So I’ve got an *@h setup at my headquarters. Everything is working quite well. I decided to setup an extension/phone for my home office so that I could use 3 digit dial to reach my office bound colleagues and bypass the pstn. The IP phone at home (xt 201, Polycom IP300) registers with the * server (filled in phone icon) gets dial tone etc. However when I place a call to a, local to the * box, SIP extension (xt 185) I get:
Connected to Asterisk 1.0.9 currently running on asterisk1 (pid = 3395)
Verbosity is at least 3
-- Executing Macro("SIP/201-5451", "exten-vm|185@default|185") in new stack
-- Executing SetVar("SIP/201-5451", "FROMCONTEXT=exten-vm") in new stack
-- Executing Macro("SIP/201-5451", "record-enable|185|IN") in new stack
-- Executing GotoIf("SIP/201-5451", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing GotoIf("SIP/201-5451", "0?5:8") in new stack
-- Goto (macro-record-enable,s,8)
-- Executing GotoIf("SIP/201-5451", "0?9:12") in new stack
-- Goto (macro-record-enable,s,12)
-- Executing DBget("SIP/201-5451", "RecEnable=RECORD-IN/185") in new stack
-- DBget: varname=RecEnable, family=RECORD-IN, key=185
-- DBget: Value not found in database.
-- Executing SetVar("SIP/201-5451", "CALLFILENAME=20051129-110810-1133291290.400") in new stack
-- Executing GotoIf("SIP/201-5451", "0?15:99") in new stack
-- Goto (macro-record-enable,s,99)
-- Executing NoOp("SIP/201-5451", "NO RECORDING NEEDED") in new stack
-- Executing Macro("SIP/201-5451", "dial|15|tr|185") in new stack
-- Executing GotoIf("SIP/201-5451", "0?4:2") in new stack
-- Goto (macro-dial,s,2)
-- Executing GotoIf("SIP/201-5451", "0?4:3") in new stack
-- Goto (macro-dial,s,3)
-- Executing SetCIDName("SIP/201-5451", "Dana Morris") in new stack
-- Executing AGI("SIP/201-5451", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- dialparties.agi: request = dialparties.agi
-- dialparties.agi: priority = 4
-- dialparties.agi: extension = s
-- dialparties.agi: language = en
-- dialparties.agi: accountcode =
-- dialparties.agi: uniqueid = 1133291290.400
-- dialparties.agi: channel = SIP/201-5451
dialparties.agi: callerid = Dana
-- dialparties.agi: context = macro-dial
-- dialparties.agi: type = SIP
-- dialparties.agi: rdnis = unknown
-- dialparties.agi: enhanced = 0.0
-- dialparties.agi: dnid = 185
dialparties.agi: Caller ID name is 'Dana Morris' number is '201'
-- dialparties.agi: Added extension 185 to extension map
-- dialparties.agi: Extension 185 cf is disabled
-- dialparties.agi: Extension 185 do not disturb is disabled
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
dialparties.agi: Extension 185 has call waiting disabled
-- dialparties.agi: DbSet CALLTRACE/185 to 201
dialparties.agi: Dial string is SIP/185|15|tr
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("SIP/201-5451", "SIP/185|15|tr") in new stack
-- Called 185
-- SIP/185-d099 is ringing
-- Nobody picked up in 15000 ms
-- Executing GotoIf("SIP/201-5451", "0?s-NOANSWER|1") in new stack
-- Executing GotoIf("SIP/201-5451", "0?s-NOANSWER|1") in new stack
-- Executing NoOp("SIP/201-5451", "Sending to Voicemail box 185@default") in new stack
-- Executing Macro("SIP/201-5451", "vm|185@default|NOANSWER") in new stack
-- Executing Goto("SIP/201-5451", "s-NOANSWER|1") in new stack
-- Goto (macro-vm,s-NOANSWER,1)
-- Executing VoiceMail("SIP/201-5451", "u185@default") in new stack
-- Playing 'voicemail/default/185/unavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/185/INBOX/msg0001 format: wav49, 0x9086e90
-- x=1, open writing: /var/spool/asterisk/voicemail/default/185/INBOX/msg0001 format: wav, 0x90a62b0
-- User hung up
== Spawn extension (macro-vm, s-NOANSWER, 1) exited non-zero on 'SIP/201-5451' in macro 'vm'
== Spawn extension (macro-exten-vm, s, 7) exited non-zero on 'SIP/201-5451' in macro 'exten-vm'
== Spawn extension (from-internal, 185, 1) exited non-zero on 'SIP/201-5451'
-- Executing Macro("SIP/201-5451", "hangupcall") in new stack
-- Executing ResetCDR("SIP/201-5451", "w") in new stack
-- Executing NoCDR("SIP/201-5451", "") in new stack
-- Executing Wait("SIP/201-5451", "5") in new stack
-- Executing Hangup("SIP/201-5451", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/201-5451' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-5451'
This causes the called extension to ring, and regardless of the line being picked up, or being moved to VM there is no audio on the calling end.
If someone at the office dials my soho extension, my soho phone rings, but the caller gets an “unavailable” message directly after placing the call, no ringing at all. Below is the output of that scenario:
-- Executing Macro("SIP/199-a7ab", "exten-vm|201@default|201") in new stack
-- Executing SetVar("SIP/199-a7ab", "FROMCONTEXT=exten-vm") in new stack
-- Executing Macro("SIP/199-a7ab", "record-enable|201|IN") in new stack
-- Executing GotoIf("SIP/199-a7ab", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing GotoIf("SIP/199-a7ab", "0?5:8") in new stack
-- Goto (macro-record-enable,s,8)
-- Executing GotoIf("SIP/199-a7ab", "0?9:12") in new stack
-- Goto (macro-record-enable,s,12)
-- Executing DBget("SIP/199-a7ab", "RecEnable=RECORD-IN/201") in new stack
-- DBget: varname=RecEnable, family=RECORD-IN, key=201
-- DBget: Value not found in database.
-- Executing SetVar("SIP/199-a7ab", "CALLFILENAME=20051129-111439-1133291679.410") in new stack
-- Executing GotoIf("SIP/199-a7ab", "0?15:99") in new stack
-- Goto (macro-record-enable,s,99)
-- Executing NoOp("SIP/199-a7ab", "NO RECORDING NEEDED") in new stack
-- Executing Macro("SIP/199-a7ab", "dial|15|tr|201") in new stack
-- Executing GotoIf("SIP/199-a7ab", "0?4:2") in new stack
-- Goto (macro-dial,s,2)
-- Executing GotoIf("SIP/199-a7ab", "0?4:3") in new stack
-- Goto (macro-dial,s,3)
-- Executing SetCIDName("SIP/199-a7ab", "Mila Hiles") in new stack
-- Executing AGI("SIP/199-a7ab", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- dialparties.agi: request = dialparties.agi
-- dialparties.agi: priority = 4
-- dialparties.agi: extension = s
-- dialparties.agi: language = en
-- dialparties.agi: accountcode =
-- dialparties.agi: uniqueid = 1133291679.410
-- dialparties.agi: channel = SIP/199-a7ab
dialparties.agi: callerid = Mila
-- dialparties.agi: context = macro-dial
-- dialparties.agi: type = SIP
-- dialparties.agi: rdnis = unknown
-- dialparties.agi: enhanced = 0.0
-- dialparties.agi: dnid = 201
dialparties.agi: Caller ID name is 'Mila Hiles' number is '199'
-- dialparties.agi: Added extension 201 to extension map
-- dialparties.agi: Extension 201 cf is disabled
-- dialparties.agi: Extension 201 do not disturb is disabled
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
dialparties.agi: Extension 201 has call waiting disabled
-- dialparties.agi: DbSet CALLTRACE/201 to 199
dialparties.agi: Dial string is SIP/201|15|tr
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("SIP/199-a7ab", "SIP/201|15|tr") in new stack
-- Called 201
== No one is available to answer at this time
-- Executing GotoIf("SIP/199-a7ab", "0?s-NOANSWER|1") in new stack
-- Executing GotoIf("SIP/199-a7ab", "0?s-NOANSWER|1") in new stack
-- Executing NoOp("SIP/199-a7ab", "Sending to Voicemail box 201@default") in new stack
-- Executing Macro("SIP/199-a7ab", "vm|201@default|NOANSWER") in new stack
-- Executing Goto("SIP/199-a7ab", "s-NOANSWER|1") in new stack
-- Goto (macro-vm,s-NOANSWER,1)
-- Executing VoiceMail("SIP/199-a7ab", "u201@default") in new stack
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/1' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/201/INBOX/msg0001 format: wav49, 0x90a0438
-- x=1, open writing: /var/spool/asterisk/voicemail/default/201/INBOX/msg0001 format: wav, 0x9084cc0
-- User hung up
== Spawn extension (macro-vm, s-NOANSWER, 1) exited non-zero on 'SIP/199-a7ab' in macro 'vm'
== Spawn extension (macro-exten-vm, s, 7) exited non-zero on 'SIP/199-a7ab' in macro 'exten-vm'
== Spawn extension (from-internal, 201, 1) exited non-zero on 'SIP/199-a7ab'
-- Executing Macro("SIP/199-a7ab", "hangupcall") in new stack
-- Executing ResetCDR("SIP/199-a7ab", "w") in new stack
-- Executing NoCDR("SIP/199-a7ab", "") in new stack
-- Executing Wait("SIP/199-a7ab", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/199-a7ab' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/199-a7ab'
If I dial a PSTN call (my cell phone) from the soho phone, it goes out through * box, rings my cell phone. If I answer my cell phone, I get no audio, and soho phone doesn’t get any audio either, but the call continues along as if audio was being transmitted to and fro.
Similarily, dialing into voicemail or any other IVR’s on the * box from the soho line, greeted with no audio of any kind.
I hope it’s just a config, and not an issue with bandwidth.
Asterisk Box = T1 1500/1500
Home = Cable 1500/4500
Thanks for any assistance.
day
Asterisk@home v1.5
Asterisk 1.0.9 behind router, w/ one to one NAT giving * box a public IP
SOHO public IP