search for: teledata

Hi all, I have an Asterisk server connected in a PABX (TELEDATA) by channel Unicall.. I`m having problem when somebody call from PABX to Asterisk.. Eg: When somebody dial 1234, I received 1111112222333333444444 in the Asterisk CLI... If somebody can help me... or already saw this... Everton Goularth Uberlandia - MG - Brazil ______________________...

Does anybody knows if it's posible or if there is some develoment in course to be able to use longer transmit packet sizes (as long as I know this is fixed in 20ms now) with the compressed voip codecs in asterisk (g729, g726, gsm, etc). I need to use asterisk to connect remote sip clients with 24kb bandwidth lines and I'm using a licences g729 codec but because I can't increase

Just by the way, do anybody knows if call pickup of a call to a sip extension from a mgcp phone is supposed to work (even if sip keeps ringing). The scenary is as follows: 3@mgcp02 (ext 136) calls sip/julia (ext 133) and after It starts ringing 2@mgcp02 (ext 135) dials *8. Nothing happens, only 135 gets congestion tone, 133 keeps ringing and in the asterisk console I get: --

Hello all We are running an Asterisk server using 24 extensions with an AB1 channel bank and a T1 digium card, plus a TDM400 with 4 fxo modules (using only modules 2,3,4). We have been using the pbx with this configuration for some time (asterisk 1.0rc1,1.0rc2). A few weeks ago we have installed a mirrored disk (raid 1) in this server and we have been having some problems with sporadic

Hello all We have tested for a mounth or two an asterisk PBX using one T1 channel bank with 24 fxs and one TDM400P digium card with 4 FXO modules. This worked with minor problems, the most notorious being some sporadic static noice or failure in the first FXO module on the wildcard. Now we have a client with 12 pstn lines and 48 extensions and we are trying to deploy an Asterisk PBX server

More than one hundred messages related to *8 or call pickup problem in last 6 months!! Please someone in the development team could clarify this and make himself responsible for the response. By now It seems a bad joke. We have spent thousand dollars with hardware, sip phones, working men hours, and with digium stuff (E1, fxo, fxs cards etc) and we have had the *8 problem (sip callee ringing

Hello all I found a bug in the chan_capi driver (really a not implemented message handling and then a false error condition) and I guess I have wrote a patch to fix it (basically I searched the internet for other capi open source implementation an borrowed the code snippet) but I don't know where to send the report and bugfix. I also found some miss-behaviours that I would like to share

Dear Mailinglist-User currently we`re working with an IP-PBX, based on Asterisk, with SIP, H.323 and ISDN-Capabilities. SIP and ISDN works fine, but H.323 not. In our first test, we started to connect Asterisk to an Cisco IOS-Gatekeeper with the "chan_oh323" (version 0.6.5). We successfully tested in/egress calls without any problems. But when we started to connect our Asterisk

Hello all, I'm having problems getting chan_capi 0.3.5 to work well with an Eicon Diva Server card using using the driver from linux kernel both 2.6.10 and 2.6.12 (vanilla versions). I have a (really two) producci?n system(s) running chan_capi in another identical Eicon Card using kernel 2.4.27 and the Diva Server drivers from Eicon. I installed an compiled the source level rpm