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..."unload" and "load". I didn't have much luck looking through the archives or wiki on this, but I was wondering if anyone had any suggestions. Any feedback is appreciated. Thanks much, Franklin Webb Assistant IT Project Leader Inter Medi@ Marketing Solutions 610-701-9670 fwebb@imminc.com

...s anyone run into this issue any maybe have a solution, or does anyone know of a good way to get that call back onto the Asterisk switch from another extension prior to calling chanspy? Thanks much, Franklin Webb -- Franklin Webb Asst Project Manager Inter Medi@ Marketing Solutions 610-701-9670 fwebb at imminc.com

Greetings: I'm in the situation where I'm trying to splash information picked off by an asterisk IVR into a Cisco call center environment. I'm under the impression that the ONLY way to do this is to setup socket connections with the Cisco "voice processor", or CVP, and send packets corresponding to GED-125. Cisco has a detailed 100+-page document detailing the internals of

Greetings all, We are settng up a fair sized call center on Asterisk, but we are having some issues with our agents not knowing if they have logged in and logged out. Prior to beginning our migration to VoIP the agents logged into our nortel phones and confirmation was displayed on the phone. My question is has anyone out there done anything from Asterisk that can change the display on

Hi, I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is "cut". Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? my extension.conf for meetme: ;switch =>

Greetings fellow list members, I have what I think is a relatively simple question, but it did not appear to be addressed on the wiki. I am trying to setup a queue so that it plays an estimated holdtime announcement, but not a queue position announcement. Currently my dialplan does both, and while I know how to take out the estimated holdtime without affecting the queue position

Fellow list members, I have run into an issue where I encounter a delay at the beginning of a phone conversation when I make outgoing calls through Asterisk with an Aastra 9133i hardphone. This is most noticable when I call a voicemail system with the Aasta and then with a land line or other VoIP phone. The first word or two of the voicemail message is generally cut off. According to

Greetings fellow list members, I am trying to add some tricky functionality to Asterisk dialplan and I was curious if anyone else has come up with a solution to something like this. Basically I have phone representatives that log into one of several queues (not using chan Agent, we log in by the extension), and frequently these agents have to make attended transfer calls to outside numbers.