similar to: Inbound/Outbound undesired behavior

Hi to All, I need to implement an automatic telephone messaging system that works like this: -the system generates the call based on mysql records or any database -when the client answer the phone, the Asterisk PBX playback a recorded message -when finish, hang up the channel. Only for voice messages not SMS. Exists some application based on Asterisk that makes this, or any code to

Hi to all, I actually have an asterisk server configured to write CDR mysql records in the same machine (localhost), but I want to write this records to another machine also in mysql at the same time, It is possible? It means that I want save the records in both machines. Thanks in advance. Ricardo Melendez -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello all, I have asterisk 1.4.26 working on CentOS 5.3 and I have the following problem: - when I restart asterisk all the members of the queue are Invalid. - when I make a call to one of the members, of the queue, and then check the state, it turns to "Not in use" for the called phone, and the queue works fine for that member after. - after doing a module reload of the

I'm curious what asterisk solutions there are out there for inbound call centers with multiple DIDs. I'm looking for solutions for a setup where single system may have 1000 DIDs going to it, one for each account. Each account may not get that many calls. Solutions that will all reporting on calls coming into different accounts, call routing for queues based on distribution groups. Like

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

Hi all, We have an 8 agent support desk setup with 2 call queues running Asterisk 1.4.5. Every so often agents will receive a call from the queue that only rings once not allowing them time to answer. The call doesn't seem to be dropped, just seems to go to voicemail. The agents are also mentioning they do not receive the 30 second wrapuptime I have specified in queues.conf. We're

Hi to All, I dont know much about PCI express slots in newer Servers, my doubt is if the Digium and Sangoma PCI express cards, are compatible with the x8 PCI express slots that come in the HP Proliant ML150 G5 server. Thanks in advance. Ricardo -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hello all, I am using R to perform certain calculations on huge amounts of data. In short I need a function that does the mode function, ie returns the most common element. I looked at the mode function in R but it seems to return the type of the data element you give it. Does such a method exist? I have tried googling this to no avail as all the results lead me back to the mode function I do

Hello list, hope someone can help me - I'm going crazy using the FXO port a SPA3012. I would like the SPA 3012 to act as a simple FXO port to an Asterisk, that is, once it detects a call, it should simply send it over to the local Asterisk server. No intelligent routing, PIN, anything else.... I configured it like this: PSTN-To-VoIP Gateway Setup PSTN-To-VoIP Gateway Enable: yes PSTN

Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in

Hello List, In PHPAGI, I'm using the Astrisk Manager function send_request() to originate an outbound call. I want to execute the remaining PHP code after the call gets executed (depending on user input). But presently the call originates in a different context and asterisk executes the remaining code in parallel. Is there a way in which I can pause the code execution until the call is

So I've got a bit of a head scratcher. Wanted to get some insight. I've got a PBX running 12.3.0 We're a ULAW shop from end to end. But I've been playing with G722 just for fun. I've got a Yealink T46G on my desk, And my colleague, A Polycom IP650 (Same office). Basically, Whenever I make an outbound call to a destination to something not G722 ready, I get no

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

Hi, I am using the following script to limit my outbound traffic. This scipt runs on a box behind my firewall. It limits my outbound passive ftp traffic to 39K perfectly....just like i want. However, i just noticed that it is also limiting uploads coming to my server. Is there something I can change to make it not limit uploads to my server? #!/bin/bash #shaping passive ftp traffic #

Greetings list- I have a box with a single FXO card in it. I'm able to dial out ZAP/1 with no problems and as expected. However, I would like inbound calls on that POTS line to go unanswered by Asterisk since I have other equipment on the line. I've setup zapata.conf for the channel without a context but the line is still answered. I've also setup a blank context with the same result.

Is there a way to specify the outbound leg Call-ID? -- Eric Chamberlain

I have a strange problem in that when I receive a call through the X100P which is forwarded to my budgetone 100 then the voice quality is perfect both directions. However, if I make a call out from the budgetone to the same caller via the X100P the sound level is a lot lower and the quality a lot poorer. I've had to set the rx tx gain to 1.5 or I can hardly hear at all. Any ideas what is

Hello I get this warning all the time when I am using iax2 for inbound calls or outbound. Aug 24 13:48:41 WARNING[-1105474640]: chan_iax2.c:4873 socket_read: Error: Resource temporarily unavailable I get the calls and the sound is fine. But the screen on the cli is full of these warnings and Error: What can I do to fix this. I get it when using calls to iaxtel, FWD, VoicePulse, Nufone and