXX,n,Set(PARKINGDYNCONTEXT=${account}-Lot)
;exten = _XXX,n,Set(PARKINGEXTEN=99)
exten = _XXX,n,Park()
[MSIP-DynParkPickup]
exten = _NXX,1,ParkedCall(${EXTEN},${account}-Lot)
exten = _NXX,hint,park:$EXTEN@${account}-Lot
I think you have a couple issues with this dialplan:
1) Many issues with parking lots have been fixed so I recommend at least
v1.8.7 or v10.1. Multiple parkinglots before these versions was very
broken.
2) PARKINGDYNAMIC is the parkinglot to use as a template for a dynamically
created parkinglot. It is not the parkinglot to park the call.
3) The dialplan you show here is not specifying a parkinglot so it will be a
parkinglot set when the channel is created or the default. If the selected
parkinglot does not exist it will then be dynamically created.
See the CLI core show application Park documentation.
4) You are likely running into
https://issues.asterisk.org/jira/browse/ASTERISK-19322
Richard
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[asterisk-users] Phone Inventory
Hi there I have just took a support of a customer with hundreds of IP phones, mostly Polycom with mixed models. Is there a way to query asterisk or any other command to retrieve the inventory of all connected phones. i.e. Phone Type and Phone Model, say Polycom SPIP331 or so Thanks Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] p-associated-uri in 200OK
Hi, Can someone share how can I configure asterisk to get P-Associated-Uri header in 200 Ok to the REGISTER. Thanks, Amit -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] p-associated-uri in 200OK
Am 23.02.12 07:18, schrieb Goyal, Amit: Hi, Can someone share how can I configure asterisk to get P-Associated-Uri header in 200 Ok to the REGISTER. Thanks, Amit Hello amit, AFAIK P-Associated-Uri is not supported by asterisk in any version so you cannot configure it. I even dont think that this header field and its usage typically fits to how asterisk is used, but i might be wrong about this ;) best regards Stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
