similar to: app_queue bug with call transfer

Anyone have any example scripts in PHP that connect to the manager? I'm not really a much of a programmer so I could use boost. Once I can figure out how to get it to login properly, I'll be ok from there. Thanks, Kevin _____________________________________________________________ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com

Hi. There is not much info on the FWD site about this. What codecs do they use? When I try to connect with X-Lite, it works with GSM. When I try to call out with *, it wants G729. I have disallow=all and allow=gsm in the sip.conf. I end up getting errors: Unable to find a path from G729A to GSM Unable to find a path from GSM to G729A What's up with that? I was able to make a call once

Hi. I just tried updating asterisk and I guess I broke something. Here's the log: Unable to find a path from G729A to SLINR Unable to find a path from ULAW to G729A Any ideas on what I should try? I tried nuking all the zaptel stuff in the system and the source and started over agian. Also nuked the asterisk config files.... I saw this asked once before but there was no reply :-/

Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: debian:~# sphinx2-simple2 sphinx2-simple: Demo CMU Sphinx2

All, I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Regards, Hans -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may

Sorry, this is a re-post. I sent the message from the wrong e-mail address. I was checking up on the cvs changes. Hi. I just tried setting up the send url function of Queue app, but it did not seem to work with gnophone. I must have missed something. It never tries to go to the page. I have this in extensions.conf: exten => 1,3,Queue(support,t,http://www.yahoo.com) It shows in the

Hello everyone, I have done my reading & research and everything I try is coming to different errors, so I am going to beg & pray someone here can help me with my problem. I appericiate any help in advance! I am running Samba 2.2.11 on a RedHat Linux 7.3 Server, connected to a network of Windows 2000 & XP Machines. I want to configure Samba to be the Domain Controller for my other

iaxComm is a cross-platform IAX2 softphone available for Win32 and Linux. Win32 and Linux binaries as well as the LGPL source are available at: http://iaxclient.sourceforge.net Recent improvements are a less cluttered user interface, audible ringback and audible outgoing ring, and of course IAX2 protocol support. iaxComm is based upon the wxWindow GUI framework and compiles on Microsoft

Hello, Anyone aware of any CRM products projects that intagrete with *? Or that integrate with any telephony products? Is there some open API for such integration, or are they all proprietory? Thanks -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

I would like to release prepaid application. But I have a small problem, we are using their Cisco prompts (nice lady voice) And I do not know if it is ok to release it. Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/db22d880/attachment.htm

Is there a way to clear the Queue stats? That is with out restarting *? I'd like to reset them daily and don't see a way to do that. Unless the only way is maybe a cron restart asterisk like every weekday @ 04:00?

Hi, I am using SJPhone here for testing ivr with Asterisk. I haven't seen any problem here yet. I have tried different things for that and finally got it working. I am not an expert to explain more about that, but here is the general section form my sip.conf. dont know whether it will help... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ;

I've searched through the lists, but can't seem to find a reference to someone that said they had a time and temperature application for *. Voip-info wiki would be a good place for it as a Festival example!!! Can someone remind me where it's at? TIA. Tim Thompson

I noticed a few issues with app_queue just wanted to know if its sip related or ata186 related: Ext 111 and Ext 112 are dynamically loged into the queue via AddQueueMember. Call hits queue with fewestcalls routing. Rings ext 111 if 111 doesn't answer. It rings ext 112. If for some reason ext 112 doesn't answer it rings back to 111. Again at this point ext 111 isn't answered it

Anyone seen this? It ocurred on a 'reload app_queue.so' command. Asterisk version is 1.2.9.1. Tried again, but it was not immediately reproducable. Doug. (gdb) bt #0 reload_queues () at app_queue.c:3339 #1 0xb778a7a8 in reload () at app_queue.c:4012 #2 0x0805bb44 in ast_module_reload (name=0x8137cc7 "app_queue.so") at loader.c:257 #3 0x08092b3f in handle_reload (fd=33,

Hey guys, i'm wondering if there is any good way to get app_queue working in real roundrobin strategy. The idea is to specify a call list of, lets say, 3 agants. Those agents should always be called in the correct defined order. So all calls have to get the following agent priority: 1st Agent -> 2nd Agent -> 3rd Agent I've actually solved that by defining penelty for the accounts,

The new app_groupcount looks great for most applications but it a is a step back for call queueing... since app_queue calls physical interfaces and not extensions, app_groupcont can't be used to limit the calls passed to a dynamically added agent. I presently use the broken sip incominglimit feature (even though it's less than ideal as it also limits outgoing calls preventing

hi, our customer reports problem when 2 agents answer the call in the same time faster operator (device) answer the call, but the second is showed up (on device) and call is without sound asterisk 13.9/app_queue with strategy ringall/operators via Local channel with sip device (chan_sip) do you have any tips/info before i will dig deep into logs/debug? checked google&issues.asterisk.org

First of all I would like to thank Mark for getting roundrobin to go roundrobin. Good job. Now we have some options here for leastrecent and fewestcalls strategy. It needs some work on the logic and Mark recommend that I ask the list and get some input before he makes any changes to it. fewestcalls from what I have seen would always ring the agent with the fewestcalls first then go into

Hi, This is the output from queue show 28: 47 (DAHDI/g0/12345678) (realtime) (Ringing) has taken no calls yet Why is the devicestate "Ringing" when no channels is calling this number, and the queue says "has taken no calls yet"? Is it picking up the general state of a random channel on g0 in dahdi? Or what is happening? It only seems to happen with this particular