[Asterisk-Users] SER+ASTERISK voicemail
Hello, I set SER as sip proxy and ASTERISK as voicemail server (ARA) and serweb as TUI (telephone user interface) . Serweb | Ua---ser---asterisk voicemail | | Mysql DB I add user agents with address sip:[EMAIL PROTECTED] + aliases sip:[EMAIL PROTECTED] where 123 is mailbox I can forward voice messages to Asterisk with failure route for status 408 or 486. However I can't do it for offline users because of SER look for addresses like sip:[EMAIL PROTECTED] not sip:[EMAIL PROTECTED] where 123 is mailbox How could I solve this problem if possible ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER + ASTERISK voicemail
Hello,
Thanks for help it's ok with static file
voicemail.conf
However something is wrong with ARA .
app_voicemail search entries in voicemail.conf ?!
I set apps/Makefile for USE_ODBC_STORAGE.
Regards
Harry
//
Connected to Asterisk CVS-HEAD currently running on
serveur1 (pid = 2584)
Verbosity is at least 3
-- Executing VoiceMail(SIP/asterisk-8db8, b84)
in new stack
Aug 29 16:11:40 WARNING[7947]: app_voicemail.c:2602
leave_voicemail: No entry in voicemail config file
for '84'
Aug 29 16:11:50 WARNING[7947]: pbx.c:2336
__ast_pbx_run: Timeout, but no rule 't' in context
'loc al'
serveur1*CLI odbc show
Name: asterisk
DSN: asterisk
Connected: yes
serveur1*CLI
///
--- Steve Blair [EMAIL PROTECTED] a écrit :
You'll want some rules in your sip.conf to handle
the connection from
SER. A
starting point might be:
[ser ip addr:ser port ?= 5060]
type=peer
context=my sip context name
tos=lowdelay; tos delay
allow=ulaw ; dtmfmode=inband
only works with ulaw
or alaw!
dtmfmode=inband; Choices are
inband, rfc2833, or info
You'll then want some rules in extensions.conf to
accept the call and
redirect it
to mailboxes defined in your voicemail.conf or in
MySQL. Something like:
[general]
context=my sip context name
switch = Realtime/my sip context
name@extensions
static=yes
[my sip context name]
exten = _uX,1,VoiceMail(${EXTEN}@my sip
context name)
exten = _X,1,VoiceMail(${EXTEN}@my sip
context name)
exten = _bX,1,VoiceMail(${EXTEN}@my sip
context name))
exten = #,2,Hangup ; Hang
them up.
Steve
harry gaillac wrote:
Hello,
I try set Ua---SERAsterisk (voicemail/ARA)
|
Ua
ser stable
asterisk cvs head
I read
http://mail.iptel.org/pipermail/serusers/2005-February/015997.html
to forward unavailable or busy sip agents to
asterisk
voicemail in failure route.
How may I configure extensions.conf and ser.cfg ?
I have been trying without success!
Regards
Harry
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nouveau Yahoo! Messenger
Téléchargez cette version sur
http://fr.messenger.yahoo.com
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[Asterisk-Users] SER + ASTERISK voicemail
Hello, I try set Ua---SERAsterisk (voicemail/ARA) | Ua ser stable asterisk cvs head I read http://mail.iptel.org/pipermail/serusers/2005-February/015997.html to forward unavailable or busy sip agents to asterisk voicemail in failure route. How may I configure extensions.conf and ser.cfg ? I have been trying without success! Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER + ASTERISK voicemail
You'll want some rules in your sip.conf to handle the connection from
SER. A
starting point might be:
[ser ip addr:ser port ?= 5060]
type=peer
context=my sip context name
tos=lowdelay; tos delay
allow=ulaw ; dtmfmode=inband only works with ulaw
or alaw!
dtmfmode=inband; Choices are inband, rfc2833, or info
You'll then want some rules in extensions.conf to accept the call and
redirect it
to mailboxes defined in your voicemail.conf or in MySQL. Something like:
[general]
context=my sip context name
switch = Realtime/my sip context name@extensions
static=yes
[my sip context name]
exten = _uX,1,VoiceMail(${EXTEN}@my sip context name)
exten = _X,1,VoiceMail(${EXTEN}@my sip context name)
exten = _bX,1,VoiceMail(${EXTEN}@my sip context name))
exten = #,2,Hangup ; Hang them up.
Steve
harry gaillac wrote:
Hello,
I try set Ua---SERAsterisk (voicemail/ARA)
|
Ua
ser stable
asterisk cvs head
I read
http://mail.iptel.org/pipermail/serusers/2005-February/015997.html
to forward unavailable or busy sip agents to asterisk
voicemail in failure route.
How may I configure extensions.conf and ser.cfg ?
I have been trying without success!
Regards
Harry
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com
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[Asterisk-Users] SER - Asterisk voicemail on busy/unavailable. Anyone did it? (googling says NO)
Hello all! I googled lists.digium.com and ser mailing list, but did
not find any working configuration of asterisk used as voicemail for
SER. This is my config
if (uri==myself) {
if (method==REGISTER) {
save(location);
log (1, Registered\n);
break;
};
if (lookup(location)) {
log (1, *** IP to IP call *);
if (method == INVITE){
setflag (1);
t_on_failure(1);
t_relay();
sl_send_reply (180, Ringing);
setflag (1);
break;
}
if (!t_relay()) {
sl_send_reply(404, Not Found);
break;
};
#};
break;
};
failure_route[1] {
revert_uri();
forward(69.70.x.x,5060);
break();
}
Asterisk sip.conf:
[ser]
host=69.70.x.x
context=ser
type=friend
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723.1
allow=gsm
allow=ilbc
nat=yes
extensions.conf:
[ser]
include = vm
include = messagecenter
[vm]
exten = _9.,1,VoiceMail(u${EXTEN})
exten = _9.,2,Hangup
[messagecenter]
exten = 555,1,Answer
exten = 555,2,Wait(1)
exten = 555,3,VoiceMailMain(default)
exten = 555,4,Hangup
exten = _555X.,1,Answer; can dial 555exten
to skip 'mailbox' prompt. Useful for speedial.
exten = _555X.,2,Wait(1)
exten = _555X.,3,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])
exten = _555X.,4,Hangup
All SER calls 9xxx must go to asterisk, and it does, but I get the
following in aster log:
to 69.70.7.174:5060
Mar 6 18:41:36 WARNING[3539]: chan_sip.c:695 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for seqno 1
(Non-critical Response)
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/home/asterisk/var/spool/asterisk/voicemail/default/900/INBOX/msg0005
format: wav49, 0x814cb60
-- x=1, open writing:
/home/asterisk/var/spool/asterisk/voicemail/default/900/INBOX/msg0005
format: gsm, 0x814d068
-- x=2, open writing:
/home/asterisk/var/spool/asterisk/voicemail/default/900/INBOX/msg0005
format: wav, 0x8144980
Mar 6 18:41:45 WARNING[3539]: app.c:619 ast_play_and_record: No audio
available on SIP/69.70.x.x-08149a98??
-- User hung up
== Spawn extension (ser, 900, 1) exited non-zero on 'SIP/69.70.x.x-08149a98'
Destroying call '[EMAIL PROTECTED]'
If I use rewritehostport instead of forward, the call does not reach asterisk:
failure_route[1] {
revert_uri();
rewritehostport(69.70.x.x:5060);
t_relay()
break();
SER log:
4(11513) *** IP to IP call * 1(11506) ERROR:
t_forward_nonack: no branched for fwding
1(11506) ERROR: w_t_relay (failure mode): forwarding failed
3(11512) *** IP to IP call * 2(11509) Bye
Is there a way to do append_branch([EMAIL PROTECTED]) ?
Anyone did it? Reply pls with your config files!!
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Re: [Asterisk-Users] SER - Asterisk voicemail on busy/unavailable. Anyone did it? (googling says NO)
If I use rewritehostport instead of forward, the call does not reach asterisk:
failure_route[1] {
revert_uri();
rewritehostport(69.70.x.x:5060);
t_relay()
break();
SER log:
Your failure route should read:
failure_route[1] {
revert_uri();
rewritehostport(69.70.x.x:5060);
append_branch(); ==YOU MISSED THIS
t_relay()
break();
--
Andres
Network Admin
http://www.telesip.net
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[Asterisk-Users] SER Asterisk Voicemail
Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java Rockx using sipsak for sending mwi sip notify messages to the phone but is there a simpler way which I am blindly ignoring?? Thank you in advance, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Asterisk Voicemail
The sipsak way simply lites the MWI (or not) to indicate a message is waiting. You need to provide instructions in extensions.conf that route the call into voicemailmain. I use exten = 68007,1,VoicemailMain exten = 68007,2,Hangup -Steve Aisling O'Driscoll wrote: Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java Rockx using sipsak for sending mwi sip notify messages to the phone but is there a simpler way which I am blindly ignoring?? Thank you in advance, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
