thr3ads.net - asterisk users - [Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird [Jun 2006]

If this information is useful, please help other people find it:
Share via:

Isaac Xiao

2006-Jun-26 07:32 UTC

[Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird

Hi,

 

Does any one experience that SIP phone to SIP phone (Polycom phone)
calls can't hear each other, but Monitor application records both end's
voices. It also happens in group pickup calls. Zap calls to queue (Local
channel) also experience this problem (sometimes, our SIP phone can't
hear any voice from incoming Zap calls when pickup, sometimes this
happens after 10-50 seconds' talk). It is weird.

 

Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Executing
Dial("Local/7188@from-internal-7036,2", "SIP/7188|30|trWwT")
in new
stack
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on RTP to 0
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on VRTP to 0
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Outgoing Call for 7188
Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Called 7188
Jun 26 16:53:35 VERBOSE[8287] logger.c: --
Local/7188@from-internal-7036,1 is ringing
Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'2978d85271fdaf3d0ac2e5b244e78773@192.168.2.66' Request 102: Found
Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'2978d85271fdaf3d0ac2e5b244e78773@192.168.2.66' Request 102: Found
Jun 26 16:53:35 DEBUG[2957] channel.c: Avoiding initial deadlock for
'SIP/7188-6b1f'
Jun 26 16:53:35 VERBOSE[8290] logger.c: -- SIP/7188-6b1f is ringing
Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Acked pending invite 102
Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Stopping retransmission on
'2978d85271fdaf3d0ac2e5b244e78773@192.168.2.66' of Request 102: Match
Found
Jun 26 16:53:37 DEBUG[2966] chan_sip.c: build_route: Contact hop: 
Jun 26 16:53:37 VERBOSE[8290] logger.c: -- SIP/7188-6b1f answered
Local/7188@from-internal-7036,2
Jun 26 16:53:37 DEBUG[8287] app_queue.c: Dunno what to do with control
type -1
Jun 26 16:53:37 VERBOSE[8287] logger.c: --
Local/7188@from-internal-7036,1 answered Zap/13-1
Jun 26 16:53:37 DEBUG[8287] chan_zap.c: Set option TONE VERIFY, mode:
MUTECONF(1) on Zap/13-1
Jun 26 16:53:37 VERBOSE[8287] logger.c: -- Stopped music on hold on
Zap/13-1
Jun 26 16:53:37 DEBUG[8287] channel.c: Scheduling timer at 0 sample
intervals
Jun 26 16:54:02 DEBUG[8290] channel.c: Didn't get a frame from channel:
SIP/7188-6b1f
Jun 26 16:54:02 DEBUG[8290] channel.c: Bridge stops bridging channels
Local/7188@from-internal-7036,2 and SIP/7188-6b1f
Jun 26 16:54:02 DEBUG[8290] chan_sip.c: update_call_counter(7188) -
decrement call limit counter
Jun 26 16:54:02 DEBUG[8290] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial,
s, 10) exited non-zero on 'Local/7188@from-internal-7036,2' in macro
'dial'
Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial,
s, 10) exited non-zero on 'Local/7188@from-internal-7036,2' in macro
'exten-vm'
Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial,
s, 10) exited non-zero on 'Local/7188@from-internal-7036,2'
Jun 26 16:54:02 DEBUG[8290] res_monitor.c: monitor executing ( nice -n
19 soxmix
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901-in.gsm"
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901-out.gsm"
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901.gsm"
&& rm
-f "/var/spool/asterisk/monitor/20060626-165333-1151304813.901-"* )
&
Jun 26 16:54:02 DEBUG[8287] channel.c: Didn't get a frame from channel:
Local/7188@from-internal-7036,1
Jun 26 16:54:02 DEBUG[8287] channel.c: Bridge stops bridging channels
Zap/13-1 and Local/7188@from-internal-7036,1
Jun 26 16:54:02 VERBOSE[8287] logger.c: == Spawn extension (ext-queues,
7141, 6) exited non-zero on 'Zap/13-1'
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value:
ON(1) on Zap/13-1
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Hangup: channel: 13 index = 0,
normal = 27, callwait = -1, thirdcall = -1
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Not yet hungup... Calling hangup
once with icause, and clearing call
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on
channel 13
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/13-1
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Updated conferencing on 13, with
0 conference users
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value:
OFF(0) on Zap/13-1
Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on
channel 13
Jun 26 16:54:02 VERBOSE[8287] logger.c: -- Hungup 'Zap/13-1'

 

Isaac Xiao

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060626/092a4a5f/attachment.htm

C F

2006-Jun-26 07:51 UTC

head link

[Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird

I have seen this when Polycom has to communicate with none polycom
phones and a transfer is initiated to a polycom, unless the Polycom
presses Hold and then unhold, there is only one way audio, this is
without NAT involved. There might also be other cases when this
happens. My workaround is to add canreinvite=no


On 6/26/06, Isaac Xiao <isaac.x@kvbkunlun.com>
wrote:>
>
>
>
> Hi,
>
>
>
> Does any one experience that SIP phone to SIP phone (Polycom phone) calls
> can't hear each other, but Monitor application records both end's
voices. It
> also happens in group pickup calls. Zap calls to queue (Local channel) also
> experience this problem (sometimes, our SIP phone can't hear any voice
from
> incoming Zap calls when pickup, sometimes this happens after 10-50
seconds'
> talk). It is weird.
>
>
>
> Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Executing
> Dial("Local/7188@from-internal-7036,2",
"SIP/7188|30|trWwT") in new stack
>  Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on RTP to 0
>  Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on VRTP to 0
>  Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Outgoing Call for 7188
>  Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Called 7188
>  Jun 26 16:53:35 VERBOSE[8287] logger.c: -- Local/7188@from-internal-7036,1
> is ringing
>  Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping
> retransmission (but retaining packet) on
> '2978d85271fdaf3d0ac2e5b244e78773@192.168.2.66' Request
> 102: Found
>  Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping
> retransmission (but retaining packet) on
> '2978d85271fdaf3d0ac2e5b244e78773@192.168.2.66' Request
> 102: Found
>  Jun 26 16:53:35 DEBUG[2957] channel.c: Avoiding initial deadlock for
> 'SIP/7188-6b1f'
>  Jun 26 16:53:35 VERBOSE[8290] logger.c: -- SIP/7188-6b1f is ringing
>  Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Acked pending invite 102
>  Jun 26 16:53:37 DEBUG[2966] chan_sip.c: Stopping retransmission on
> '2978d85271fdaf3d0ac2e5b244e78773@192.168.2.66' of Request
> 102: Match Found
>  Jun 26 16:53:37 DEBUG[2966] chan_sip.c: build_route: Contact hop:
>  Jun 26 16:53:37 VERBOSE[8290] logger.c: -- SIP/7188-6b1f answered
> Local/7188@from-internal-7036,2
>  Jun 26 16:53:37 DEBUG[8287] app_queue.c: Dunno what to do with control
type
> -1
>  Jun 26 16:53:37 VERBOSE[8287] logger.c: -- Local/7188@from-internal-7036,1
> answered Zap/13-1
>  Jun 26 16:53:37 DEBUG[8287] chan_zap.c: Set option TONE VERIFY, mode:
> MUTECONF(1) on Zap/13-1
>  Jun 26 16:53:37 VERBOSE[8287] logger.c: -- Stopped music on hold on
> Zap/13-1
>  Jun 26 16:53:37 DEBUG[8287] channel.c: Scheduling timer at 0 sample
> intervals
>  Jun 26 16:54:02 DEBUG[8290] channel.c: Didn't get a frame from
channel:
> SIP/7188-6b1f
>  Jun 26 16:54:02 DEBUG[8290] channel.c: Bridge stops bridging channels
> Local/7188@from-internal-7036,2 and SIP/7188-6b1f
>  Jun 26 16:54:02 DEBUG[8290] chan_sip.c: update_call_counter(7188) -
> decrement call limit counter
>  Jun 26 16:54:02 DEBUG[8290] app_dial.c: Exiting with DIALSTATUS=ANSWER.
>  Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s,
> 10) exited non-zero on 'Local/7188@from-internal-7036,2' in macro
'dial'
>  Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s,
> 10) exited non-zero on 'Local/7188@from-internal-7036,2' in macro
'exten-vm'
>  Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s,
> 10) exited non-zero on 'Local/7188@from-internal-7036,2'
>  Jun 26 16:54:02 DEBUG[8290] res_monitor.c: monitor executing ( nice -n 19
> soxmix
>
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901-in.gsm"
>
"/var/spool/asterisk/monitor/20060626-165333-1151304813.901-out.gsm"
> "/var/spool/asterisk/monitor/20060626-165333-1151304813.901.gsm"
> && rm -f
> "/var/spool/asterisk/monitor/20060626-165333-1151304813.901-"*
> ) &
>  Jun 26 16:54:02 DEBUG[8287] channel.c: Didn't get a frame from
channel:
> Local/7188@from-internal-7036,1
>  Jun 26 16:54:02 DEBUG[8287] channel.c: Bridge stops bridging channels
> Zap/13-1 and Local/7188@from-internal-7036,1
>  Jun 26 16:54:02 VERBOSE[8287] logger.c: == Spawn extension (ext-queues,
> 7141, 6) exited non-zero on 'Zap/13-1'
>  Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value:
ON(1)
> on Zap/13-1
>  Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Hangup: channel: 13 index = 0,
> normal = 27, callwait = -1, thirdcall = -1
>  Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Not yet hungup... Calling hangup
> once with icause, and clearing call
>  Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on
> channel 13
>  Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option TDD MODE, value: OFF(0)
> on Zap/13-1
>  Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Updated conferencing on 13, with 0
> conference users
>  Jun 26 16:54:02 DEBUG[8287] chan_zap.c: Set option AUDIO MODE, value:
> OFF(0) on Zap/13-1
>  Jun 26 16:54:02 DEBUG[8287] chan_zap.c: disabled echo cancellation on
> channel 13
>  Jun 26 16:54:02 VERBOSE[8287] logger.c: -- Hungup 'Zap/13-1'
>
>
>
> Isaac Xiao
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>

Isaac Xiao

2006-Jun-26 16:37 UTC

head link

[Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird

Our phone all Polycom phone and we use *'s transfer function rather than
phone's one. We also has canreinvite=no. I believe that it is something
wrong with Call Bridge between two channels(ZAP/SIP/Local). Before we
didn't disable autofallthrough (default is yes), we also experienced
call drop.
>I have seen this when Polycom has to communicate with none polycom
>phones and a transfer is initiated to a polycom, unless the Polycom
>presses Hold and then unhold, there is only one way audio, this is
>without NAT involved. There might also be other cases when this
>happens. My workaround is to add canreinvite=no
Isaac Xiao

Possibly Parallel Threads

Search for more reasonably related threads

asterisk users - Jun 2006 - 1.2.9.1 SIP/Local/Queue behaviours weird

[Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird

[Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird

[Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird

Possibly Parallel Threads