[Asterisk-Users] Polycom 601 Message Center
Kevin Smith
kevin.smith at mercury.net
Sat Mar 25 12:39:24 MST 2006
As far as I can tell everything is pretty much the same. Below is the
debug output for a particular phone I left a voicemail for. Maybe I am
missing something that I am just not seeing. Otherwise I'm still not
getting a count, but the other notifications are still working.
Thanks again,
Kevin
Here is the phone.cfg section:
<msg msg.bypassInstantMessage="0">
<mwi msg.mwi.1.subscribe="9897943727 at Mercury-Network-Emp"
msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="6245*"
msg.mwi.2.subscribe="" msg.mwi.2.callBackMode="disabled"
msg.mwi.2.callBack="" msg.mwi.3.subscribe=""
msg.mwi.3.callBackMode="disabled" msg.mwi.3.callBack=""
msg.mwi.4.subscribe="" msg.mwi.4.callBackMode="disabled"
msg.mwi.4.callBack="" msg.mwi.5.subscribe=""
msg.mwi.5.callBackMode="disabled" msg.mwi.5.callBack=""
msg.mwi.6.subscribe="" msg.mwi.6.callBackMode="disabled"
msg.mwi.6.callBack=""/>
</msg>
<nat nat.ip="" nat.signalPort="" nat.mediaPortStart=""/>
Here is the sip.conf for *. The Mercury-Defaults, is just some simple
rules for the sip that I applied to everyone. But the mailboxes needed
to be different for obvious reasons.
[9897943727](Mercury-Defaults)
mailbox=9897943727 at Mercury-Network-Emp
---
-- SIP/9897943727-2689 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing VoiceMail("Zap/1-1", "u9897943727 at Mercury-Network-Emp")
in new stack
Destroying call '2dca929b0f9b73282fcf1bad61ad8dbe at 64.7.161.26'
-- Playing
'/var/spool/asterisk/voicemail/Mercury-Network-Emp/9897943727/unavail'
(language 'en')
12 headers, 0 lines
Reliably Transmitting (no NAT) to 64.7.177.102:5060:
OPTIONS sip:9897943727queue at 64.7.177.102;transport=udp SIP/2.0
Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK0cf03d0b;rport
From: "asterisk" <sip:asterisk at 64.7.161.26>;tag=as18caf65c
To: <sip:9897943727queue at 64.7.177.102;transport=udp>
Contact: <sip:asterisk at 64.7.161.26>
Call-ID: 74e5b7df16de35c2751badd80b3f84d3 at 64.7.161.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 25 Mar 2006 19:30:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
voip-1*CLI>
<-- SIP read from 64.7.177.102:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK0cf03d0b;rport
From: "asterisk" <sip:asterisk at 64.7.161.26>;tag=as18caf65c
To: <sip:9897943727queue at 64.7.177.102;transport=udp>;tag=88909039-CD314322
CSeq: 102 OPTIONS
Call-ID: 74e5b7df16de35c2751badd80b3f84d3 at 64.7.161.26
Contact: <sip:9897943727queue at 64.7.177.102>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041
Content-Length: 0
--- (10 headers 0 lines)---
Destroying call '74e5b7df16de35c2751badd80b3f84d3 at 64.7.161.26'
-- Playing 'vm-intro' (language 'en')
12 headers, 0 lines
Reliably Transmitting (no NAT) to 64.7.177.102:5060:
OPTIONS sip:9897943727 at 64.7.177.102;transport=udp SIP/2.0
Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK6f26102d;rport
From: "asterisk" <sip:asterisk at 64.7.161.26>;tag=as57826c20
To: <sip:9897943727 at 64.7.177.102;transport=udp>
Contact: <sip:asterisk at 64.7.161.26>
Call-ID: 7bd39b76301000695289057f795d5e64 at 64.7.161.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 25 Mar 2006 19:30:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
voip-1*CLI>
<-- SIP read from 64.7.177.102:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK6f26102d;rport
From: "asterisk" <sip:asterisk at 64.7.161.26>;tag=as57826c20
To: <sip:9897943727 at 64.7.177.102;transport=udp>;tag=32013F46-4AA2FB
CSeq: 102 OPTIONS
Call-ID: 7bd39b76301000695289057f795d5e64 at 64.7.161.26
Contact: <sip:9897943727 at 64.7.177.102>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041
Content-Length: 0
--- (10 headers 0 lines)---
Destroying call '7bd39b76301000695289057f795d5e64 at 64.7.161.26'
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/Mercury-Network-Emp/9897943727/INBOX/msg0003
format: wav49, 0x8c77e68
-- x=1, open writing:
/var/spool/asterisk/voicemail/Mercury-Network-Emp/9897943727/INBOX/msg0003
format: gsm, 0x8c5c408
-- x=2, open writing:
/var/spool/asterisk/voicemail/Mercury-Network-Emp/9897943727/INBOX/msg0003
format: wav, 0x8ca0480
-- User ended message by pressing #
-- Playing 'auth-thankyou' (language 'en')
-- Playing 'vm-review' (language 'en')
-- Saving message as is
-- Playing 'vm-msgsaved' (language 'en')
12 headers, 3 lines
Reliably Transmitting (no NAT) to 64.7.177.102:5060:
NOTIFY sip:9897943727 at 64.7.177.102;transport=udp SIP/2.0
Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK54a0c577;rport
From: "asterisk" <sip:asterisk at 64.7.161.26>;tag=as5fddfb64
To: <sip:9897943727 at 64.7.177.102;transport=udp>
Contact: <sip:asterisk at 64.7.161.26>
Call-ID: 23167d0f77cc64fd48bc1cc80b015249 at 64.7.161.26
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 92
Messages-Waiting: yes
Message-Account: sip:asterisk at 64.7.161.26
Voice-Message: 4/0 (0/0)
---
Scheduling destruction of call
'23167d0f77cc64fd48bc1cc80b015249 at 64.7.161.26' in 15000 ms
voip-1*CLI>
<-- SIP read from 64.7.177.102:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK54a0c577;rport
From: "asterisk" <sip:asterisk at 64.7.161.26>;tag=as5fddfb64
To: <sip:9897943727 at 64.7.177.102;transport=udp>;tag=87CF00E0-8AABB82D
CSeq: 102 NOTIFY
Call-ID: 23167d0f77cc64fd48bc1cc80b015249 at 64.7.161.26
Contact: <sip:9897943727 at 64.7.177.102>
Event: message-summary
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041
Content-Length: 0
--- (10 headers 0 lines)---
Destroying call '23167d0f77cc64fd48bc1cc80b015249 at 64.7.161.26'
-- Channel 0/1, span 1 got hangup request
-- Hungup 'Zap/1-1'
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