[asterisk-users] Polycom 601 loop

dave cantera david.cantera at iacnet.net
Mon Mar 26 15:01:32 MST 2007


nathan,
can you post your extensions.conf file [to-sip], and your sip.conf 
section for extension 201... ie [201]?
it looks like, perhaps, it is a dialplan problem...
daveC

Nathan Bell wrote:
> I tried to add a couple of SIP phones (polycom 601s) to my existing 
> asterisk installation. I can successfully make a call from the SIP 
> phone to any other phone (inside or outside), but I can not make any 
> calls to a SIP phone. Attached are the pertinent parts of sip.conf and 
> extensions.conf.
>
> The log starts off normal with:
> Mar 26 09:51:15 DEBUG[4885] chan_zap.c: DTMF digit: 2 on Zap/55-1
> Mar 26 09:51:15 DEBUG[4885] chan_zap.c: DTMF digit: 0 on Zap/55-1
> Mar 26 09:51:15 DEBUG[4885] chan_zap.c: DTMF digit: 1 on Zap/55-1
> Mar 26 09:51:18 DEBUG[3896] chan_sip.c: Auto destroying call 
> 'a5306fd4-bcea3062-caa16c03 at 192.168.3.2'
> Mar 26 09:51:18 DEBUG[4885] chan_zap.c: Enabled echo cancellation on 
> channel 55
> Mar 26 09:51:18 VERBOSE[4885] logger.c: -- Executing Goto("Zap/55-1", 
> "to-sip|201|1") in new stack
> Mar 26 09:51:18 VERBOSE[4885] logger.c: -- Goto (to-sip,201,1)
> Mar 26 09:51:18 VERBOSE[4885] logger.c: -- Executing Dial("Zap/55-1", 
> "SIP/201 at 192.168.2.13|120") in new stack
> Mar 26 09:51:18 DEBUG[4885] chan_sip.c: Outgoing Call for 201
> Mar 26 09:51:18 VERBOSE[4885] logger.c: -- Called 201 at 192.168.2.13
> Mar 26 09:51:18 DEBUG[3896] chan_sip.c: Acked pending invite 102
> Mar 26 09:51:18 DEBUG[3896] chan_sip.c: Stopping retransmission on 
> '10e2724033eccc6458c1f3cc7d9e1088 at 192.168.2.13' of Request 102: Match 
> Found
> Mar 26 09:51:18 VERBOSE[3896] logger.c: -- Got SIP response 482 "Loop 
> Detected" back from 192.168.2.13
> Mar 26 09:51:18 DEBUG[3896] chan_sip.c: Hairpin detected, setting up 
> call forward for what it's worth
> Mar 26 09:51:18 VERBOSE[4885] logger.c: -- Now forwarding Zap/55-1 to 
> 'Local/201 at from-sip' (thanks to SIP/192.168.2.13-08e24bd0)
> Mar 26 09:51:18 DEBUG[4885] chan_sip.c: update_call_counter(201) - 
> decrement call limit counter
>
> After that it will loop hundreds of times with a block like this in 
> the log:
>
> Mar 26 09:51:18 VERBOSE[4888] logger.c:     -- Executing 
> Goto("Local/201 at from-sip-661a,2", "to-sip|201|1") in new stack
> Mar 26 09:51:18 VERBOSE[4888] logger.c:     -- Goto (to-sip,201,1)
> Mar 26 09:51:18 VERBOSE[4888] logger.c:     -- Executing 
> Dial("Local/201 at from-sip-661a,2", "SIP/201 at 192.168.2.13|120") in new 
> stack
> Mar 26 09:51:18 DEBUG[4888] chan_sip.c: Outgoing Call for 201
> Mar 26 09:51:18 DEBUG[3896] chan_sip.c: Failed to grab lock, trying 
> again...
> Mar 26 09:51:18 VERBOSE[4888] logger.c:     -- Called 201 at 192.168.2.13
> Mar 26 09:51:18 NOTICE[4888] channel.c: Dropping incompatible voice 
> frame on Local/201 at from-sip-661a,2 of format ulaw since our native 
> format has changed to slin
> Mar 26 09:51:18 DEBUG[3896] chan_sip.c: Acked pending invite 102
> Mar 26 09:51:18 DEBUG[3896] chan_sip.c: Stopping retransmission on 
> '4bf4e48d35343cad482650206ce86df0 at 192.168.2.13' of Request 102: Match 
> Found
> Mar 26 09:51:18 VERBOSE[3896] logger.c:     -- Got SIP response 482 
> "Loop Detected" back from 192.168.2.13
> Mar 26 09:51:18 DEBUG[3896] chan_sip.c: Hairpin detected, setting up 
> call forward for what it's worth
> Mar 26 09:51:18 VERBOSE[4888] logger.c:     -- Now forwarding 
> Local/201 at from-sip-661a,2 to 'Local/201 at from-sip' (thanks to 
> SIP/192.168.2.13-08da2240)
> Mar 26 09:51:18 DEBUG[4888] chan_sip.c: update_call_counter(201) - 
> decrement call limit counter
>
> (intertwined two parts, I know, but it's all the same messages)
>
> Eventually, this given:
>
> Mar 26 09:51:20 WARNING[6217] rtp.c: Unable to allocate socket: Too 
> many open files
> Mar 26 09:51:20 WARNING[6217] acl.c: Cannot create socket
> Mar 26 09:51:20 WARNING[6217] channel.c: Channel allocation failed: 
> Can't create alert pipe!
> Mar 26 09:51:20 WARNING[6217] chan_sip.c: Unable to allocate SIP 
> channel structure
> Mar 26 09:51:20 NOTICE[6217] app_dial.c: Unable to create channel of 
> type 'SIP' (cause 0 - Unknown)
> Mar 26 09:51:20 VERBOSE[6217] logger.c:   == Everyone is 
> busy/congested at this time (1:0/0/1)
> Mar 26 09:51:20 DEBUG[6217] app_dial.c: Exiting with 
> DIALSTATUS=CHANUNAVAIL.
> Mar 26 09:51:20 VERBOSE[6217] logger.c:     -- Executing 
> Playback("Local/201 at from-sip-d3ce,2", "tt-allbusy") in new stack
> Mar 26 09:51:20 DEBUG[6217] channel.c: Scheduling timer at 160 sample 
> intervals
> Mar 26 09:51:20 VERBOSE[6217] logger.c:     -- Playing 'tt-allbusy' 
> (language 'en')
>
> After that, it will give back a response like this for each loop:
>
> Mar 26 09:51:20 VERBOSE[6214] logger.c:     -- 
> Local/201 at from-sip-d3ce,1 answered Local/201 at from-sip-478c,2
>
> Then finally it will give this block for each loop:
>
> Mar 26 09:51:20 DEBUG[6208] channel.c: Got clone lock for masquerade 
> on 'Local/201 at from-sip-d3ce,1' at 0x8de2084
> Mar 26 09:51:20 DEBUG[6208] channel.c: Putting channel 
> Local/201 at from-sip-d3ce,1 in 64/64 formats
> Mar 26 09:51:20 DEBUG[6208] channel.c: Released clone lock on 
> 'Local/201 at from-sip-ee33,1<ZOMBIE>'
> Mar 26 09:51:20 DEBUG[6208] channel.c: Done Masquerading 
> Local/201 at from-sip-d3ce,1 (6)
> Mar 26 09:51:20 DEBUG[6208] channel.c: Planning to masquerade channel 
> Local/201 at from-sip-d3ce,1 into the structure of Local/201 at from-sip-5cc0,1
> Mar 26 09:51:20 DEBUG[6208] channel.c: Done planning to masquerade 
> channel Local/201 at from-sip-d3ce,1 into the structure of 
> Local/201 at from-sip-5cc0,1
> Mar 26 09:51:20 DEBUG[6208] chan_local.c: Not posting to queue since 
> already masked on 'Local/201 at from-sip-5cc0,2'
> Mar 26 09:51:20 DEBUG[6208] channel.c: Didn't get a frame from 
> channel: Local/201 at from-sip-5cc0,2
> Mar 26 09:51:20 DEBUG[6208] channel.c: Bridge stops bridging channels 
> Local/201 at from-sip-5cc0,2 and Local/201 at from-sip-5cc0,1<ZOMBIE>
> Mar 26 09:51:20 DEBUG[6208] app_dial.c: Exiting with DIALSTATUS=ANSWER.
> Mar 26 09:51:20 VERBOSE[6208] logger.c:   == Spawn extension (to-sip, 
> 201, 1) exited non-zero on 'Local/201 at from-sip-5cc0,2'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is '(null)'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is '(null)'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is '201'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is 'to-sip'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is 
> 'Local/201 at from-sip-5cc0,2'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is 
> 'Local/201 at from-sip-ee33,1'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is 'Dial'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is 
> 'SIP/201 at 192.168.2.13|120'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is '2007-03-26 
> 09:51:20'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is '2007-03-26 
> 09:51:20'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is '2007-03-26 
> 09:51:20'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is '0'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is '0'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is 'ANSWERED'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is 'DOCUMENTATION'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is '(null)'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is '1174924280.41882'
> Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is '(null)'
>
> For a complete log (1.7 mb) of a single call to the extension, see 
> http://www.actarg.com/all_log
>
> The polycoms are running bootrom 3.2.2.0019 and application version 
> 1.6.7.0098. Any help on this would be greatly appreciated.
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