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/[EMAIL PROTECTED],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
'[EMAIL PROTECTED]'
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/[EMAIL PROTECTED]|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 [EMAIL PROTECTED]
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
'[EMAIL PROTECTED]' 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/[EMAIL PROTECTED]' (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/[EMAIL PROTECTED],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/[EMAIL PROTECTED],2", "SIP/[EMAIL PROTECTED]|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 [EMAIL PROTECTED]
Mar 26 09:51:18 NOTICE[4888] channel.c: Dropping incompatible voice
frame on Local/[EMAIL PROTECTED],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
'[EMAIL PROTECTED]' 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/[EMAIL PROTECTED],2 to 'Local/[EMAIL PROTECTED]' (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/[EMAIL PROTECTED],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/[EMAIL PROTECTED],1 answered Local/[EMAIL PROTECTED],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/[EMAIL PROTECTED],1' at 0x8de2084
Mar 26 09:51:20 DEBUG[6208] channel.c: Putting channel
Local/[EMAIL PROTECTED],1 in 64/64 formats
Mar 26 09:51:20 DEBUG[6208] channel.c: Released clone lock on
'Local/[EMAIL PROTECTED],1<ZOMBIE>'
Mar 26 09:51:20 DEBUG[6208] channel.c: Done Masquerading
Local/[EMAIL PROTECTED],1 (6)
Mar 26 09:51:20 DEBUG[6208] channel.c: Planning to masquerade channel
Local/[EMAIL PROTECTED],1 into the structure of
Local/[EMAIL PROTECTED],1
Mar 26 09:51:20 DEBUG[6208] channel.c: Done planning to masquerade
channel Local/[EMAIL PROTECTED],1 into the structure of
Local/[EMAIL PROTECTED],1
Mar 26 09:51:20 DEBUG[6208] chan_local.c: Not posting to queue since
already masked on 'Local/[EMAIL PROTECTED],2'
Mar 26 09:51:20 DEBUG[6208] channel.c: Didn't get a frame from
channel: Local/[EMAIL PROTECTED],2
Mar 26 09:51:20 DEBUG[6208] channel.c: Bridge stops bridging channels
Local/[EMAIL PROTECTED],2 and Local/[EMAIL PROTECTED],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/[EMAIL PROTECTED],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/[EMAIL PROTECTED],2'
Mar 26 09:51:20 DEBUG[6208] pbx.c: Function result is
'Local/[EMAIL PROTECTED],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/[EMAIL PROTECTED]|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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