joe,
when I have problems with audio and other connections seem to work, I always look for a codec incompatibility... use 'sip set debug peer <extension>' and look for the codec handshaking... make sure both extensions have a compatible codec choice...
daveC

Using INVITE request as basis request - [EMAIL PROTECTED]
Found user '401'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP video format 99
Peer audio RTP is at port 192.168.15.100:5004

*Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format GSM for ID 3
Found description format H264 for ID 99

*Capabilities: us - 0x20000e (gsm|ulaw|alaw|h264), peer - audio=0x20000e (gsm|ulaw|alaw|h264)/video=0x200000 (h264), combined - 0x20000e (gsm|ulaw|alaw|h264)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.15.100:5004
Peer video RTP is at port 192.168.15.100:5006
Looking for 404 in inbound-video (domain sip3701.ibsonecall.com)
list_route: hop: <sip:[EMAIL PROTECTED]:5060;user=phone>



Joe Acquisto wrote:
Steve Totaro <[EMAIL PROTECTED]> Wrote: 4/4/2007 8:44 PM:
Joe Acquisto wrote:
Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite softphones, for eval/testing. They do get registered, and can call each other, but mostly get no audio, sometimes one way audio.

Suggestions/fixes?

joe a.
Is there NAT on both sides? Are you using qualify? Paint a clearer picture.



Sorry, I missed your reply, till now.

------------------switch
     |      |     |----phones
     |      |---------asterisk box
     
|---------------IPcop------------|---internet-----|-----home/remote-office----|----sip
 phone
                                                                            
|-----ditto

Hope that is intelligible.

joe a

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