[asterisk-users] Polycom 601 loop
Nathan Bell
nathanb at actarg.com
Mon Mar 26 14:15:08 MST 2007
Pertinent part of extensions.conf:
; from outside T1
[from-ptsn]
exten => s,1,Answer()
include => cac-ext
include => sip-ext
include => intertel-ext
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()
; from sip lines
[from-sip]
include => internal
; generic interal route
[internal]
exten => s,1,Answer()
include => cac-ext
include => sip-ext
include => intertel-ext
include => to-ptsn
; check if extension is to sip
[sip-ext]
exten => _20X,1,Goto(to-sip,${EXTEN},1)
; send call to sip
[to-sip]
exten => _X.,1,Dial(SIP/${EXTEN}@192.168.2.13,120)
exten => _X.,2,Playback(vm-nobodyavail)
exten => _X.,3,Hangup()
exten => _X.,102,Playback(tt-allbusy)
exten => _X.,103,Hangup()
cac-ext, intertel-ext are for our CAC channel bank and our inter-tel pbx
extensions. to-ptsn just routes all remaining calls to our outside T1.
There are also from-cac and from-intertel contexts that are identical to
from-sip, where the only line is include => internal.
Here's the 201 extension in sip.conf:
[201]
type=friend ; Friends place calls and receive calls
context=from-sip ; Context for incoming calls from this user
secret=asteriskpassword
host=dynamic ; This peer register with us
callerid=John Doe <201>
disallow=all
allow=ulaw ; dtmfmode=inband only works with ulaw or
alaw!
progressinband=no ; Polycom phones don't work properly with
"never"
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
nat=no ; there is not NAT between phone and Asterisk
canreinvite=no ; disallow RTP voice traffic to bypass
Asterisk
The global part of sip.conf is in my other e-mail "SIP registration"
where in I tell my tale of SIP registration woes ... perhaps both my
problems are one and the same?
dave cantera wrote:
> 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.
>> _______________________________________________
>> --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
>>
>>
>>
>
More information about the asterisk-users
mailing list