[asterisk-users] WHAT happened to AgentMonitorOutgoing(c) in Asterisk 1.4.5 ??
Recently installed 1.4.5, and still unfortunate to find out that the 'c'
option in AgentMonitorOutgoing() still doesn't work ('c' - change the CDR so
that the source of the call is 'Agent/agent_id'). It wont change the source
channel column in the CDR to 'Agent/agent_id' like it used to in 1.4.2 and
other releases older than that.
I hope this gets fixed in the next release, we need this for our call
monitoring script.
http://bugs.digium.com/view.php?id=10011
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Re: [asterisk-users] Error: Unable to allocate RTCP socket: Too manyopen files
This was a bug 1.4.4 It has now been fixed in Asterisk 1.4.5 Stuart Bennett wrote: Hi Yusuf A friend of mine had the same problem with a high volume site.. The problem lies with a limitation in Linux. Linux will only allow a certain amount of open files at a time. You will need to add the following line before running asterisk. ulimit -n 32768 That will set the max open files to 32768 for you.. The default is 1024, so I am sure there should be enough once setting 32768... I hope this helps.. Think it is the same problem... Give it a bash.. Stuart Bennett Technical Engineer Electrodynamics Frontline Software (Pty) Ltd Nortel and Asterisk Software Solutions http://www.electrodynamics.biz -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yusuf Sent: 15 June 2007 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Error: Unable to allocate RTCP socket: Too manyopen files Hi, I have a Intel Xeon Dual Core server, with 3 GB RAM, running Centos 5.0, Asterisk 1.4.4 and mysql 5.0. It is a kinda high-traffic box, with about 60 concurrent calls. The profile of calls on this box are: Incoming: via a Sangoma A101 via SIP from anothjer SIP server Outgoing all calls that come in are sent out via SIP to yet another SIP server. This morning I has this error: (edited) Executing [EMAIL PROTECTED]:37] Dial(Zap/11-1, SIP/[EMAIL PROTECTED]|40|L(360)) in new stack -- Setting call duration limit to 3600 seconds. -- Called [EMAIL PROTECTED] -- Call on SIP/10.65.138.105-0a67bbd8 left from hold -- SIP/10.65.138.105-0a67bbd8 answered SIP/sipCloverCSC-b7eba8a8 -- Packet2Packet bridging SIP/sipClCSC-b7eba8a8 and SIP/10.65.138.105-0a67bbd8 [Jun 15 09:21:48] WARNING[5306]: channel.c:768 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! [Jun 15 09:21:48] WARNING[5306]: chan_sip.c:3783 sip_new: Unable to allocate AST channel structure for SIP channel [Jun 15 09:21:48] NOTICE[5306]: chan_sip.c:13662 handle_request_invite: Unable to create/find SIP channel for this INVITE -- SIP/iswitch-0a69fb70 is ringing -- Call on SIP/iswitch-0a69fb70 left from hold -- SIP/iswitch-0a69fb70 is making progress passing it to SIP/sipClCSC-b7e2ec78 -- Call on SIP/iswitch-0a569528 left from hold -- SIP/iswitch-0a569528 answered Zap/9-1 [Jun 15 09:21:49] WARNING[5306]: channel.c:768 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! [Jun 15 09:21:49] WARNING[5306]: chan_sip.c:3783 sip_new: Unable to allocate AST channel structure for SIP channel [Jun 15 09:21:49] NOTICE[5306]: chan_sip.c:13662 handle_request_invite: Unable to create/find SIP channel for this INVITE -- SIP/10.65.138.103-0a8c4000 is ringing -- Call on SIP/10.65.138.103-0a8c4000 left from hold -- SIP/10.65.138.103-0a8c4000 is making progress passing it to SIP/sipClCSC-b7e62f28 -- SIP/10.65.138.103-0a8c4000 is ringing -- Call on SIP/10.65.138.103-0a8c4000 left from hold -- SIP/10.65.138.103-0a8c4000 is making progress passing it to SIP/sipClCSC-b7e62f28 -- Call on SIP/10.65.138.103-0a8c4000 left from hold -- SIP/10.65.138.103-0a8c4000 answered SIP/sipCloverCSC-b7e62f28 -- Packet2Packet bridging SIP/sipCloverCSC-b7e62f28 and SIP/10.65.138.103-0a8c4000 == Spawn extension (iaxClover, 0722269331, 37) exited non-zero on 'SIP/sipClCSC-b7e4cd58' -- Executing [EMAIL PROTECTED]:52] GotoIf(Zap/1-1, 0 ? 60) in new stack -- Executing [EMAIL PROTECTED]:53] Dial(Zap/1-1, SIP/iswitch/27117973000|40|L(360)) in new stack -- Setting call duration limit to 3600 seconds. -- Called iswitch/27117973000 [Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:04] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files [Jun 15 09:22:04] WARNING[5306]: chan_sip.c:4242 sip_alloc: Unable to create RTP audio session: Too many open files [Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:05] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:05] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files [Jun 15 09:22:05] WARNING[5306]: chan_sip.c:4242 sip_alloc: Unable to create RTP audio session: Too many open files [Jun 15 09:22:05] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:06] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:06] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files So I stopped Asterisk. I am going to increase the ulimit, also increasing the RTP range, from the default of 1 - 2. I had SElinux on
Re: [asterisk-users] Inline record
Hi Rob, (and Drew)
Thanks for that info, it helped a lot.
I've edited featuremap as detailed, and show features gives:
ubiphone*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #
Attended Transfer
One Touch Monitor *1
Disconnect Call * *
I've added the variable to [general] (although I think it should be = instead
of = according to the docs, and I've modified my Dial string to:
exten = _6.,3,Dial(${TRUNK2}/${EXTEN:1},,wW)
But on an call, I still although the DTMF is heard, it doesn't do anything that
I can tell: (numbers hidden)
Everyone is busy/congested at this time (1:0/0/1)
-- Executing Dial(SIP/227-08865c90, IAX2/ubigradout/***||wW) in
new stack
-- Called ubigradout/
-- Call accepted by 193.111.201.75 (format ulaw)
-- Format for call is ulaw
-- IAX2/ubigradout-16385 is ringing
-- IAX2/ubigradout-16385 is making progress passing it to SIP/227-08865c90
-- IAX2/ubigradout-16385 stopped sounds
-- IAX2/ubigradout-16385 answered SIP/227-08865c90
Jun 20 09:45:40 DTMF[22469]: channel.c:2350 ast_write: IAX2/ubigradout-16385 : *
Jun 20 09:45:40 DTMF[22469]: channel.c:2350 ast_write: IAX2/ubigradout-16385 : 1
-- Hungup 'IAX2/ubigradout-16385'
I'm expecting to see something about recording, and then a file to appear in
the monitor or recordings directory.
I've restarted A*k as well.. I'll try playing with which keys to use and see
if it's a dtmf issue..
A.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall
Sent: 19 June 2007 19:48
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inline record
In the features.conf file, under featuremap, add automon = *1
Then in extensions.conf...
[general]
DYNAMIC_FEATURES=automon ; Auto Monitor Calls by pressing *1
now if you press *1 while on a call, it will begin recording. Press *1 again
and it will complete the recording.
Rob
Drew Gibson wrote:
Adrian Marsh wrote:
Hi All,
Is there a way to have A*k record calls on-the-fly, at the users
request? i.e. a possible scenario:
Party A calls Party B
During the call, Party A wants to start recording the call, so presses
*, A*k announces recording.. and starting MixMonitor to a file.
Once the call is finished, then A*k emails a copy of the .wav file
over...
I know that meetme can record calls, and I've been able to record calls
from the beginning using Record and MixRecord, but can't see with Dial
how you'd have A*k listen for the *.
I know that voicemail can email saved messages
So I'm guessing this is a mix of the two..
Cheers,
Adrian
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automon
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+features.conf
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Re: [asterisk-users] ipv6 on Asterisk
Jason Ma wrote: Hi guys, Does anybody try to install IPV6 support on asterisk?I just found a patch for that but it is released on 2005,I have no idea if there is new version to support ipv6 or new patches,please advise.Thanks a lot. It is a very desirable feature that will solve a lot of problems, but for one reason or another it has largely been ignored. ___ --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] Inline record
Scrap that... Tried the Set() method and it worked, so then I moved it from
[general] to [globals] and it does now record the calls.
A.
-Original Message-
From: Adrian Marsh
Sent: 20 June 2007 10:06
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Inline record
Hi Rob, (and Drew)
Thanks for that info, it helped a lot.
I've edited featuremap as detailed, and show features gives:
ubiphone*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #
Attended Transfer
One Touch Monitor *1
Disconnect Call * *
I've added the variable to [general] (although I think it should be =
instead of = according to the docs, and I've modified my Dial string to:
exten = _6.,3,Dial(${TRUNK2}/${EXTEN:1},,wW)
But on an call, I still although the DTMF is heard, it doesn't do anything
that I can tell: (numbers hidden)
Everyone is busy/congested at this time (1:0/0/1)
-- Executing Dial(SIP/227-08865c90, IAX2/ubigradout/***||wW)
in new stack
-- Called ubigradout/
-- Call accepted by 193.111.201.75 (format ulaw)
-- Format for call is ulaw
-- IAX2/ubigradout-16385 is ringing
-- IAX2/ubigradout-16385 is making progress passing it to
SIP/227-08865c90
-- IAX2/ubigradout-16385 stopped sounds
-- IAX2/ubigradout-16385 answered SIP/227-08865c90
Jun 20 09:45:40 DTMF[22469]: channel.c:2350 ast_write: IAX2/ubigradout-16385
: *
Jun 20 09:45:40 DTMF[22469]: channel.c:2350 ast_write: IAX2/ubigradout-16385
: 1
-- Hungup 'IAX2/ubigradout-16385'
I'm expecting to see something about recording, and then a file to appear in
the monitor or recordings directory.
I've restarted A*k as well.. I'll try playing with which keys to use and
see if its a dtmf issue..
A.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall
Sent: 19 June 2007 19:48
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inline record
In the features.conf file, under featuremap, add automon = *1
Then in extensions.conf...
[general]
DYNAMIC_FEATURES=automon ; Auto Monitor Calls by pressing *1
now if you press *1 while on a call, it will begin recording. Press *1 again
and it will complete the recording.
Rob
Drew Gibson wrote:
Adrian Marsh wrote:
Hi All,
Is there a way to have A*k record calls on-the-fly, at the users
request? i.e. a possible scenario:
Party A calls Party B
During the call, Party A wants to start recording the call, so presses
*, A*k announces recording.. and starting MixMonitor to a file.
Once the call is finished, then A*k emails a copy of the .wav file
over...
I know that meetme can record calls, and I've been able to record calls
from the beginning using Record and MixRecord, but can't see with Dial
how you'd have A*k listen for the *.
I know that voicemail can email saved messages
So I'm guessing this is a mix of the two..
Cheers,
Adrian
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http://www.voip-info.org/wiki/index.php?page=Asterisk+config+features.conf
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[asterisk-users] mISDN problem
Hello everybody.
I have an other problem with mISDN.
The outgoing calls goes perfect, but the incoming no.
When people call in the CLI puts that:
*CLI Jun 20 12:32:08 WARNING[2315]: chan_misdn.c:4920 chan_misdn_log:
Extension can never match, so disconnecting
this is my extensions.conf:
[general]
static=yes
writeprotect=yes
[SOME]
exten = 101,1,Dial(SIP/101,30,Ttm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,Ttm)
exten = 102,2,Hangup
include = outgoing_RDSI
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,tTwW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[outgoing_RDSI]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,tTwW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/101,30,Ttm)
[incoming]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/101,30,Ttm)
and my misdn.conf this:
[general]
misdn_init=/etc/misdn- init.conf
debug=0
ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log
bridging=no
stop_tone_after_first_digit=yes
append_digits2exten=yes
dynamic_crypt=no
crypt_prefix=**
crypt_keys=test,muh
[default]
context=misdn
language=es
musicclass=default
senddtmf=yes
far_alerting=no
allowed_bearers=all
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=0
te_choose_channel=no
pmp_l1_check=no
reject_cause=16
need_more_infos=no
nttimeout=no
method=standard
dialplan=0
localdialplan=0
cpndialplan=0
early_bconnect=yes
incoming_early_audio=no
nodialtone=no
presentation=-1
screen=-1
jitterbuffer=4000
jitterbuffer_upper_threshold=0
hdlc=no
[isdn]
ports=1
context=incoming
msns=*
I don't know if the [isdn] is well
someone how has the mISDN?¿
thanks for all
Josu Lazkano
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Re: [asterisk-users] atxfer attended transfer feature
--- Don Pobanz [EMAIL PROTECTED] wrote: I would like to know if atxfer is supported This was a little confusing for me also. A week or so ago, someone pointed out that you need to include featuremap in your extensions.conf Thanks Don I'll try that. It surprises me that such an important feature is vaguely documented in *. Pinpoint customers who are looking for what you sell. http://searchmarketing.yahoo.com/ ___ --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] Asterisk GUI
Hi List; My Question was: From where I can download the Asterisk GUI, a lot of replies we received but I did not receive from where I download it and how I compile it. Regards Bilal Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. http://new.toolbar.yahoo.com/toolbar/features/mail/index.php ___ --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] Inline record
Ah...
One question though - Obviously doesn't work for Meetme.. I know I can
pre-program meetme to record conferences, but I don't see how to let users
start the record on-the-fly.
Nothing at http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe seems to suggest
it can be done..
Can it?
A.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Marsh
Sent: 20 June 2007 10:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inline record
Scrap that... Tried the Set() method and it worked, so then I moved it from
[general] to [globals] and it does now record the calls.
A.
-Original Message-
From: Adrian Marsh
Sent: 20 June 2007 10:06
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Inline record
Hi Rob, (and Drew)
Thanks for that info, it helped a lot.
I've edited featuremap as detailed, and show features gives:
ubiphone*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #
Attended Transfer
One Touch Monitor *1
Disconnect Call * *
I've added the variable to [general] (although I think it should be =
instead of = according to the docs, and I've modified my Dial string to:
exten = _6.,3,Dial(${TRUNK2}/${EXTEN:1},,wW)
But on an call, I still although the DTMF is heard, it doesn't do anything
that I can tell: (numbers hidden)
Everyone is busy/congested at this time (1:0/0/1)
-- Executing Dial(SIP/227-08865c90, IAX2/ubigradout/***||wW)
in new stack
-- Called ubigradout/
-- Call accepted by 193.111.201.75 (format ulaw)
-- Format for call is ulaw
-- IAX2/ubigradout-16385 is ringing
-- IAX2/ubigradout-16385 is making progress passing it to
SIP/227-08865c90
-- IAX2/ubigradout-16385 stopped sounds
-- IAX2/ubigradout-16385 answered SIP/227-08865c90
Jun 20 09:45:40 DTMF[22469]: channel.c:2350 ast_write: IAX2/ubigradout-16385
: *
Jun 20 09:45:40 DTMF[22469]: channel.c:2350 ast_write: IAX2/ubigradout-16385
: 1
-- Hungup 'IAX2/ubigradout-16385'
I'm expecting to see something about recording, and then a file to appear in
the monitor or recordings directory.
I've restarted A*k as well.. I'll try playing with which keys to use and
see if it's a dtmf issue..
A.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall
Sent: 19 June 2007 19:48
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inline record
In the features.conf file, under featuremap, add automon = *1
Then in extensions.conf...
[general]
DYNAMIC_FEATURES=automon ; Auto Monitor Calls by pressing *1
now if you press *1 while on a call, it will begin recording. Press *1 again
and it will complete the recording.
Rob
Drew Gibson wrote:
Adrian Marsh wrote:
Hi All,
Is there a way to have A*k record calls on-the-fly, at the users
request? i.e. a possible scenario:
Party A calls Party B
During the call, Party A wants to start recording the call, so presses
*, A*k announces recording.. and starting MixMonitor to a file.
Once the call is finished, then A*k emails a copy of the .wav file
over...
I know that meetme can record calls, and I've been able to record calls
from the beginning using Record and MixRecord, but can't see with Dial
how you'd have A*k listen for the *.
I know that voicemail can email saved messages
So I'm guessing this is a mix of the two..
Cheers,
Adrian
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automon
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+features.conf
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[asterisk-users] zlib1g
Hi List; Why I need zlib1g to do installation for Zaptel? Will zlib1g do compression or it will what extactly do during the installation process? Regards Bilal Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --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] Asterisk GUI
http://www.tuxtone.com/index.php/VOIP:Asterisk_Install_Script On 6/20/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; My Question was: From where I can download the Asterisk GUI, a lot of replies we received but I did not receive from where I download it and how I compile it. Regards Bilal Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. http://new.toolbar.yahoo.com/toolbar/features/mail/index.php ___ --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 -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [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] zlib1g
On Wed, Jun 20, 2007 at 04:42:59AM -0700, bilal ghayyad wrote: Hi List; Why I need zlib1g to do installation for Zaptel? Will zlib1g do compression or it will what extactly do during the installation process? I don't think you need it. Where does it say you need it? You'll need it for building Asterisk later, though. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Asterisk GUI
On Wed, Jun 20, 2007 at 04:25:44AM -0700, bilal ghayyad wrote: Hi List; My Question was: From where I can download the Asterisk GUI, a lot of replies we received but I did not receive from where I download it and how I compile it. svn co http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Query regarding connecting PABX with Application server
Dear all, We are connecting the PABX with Application server.What we are trying is that when a nbr 1800 (this is not registered in PABX) is dialled the pabx should route the call to Application server .The PABX should also have intelligence to route the call by itself for its registered clients.For this scenario to work please guide us what are the files we need to change and other necessary details.If possible please also provide us the configuration that needs to be set up in XLITE sip soft phone for this service. Kindly do the needful. Thanks and regards, S.Ravi___ --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] asterisk + mediant 2000
Dear All I am new in this list right now i am working on asterisk server and deploying asterisk PBX in my organization now i have alread setup Avaya PBX and i want to intergrate my asterisk through mediant 2000 [asterisk]-[mediant 2k]E1-trunk--[Avaya] this is my setup now i want to create dialpan so how to forward call in to existing avaya setup means i have not good knowledge of dialpan routing call is there any configuration example to router call on asterisk this is possible to do in this setup suggest me Regards Satish Patel - Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. ___ --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] Res: Record CDR in a Oracle database
Hi All, Thank's for your hint Tim Panton I could connect my asterisk machine to my oracle machine. I used unixODBC-2.2.11.tar.gz, oracle-instantclient-basic-10.2.0.3-1.i386.rpm, oracle-instantclient-sqlplus-10.2.0.3-1.i386.rpm and the drive from www.oracle.com (odbc-oracle-3.1.0-linux-x86-glibc.tar) to configure my asterisk machine. I can connect to my oracle machine with isql and in the asterisk CLI I can see that the odbc is connected to oracle with the following command: asterisk*CLI odbc show Name: oracle DSN: oracle Connected: yes And I can see with netstat in my oracle machine that the my asterisk machine is connected in the DB oracle. But when I finish a call I received the following error: cdr_odbc: Connected to oracle cdr_odbc: Error in Query -1 cdr_odbc: Query FAILED Call not logged! cdr_odbc: Reconnecting to dsn oracle cdr_odbc: Connected to oracle cdr_odbc: Trying Query again! cdr_odbc: Error in Query -2 cdr_odbc: Query FAILED Call not logged! Does anyone have any ideia? Did anyone have this error?? or know how can I do this? Thank's in advanced... Everton Goularth GoVoIP - Uberlandia - MG Brasil ___ Yahoo! Mail - Sempre a melhor opção para você! Experimente já e veja as novidades. http://br.yahoo.com/mailbeta/tudonovo/ ___ --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] Agent auto congesting
Hello, I Have an agent on a queue, evry thing works normally, but after a time (about 5 minutes) my agent is pauses (agent is still regitred but can't takes calls), on Astrisk console i have the message : [Jun 20 11:55:12] NOTICE[8803]: chan_sip.c:2757 auto_congest: Auto-congesting SIP/anna-08215f68 -- SIP/anna-08215f68 is circuit-busy -- Nobody picked up in 8000 ms I think that my agent will be busy after a timeout.!!, why ?, and how can 'i modify this timeout??? If you have any idea to resolve this problem, thanks to write me. Rachid ___ --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] ChanSpy SIP
For anyone experiencing the same problem, I was able to make SpyChan work on SIP extensions using the b and v options. exten = _**.,1,ChanSpy(IAX2/1654|bv(4)) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nunez Sent: Tuesday, June 19, 2007 8:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ChanSpy SIP Has anyone succesfully tried using ChanSpy on SIP channels with the latest Asterisk 1.4? I tried ChanSpy(SIP/5060) to monitor SIP extension 5060 and the console displays, Monitoring Sip/5060, but I don't hear anything. I am able to monitor Zap channels. ___ --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] Asterisk RealTime
Hello All, I manage to configure Asterisk RealTime and now it loads the SIP users/peers from MySQL DB. The table I am using is of A2Billing DB cc_sip_buddies. Now the only problem I am facing is incoming calls are failing... The ATA which is assigned this DID number is behind NAT and according to Olle's explanations he said *there's no support for NAT keep-alives (qualify=) or voicemail indications* for these peers. http://www.voip-info.org/wiki/view/Asterisk+RealTime So does this mean that I can not have any ATA behind NAT with this kind of setup? Below is the error message. NOTE: The same setup used to work when I was using flat file config. Here is my extconfig.conf: - [settings] sipusers = mysql,mya2billing,cc_sip_buddies sippeers = mysql,mya2billing,cc_sip_buddies Here is my res_mysql.conf: - [general] dbhost = 127.0.0.1 dbname = mya2billing dbuser = billinguser dbpass = 000eFm500F9E36 dbport = 3306 dbsock = /tmp/mysql.sock When I do sip show peers on the *CLI I can see my SIP user: - hyperion*CLI sip show peers Name/username HostDyn Nat ACL Port Status 2486543210/248654321 69.148.36.78 D N 38813OK (20 ms) 1 sip peers [1 online , 0 offline] Error message while receiving the call: - -- AGI Script Executing Application: (DIAL) Options: (SIP/2486543210|60|HL(360:61000:3)) -- Limit Data for this call: -- - timelimit = 360 -- - play_warning = 61000 -- - play_to_caller= yes -- - play_to_callee= no -- - warning_freq = 3 -- - start_sound = UNDEF -- - warning_sound = timeleft -- - end_sound = UNDEF Jun 20 09:49:58 NOTICE[24952]: app_dial.c:1069 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) a2billing.php|1|did: file:Class.A2Billing.php - line:634 - [CARD STATUS UPDATE : UPDATE cc_card SET inuse=inuse-1 WHERE username='2486543210'] -- AGI Script a2billing.php completed, returning 0 Any advice... Cheers, Nitesh ___ --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] Phantom Calls
Lee Jenkins wrote on 6/19/07 9:56 AM: Vadim Berezniker wrote: Enable verbose logging for the asterisk log Set verbose level to 4 Review the log file for anything that looks like a phantom call. There should be enough information to get some idea of why this is happening. Thanks, I have done this yesterday by setting up putty to log to a file, but the customer employees have inadvertently shut it down on a couple of a occasions :) Hopefully it will be running when this happens again so I can try to track down the problem. You should be able to tell it to log to a file in addition to the console in logger.conf. Something like: full = notice,warning,error,verbose Then it should show up in /var/log/asterisk/full and you wouldn't need to keep a session open to the console to see it, just go back and look at the file later. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --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] hanging up
Is there anyway on knowing in the h extension if a call has been ended as a result of a transfer ? i.e. 1) A calls B. 2) B transfers A to C. 3) B gets hung up. 4) A talks to C at (3) i need to know if this is a normal hangup (A or B has hung up) or if it is a result of the transfer. Julian. ___ --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] DTMF doesn't work between Asterisk and Cisco SIP Proxy
Hi buddies, I encountered DTMF issue when I tried to place call from x-lite to a sip conference serice,here is the diagram. X-liteAsterisk---Cisco SIP proxySIP Conference service The Call can be established,and I can hear from x-lite the prompt of the conference,but when I input any digits,nothing happened,the conference service did not recognize my input.At the same time,in the log of asterisk,I can find that asterisk recognized all the digitsI tried rfc2833,inband,info in the dtmfmode parameter,but did not work ,I'm not sure whether asterisk send the right dtmf to cisco proxy,how can I track that? I made another test,dialing from x-lite registered with Cisco proxy to voicemail service of Asterisk. x-liteCisco SIP proxyAsterisk---Voicemail service Both the call and dtmf worked fine,I can input my mailbox number and password and listen my voicemail.both rfc2933 and inband worked in this situation,but not info. My Asterisk is 1.4.4 with asterisk now,I did not configure dtmfmode in the section of xlite and the trunk to cisco proxy,just configure the dtmfmode in sip.conf. When I used rfc2833,I can see the log in asterisk as : [2007-06-19 16:01:40] DTMF[8925] channel.c: DTMF begin '2' received on SIP/-08269470 [2007-06-19 16:01:41] DTMF[8925] channel.c: DTMF end '2' received on SIP/-08269470, duration 160 ms [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF begin '1' received on SIP/-08269470 [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF end '1' received on SIP/-08269470, duration 140 ms and when I used inband,I can see : [2007-06-19 15:55:21] DTMF[8852] channel.c: DTMF end '2' received on SIP/-09d916c0, duration 0 ms [2007-06-19 15:55:22] DTMF[8852] channel.c: DTMF end '1' received on SIP/-09d916c0, duration 0 ms Is that right?Can I check what digits that asterisk sent out ? How can I track where is wrong with the dtmf?Did asterisk send dtmf to Cisco proxy correctly? I really have no idea about that.Please advise.Thank you very much ___ --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] hanging up
at (3) i need to know if this is a normal hangup (A or B has hung up) or if it is a result of the transfer. It's normal, who start's the conference can't hangup. On 6/20/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: Is there anyway on knowing in the h extension if a call has been ended as a result of a transfer ? i.e. 1) A calls B. 2) B transfers A to C. 3) B gets hung up. 4) A talks to C at (3) i need to know if this is a normal hangup (A or B has hung up) or if it is a result of the transfer. Julian. ___ --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 -- Gabriel Lopes ___ --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] Phantom Calls
Dave Miller wrote: Lee Jenkins wrote on 6/19/07 9:56 AM: Vadim Berezniker wrote: Enable verbose logging for the asterisk log Set verbose level to 4 Review the log file for anything that looks like a phantom call. There should be enough information to get some idea of why this is happening. Thanks, I have done this yesterday by setting up putty to log to a file, but the customer employees have inadvertently shut it down on a couple of a occasions :) Hopefully it will be running when this happens again so I can try to track down the problem. You should be able to tell it to log to a file in addition to the console in logger.conf. Something like: full = notice,warning,error,verbose Then it should show up in /var/log/asterisk/full and you wouldn't need to keep a session open to the console to see it, just go back and look at the file later. Nice tip, Dave. Thanks, -- Warm Regards, Lee ___ --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] asterisk with mediant 2000 trunk
Dear All I want to integrate asterisk with mediant so anybody have configuration for this setup [asterisk]--[mediant]--[avaya] this is my setup so what is the basic configuration for this setup - Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase.___ --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] Phantom Calls
Vadim Berezniker wrote: Enable verbose logging for the asterisk log Set verbose level to 4 Review the log file for anything that looks like a phantom call. There should be enough information to get some idea of why this is happening. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Monday, June 18, 2007 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Phantom Calls Stephen Bosch wrote: Lee Jenkins wrote: I have a client that is having problems with phantom calls. I have not been able to see it happen myself, but they say when it happens, the display on the phone (polycom 301's) says Device is calling, but when they answer the phone, there is only silence and then they hang back up and it sometimes rings again. I've been trying to track this down for a couple of weeks now with no success yet. If anyone can lend a suggestion or a pointer to look for, I would greatly appreciate it. I've tried using WaitForRing() in case it is bad signaling coming from the phone company. But that has not helped. Below is the CLI output when this issue happened. As you can see, I am using WaitForRing() to discourage phantom calls. Every time this has happened, there appears to be an error getting caller ID. I'm thinking that if I insert a Wait(1/2) before Answer, that may resolve the problems with Caller ID as it looks like Asterisk is not waiting long enough for the CID to come in. Whether or not that will fix the problem with phantom calls remains to be seen after I make the changes. Also notice, the line: localhost*CLI -- Got SIP response 400 Bad Request back from 192.168.1.216. What does 400 Bad Request usually mean for sip? Generic message or something that would provide a clue? localhost*CLI -- Starting simple switch on 'Zap/3-1' localhost*CLI Jun 21 10:44:48 NOTICE[11257]: callerid.c:325 callerid_feed: Caller*ID failed checksum localhost*CLI Jun 21 10:44:51 NOTICE[11257]: chan_zap.c:6233 ss_thread: Got event 18 (Ring Begin)... localhost*CLI Jun 21 10:44:53 NOTICE[11257]: chan_zap.c:6233 ss_thread: Got event 2 (Ring/Answered)... -- Executing WaitForRing(Zap/3-1, 1) in new stack localhost*CLI -- Got a ring after the timeout -- Executing Answer(Zap/3-1, ) in new stack -- Executing Ringing(Zap/3-1, ) in new stack -- Executing SetMusicOnHold(Zap/3-1, default) in new stack -- Executing Goto(Zap/3-1, check_time|s|1) in new stack -- Goto (check_time,s,1) -- Executing Set(Zap/3-1, FAIL_MENU=daytime|TIMEOUT_MENU=daytime) in new stack -- Executing GotoIfTime(Zap/3-1, 08:30-17:00|mon-fri|*|*|?daytime|s|1) in new stack -- Goto (daytime,s,1) -- Executing Set(Zap/3-1, TIMEOUT(response)=1) in new stack -- Response timeout set to 1 -- Executing Dial(Zap/3-1, SIP/114SIP/115SIP/116SIP/117|20|tr) in new stack -- Called 114 -- Called 115 -- Called 116 -- Called 117 localhost*CLI -- SIP/117-0a0718d8 is ringing localhost*CLI -- SIP/116-0a06c398 is ringing localhost*CLI -- SIP/115-0a057678 is ringing localhost*CLI -- SIP/114-0a066c58 is ringing localhost*CLI -- Nobody picked up in 2 ms -- Executing BackGround(Zap/3-1, custom/no-answer) in new stack -- Playing 'custom/no-answer' (language 'en') localhost*CLI -- Got SIP response 400 Bad Request back from 192.168.1.216 localhost*CLI -- Timeout on Zap/3-1 == CDR updated on Zap/3-1 -- Executing Goto(Zap/3-1, daytime|s|1) in new stack -- Goto (daytime,s,1) -- Executing Set(Zap/3-1, TIMEOUT(response)=1) in new stack -- Response timeout set to 1 -- Executing Dial(Zap/3-1, SIP/114SIP/115SIP/116SIP/117|20|tr) in new stack -- Called 114 -- Called 115 -- Called 116 -- Called 117 localhost*CLI -- SIP/115-0a076e18 is ringing localhost*CLI -- SIP/116-0a07c358 is ringing localhost*CLI -- SIP/114-0a06c398 is ringing localhost*CLI -- SIP/117-0a081898 is ringing localhost*CLI -- Nobody picked up in 2 ms -- Executing BackGround(Zap/3-1, custom/no-answer) in new stack -- Playing 'custom/no-answer' (language 'en') localhost*CLI -- Got SIP response 400 Bad Request back from 192.168.1.216 localhost*CLI -- Timeout on Zap/3-1 == CDR updated on Zap/3-1 -- Executing Goto(Zap/3-1, daytime|s|1) in new stack -- Goto (daytime,s,1) -- Executing Set(Zap/3-1, TIMEOUT(response)=1) in new stack -- Response timeout set to 1 -- Executing Dial(Zap/3-1, SIP/114SIP/115SIP/116SIP/117|20|tr) in new stack -- Called 114 -- Called 115 -- Called 116 -- Called 117 localhost*CLI -- SIP/116-0a07c358 is ringing localhost*CLI -- SIP/117-0a0718d8 is ringing localhost*CLI -- SIP/115-0a057678 is ringing localhost*CLI -- SIP/114-0a066c58 is ringing localhost*CLI == Spawn extension
[asterisk-users] X-Lite problems on basic asterisk setup
I'm trying to setup my first Asterisk setup on a CentOS 5 installation on VMWare Workstation 6. Got two Linksys SPA941s working fine. But X-Lite softphones can't answer phone calls, and when one of them calls on of the Linksys phones they connect but neither party can hear hear the other. I noticed that the Linksys phones are connected via Native bridging while the X-Lite ones are connected via Packet2Packet bridging. Also, on the X-Lite phones there is a about a 30 second lag between when the X-Lite client hits dial/call and when the called party starts ringing. ::Asterisk setup:: Asterisk 1.4.4 Zaptel 1.4.3 (only ztdummy compiled) Asterisk Addons 1.4.1 CentOS 5 VMWare Workstation 6 ::sip.conf:: [Linksys01] type=friend secret=ledzep context=default host=dynamic mailbox=6445 [X-Lite01] type=friend secret=rammerjammer context=default host=dynamic dtmfmode=rfc2833 mailbox=2070 canreinvite=yes nat=no [Linksys02] type=friend secret=bigben context=default host=dynamic mailbox=6368 qualify=yes ::extenstions.conf:: [default] include = demo exten = 6445,1,Dial(SIP/Linksys01,20) exten = 6445,n,Voicemail(u6445) exten = 2070,1,Dial(SIP/X-Lite01,20) exten = 2070,n,Voicemail(u2070) exten = 2070,n,HangUp() exten = 6368,1,Answer exten = 6368,n,Ringing exten = 6368,n,Dial(SIP/Linksys02,20) exten = 6368,n,Voicemail(u6368) exten = 6368,n,HangUp() --- Andrew Stewart ___ --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] How to Create Custom Context
Hello All, Is there any way to write a custom context, where first it checks internally to see if the SIP User exist with same DID number, if it does route the call internally like calling from one extension to another extension, else pass the call to A2Billing to do the billing and use the default outbound trunks to terminate the call. Reason for this is because I have generated my SIP User with real US DID and I would like to keep the cost minimum by not sending the local calls out on Trunk and receive it back... For example, [custom-a2billing] exten = _XX,1,Answer exten = _XX,2,Wait,2 exten = _XX,3,_ _ _ _ _ _ _ ; What to fill here to check for local calls and it not found send to A2Billing.php exten = _XX,4,DeadAGI(a2billing.php|1) exten = _XX,5,Wait,2 exten = _XX,6,Hangup Any suggestions...? Cheers, Nitesh ___ --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] Forcing Dial application to skip if called server is unreachable
On Wed, 20 Jun 2007, [EMAIL PROTECTED] wrote: Is it possible to force the Dial function to skip to the next priority if it doesn't find the server of the called contact within a few seconds? I know I can use: Dial(Technology/resource[Tech2/resource2...][|timeout][|options][|URL]) where I can use some short timeout in the timeout option, but if I do so, when some call is well succeeded, it will only ring for that time! I think you basically have to pick one or the other. Either set a long timeout (15-30 sec, e.g. Dial(SIP/whatever,20) or don't use this feature. The good news is that if the destination SIP server is actually unreachable, Dial() should return almost instantly, at which point it should jump to the failure priority. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] different codec for different extensions
Hi All, I am wondering that how I can setup different codec for different extensions in my dial plan. scanario will when user X (Sip) call 111 extension in default context. The Asterisk response should be in GSM codec When user X (Sip) call 222 extension in default context. the Asterisk response should be in G711 Codec Actually I want to setup an extension for FAX receiving (rx_fax) and other for IVR. when your call FAX extension the codec will be G711 and when user call IVR the codec must be GSM Please help me Thanks Nasir Iqbal ___ --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] mISDN problem
You have only one extension in the [incoming] context and that is 's'. You probably need a different one -- the one the telco sends you... Ideas: 1. Try using a generic wildcard such as '_X.' instead of 's', then check the CLI after incrementing verbosity to at least 3 (BTW: don't forget reloading extensions!) 2. Enable misdn debugging to leve 3 and check its log at /var/log/asterisk/misdn.log. You will have the destination extension as the dad field, IIRC. Good luck -- Ex Vito On 6/20/07, Josu Lazkano [EMAIL PROTECTED] wrote: Hello everybody. I have an other problem with mISDN. The outgoing calls goes perfect, but the incoming no. When people call in the CLI puts that: *CLI Jun 20 12:32:08 WARNING[2315]: chan_misdn.c:4920 chan_misdn_log: Extension can never match, so disconnecting ___ --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] Forcing Dial application to skip if called server is unreachable
...not really sure, maybe ChanIsAvail can be of use ? -- Ex Vito On 6/20/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Is it possible to force the Dial function to skip to the next priority if it doesn't find the server of the called contact within a few seconds? I know I can use: Dial(Technology/resource[Tech2/resource2...][|timeout][|options][|URL]) where I can use some short timeout in the timeout option, but if I do so, when some call is well succeeded, it will only ring for that time! Any ideas? ___ --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] Asterisk RealTime
On 6/20/07, Nitesh Divecha [EMAIL PROTECTED] wrote: So does this mean that I can not have any ATA behind NAT with this kind of setup? Below is the error message. You may want to check out the 'rtcachefriends' option in sip.conf, and see if that solves your problem. -Jared ___ --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] Firewall on AsteriskNow
hi, Is it easy to add to the AsteriskNow system? I am looking to use it to replace my older Asterisk box but I put my asterisk box on the internet and restrict access to specific IP addresses with APF firewall.. So it would be nice if I could install APF on AsteriskNow.. Thanks.. ___ --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] Ex-Girlfriend Logic in 1.4.4
Hi,
exten = 5000/19256002182,1,Answer
exten = 5000/19256002182,n,Wait(1)
exten = 5000/19256002182,n,NoOp(${CALLERID(num)})
exten = 5000/19256002182,n,Playback(tt-monkeys)
nothing appears on the console and I get no match. You can see the ca
Try with underscore before extension like.
exten = _5000/19256002182,1,Answer
Nasir Iqbal
ICT Innovations
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Re: [asterisk-users] Asterisk GUI
On Jun 19, 2007, at 12:37 PM, Senad Jordanovic wrote: Tom Rymes wrote: [snip] How many times does it have to be said? Don't feed the trolls! Tom Tom...Who in your opinion is a troll? Senad Well, technically, I was calling the original post a troll, not the original poster. More specifically, the usage of troll I am referring to resembles the fishing technique more than the mythological creature. Basically, a troll in this context is a post that someone makes simply for the purpose of starting a heated discussion on a very touchy subject. In other words, the original poster is trolling for people who will get all bent out of shape about their post and fire back a heated response. For example, a user could post a message to the list asking I'm new to Linux and Asterisk. Should edit my dialplan by hand, use FreePBX, or buy a commercial solution? Imagine the response as you tried to convince them to buy PBXWare, FreePBX users try to convince them that they should start out using FreePBX, and others go on about how hand coding a dialplan is the one-true-way® to learn Asterisk. Generally, the original poster is just looking to get everyone stirred up over nothing. In other words, Paul's original post of GUI bad! CLI good! was just the sort of post that is going to get folks fired up re-re-restarting the age-old discussion of which is better: CLI or GUI. Basically, it could be like posting any of the following: - Which is better: emacs or vi? - Which linux distribution is the best? - Which is better: Macs or Windows? All of these questions share the following: 1.) They have no right answer (macs are better for some, Windows for others, and linux for others still, not to mention OS/2, BSD, etc) 2.) People on the various sides of the debate have extremely strong feelings on the matter 3.) Nobody is likely to be convinced that the other side is right and that they are wrong. 4.) They have all been discussed thousands of times before, and nothing new is likely to be said on the matter. 5.) The only purpose served by the discussion, due to the reasons above, is to clutter up the mailing list. 6.) Any discussion thread regarding these sorts of topics is best avoided. For a more thorough description of an internet troll, see the following wikipedia article:http://en.wikipedia.org/wiki/Troll_% 28internet%29 In other words, if you see a post that is just going to result in a re-rehashing of the last rehash of a specific subject, just hit the delete key instead of clogging up the mailing list with yet another thread on whether a GUI or a CLI is better. (for example). In Paul's defense, it looked to me like his original post was simply a joke that was misunderstood. (I thought it was funny, anyway) I suppose I should take my own advice on this one, but sometimes I guess we all just can't resist. grin Tom ___ --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] Dynamically adding Context in dialplan?
Hi, How about using asterisk real time ? http://www.voip-info.org/wiki-Asterisk+RealTime We can write a switch command for existing context but I want to new context dynamically. From asterisk CLI we can add extensions in dial-plan dynamically using dialplan add extension command. but how we can dynamically create a context in dialplan. is that possible? Nasir Iqbal ___ --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] Asterisk GUI
Tom, I disagree with your argument for a number of reasons. Each of these reasons should be more than enough to convince you I'm correct and you should do it my way and only my way. And for the record, VI and CLI. Rob Tom Rymes wrote: On Jun 19, 2007, at 12:37 PM, Senad Jordanovic wrote: Tom Rymes wrote: [snip] How many times does it have to be said? Don't feed the trolls! Tom Tom...Who in your opinion is a troll? Senad Well, technically, I was calling the original post a troll, not the original poster. More specifically, the usage of troll I am referring to resembles the fishing technique more than the mythological creature. Basically, a troll in this context is a post that someone makes simply for the purpose of starting a heated discussion on a very touchy subject. In other words, the original poster is trolling for people who will get all bent out of shape about their post and fire back a heated response. For example, a user could post a message to the list asking I'm new to Linux and Asterisk. Should edit my dialplan by hand, use FreePBX, or buy a commercial solution? Imagine the response as you tried to convince them to buy PBXWare, FreePBX users try to convince them that they should start out using FreePBX, and others go on about how hand coding a dialplan is the one-true-way® to learn Asterisk. Generally, the original poster is just looking to get everyone stirred up over nothing. In other words, Paul's original post of GUI bad! CLI good! was just the sort of post that is going to get folks fired up re-re-restarting the age-old discussion of which is better: CLI or GUI. Basically, it could be like posting any of the following: - Which is better: emacs or vi? - Which linux distribution is the best? - Which is better: Macs or Windows? All of these questions share the following: 1.) They have no right answer (macs are better for some, Windows for others, and linux for others still, not to mention OS/2, BSD, etc) 2.) People on the various sides of the debate have extremely strong feelings on the matter 3.) Nobody is likely to be convinced that the other side is right and that they are wrong. 4.) They have all been discussed thousands of times before, and nothing new is likely to be said on the matter. 5.) The only purpose served by the discussion, due to the reasons above, is to clutter up the mailing list. 6.) Any discussion thread regarding these sorts of topics is best avoided. For a more thorough description of an internet troll, see the following wikipedia article:http://en.wikipedia.org/wiki/Troll_% 28internet%29 In other words, if you see a post that is just going to result in a re-rehashing of the last rehash of a specific subject, just hit the delete key instead of clogging up the mailing list with yet another thread on whether a GUI or a CLI is better. (for example). In Paul's defense, it looked to me like his original post was simply a joke that was misunderstood. (I thought it was funny, anyway) I suppose I should take my own advice on this one, but sometimes I guess we all just can't resist. grin Tom ___ --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] mISDN problem
Thank yo very much, it works! I had a 0 before the number, in misdn.conf -natiolapreffix=0 2007/6/20, Ex Vitorino [EMAIL PROTECTED]: You have only one extension in the [incoming] context and that is 's'. You probably need a different one -- the one the telco sends you... Ideas: 1. Try using a generic wildcard such as '_X.' instead of 's', then check the CLI after incrementing verbosity to at least 3 (BTW: don't forget reloading extensions!) 2. Enable misdn debugging to leve 3 and check its log at /var/log/asterisk/misdn.log. You will have the destination extension as the dad field, IIRC. Good luck -- Ex Vito On 6/20/07, Josu Lazkano [EMAIL PROTECTED] wrote: Hello everybody. I have an other problem with mISDN. The outgoing calls goes perfect, but the incoming no. When people call in the CLI puts that: *CLI Jun 20 12:32:08 WARNING[2315]: chan_misdn.c:4920 chan_misdn_log: Extension can never match, so disconnecting ___ --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] Asterisk GUI
I would have been convinced if you had not top-posted! heh Rob Schall wrote: Tom, I disagree with your argument for a number of reasons. Each of these reasons should be more than enough to convince you I'm correct and you should do it my way and only my way. And for the record, VI and CLI. Rob Tom Rymes wrote: On Jun 19, 2007, at 12:37 PM, Senad Jordanovic wrote: Tom Rymes wrote: [snip] How many times does it have to be said? Don't feed the trolls! Tom Tom...Who in your opinion is a troll? Senad Well, technically, I was calling the original post a troll, not the original poster. More specifically, the usage of troll I am referring to resembles the fishing technique more than the mythological creature. Basically, a troll in this context is a post that someone makes simply for the purpose of starting a heated discussion on a very touchy subject. In other words, the original poster is trolling for people who will get all bent out of shape about their post and fire back a heated response. For example, a user could post a message to the list asking I'm new to Linux and Asterisk. Should edit my dialplan by hand, use FreePBX, or buy a commercial solution? Imagine the response as you tried to convince them to buy PBXWare, FreePBX users try to convince them that they should start out using FreePBX, and others go on about how hand coding a dialplan is the one-true-way® to learn Asterisk. Generally, the original poster is just looking to get everyone stirred up over nothing. In other words, Paul's original post of GUI bad! CLI good! was just the sort of post that is going to get folks fired up re-re-restarting the age-old discussion of which is better: CLI or GUI. Basically, it could be like posting any of the following: - Which is better: emacs or vi? - Which linux distribution is the best? - Which is better: Macs or Windows? All of these questions share the following: 1.) They have no right answer (macs are better for some, Windows for others, and linux for others still, not to mention OS/2, BSD, etc) 2.) People on the various sides of the debate have extremely strong feelings on the matter 3.) Nobody is likely to be convinced that the other side is right and that they are wrong. 4.) They have all been discussed thousands of times before, and nothing new is likely to be said on the matter. 5.) The only purpose served by the discussion, due to the reasons above, is to clutter up the mailing list. 6.) Any discussion thread regarding these sorts of topics is best avoided. For a more thorough description of an internet troll, see the following wikipedia article:http://en.wikipedia.org/wiki/Troll_% 28internet%29 In other words, if you see a post that is just going to result in a re-rehashing of the last rehash of a specific subject, just hit the delete key instead of clogging up the mailing list with yet another thread on whether a GUI or a CLI is better. (for example). In Paul's defense, it looked to me like his original post was simply a joke that was misunderstood. (I thought it was funny, anyway) I suppose I should take my own advice on this one, but sometimes I guess we all just can't resist. grin Tom ___ --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
Re: [asterisk-users] asterisk hang (Critical Response)
I have the SAME problem: libpri 1.2.4, zaptel 1.2.18, Asterisk 1.2.18. , Aprox 150 SIP extensions. The machine works fine for about 6 to 12 hours without any issue, but appears the same error and the asterisk hangs definitly. I must restart the machine to have asterisk up un running without problems... Please, any idea ?? Rilawich Ango escribió: 1.2.10 On 6/19/07, Doug [EMAIL PROTECTED] wrote: At 02:08 6/17/2007, Rilawich Ango wrote: HI all, Recently, I got the following message from CLI and finally the asterisk will hang. Anyone can tell me how to fix the problem or why it will happen. Thanks. Version? Also: http://www.google.com/search?q=Avoiding+initial+deadlock+channel+lock+for+SIP+sipsock_read Jun 17 14:18:02 DEBUG[24573] channel.c: Avoiding initial deadlock for 'SIP/1127-008d65f0' Jun 17 14:22:45 ERROR[24696]: chan_sip.c:11337 sipsock_read: We could NOT get the channel lock for SIP/1589-0087cdd0! Jun 17 14:22:45 ERROR[24696]: chan_sip.c:11338 sipsock_read: SIP MESSAGE JUST IGNORED: CANCEL Jun 17 14:22:45 ERROR[24696]: chan_sip.c:11339 sipsock_read: BAD! BAD! BAD! Jun 17 14:28:04 WARNING[25368]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 103 (Critical Response) ___ --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
Re: [asterisk-users] X-Lite problems on basic asterisk setup
Packet sniffer found the problem. RTP was firewalled on the Asterisk box. Fixed it using the Asterisk firewall rules page on the wiki http://www.voip-info.org/wiki-Asterisk+firewall+rules. The 30 second lag on the dialing has something to do with using the domain name instead of the IP address of the asterisk server in the SIP config on X-Lite. The call goes immediately when I set the domain to the IP address of the asterisk box. Thanks for your help. Rob Schall wrote: This typically happens when the phone is natting or there is a firewall between the phone and the asterisk server. The connection is made via sip (5060), but the voice is over ports 1-2 (RTP). Most likely, the sip connection is succeeding, since you are connecting, but the actual voice is failing to transfer over RTP. if this is the case, I would aim to use IAX since it was made for this type of use. If the phone is on the same network as the asterisk server, and you are still having issues, use a packet sniffer and watch the traffic on both ends. You should be able to receive every packet that is sent. Most likely in this case though, you will only see those 5060 packets making it. Rob Andrew Stewart wrote: I'm trying to setup my first Asterisk setup on a CentOS 5 installation on VMWare Workstation 6. Got two Linksys SPA941s working fine. But X-Lite softphones can't answer phone calls, and when one of them calls on of the Linksys phones they connect but neither party can hear hear the other. I noticed that the Linksys phones are connected via Native bridging while the X-Lite ones are connected via Packet2Packet bridging. Also, on the X-Lite phones there is a about a 30 second lag between when the X-Lite client hits dial/call and when the called party starts ringing. ::Asterisk setup:: Asterisk 1.4.4 Zaptel 1.4.3 (only ztdummy compiled) Asterisk Addons 1.4.1 CentOS 5 VMWare Workstation 6 ::sip.conf:: [Linksys01] type=friend secret=ledzep context=default host=dynamic mailbox=6445 [X-Lite01] type=friend secret=rammerjammer context=default host=dynamic dtmfmode=rfc2833 mailbox=2070 canreinvite=yes nat=no [Linksys02] type=friend secret=bigben context=default host=dynamic mailbox=6368 qualify=yes ::extenstions.conf:: [default] include = demo exten = 6445,1,Dial(SIP/Linksys01,20) exten = 6445,n,Voicemail(u6445) exten = 2070,1,Dial(SIP/X-Lite01,20) exten = 2070,n,Voicemail(u2070) exten = 2070,n,HangUp() exten = 6368,1,Answer exten = 6368,n,Ringing exten = 6368,n,Dial(SIP/Linksys02,20) exten = 6368,n,Voicemail(u6368) exten = 6368,n,HangUp() --- Andrew Stewart ___ --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 -- --- Andrew Stewart [EMAIL PROTECTED] (205) 585-2980 - cell ___ --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] zlib1g
Dear Cohen; In this link: http://www.asteriskguru.com/tutorials/wildcard_tdm400p.html In the subject: 2.Installation, then in the sub title: Zaptel Installation Please advise. ___ You snooze, you lose. Get messages ASAP with AutoCheck in the all-new Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_html.html ___ --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] Gigabit SIP Phones
I could use it as I've got gig network everywhere but most are through the phone port since there's only one jack per desk and no time to upgrade.. It stays that way until affordable gig phones exist to justify the upgrade D Dave Bour Desktop Solution Center 905.381.0077 [EMAIL PROTECTED] For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Sent: Tue Jun 12 18:12:13 2007 Subject: Re: [asterisk-users] Gigabit SIP Phones Quoting Erik Anderson [EMAIL PROTECTED]: On 6/12/07, Olivier [EMAIL PROTECTED] wrote: Hello, Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone. Did I miss something ? I don't know of any other GE phones. However... Why in the world would you ever need GigE sip phones? unless you're using a built in 2 port switch in it or something I can't see the need either - the phone itself doesn't even approach 10mb/s, let alone 1000. -Erik ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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] Ex-Girlfriend Logic in 1.4.4
For real? I thought _ was to tell asterisk it was time for some pattern
matching:
; exact extension, exact cid
exten = 5000/19256002182,1,Answer
; any extension beginning with 5, from specific cid only
exten = _5./19256002182,1,Answer
; match exactly extension 5000, but anyone calling from
; (925) 600- matches
exten = 5000/_1925600.,1,Answer
; match anyone calling any extension beginning with 5 FROM any cid
; in the (925) 600- block
exten = _5./_1925600.,1,Answer
are the ways I've always used the underscore.
Doug, sorry I didn't have anything to help with your problem. I just
wanted to get some clarification of this poster's statement, to either
help myself or 10,000 other readers, I'm not sure who, yet...
Mojo
Nasir Iqbal wrote:
Hi,
exten = 5000/19256002182,1,Answer
exten = 5000/19256002182,n,Wait(1)
exten = 5000/19256002182,n,NoOp(${CALLERID(num)})
exten = 5000/19256002182,n,Playback(tt-monkeys)
nothing appears on the console and I get no match. You can see the ca
Try with underscore before extension like.
exten = _5000/19256002182,1,Answer
Nasir Iqbal
ICT Innovations
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Re: [asterisk-users] Improving Asterisk's DNS support
I could understand if it couldn't register to an ITSP or similar. But, (I had this happen today) asterisk takes forever to start up and SIP phones can not register to it. DNS should not need to be used for anything in asterisk except registering to VOIP providers and maybe external SQL from the dialplan. If there are reverse lookups being done, I do not see the output of it. -- -- Steven http://www.glimasoutheast.org Remco Post [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Kristian Kielhofner wrote: Hello everyone, After several years of using Asterisk I have always been frustrated by the support for DNS. I have seen all kinds of strange behavior when Asterisk is used on a system with iffy DNS servers: - no failover to other DNS servers in /etc/resolv.conf (might be a C library thing) wasn't there some setting for that? I run a dns caching deamon om my * box (speeds up enum lookups big time), but i seem to recall that some dns settings could be made - chan_sip will sometimes mark even local SIP peers as unreachable during/after any DNS problems - why? because your * can't resolve the names any more? - dnsmgr doesn't support SIP (yikes!): http://bugs.digium.com/view.php?id=9153 - other randomness (please contribute your own experiences) What can we do about improving this situation? At the very least we need to extend DNS manager support to SIP. I'm willing to pay for this and any other Asterisk DNS improvements. Any other ideas? -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] install Asterisk-addons 1.4.2
Hi, I am trying to install the Asterisk-addons-1.4.2, and when I make install it prompt me such error messages make[1]: Entering directory `/usr/src/asterisk-addons/asterisk-ooh323c' cp .libs/libchan_h323.so.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory make[1]: *** [install] Error 1 make[1]: Leaving directory `/usr/src/asterisk-addons/asterisk-ooh323c' make: *** [install] Error 2 How to solve it out? clive chan Alpha Trilogies Networks Sdn Bhd Tel : 04 - 647 288 Ext: 338 Tel : 04 - 647 2999 Mobile : 012 - 408 6376 email : [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
[asterisk-users] Asterisk 1.2.0 addon Radius
Hai all. I'm currently developing a softswitch with asterisk server. But i have a few problems here. Anyone can help me? 1. Is there any add on for asterisk 1.2.0 to connect it to Freeradius server? 2. As far as i know, i need a file named cdr_radius.c and compile my asterisk again, but when i did this, my asterisk suddenly error. Any time i start the service, it will terminate after a few second. Any one know why? Any solutions? 3. For this problem, i recompile my asterisk 1.2.0 without the cdr_radius.c file (which took almost 1 hour). But then, i can't connect my softphone to Asterisk (which is no problem before). It always prompt 408 - request timeout. Any one know why? Any solutions? I just really don't know the cause. I've search through anywhere but there's nothing useful. But, thanks for helping me. Regards, Igor Need Mail bonding? Go to the Yahoo! Mail QA for great tips from Yahoo! Answers users. http://answers.yahoo.com/dir/?link=listsid=396546091___ --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] zlib1g
On Wed, Jun 20, 2007 at 03:32:19PM -0700, bilal ghayyad wrote: Dear Cohen; In this link: http://www.asteriskguru.com/tutorials/wildcard_tdm400p.html In the subject: 2.Installation, then in the sub title: Zaptel Installation Please advise. My advice: don't use obsolete doucmentation. That incorrect recommenndation is not the only mistake in that page. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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
