I'm using ulaw, but seeing this problem as well.
Are you using CVS? I would swear it didn't do this to me in earlier tests, but
it is doing it now. I will try to track down the specific change tonight ...
My solution for now is to Answer() the call before dialing out. I changed all
of my outbound dialing rules from:
[trunklocal]
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
To:
[trunklocal]
exten => _9NXXXXXX,1,Answer
exten => _9NXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
This seems to fix it, and I haven't identified any side effects.
I need to do this anyway to workaround an early-media problem I have.
Does it work for you after this change?
-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Herrick
Sent: Saturday, April 30, 2005 8:49 AM
To: [email protected]
Subject: [Asterisk-Users] Polycom IP500 Forward problem codec issue
Polycom IP500 Forward problem codec issue
All,
I’m running the Polycom IP500 phones at several sites. My * server is
at a collocation site and I have complete control of the T1’s running to the
remote sites with the IP500 phones. Connectivity to the PSTN is
through a Cisco 2600 with a PRI card. During initial testing I ran
G711/ulaw but have added G729 licenses to the system.
Problem: When the forwarding function on the Polycom phones is enabled the
forward/transfer does work but the caller does not hear any ringing.
During the time that the caller should hear ringing the * console produces
pages of errors.
<snip>
…..
Apr 30 08:41:03 NOTICE[2813]: channel.c:1314 ast_read: Dropping incompatible
voice frame on Local/[EMAIL PROTECTED],2 of format g729 since our native format
has changed to ulaw Apr 30 08:41:03 NOTICE[2813]: channel.c:1314 ast_read:
Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format g729
since our native format has changed to ulaw …..
</snip>
I have tested this with the phones behind a PIX firewall with NAT, behind a PIX
firewall without NAT, and without a firewall at all. Nat is not the problem.
In the SIP.conf canreinvite=no so all traffic should be passing through the *
server.
The problem seems to be in the translation of the G729 packets from the
phone to the G711 packets to the router. Only during the forwarding
process is this a problem.
Here is a snip from the console when it worked.
(Note: it worked because I was ringing two phones with this line in my
extensions.conf (exten => ------6081,1,Dial(SIP/------6081&SIP/------6091,20)
=========<SNIP>
-- Executing Goto("SIP/---.----.241.35-40400490", "TPN|------6081|1") in new
stack
-- Goto (TPN,------6081,1)
-- Executing Dial("SIP/---.---.241.35-40400490",
"SIP/------6081&SIP/------6091|20") in new stack
-- Called ------6081
-- Called ------6091
-- Got SIP response 302 "Moved Temporarily" back from ------.92.27
-- Now forwarding SIP/---.---.---.35-40400490 to 'Local/[EMAIL PROTECTED]'
(thanks toSIP/------6091-6268)
-- Executing Dial("Local/[EMAIL PROTECTED],2",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/------6081-e558 is ringing
-- SIP/---.---.241.35-f522 is making progress passing it to
Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 is making progress passing it to
SIP/---.---.241.35-40400490
-- SIP/---.---.241.35-f522 answered Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 answered SIP/---.---.---.35-40400490
== Spawn extension (TPN, ------6081, 1) exited non-zero on 'Local/[EMAIL
PROTECTED],2<ZOMBIE>'
-- Attempting native bridge of SIP/---.---.241.35-40400490 and
SIP/---.---.241.35-f522
==========</SNIP>
Now here is the console output with a single phone defined in the
extensions.conf (exten => ------6081,1,Dial(SIP/------6091,20)
*********<SNIP>
Asterisk-A*CLI>
-- Executing Goto("SIP/---.---.241.35-40418730", "Charity|------3263|1") in new
stack
-- Goto (Charity,-------263,1)
-- Executing Dial("SIP/---.---.241.35-40418730", "SIP/------3263|18") in new
stack
-- Called ------3263
-- Got SIP response 302 "Moved Temporarily" back from ---.---.243.5
-- Now forwarding SIP/---.---.241.35-40418730 to 'Local/[EMAIL PROTECTED]'
(thanks to SIP/------3263-f670)
-- Executing Dial("Local/[EMAIL PROTECTED],2",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/---.---.241.35-36ca is making progress passing it to
Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 is making progress passing it to
SIP/---.---.241.35-40418730 Apr 29 11:30:03 NOTICE[2197]: channel.c:1314
ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of
format
g729 since our native format has changed to ulaw … …<pages of the same error> …
Apr 29 11:19:18 NOTICE[2185]: channel.c:1314 ast_read: Dropping incompatible
voice frame on Local/[EMAIL PROTECTED],2 of format
g729 since our native format has changed to ulaw
-- SIP/---.---.241.35-4e1f answered Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 answered SIP/---.---.241.35-40400490
-- Attempting native bridge of SIP/---.---.241.35-40400490 and
SIP/---.---.241.35-4e1f == Spawn exten (Charity, -------0059, 1) exited
non-zero on 'Local/[EMAIL PROTECTED],2'
*********</SNIP>
I’m sure I could change everything to ulaw G711 the problem would go away but I
do not want to do that.
Any Ideas?
Thanks
Scott H
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