I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com 101 => 1234,Jason Madden,jason@siskiyoutech.com 102 => 1234,Melinda Garland,melinda@siskiyoutech.com Sean Garland, MCP+I, A+ Siskiyou Technology Consultants
Sean, I use the sendmail app on the pbx itself (redhat 9.1) with the serveremail=localhost Not a lot of overhead on this process, of course sendmail needs to be able to route to the internet to send out mail, so this can't be a private subnet only pbx. -Bryan -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Sean Garland Sent: Friday, June 11, 2004 3:03 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail problem I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com 101 => 1234,Jason Madden,jason@siskiyoutech.com 102 => 1234,Melinda Garland,melinda@siskiyoutech.com Sean Garland, MCP+I, A+ Siskiyou Technology Consultants _______________________________________________ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.701 / Virus Database: 458 - Release Date: 6/7/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.701 / Virus Database: 458 - Release Date: 6/7/2004
How do you specify sendmail, or any mail program? I changed the servermail= to equal my in-house exchange server, and allowed relaying by it's the pbx's IP address, but I still don't understand how it know where to send or what program it uses.. Thanks Sean -----Original Message----- From: public [mailto:public@iwantmonkey.com] Sent: Friday, June 11, 2004 2:50 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Voicemail problem Sean, I use the sendmail app on the pbx itself (redhat 9.1) with the serveremail=localhost Not a lot of overhead on this process, of course sendmail needs to be able to route to the internet to send out mail, so this can't be a private subnet only pbx. -Bryan -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Sean Garland Sent: Friday, June 11, 2004 3:03 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail problem I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com 101 => 1234,Jason Madden,jason@siskiyoutech.com 102 => 1234,Melinda Garland,melinda@siskiyoutech.com Sean Garland, MCP+I, A+ Siskiyou Technology Consultants _______________________________________________ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.701 / Virus Database: 458 - Release Date: 6/7/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.701 / Virus Database: 458 - Release Date: 6/7/2004 _______________________________________________ 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
I setted up my * mailbox. Howerver, when I access my mailbox by extension 8.
I cannot hear the prompt to input mailbox number and PIN. * console tells
me: "RFC3389 support imcomplete. Turn off on client if possible". Here
is
the complete log. Can anybody tell me how to let it work.
Thank you,
Wei
Jan 6 10:43:03 WARNING[6150]: chan_sip.c:2771 process_sdp: No compatible
codecs !
Jan 6 10:43:04 WARNING[6150]: chan_sip.c:2771 process_sdp: No compatible
codecs !
-- Executing Ringing("SIP/2201-76bf", "") in new stack
-- Executing Wait("SIP/2201-76bf", "2") in new stack
-- Executing VoiceMailMain("SIP/2201-76bf", "") in new
stack
-- Playing 'vm-login' (language 'en')
-- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back
from
192.
168.1.102
RFC3389: 1 bytes, level 4...
Jan 6 10:43:08 NOTICE[23567]: rtp.c:289 process_rfc3389: RFC3389 support
incomp lete. Turn off on client if possible
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
-- Username not entered
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
-- Timeout on SIP/2201-76bf
== CDR updated on SIP/2201-76bf
-- Executing Goto("SIP/2201-76bf", "#|1") in new stack
-- Goto (default,#,1)
-- Executing Playback("SIP/2201-76bf", "demo-thanks") in
new stack
-- Playing 'demo-thanks' (language 'en')
RFC3389: 1 bytes, level 4...
Jan 6 10:43:34 WARNING[23567]: file.c:548 ast_readaudio_callback:
Failed to wri
te frame
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Dear Mr. Alakavuk, I turned off VAD and silence suppression, the problem is still there. Are there any way I can try? Thank you. Wei turning off VAD and silence suppression at the client can solve this problem. Yusuf Alakavuk Teknik Danisman - Technical Consultant Grid Bilisim Teknolojileri A.S. Kustepe Mahallesi Leylak Sokak Murat Is Merkezi A Blok Kat:2 Daire:9 34387 Sisli Istanbul Turkiye Tel : +90 (212) 336 92 55 Fax : +90 (212) 266 25 50 _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wei Su Sent: 06 Ocak 2005 Persembe 20:56 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail problem I setted up my * mailbox. Howerver, when I access my mailbox by extension 8. I cannot hear the prompt to input mailbox number and PIN. * console tells me: "RFC3389 support imcomplete. Turn off on client if possible". Here is the complete log. Can anybody tell me how to let it work. Thank you, Wei
I'm trying the voicemail but I can't receive nothing in my mail account, the message records well but it does not seem to deliver anything... what I'm doing wrong? Thanxs!
Hello ! I am using asterisk at home 1.5, and i have a really big program. I setup multiple extensions, all with voicemail feature, but the voicemail does not kick in at all. Any idea what might be wrong ? I am allowing all possible codecs, but i cannot see what is possible wrong. Thank you in advance ! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050910/bf8154b9/attachment.htm
Do you mean voicemail isn't kicking in after a SIP phone has been
called, or it isn't kicking in when your trying to check voicemail?
Here's my working voicemail related configs.
sip.conf:
[1000]
type=friend
username=1000
secret=xxxxxxx
callerid=1000
mailbox=1000@internal
host=dynamic
context=internal
canreinvite=yes
nat=no
dtmfmode=rfc2833
qualify=yes
extensions.conf:
[globals]
; Define global variables here
PHONE1=SIP/2001
PHONE1VM=2001
[internal]
exten => *5,1,VoicemailMain(${CALLERIDNUM})
vociemail.conf:
[internal]
1000 => 3434223288,1000,user@domain.com
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Narcis
GRATIANU
Sent: Sunday, 11 September 2005 12:03 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Voicemail problem
Hello !
I am using asterisk at home 1.5, and i have a really big
program. I setup multiple extensions, all with voicemail feature, but
the voicemail does not kick in at all. Any idea what might be wrong ? I
am allowing all possible codecs, but i cannot see what is possible
wrong.
Thank you in advance !
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I have just setup my OPENSER to work with the asterisk 1.2.2.
I've set extension 400 in extension.conf to point to the VoicemailMain()
application
The entire program works fine, but there seems to be some problem
whenever the call is hangup, either by pushing # to exit the
VoicemailMain() apps or by hanging the phone. If the # button is push,
should Asterisk send something back to tell OPENSER to hang up the party
?
Here's the log of verbose level 3
Asterisk*CLI>
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-opts' (language 'en')
-- Playing 'vm-goodbye' (language 'en')
-- Executing Playback("SIP/210.23.1.139-081ee3d8",
"Goodbye") in new
stack
Feb 9 15:05:06 WARNING[23242]: file.c:509 ast_openstream_full: File
Goodbye does not exist in any format
Feb 9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile: Unable to
open Goodbye (format alaw): No such file or dire
ctory
Feb 9 15:05:06 WARNING[23242]: app_playback.c:132 playback_exec:
ast_streamfile failed on SIP/203.125.68.66-081ee3d8
for Goodbye
-- Executing Hangup("SIP/203.125.68.66-081ee3d8", "") in
new stack
== Spawn extension (default, 400, 3) exited non-zero on
'SIP/203.125.68.66-081ee3d8'
Asterisk*CLI>
Any idea what is this all about ?
Regards,
Sam
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Hello All I attempt use Asterisk with Tormenta 2 cards for connections to channel bank ( mux FMX2 with FXS cards which manufacture SIEMENS ) but it doesnt work. At cas signaling for 1 channel which use for connecting have ABCD 0101 ( mux have ABCD 1001) at idle state May I correct cas signaling ? Best regards Viktor Tatianin vtatian@druzhba.lviv.ua -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060209/20990f00/attachment.htm
Hey guys,
Any hint at all ?
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Sam Lee
Sent: Thursday, February 09, 2006 3:30 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voicemail Problem
I have just setup my OPENSER to work with the asterisk 1.2.2.
I've set extension 400 in extension.conf to point to the VoicemailMain()
application
The entire program works fine, but there seems to be some problem
whenever the call is hangup, either by pushing # to exit the
VoicemailMain() apps or by hanging the phone. If the # button is push,
should Asterisk send something back to tell OPENSER to hang up the party
?
Here's the log of verbose level 3
Asterisk*CLI>
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-opts' (language 'en')
-- Playing 'vm-goodbye' (language 'en')
-- Executing Playback("SIP/210.23.1.139-081ee3d8",
"Goodbye") in new
stack
Feb 9 15:05:06 WARNING[23242]: file.c:509 ast_openstream_full: File
Goodbye does not exist in any format
Feb 9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile: Unable to
open Goodbye (format alaw): No such file or dire
ctory
Feb 9 15:05:06 WARNING[23242]: app_playback.c:132 playback_exec:
ast_streamfile failed on SIP/203.125.68.66-081ee3d8
for Goodbye
-- Executing Hangup("SIP/203.125.68.66-081ee3d8", "") in
new stack
== Spawn extension (default, 400, 3) exited non-zero on
'SIP/203.125.68.66-081ee3d8'
Asterisk*CLI>
Any idea what is this all about ?
Regards,
Sam
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Strange thing that , its there !
root@asterisk:/home/sam# ls /var/lib/asterisk/sounds/goodbye.gsm
/var/lib/asterisk/sounds/goodbye.gsm
root@asterisk:/home/sam#
That's why i found it very strange. Thanks for replying. Are there any
other ideas ?
Regards,
Sam
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wojciech
Tryc
Sent: Friday, February 10, 2006 9:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail Problem
You don't have 'vm-goodbye' voice file. Check under
/var/lib/asterisk/sounds....
Wojtek
----- Original Message -----
From: Sam Lee <mailto:sam.lee@super.net.sg>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<mailto:asterisk-users@lists.digium.com>
Sent: Thursday, February 09, 2006 8:38 PM
Subject: RE: [Asterisk-Users] Voicemail Problem
Hey guys,
Any hint at all ?
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Sam Lee
Sent: Thursday, February 09, 2006 3:30 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voicemail Problem
I have just setup my OPENSER to work with the asterisk 1.2.2.
I've set extension 400 in extension.conf to point to the
VoicemailMain() application
The entire program works fine, but there seems to be some
problem whenever the call is hangup, either by pushing # to exit the
VoicemailMain() apps or by hanging the phone. If the # button is push,
should Asterisk send something back to tell OPENSER to hang up the party
?
Here's the log of verbose level 3
Asterisk*CLI>
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-opts' (language 'en')
-- Playing 'vm-goodbye' (language 'en')
-- Executing Playback("SIP/210.23.1.139-081ee3d8",
"Goodbye") in new stack
Feb 9 15:05:06 WARNING[23242]: file.c:509 ast_openstream_full:
File Goodbye does not exist in any format
Feb 9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile:
Unable to open Goodbye (format alaw): No such file or dire
ctory
Feb 9 15:05:06 WARNING[23242]: app_playback.c:132
playback_exec: ast_streamfile failed on SIP/203.125.68.66-081ee3d8
for Goodbye
-- Executing Hangup("SIP/203.125.68.66-081ee3d8", "")
in new
stack
== Spawn extension (default, 400, 3) exited non-zero on
'SIP/203.125.68.66-081ee3d8'
Asterisk*CLI>
Any idea what is this all about ?
Regards,
Sam
________________________________
_______________________________________________
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It is also there ..
root@asterisk:/home/sam# ls /var/lib/asterisk/sounds/vm-goodbye.gsm
/var/lib/asterisk/sounds/vm-goodbye.gsm
root@asterisk:/home/sam#
Regards,
Sam
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wojciech
Tryc
Sent: Friday, February 10, 2006 10:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail Problem
You are looking for vn-goodbye, most likely under sounds/vm
W
----- Original Message -----
From: Sam Lee <mailto:sam.lee@super.net.sg>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<mailto:asterisk-users@lists.digium.com>
Sent: Thursday, February 09, 2006 9:21 PM
Subject: RE: [Asterisk-Users] Voicemail Problem
Strange thing that , its there !
root@asterisk:/home/sam# ls /var/lib/asterisk/sounds/goodbye.gsm
/var/lib/asterisk/sounds/goodbye.gsm
root@asterisk:/home/sam#
That's why i found it very strange. Thanks for replying. Are
there any other ideas ?
Regards,
Sam
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wojciech
Tryc
Sent: Friday, February 10, 2006 9:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail Problem
You don't have 'vm-goodbye' voice file. Check under
/var/lib/asterisk/sounds....
Wojtek
----- Original Message -----
From: Sam Lee <mailto:sam.lee@super.net.sg>
To: Asterisk Users Mailing List - Non-Commercial
Discussion <mailto:asterisk-users@lists.digium.com>
Sent: Thursday, February 09, 2006 8:38 PM
Subject: RE: [Asterisk-Users] Voicemail Problem
Hey guys,
Any hint at all ?
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Sam Lee
Sent: Thursday, February 09, 2006 3:30 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voicemail Problem
I have just setup my OPENSER to work with the asterisk
1.2.2.
I've set extension 400 in extension.conf to point to the
VoicemailMain() application
The entire program works fine, but there seems to be
some problem whenever the call is hangup, either by pushing # to exit
the VoicemailMain() apps or by hanging the phone. If the # button is
push, should Asterisk send something back to tell OPENSER to hang up the
party ?
Here's the log of verbose level 3
Asterisk*CLI>
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-opts' (language 'en')
-- Playing 'vm-goodbye' (language 'en')
-- Executing Playback("SIP/210.23.1.139-081ee3d8",
"Goodbye") in new stack
Feb 9 15:05:06 WARNING[23242]: file.c:509
ast_openstream_full: File Goodbye does not exist in any format
Feb 9 15:05:06 WARNING[23242]: file.c:821
ast_streamfile: Unable to open Goodbye (format alaw): No such file or
dire
ctory
Feb 9 15:05:06 WARNING[23242]: app_playback.c:132
playback_exec: ast_streamfile failed on SIP/203.125.68.66-081ee3d8
for Goodbye
-- Executing Hangup("SIP/203.125.68.66-081ee3d8",
"") in new stack
== Spawn extension (default, 400, 3) exited non-zero
on 'SIP/203.125.68.66-081ee3d8'
Asterisk*CLI>
Any idea what is this all about ?
Regards,
Sam
________________________________
_______________________________________________
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________________________________
_______________________________________________
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(sent this earlier with my gmail account, think there is a problem there so I'm sending it here. If anyone replied to this, please resend it to this email address. thanks alot) I am new to asterisk and I'm setting up a test box to flesh out a switchover we're going to do at work. Right now I'm working on voicemail. I can leave a message fine, but when I attempt to listen to messages, I am having trouble. I can dial an extension for voice mail main, login, and it'll tell me how many messages I have. I press one and it will give me the date/time for the message but when playback would normally start, VMMain hangs up. The message in the * console is: (removed sound playback lines) Parsing '/var/spool/asterisk/voicemail/default/105/Old/ msg0000.txt': Found (playback some sound files) == Spawn extension (internal, 500, 1) exited non-zero on 'SIP/ joseph-0e7f' At this point it hangs up on me. I'm using X-Lite softphone for testing (can't buy any phones till we're sure we have it all working right :) ) Thanks in advance for any help you guys can give. I'll include the appropriate config stuff below as well: sip.conf: (nothing in [general]) [joseph] ;exten105 type=friend secret=welcome qualify=yes nat=no host=dynamic canreinvite=no context=internal mailbox=105@default voicemail.conf: [default] 105 => 1234,Joseph Blake,root@localhost extensions.conf: [internal] exten => 1,1,Answer() exten => 1,2,Playback(all-your-base) exten => 1,3,Hangup() exten => 105,1,Answer() exten => 105,2,Dial(SIP/joseph,30) exten => 105,3,VoiceMail(105@default) exten => 105,4,Hangup() exten => 500,1,VoiceMailMain()
Hi,
when I call the voicemail app, it starts and die suddenly. Has anyone
already had this problem?
Log:
app.c:644 ast_play_and_record: No audio available on SIP/XXXX-6fca??
-- User hung up
Tks,
D.K.
Hi Daniel can you give us more information so that it would be easy to debug. like voice mail configuration etc Thanks, GIridhar Bandi. On 4/18/06, Daniel Korndorfer <daniel.korndorfer@gmail.com> wrote:> > Hi, > when I call the voicemail app, it starts and die suddenly. Has anyone > already had this problem? > > Log: > app.c:644 ast_play_and_record: No audio available on SIP/XXXX-6fca?? > -- User hung up > > Tks, > D.K. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060419/de8e5969/attachment.htm