Re: [asterisk-users] ACD problem
I am not sure I understand the required routing pattern, but I'm sure queues are your friends, as you can dynamically add and remove member and you can have a first-level queue easily move fall-through to another queue in case all members should be busy or none should be available. Plus by using queues you decouple the what you want to do from the who is doing it. 2013/4/10 Tommy Cooper tomcoope...@yahoo.com Hi, I am working on a small inbound call center solution that uses an ACD system. I might add an IVR system later on. I only have 2 extensions set up (extensions 1000 and 1001), I want the system to put new calls in a queue if both extensions are busy. I am currently subscribed with a SIP trunk provider and can successfully recieve calls. I want to design a system where customers can call my number, that call will then be directed to either extension 1000 or 1001. If both extensions are in use, I want that 3rd call to be queued. I don't think that the config below will direct calls to extension 1001 because the second line states that any incomming calls should be routed to extension 1000. How do I change this so that calls are directed to all of my exensions? extensions.conf [from-myprovider] exten = *DID number*,1,Answer exten = *DID number*,2,Dial(SIP/1000) exten = *DID number*,3,Queue(support) ;not sure if this line belongs here exten = *DID number*,4,Hangup queues.conf [general] [support] musicclass=default strategy=rrmemory joinempty=no leavewhenempty=yes ringinuse=no Member = SIP/1000 Member = SIP/1001 agent = 1000,1000 agent = 1001,1001 When using the current config the caller will listen to the 'music on hold' until the agent answers but calls are only being forwarded to extension 1000 as stated above -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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] Logging SIP connection status for review
On Wed, 2013-04-10 at 11:06 -0700, Carlos Alvarez wrote: On Wed, Apr 10, 2013 at 11:02 AM, Steve Edwards asterisk@sedwards.com wrote: dumpcap can capture all of the SIP (and RTP) packets into a series of files without a huge performance hit. A cron job can pbzip2 the files and delete if over x days old. That's completely different. We already run a good packet capture system. What I want to see is SIP registration statuses and latency logged about once a minute. We do that now by doing a 'sip show peers like x' and putting it in a text file. I can then correlate issues with times of high latency or unreachable phones. I'd just like to see more reporting and the ability to correlate times and such. How about using your current scripts and then pushing the data into Graphite? http://kaivanov.blogspot.co.uk/2012/02/how-to-install-and-use-graphite.html Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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 11.2.1 / dahdi destroy channel / asterisk crashes
Hi, I have the following setup: Ubuntu 12.04.02 LTS (64 bit) Asterisk 11.2.1 Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected) WANPIPE Release: 3.5.28 DAHDI Version: 2.6.1 Echo Canceller: HWEC libpri version: 1.4.12 I call via sip into the dialplan. Then I do a Dial(DAHDI/g1/voicenumber,r). The call is bridged and everything is fine. dahdi show channels shows me, that channel 1 is used for the outcall. Then I try to hangup the outcall via dahdi destroy channel 1. Asterisk crahes immediatly. No message is logged (verbose is 10 and debug is 10). I get disconnected from the atserisk cli at this moment: vlr-3*CLI dahdi destroy channel 1 vlr-3*CLI Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups voxi@vlr-3:/tmp$ Is this a bug or is this my fault? Best regards -Thorsten- -- _ -- 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 11.2.1 / dahdi destroy channel / asteriskcrashes
CLIchannel request hangup DAHDI/1-1 Would work. But 'dahdi destroy channel 1' shouldn't segfault asterisk. Alec -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten Göllner Sent: Thursday, 11 April 2013 8:57 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 11.2.1 / dahdi destroy channel / asteriskcrashes Hi, I have the following setup: Ubuntu 12.04.02 LTS (64 bit) Asterisk 11.2.1 Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected) WANPIPE Release: 3.5.28 DAHDI Version: 2.6.1 Echo Canceller: HWEC libpri version: 1.4.12 I call via sip into the dialplan. Then I do a Dial(DAHDI/g1/voicenumber,r). The call is bridged and everything is fine. dahdi show channels shows me, that channel 1 is used for the outcall. Then I try to hangup the outcall via dahdi destroy channel 1. Asterisk crahes immediatly. No message is logged (verbose is 10 and debug is 10). I get disconnected from the atserisk cli at this moment: vlr-3*CLI dahdi destroy channel 1 vlr-3*CLI Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups voxi@vlr-3:/tmp$ Is this a bug or is this my fault? Best regards -Thorsten- -- _ -- 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] PRI DEBUG
hi,
strange behaviour while trying to use pri debugging on asterisk 11.x ...
please take a look:
bas1104*CLI pri show version
libpri version: 1.4.13
bas1104*CLI dahdi show version
DAHDI Version: 2.6.1 Echo Canceller: HWEC
bas1104*CLI help pri
*pri intense debug span*no description available
pri service disable channel Remove a channel from service
pri service enable channel Return a channel to service
*pri set debug {on|off*|hex|inte Enables PRI debugging on a span
pri set debug file Sends PRI debug output to the specified file
pri show channels Displays PRI channel information
*pri show debug*Displays current PRI debug settings
pri show spans Displays PRI span information
pri show span Displays PRI span information
pri show version Displays libpri version
bas1104*CLI help dahdi
dahdi destroy channel Destroy a channel
dahdi restart Fully restart DAHDI channels
dahdi set dnd Sets/resets DND (Do Not Disturb) mode on
a channel
dahdi set hwgain Set hardware gain on a channel
dahdi set swgain Set software gain on a channel
dahdi show cadences List cadences
dahdi show channels [group|con Show active DAHDI channels
dahdi show channel Show information on a channel
dahdi show status Show all DAHDI cards status
dahdi show version Show the DAHDI version in use
/
//currently all debug off:/
bas1104*CLI pri show debug
Span 1: Debug: No Intense: No
Span 2: Debug: No Intense: No
Span 3: Debug: No Intense: No
Span 4: Debug: No Intense: No
/
//switching it on (which currently works as expected)/
bas1104*CLI pri intense debug span 1
Enabled debugging on span 1
/
//
//oops, still shows no debug but it IS activated.../
bas1104*CLI pri show debug
Span 1: Debug: No Intense: No
Span 2: Debug: No Intense: No
Span 3: Debug: No Intense: No
Span 4: Debug: No Intense: No
/
//huh... how to disable it again? on some machines I can do so with pri
no debug span nr but not here... gives same result (no//
//such command) and debug is still enabled.../
bas1104*CLI pri set debug off
No such command 'pri set debug off' (type 'core show help pri set' for
other possible commands)
bas1104*CLI
so... whats the right way to disable pri debugging?
thx,
yves
--
_
-- 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 11.2.1 / dahdi destroy channel / asterisk crashes
Hi, I can reproduce your report (11.0.1, libpri 1.4.13, dahdi 2.6.1) and would say it is a bug... To remotely hang up a call use * **hangup request channel* where channel is the exact id of your channel as you would receive it via *core show channels* yves Am 11.04.2013 10:56, schrieb Thorsten Göllner: Hi, I have the following setup: Ubuntu 12.04.02 LTS (64 bit) Asterisk 11.2.1 Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected) WANPIPE Release: 3.5.28 DAHDI Version: 2.6.1 Echo Canceller: HWEC libpri version: 1.4.12 I call via sip into the dialplan. Then I do a Dial(DAHDI/g1/voicenumber,r). The call is bridged and everything is fine. dahdi show channels shows me, that channel 1 is used for the outcall. Then I try to hangup the outcall via dahdi destroy channel 1. Asterisk crahes immediatly. No message is logged (verbose is 10 and debug is 10). I get disconnected from the atserisk cli at this moment: vlr-3*CLI dahdi destroy channel 1 vlr-3*CLI Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups voxi@vlr-3:/tmp$ Is this a bug or is this my fault? Best regards -Thorsten- -- _ -- 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] Dropping call because extensions '200', 's' and 'i' doesn't exists
hello all i,m newbie in asterisk and now want to sip and h323 connection. this is my scenario: phone(ext100)---freepbx---sip---system1---H323---system2---freepbx---phone(ext200) when i call 100 from 200, every thing is ok and phone is ringing but when i call 200 from 100, it says service unavailable. i debug asterisk in my system 2 and see below message: Dropping call because extensions '200', 's' and 'i' doesn't exists in context [from-trunk] i googled about this message and found that file extensions_mor_h323.conf should be included into /etc/asterisk/extensions_mor.conf. but i don't have any extensions_mor.conf file at all!!! is extensions_mor.conf really necessary to fix my problem?if yes, how i have connection in one way without this file? if no, how i can fix this problem? thanks in advance 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
Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists
On Thursday 11 April 2013, s m wrote: when i call 100 from 200, every thing is ok and phone is ringing but when i call 200 from 100, it says service unavailable. i debug asterisk in my system 2 and see below message: Dropping call because extensions '200', 's' and 'i' doesn't exists in context [from-trunk] OK. What do you have in the [from-trunk] context in your extensions.conf ? -- 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] Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes
Thanks! I do not have experience with bug reporting. Is that neccessary in that case? Where can I open a ticket for it (if neccessary)? Am 11.04.2013 12:23, schrieb Yves A.: Hi, I can reproduce your report (11.0.1, libpri 1.4.13, dahdi 2.6.1) and would say it is a bug... To remotely hang up a call use * **hangup request channel* where channel is the exact id of your channel as you would receive it via *core show channels* yves Am 11.04.2013 10:56, schrieb Thorsten Göllner: Hi, I have the following setup: Ubuntu 12.04.02 LTS (64 bit) Asterisk 11.2.1 Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected) WANPIPE Release: 3.5.28 DAHDI Version: 2.6.1 Echo Canceller: HWEC libpri version: 1.4.12 I call via sip into the dialplan. Then I do a Dial(DAHDI/g1/voicenumber,r). The call is bridged and everything is fine. dahdi show channels shows me, that channel 1 is used for the outcall. Then I try to hangup the outcall via dahdi destroy channel 1. Asterisk crahes immediatly. No message is logged (verbose is 10 and debug is 10). I get disconnected from the atserisk cli at this moment: vlr-3*CLI dahdi destroy channel 1 vlr-3*CLI Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups voxi@vlr-3:/tmp$ Is this a bug or is this my fault? Best regards -Thorsten- -- -- _ -- 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] Dropping call because extensions '200', 's' and 'i' doesn't exists
this is my [from-trunk] extension:
[from-trunk]
exten=_2.,1,Dial(SIP/to-232/2${EXTEN:1})
and this is [to-231] in sip_additional.conf:
[to-232]
host=192.168.0.232
type=peer
qualify=yes
and 192.168.0.232 in the ip address of my freepbx.
On 4/11/13, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Thursday 11 April 2013, s m wrote:
when i call 100 from 200, every thing is ok and phone is ringing but
when i call 200 from 100, it says service unavailable.
i debug asterisk in my system 2 and see below message:
Dropping call because extensions '200', 's' and 'i' doesn't exists
in context [from-trunk]
OK. What do you have in the [from-trunk] context in your extensions.conf ?
--
AJS
Answers come *after* questions.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists
hi,
try
exten= _2.,1,Dial(SIP/to-232/2${EXTEN:1})
Note space before underscore.
On Thu, Apr 11, 2013 at 2:50 PM, s m sam.gh1...@gmail.com wrote:
this is my [from-trunk] extension:
[from-trunk]
exten=_2.,1,Dial(SIP/to-232/2${EXTEN:1})
and this is [to-231] in sip_additional.conf:
[to-232]
host=192.168.0.232
type=peer
qualify=yes
and 192.168.0.232 in the ip address of my freepbx.
On 4/11/13, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Thursday 11 April 2013, s m wrote:
when i call 100 from 200, every thing is ok and phone is ringing but
when i call 200 from 100, it says service unavailable.
i debug asterisk in my system 2 and see below message:
Dropping call because extensions '200', 's' and 'i' doesn't exists
in context [from-trunk]
OK. What do you have in the [from-trunk] context in your
extensions.conf ?
--
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
--
_
-- 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 11.2.1 / dahdi destroy channel / asteriskcrashes
- Original Message - CLIchannel request hangup DAHDI/1-1 Would work. But 'dahdi destroy channel 1' shouldn't segfault asterisk. The dahdi destroy channel command is *only* for use when you know what your doing. Even then I would not recommend ever using that command. The CLI help for that command shows: Usage: dahdi destroy channel chan num DON'T USE THIS UNLESS YOU KNOW WHAT YOU ARE DOING. Immediately removes a given channel, whether it is in use or not. So if that channel were in use then I would expect to get a segfault because that channel is unconditionally removed from the system and cannot be used again until Asterisk is restarted. Alec -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten Göllner Sent: Thursday, 11 April 2013 8:57 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 11.2.1 / dahdi destroy channel / asteriskcrashes Hi, I have the following setup: Ubuntu 12.04.02 LTS (64 bit) Asterisk 11.2.1 Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected) WANPIPE Release: 3.5.28 DAHDI Version: 2.6.1 Echo Canceller: HWEC libpri version: 1.4.12 I call via sip into the dialplan. Then I do a Dial(DAHDI/g1/voicenumber,r). The call is bridged and everything is fine. dahdi show channels shows me, that channel 1 is used for the outcall. Then I try to hangup the outcall via dahdi destroy channel 1. Asterisk crahes immediatly. No message is logged (verbose is 10 and debug is 10). I get disconnected from the atserisk cli at this moment: vlr-3*CLI dahdi destroy channel 1 vlr-3*CLI Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups voxi@vlr-3:/tmp$ Is this a bug or is this my fault? It is the wrong command. Richard -- _ -- 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] Setting a CDR field from using feature codes...
hi, you have not assign any value to CDR(userfield). try code = #111,self,SET(CDR(userfield)=111) On Thu, Apr 11, 2013 at 12:53 AM, Carlos Chavez cur...@telecomabmex.comwrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I am trying to set the CDR(userfield) to a certain vaule using the application map of features.conf but I am not able to do it. When I receive a call I would like to tag it with a client code (3 digit numeric) so I can referenci it later from the CDR. I have edited features.conf with something like: code = #111,self,SET(CDR(userfield(111)) or code = #111,self,AGI(code.agi) The DYNAMIC_FEATURES variable is in the globals section and includes the application map name. When I do a features reload I can see everything loads and when I dial the code during a call I can see a message like: - -- Feature Found: code exten: code The problem is that my CDR variable is not being written to. The first example does not show anything on screen. For the second when I turn agi debug on I can see: SIP/2001-0003AGI Rx SET VARIABLE CDR(userfield) 111 But when I hang up neither my h extension or the CDR itself will show the value I set, it is empty. I do not know what I am doing wrong or maybe CDR variables are not available from features? - -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.18 (Darwin) Comment: GPGTools - http://gpgtools.org Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ iEYEARECAAYFAlFl7VYACgkQqmNh+MyHzx7SzACggvfeVZEE70JhVUXjzEvCTTg9 d2gAoJWAYR7cBI7DCfbL47s6afIiZB9G =SJlv -END PGP SIGNATURE- -- _ -- 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] Voicemail Prepend not working properly on 1.8.18
Hi, I have a problem with forwarding a voicemail and prepending a message to it. If a user just forwards a voicemail, everything works fine. However, if a user prepends a message to the voicemail when forwarding, the voicemail that is forwarded only contains the prepended message and not the original voicemail message. Also, I continue to have voicemails and recordings that are recording the '#' to end the message. Thanks. -- James -- _ -- 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] Setting a CDR field from using feature codes...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 4/11/13 11:18 AM, Asghar Mohammad wrote: hi, you have not assign any value to CDR(userfield). try code = #111,self,SET(CDR(userfield)=111) On Thu, Apr 11, 2013 at 12:53 AM, Carlos Chavez cur...@telecomabmex.com mailto:cur...@telecomabmex.com wrote: I am trying to set the CDR(userfield) to a certain vaule using the application map of features.conf but I am not able to do it. When I receive a call I would like to tag it with a client code (3 digit numeric) so I can referenci it later from the CDR. I have edited features.conf with something like: code = #111,self,SET(CDR(userfield(111)) or code = #111,self,AGI(code.agi) The DYNAMIC_FEATURES variable is in the globals section and includes the application map name. When I do a features reload I can see everything loads and when I dial the code during a call I can see a message like: -- Feature Found: code exten: code The problem is that my CDR variable is not being written to. The first example does not show anything on screen. For the second when I turn agi debug on I can see: SIP/2001-0003AGI Rx SET VARIABLE CDR(userfield) 111 But when I hang up neither my h extension or the CDR itself will show the value I set, it is empty. I do not know what I am doing wrong or maybe CDR variables are not available from features? That was a copy/paste error on my part. The line is as you put it but I cannot get the value after. - -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.18 (Darwin) Comment: GPGTools - http://gpgtools.org Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ iEYEARECAAYFAlFm56gACgkQqmNh+MyHzx6VpwCePy+X5YzFX68fTbTDtqXRe3PO kvMAn3mEXOddPyd9wu/HTRu7QjPAv5xJ =rbhr -END PGP SIGNATURE- -- _ -- 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] Is there a php script to analyse and show call detail reports from Asterisk CDR?
hi, it is not difficult in php and mysql i have created a simple billing system for my wholesale postpay clients without any AGI. it report ACD ASR all calls ANSWERD calls filter by date by callerid etc. do billing as soon as call end. for billing i am using mysql trigger. report live calls. 2 interfaces 1 for admin and other for clients, every client can login with his accountcode and password and can see live calls cdr report billing etc. i am still working on this so codes are not clean. if someone need to create a new interface i can help. On Wed, Apr 10, 2013 at 11:22 PM, Daniel - Asterisk earohua...@gmail.comwrote: Hello Brynjolfur Thorvardsson, Can I take a look at you CDR reporting tool? I'm planning on using it on Postgresql but MySQL could be used too. Thank you! Elder D. Arohuanca dCAP Lima - Peru On Fri, Feb 10, 2012 at 11:55 AM, asterisk jobs asteriskcod...@gmail.comwrote: No, that doesn't do the job I specifically asked and installation instructions are all over the place... Thanks though. On Fri, Feb 10, 2012 at 11:36 AM, Tim Nelson tnel...@rockbochs.comwrote: - Original Message - Yes, this is exactly what I am looking for - hopefully in English :-) Date or range selection would make this perfect. I have been looking for something like this for quite a while but there is none. I would really appreciate it if you share this with me. Question here, does the .php code read from database and displays or does it analyse the custom-cdr.csv file? Don't forget about the ever-popular Asterisk-stat and the newly revised cdr-stats projects, both web based, proven, and work fantastic: http://www.areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 http://www.cdr-stats.org/ --Tim -- _ -- 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 -- _ -- 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] PRI DEBUG
- Original Message -
hi,
strange behaviour while trying to use pri debugging on asterisk 11.x
...
please take a look:
bas1104*CLI pri show version
libpri version: 1.4.13
bas1104*CLI dahdi show version
DAHDI Version: 2.6.1 Echo Canceller: HWEC
bas1104*CLI help pri
pri intense debug span no description available
pri service disable channel Remove a channel from service
pri service enable channel Return a channel to service
pri set debug {on|off |hex|inte Enables PRI debugging on a span
pri set debug file Sends PRI debug output to the specified file
pri show channels Displays PRI channel information
pri show debug Displays current PRI debug settings
pri show spans Displays PRI span information
pri show span Displays PRI span information
pri show version Displays libpri version
bas1104*CLI help dahdi
dahdi destroy channel Destroy a channel
dahdi restart Fully restart DAHDI channels
dahdi set dnd Sets/resets DND (Do Not Disturb) mode on a channel
dahdi set hwgain Set hardware gain on a channel
dahdi set swgain Set software gain on a channel
dahdi show cadences List cadences
dahdi show channels [group|con Show active DAHDI channels
dahdi show channel Show information on a channel
dahdi show status Show all DAHDI cards status
dahdi show version Show the DAHDI version in use
currently all debug off:
bas1104*CLI pri show debug
Span 1: Debug: No Intense: No
Span 2: Debug: No Intense: No
Span 3: Debug: No Intense: No
Span 4: Debug: No Intense: No
switching it on (which currently works as expected)
bas1104*CLI pri intense debug span 1
Enabled debugging on span 1
oops, still shows no debug but it IS activated...
It activated a different mode of debug than what you expected
because that command is an alias that was not updated.
bas1104*CLI pri show debug
Span 1: Debug: No Intense: No
Span 2: Debug: No Intense: No
Span 3: Debug: No Intense: No
Span 4: Debug: No Intense: No
huh... how to disable it again? on some machines I can do so with
pri no debug span nr but not here... gives same result (no
such command) and debug is still enabled...
bas1104*CLI pri set debug off
No such command 'pri set debug off' (type 'core show help pri set'
for other possible commands)
bas1104*CLI
so... whats the right way to disable pri debugging?
The correct command is pri set debug {on|off|intense} span x.
The pri intense debug span x command is an alias for
pri set debug 2 span x that didn't get updated when the real
command was changed to pri set debug intense span x.
This will show the help you need:
bas1104*CLI help pri set debug off span
Richard
--
_
-- 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] Setting a CDR field from using feature codes...
i am using
exten = _XXX.,n,Set(CDR(cli_name)=${CHANNEL(peerip)})
cli_name is field in mysql and it work fine.
show me cli output without AGI.
On Thu, Apr 11, 2013 at 6:41 PM, Carlos Chavez cur...@telecomabmex.comwrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 4/11/13 11:18 AM, Asghar Mohammad wrote:
hi, you have not assign any value to CDR(userfield). try code =
#111,self,SET(CDR(userfield)=111)
On Thu, Apr 11, 2013 at 12:53 AM, Carlos Chavez
cur...@telecomabmex.com mailto:cur...@telecomabmex.com wrote:
I am trying to set the CDR(userfield) to a certain vaule using the
application map of features.conf but I am not able to do it. When
I receive a call I would like to tag it with a client code (3
digit numeric) so I can referenci it later from the CDR. I have
edited features.conf with something like:
code = #111,self,SET(CDR(userfield(111))
or
code = #111,self,AGI(code.agi)
The DYNAMIC_FEATURES variable is in the globals section and
includes the application map name. When I do a features reload I
can see everything loads and when I dial the code during a call I
can see a message like:
-- Feature Found: code exten: code
The problem is that my CDR variable is not being written to. The
first example does not show anything on screen. For the second
when I turn agi debug on I can see:
SIP/2001-0003AGI Rx SET VARIABLE CDR(userfield) 111
But when I hang up neither my h extension or the CDR itself will
show the value I set, it is empty. I do not know what I am doing
wrong or maybe CDR variables are not available from features?
That was a copy/paste error on my part. The line is as you put it
but I cannot get the value after.
- --
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
-BEGIN PGP SIGNATURE-
Version: GnuPG/MacGPG2 v2.0.18 (Darwin)
Comment: GPGTools - http://gpgtools.org
Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/
iEYEARECAAYFAlFm56gACgkQqmNh+MyHzx6VpwCePy+X5YzFX68fTbTDtqXRe3PO
kvMAn3mEXOddPyd9wu/HTRu7QjPAv5xJ
=rbhr
-END PGP SIGNATURE-
--
_
-- 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] Setting a CDR field from using feature codes...
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
When I execute without using the AGI method I get no output on the CLI
at all.
On 4/11/13 11:54 AM, Asghar Mohammad wrote:
i am using exten =
_XXX.,n,Set(CDR(cli_name)=${CHANNEL(peerip)}) cli_name is field
in mysql and it work fine. show me cli output without AGI.
On Thu, Apr 11, 2013 at 6:41 PM, Carlos Chavez
cur...@telecomabmex.com mailto:cur...@telecomabmex.com wrote:
On 4/11/13 11:18 AM, Asghar Mohammad wrote:
hi, you have not assign any value to CDR(userfield). try code =
#111,self,SET(CDR(userfield)=111)
On Thu, Apr 11, 2013 at 12:53 AM, Carlos Chavez
cur...@telecomabmex.com mailto:cur...@telecomabmex.com
mailto:cur...@telecomabmex.com mailto:cur...@telecomabmex.com
wrote:
I am trying to set the CDR(userfield) to a certain vaule using
the application map of features.conf but I am not able to do it.
When I receive a call I would like to tag it with a client code
(3 digit numeric) so I can referenci it later from the CDR. I
have edited features.conf with something like:
code = #111,self,SET(CDR(userfield(111))
or
code = #111,self,AGI(code.agi)
The DYNAMIC_FEATURES variable is in the globals section and
includes the application map name. When I do a features reload
I can see everything loads and when I dial the code during a call
I can see a message like:
-- Feature Found: code exten: code
The problem is that my CDR variable is not being written to. The
first example does not show anything on screen. For the second
when I turn agi debug on I can see:
SIP/2001-0003AGI Rx SET VARIABLE CDR(userfield) 111
But when I hang up neither my h extension or the CDR itself will
show the value I set, it is empty. I do not know what I am
doing wrong or maybe CDR variables are not available from
features?
That was a copy/paste error on my part. The line is as you put it
but I cannot get the value after.
--
_
- -- 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
- --
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
-BEGIN PGP SIGNATURE-
Version: GnuPG/MacGPG2 v2.0.18 (Darwin)
Comment: GPGTools - http://gpgtools.org
Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/
iEYEARECAAYFAlFm7e8ACgkQqmNh+MyHzx6POwCeLtZtIH42LgTPE/N0/l7kpfDP
XpkAnRqtgX6iFhaGzn29B+rjFhXd6tIv
=VW+3
-END PGP SIGNATURE-
--
_
-- 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] Setting a CDR field from using feature codes...
how you are executing?
show me your full context and how call enter in context.
On Thu, Apr 11, 2013 at 7:07 PM, Carlos Chavez cur...@telecomabmex.comwrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
When I execute without using the AGI method I get no output on the CLI
at all.
On 4/11/13 11:54 AM, Asghar Mohammad wrote:
i am using exten =
_XXX.,n,Set(CDR(cli_name)=${CHANNEL(peerip)}) cli_name is field
in mysql and it work fine. show me cli output without AGI.
On Thu, Apr 11, 2013 at 6:41 PM, Carlos Chavez
cur...@telecomabmex.com mailto:cur...@telecomabmex.com wrote:
On 4/11/13 11:18 AM, Asghar Mohammad wrote:
hi, you have not assign any value to CDR(userfield). try code =
#111,self,SET(CDR(userfield)=111)
On Thu, Apr 11, 2013 at 12:53 AM, Carlos Chavez
cur...@telecomabmex.com mailto:cur...@telecomabmex.com
mailto:cur...@telecomabmex.com mailto:cur...@telecomabmex.com
wrote:
I am trying to set the CDR(userfield) to a certain vaule using
the application map of features.conf but I am not able to do it.
When I receive a call I would like to tag it with a client code
(3 digit numeric) so I can referenci it later from the CDR. I
have edited features.conf with something like:
code = #111,self,SET(CDR(userfield(111))
or
code = #111,self,AGI(code.agi)
The DYNAMIC_FEATURES variable is in the globals section and
includes the application map name. When I do a features reload
I can see everything loads and when I dial the code during a call
I can see a message like:
-- Feature Found: code exten: code
The problem is that my CDR variable is not being written to. The
first example does not show anything on screen. For the second
when I turn agi debug on I can see:
SIP/2001-0003AGI Rx SET VARIABLE CDR(userfield) 111
But when I hang up neither my h extension or the CDR itself will
show the value I set, it is empty. I do not know what I am
doing wrong or maybe CDR variables are not available from
features?
That was a copy/paste error on my part. The line is as you put it
but I cannot get the value after.
--
_
- -- 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
- --
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
-BEGIN PGP SIGNATURE-
Version: GnuPG/MacGPG2 v2.0.18 (Darwin)
Comment: GPGTools - http://gpgtools.org
Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/
iEYEARECAAYFAlFm7e8ACgkQqmNh+MyHzx6POwCeLtZtIH42LgTPE/N0/l7kpfDP
XpkAnRqtgX6iFhaGzn29B+rjFhXd6tIv
=VW+3
-END PGP SIGNATURE-
--
_
-- 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] Setting a CDR field from using feature codes...
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Right now it is a simple call between 2 extensions. The receiving
extension dials the code. The 3rd line of my h extension is a
Noop(${CRD(userfield)})
pbxoficina*CLI features reload
== Parsing '/etc/asterisk/features.conf': == Found
== Registered Feature 'cita1'
== Mapping Feature 'cita1' to app 'SET(CDR(userfield)=111)' with
code '#111'
== Registered Feature 'cita2'
== Mapping Feature 'cita2' to app 'Noop(${CDR(src)})' with code '#112'
== Registered Feature 'cita3'
== Mapping Feature 'cita3' to app 'AGI(pin.agi,113)' with code '#113'
== Registered group 'cita'
== Registered feature 'cita1' for group 'cita' at exten '#111'
== Registered feature 'cita2' for group 'cita' at exten '#112'
== Registered feature 'cita3' for group 'cita' at exten '#113'
-- Added extension '700' priority 1 to parkedcalls
-- Added extension '701' priority -1 to parkedcalls
-- Added extension '702' priority -1 to parkedcalls
-- Added extension '703' priority -1 to parkedcalls
-- Added extension '704' priority -1 to parkedcalls
-- Added extension '705' priority -1 to parkedcalls
-- Added extension '706' priority -1 to parkedcalls
-- Added extension '707' priority -1 to parkedcalls
-- Added extension '708' priority -1 to parkedcalls
-- Added extension '709' priority -1 to parkedcalls
-- Added extension '710' priority -1 to parkedcalls
-- Added extension '711' priority -1 to parkedcalls
-- Added extension '712' priority -1 to parkedcalls
-- Added extension '713' priority -1 to parkedcalls
-- Added extension '714' priority -1 to parkedcalls
-- Added extension '715' priority -1 to parkedcalls
-- Added extension '716' priority -1 to parkedcalls
-- Added extension '717' priority -1 to parkedcalls
-- Added extension '718' priority -1 to parkedcalls
-- Added extension '719' priority -1 to parkedcalls
-- Added extension '720' priority -1 to parkedcalls
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [2001@oficina:1] Macro(SIP/2003-000e,
stdexten,2001,SIP/2001) in new stack
-- Executing [s@macro-stdexten:1] NoOp(SIP/2003-000e,
LLamada a extension estandar 2001) in new stack
-- Executing [s@macro-stdexten:2] NoOp(SIP/2003-000e,
LLamada desde: Carlos Chavez 2003) in new stack
-- Executing [s@macro-stdexten:3] GotoIf(SIP/2003-000e,
0?UNAVAIL) in new stack
-- Executing [s@macro-stdexten:4] GotoIf(SIP/2003-000e,
0?DESVIO) in new stack
-- Executing [s@macro-stdexten:5] GotoIf(SIP/2003-000e,
0?FOLLOWME) in new stack
-- Executing [s@macro-stdexten:6] Dial(SIP/2003-000e,
SIP/2001,25,tWw) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/2001
== Extension Changed 2001[hints] new state Ringing for Notify User 4000
-- SIP/2001-000f is ringing
-- SIP/2001-000f answered SIP/2003-000e
== Extension Changed 2001[hints] new state InUse for Notify User 4000
[Apr 11 11:56:44] WARNING[5184]: translate.c:206 framein: no samples
for ulawtolin
-- Launched AGI Script /var/lib/asterisk/agi-bin/pin.agi
SIP/2003-000eAGI Tx agi_request: pin.agi
SIP/2003-000eAGI Tx agi_channel: SIP/2003-000e
SIP/2003-000eAGI Tx agi_language: en
SIP/2003-000eAGI Tx agi_type: SIP
SIP/2003-000eAGI Tx agi_uniqueid: 1365699403.18
SIP/2003-000eAGI Tx agi_version: 1.8.15.0
SIP/2003-000eAGI Tx agi_callerid: 2003
SIP/2003-000eAGI Tx agi_calleridname: Carlos Chavez
SIP/2003-000eAGI Tx agi_callingpres: 0
SIP/2003-000eAGI Tx agi_callingani2: 0
SIP/2003-000eAGI Tx agi_callington: 0
SIP/2003-000eAGI Tx agi_callingtns: 0
SIP/2003-000eAGI Tx agi_dnid: 2001
SIP/2003-000eAGI Tx agi_rdnis: unknown
SIP/2003-000eAGI Tx agi_context: macro-stdexten
SIP/2003-000eAGI Tx agi_extension: s
SIP/2003-000eAGI Tx agi_priority: 6
SIP/2003-000eAGI Tx agi_enhanced: 0.0
SIP/2003-000eAGI Tx agi_accountcode: general
SIP/2003-000eAGI Tx agi_threadid: 139796748805888
SIP/2003-000eAGI Tx agi_arg_1: 113
SIP/2003-000eAGI Tx
SIP/2003-000eAGI Rx VERBOSE Codigo: 113 3
-- pin.agi,113: Codigo: 113
SIP/2003-000eAGI Tx 200 result=1
SIP/2003-000eAGI Rx SET VARIABLE CDR(userfield) 113
SIP/2003-000eAGI Tx 200 result=1
-- SIP/2003-000eAGI Script pin.agi completed, returning 0
-- Executing [h@oficina:1] NoOp(SIP/2003-000e, Colgar
llamada de 2003 en OFICINA) in new stack
-- Executing [h@oficina:2] NoOp(SIP/2003-000e, 2003) in
new stack
-- Executing [h@oficina:3] NoOp(SIP/2003-000e, ) in new stack
On 4/11/13 12:24 PM, Asghar Mohammad wrote:
how you are executing? show me your full context and how call enter
in context.
On Thu, Apr 11, 2013 at 7:07 PM, Carlos Chavez
cur...@telecomabmex.com mailto:cur...@telecomabmex.com wrote:
When I execute without using the
Re: [asterisk-users] Setting a CDR field from using feature codes...
you should set variable in extensions.conf not in features.conf
On Thu, Apr 11, 2013 at 7:34 PM, Carlos Chavez cur...@telecomabmex.comwrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Right now it is a simple call between 2 extensions. The receiving
extension dials the code. The 3rd line of my h extension is a
Noop(${CRD(userfield)})
pbxoficina*CLI features reload
== Parsing '/etc/asterisk/features.conf': == Found
== Registered Feature 'cita1'
== Mapping Feature 'cita1' to app 'SET(CDR(userfield)=111)' with
code '#111'
== Registered Feature 'cita2'
== Mapping Feature 'cita2' to app 'Noop(${CDR(src)})' with code '#112'
== Registered Feature 'cita3'
== Mapping Feature 'cita3' to app 'AGI(pin.agi,113)' with code '#113'
== Registered group 'cita'
== Registered feature 'cita1' for group 'cita' at exten '#111'
== Registered feature 'cita2' for group 'cita' at exten '#112'
== Registered feature 'cita3' for group 'cita' at exten '#113'
-- Added extension '700' priority 1 to parkedcalls
-- Added extension '701' priority -1 to parkedcalls
-- Added extension '702' priority -1 to parkedcalls
-- Added extension '703' priority -1 to parkedcalls
-- Added extension '704' priority -1 to parkedcalls
-- Added extension '705' priority -1 to parkedcalls
-- Added extension '706' priority -1 to parkedcalls
-- Added extension '707' priority -1 to parkedcalls
-- Added extension '708' priority -1 to parkedcalls
-- Added extension '709' priority -1 to parkedcalls
-- Added extension '710' priority -1 to parkedcalls
-- Added extension '711' priority -1 to parkedcalls
-- Added extension '712' priority -1 to parkedcalls
-- Added extension '713' priority -1 to parkedcalls
-- Added extension '714' priority -1 to parkedcalls
-- Added extension '715' priority -1 to parkedcalls
-- Added extension '716' priority -1 to parkedcalls
-- Added extension '717' priority -1 to parkedcalls
-- Added extension '718' priority -1 to parkedcalls
-- Added extension '719' priority -1 to parkedcalls
-- Added extension '720' priority -1 to parkedcalls
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [2001@oficina:1] Macro(SIP/2003-000e,
stdexten,2001,SIP/2001) in new stack
-- Executing [s@macro-stdexten:1] NoOp(SIP/2003-000e,
LLamada a extension estandar 2001) in new stack
-- Executing [s@macro-stdexten:2] NoOp(SIP/2003-000e,
LLamada desde: Carlos Chavez 2003) in new stack
-- Executing [s@macro-stdexten:3] GotoIf(SIP/2003-000e,
0?UNAVAIL) in new stack
-- Executing [s@macro-stdexten:4] GotoIf(SIP/2003-000e,
0?DESVIO) in new stack
-- Executing [s@macro-stdexten:5] GotoIf(SIP/2003-000e,
0?FOLLOWME) in new stack
-- Executing [s@macro-stdexten:6] Dial(SIP/2003-000e,
SIP/2001,25,tWw) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/2001
== Extension Changed 2001[hints] new state Ringing for Notify User 4000
-- SIP/2001-000f is ringing
-- SIP/2001-000f answered SIP/2003-000e
== Extension Changed 2001[hints] new state InUse for Notify User 4000
[Apr 11 11:56:44] WARNING[5184]: translate.c:206 framein: no samples
for ulawtolin
-- Launched AGI Script /var/lib/asterisk/agi-bin/pin.agi
SIP/2003-000eAGI Tx agi_request: pin.agi
SIP/2003-000eAGI Tx agi_channel: SIP/2003-000e
SIP/2003-000eAGI Tx agi_language: en
SIP/2003-000eAGI Tx agi_type: SIP
SIP/2003-000eAGI Tx agi_uniqueid: 1365699403.18
SIP/2003-000eAGI Tx agi_version: 1.8.15.0
SIP/2003-000eAGI Tx agi_callerid: 2003
SIP/2003-000eAGI Tx agi_calleridname: Carlos Chavez
SIP/2003-000eAGI Tx agi_callingpres: 0
SIP/2003-000eAGI Tx agi_callingani2: 0
SIP/2003-000eAGI Tx agi_callington: 0
SIP/2003-000eAGI Tx agi_callingtns: 0
SIP/2003-000eAGI Tx agi_dnid: 2001
SIP/2003-000eAGI Tx agi_rdnis: unknown
SIP/2003-000eAGI Tx agi_context: macro-stdexten
SIP/2003-000eAGI Tx agi_extension: s
SIP/2003-000eAGI Tx agi_priority: 6
SIP/2003-000eAGI Tx agi_enhanced: 0.0
SIP/2003-000eAGI Tx agi_accountcode: general
SIP/2003-000eAGI Tx agi_threadid: 139796748805888
SIP/2003-000eAGI Tx agi_arg_1: 113
SIP/2003-000eAGI Tx
SIP/2003-000eAGI Rx VERBOSE Codigo: 113 3
-- pin.agi,113: Codigo: 113
SIP/2003-000eAGI Tx 200 result=1
SIP/2003-000eAGI Rx SET VARIABLE CDR(userfield) 113
SIP/2003-000eAGI Tx 200 result=1
-- SIP/2003-000eAGI Script pin.agi completed, returning 0
-- Executing [h@oficina:1] NoOp(SIP/2003-000e, Colgar
llamada de 2003 en OFICINA) in new stack
-- Executing [h@oficina:2] NoOp(SIP/2003-000e, 2003) in
new stack
-- Executing [h@oficina:3] NoOp(SIP/2003-000e, ) in new stack
On 4/11/13 12:24 PM, Asghar
Re: [asterisk-users] PRI DEBUG
thanks, that command syntax works.
yves
Am 11.04.2013 18:51, schrieb Richard Mudgett:
- Original Message -
hi,
strange behaviour while trying to use pri debugging on asterisk 11.x
...
please take a look:
bas1104*CLI pri show version
libpri version: 1.4.13
bas1104*CLI dahdi show version
DAHDI Version: 2.6.1 Echo Canceller: HWEC
bas1104*CLI help pri
pri intense debug span no description available
pri service disable channel Remove a channel from service
pri service enable channel Return a channel to service
pri set debug {on|off |hex|inte Enables PRI debugging on a span
pri set debug file Sends PRI debug output to the specified file
pri show channels Displays PRI channel information
pri show debug Displays current PRI debug settings
pri show spans Displays PRI span information
pri show span Displays PRI span information
pri show version Displays libpri version
bas1104*CLI help dahdi
dahdi destroy channel Destroy a channel
dahdi restart Fully restart DAHDI channels
dahdi set dnd Sets/resets DND (Do Not Disturb) mode on a channel
dahdi set hwgain Set hardware gain on a channel
dahdi set swgain Set software gain on a channel
dahdi show cadences List cadences
dahdi show channels [group|con Show active DAHDI channels
dahdi show channel Show information on a channel
dahdi show status Show all DAHDI cards status
dahdi show version Show the DAHDI version in use
currently all debug off:
bas1104*CLI pri show debug
Span 1: Debug: No Intense: No
Span 2: Debug: No Intense: No
Span 3: Debug: No Intense: No
Span 4: Debug: No Intense: No
switching it on (which currently works as expected)
bas1104*CLI pri intense debug span 1
Enabled debugging on span 1
oops, still shows no debug but it IS activated...
It activated a different mode of debug than what you expected
because that command is an alias that was not updated.
bas1104*CLI pri show debug
Span 1: Debug: No Intense: No
Span 2: Debug: No Intense: No
Span 3: Debug: No Intense: No
Span 4: Debug: No Intense: No
huh... how to disable it again? on some machines I can do so with
pri no debug span nr but not here... gives same result (no
such command) and debug is still enabled...
bas1104*CLI pri set debug off
No such command 'pri set debug off' (type 'core show help pri set'
for other possible commands)
bas1104*CLI
so... whats the right way to disable pri debugging?
The correct command is pri set debug {on|off|intense} span x.
The pri intense debug span x command is an alias for
pri set debug 2 span x that didn't get updated when the real
command was changed to pri set debug intense span x.
This will show the help you need:
bas1104*CLI help pri set debug off span
Richard
--
_
-- 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
