Re: [asterisk-users] realtime config for general settings in sip.conf
Hello, For realtime configuration, in /etc/asterisk/extconfig.conf file, what should be the family name to configure general sip.conf parameters. => ,, thanks, Kamlesh > From: i...@pack-net.co.uk > To: asterisk-users@lists.digium.com > Date: Wed, 2 May 2012 13:59:58 +0100 > Subject: Re: [asterisk-users] realtime config for general settings in sip.conf > > On Wed, 2012-05-02 at 12:04 +, Kamlesh Kumar wrote: > > Hi, > > > > I need to configure global parameters in sip.conf like rtptimeout, > > rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real > > time architecture. Please suggest the way to do it. > > > > thanks, > > Kamlesh > > > > Hi > > You can set defaults in the column definitions and you can still set > globals in the sip.conf > > Ish > > -- > Ishfaq Malik > Department: VOIP Support > Company: Packnet Limited > t: +44 (0)845 004 4994 > f: +44 (0)161 660 9825 > e: i...@pack-net.co.uk > w: http://www.pack-net.co.uk > > Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET > NORTH, MANCHESTER > SCIENCE PARK, MANCHESTER, M156SE > COMPANY REG NO. 04920552 > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk AMI SIP channel detect phone ringing
2012/5/3 JIMMY GATHAGE : > I am using a SIP trunk to make outgoing calls. Outgoing calls are > going through okay. I am using the AMI to Originate a call. The > channel is not returning any event when the phone on the PSTN is > ringing. How can i detect the phone ringing on the SIP channel? It is possible you made synchronous origination. Did you specified Async: true header in origination action? If I remember correctly, synchronous origination blocks AMI session until it done origination routine. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk AMI SIP channel detect phone ringing
On 05/02/2012 04:41 PM, JIMMY GATHAGE wrote: Hey guys, I am using a SIP trunk to make outgoing calls. Outgoing calls are going through okay. I am using the AMI to Originate a call. The channel is not returning any event when the phone on the PSTN is ringing. How can i detect the phone ringing on the SIP channel? If your SIP provider is not sending you '180 Ringing' responses, then your only choice would be look into a 'call progress detection' package that can listen to the incoming audio and analyze it for ring-back. Unfortunately these are not terribly reliable, because ring-back tones vary greatly, and they might not even be traditional ring-back (many mobile providers offer 'music ringback' to their subscribers). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk AMI SIP channel detect phone ringing
Hey guys, I am using a SIP trunk to make outgoing calls. Outgoing calls are going through okay. I am using the AMI to Originate a call. The channel is not returning any event when the phone on the PSTN is ringing. How can i detect the phone ringing on the SIP channel? Am desperate. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10.4.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 10.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.4.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Prevent chanspy from binding to zombie channels (Closes issue ASTERISK-19493. Reported by lvl) * --- Fix Dial m and r options and forked calls generating warnings for voice frames. (Closes issue ASTERISK-16901. Reported by Chris Gentle) * --- Remove ISDN hold restriction for non-bridged calls. (Closes issue ASTERISK-19388. Reported by Birger Harzenetter) * --- Fix copying of CDR(accountcode) to local channels. (Closes issue ASTERISK-19384. Reported by jamicque) * --- Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors (Closes issue ASTERISK-19303. Reported by Jon Tsiros) * --- Eliminate double close of file descriptor in manager.c (Closes issue ASTERISK-18453. Reported by Jaco Kroon) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.4.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.12.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.12.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.12.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Prevent chanspy from binding to zombie channels (Closes issue ASTERISK-19493. Reported by lvl) * --- Fix Dial m and r options and forked calls generating warnings for voice frames. (Closes issue ASTERISK-16901. Reported by Chris Gentle) * --- Remove ISDN hold restriction for non-bridged calls. (Closes issue ASTERISK-19388. Reported by Birger Harzenetter) * --- Fix copying of CDR(accountcode) to local channels. (Closes issue ASTERISK-19384. Reported by jamicque) * --- Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors (Closes issue ASTERISK-19303. Reported by Jon Tsiros) * --- Eliminate double close of file descriptor in manager.c (Closes issue ASTERISK-18453. Reported by Jaco Kroon) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.12.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] parsing issue
Or even Hangup(-${Z})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller
Sent: Wednesday, May 02, 2012 2:11 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] parsing issue
On Wed, May 02, 2012 at 01:48:04PM -0400, CDR wrote:
> I get an error when I execute this code exten =>
> rejected,n,Hangup($[-1*${Z}])
>
> May 2 13:42:09] WARNING[23128]: ast_expr2.fl:468 ast_yyerror:
> ast_yyerror(): syntax error: syntax error, unexpected $end, expecting
> '-' or '!' or '(' or ''; Input:
> -1*
>
> The variable "Z" has a negative number, which is the code that I need
> to use in the hangup.
> Any idea how can I do this? There is no ABS() function in Asterisk. I
> already filed a request for it but it turns up that it will cost me
> money. How can I remove the sign from a number?
If you're sure Z is non-null and negative, then
exten => rejected,n,Hangup($[-(${Z})])
--
Barry
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Re: [asterisk-users] parsing issue
On Wed, May 02, 2012 at 01:48:04PM -0400, CDR wrote:
> I get an error when I execute this code
> exten => rejected,n,Hangup($[-1*${Z}])
>
> May 2 13:42:09] WARNING[23128]: ast_expr2.fl:468 ast_yyerror:
> ast_yyerror(): syntax error: syntax error, unexpected $end, expecting
> '-' or '!' or '(' or ''; Input:
> -1*
>
> The variable "Z" has a negative number, which is the code that I need
> to use in the hangup.
> Any idea how can I do this? There is no ABS() function in Asterisk. I
> already filed a request for it but it turns up that it will cost me
> money. How can I remove the sign from a number?
If you're sure Z is non-null and negative, then
exten => rejected,n,Hangup($[-(${Z})])
--
Barry
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Re: [asterisk-users] parsing issue
Have you tried the MATH() function?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of CDR
Sent: Wednesday, May 02, 2012 1:48 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] parsing issue
I get an error when I execute this code
exten => rejected,n,Hangup($[-1*${Z}])
May 2 13:42:09] WARNING[23128]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected $end, expecting '-' or
'!' or '(' or ''; Input:
-1*
The variable "Z" has a negative number, which is the code that I need to use
in the hangup.
Any idea how can I do this? There is no ABS() function in Asterisk. I already
filed a request for it but it turns up that it will cost me money. How can I
remove the sign from a number?
Philip
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Re: [asterisk-users] parsing issue
3 possibilities - 1 use an AGI; 2 use a system command 3 (hopefully the
simplest) just use a Set command. Which one you actually end up using may
depend on your Asterisk flavor.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of CDR
Sent: Wednesday, May 02, 2012 12:48 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] parsing issue
I get an error when I execute this code
exten => rejected,n,Hangup($[-1*${Z}])
May 2 13:42:09] WARNING[23128]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected $end, expecting '-'
or '!' or '(' or ''; Input:
-1*
The variable "Z" has a negative number, which is the code that I need to
use in the hangup.
Any idea how can I do this? There is no ABS() function in Asterisk. I
already filed a request for it but it turns up that it will cost me money.
How can I remove the sign from a number?
Philip
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Re: [asterisk-users] detecting intl. CLI with +
Eric,
You were right.
Thanks :)
Michael
On Wed, May 2, 2012 at 7:25 PM, Eric Wieling wrote:
> If you have quotes on one side of the = sign, then you need quotes on the
> other side. In your dialplan line you are comparing + with "+". A plus
> sign is not equal to quote plus sign quote
>
> exten => _X., n, Set(CALLERID(num)=${IF($["${CALLERID(num):0:1}" =
> "+"]?${CALLERID(num)}:0${CALLERID(num)})})
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Michael
> Sent: Wednesday, May 02, 2012 12:20 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] detecting intl. CLI with +
>
> Hello asterisk users,
>
> I need to convert the CLI received according to national/international
> format:
>
> 55-555- to 055-555- (add 0 in the beginning)
> +55-55-555- to +55-55-555- (remains unchanged)
>
> I put the following line in my dial plan:
> exten => _X., n, Set(CALLERID(num)=${IF($[ ${CALLERID(num):0:1} =
> "+"]?${CALLERID(num)}:0${CALLERID(num)})})
>
> But I get these error messages:
> [May 2 17:05:43] WARNING[1494]: ast_expr2.fl:468 ast_yyerror:
> ast_yyerror(): syntax error: syntax error, unexpected '+', expecting $end;
> Input:
> + = "+"
> ^
> [May 2 17:05:43] WARNING[1494]: ast_expr2.fl:472 ast_yyerror: If you have
> questions, please refer to
> https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
> [May 2 17:05:43] WARNING[1494]: func_logic.c:192 acf_if: Syntax
> IF(?[][:]) (expr must be non-null, and either or
> must be non-null)
> [May 2 17:05:43] WARNING[1494]: func_logic.c:193 acf_if: In this
> case, ='', ='+555', and ='0+555'
>
> Can anyone suggest the proper syntax? I tried the + with no quotes, single
> quotes '+' and double quotes"+" and nothing worked.
>
> Thanks
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> http://www.asterisk.org/hello
>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] detecting intl. CLI with +
If you have quotes on one side of the = sign, then you need quotes on the other
side. In your dialplan line you are comparing + with "+". A plus sign is not
equal to quote plus sign quote
exten => _X., n, Set(CALLERID(num)=${IF($["${CALLERID(num):0:1}" =
"+"]?${CALLERID(num)}:0${CALLERID(num)})})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael
Sent: Wednesday, May 02, 2012 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] detecting intl. CLI with +
Hello asterisk users,
I need to convert the CLI received according to national/international format:
55-555- to 055-555- (add 0 in the beginning)
+55-55-555- to +55-55-555- (remains unchanged)
I put the following line in my dial plan:
exten => _X., n, Set(CALLERID(num)=${IF($[ ${CALLERID(num):0:1} =
"+"]?${CALLERID(num)}:0${CALLERID(num)})})
But I get these error messages:
[May 2 17:05:43] WARNING[1494]: ast_expr2.fl:468 ast_yyerror: ast_yyerror():
syntax error: syntax error, unexpected '+', expecting $end; Input:
+ = "+"
^
[May 2 17:05:43] WARNING[1494]: ast_expr2.fl:472 ast_yyerror: If you have
questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
[May 2 17:05:43] WARNING[1494]: func_logic.c:192 acf_if: Syntax
IF(?[][:]) (expr must be non-null, and either or
must be non-null)
[May 2 17:05:43] WARNING[1494]: func_logic.c:193 acf_if: In this case,
='', ='+555', and ='0+555'
Can anyone suggest the proper syntax? I tried the + with no quotes, single
quotes '+' and double quotes"+" and nothing worked.
Thanks
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Re: [asterisk-users] detecting intl. CLI with +
I think you need to "escape" the + "\+"
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael
Sent: Wednesday, May 02, 2012 11:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] detecting intl. CLI with +
Hello asterisk users,
I need to convert the CLI received according to national/international
format:
55-555- to 055-555- (add 0 in the beginning)
+55-55-555- to +55-55-555- (remains unchanged)
I put the following line in my dial plan:
exten => _X., n, Set(CALLERID(num)=${IF($[ ${CALLERID(num):0:1} =
"+"]?${CALLERID(num)}:0${CALLERID(num)})})
But I get these error messages:
[May 2 17:05:43] WARNING[1494]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected '+', expecting $end;
Input:
+ = "+"
^
[May 2 17:05:43] WARNING[1494]: ast_expr2.fl:472 ast_yyerror: If you have
questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
[May 2 17:05:43] WARNING[1494]: func_logic.c:192 acf_if: Syntax
IF(?[][:]) (expr must be non-null, and either or
must be non-null)
[May 2 17:05:43] WARNING[1494]: func_logic.c:193 acf_if: In this
case, ='', ='+555', and ='0+555'
Can anyone suggest the proper syntax? I tried the + with no quotes, single
quotes '+' and double quotes"+" and nothing worked.
Thanks
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[asterisk-users] detecting intl. CLI with +
Hello asterisk users,
I need to convert the CLI received according to national/international
format:
55-555- to 055-555- (add 0 in the beginning)
+55-55-555- to +55-55-555- (remains unchanged)
I put the following line in my dial plan:
exten => _X., n, Set(CALLERID(num)=${IF($[ ${CALLERID(num):0:1} =
"+"]?${CALLERID(num)}:0${CALLERID(num)})})
But I get these error messages:
[May 2 17:05:43] WARNING[1494]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected '+', expecting $end;
Input:
+ = "+"
^
[May 2 17:05:43] WARNING[1494]: ast_expr2.fl:472 ast_yyerror: If you have
questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
[May 2 17:05:43] WARNING[1494]: func_logic.c:192 acf_if: Syntax
IF(?[][:]) (expr must be non-null, and either or
must be non-null)
[May 2 17:05:43] WARNING[1494]: func_logic.c:193 acf_if: In this
case, ='', ='+555', and ='0+555'
Can anyone suggest the proper syntax? I tried the + with no quotes, single
quotes '+' and double quotes"+" and nothing worked.
Thanks
--
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[asterisk-users] hangup problem on T1 span
Hello all,
I'm trying to solve a problem on a T1 span setup wherein calls are
apparently not hanging up properly.
The system in question is using a Xorcom Astribank with 1 full and 1
partial T1 span, and running Asterisk 1.4.36.
The symptom is that when a call hangs up on a DAHDI channel (according to
Asterisk), and another outgoing call tries to open a new channel on the
same line as the hung-up call within approximately a minute of the hangup,
the new call gets a congestion notice ("all circuits busy") from
asterisk. After about a minute passes after the hangup, the line becomes
available again. So it seems like the channels are not hanging up when
Asterisk tells them to, and Asterisk doesn't know it.
I suspected a signaling issue, and this appeared confirmed when I
discovered that the signalling was set in chan_dahdi.conf as "fxs_ks" (this
installation had been converted from analog lines by another company; I
guess that was an oversight?).
So I changed it to pri_cpe, as my reading of the docs indicated was proper.
After this change and restarting everything, though, the symptoms persist.
So I figure that either my reading of the docs is wrong (and therefore
pri_cpe is not the right signaling) OR something totally unrelated is going
on.
Can someone please clue me in here? I am a bit at a loss. Let me know if
you need further information about the system/environment.
Regards,
Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729
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Re: [asterisk-users] realtime config for general settings in sip.conf
On Wed, 2012-05-02 at 12:04 +, Kamlesh Kumar wrote: > Hi, > > I need to configure global parameters in sip.conf like rtptimeout, > rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real > time architecture. Please suggest the way to do it. > > thanks, > Kamlesh > Hi You can set defaults in the column definitions and you can still set globals in the sip.conf Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 8 and mixmonitor
On Wed, May 2, 2012 at 3:44 PM, SamyGo wrote: >> I can't figure out if it's a known issue, or a new bug. > > > Or a new feature !! > > Can you share the dialplan code where you are executing the mixmon > application ! I use it using the manager: DEBUG ami 2012-05-02 15:43:53 Sending AMI action: >>> Action: monitor >>> ActionID: pZWLT4gi-3OFk-emUX-6xiu-4GptK0BXJXuR >>> Channel: Local/leg_a@some-context-9710;2 >>> File: /var/spool/asterisk/recordings/wav/121e0009a9327900c0b9b3d5d5db7426 >>> Mix: 0 >>> (I edited the number and context name) > > Regards, > Sammy. > > > On Wed, May 2, 2012 at 5:09 PM, ik wrote: >> >> Hello, >> >> I have weird issue with Asterisk 8 lately. >> >> When I call MixMonitor without mixing the channels, it changes the >> sides of "in" and "out". >> Sometimes the first leg of the call is "in" and sometimes it's "out". >> >> I can't figure out if it's a known issue, or a new bug. >> >> I'm using Asterisk 8.11.1 >> >> Any ideas how can I figure out what is leg is what file ? >> >> Thanks, >> Ido >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 8 and mixmonitor
> > I can't figure out if it's a known issue, or a new bug. Or a new feature !! Can you share the dialplan code where you are executing the mixmon application ! Regards, Sammy. On Wed, May 2, 2012 at 5:09 PM, ik wrote: > Hello, > > I have weird issue with Asterisk 8 lately. > > When I call MixMonitor without mixing the channels, it changes the > sides of "in" and "out". > Sometimes the first leg of the call is "in" and sometimes it's "out". > > I can't figure out if it's a known issue, or a new bug. > > I'm using Asterisk 8.11.1 > > Any ideas how can I figure out what is leg is what file ? > > Thanks, > Ido > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 8 and mixmonitor
>> I'm using Asterisk 8.11.1 As far as I'm aware, there is no Asterisk 8. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime config for general settings in sip.conf
2012/5/2 Kamlesh Kumar > Hi, > > I need to configure global parameters in sip.conf like rtptimeout, > rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real time > architecture. Please suggest the way to do it. > > thanks, > Kamlesh > > For what I have discovered, it is not possible. I hope to be wrong, but the sip.conf realtime is limited to peers (or users) registering on the box. It is not suitable even for defining trunks to be used by asterisk. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 8 and mixmonitor
Hello, I have weird issue with Asterisk 8 lately. When I call MixMonitor without mixing the channels, it changes the sides of "in" and "out". Sometimes the first leg of the call is "in" and sometimes it's "out". I can't figure out if it's a known issue, or a new bug. I'm using Asterisk 8.11.1 Any ideas how can I figure out what is leg is what file ? Thanks, Ido -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] realtime config for general settings in sip.conf
Hi, I need to configure global parameters in sip.conf like rtptimeout, rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real time architecture. Please suggest the way to do it. thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerId back to incoming
I'm currently doing some testing with Asterisk ( 1.8.11.0) on RHEL6 using realtime for sippeers, sipusers and musiconhold I have Avaya definity <-> PRI E1 <-> Asterisk 1 <-> IAX2 <-> Asterisk 2 I have peers (sip) snom 821s on both Asterisk 1 and 2 all calls working between all systems. CallerID from Asterisk to Avaya is working correctly. The problem is a caller from Avaya to Asterisk displays correctly the CID of the Asterisk Extension to the calling party on the Avaya but only if the peer is on Asterisk 1. If the peer is on Asterisk 2 only the CID of the PRI on the avaya side is displayed. I hope this makes sense. I'm not sure where to start looking or whether its even possible. I can of course supply any of the configs that may help. Cheers Stephen Collier -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
