search for: converstaion

Displaying 5 results from an estimated 5 matches for "converstaion".

Did you mean: conversation
2003 Sep 22
1
Switch between calls without initiating a threeway converstaion
I was just wondering if there was a way that you could have two calls on one line and switch between the two without initiating a threeway conversation? I would imagine that Flash is the way to do this, but when I Flash twice, a 3-way call is initiated. If I turn threeway off, then I can't transfer. Also, is it possible to hang up one of the calls, and then continue talking to the second
2006 Nov 03
4
Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is?
...realize that other parth has started saying 'hello, hello' because they can't hear you. But you are hearing them loud and clear. Now you are on static IP with dyndns FQDN. externip and localnet settings in sip.conf (do we need them for static IP?). After about 15-20 seconds, again 2-way converstaion is established again. IAX trunk, SIP extension, no NAT. 3. Conversation goes one way for 15-20 sec during the most important part of the conversation (Murphy's Law). You are on a static IP with no dyndns enrty. Trunk is ZAP on PRI, extensions SIP. NAT present but router properly configures for...
2005 Jul 14
1
Wire Tapping on Asterisk
I'm new to asterisk. I would like to ask if there's a feature in asterisk wherein you can monitor ongoing calls, some kinda like tapping into active phone calls? It must have this feature but I do not know where to get some reference to set this up or test this. Can anyone share me some sites as reference? thnx...
2006 Nov 06
1
Audio goes one way during the call for a fewseconds. Is it RTP, NAT, dyndns, or what it is?
...hello, hello' because they > can't hear > > you. But you are hearing them loud and clear. Now you are > on static IP with dyndns FQDN. > > externip and localnet settings in sip.conf (do we need them > for static IP?). > > After about 15-20 seconds, again 2-way converstaion is > established again. > > IAX trunk, SIP extension, no NAT. > > > > 3. Conversation goes one way for 15-20 sec during the most > important > > part of the conversation (Murphy's Law). You are on a > static IP with > > no dyndns enrty. Trunk is ZAP o...
2004 Jul 07
8
Voicemail volume
...lo, When I listen to a voicemail message, the recorded message is played back at extremely low volume. All the supporting prompts are at the correct volume, it's just the incoming recorded message that is played back almost inaudibly quiet. There's no problem with the volume during normal converstaions so I'm thinking this must be specific to the Voicemail application. I've tried changing the recording format from wav to gsm, but that didn't help. I've also tried adjusting rxgain and txgain in zapata.conf, but that doesn't improve things either. Help! Thanks Bob