Re: [asterisk-users] SIP Dial Command to a non-Asterisk url

2007-05-30 Thread Gavin Henry
This is what is shown when the call connects with: sip show channel The conference suite from another provider on internal IP is waiting for an ACK on port 5605, but * is sending it back to port 2289 Internal between Asterisk and another Conference suite: * SIP Call Direction:

Re: [asterisk-users] SIP Dial Command to a non-Asterisk url

2007-05-26 Thread Gavin Henry
On 23/05/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: Does the non-Asterisk server _answer_ the line? :) Hi, sorry. I have been away on site doing 8 work ;-) Yes, it does. We've done a packet trace and it appears that * sends an ACK back on the wrong port, i.e. not 5605 like a

Re: [asterisk-users] SIP Dial Command to a non-Asterisk url

2007-05-26 Thread Gavin Henry
On 23/05/07, Nick Seraphin [EMAIL PROTECTED] wrote: The 2 most common problems I've seen for no audio in one or both directions is usually either a firewall (which you already said you don't have) or a CODEC problem. Make sure both sides are negotiating the same CODEC. I've often seen

[asterisk-users] SIP Dial Command to a non-Asterisk url

2007-05-23 Thread Gavin Henry
Dear All, I have a tiny dial plan like: [testing] exten = 454,s,Ringing() exten = 454,n,Wait(4) exten = 454,n,Dial(SIP/[EMAIL PROTECTED]:5605,10) exten = 454,n,Hangup This connects fine when I dial 454 from any extension in my system, but there is never any audio? Where can I start to look

Re: [asterisk-users] SIP Dial Command to a non-Asterisk url

2007-05-23 Thread Alex Balashov
Gavin, Does the Asterisk server's route to 192.168.45.18 traverse a firewall or router that may be blocking non-SIP ports that are dynamically allocated? SDP -- part of the SIP INVITE transaction payload -- negotiates arbitrary ports between the two endpoints for actually passing media.

Re: [asterisk-users] SIP Dial Command to a non-Asterisk url

2007-05-23 Thread Gavin Henry
On 23/05/07, Alex Balashov [EMAIL PROTECTED] wrote: Gavin, Hi. Does the Asterisk server's route to 192.168.45.183 traverse a firewall or router that may be blocking non-SIP ports that are dynamically allocated? Nope, all internal. SDP -- part of the SIP INVITE transaction

Re: [asterisk-users] SIP Dial Command to a non-Asterisk url

2007-05-23 Thread Mojo with Horan Company, LLC
Does the non-Asterisk server _answer_ the line? :) Gavin Henry wrote: Dear All, I have a tiny dial plan like: [testing] exten = 454,s,Ringing() exten = 454,n,Wait(4) exten = 454,n,Dial(SIP/[EMAIL PROTECTED]:5605,10) exten = 454,n,Hangup This connects fine when I dial 454 from any extension

Re: [asterisk-users] SIP Dial Command to a non-Asterisk url

2007-05-23 Thread Nick Seraphin
The 2 most common problems I've seen for no audio in one or both directions is usually either a firewall (which you already said you don't have) or a CODEC problem. Make sure both sides are negotiating the same CODEC. I've often seen situations where something like the Asterisk server will