Problem with outgoing call

Asterisk Asterisk Support
1

Hello

I have asterisk 1.4.14, few PA168 and gateway: Sip-Gateway-MP-408/SIP 2.2.9.14263. All device are in the same subnet.

I have problem with outgoing call from my asterisk. Incoming are ok. Let’s say that the problem is more less with 30% of all outgoing call, i don’t hear any ringing in phone. Sometimes when i try second time to call to the same number the call is ok. I saw logs, debug outputs but i don’t see anything interesting. I don’t have any idea how figure out this and what next step of debuging i should do :frowning:

my sip.conf

allow=g729
[authentication]
canreinvite=no
disallow=all
domain=pbx.mobiwide.com
[general]
register=>121114020:pass_secret@eneo.integralnet.com/121114020

my extension.conf:

.51
mt@192.168.1.51's password: 
Linux dell 2.6.18-5-686 #1 SMP Wed Oct 3 00:12:50 UTC 2007 i686

The programs included with the Debian GNU/Linux system are free software;
the exact distribution terms for each program are described in the
individual files in /usr/share/doc/*/copyright.

Debian GNU/Linux comes with ABSOLUTELY NO WARRANTY, to the extent
permitted by applicable law.
Last login: Tue Mar 30 10:02:00 2010 from iux130.internetdsl.tpnet.pl
mt@dell:~$ su
Password: 
dell:/home/mt# cat /etc/asterisk/extensions.conf | grep -v ";" | sort























autofallthrough=yes
clearglobalvars=no
[default]
DYNAMIC_FEATURES=>automon#apps
exten = _0.,1,Dial(SIP/audiocodes/${EXTEN},,W)
exten = _0.,1,Set(CDR(userfield)=OUT_HRK)
exten = _0.,2,Dial(SIP/integral_122984029/${EXTEN})
exten = _0.,2,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?4:)
exten = _0.,3,Hangup
exten = 101,1,Dial(SIP/audiocodes/${EXTEN})
exten = 103,1,Dial(SIP/audiocodes/${EXTEN})
exten => 1,1,Answer
exten = 120,1,Dial(SIP/audiocodes/${EXTEN})
exten => 122984022,1,Goto(default,120,1)
exten => 122984023,1,Goto(incoming_ivr,1,1)
exten => 122984029,1,Dial(SIP/200,5)
exten => 122984029,2,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?4:)
exten => 122984029,3,Hangup
exten => 122984029,4,Dial(SIP/integral_122984029/0509370209&SIP/200)
exten => _12298402X,1,Set(CDR(userfield)=IN_${EXTEN})
exten => 1,2,Set(TIMEOUT(digit)=5)
exten => 1,3,Set(TIMEOUT(response)=10)
exten => 1,4,Background(/tmp/asterisk-recording)
exten => 1,5,WaitExten(10|m(default))
exten => 1,6,Goto(4)
exten = 500,1,Meetme(1)
exten = 501,1,Meetme(2)
  exten => 705,3,Wait(2)
  exten => 705,5,wait(2)
  exten => 705,6,Hangup
exten = _80.,1,Set(CALLFILENAME=${CALLERID(number)}-${EXTEN:1}-${UNIQUEID}.wav|V(3)) 
exten = _80.,n,Dial(SIP/audiocodes/${EXTEN:1})
exten = _80.,n,MixMonitor(${CALLFILENAME})
exten = _90.,1,Dial(SIP/integral_122984020/${EXTEN:1})
exten => i,1,Hangup
exten => s,1,Dial(${ARG1},15)
exten => s,1,GotoIf($["${XAD}" = "0" | "${XAD}" = ""]?startrec:stoprec) 
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s,n,MacroExit 
exten => s,n,MacroExit 
exten => s,n,Monitor(wav,${FILENAME},m) 
exten => s-NOANSWER,1,Dial(SIP/100&SIP/audiocodes/101&SIP/102&SIP/104&SIP/107&SIP/108)
exten => s,n,Set(FILENAME=${TIMESTAMP}-OUT${CALLERID(number)}-^-${UNIQUEID}) 
exten => s,n,Set(MONITOR_EXEC_ARGS=&& nice -n 19 /usr/local/bin/lame -b 96 -t -F -m m --bitwidth 16 --quiet "/var/spool/asterisk/monitor/${FILENAME}.wav" "/var/spool/asterisk/monitor/${FILENAME}.mp3" && rm -f "/var/spool/asterisk/monitor/${FILENAME}.wav") 
exten => s,n,Set(XAD=0) 
exten => s,n,Set(XAD=1) 
exten => s,n(stoprec),StopMonitor 
exten => _XXX,1,Goto(default,${EXTEN},1)
exten = _XXX,1,Goto(default,${EXTEN},1)
exten = _XXXXXXXXX,1,Dial(SIP/integral_122984020/0${EXTEN})
[general]
[globals]
[hrk]
[incoming_ivr]
[macro-apprecord] 
[macro-dialandhunt]
static=yes
writeprotect=no
dell:/home/mt# 
dell:/home/mt# cat /etc/asterisk/extensions.conf | grep -v ";" | sort























autofallthrough=yes
clearglobalvars=no
[default]
DYNAMIC_FEATURES=>automon#apps
exten = _0.,1,Dial(SIP/audiocodes/${EXTEN},,W)
exten = _0.,1,Set(CDR(userfield)=OUT_HRK)
exten = _0.,2,Dial(SIP/integral_122984029/${EXTEN})
exten = _0.,2,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?4:)
exten = _0.,3,Hangup
exten = 101,1,Dial(SIP/audiocodes/${EXTEN})
exten = 103,1,Dial(SIP/audiocodes/${EXTEN})
exten => 1,1,Answer
exten = 120,1,Dial(SIP/audiocodes/${EXTEN})
exten => 122984022,1,Goto(default,120,1)
exten => 122984023,1,Goto(incoming_ivr,1,1)
exten => 122984029,1,Dial(SIP/200,5)
exten => 122984029,2,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?4:)
exten => 122984029,3,Hangup
exten => 122984029,4,Dial(SIP/integral_122984029/0509370209&SIP/200)
exten => _12298402X,1,Set(CDR(userfield)=IN_${EXTEN})
exten => 1,2,Set(TIMEOUT(digit)=5)
exten => 1,3,Set(TIMEOUT(response)=10)
exten => 1,4,Background(/tmp/asterisk-recording)
exten => 1,5,WaitExten(10|m(default))
exten => 1,6,Goto(4)
exten = 500,1,Meetme(1)
exten = 501,1,Meetme(2)
  exten => 705,3,Wait(2)
  exten => 705,5,wait(2)
  exten => 705,6,Hangup
exten = _80.,1,Set(CALLFILENAME=${CALLERID(number)}-${EXTEN:1}-${UNIQUEID}.wav|V(3)) 
exten = _80.,n,Dial(SIP/audiocodes/${EXTEN:1})
exten = _80.,n,MixMonitor(${CALLFILENAME})
exten = _90.,1,Dial(SIP/integral_122984020/${EXTEN:1})
exten => i,1,Hangup
exten => s,1,Dial(${ARG1},15)
exten => s,1,GotoIf($["${XAD}" = "0" | "${XAD}" = ""]?startrec:stoprec) 
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s,n,MacroExit 
exten => s,n,MacroExit 
exten => s,n,Monitor(wav,${FILENAME},m) 
exten => s-NOANSWER,1,Dial(SIP/100&SIP/audiocodes/101&SIP/102&SIP/104&SIP/107&SIP/108)
exten => s,n,Set(FILENAME=${TIMESTAMP}-OUT${CALLERID(number)}-^-${UNIQUEID}) 
exten => s,n,Set(MONITOR_EXEC_ARGS=&& nice -n 19 /usr/local/bin/lame -b 96 -t -F -m m --bitwidth 16 --quiet "/var/spool/asterisk/monitor/${FILENAME}.wav" "/var/spool/asterisk/monitor/${FILENAME}.mp3" && rm -f "/var/spool/asterisk/monitor/${FILENAME}.wav") 
exten => s,n,Set(XAD=0) 
exten => s,n,Set(XAD=1) 
exten => s,n(stoprec),StopMonitor 
exten => _XXX,1,Goto(default,${EXTEN},1)
exten = _XXX,1,Goto(default,${EXTEN},1)
exten = _XXXXXXXXX,1,Dial(SIP/integral_122984020/0${EXTEN})
[general]
[globals]
[hrk]
[incoming_ivr]
[macro-apprecord] 
[macro-dialandhunt]
static=yes
writeprotect=no
dell:/home/mt# cat /etc/asterisk/extensions.conf | grep -v ";" 
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no

[globals]
DYNAMIC_FEATURES=>automon#apps


[default]
exten = 500,1,Meetme(1)

exten = 120,1,Dial(SIP/audiocodes/${EXTEN})
exten = 103,1,Dial(SIP/audiocodes/${EXTEN})
exten = 101,1,Dial(SIP/audiocodes/${EXTEN})

exten = _0.,1,Dial(SIP/audiocodes/${EXTEN},,W)
exten = _0.,2,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?4:)
exten = _0.,3,Hangup

exten = _XXXXXXXXX,1,Dial(SIP/integral_122984020/0${EXTEN})
exten = _90.,1,Dial(SIP/integral_122984020/${EXTEN:1})

exten = _80.,1,Set(CALLFILENAME=${CALLERID(number)}-${EXTEN:1}-${UNIQUEID}.wav|V(3)) 
exten = _80.,n,MixMonitor(${CALLFILENAME})
exten = _80.,n,Dial(SIP/audiocodes/${EXTEN:1})



exten => 122984029,1,Dial(SIP/200,5)
exten => 122984029,2,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?4:)
exten => 122984029,3,Hangup
exten => 122984029,4,Dial(SIP/integral_122984029/0509370209&SIP/200)



exten => 122984022,1,Goto(default,120,1)
exten => 122984023,1,Goto(incoming_ivr,1,1)
exten => _12298402X,1,Set(CDR(userfield)=IN_${EXTEN})



  exten => 705,3,Wait(2)
  exten => 705,5,wait(2)
  exten => 705,6,Hangup

exten => i,1,Hangup

[macro-dialandhunt]
exten => s,1,Dial(${ARG1},15)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Dial(SIP/100&SIP/audiocodes/101&SIP/102&SIP/104&SIP/107&SIP/108)

[incoming_ivr]
exten => 1,1,Answer
exten => 1,2,Set(TIMEOUT(digit)=5)
exten => 1,3,Set(TIMEOUT(response)=10)
exten => 1,4,Background(/tmp/asterisk-recording)
exten => 1,5,WaitExten(10|m(default))
exten => 1,6,Goto(4)
exten => _XXX,1,Goto(default,${EXTEN},1)


[hrk]

exten = _0.,1,Set(CDR(userfield)=OUT_HRK)
exten = _0.,2,Dial(SIP/integral_122984029/${EXTEN})
exten = 501,1,Meetme(2)
exten = _XXX,1,Goto(default,${EXTEN},1)

[macro-apprecord] 
exten => s,1,GotoIf($["${XAD}" = "0" | "${XAD}" = ""]?startrec:stoprec) 
exten => s,n,Set(XAD=1) 
exten => s,n,Set(FILENAME=${TIMESTAMP}-OUT${CALLERID(number)}-^-${UNIQUEID}) 
exten => s,n,Set(MONITOR_EXEC_ARGS=&& nice -n 19 /usr/local/bin/lame -b 96 -t -F -m m --bitwidth 16 --quiet "/var/spool/asterisk/monitor/${FILENAME}.wav" "/var/spool/asterisk/monitor/${FILENAME}.mp3" && rm -f "/var/spool/asterisk/monitor/${FILENAME}.wav") 
exten => s,n,Monitor(wav,${FILENAME},m) 
exten => s,n,MacroExit 
exten => s,n(stoprec),StopMonitor 
exten => s,n,Set(XAD=0) 
exten => s,n,MacroExit 
dell:/home/mt# 
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dell:/home/mt# 
dell:/home/mt# 
dell:/home/mt# cat /etc/asterisk/extensions.conf | grep -v ";"  | more
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no

[globals]
DYNAMIC_FEATURES=>automon#apps


[default]
exten = 500,1,Meetme(1)

exten = 120,1,Dial(SIP/audiocodes/${EXTEN})
exten = 103,1,Dial(SIP/audiocodes/${EXTEN})
exten = 101,1,Dial(SIP/audiocodes/${EXTEN})

exten = _0.,1,Dial(SIP/audiocodes/${EXTEN},,W)
exten = _0.,2,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?4:)
exten = _0.,3,Hangup

exten = _XXXXXXXXX,1,Dial(SIP/integral_122984020/0${EXTEN})
exten = _90.,1,Dial(SIP/integral_122984020/${EXTEN:1})

exten = _80.,1,Set(CALLFILENAME=${CALLERID(number)}-${EXTEN:1}-${UNIQUEID}.wav|V(3)) 
exten = _80.,n,MixMonitor(${CALLFILENAME})
exten = _80.,n,Dial(SIP/audiocodes/${EXTEN:1})

and my logs:

working ok:

[Mar 26 11:22:53] VERBOSE[18111] logger.c:     -- Executing [0606827530@default:1] Dial("SIP/100-081e7368", "SIP/audiocodes/0606827530||W") in new stack
[Mar 26 11:22:53] VERBOSE[18111] logger.c:     -- Called audiocodes/0606827530
[Mar 26 11:22:57] VERBOSE[18111] logger.c:     -- SIP/audiocodes-081dd148 is ringing
[Mar 26 11:23:01] NOTICE[18111] cdr.c: CDR on channel 'SIP/audiocodes-081dd148' not posted
[Mar 26 11:23:01] VERBOSE[18111] logger.c:   == Spawn extension (default, 0606827530, 1) exited non-zero on 'SIP/100-081e7368'

not ringing:

[Mar 26 11:23:20] VERBOSE[18112] logger.c:     -- Executing [0608136204@default:1] Dial("SIP/100-081e5150", "SIP/audiocodes/0608136204||W") in new stack
[Mar 26 11:23:20] VERBOSE[18112] logger.c:     -- Called audiocodes/0608136204
[Mar 26 11:24:01] VERBOSE[18112] logger.c:     -- SIP/audiocodes-081dfb30 answered SIP/100-081e5150
[Mar 26 11:24:01] VERBOSE[18112] logger.c:     -- Packet2Packet bridging SIP/100-081e5150 and SIP/audiocodes-081dfb30
[Mar 26 11:24:06] VERBOSE[18112] logger.c:   == Spawn extension (default, 0608136204, 1) exited non-zero on 'SIP/100-081e5150'

ringing:

[Mar 26 11:24:13] VERBOSE[18113] logger.c:     -- Executing [0608136204@default:1] Dial("SIP/100-081dd148", "SIP/audiocodes/0608136204||W") in new stack
[Mar 26 11:24:13] VERBOSE[18113] logger.c:     -- Called audiocodes/0608136204
[Mar 26 11:24:20] VERBOSE[18113] logger.c:     -- SIP/audiocodes-081e7368 is ringing
[Mar 26 11:24:22] NOTICE[18113] cdr.c: CDR on channel 'SIP/audiocodes-081e7368' not posted
[Mar 26 11:24:22] VERBOSE[18113] logger.c:   == Spawn extension (default, 0608136204, 1) exited non-zero on 'SIP/100-081dd148'

not ringing:

[Mar 26 11:24:44] VERBOSE[18114] logger.c:     -- Executing [0694457421@default:1] Dial("SIP/100-081dd148", "SIP/audiocodes/0694457421||W") in new stack
[Mar 26 11:24:44] VERBOSE[18114] logger.c:     -- Called audiocodes/0694457421
[Mar 26 11:25:16] NOTICE[18114] cdr.c: CDR on channel 'SIP/audiocodes-081e10b8' not posted
[Mar 26 11:25:16] VERBOSE[18114] logger.c:   == Spawn extension (default, 0694457421, 1) exited non-zero on 'SIP/100-081dd148'

not ringing first time:

[Mar 26 11:23:20] VERBOSE[18112] logger.c:     -- Executing [0608136204@default:1] Dial("SIP/100-081e5150", "SIP/audiocodes/0608136204||W") in new stack
[Mar 26 11:23:20] VERBOSE[18112] logger.c:     -- Called audiocodes/0608136204
[Mar 26 11:24:01] VERBOSE[18112] logger.c:     -- SIP/audiocodes-081dfb30 answered SIP/100-081e5150
[Mar 26 11:24:01] VERBOSE[18112] logger.c:     -- Packet2Packet bridging SIP/100-081e5150 and SIP/audiocodes-081dfb30
[Mar 26 11:24:06] VERBOSE[18112] logger.c:   == Spawn extension (default, 0608136204, 1) exited non-zero on 'SIP/100-081e5150'

ringing second time

[Mar 26 11:24:13] VERBOSE[18113] logger.c:     -- Executing [0608136204@default:1] Dial("SIP/100-081dd148", "SIP/audiocodes/0608136204||W") in new stack
[Mar 26 11:24:13] VERBOSE[18113] logger.c:     -- Called audiocodes/0608136204
[Mar 26 11:24:20] VERBOSE[18113] logger.c:     -- SIP/audiocodes-081e7368 is ringing
[Mar 26 11:24:22] NOTICE[18113] cdr.c: CDR on channel 'SIP/audiocodes-081e7368' not posted
[Mar 26 11:24:22] VERBOSE[18113] logger.c:   == Spawn extension (default, 0608136204, 1) exited non-zero on 'SIP/100-081dd148'
* SIP Call
  Curr. trans. direction:  Incoming
  Call-ID:                me50wxg57ae5RbEX@192.168.1.70
  Owner channel ID:       SIP/102-081e4a78
  Our Codec Capability:   256
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   1294
  Joint Codec Capability:   256
  Format:                 0x100 (g729)
  MaxCallBR:              384 kbps
  Theoretical Address:    192.168.1.70:1720
  Received Address:       192.168.1.70:1720
  SIP Transfer mode:      open
  NAT Support:            RFC3581
  Audio IP:               192.168.1.51 (local)
  Our Tag:                as6ffb1bc4
  Their Tag:              7faGI9XGyvRxfupH
  SIP User agent:         PA168S V1.57.006 CFG0
  Username:               102
  Peername:               102
  Original uri:           sip:102@192.168.1.70:1720
  Caller-ID:              102
  Need Destroy:           0
  Last Message:           Rx: INVITE
  Promiscuous Redir:      No
  Route:                  sip:102@192.168.1.70:1720
  DTMF Mode:              rfc2833
  SIP Options:            replaces replace





  * SIP Call
  Curr. trans. direction:  Incoming
  Call-ID:                me50wxg57ae5RbEX@192.168.1.70
  Owner channel ID:       SIP/102-081e4a78
  Our Codec Capability:   256
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   1294
  Joint Codec Capability:   256
  Format:                 0x100 (g729)
  MaxCallBR:              384 kbps
  Theoretical Address:    192.168.1.70:1720
  Received Address:       192.168.1.70:1720
  SIP Transfer mode:      open
  NAT Support:            RFC3581
  Audio IP:               192.168.1.51 (local)
  Our Tag:                as6ffb1bc4
  Their Tag:              7faGI9XGyvRxfupH
  SIP User agent:         PA168S V1.57.006 CFG0
  Username:               102
  Peername:               102
  Original uri:           sip:102@192.168.1.70:1720
  Caller-ID:              102
  Need Destroy:           0
  Last Message:           Rx: ACK
  Promiscuous Redir:      No
  Route:                  sip:102@192.168.1.70:1720
  DTMF Mode:              rfc2833
  SIP Options:            replaces replace

  == Spawn extension (default, 0600660713, 1) exited non-zero on 'SIP/102-081e4a78'



  Curr. trans. direction:  Incoming
  Call-ID:                9VSEDmHi8eNB0L6n@192.168.1.70
  Owner channel ID:       SIP/102-081dd148
  Our Codec Capability:   256
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   1294
  Joint Codec Capability:   256
  Format:                 0x100 (g729)
  MaxCallBR:              384 kbps
  Theoretical Address:    192.168.1.70:1720
  Received Address:       192.168.1.70:1720
  SIP Transfer mode:      open
  NAT Support:            RFC3581
  Audio IP:               192.168.1.51 (local)
  Our Tag:                as46ee0d50
  Their Tag:              Gfy99QDtB8yfhPtx
  SIP User agent:         PA168S V1.57.006 CFG0
  Username:               102
  Peername:               102
  Original uri:           sip:102@192.168.1.70:1720
  Caller-ID:              102
  Need Destroy:           0
  Last Message:           Rx: INVITE
  Promiscuous Redir:      No
  Route:                  sip:102@192.168.1.70:1720
  DTMF Mode:              rfc2833
  SIP Options:            replaces replace

dell*CLI> sip show channels

Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold Last Message
192.168.1.68     (None)      41bJ80reOF7  00101/63141  unkn  No Rx: REGISTER
192.168.1.50     0500160713  37f841794ef  00102/00000  g729  No Init: INVITE
192.168.1.70     102         HFqFMxnjnc9  00101/00001  g729  No Rx: INVITE

dell*CLI>

  • SIP Call
    Curr. trans. direction: Outgoing
    Call-ID: 37f841794ef889aa420e9c5e5fd4dd26@192.168.1.51
    Owner channel ID: SIP/audiocodes-081dfb30
    Our Codec Capability: 256
    Non-Codec Capability (DTMF): 1
    Their Codec Capability: 256
    Joint Codec Capability: 256
    Format: 0x100 (g729)
    MaxCallBR: 384 kbps
    Theoretical Address: 192.168.1.50:1720
    Received Address: 192.168.1.50:1720
    SIP Transfer mode: open
    NAT Support: RFC3581
    Audio IP: 192.168.1.51 (local)
    Our Tag: as5262b4e8
    Their Tag: 139ea90-0-13c4-1111e0-e3d3bf6-1111e0
    SIP User agent: Audiocodes-Sip-Gateway-MP-408/SIP 2.2.9.14263
    Username: 0500160713
    Peername: audiocodes
    Original uri: sip:0600660713@192.168.1.50:1720
    Need Destroy: 0
    Last Message: Tx: ACK
    Promiscuous Redir: No
    Route: sip:0600660713@192.168.1.50:1720
    DTMF Mode: rfc2833
    SIP Options: (none)

    – Started music on hold, class ‘default’, on SIP/audiocodes-081dfb30
    – Stopped music on hold on SIP/audiocodes-081dfb30
    == Spawn extension (default, 0600660713, 1) exited non-zero on ‘SIP/102-081e4a78’
    [/code]

[Mar 29 09:47:11] VERBOSE[20407] logger.c:   == Spawn extension (default, 0600660713, 1) exited non-zero on 'SIP/102-081eda38'
[Mar 29 09:47:19] VERBOSE[20408] logger.c:     -- Executing [0600660713@default:1] Dial("SIP/102-081eda38", "SIP/audiocodes/0600660713||W") in new stack
[Mar 29 09:47:19] VERBOSE[20408] logger.c:     -- Called audiocodes/0600660713
[Mar 29 09:47:50] NOTICE[20408] cdr.c: CDR on channel 'SIP/audiocodes-081f2f30' not posted
[Mar 29 09:47:50] VERBOSE[20408] logger.c:   == Spawn extension (default, 0600660713, 1) exited non-zero on 'SIP/102-081eda38'
[Mar 29 09:48:00] VERBOSE[20409] logger.c:     -- Executing [0600660713@default:1] Dial("SIP/102-081eda38", "SIP/audiocodes/0600660713||W") in new stack
[Mar 29 09:48:00] VERBOSE[20409] logger.c:     -- Called audiocodes/0600660713
[Mar 29 09:48:21] NOTICE[20409] cdr.c: CDR on channel 'SIP/audiocodes-081f19a8' not posted
[Mar 29 09:48:21] VERBOSE[20409] logger.c:   == Spawn extension (default, 0600660713, 1) exited non-zero on 'SIP/102-081eda38'
[Mar 29 09:48:35] VERBOSE[20410] logger.c:     -- Executing [0600660713@default:1] Dial("SIP/102-081eda38", "SIP/audiocodes/0600660713||W") in new stack
[Mar 29 09:48:35] VERBOSE[20410] logger.c:     -- Called audiocodes/0600660713

logs in working time:

Code: Select all
    [Mar 26 11:22:53] VERBOSE[18111] logger.c:     -- Executing [0606827530@default:1] Dial("SIP/100-081e7368", "SIP/audiocodes/0606827530||W") in new stack
    [Mar 26 11:22:53] VERBOSE[18111] logger.c:     -- Called audiocodes/0606827530
    [Mar 26 11:22:57] VERBOSE[18111] logger.c:     -- SIP/audiocodes-081dd148 is ringing
    [Mar 26 11:23:01] NOTICE[18111] cdr.c: CDR on channel 'SIP/audiocodes-081dd148' not posted
    [Mar 26 11:23:01] VERBOSE[18111] logger.c:   == Spawn extension (default, 0606827530, 1) exited non-zero on 'SIP/100-081e7368'

not working:

Code: Select all
    [Mar 26 11:23:20] VERBOSE[18112] logger.c:     -- Executing [0608136204@default:1] Dial("SIP/100-081e5150", "SIP/audiocodes/0608136204||W") in new stack
    [Mar 26 11:23:20] VERBOSE[18112] logger.c:     -- Called audiocodes/0608136204
    [Mar 26 11:24:01] VERBOSE[18112] logger.c:     -- SIP/audiocodes-081dfb30 answered SIP/100-081e5150
    [Mar 26 11:24:01] VERBOSE[18112] logger.c:     -- Packet2Packet bridging SIP/100-081e5150 and SIP/audiocodes-081dfb30
    [Mar 26 11:24:06] VERBOSE[18112] logger.c:   == Spawn extension (default, 0608136204, 1) exited non-zero on 'SIP/100-081e5150'

now working:

Code: Select all
    [Mar 26 11:24:13] VERBOSE[18113] logger.c:     -- Executing [0608136204@default:1] Dial("SIP/100-081dd148", "SIP/audiocodes/0608136204||W") in new stack
    [Mar 26 11:24:13] VERBOSE[18113] logger.c:     -- Called audiocodes/0608136204
    [Mar 26 11:24:20] VERBOSE[18113] logger.c:     -- SIP/audiocodes-081e7368 is ringing
    [Mar 26 11:24:22] NOTICE[18113] cdr.c: CDR on channel 'SIP/audiocodes-081e7368' not posted
    [Mar 26 11:24:22] VERBOSE[18113] logger.c:   == Spawn extension (default, 0608136204, 1) exited non-zero on 'SIP/100-081dd148'

and not working:

Code: Select all
    [Mar 26 11:24:44] VERBOSE[18114] logger.c:     -- Executing [0694457421@default:1] Dial("SIP/100-081dd148", "SIP/audiocodes/0694457421||W") in new stack
    [Mar 26 11:24:44] VERBOSE[18114] logger.c:     -- Called audiocodes/0694457421
    [Mar 26 11:25:16] NOTICE[18114] cdr.c: CDR on channel 'SIP/audiocodes-081e10b8' not posted
    [Mar 26 11:25:16] VERBOSE[18114] logger.c:   == Spawn extension (default, 0694457421, 1) exited non-zero on 'SIP/100-081dd148'
2

[quote=“dark1965”]my sip.conf

allow=g729 [authentication] canreinvite=no disallow=all domain=pbx.mobiwide.com [general] register=>121114020:pass_secret@eneo.integralnet.com/121114020 [/quote]
When Asterisk reads your sip.conf file and encounters disallow=all statement, it removes all CoDec. In this case, the allow=g729 statement will no longer has any impact as if it is not there.

3

It’s bug in coping to forum, really i have sip.conf in this order:

[general]
domain=pbx.mobiwide.com
disallow=all
allow=g729
register=>121114020:pass_secret@eneo.integralnet.com/121114020
canreinvite=no
[authentication]

and i still haven’t any idea how i should debug more this case…

4

[quote=“dark1965”]It’s bug in coping to forum, really i have sip.conf in this order:

[general]
domain=pbx.mobiwide.com
disallow=all
allow=g729
register=>121114020:pass_secret@eneo.integralnet.com/121114020
canreinvite=no
[authentication]

and i still haven’t any idea how i should debug more this case…[/quote]
Since you only allow=g729, make sure the call you place supports G729 CoDec. If your outbound call goes to a PSTN network, be sure to check with your VoSP which CoDecs are supported.

5

All of my outgoing call should go via Sip-Gateway-MP-408/SIP 2.2.9.14263

it’s my extension.conf

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no

[globals]
DYNAMIC_FEATURES=>automon#apps


[default]
exten = 500,1,Meetme(1)

exten = 120,1,Dial(SIP/audiocodes/${EXTEN})
exten = 103,1,Dial(SIP/audiocodes/${EXTEN})
exten = 101,1,Dial(SIP/audiocodes/${EXTEN})

exten = _0.,1,Dial(SIP/audiocodes/${EXTEN},,W)
exten = _0.,2,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?4:)
exten = _0.,3,Hangup

exten = _XXXXXXXXX,1,Dial(SIP/integral_122984020/0${EXTEN})
exten = _90.,1,Dial(SIP/integral_122984020/${EXTEN:1})

exten = _80.,1,Set(CALLFILENAME=${CALLERID(number)}-${EXTEN:1}-${UNIQUEID}.wav|V(3))
exten = _80.,n,MixMonitor(${CALLFILENAME})
exten = _80.,n,Dial(SIP/audiocodes/${EXTEN:1})

my sip show peers:

dell*CLI> sip show pee
peers  peer
dell*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
integral_122984023/122984  87.204.129.4         N      5060     OK (1 ms)
integral_122984022/122984  87.204.129.4         N      5060     OK (1 ms)
integral_122984020/122984  87.204.129.4         N      5060     OK (1 ms)
integral_122984029/122984  87.204.129.4         N      5060     OK (1 ms)
200                        (Unspecified)    D          0        UNKNOWN
114                        (Unspecified)    D          0        UNKNOWN
113                        (Unspecified)    D          0        UNKNOWN
112                        (Unspecified)    D          0        UNKNOWN
111                        (Unspecified)    D          0        UNKNOWN
110/110                    192.168.1.64     D          1720     OK (48 ms)
109                        (Unspecified)    D          0        UNKNOWN
108/108                    192.168.1.68     D          1720     OK (49 ms)
107/107                    192.168.1.67     D          1720     OK (57 ms)
106/106                    192.168.1.66     D          1720     OK (49 ms)
105                        (Unspecified)    D          0        UNKNOWN
104                        (Unspecified)    D          0        UNKNOWN
102/102                    192.168.1.70     D          1720     OK (48 ms)
100/100                    192.168.1.142    D          5060     OK (71 ms)
audiocodes/audiocodes      192.168.1.50     D          1720     OK (23 ms)
6

What CoDec is this device configured with?

7

dell*CLI> sip show peer audiocodes


  * Name       : audiocodes
  Secret       : <Not set>
  MD5Secret    : <Not set>
  Context      : default
  Subscr.Cont. : <Not set>
  Language     :
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 1
  Pickupgroup  : 1
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "New User" <>
  MaxCallBR    : 384 kbps
  Expire       : 2901
  Insecure     : port,invite
  Nat          : RFC3581
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       :
  Addr->IP     : 192.168.1.50 Port 1720
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: audiocodes
  SIP Options  : (none)
  Codecs       : 0x100 (g729)
  Codec Order  : (g729:20)
  Auto-Framing:  No
  Status       : OK (23 ms)
  Useragent    :
  Reg. Contact : sip:audiocodes@192.168.1.50:1720
dell*CLI>

i tested alaw

disallow=all
allow=g729
;allow=ulaw                     ; Allow codecs in order of preference
;allow=alaw

but it didn’t help