Bryan (JT) Ayers
2005-Jun-15 11:09 UTC
[Asterisk-Users] SIP transfer/REFER to voicemail problem
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 __________________________________ Discover Yahoo! Find restaurants, movies, travel and more fun for the weekend. Check it out! http://discover.yahoo.com/weekend.html
For anyone else who might run into this, I got around the transferring to voicemail problem by putting a "canreinvite=no" line into the section for each caller's SIP address in sip.conf. Not ideal, but it works. I also had to add a "dtmfmode=inband" for my Mediatrix 1204 addresses to be able to access the voicemail commands. ------------------------------------------------------------------ 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 __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com