RE: [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 = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
To:
[trunklocal]
exten = _9NXX,1,Answer
exten = _9NXX,n,Dial(${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: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom IP500 Forward problem codec issue
Polycom IP500 Forward problem codec issue
All,
Im running the Polycom IP500 phones at several sites. My * server is
at a collocation site and I have complete control of the T1s 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/--6081SIP/--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/--6081SIP/--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],2ZOMBIE'
-- 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
Re: [Asterisk-Users] Polycom IP500 Forward problem codec issue
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 = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
To:
[trunklocal]
exten = _9NXX,1,Answer
exten = _9NXX,n,Dial(${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: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom IP500 Forward problem codec issue
Polycom IP500 Forward problem codec issue
All,
Im running the Polycom IP500 phones at several sites. My * server is
at a collocation site and I have complete control of the T1s 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/--6081SIP/--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/--6081SIP/--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],2ZOMBIE' -- 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
Re: [Asterisk-Users] Polycom IP500 Forward problem codec issue
Joe and Charlie,
YES, that fixed the problem. I did move the whole network to G729 but
it was never a codec problem.
I'm not running CVS, it's 1.0.3 at the moment.
Thanks
Scott H
Joe Baptista wrote:
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 = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
To:
[trunklocal]
exten = _9NXX,1,Answer
exten = _9NXX,n,Dial(${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: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom IP500 Forward problem codec issue
Polycom IP500 Forward problem codec issue
All,
Im running the Polycom IP500 phones at several sites. My * server is
at a collocation site and I have complete control of the T1s 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/--6081SIP/--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/--6081SIP/--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],2ZOMBIE' -- 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
[Asterisk-Users] Polycom IP500 Forward problem codec issue
Polycom IP500 Forward problem codec issue All, Im running the Polycom IP500 phones at several sites. My * server is at a collocation site and I have complete control of the T1s 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/--6081SIP/--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/--6081SIP/--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],2ZOMBIE' -- 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 Im 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
