[asterisk-users] Compiling new version libpri
If I upgrade libpri 1.4.0 to 1.4.1, do I then need to recompile asterisk even though I'm not upgrading asterisk? -- ----- Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC version for cdr?
I didn't have libtool-ltdl-devel. Once I install the devel package, it finished the configuration. Thanks James, Jared and Kai-Uwe for the responses. On 10/1/07, Kai-Uwe Jensen <[EMAIL PROTECTED]> wrote: > If this is on a RedHat-type system (EL, Fedora, but also CentOS), make > sure you have a symlink in place for libltdl.so. Even though I also > had the libtool-ltdl package installed, it only provided libs and > links for /usr/lib/libltdl.so..3.1.4 and libltdl.so.3. It did not > create a symlink to a plain-jane libltdl.so library, which is what was > needed here to successfully ./configure. > > On 10/1/07, Chris Stinson <[EMAIL PROTECTED]> wrote: > > The libtool-ltdl package is installed. > -- - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC version for cdr?
The libtool-ltdl package is installed. On 10/1/07, Jared Smith <[EMAIL PROTECTED]> wrote: > On Mon, 2007-10-01 at 12:52 -0500, Chris Stinson wrote: > > I'm having an error when I try to ./configure asterisk using > > --with-odbc=/usr/lib. Below is the version of each application and the > > ./configure below that. Any help would be appreciated. > > The autoconf magic in Asterisk looks for a shared library provided by > the libtool-ltdl package (at least under Red Hat, CentOS, and Fedora), > and won't detect the ODBC libraries without it. (Yes, the build system > *should* be a little more informative about this.) > > -- > Jared Smith > Community Relations Manager > Digium, Inc. > > -- - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ODBC version for cdr?
I'm having an error when I try to ./configure asterisk using --with-odbc=/usr/lib. Below is the version of each application and the ./configure below that. Any help would be appreciated. unixODBC-2.2.11-7.1 unixODBC-devel-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2 mysql-5.0.22-2.1 Contents of odbcinst.ini # Driver from the MyODBC package # Setup from the unixODBC package [MySQL] Description = ODBC for MySQL Driver = /usr/lib/libmyodbc.so Setup = /usr/lib/libodbcmyS.so FileUsage = 1 checking for SQLConnect in -lodbc... no configure: *** configure: *** The unixODBC installation on this system appears to be broken. configure: *** Either correct the installation, or run configure configure: *** without explicitly specifying --with-odbc -- ----- Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ODBC version
What version of ODBC does asterisk 1.4 need? -- - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] choppy playback
I'm having an issue with an asterisk install that anything recorded (in .gsm format) and all of the pre-recorded .gsm files are choppy. All calls into the asterisk box is fine and any voice mails left in a box are fine as well. It's just the playback of any recorded message and any of the pre-recorded files. Anyone have an idea what might be going on? The only problem is the playback of .gsm files. -- ----- Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] ___ --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] fax2mail
Yes you are probably right but I don't know how to rotate the fax in fax2mail. I was hoping someone here on the list had to do it and would post the solution :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Friday, January 13, 2006 3:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] fax2mail On Fri, 2006-01-13 at 15:03 -0600, Chris Stinson wrote: > Anyone here use fax2mail? Every fax get's flipped 90 degress. I was just > wondering if anyone else had this issue and how they resolved it. I generate the emails myself, I posted the macro I use to do this a week or two ago (check the archives) and dont have that problem. This makes me think that there is an option set to rotate somewhere within fax2email ... -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group ___ --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] fax2mail
Anyone here use fax2mail? Every fax get's flipped 90 degress. I was just wondering if anyone else had this issue and how they resolved it. ----- Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] ___ --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] changing "Nobody picked up in 30000 m"
In extension.conf exten => XXX,1,Dial(SIP/XXX|30) Change the 30 to 40 and the phone will ring for 4ms. The |30 is how long to ring the interface. I'm using SIP here. This is one way to change that amount but I don't know what your configuration looks like. - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: Friday, July 08, 2005 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] changing "Nobody picked up in 3 m" Where are you trying to change this? Darren Wiebe [EMAIL PROTECTED] wassim darwish wrote: >i dont know how to edit the the time for ringing >"3ms" to "4ms",please help me. > >__ >Do You Yahoo!? >Tired of spam? Yahoo! Mail has the best spam protection around >http://mail.yahoo.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 > > > ___ 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 ___ 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] Music oh hold
Does your default look like this in
musiconhold.conf, default => quietmp3:/var/lib/asterisk/mohmp3
If so, do you have any music in the
directory mohmp3?
-
Chris Stinson
Network Operations Center
ISDN-Net, Inc.
615-221-4200 x103
[EMAIL PROTECTED]
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis
Sent: Wednesday, June 29, 2005
12:35 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Music oh
hold
Sorry, i also tried this:
exten => 6000,1,Answer
exten => 6000,2,MusicOnHold(default)
and i got this result:
*CLI> --
Executing Answer("SIP/2391-8cdd", "") in new stack
-- Executing MusicOnHold("SIP/2391-8cdd",
"default") in new stack
Jun 29 19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start
music on hold (class 'default') on channel SIP/2391-8cdd
== Spawn extension (local, 6000, 2) exited non-zero on 'SIP/2391-8cdd'
Any ideas ?
Thanks
Giordano
Da:
Giordano Grandis
Inviato: mercoledì 29 giugno 2005
19.27
A: asterisk-users@lists.digium.com
Oggetto:
Hi, I installed mpg123 v0.59r without error and defined as
defaut folder /var/lib/asterisk/mohmp3. When i set a call on hold everythinghs
seem ok, but i cannot hear music. I'm using asterisk 1.0.8
*CLI> -- Executing
Dial("SIP/2339-4da6", "SIP/2391|60|Thtr") in new stack
-- Called 2391
-- SIP/2391-79a0 is ringing
-- Saved useragent "PA168S" for peer 2319
-- SIP/2391-79a0 answered SIP/2339-4da6
-- Attempting native bridge of SIP/2339-4da6 and
SIP/2391-79a0
-- Started music on hold, class 'default', on SIP/2339-4da6
-- Stopped music on hold on SIP/2339-4da6
== Spawn extension (local, 2391, 1) exited non-zero on 'SIP/2339-4da6'
Anyone can help me please ?
Thanks
Giordano
___
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] Group/Broadcast Voicemail
Were you guys able to figure this out? - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zen Kato Sent: Wednesday, June 22, 2005 4:45 PM To: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Group/Broadcast Voicemail Hi, Please see inline: In Message-ID: <[EMAIL PROTECTED]> Robert Goodyear <[EMAIL PROTECTED]> wrote : > > On Jun 22, 2005, at 1:50 PM, Zen Kato wrote: > > > Hi Robert, > > > >> Let me guess... mailbox 5103 or 5203 were the last in the list to > >> receive it? > > > > Every trials(1-6) I got only 51 mailboxes copied. My quick guess is > > 256/5(u0103 and &xx03s)=51...1, so changing tmp[256] to tmp[4096] > > does not work. 'Pseudo-diagram' as you mentioned before(6/8/05) > > is desirable for expandability, but it also did not work. > > > > > > So what about the variable BASEMAXINLINE? Did you change that and > recompile yet? Yes, I changed #define BASEMAXLINE on step 5(line 80) and step 6(line 82) and recomiled each case. Regards, Zen Kato ___ 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 ___ 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] SIP call doesn't execute the 's'-extension
You only have a 1 in the javaAgi context and you aren't point the javaAgi to any other contexts, pressing anyting else but 1 will get a not found error because you only have 1 defined. If you want the call to continue you need to send it to another context or add more to the javaAgi context. Tobias Wolf wrote: Hi, i have just started to configure access to the * over SIP-Phones. Therefore I have defined this SIP-Phone in sip.conf: [tobias] type=friend username=tobias secret=tobias auth=md5 host=dynamic reinvite=no dtmfmode=inband callerid="Tobias" <1087006> allow=all context=javaAgi dtmfmode=rfc2833 As you can see i am directing calls from this user to the context [javaAgi] which is defined here in extension.conf: [javaAgi] exten => s,1,Answer() exten => s,2,Playback(code1000) exten => s,3,Hangup() exten => 1,1,Answer() exten => 1,2,Playback(code1000) exten => 1,3,Hangup() If i dial 1 on my SIP Phone everything works as suspected, the call is answered and the gsm-file is played. My understanding of the 's'-extension is, that it is executed then a call comes in an there is no extension wich matches the called number. But if i dial a random number i get an "404 Not found" error. Here is an snippet of what * tells me on sip debug, but i can't get a clue out of it: 12 headers, 13 lines Using latest request as basis request Sending to 10.3.4.98 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 10.3.4.98:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x60e(GSM|ULAW|ALAW|SPEEX|ILBC)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user 'tobias' Looking for 2 in javaAgi Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.3.4.98:5060;branch=z9hG4bK102F7CD4855F4C4E927C3398E3C57BF4 From: Tobias ;tag=2760968676 To: ;tag=as396962de Call-ID: [EMAIL PROTECTED] CSeq: 58303 INVITE User-Agent: evision PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 Perhaps anyone can point me to the right direction ?? Tobias ___ 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 -- - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [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
Re: [Asterisk-Users] Group/Broadcast Voicemail
I did change char tmp[4096], *ext; to 4096 but there's also the same
line under vm_execmain but I really don't know anything about
programming. I only saw the same line.
Robert Goodyear wrote:
On Jun 9, 2005, at 5:14 PM, Chris Stinson wrote:
Robert Goodyear wrote:
On Jun 9, 2005, at 2:55 PM, Robert Goodyear wrote:
On Jun 9, 2005, at 12:45 PM, Chris Stinson wrote:
I was told to change "in app_voicemail.c in the function vm_exec
set the tmp[256] to be tmp[4096]" in an earlier replay so I did.
"static int vm_exec(struct ast_channel *chan, void *data)
{
int res=0, silent=0, busy=0, unavail=0;
struct localuser *u;
char tmp[4096], *ext;"
I guess it has to be changed somewhere else. It's on 4096 right now
under the vm_exec. Evidently it needs to be changed elsewhere.
Noted, but I was wondering if you could try to shorten the arguments
to see if that is, in fact, the issue before mucking around with
source and recompiling.
In the spirit of the aforementioned mucking around, it feels like
BASEMAXINLINE might be the culprit. I am NOT a C guy, but just
looking at it and then where BASEMAXINLINE is called (linked list of
users) looks like it might pay off. Try messing with that constant
and see what blows up :-)
-Rob.
Well, since I don't know jack about programming I will try to cut it
down some :)
So... any luck? If you can't adjust that list of users in the dialplan,
let me know and I'll play with the code and recompile.
/rg
Robert Goodyear
Brand Up LLC
http://www.brand-up.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
___
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] Group/Broadcast Voicemail
No, I can't get characters below 256.
Robert Goodyear wrote:
On Jun 9, 2005, at 5:14 PM, Chris Stinson wrote:
Robert Goodyear wrote:
On Jun 9, 2005, at 2:55 PM, Robert Goodyear wrote:
On Jun 9, 2005, at 12:45 PM, Chris Stinson wrote:
I was told to change "in app_voicemail.c in the function vm_exec
set the tmp[256] to be tmp[4096]" in an earlier replay so I did.
"static int vm_exec(struct ast_channel *chan, void *data)
{
int res=0, silent=0, busy=0, unavail=0;
struct localuser *u;
char tmp[4096], *ext;"
I guess it has to be changed somewhere else. It's on 4096 right now
under the vm_exec. Evidently it needs to be changed elsewhere.
Noted, but I was wondering if you could try to shorten the arguments
to see if that is, in fact, the issue before mucking around with
source and recompiling.
In the spirit of the aforementioned mucking around, it feels like
BASEMAXINLINE might be the culprit. I am NOT a C guy, but just
looking at it and then where BASEMAXINLINE is called (linked list of
users) looks like it might pay off. Try messing with that constant
and see what blows up :-)
-Rob.
Well, since I don't know jack about programming I will try to cut it
down some :)
So... any luck? If you can't adjust that list of users in the dialplan,
let me know and I'll play with the code and recompile.
/rg
Robert Goodyear
Brand Up LLC
http://www.brand-up.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
___
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] Group/Broadcast Voicemail
Well, since I don't know jack about programming I will try to cut it
down some :)
Robert Goodyear wrote:
On Jun 9, 2005, at 2:55 PM, Robert Goodyear wrote:
On Jun 9, 2005, at 12:45 PM, Chris Stinson wrote:
I was told to change "in app_voicemail.c in the function vm_exec set
the tmp[256] to be tmp[4096]" in an earlier replay so I did.
"static int vm_exec(struct ast_channel *chan, void *data)
{
int res=0, silent=0, busy=0, unavail=0;
struct localuser *u;
char tmp[4096], *ext;"
I guess it has to be changed somewhere else. It's on 4096 right now
under the vm_exec. Evidently it needs to be changed elsewhere.
Noted, but I was wondering if you could try to shorten the arguments
to see if that is, in fact, the issue before mucking around with
source and recompiling.
In the spirit of the aforementioned mucking around, it feels like
BASEMAXINLINE might be the culprit. I am NOT a C guy, but just looking
at it and then where BASEMAXINLINE is called (linked list of users)
looks like it might pay off. Try messing with that constant and see what
blows up :-)
-Rob.
Robert Goodyear
Brand Up LLC
http://www.brand-up.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
___
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] Group/Broadcast Voicemail
I was told to change "in app_voicemail.c in the function vm_exec set the
tmp[256] to be tmp[4096]" in an earlier replay so I did.
"static int vm_exec(struct ast_channel *chan, void *data)
{
int res=0, silent=0, busy=0, unavail=0;
struct localuser *u;
char tmp[4096], *ext;"
I guess it has to be changed somewhere else. It's on 4096 right now
under the vm_exec. Evidently it needs to be changed elsewhere.
Robert Goodyear wrote:
On Jun 9, 2005, at 7:33 AM, Chris Stinson wrote:
Here's what it looks like Robert
-- Executing VoiceMail("SIP/6153245827-0a2e",
"[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&
[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&8
[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&83
[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&840
@mcdstores&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&845@
mcdstores&[EMAIL PROTECTED]@mcdstores") in new stack
-- Playing 'vm-intro' (language 'en')
-- SIP/6153245805-d694 answered SIP/207.65.117.4-bf434468
Do you think there's any coincidence that exten 838, where you indicate
the last vm is copied to, falls right around character 256 of that
argument?
I would experiment by temporarily shortening the contexts to q (for
headquarters) and s (for stores) and trying again. That would shorten
the argument you're sending to the vm app considerably and would give
proof if this is or isn't the issue.
Let me know... I'm very curious now!
Robert Goodyear
Brand Up LLC
http://www.brand-up.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
--
-
Chris Stinson
Network Operations Center
ISDN-Net, Inc.
615-221-4200 x103
[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
Re: [Asterisk-Users] Group/Broadcast Voicemail
Here's what it looks like Robert
-- Executing VoiceMail("SIP/6153245827-0a2e",
"[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]&[EMAIL PROTECTED]@mcdstores")
in new stack
-- Playing 'vm-intro' (language 'en')
-- SIP/6153245805-d694 answered SIP/207.65.117.4-bf434468
-- Attempting native bridge of SIP/207.65.117.4-bf434468 and
SIP/6153245805-d694
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/mcdhq/801/INBOX/msg format: wav49,
0x958fc80
-- x=1, open writing:
/var/spool/asterisk/voicemail/mcdhq/801/INBOX/msg format: gsm, 0x9590c48
-- x=2, open writing:
/var/spool/asterisk/voicemail/mcdhq/801/INBOX/msg format: wav, 0x94e4358
-- User hung up
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message:
Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED]
You can see there's about 33 voicemail accounts but it will only copy to
about 22 of the boxes.
Robert Goodyear wrote:
On Jun 8, 2005, at 6:14 PM, Chris Stinson wrote:
So, anyone else have any ideas? I tried the below suggestion and it's
still only sending out 20 of the 32 voicemails.
C F wrote:
did you recompile afterwards? by doing make clean make make install
On 5/2/05, Chris Stinson <[EMAIL PROTECTED]> wrote:
Still only doing 20 voicemails. Thanks for the suggestion.
-
Here's a weird idea. Can you put each group of 20 users into a
distribution group whose distributOR is a member of a distribution group
itself?
Pseudo-diagram, assuming: 400 is the master VM broadcaster and 5600
through 5631 a
Re: [Asterisk-Users] Group/Broadcast Voicemail
Tried that. Didn't work. Robert Goodyear wrote: On Jun 8, 2005, at 6:14 PM, Chris Stinson wrote: So, anyone else have any ideas? I tried the below suggestion and it's still only sending out 20 of the 32 voicemails. C F wrote: did you recompile afterwards? by doing make clean make make install On 5/2/05, Chris Stinson <[EMAIL PROTECTED]> wrote: Still only doing 20 voicemails. Thanks for the suggestion. - Here's a weird idea. Can you put each group of 20 users into a distribution group whose distributOR is a member of a distribution group itself? Pseudo-diagram, assuming: 400 is the master VM broadcaster and 5600 through 5631 are your 32 users. exten => 400,1,VoiceMail(u401&402&403) exten => 401,1,VoiceMail(u5600&5601&5602...&5619) exten => 402,1,VoiceMail(u5620&5621&5622...&5639) Wonder if that would work? Robert Goodyear Brand Up LLC http://www.brand-up.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 ___ 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] Group/Broadcast Voicemail
So, anyone else have any ideas? I tried the below suggestion and it's still only sending out 20 of the 32 voicemails. C F wrote: did you recompile afterwards? by doing make clean make make install On 5/2/05, Chris Stinson <[EMAIL PROTECTED]> wrote: Still only doing 20 voicemails. Thanks for the suggestion. ----- Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] Eric Wieling aka ManxPower wrote: in app_voicemail.c in the function vm_exec set the tmp[256] to be tmp[4096] Chris Stinson wrote: I have one with 33. but I can't get the voicemail to copy to more than 20 mailboxes. Eric Wieling aka ManxPower wrote: Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes? ___ 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 ___ 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 ___ 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
[Asterisk-Users] Voicemails not deleting
Has anyone experienced when someone listens to a voicemail and they press 7 and it tells the user the voicemail has been delete but it only put it in the Old folder? I have noticed that when deleting a voicemail and I leave one at the same time there's an error message can't write to file and the voicemail doesn't delete. ___ 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
[Asterisk-Users] DISA
We are using DISA with local SIP users. The user enters in a 2 digit
code then they get a dialtone and the phone dials out. The problem is
that the calls waits 10 seconds after the outgoing number is dialed, no
matter what I put for the timeout values. Anyone else using DISA that
has run into this?
exten => _2X,1,Answer
exten => _2X,2,DigitTimeout(2)
exten => _2X,3,ResponseTimeout(2)
exten => _2X,4,SetAccount(1${EXTEN})
exten => _2X,5,SetCDRUserField(${SIPCALLID})
exten => _2X,6,DISA(no-password|)
--
-
Chris Stinson
Network Operations Center
ISDN-Net, Inc.
615-221-4200 x103
[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
[Asterisk-Users] DISA
We are using DISA with local SIP users. The user enters in a 2 digit
code then they get a dialtone and the phone dials out. The problem is
that the calls waits 10 seconds after the outgoing number is dialed, no
matter what I put for the timeout values. Anyone else using DISA that
has run into this?
exten => _2X,1,Answer
exten => _2X,2,DigitTimeout(2)
exten => _2X,3,ResponseTimeout(2)
exten => _2X,4,SetAccount(1${EXTEN})
exten => _2X,5,SetCDRUserField(${SIPCALLID})
exten => _2X,6,DISA(no-password|)
--
-
Chris Stinson
Network Operations Center
ISDN-Net, Inc.
615-221-4200 x103
[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
Re: [Asterisk-Users] Group/Broadcast Voicemail
Yes, I recompiled asterisk. - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] C F wrote: did you recompile afterwards? by doing make clean make make install On 5/2/05, Chris Stinson <[EMAIL PROTECTED]> wrote: Still only doing 20 voicemails. Thanks for the suggestion. - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] Eric Wieling aka ManxPower wrote: in app_voicemail.c in the function vm_exec set the tmp[256] to be tmp[4096] Chris Stinson wrote: I have one with 33. but I can't get the voicemail to copy to more than 20 mailboxes. Eric Wieling aka ManxPower wrote: Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes? ___ 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 ___ 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 ___ 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] Group/Broadcast Voicemail
Still only doing 20 voicemails. Thanks for the suggestion. - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] Eric Wieling aka ManxPower wrote: in app_voicemail.c in the function vm_exec set the tmp[256] to be tmp[4096] Chris Stinson wrote: I have one with 33. but I can't get the voicemail to copy to more than 20 mailboxes. Eric Wieling aka ManxPower wrote: Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes? ___ 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] Group/Broadcast Voicemail
I have one with 33. but I can't get the voicemail to copy to more than 20 mailboxes. ----- Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] Eric Wieling aka ManxPower wrote: Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes? ___ 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 ___ 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
[Asterisk-Users] Voicemail Broadcasts
Is there a limit to how many voicemail boxes you can copy a voicemail to? I have a group that has about 40 members and it only copies to voicemail to 20 of them. ___ 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
[Fwd: Re: [Fwd: [Asterisk-Users] Voicemails stopping]]
The voicemail is there. When a user hits hit message button and enters in his password the voicemail will play but sometimes will stop and go back to the menu. The user will hit 5 (replay the current message) and the voicemail will either play all the way through or stop again. The user will press 5 once again and the voicemail will go all the way through. Nobody has had this problem before? Original Message Subject: Re: [Fwd: [Asterisk-Users] Voicemails stopping] Date: Wed, 27 Apr 2005 13:37:15 -0700 From: Michael D Schelin <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion Organization: SHELCOMM To: Asterisk Users Mailing List - Non-Commercial Discussion References: <[EMAIL PROTECTED]> Chris, Try upgrading Lenux. I did mine with Yum Update and now I got voice. Chris Stinson wrote: Has anyone else had this issue? Original Message Subject: [Asterisk-Users] Voicemails stopping Date: Tue, 26 Apr 2005 13:04:55 -0500 From: Chris Stinson <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Organization: ISDN-Net, Inc. To: asterisk-users@lists.digium.com Has anyone ever had an issue with a voicemail cutting off and then going to the menu, then by pressing 5 the voicemail will play a bit further then cut off again? After hitting 5 once more it will play the rest of the voicemail message. ___ 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 ___ 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: [Fwd: [Asterisk-Users] Voicemails stopping]
The voicemail is there. When a user hits hit message button and enters in his password the voicemail will play but sometimes will stop and go back to the menu. The user will hit 5 (replay the current message) and the voicemail will either play all the way through or stop again. The user will press 5 once again and the voicemail will go all the way through. Nobody has had this problem before? Michael D Schelin wrote: Chris, Try upgrading Lenux. I did mine with Yum Update and now I got voice. Chris Stinson wrote: Has anyone else had this issue? Original Message Subject: [Asterisk-Users] Voicemails stopping Date: Tue, 26 Apr 2005 13:04:55 -0500 From: Chris Stinson <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Organization: ISDN-Net, Inc. To: asterisk-users@lists.digium.com Has anyone ever had an issue with a voicemail cutting off and then going to the menu, then by pressing 5 the voicemail will play a bit further then cut off again? After hitting 5 once more it will play the rest of the voicemail message. ___ 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 ___ 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
[Fwd: [Asterisk-Users] Voicemails stopping]
Has anyone else had this issue? Original Message Subject: [Asterisk-Users] Voicemails stopping Date: Tue, 26 Apr 2005 13:04:55 -0500 From: Chris Stinson <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Organization: ISDN-Net, Inc. To: asterisk-users@lists.digium.com Has anyone ever had an issue with a voicemail cutting off and then going to the menu, then by pressing 5 the voicemail will play a bit further then cut off again? After hitting 5 once more it will play the rest of the voicemail message. -- - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [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 -- - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [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
[Asterisk-Users] Voicemails stopping
Has anyone ever had an issue with a voicemail cutting off and then going to the menu, then by pressing 5 the voicemail will play a bit further then cut off again? After hitting 5 once more it will play the rest of the voicemail message. -- - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [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
