[Asterisk-Users] Caller ID
I have a quick Caller*ID question.
I have an inbound call to my PBX which I am attempting to bridge with
a PSTN number (specifically my cell phone, so when someone dials my
extension the cell phone rings).
In my extentions.conf I have:
; Daniel -- 1102
exten = 1102,1,Answer()
exten = 1102,2,Set(DIALEDNUM=1102)
exten = 1102,3,Wait(2)
exten = 1102,4,Playback(pls-wait-connect-call)
exten = 1102,5,Wait(1)
exten = 1102,6,Dial(SIP/2102SIP/3102SIP/4102SIP/[EMAIL PROTECTED],
33,mj)
exten = 1102,7,Voicemail(su{$EXTEN})
exten = 1102,8,Hangup()
exten = 1102,106,Voicemail(sb{$EXTEN})
exten = 1102,107,Hangup()
where porta is my SIP account with the company that provides my
PSTN connection. I know for a fact that I can set any caller ID I
want (because I've done it with ATAs) and my carrier will pass it;
however, my question is, how do I get my asterisk box to pass the
original Call*ID instead of the number assigned to me by my provider?
this is the entry in sip.conf
[porta]
type=peer
secret=corbe9845
username=portasip
host=68.145.125.95
;fromuser=17862065496
fromdomain=66.165.175.35
insecure=very
;nat=yes
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[Asterisk-Users] Re: musiconhold errors in 1.2.0-beta1
Has anyone run into this problem yet? -Daniel On 9/9/05, Daniel Corbe [EMAIL PROTECTED] wrote: I'm getting a FLOOD of these types of messages on my MAC OS X box: Sep 9 14:46:31 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:37 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:37 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:43 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! -Daniel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] musiconhold errors in 1.2.0-beta1
I'm getting a FLOOD of these types of messages on my MAC OS X box: Sep 9 14:46:31 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:37 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:37 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:43 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! -Daniel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Monitoring Tools
Hello, I'm currently researching a project that would enable us to pull the actual signaling (SIP conversation) along with our CDRs The best way I can tell to approach this is to set up a server on a SPAN port which mirrors all my proxy servers' traffic. I was curious if anyone else has ever used this approach; and if so, what tools were being used to filter out and subsequently log the signaling (SIP) traffic. If not, is there a better way to approach the problem? Thanks. -Daniel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loop Detection
I keep getting the same answer from people Well the SIP implementation is fine if you use XXX IP Phone so obviously Asterisk was never designed to be used as a TDM gateway but merely as a PBX server only. On 4/17/05, Cameron Beattie [EMAIL PROTECTED] wrote: This is very interesting to me since I am in the process of setting up SER to Asterisk in a similar scenario. I'm surprised there haven't been more posts. Maybe include SER - Asterisk in the title. There are other posters on the list who use SER and Asterisk together who surely must have encountered (overcome?) this problem since it is so fundamental. Perhaps a bug should be raised? Regards Cameron - Original Message - From: Daniel Corbe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 14, 2005 7:29 AM Subject: [Asterisk-Users] Loop Detection Hello, Is there any way to turn Loop Detection off or tune the params a bit? I am having an issue with Call Forwarding on my SIP Proxy Server which is causing me great pains. Here is the issue: 1) I have a SIP UA which registers with a SER proxy server. 2) I have an Asterisk TDM gateway in my network, also which registers with SER 3) A call comes in through the PSTN to the Asterisk Gateway. The Asterisk gateway sends the call to SER destined for my SIP UA 4) SER sees that the SIP UA has call forwarding enabled so it creates a new outbound call with the same Call ID but it has a different TAG= line and Max-Forwards is set to 70. 5) Since the fowarding number is out on the PSTN, SER routes the call back through the same * gateway. 6) Asterisk rejects the phone call with Loop Detected According to my interpretation of the RFC, it is more correct to base loop detection off of the TAG= than it is off of the Call ID. Having said that, SER also sets the Max Forwards on the call. Is there any way at all to get Asterisk to either base its loop detection off the TAG= or respect the Max-Forwards setting? I've also attached a libpcap packet dump of a phone call. 389.764074 62.25.108.211 - 66.165.175.44 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 389.885825 66.165.175.44 - 62.25.108.211 SIP Status: 401 Unauthorized 389.885999 62.25.108.211 - 66.165.175.44 SIP Request: ACK sip:[EMAIL PROTECTED] 389.886104 62.25.108.211 - 66.165.175.44 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 390.145261 66.165.175.44 - 62.25.108.211 SIP Status: 100 trying -- your call is important to us 390.257658 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 390.257706 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 390.801964 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 390.802007 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 391.901785 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 391.901829 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 393.991808 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 393.991851 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 401.223872 62.25.108.211 - 66.165.175.44 SIP Request: CANCEL sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Re: Loop Detection
Doug,
Last night I attempted exactly what you just described. I commented
the if { block in chan_sip.c which contains the loop detection code
and tried to place a call. This caused the Asterisk server to dump
core.
I did a little more research into the subject and found this:
Apparently Asterisk's Call ID behavior is slightly modified if you go
into chan_sip.c and set the variable pedanticsipchecking to 1.
This causes Asterisk to not only compare the Call ID of the call but
the TAG= line in the header.
This behavior is more RFC compliant than the default behavior; however
after enabling this and placing a few test calls I seem to get
extremely long delays in establishing the forwarded leg of my calls
and the RTP stream is not being relayed correctly.
I'm currently at a loss. I am seriously considering replacing this
Asterisk TDM gateway with a Cisco 5350.
-Daniel
On 4/15/05, Doug Meredith [EMAIL PROTECTED] wrote:
Daniel Corbe [EMAIL PROTECTED] wrote:
Is there any way to turn Loop Detection off or tune the params a bit?
I am having an issue with Call Forwarding on my SIP Proxy Server which
is causing me great pains.
All I can do is sympathize. The same problem occurs when a call comes
in through Asterisk, gets sent to SER, then comes back to Asterisk 20
seconds later for voicemail.
I have contemplated just commenting out the check in chan_sip.c, but I
haven't tried this. Not sure if this might cause other problems.
Asterisk has many SIP deficiencies. Asterisk has been built as a
monolithic PBX, and it seems to do okay using SIP phones as channels.
If you want Asterisk to simply act as a SIP UA, you are going to run
into a whole slew of problems. I'm not holding my breath waiting for
this to change.
Doug
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[Asterisk-Users] Loop Detection
Hello, Is there any way to turn Loop Detection off or tune the params a bit? I am having an issue with Call Forwarding on my SIP Proxy Server which is causing me great pains. Here is the issue: 1) I have a SIP UA which registers with a SER proxy server. 2) I have an Asterisk TDM gateway in my network, also which registers with SER 3) A call comes in through the PSTN to the Asterisk Gateway. The Asterisk gateway sends the call to SER destined for my SIP UA 4) SER sees that the SIP UA has call forwarding enabled so it creates a new outbound call with the same Call ID but it has a different TAG= line and Max-Forwards is set to 70. 5) Since the fowarding number is out on the PSTN, SER routes the call back through the same * gateway. 6) Asterisk rejects the phone call with Loop Detected According to my interpretation of the RFC, it is more correct to base loop detection off of the TAG= than it is off of the Call ID. Having said that, SER also sets the Max Forwards on the call. Is there any way at all to get Asterisk to either base its loop detection off the TAG= or respect the Max-Forwards setting? I've also attached a libpcap packet dump of a phone call. 389.764074 62.25.108.211 - 66.165.175.44 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 389.885825 66.165.175.44 - 62.25.108.211 SIP Status: 401 Unauthorized 389.885999 62.25.108.211 - 66.165.175.44 SIP Request: ACK sip:[EMAIL PROTECTED] 389.886104 62.25.108.211 - 66.165.175.44 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 390.145261 66.165.175.44 - 62.25.108.211 SIP Status: 100 trying -- your call is important to us 390.257658 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 390.257706 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 390.801964 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 390.802007 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 391.901785 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 391.901829 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 393.991808 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 393.991851 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 401.223872 62.25.108.211 - 66.165.175.44 SIP Request: CANCEL sip:[EMAIL PROTECTED] voip4u-2005040401.dump Description: Binary data ___ 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] 7905 example configs
Anyone have the example.txt file that Cisco's documentation on the 7905 IP phone keeps referring to? Or can someone possibly share a fairly complete example config file for these phones with me? Thanks! -Daniel ___ 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] * Call Monitoring
I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? -Daniel ___ 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] * Call Monitoring
Okay
here's a quick and dirty little perl script to monitor the PRI Status
and mimic nagios plugin output.
-Daniel
On Mon, 21 Feb 2005 07:50:45 -0600, Brian Roy [EMAIL PROTECTED] wrote:
On Mon, 21 Feb 2005 08:00:40 -0500, Daniel Corbe
[EMAIL PROTECTED] wrote:
I need to make sure the PRIs connected to my box stay up and I need to
make sure calls are not failing for any reason. Are there any *
monitoring packages like this?
There aren't any specific tools that do exactly what you want afaik.
It wouldn't take much to taylor a few things yourself though.
As for the PRI processing calls. You could always drop a call file in
from the cron every 10 minutes that makes a call out and back in. Then
you you can run a script that looks over your CDR to verify that the
call was received. Have it call a specific context or application to
look for.
As for calls failing this could be a challange. What do you consider
failing? You could use something like my-swatch to tail the log file
looking for certain patterns. PRI alarms would be an obvious.
Might take you a day or so to get these things going, but it would be
well worth your time and piece of mind.
-Chuji
#!/usr/bin/perl
###
# Michael Jastremski
# Monitor Asterisk PBX via Manager Interface
# http://megaglobal.net/docs/
###
# Based upon:
#
# TACI - Trivial Asterisk Call Interface v.02
# Last update 3/30/2004
# Tony Wasson [EMAIL PROTECTED]
#
#
# Modified by Daniel Corbe to monitor PRI spans
# [EMAIL PROTECTED]
#
# -Daniel
#
$ENV{'PATH'}='';
$ENV{'BASH_ENV'}='';
$ENV{'ENV'}='';
$| = 1;
use Net::Telnet ();
use File::Basename;
use lib /usr/local/nagios/libexec;
use utils qw(%ERRORS);
my $mgr_user = nagios;
my $mgr_secret = XyXyXyXyXy;
my $failed = 0;
my $reason = undef;
my $server_ip = 127.0.0.1;
my $prispan = $ARGV[0];
$tn = new Net::Telnet (Port = 5038,
Prompt = '/.*[\$%#] $/',
Output_record_separator = '',
Errmode= 'return'
);
$tn-open($server_ip);
$tn-waitfor('/0\n$/');
$tn-print(Action: Login\nUsername: $mgr_user\nSecret: $mgr_secret\n\n);
unless($tn-waitfor('/Authentication accept*/'))
{
$failed = 1;
$reason = Failed Connect;
}
else
{
$tn-print(Action: Command\n);
$tn-print(Command: pri show span $prispan\n\n);
#Response: Follows
#Primary D-channel: 24
#Status: Provisioned, Up, Active
unless($tn-waitfor('/Response: Follows\nPrimary D-channel: (.*)?\nStatus:
Provisioned, Up, Active/'))
{
$failed = 1;
$reason = PRI Span # . $prispan . is down;
}
else
{
$tn-print(Action: Logoff\n\n);
}
}
print PRI Span #$prispan is up\n unless $failed;
print $reason\n if $failed;
exit $ERRORS{'CRITICAL'} if $failed;
exit $ERRORS{'OK'};
exit 0;
__END__
TODO:
-- Maybe check other variables?
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Re: [Asterisk-Users] * Call Monitoring
Yeah, I'd be interested in porting your work so it runs under nagios. Please post your results when you're finished. -Daniel On Tue, 22 Feb 2005 02:54:22 +1100, Adam Goryachev [EMAIL PROTECTED] wrote: On Mon, 2005-02-21 at 08:00 -0500, Daniel Corbe wrote: I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? Interesting you should ask this today... I got to work this morning and was wondering why some of my calls were still diverting to my mobile. Eventually I realised that they were diverting on no answer. A restart of asterisk, reload of modules etc made no differences, I couldn't do anything with the line. Eventually I worked out it was a telco problem (no dialtone/etc) so I logged the fault. I looked at zttool and it showed a red alarm... In around 10-20 minutes I hacked zttool.c and converted it into a very basic cli version (which doesn't need newt) and would just dump the current status of all the spans. Similar to what you see on screen when you first start zttool. Then, I threw together some simple shell scripting to analyse/send the report to BigBrother (www.bb4.org). So far it is working nicely, by tomorrow night (yes, 27 hours after reporting it) hopefully my line should come back, and the alarm should change to OK... I'll put the package etc onto www.deadcat.net (BB addons website) and drop a post here when it is done. Will also put it onto www.websitemanagers.com.au/asterisk/ BTW, I did need to suid the zttool-cli command to root, as the normal BB user doesn't have the needed permissions. I haven't looked into this, but if anyone has a suggestion on a better way to do this, feel free to let me know. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Codec negotiation problems
My PBX seems to have just started showing wierd codec negotiation problems. I'm not all of a sudden getting this on certain phone numbers on my system: Feb 8 22:19:19 NOTICE[1125329728]: channel.c:1683 ast_set_read_format: Unable to find a path from ULAW to G729A Feb 8 22:19:19 NOTICE[1125329728]: channel.c:1650 ast_set_write_format: Unable to find a path from G729A to ULAW -- SIP/2101-aaf2 answered SIP/19544342000-375a -- Attempting native bridge of SIP/19544342000-375a and SIP/2101-aaf2 -- Attempting native bridge of SIP/19544342000-375a and SIP/2101-aaf2 Feb 8 22:19:19 NOTICE[1225991488]: channel.c:1683 ast_set_read_format: Unable to find a path from G729A to ULAW Feb 8 22:19:19 NOTICE[1225991488]: channel.c:1650 ast_set_write_format: Unable to find a path from ULAW to G729A Feb 8 22:19:19 WARNING[1225991488]: chan_sip.c:1797 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Regards, Daniel ___ 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] g729
Hello, I've inherited a (now) broken asterisk implementation. It seems as if there are currently codec tanscoding issues in this box. Specifically I am receving calls from a SIP proxy in G.729 and attempting to transcode them to ULAW. My asterisk installation was working up until yesterday. The information I have found on the voip-Asterisk wiki is for a third party open-source implementation of asterisk (which doesn't seem to be what was deployed on this asterisk box at all); however there are glaring warnings all over the place about possible legal implications of using this codec. Moreover the words GPL Violation are printed and duplicated many times over in the form of an E-Mail from Mark Spencer attached to the Wiki. The documentation does not; however tell me the correct and 100% legal way to license and implement G.729. In short this server is broken and I don't know what to do because I'm afraid of possibly being in direct violation of the GPL by following the voip-Asterisk wiki's documentation. Can someone kindly point me to an RTFM in the form of Digium-supported licensing options for G.729 and some technical documentation on how to acctually do the implementation? Or perhaps someone could suggest a fix for my current issues: Here's a log snippet from my asterisk console: Feb 8 22:19:19 NOTICE[1125329728]: channel.c:1683 ast_set_read_format: Unable to find a path from ULAW to G729A Feb 8 22:19:19 NOTICE[1125329728]: channel.c:1650 ast_set_write_format: Unable to find a path from G729A to ULAW -- SIP/2101-aaf2 answered SIP/19544342000-375a -- Attempting native bridge of SIP/19544342000-375a and SIP/2101-aaf2 -- Attempting native bridge of SIP/19544342000-375a and SIP/2101-aaf2 Feb 8 22:19:19 NOTICE[1225991488]: channel.c:1683 ast_set_read_format: Unable to find a path from G729A to ULAW Feb 8 22:19:19 NOTICE[1225991488]: channel.c:1650 ast_set_write_format: Unable to find a path from ULAW to G729A Feb 8 22:19:19 WARNING[1225991488]: chan_sip.c:1797 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Now both channels in question have allow=ulaw and allow=g729 Any help at all would be appriciated. Regards, Daniel ___ 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] Concurrent calls
Is there any way to quickly poll an asterisk server for concurrent call count? Preferably from like a perl or PHP script. -Daniel ___ 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] Re: Concurrent calls
:) On Thu, 3 Feb 2005 10:41:37 -0500, Daniel Corbe [EMAIL PROTECTED] wrote: Is there any way to quickly poll an asterisk server for concurrent call count? Preferably from like a perl or PHP script. -Daniel ___ 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] My dialplan just stopped working one day
Hrm,
All of a sudden for some reason Wait() and Playback() are returning
non-zero and its causing calls on my inbound SIP leg not to complete.
I'm not sure why
-- Executing Answer(SIP/2181-4518, ) in new stack
-- Executing Playback(SIP/2181-4518, silence/1) in new stack
-- Playing 'silence/1' (language 'en')
== Spawn extension (inbound, 2181, 2) exited non-zero on 'SIP/2181-4518'
-- Executing Hangup(SIP/2181-4518, ) in new stack
== Spawn extension (inbound, h, 1) exited non-zero on 'SIP/2181-4518'
Extentions.conf:
[default]
; Main menu
exten = main,1,Background(thank-you-for-calling)
exten = main,2,Background(silence/1)
exten = main,3,Background(if-u-know-ext-dial)
exten = main,4,Background(silence/1)
exten = main,5,Background(for-tech-support)
exten = main,6,Background(press-1)
exten = main,7,Background(silence/1)
exten = main,8,Background(to-hear-menu-again)
exten = main,9,Background(press-9)
exten = h,1,Hangup
exten = t,1,Goto(default,main,1)
exten = i,1,Playback(invalid)
exten = i,2,Playback(goodbye)
exten = i,3,Goto(default,main,1)
exten = T,1,Playback(goodbye)
exten = T,2,Hangup
; Use dialed an extention
exten = _2XXX,1,Goto(extentions,${EXTEN},1)
[inbound]
; This is the list if inbound lines
exten = 2181,1,Answer
exten = 2181,2,Playback(silence/1)
exten = 2181,3,Goto(default,main,1)
exten = 2181,3,Hangup
exten = h,1,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
exten = T,1,Hangup
[extentions]
exten = 2000,1,Dial(SIP/2000,20)
exten = 2000,2,Voicemail(u2000)
exten = 2000,102,Voicemail(b2000)
exten = 2000,103,Hangup
exten = 2001,1,Dial(SIP/2001,20)
exten = 2001,2,Voicemail(u2001)
exten = 2001,102,Voicemail(b2001)
exten = 2001,103,Hangup
;exten = 2002,1,Dial(IAX2/iaxphone,20)
;exten = 2002,2,Voicemail(u2002)
;exten = 2002,102,Voicemail(b2002)
;exten = 2002,103,Hangup
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
exten = h,1,Hangup
exten = t,1,Hangup
exten = i,1,Playback(invalid)
exten = i,2,Goto(default,main,3)
exten = T,1,Playback(goodbye)
exten = T,2,Hangup
exten = s,1,Goto(default,main,3)
Sip.conf:
[general]
bindaddr = 0.0.0.0
allow = all
context = bogon-calls
register = 2181:[EMAIL PROTECTED]/2181
[2000]
type=friend
username=2000
secret=yoyoyodawg
host=dynamic
nat=yes
canreinvite=no
context=outbound
mailbox=100
disallow=all
allow=gsm
[2001]
type=friend
username=2001
secret=yoyoyodawg
host=dynamic
context=outbound
mailbox=101
nat=yes
[2181]
type=friend
secret=yoyoyodawg
host=66.165.175.227
context=inbound
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=g729
allow=ilbc
allow=speex
allow=ulaw
allow=alaw
username=2181
fromuser=2181
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[Asterisk-Users] Open Source SCGP
Hey, Someone told me an open source SGCP gateway was created for the Asterisk project. I'm looking for a little more information. I have two VG248s that I'd like to attach to my VoIP network; however, Cisco's documentation seems to indicate that Cisco CallManager is required for these things to operate. Cisco CallManager being a 10,000 dollar application, I would like to find any open source alternatives. Regards, Daniel -- Daniel Corbe, CCNP Senior Network Engineer Results Technologies, Inc. 952-921-2400 x104 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
