similar to: Asterisk Queue

We've been using RNK Telecom for terminations for our SIP service, but their billing is questionable, they've been in breach of contract multiple times, and when I brought it to their attention, Russ Man, our 'friendly' account manager told me, and I quote "If you are having that much of a problem..please find another carrier." With customer service like that, I've

Hi folks, I know this isn't an Asterisk question, but I'm really desperate and wondering if someone could help me. I apologise for the off-topic post. Cisco phones connected to CallManager can forward calls. But when they do, CallManager conserves the originating caller's ANI in the new leg that is built. I cannot find a way to get it to rewrite the ANI to be that of the phone.

Or, in full: [Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The device state of this queue member, SIP/612, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. So, what do I check in UPGRADE.txt? This is with Asterisk 1.4.11

[Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device state of this queue member, SIP/3617001000, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. [Cary Fitch] We are running 1.4.22 and this message popped up in console. It could be causing our Queues announcement problem, because if all members

Martin Pycko <martinp@digium.com> writes: > sure, use the 'n' option of the queue and put voicemail app as the next > priority Will that work? From my read of the code, the timeout parameter is only checked while the call is being sent to an agent's phone (inside the try_calling function). The timeout doesn't seem to be checked while the user is waiting to get to

Hi, I'm a little surprised, up until 1.4.22 my agents where using an IAX channel to ZoIPer Softphone, however since after the upgrade to .22 we experienced a problem with hangup failure between zoiper and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i made them switch to SIP instead. Weird thing is that the 'Not In Use' warning message keep showing

When working with call queues, if an agent is logged in via AgentCallbackLogin and the extension they are registered at becomes "unavailable" (from a bad connection, or something of the like), calls routed to that extension seemed to be accepted by it, so if the next action for that extension is to go to voicemail, the caller in the queue is sent to the extensions voicemail. Even worse,

Hi, I'm using the following setup : Alice ---- IPPhone ------<LAN>----- Media gateway ----<PSTN> ------- Phone ---- Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably,

Hello list, asterisk 1.4.30 2 situations in which call-limit should work, but it does not : [Jul 8 09:15:49] WARNING[11132]: app_queue.c:3272 try_calling: The device state of this queue member, test12, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. In sip.conf I have : limitonpeer = yes In my realtime sip_buddies

On the wiki it say's that if you use the Goto commands you need to replace ',' with '|' in the app data field. But in the examples it uses '|' in place of ',' in the Dial command also in a couple of places. Is it safe to replace ',' with '|' everywhere in the app data field when using realtime? Or should I still to substituting ','

X101P with Ambient md3200 chip on it, with the zaptel wcfxo driver.... Just an indication of how many people have got this to work would be useful. Cheers Mark.

How "real time" is realtime? If the extensions.conf is stored in the database, does * query it row by row or is it "cached"? In other words, given the following exerpt: exten => 5001,1,Dial(IAX2/test@test/s,30,g) exten => 5001,2,Voicemail(u5001) exten => 5001,102,Voicemail(b5001) exten => 5001,103,Hangup exten => 5002,1,Dial(IAX2/test2@test2/s,30) exten =>

2023-11-08 18:14:13] ERROR[571246][C-000017e2] : Got 19 backtrace records # 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed() # 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref() # 2: [0x58e660] asterisk stasis_cache.c:824 update_create() # 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec() # 4: [0x586b90] asterisk stasis.c:1380 dispatch_message() # 5: [inlined] asterisk

Dear all, I am trying to prepare some figures where I'd like the color of the plot region (i.e., the area bounded by the axes) to be different from the margin area (i.e., the area of the figure not within the axes region). I can use "bg" but that changes the background of everything? How can I accomplish what I want? Thanks, Ramon -- Ramón Díaz-Uriarte Triana 47 28016 Madrid

HI I bought 20 license from Digium and install in my server and b'coz of some problem I've to change my server is it possible that I can use those lice and register again in my new server ? Is it possible that I'll be able to use those lice in my old box also ? thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I've two boxes connected over IAX2 trunk before IAX I was using SIP trunk and they were working fine b'coz of bandwidth issue I changed from SIP to IAX now I'm facing a strange problem after some time on the cli of my asterisk box I see lots of messages of IAX2 trunk and b'coz of that my agents are not able to hear any thing and I've restart my * box. Please guide me what I

Hi, My laptop runs a local IMAP server so that I'm immune from breaking my mail client, and when playing around with powertop I discovered that Dovecot (in particular imap-login) wakes up every 1000ms and then every 50ms even when no clients are connected. A bit of stracing shows this: gettimeofday({1180875699, 667917}, NULL) = 0 <0.000010> poll([{fd=7,

Hello everybody, i'm new en icecast, in fact, i don't know nothing about this software. I'm Nicaraguan, in Central America. This Radio http://www.radio-corporacion.com/principal.php is a local radio. They use icecast from http://radioneer.com:8750/, but I don?t know what to do for add my radio http://www.radiocamoapa.com to http://radioneer.com:8750/ and i can use its streaming

Hi, After playing around with queues a bunch on 1.4.26.2, I noticed a few things, which the patch below addresses. It addresses: - Callers in position 0 will hear periodic/position announcements at a very different rate than all other callers. -- Announcements while in position 0 could be delayed up to "timeout+retry" seconds. -- This patch reduces that possible delay to only

I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. #> thread apply all bt ........ ........ Thread 6 (process 20135): #0