search for: siganlling

Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! ======================================== pbx1*CLI> zap

hi; i setup a Meetme conference room and i notice the following behavior: if A calls confroom over PSTN channel 1 B call confroom over PSTN channel 2 C calls confroom over SIP/Ethernet then i have all of them talking and the media stream mixed by asterisk. However, if i hang up A, channel 1 is still ocuppied (i try dialing inbound again on channel and it continues to give a busy siganl) any

Hello All, I have installed Asterisk 1.6 with openVPN in the same machine. I have set up a VPN connection between 2 SIP clients and Asterisk using x-lite. The 2 clients connects to Asterisk. SIP signaling goes ok over the vpn tunnel. When attempting to make a call between the clients, the siganling part of the call goes well. But, when the call is set up, some RTP packets are exchanged at

...39;s PBX. It is working but no names are exchanged. From the debug mode I see that the Cisco sends the display name (which does not appear on the Nortel's phones) and the Nortel does not bother to send it at all. I recall that when I had a pilot with Cisco CCM two years ago we had to set the siganlling to ESGF on the Nortel and use MGCP on Cisco (since MGCP protocol forces the Cisco to use ESGF signalling). We could not use H.323 as it forces the Cisco to use ISGF. I suspect that SIP is the same, but setting ISGF signalling on Nortel doesn't help. Anyone had some luck with this configurati...

Hi. I have a x100p card installed on my asterisk box... my zapata.conf file includes the following lines: [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 echocancel=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 Basically, the zapata.conf file generated by make samples. Then in my extensions.conf I have this: [default] include => demo And demo is

https://bugzilla.samba.org/show_bug.cgi?id=4132 Summary: Does not delete partial files upon completion of transfer Product: rsync Version: 2.6.8 Platform: Other OS/Version: All Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org

I have a TDM400 with an FXS and FXO interface. I have adjusted the TX and RX gain for both the FXS and FXO channels so that everything is balanced. I hear myself very loudly with no lag. I also get alot of background noise when the bridge is formed between the FXS and FXO channels. I have read the following articles: http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation