similar to: AGI Delimiter in 1.6

Hello list, what is the correct syntax ? exten => s,n,Queue(${queuename},,,,${timeout},cleanpickup.agi^${CHANNEL}) [Sep 13 10:23:58] WARNING[23551]: res_agi.c:886 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/cleanpickup.agi^SIP/329909007906-0000017a': File does not exist. Kind regards, Jonas. -------------- next part -------------- An HTML attachment was

I am running dovecot 2.2 changeset 16983:d925674c4329 on AIX. We use mbox files with the "/" delimter. I have an imap protocol question. When I do: 1 list directory1/ * 1 OK List completed. Is that the correct answer? If I do 1 list "" * it shows * LIST (\Noselect \HasChildren) "/" "directory1" I forget why I think that even with the delimiter, I

Hello everybody: This could be a stupid question, or may be not; I'm not sure 'cause I have not a very wide experience working with Asterisk, actually I just started last week. I need to make an IVR system work and I choose working with AGIs, written in Perl. The available documentation I've found show it as a very simple proccess, but it doesn't work for me... and I

Hi, Using 1.4.26.1 & DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] & same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? Thanks, --Barry

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

Hi, I noticed that when dial terminates it does not return to the dialplan, and therefore can not execute any entry after Dial(). Is there any trick to overcome this limitation ? How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if I can not execute anything after Dial()? I made a workaround with DeadAGI (below) but it is unreliable: if 2 calls end

Imagine that you have this code: exten => _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})})) If ${QueueName} happens to be unset, this will cause a warning: [Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187 queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an argument: queuename The obvious solution: exten => _X!,n,ExecIf($["${QueueName}" !=

Hi folks, I use safe asterisk to startup and run asterisk in the background. In Safe_asterisk script, there is a parameter (right at the top ), CONSOLE which I can set to no or something. If it is no asterisk startup as asterisk -vvvg , if it is set to something the asterisk startup as asterisk -vvvg -c. Now I am running an agi script when calls get hung-up. That is in my extensions.conf I call

Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: ---------------------------------- [CC] res_adsi.c -> res_adsi.o [LD] res_adsi.o -> res_adsi.so [CC] res_agi.c -> res_agi.o [LD] res_agi.o -> res_agi.so [CC] res_clioriginate.c -> res_clioriginate.o

I want to combine a selection of data from two tables and display it as list. Note that this isn''t a join. i''m looking for something like this: Table 1: Sheep Name, Age, Farmer, etc. (sheep specific columns) Table 2: Cows: Name, Age, Farmer, etc. (cow specific columns) For my display I nneed something like this: Farmer Jone''s Livestock: TYPE NAME AGE Cow

Dear all after an upgrade to 1.6 from 1.4 (as explained in the UPGRADE-1.6.txt file) the | delimiter is not working by default. I've added a compat section in asterisk.conf a [options] dontwarn = yes [compat] pbx_realtime=1.4 res_agi=1.4 app_set=1.4 And restarted Asterisk, but i still have problem to have the | delimiter working, [Jun 9 23:20:54] DEBUG[11744]: pbx.c:3122

It would appear that using :include on a model that has_many with an :order on another model makes it so that the order is ignored. Is this supposed to happen? Strikes me as odd behavior. For example class Farmer < ActiveRecord::Base has_many :cows, :order => ''position'' end class Cow < ActiveRecord::Base acts_as_list :scope => :farmer end If I do

Hello guys, I have these models class Farmer < ActiveRecord::Base has_many :cows end class Cow < ActiveRecord::Base set_table_name :cows belongs_to :farmer end ... graph = Farmer.find(x) # Retrieves four cows of x graph.cows[1].name = "Trottolina" graph.save while adding a new cow to collection works perfectly fine, ActiveRecord doesn''t

On 4/20/2023 7:16 AM, Rhys Hanrahan wrote: > > Hi All, > > I’m having issues figuring out how to set no DTMF timeout on the AGI > GET DATA command as “0” still has a multi-second timeout if no input > is given. I am trying to find the source code of the AGI command to > figure it out, but I can’t for the life of me find the underlying > source code. The closest I’ve

Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy

I have a need to have a GUID (for example, bcd47ccc-d7c9-ddb6-dc11-6746a770d77d [36 characters long including the "-"]) generated in the dialplan. Is there any asterisk function that would do this ? I would prefer not to have to shell out every time a call comes in. Julian

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I am running 1.6.0.15 and am trying to get IMAP storage working. I have had no trouble doing so, except that I wish to create a subfolder in my account for voicemail, such that I have: #voicemail/ INBOX Old Family Friends Work I can set IMAPFOLDER=#voicemail.INBOX in voicemail.conf and successfully get voicemail to work as expected.

While reading about how to maximize capabilities in asterisk i have read about SER and OpenSER. The sites do not explain to newbies (maybe that's on purpose) what are the benefits of using those products tied with asterisk (or is SER an asterisk replacement??) Can someone give me an idea of what's the usage for open(ser) and asterisk? is it for scalability? should I run it in the same

Hi, my Asterisk 1.2.9.1 suddenly freezed ("stop now" did not work!!) . I found the following errors inside /var/log/asterisk/message: Feb 14 14:55:41 ERROR[11273] rtp.c: Unable to allocate socket: Too many open files Feb 14 14:55:41 WARNING[11273] chan_sip.c: Unable to create RTP audio session: Too many open files Feb 14 14:55:43 WARNING[11273] acl.c: Cannot create socket Feb 14

Hello everyone, I have a problem with a FastAGI connection, could you help me fix this problem please? Here is my log: [2015-12-15 16:17:09] WARNING[23936][C-00000015]: res_agi.c:1658 handle_connection: Connecting to '10.171.54.149:9110' failed for url