thr3ads.net - asterisk users - [Asterisk-Users] Polycom IP500 Forward problem codec issue [Apr 2005]

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Scott Herrick

2005-Apr-30 07:49 UTC

[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/-------0509@TPN-498a,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/-------0509@TPN-498a,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/--------0509@TPN' (thanks toSIP/------6091-6268)
  -- Executing Dial("Local/-------0509@TPN-48f0,2", 
"SIP/-------0509@---.---.-41.35") in new stack
  -- Called ------0509@---.---.241.35
  -- SIP/------6081-e558 is ringing
  -- SIP/---.---.241.35-f522 is making progress passing it to 
Local/-------0509@TPN-48f0,2
  -- Local/-------0509@TPN-48f0,1 is making progress passing it to 
SIP/---.---.241.35-40400490
  -- SIP/---.---.241.35-f522 answered Local/-------0509@TPN-48f0,2
  -- Local/-------0509@TPN-48f0,1 answered SIP/---.---.---.35-40400490
  == Spawn extension (TPN, ------6081, 1) exited non-zero on 
'Local/-------0509@TPN-48f0,2<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/-------0059@Charity' (thanks to SIP/------3263-f670)
-- Executing Dial("Local/-------0059@Charity-da6c,2", 
"SIP/------0059@---.---.241.35") in new stack
  -- Called ------0059@---.---.241.35
  -- SIP/---.---.241.35-36ca is making progress passing it to 
Local/-------0059@Charity-da6c,2
  -- Local/-------0059@Charity-da6c,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/-------0059@Charity-da6c,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/-------0059@Charity-5686,2 of format 
g729 since our native format has changed to ulaw
     -- SIP/---.---.241.35-4e1f answered Local/-------0059@Charity-5686,2
     -- Local/-------0059@Charity-5686,1 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/-------0059@Charity-5686,2'

*********</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

Charlie Watts

2005-May-02 07:31 UTC

head link

[Asterisk-Users] Polycom IP500 Forward problem codec issue

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: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Scott Herrick
Sent: Saturday, April 30, 2005 8:49 AM
To: asterisk-users@lists.digium.com
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/-------0509@TPN-498a,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/-------0509@TPN-498a,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/--------0509@TPN' (thanks toSIP/------6091-6268)
  -- Executing Dial("Local/-------0509@TPN-48f0,2",
"SIP/-------0509@---.---.-41.35") in new stack
  -- Called ------0509@---.---.241.35
  -- SIP/------6081-e558 is ringing
  -- SIP/---.---.241.35-f522 is making progress passing it to
Local/-------0509@TPN-48f0,2
  -- Local/-------0509@TPN-48f0,1 is making progress passing it to
SIP/---.---.241.35-40400490
  -- SIP/---.---.241.35-f522 answered Local/-------0509@TPN-48f0,2
  -- Local/-------0509@TPN-48f0,1 answered SIP/---.---.---.35-40400490
  == Spawn extension (TPN, ------6081, 1) exited non-zero on
'Local/-------0509@TPN-48f0,2<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/-------0059@Charity' (thanks to SIP/------3263-f670)
-- Executing Dial("Local/-------0059@Charity-da6c,2",
"SIP/------0059@---.---.241.35") in new stack
  -- Called ------0059@---.---.241.35
  -- SIP/---.---.241.35-36ca is making progress passing it to
Local/-------0059@Charity-da6c,2
  -- Local/-------0059@Charity-da6c,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/-------0059@Charity-da6c,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/-------0059@Charity-5686,2 of format
g729 since our native format has changed to ulaw
     -- SIP/---.---.241.35-4e1f answered Local/-------0059@Charity-5686,2
     -- Local/-------0059@Charity-5686,1 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/-------0059@Charity-5686,2'

*********</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

-- 
No virus found in this outgoing message.
http://www.avg-antivirus.net/
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.1 - Release Date: 5/2/2005

asterisk users - Apr 2005 - Polycom IP500 Forward problem codec issue

[Asterisk-Users] Polycom IP500 Forward problem codec issue

[Asterisk-Users] Polycom IP500 Forward problem codec issue