joe,
look for the codec negociation... I have a similar problem where the
endpoints could not agree on the codec and thus no audio went through.
in 1.4.X
CLI> sip set debug peer <extension>
yields,
Audio is at 10.10.15.15 port 15342
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Reliably Transmitting (no NAT) to 10.10.15.219:5060:
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
make sure both endpoints have at least one codec that is the same...
if not, adjust your sip.conf for both endpoints.
daveC
Joe Acquisto wrote:
Hi.
Is there a way to isolate what shows on CLI to just the conversation with that
extension? There appears to be a lot of stuff unrelated to this extension.
Packet traces are not out of the question, but cannot be done today.
joe a.
"Yossi Ben Hagai" <[EMAIL PROTECTED]> Wrote: 4/9/2007 12:56 PM:
Hi Joe,
The debug trace you've enclosed is a NOTIFY message sent from * for the
message waiting feature - and is not related to the call.
You can however tell that something is wrong since the message is being
retransmitted since the server didn't receive 200 OK in reply - while it
could be due to the client being offline or not supporting this feature
It
could imply a NAT issue so try to recheck your NAT configs.
can you post a full trace (starting with the INVITE message)? also you
can
try to run a sniffer trace on the client side to see if it
receives/sends
the messages correctly.
Joss.
On 4/9/07, Joe Acquisto <[EMAIL PROTECTED]> wrote:
I never get this far, apparently. While the connection seems to be made,
and calls can be "completed" (rings, answers) there is no audio. On CLI, I
can see what appears to be call being made and connected. These are x-lite
phones (for testing, one hopes) there appears to be no codec selection
available.
I see no CODEC dialog. What I see is six iterations of the below:
. . . .
---
Retransmitting #6 (NAT) to xx.xx.xx.xx:64909:
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK363305c9;rport
From: "nnnnn"<sip:[EMAIL PROTECTED];tag=as67e5c857
To: "nnnnn"<sip:[EMAIL PROTECTED]>;tag=9c58a77e
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: MjY1NTgyYzQ0YTdjNmM3NTJkODE2ODM1ZmNhMWE3OGE.
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: terminated;reason=timeout
Content-Length: 0
-----
Does this imply anyting to anyone?
Call can be made, after this.
joe a.
******
dave cantera <[EMAIL PROTECTED]> Wrote: 4/7/2007 3:53 PM:
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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