[Asterisk-Users] SIP transfer/REFER to voicemail problem

[Bryan (JT) Ayers]

I've google for hours trying to find a discussion of a similar problem as the
one I'm having, so forgive me if this has come up before.  If it has, please
point me in the right direction!

The problem occurs when a caller (A) is transferred by an intermediary party
(B) to voicemail (Voicemail or VoicemailMain), either directly or by being
taken to voicemail when the callee (C) doesn't answer.  Caller (A) hears the
Asterisk voicemail prompts, but the voicemail application doesn't hear any
audio or DTMF.

Easy to duplicate:
1.) A -> B (INVITE)
2.) B -> C (REFER A to C)
3.) A -> C

More descriptive:
1.)  Caller (A) calls intermediary (B).  (B can be any SIP user agent)
2.)  Intermediary (B) REFERs caller (A) to callee (C)
3.)  C is either a SIP UA which times-out and Asterisk takes to Voicemail, or
an extension tied to VoicemailMain.

I've come across a thread saying that the Asterisk voicemail system only uses
the GSM codec, but if this were the problem, then how can the caller (using
mu-law) hear the voicemail prompts?  Would Asterisk be doing a half duplex
protocol conversion?

Any insight would be greatly appreciated!!


Current configuration:
Fedora Core 1
Asterisk - 1.0.7 (had same problem on 1.0.6)
SJPhone - 1.50.271d, Mar 11 2005  (WinXP)
XLite - 1103m build stamp 14262  (WinXP)
Zultys Zip2 - ZUTS 3.52


sip.conf exerpt:

[6003]			; (A)
type=friend
regexten=6003
username=6003
host=dynamic
disallow=all
;allow=gsm
allow=ulaw

[6004]			; (C)
type=friend
regexten=6004
username=6004
host=dynamic
disallow=all
;allow=gsm
allow=ulaw

[2101]			; (B)
type=friend
regexten=2101
username=2101
host=dynamic
disallow=all
;allow=gsm
allow=ulaw


extensions.conf exerpt:

exten => 6003,1,Dial(SIP/1003,15)
exten => 6003,2,Voicemail(u1003)
exten => 6003,102,Voicemail(b1003)

exten => 6004,1,Dial(SIP/1004,5)
exten => 6004,2,Voicemail(u1004)
exten => 6004,102,Voicemail(b1004)

exten => 2101,1,Dial(SIP/2101)

exten => 8500,1,VoicemailMain
exten => 8500,2,Hangup


Asterisk (-dvvvvvvgc) with sip debug on (REFER-ing caller to VoicemailMain) :

   -- No username but # key pressed. Using CID '6003'
   -- Playing 'vm-password' (language 'en')
Urgent handler
   -- Incorrect password '' for user '6003' (context = ,any)
   -- Playing 'vm-incorrect-mailbox' (language 'en')
Urgent handler


		
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