Re: [Asterisk-Users] incominglimit stuck in app_queue

I've seen this same thing. But it doesn't happen only for phones using the queue I believe it is a bug in the chan_sip driver. What I have found is that when a phone sip phone is unplugged/not registered and a call comes in it increments the counter and doesn't reset the counter when the phone rere

[Asterisk-Users] Dialing long-distance locally

nd not a completed call. This was his best guess at to what was happening. How is this normally handled by Asterisk? -- Any help is much appreciated Paul Lambert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] frames/packet

I'm thinking about SIP as well. Martin Pycko wrote: > > Unless you use IAX2 trunking that would limit 32 calls to 50 packets/sec. > > Martin > > On Wed, 3 Sep 2003, Paul Lambert wrote: > > > "Not yet." implies that it is coming. I know it would help

Re: [Asterisk-Users] frames/packet

Steve Underwood wrote: > > Paul Lambert wrote: > > >"Not yet." implies that it is coming. > > > Look at the latency it causes, and you will see its not that useful. > > >I know it would help on Internet > >connections such as fixed wireless and c

Re: [Asterisk-Users] IAX and frames/packet

Thanks, you guys know your stuff! -- Paul Paul Lambert wrote: > > When multiple calls are in session between two IAX servers do the voice > frames from the various calls get put into a single packet to conserve > on total packet rate? > ___

[Asterisk-Users] IAX and frames/packet

When multiple calls are in session between two IAX servers do the voice frames from the various calls get put into a single packet to conserve on total packet rate? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

Re: [Asterisk-Users] frames/packet

always sends 20 ms of voice data per packet. > > regards > Martin > > On Wed, 3 Sep 2003, Paul Lambert wrote: > > > Noticed that I can adjust the number if frames/packet on the GrandStream > > phone. Can * do the same? > > __

Re: [Asterisk-Users] One way voice through NAT

Thanks, that worked. "WipeOut ." wrote: > > > I'm connecting and can place calls to and from my SIP phone that is > > behind a firewall, can hear audio from the SIP on the PSTN line but > > can't hear audio on the SIP phone from the PSTN line. Anyone else > > experience this? > > Try adding "nat

[Asterisk-Users] One way voice through NAT

I'm connecting and can place calls to and from my SIP phone that is behind a firewall, can hear audio from the SIP on the PSTN line but can't hear audio on the SIP phone from the PSTN line. Anyone else experience this? ___ Asterisk-Users mailing list [EMA

[Asterisk-Users] frames/packet

Noticed that I can adjust the number if frames/packet on the GrandStream phone. Can * do the same? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] STUN server from Vovida

Not sure if it's alright to talk about this here??? compiled the STUN server from Vovida on RedHat 7.3. Looks simple to configure. It isn't starting...it tries to for a long time and then just craps out. Here is my config:/etc/sysconfig/stund #!/bin/echo Not to execute. # Path to stund STUND=/usr

Re: [Asterisk-Users] Codec?

Thanks, I didn't realize that was the code for the codec. So, How do I know what codec that code translates to? -- Paul "WipeOut ." wrote: > At the console while a call is in progress run "sip show channels" and look in the > format column.. > > > How can I tell what codec a SIP session is usin

[Asterisk-Users] Codec?

How can I tell what codec a SIP session is using? -- Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users