RE: [Asterisk-Users] Voicemail Problem
It is also there ..
[EMAIL PROTECTED]:/home/sam# ls
/var/lib/asterisk/sounds/vm-goodbye.gsm/var/lib/asterisk/sounds/vm-goodbye.gsm[EMAIL PROTECTED]:/home/sam#
Regards,Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wojciech
TrycSent: Friday, February 10, 2006 10:59 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Voicemail Problem
You are looking for vn-goodbye, most likely under
sounds/vm
W
- Original Message -
From:
Sam Lee
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 09, 2006 9:21
PM
Subject: RE: [Asterisk-Users] Voicemail
Problem
Strange thing that , its there !
[EMAIL PROTECTED]:/home/sam# ls
/var/lib/asterisk/sounds/goodbye.gsm/var/lib/asterisk/sounds/goodbye.gsm
[EMAIL PROTECTED]:/home/sam#
That's why i found it very strange. Thanks for replying.
Are there any other ideas ?
Regards,Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wojciech
TrycSent: Friday, February 10, 2006 9:59 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: 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
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 09, 2006 8:38
PM
Subject: RE: [Asterisk-Users] Voicemail
Problem
Hey guys,
Any hint at all ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam
LeeSent: Thursday, February 09, 2006 3:30 PMTo: asterisk-users@lists.digium.comSubject:
[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 stackFeb 9 15:05:06 WARNING[23242]: file.c:509
ast_openstream_full: File Goodbye does not exist in any formatFeb
9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile: Unable to open Goodbye
(format alaw): No such file or directoryFeb 9 15:05:06
WARNING[23242]: app_playback.c:132 playback_exec: ast_streamfile failed on
SIP/203.125.68.66-081ee3d8for 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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RE: [Asterisk-Users] How come I don't have the MeetMe applicationregistered?
After installing the timing source , what do I have to do to get meetme application registered? Do I have to recompile asterisk again ? I don't see the compiled meetme.so module in the directory. Regards, Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Bockman Sent: Friday, February 10, 2006 6:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How come I don't have the MeetMe applicationregistered? Anthony Azzopardi wrote: > How come I don't have the MeetMe application registered? You need a timing source. See: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy Kevin ___ --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 ___ --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
RE: [Asterisk-Users] Voicemail Problem
Strange thing that , its there !
[EMAIL PROTECTED]:/home/sam# ls
/var/lib/asterisk/sounds/goodbye.gsm/var/lib/asterisk/sounds/goodbye.gsm
[EMAIL PROTECTED]:/home/sam#
That's why i found it very strange. Thanks for replying.
Are there any other ideas ?
Regards,Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wojciech
TrycSent: Friday, February 10, 2006 9:59 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: 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
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 09, 2006 8:38
PM
Subject: RE: [Asterisk-Users] Voicemail
Problem
Hey guys,
Any hint at all ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam
LeeSent: Thursday, February 09, 2006 3:30 PMTo: asterisk-users@lists.digium.comSubject:
[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 stackFeb 9 15:05:06
WARNING[23242]: file.c:509 ast_openstream_full: File Goodbye does not exist in
any formatFeb 9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile:
Unable to open Goodbye (format alaw): No such file or
directoryFeb 9 15:05:06 WARNING[23242]: app_playback.c:132
playback_exec: ast_streamfile failed on SIP/203.125.68.66-081ee3d8for
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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RE: [Asterisk-Users] Voicemailmain() refusing connection problem
Please help for this. I really got stuck at this. After a few tries , asterisk refuses connection anymore until the previous connection timeout. Let me know if you require more info. Regards,Sam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sam LeeSent: Thursday, February 09, 2006 3:44 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Voicemailmain() refusing connection problem I've just finish setting up OPENSER with Asterisk 1.2.2 In OPENSER, i have set extension 400 to push to asterisk, which in turn run apps VoicemailMain() I noticed after the INVITE came to asterisk, it reply to OPENSER with " We're at 203.125.68.66 port 16520 ". Right after that , it will keep on " Retransmitting #1 (no NAT) to 203.125.68.66:5060: " , all the way until the 6th time when it will give up and say " Feb 9 14:18:15 WARNING[22945]: chan_sip.c:1208 retrans_pkt: Maximum retries exceeded on transmission 731b65f6-7dec21 " I don't understand, is it waiting for some reply from OPENSER which never came ? or what ? I don't know whether its the same problem, but if i call 400 a couple of times to access the VoicemailMain() without actually going in (once i've hear the password prompt, i hangup , simulating a DoS attack) , asterisk refuses to take anymore call at extension 400 for VoicemailMain() . Please let me know if you don't understand what i mean. Please help! Regards,Sam ___ --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
RE: [Asterisk-Users] Voicemail Problem
Hey guys,
Any hint at all ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam
LeeSent: Thursday, February 09, 2006 3:30 PMTo:
asterisk-users@lists.digium.comSubject: [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 stackFeb 9 15:05:06
WARNING[23242]: file.c:509 ast_openstream_full: File Goodbye does not exist in
any formatFeb 9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile:
Unable to open Goodbye (format alaw): No such file or directoryFeb
9 15:05:06 WARNING[23242]: app_playback.c:132 playback_exec: ast_streamfile
failed on SIP/203.125.68.66-081ee3d8for 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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RE: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - longdelaybetweenanswering and ringing
You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - longdelaybetweenanswering and ringing What have you set the PSTN Dialing Delay: on the PSTN Line tab (logged in as admin advanced) ? Mine is set to 1 and it works well. Chris - Original Message - From: "Anthony Rodgers" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, February 08, 2006 9:50 PM Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - long delaybetweenanswering and ringing > Hi Jean-Michel, > > We did actually try the 'r' option, but it has no effect, as Asterisk > will only supply ringing until the dialed SIP extension answers, which > it does immediately. The 4 second delay occurs between when the > SPA-3000 answers the SIP call and then places the PSTN one. I believe > that the ringing tone is provided by the PSTN at that point. > > Regards, > -- > Anthony Rodgers > Business Systems Analyst > District of North Vancouver > Web: http://www.dnv.org > RSS Feed: http://www.dnv.org/rss.asp > > > On Feb 8, 2006, at 1:41 PM, Jean-Michel Hiver wrote: > >> Anthony Rodgers a écrit : >> >> > Greetings, >> > >> > We are currently testing a Sipura SPA-3000 as a gateway from our >> > Asterisk system to a PSTN line for 911 access. We have a number of >> > locations and want to place an SPA-3000 in each, connected to a >> > PSTN line that will provide the correct ANI/ALI information to 911 >> > for >> each >> > location. >> > >> > It all works great, except for a reasonably significant (4 seconds) >> > delay between when the SPA-3000 answers the SIP call from the >> Asterisk >> > server (immediately upon dialing, according to the Asterisk CLI) >> > and the ringing tone begins (the remote phone begins ringing at >> > that same time). >> > >> > The delay is enough for users to think that the phone isn't working >> > - not what you want for 911! >> > >> > Any ideas? >> >> You could use the 'r' flag in your Dial() command to simulate a >> ringing tone instantly. This is less than ideal though. Have you done >> some SIP traces (using ngrep for examples) to look when the SIP >> 'ringing' signal is actually being sent? >> >> Cheers, >> Jean-Michel. >> >> -- >> Jean-Michel Hiver - http://ykoz.net/ >> Découvrez la Réunion des Technologies IP & Telecom >> TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE >> >> >> ___ >> --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 > > ___ > --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 > ___ --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 ___ --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
[Asterisk-Users] Voicemailmain() refusing connection problem
I've just finish setting up OPENSER with Asterisk 1.2.2 In OPENSER, i have set extension 400 to push to asterisk, which in turn run apps VoicemailMain() I noticed after the INVITE came to asterisk, it reply to OPENSER with " We're at 203.125.68.66 port 16520 ". Right after that , it will keep on " Retransmitting #1 (no NAT) to 203.125.68.66:5060: " , all the way until the 6th time when it will give up and say " Feb 9 14:18:15 WARNING[22945]: chan_sip.c:1208 retrans_pkt: Maximum retries exceeded on transmission 731b65f6-7dec21 " I don't understand, is it waiting for some reply from OPENSER which never came ? or what ? I don't know whether its the same problem, but if i call 400 a couple of times to access the VoicemailMain() without actually going in (once i've hear the password prompt, i hangup , simulating a DoS attack) , asterisk refuses to take anymore call at extension 400 for VoicemailMain() . Please let me know if you don't understand what i mean. Please help! Regards,Sam ___ --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
[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 stackFeb 9 15:05:06
WARNING[23242]: file.c:509 ast_openstream_full: File Goodbye does not exist in
any formatFeb 9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile:
Unable to open Goodbye (format alaw): No such file or directoryFeb
9 15:05:06 WARNING[23242]: app_playback.c:132 playback_exec: ast_streamfile
failed on SIP/203.125.68.66-081ee3d8for 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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