[asterisk-users] Polycom 601 loop

Mon Mar 26 09:14:30 MST 2007

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.