[Asterisk-Users] Polycom IP500 Forward problem codec issue

[Joe Baptista]

On May 2, 2005 10:31 am, Charlie Watts wrote:
> 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:

Same problem encountered here.  My solution is to answer and play a sec of 
silence before the dial proceeds - if i don't answer both parties are 
connected but can't hear each other.

joe

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

-- 
Joe Baptista
www.joebaptista.com