Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line
find the inline comment... On 07/29/2011 12:11 AM, Ishwar Sridharan wrote: The dialplan is very simple. When the call comes in, we hand the call over to adhearsion. This is how the dialplan looks: ;group 0 will be used for incoming calls EXOIN = DAHDI/g0 ;group 11 for outgoing EXOOUT = DAHDI/G11 ;This will be used by adhearsion EXOCID= [general] autofallthrough = yes ;really? clearglobalvars = no [frompstn] ;Send everything to adhearsion exten = _X.,1,Ringing exten = _X.,n,AGI(agi://127.0.0.1 http://127.0.0.1) exten = _X.,n,Hangup() ; Please try this. ; End dialplan The rest of the logic happens in adhearsion. -- Thanks, Ishwar. On Thu, Jul 28, 2011 at 6:33 PM, Nikhil d.nik...@cem-solutions.net mailto:d.nik...@cem-solutions.net wrote: Can you share the dialplan ,where SIP call is dialing... Thanks Nikhil On 07/28/2011 06:15 PM, Ishwar Sridharan wrote: Hello everybody, We have an asterisk 1.8.4.1 setup, connected to a PRI line. We're currently facing an issue where asterisk does not recognise the event when the called party declines/cuts the call. This happens specifically over calls on a PRI line. For calls over SIP, call decline event is captured properly. I wasn't able to find a solution on the asterisk-users mailing list archive. Any suggestions/help would be much appreiciated :) I can share the relevant parts of the configuration files, if needed. Here's an excerpt from asterisk logs for a SIP call. -- SIP/x- requested special control 16, passing it to SIP/x-0001 -- Started music on hold, class 'default', on SIP/x-0001 -- SIP/x- requested special control 20, passing it to SIP/x-0001 -- Got SIP response 603 Decline back from 127.0.0.1:5063 http://127.0.0.1:5063/ -- SIP/x-0001 is busy -- Stopped music on hold on SIP/x-0001 As you can see, on a SIP call, a call reject event is identified. For a call over the PRI, on the other hand, this event is not recognised. Here's an excerpt from asterisk log for a call over PRI. Call from to . -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called G11/x -- Started music on hold, class 'default', on DAHDI/i1/y -- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y -- DAHDI/i1/x-18f8 is ringing # At this point in time, x rejects the call. The event that's logged in asterisk is the following: -- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y # And the call times out after the default 30s. -- Nobody picked up in 3 ms Is there a reason why asterisk doesn't recognise the call decline, and does it need any configuration changes to enable this? Thanks for your help. -- Cheers, Ishwar. -- _ -- Bandwidth and Colocation Provided byhttp://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 -- _ -- 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] Capturing call Reject/Decline events on a PRI line
Try to Add h extensions in frompstn context and print ${HANGUPCAUSE} in that
you will receive in that ,
also read this for better implementation.
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
regards
Dhaval
On Fri, Jul 29, 2011 at 11:58 AM, Nikhil d.nik...@cem-solutions.net wrote:
**
find the inline comment...
On 07/29/2011 12:11 AM, Ishwar Sridharan wrote:
The dialplan is very simple. When the call comes in, we hand the call over
to adhearsion.
This is how the dialplan looks:
;group 0 will be used for incoming calls
EXOIN = DAHDI/g0
;group 11 for outgoing
EXOOUT = DAHDI/G11
;This will be used by adhearsion
EXOCID=
[general]
autofallthrough = yes ;really?
clearglobalvars = no
[frompstn]
;Send everything to adhearsion
exten = _X.,1,Ringing
exten = _X.,n,AGI(agi://127.0.0.1)
exten = _X.,n,Hangup() ; Please try this.
; End dialplan
The rest of the logic happens in adhearsion.
--
Thanks,
Ishwar.
On Thu, Jul 28, 2011 at 6:33 PM, Nikhil d.nik...@cem-solutions.netwrote:
Can you share the dialplan ,where SIP call is dialing...
Thanks
Nikhil
On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:
Hello everybody,
We have an asterisk 1.8.4.1 setup, connected to a PRI line.
We're currently facing an issue where asterisk does not recognise the
event when the called party declines/cuts the call. This happens
specifically over calls on a PRI line. For calls over SIP, call decline
event is captured properly.
I wasn't able to find a solution on the asterisk-users mailing list
archive. Any suggestions/help would be much appreiciated :) I can share the
relevant parts of the configuration files, if needed.
Here's an excerpt from asterisk logs for a SIP call.
-- SIP/x- requested special control 16, passing it to
SIP/x-0001
-- Started music on hold, class 'default', on SIP/x-0001
-- SIP/x- requested special control 20, passing it to
SIP/x-0001
-- Got SIP response 603 Decline back from 127.0.0.1:5063
-- SIP/x-0001 is busy
-- Stopped music on hold on SIP/x-0001
As you can see, on a SIP call, a call reject event is identified.
For a call over the PRI, on the other hand, this event is not recognised.
Here's an excerpt from asterisk log for a call over PRI.
Call from to .
-- Requested transfer capability: 0x10 - 3K1AUDIO
-- Called G11/x
-- Started music on hold, class 'default', on DAHDI/i1/y
-- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y
-- DAHDI/i1/x-18f8 is ringing
# At this point in time, x rejects the call. The event that's logged
in asterisk is the following:
-- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y
# And the call times out after the default 30s.
-- Nobody picked up in 3 ms
Is there a reason why asterisk doesn't recognise the call decline, and
does it need any configuration changes to enable this?
Thanks for your help.
--
Cheers,
Ishwar.
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[asterisk-users] call forwarding number from outside.
Hi! I need help regarding the following problem: when I receive a phone call to the PBX from the number 01234567890 rings the number 100, get up the phone, I transfer (assisted) to the number 100. When the 100 number rings, the display shows the number of those who transferred the call and not the number 01234567890. How can you solve this problem? Thanks and sorry for my English -- _ -- 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] call forwarding number from outside.
That`s the normal behavior of assisted transfers. Try a blind/non-assisted transfer, that should show the original callerid. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio Sent: Friday, July 29, 2011 2:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call forwarding number from outside. Hi! I need help regarding the following problem: when I receive a phone call to the PBX from the number 01234567890 rings the number 100, get up the phone, I transfer (assisted) to the number 100. When the 100 number rings, the display shows the number of those who transferred the call and not the number 01234567890. How can you solve this problem? Thanks and sorry for my English -- _ -- 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] call forwarding number from outside.
Thanks for the reply! I've tried and works, but isn't possible with the transfer assisted? thanks From: Mike Sent: Friday, July 29, 2011 8:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] call forwarding number from outside. That`s the normal behavior of assisted transfers. Try a blind/non-assisted transfer, that should show the original callerid. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio Sent: Friday, July 29, 2011 2:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call forwarding number from outside. Hi! I need help regarding the following problem: when I receive a phone call to the PBX from the number 01234567890 rings the number 100, get up the phone, I transfer (assisted) to the number 100. When the 100 number rings, the display shows the number of those who transferred the call and not the number 01234567890. How can you solve this problem? Thanks and sorry for my English -- _ -- 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] Why no traction for Windows version?
Hi, I haven't write any How to on it but below are some step by step instructions to run Asterisk on windows, 1-Install Cygwin. 2-Install build essentials in Cygwin. 3-Download Asterisk source (I used 1.4.x) and unzip it using tar (You may need to install tar manually as it is missing in some Cygwin default installations. Don't use windows unzip for as it will create some abnormal character in source and will make unexpected compile time errors) 4-Run bootstrap it will report any missing or lower version libs, prerequisite or tools. 5-You may need to manually install/upgrade tools like autoconf, automake etc depending on your Cygwin installation. 6-You manually need to download and compile termcap, ncurses. 7-Run configure. 8-Make menuselect and disable all non-required modules as it will save to resolve lot of not needed dependencies. 9-Run make 10-Resolve any missing reported by make. 11-After successful make run make install 12-Once make install okey you can run asterisk on Cygwin console and also directly run by double clicking on asterisk.exe in c:/Cygwin/usr/sbin/. Once you have compiled it you can copy asterisk.exe to any other system not having Cygwin installed by you have to care about following, 1-You must have to create required directories structure like Cygwin on system drive. 2-You must need to copy required Cygwin DLLs to new systems \windows\system32\ folder. You can identify required DLLs by trying to run asterisk.exe and it will report missing DLLs one by one. I did just for my experiment and fun and was able to make successful SIP calls using static files configuration. However I suggest to use SIPx, Yate or FreeSWITCH if you want to stick with windows as that have native windows ports and have all required features you need in a PABX or VoIP switch. Regards, Faisal Hanif -- _ -- 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] Questions about FMFM with linked servers
Did you tried to execute Set(CALLERID(num)=you-required-callerid)? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman Sent: Friday, July 29, 2011 1:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Questions about FMFM with linked servers All; In a linked server environment, running Asterisk 1.6 I am noticing that when a call is placed from server A to server B (via 4 digit extension) and server B ext has a FMFM to call their mobile, the mobile phone shows the default caller ID setting on server B instead of the actual caller id of the person who initiated the call on server A. This scenario, of course, works in the event a call in placed via the PSTN into Server A (or B) and rings the FMFM extension. In this case, the mobile phones sees the correct (initial) caller ID on the mobile. Thanks! --Dovey -- _ -- 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] Why no traction for Windows version?
Hi, One more thing previously there was a project named as AstWin which was maintaining asterisk's port to windows and providing an installable package of Asterisk for windows. I am not aware about current state of project but, I have installation package of Asterisk for windows version 1.2. If anyone need it contact me direct at email imfa...@gmail.com I will send the software as attachment. Regards, Faisal Hanif -- _ -- 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] Why no traction for Windows version?
On Thursday 28 Jul 2011, Gilles wrote: On Thu, 28 Jul 2011 12:04:38 +0500, Faisal Hanif fai...@vopium.com wrote: I have tried asterisk on windows XP using Cygwin and it worked fine. Would you mind explaining how to do this? I hate to sound patronising but, if you need to ask how to install Cygwin on Windows, you really shouldn't bother. You *will* find it quicker, cheaper, easier and less frustrating in the long run just to get a scrap PC and install Linux on that. Especially now there are dedicated distros which install Linux and Asterisk ready to go. What you are wanting to do is, in essence, like teaching a gerbil to bark. It's an extraordinary effort to go to when you can get puppies anywhere; and at the end of the day, it's still not, and never will be, a proper dog. -- AJS Answers come *after* questions. -- _ -- 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] hide google voice number
Thank you Terry, CallWithUs is what I am looking for, the most feature rich VoIP service!!! I hope it will not be difficult for me to have it working with Asterisk and OpenBTS (It's worth to see what OpenBTS is) On Thu, Jul 28, 2011 at 12:48 PM, Terry Brummell te...@brummell.net wrote: Yes, they used to allow it. Like CallWithUs and Voip.ms (and I'm sure other VTSP's) do. -- *From:* A.H. Jos *Sent:* Thu 7/28/2011 12:01 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] hide google voice number Do you mean that was possible to set the CID in the early days of GVoice? On Thu, Jul 28, 2011 at 4:33 PM, Terry Brummell te...@brummell.netwrote: Google Voice will show your number no matter what, there was a problem with abuse when they let you send the CID in the early days. Pretty sure there is nothing you can do about it. -- *From:* A.H. Jos *Sent:* Thu 7/28/2011 9:22 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] hide google voice number Hi list, I have Asterisk speaking with google talk, is there any way to set or at least hide my google voice number when I call others? thanks for your help, AHJos -- _ -- 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
[asterisk-users] Asterisk SIP authentication against [section] instead of username
Hello, Asterisk seems to try to authenticate incoming INVITE based on the [section] in sip.conf and not the username specified. I just removed the insecure option from my sip.conf requesting every connection to be authenticated. I added the match_auth_username=yes in the [general] section for extra security. To make it work, I have to use the same [section] identifier as username. This is really bad because if multiple provider are giving me the same username, it doesn't work. If I put the following data in sip.conf, it doesn't work. Asterisk return the following error: [2011-07-29 04:55:30] WARNING[9971]: chan_sip.c:13205 check_auth: username mismatch, have GoodProvider, digest has myusername [GoodProvider] username=myusername auth=myusername defaultuser=myusername secret=verydifficultpass type=friend host=pbx.goodprovider.com canreinvite=No dtmfmode=rfc2833 context=from-outside accountcode=GoodProvider disallow=all allow=ulaw If I put the following data in sip.conf, it does work: [myusername] username=myusername auth=myusername defaultuser=myusername secret=verydifficultpass type=friend host=pbx.goodprovider.com canreinvite=No dtmfmode=rfc2833 context=from-outside accountcode=GoodProvider disallow=all allow=ulaw I check the INVITE from the GoodProvider and it is sending myusername Am I doing something wrong or is really asterisk checking the wrong section? 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
Re: [asterisk-users] hide google voice number
Voip.ms actually offers more features. Depends on your needs. I use both as long distance carriers. My DID's are from Voip.ms. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A.H. Jos Sent: Friday, July 29, 2011 4:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hide google voice number Thank you Terry, CallWithUs is what I am looking for, the most feature rich VoIP service!!! I hope it will not be difficult for me to have it working with Asterisk and OpenBTS (It's worth to see what OpenBTS is) On Thu, Jul 28, 2011 at 12:48 PM, Terry Brummell te...@brummell.net wrote: Yes, they used to allow it. Like CallWithUs and Voip.ms (and I'm sure other VTSP's) do. From: A.H. Jos Sent: Thu 7/28/2011 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hide google voice number Do you mean that was possible to set the CID in the early days of GVoice? On Thu, Jul 28, 2011 at 4:33 PM, Terry Brummell te...@brummell.net wrote: Google Voice will show your number no matter what, there was a problem with abuse when they let you send the CID in the early days. Pretty sure there is nothing you can do about it. From: A.H. Jos Sent: Thu 7/28/2011 9:22 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] hide google voice number Hi list, I have Asterisk speaking with google talk, is there any way to set or at least hide my google voice number when I call others? thanks for your help, AHJos -- _ -- 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
[asterisk-users] X86_64 Compilation Issue
Hi, compiling up a new installation of Asterisk 1.8.5 on CentOS 6 X86_64 and am seeing the following when running the make: /usr/bin/ld: skipping incompatible /usr/lib/libpam.so when searching for -lpam /usr/bin/ld: skipping incompatible /usr/lib/libssl.so when searching for -lssl /usr/bin/ld: skipping incompatible /usr/lib/libssl.a when searching for -lssl /usr/bin/ld: skipping incompatible /usr/lib/libcrypto.so when searching for -lcrypto /usr/bin/ld: skipping incompatible /usr/lib/libcrypto.a when searching for -lcrypto How can I get Asterisk to pick up the 64 bit version of the libraries instead of the 32 bit ones ? Is it just a case of updating LD_LIBRARY_PATH ? -- Thanks, Phil -- _ -- 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] How to use these feature of Asterisk
Hi List, I want to use these features but nothing was found after googling . please give me some examples Asterisk CLI prompt Changing the CLI Prompt The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable that you set from the Unix shell before starting the Asterisk CLI (not the server). You may include the following variables, that will be replaced by the current value by Asterisk: %d Date (year-month-date) %s Asterisk system name (from asterisk.conf) %h Full hostname %H Short hostname %t Time %% Percent sign %# '#' if Asterisk is run in console mode, '' if running as remote console %Cn[;n] Change terminal foreground (and optional background) color to specified *A full list of colors may be found in include/asterisk/term.h* On Linux systems, you may also use %l1 Load average over past minute %l2 Load average over past 5 minutes %l3 Load average over past 15 minutes %l4 Process fraction (processes running / total processes) %l5 The most recently allocated pid -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- 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] How to use these feature of Asterisk
Use This Information.
You can customize the prompt a bit, if the default prompt is too dull for
you. First add these lines to */etc/asterisk/extensions.conf* in the
[globals] section:
${ENV(UNIX)}
${ENV(ASTERISK_PROMPT)}
Then in */etc/profile* on the Asterisk server, set the ASTERISK_PROMPT
values:
ASTERISK_PROMPT='%t, %l2, %h* '
export PATH USER LOGNAME MAIL HOSTNAME HISTSIZE INPUTRC ASTERISK_PROMPT
Your *export* variables will probably be different; just tack
ASTERISK_PROMPT on at the end. Reboot, run *asterisk -r* from your X
terminal, and voilá! The prompt is customized and your colors do not change:
*17:51:30, 0.54, asterisk1.alrac.net**
On Fri, Jul 29, 2011 at 4:26 PM, virendra bhati virbh...@gmail.com wrote:
Hi List,
I want to use these features but nothing was found after googling . please
give me some examples
Asterisk CLI prompt
Changing the CLI Prompt
The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable
that
you set from the Unix shell before starting the Asterisk CLI (not the
server).
You may include the following variables, that will be replaced by
the current value by Asterisk:
%d Date (year-month-date)
%s Asterisk system name (from asterisk.conf)
%h Full hostname
%H Short hostname
%t Time
%% Percent sign
%# '#' if Asterisk is run in console mode, '' if running as remote console
%Cn[;n] Change terminal foreground (and optional background) color to
specified
*A full list of colors may be found in include/asterisk/term.h
*
On Linux systems, you may also use
%l1 Load average over past minute
%l2 Load average over past 5 minutes
%l3 Load average over past 15 minutes
%l4 Process fraction (processes running / total processes)
%l5 The most recently allocated pid
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
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Re: [asterisk-users] How to use these feature of Asterisk
On Friday 29 Jul 2011, virendra bhati wrote: Hi List, I want to use these features but nothing was found after googling . please give me some examples Asterisk CLI prompt Changing the CLI Prompt The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable that you set from the Unix shell before starting the Asterisk CLI (not the server). All you have to do is set an environment variable, and then make sure that it gets passed on to the `asterisk -r`process. For example: # export ASTERISK_PROMPT=Asterisk@%h: # asterisk -vr or even just # ASTERISK_PROMPT=Asterisk@%h: asterisk -vr -- AJS Answers come *after* questions. -- _ -- 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] Why no traction for Windows version?
- Original Message - What you are wanting to do is, in essence, like teaching a gerbil to bark. It's an extraordinary effort to go to when you can get puppies anywhere; and at the end of the day, it's still not, and never will be, a proper dog. +1 I could not have said it better myself. --tim -- _ -- 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] call forwarding number from outside.
The issue with assisted transfer is that the assisting transferer is a second call Outside - A A answers A calls B to tell them they have a call (call #2 with ID of A A transfers Outside but the ID stays A Blind Transfer Outside - A A answers A blind transfers to B (1 call - keeps ID. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio Sent: Friday, July 29, 2011 2:10 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] call forwarding number from outside. Thanks for the reply! I've tried and works, but isn't possible with the transfer assisted? thanks From: Mike mailto:l...@net-wall.com Sent: Friday, July 29, 2011 8:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' mailto:asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call forwarding number from outside. That`s the normal behavior of assisted transfers. Try a blind/non-assisted transfer, that should show the original callerid. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio Sent: Friday, July 29, 2011 2:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call forwarding number from outside. Hi! I need help regarding the following problem: when I receive a phone call to the PBX from the number 01234567890 rings the number 100, get up the phone, I transfer (assisted) to the number 100. When the 100 number rings, the display shows the number of those who transferred the call and not the number 01234567890. How can you solve this problem? Thanks and sorry for my English _ -- _ -- 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] call forwarding number from outside.
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, July 29, 2011 9:06 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: jim.smith...@debsinc.com Subject: Re: [asterisk-users] call forwarding number from outside. The issue with assisted transfer is that the assisting transferer is a second call Outside - A A answers A calls B to tell them they have a call (call #2 with ID of A A transfers Outside but the ID stays A Blind Transfer Outside - A A answers A blind transfers to B (1 call - keeps ID. From the output of core show application dial: f: Force the callerid of the *calling* channel to be set as the extension associated with the channel using a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be set to anything other than the number assigned to the caller. -- _ -- 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] call forwarding number from outside.
On 07/29/2011 09:12 AM, Eric Wieling wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, July 29, 2011 9:06 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: jim.smith...@debsinc.com Subject: Re: [asterisk-users] call forwarding number from outside. The issue with assisted transfer is that the assisting transferer is a second call Outside - A A answers A calls B to tell them they have a call (call #2 with ID of A A transfers Outside but the ID stays A Blind Transfer Outside - A A answers A blind transfers to B (1 call - keeps ID. From the output of core show application dial: f: Force the callerid of the *calling* channel to be set as the extension associated with the channel using a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be set to anything other than the number assigned to the caller. In Asterisk 1.8 and later, if the phones (endpoints) support it, the connected party display on the phone will update *after* the transfer has been completed to show who the person is talking to (not the person who performed the transfer). -- 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
Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line
HI Eric, Nikhil, Thanks a lot for the responses. Bear with me a little as I'm very new to asterisk. I reproduced the problem using standard dialplan. The following are the configuration files: *chan_dahdi.conf* *[trunkgroups] [channels] language=en nationalprefix=+91 pridialplan=national ; or national or local? usecallerid=yes hidecallerid=no callwaiting=yes allow_call_waiting_calls=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;Might have to play with this. callerid=asreceived facilityenable=yes priindication=outofband cidsignalling=dtmf ; most likely dtmf based on the India link below cidstart=polarity_IN #include dahdi-channels.conf* *extensions.conf:* *[frompstn] exten = x,1,Ringing exten = x,n,Dial(Dahdi/G11/y) exten = x,n,Hangup()* When a calls x, we dial y This is what I find in the logs: -- Accepting call from 'a' to 'x on channel 0/24, span 1 -- Executing [x@frompstn:1] Ringing(DAHDI/i1/a-136, ) in new stack -- Executing [x@frompstn:2] Dial(DAHDI/i1/aa-136, Dahdi/G11/yy) in new stack -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called G11/yy -- DAHDI/i1/y-137 is proceeding passing it to DAHDI/i1/a-136 -- DAHDI/i1/y-137 is ringing # At this point, y rejected the call. Asterisk doesn't recognise this, and continues to dial for 30s(the default) before hanging up. -- DAHDI/i1/y-137 is making progress passing it to DAHDI/i1/a-136 -- Nobody picked up in 3 ms I'll try out pri intense debug during night time when the traffic on our servers is low, and update the list with the logs. In the mean time, is there anything missing in the configuration that rejected calls aren't detected? -- Thanks, Ishwar. On Fri, Jul 29, 2011 at 1:10 AM, Eric Wieling ewiel...@nyigc.com wrote: 1) You have to have channels configured for your PRI SOMEWHERE in the Asterisk DAHDI configs. 2) Can't troubleshoot when everything important is masked by an AGI script. Reproduce the problem using standard dialplan stuff. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Ishwar Sridharan Sent: Thursday, July 28, 2011 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line Hi Eric, There weren't any lines with PRI channel = in the chan_dahdi.conf However, I added the lines you'd mentioned, near the top of the file. Still, no difference in either the behaviour or the asterisk output. Please note that as soon as the call lands on asterisk, we pass the control over to adhearsion. Does that affect how events are handled in asterisk? -- Thanks, Ishwar. On Thu, Jul 28, 2011 at 6:37 PM, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk- users- boun...@lists.digium.com] On Behalf Of Nikhil Sent: Thursday, July 28, 2011 9:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line Can you share the dialplan ,where SIP call is dialing... Thanks Nikhil On 07/28/2011 06:15 PM, Ishwar Sridharan wrote: Hello everybody, We have an asterisk 1.8.4.1 setup, connected to a PRI line. We're currently facing an issue where asterisk does not recognise the event when the called party declines/cuts the call. This happens specifically over calls on a PRI line. For calls over SIP, call decline event is captured properly. I wasn't able to find a solution on the asterisk-users mailing list archive. Any suggestions/help would be much appreiciated :) I can share the relevant parts of the configuration files, if needed. Here's an excerpt from asterisk logs for a SIP call. -- SIP/x- requested special control 16, passing it to SIP/x-0001 -- Started music on hold, class 'default', on SIP/x- 0001 -- SIP/x- requested special control 20, passing it to SIP/x-0001 -- Got SIP response 603 Decline back from 127.0.0.1:5063 http://127.0.0.1:5063/ -- SIP/x-0001 is busy -- Stopped music on hold on SIP/x-0001 As you can see, on a SIP call, a call reject event is identified. For a call
Re: [asterisk-users] X86_64 Compilation Issue
On 07/29/2011 06:56 AM, --[ UxBoD ]-- wrote: Hi, compiling up a new installation of Asterisk 1.8.5 on CentOS 6 X86_64 and am seeing the following when running the make: /usr/bin/ld: skipping incompatible /usr/lib/libpam.so when searching for -lpam /usr/bin/ld: skipping incompatible /usr/lib/libssl.so when searching for -lssl /usr/bin/ld: skipping incompatible /usr/lib/libssl.a when searching for -lssl /usr/bin/ld: skipping incompatible /usr/lib/libcrypto.so when searching for -lcrypto /usr/bin/ld: skipping incompatible /usr/lib/libcrypto.a when searching for -lcrypto How can I get Asterisk to pick up the 64 bit version of the libraries instead of the 32 bit ones ? Is it just a case of updating LD_LIBRARY_PATH ? -- Thanks, Phil Did you run configure with --libdir=/usr/lib64 ? -- _ -- 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] call forwarding number from outside.
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, July 29, 2011 8:49 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call forwarding number from outside. On 07/29/2011 09:12 AM, Eric Wieling wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, July 29, 2011 9:06 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: jim.smith...@debsinc.com Subject: Re: [asterisk-users] call forwarding number from outside. The issue with assisted transfer is that the assisting transferer is a second call Outside - A A answers A calls B to tell them they have a call (call #2 with ID of A A transfers Outside but the ID stays A Blind Transfer Outside - A A answers A blind transfers to B (1 call - keeps ID. From the output of core show application dial: f: Force the callerid of the *calling* channel to be set as the extension associated with the channel using a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be set to anything other than the number assigned to the caller. In Asterisk 1.8 and later, if the phones (endpoints) support it, the connected party display on the phone will update *after* the transfer has been completed to show who the person is talking to (not the person who performed the transfer). -- 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 Couple of questions - This magic trick is contained in app_dial? Functionality is inherent to 1.8/10.X structure so we can't re-invent this in our old 1.4/1.6 installs? -- _ -- 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] call forwarding number from outside.
On 07/29/2011 10:41 AM, Danny Nicholas wrote: snip Couple of questions - This magic trick is contained in app_dial? Functionality is inherent to 1.8/10.X structure so we can't re-invent this in our old 1.4/1.6 installs? No, it's core functionality, implemented in the channel drivers and using control frames that pass through bridges. It would be a large amount of effort to implement it again in 1.4/1.6. It extends well beyond simple dialing, as it can receive updates across external protocols and pass them along, it handles call redirection, and various other features. -- 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
Re: [asterisk-users] call forwarding number from outside.
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, July 29, 2011 9:48 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call forwarding number from outside. On 07/29/2011 10:41 AM, Danny Nicholas wrote: snip Couple of questions - This magic trick is contained in app_dial? Functionality is inherent to 1.8/10.X structure so we can't re-invent this in our old 1.4/1.6 installs? No, it's core functionality, implemented in the channel drivers and using control frames that pass through bridges. It would be a large amount of effort to implement it again in 1.4/1.6. It extends well beyond simple dialing, as it can receive updates across external protocols and pass them along, it handles call redirection, and various other features. -- 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 As I suspected sigh -- _ -- 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] call forwarding number from outside.
So I can't do anything? -- From: Kevin P. Fleming kpflem...@digium.com Sent: Friday, July 29, 2011 4:48 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call forwarding number from outside. On 07/29/2011 10:41 AM, Danny Nicholas wrote: snip Couple of questions - This magic trick is contained in app_dial? Functionality is inherent to 1.8/10.X structure so we can't re-invent this in our old 1.4/1.6 installs? No, it's core functionality, implemented in the channel drivers and using control frames that pass through bridges. It would be a large amount of effort to implement it again in 1.4/1.6. It extends well beyond simple dialing, as it can receive updates across external protocols and pass them along, it handles call redirection, and various other features. -- 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 -- _ -- 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] call forwarding number from outside.
Upgrade to 1.8/10.0 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio Sent: Friday, July 29, 2011 10:04 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call forwarding number from outside. So I can't do anything? -- From: Kevin P. Fleming kpflem...@digium.com Sent: Friday, July 29, 2011 4:48 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call forwarding number from outside. On 07/29/2011 10:41 AM, Danny Nicholas wrote: snip Couple of questions - This magic trick is contained in app_dial? Functionality is inherent to 1.8/10.X structure so we can't re-invent this in our old 1.4/1.6 installs? No, it's core functionality, implemented in the channel drivers and using control frames that pass through bridges. It would be a large amount of effort to implement it again in 1.4/1.6. It extends well beyond simple dialing, as it can receive updates across external protocols and pass them along, it handles call redirection, and various other features. -- 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 -- _ -- 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] X86_64 Compilation Issue
Thank you Dave. -- Thanks, Phil - Original Message - On 07/29/2011 06:56 AM, --[ UxBoD ]-- wrote: Hi, compiling up a new installation of Asterisk 1.8.5 on CentOS 6 X86_64 and am seeing the following when running the make: /usr/bin/ld: skipping incompatible /usr/lib/libpam.so when searching for -lpam /usr/bin/ld: skipping incompatible /usr/lib/libssl.so when searching for -lssl /usr/bin/ld: skipping incompatible /usr/lib/libssl.a when searching for -lssl /usr/bin/ld: skipping incompatible /usr/lib/libcrypto.so when searching for -lcrypto /usr/bin/ld: skipping incompatible /usr/lib/libcrypto.a when searching for -lcrypto How can I get Asterisk to pick up the 64 bit version of the libraries instead of the 32 bit ones ? Is it just a case of updating LD_LIBRARY_PATH ? -- Thanks, Phil Did you run configure with --libdir=/usr/lib64 ? -- _ -- 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
[asterisk-users] Accept the dtmf input in call patch
Hi team, Is it possible to capture dtmf input once call is patched between a-party and b-party? Also on dtmf input issue hangup request to b-party with out disconnecting A-party. How is this scenario implemented in dialplan? Thanks Vinod Dharashive Sent from BlackBerry® on Airtel -- _ -- 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] call forwarding number from outside.
ok I'll do it Monday, and how you handle it with the version 1.10? Thanks -- From: Danny Nicholas da...@debsinc.com Sent: Friday, July 29, 2011 5:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call forwarding number from outside. Upgrade to 1.8/10.0 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio Sent: Friday, July 29, 2011 10:04 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call forwarding number from outside. So I can't do anything? -- From: Kevin P. Fleming kpflem...@digium.com Sent: Friday, July 29, 2011 4:48 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call forwarding number from outside. On 07/29/2011 10:41 AM, Danny Nicholas wrote: snip Couple of questions - This magic trick is contained in app_dial? Functionality is inherent to 1.8/10.X structure so we can't re-invent this in our old 1.4/1.6 installs? No, it's core functionality, implemented in the channel drivers and using control frames that pass through bridges. It would be a large amount of effort to implement it again in 1.4/1.6. It extends well beyond simple dialing, as it can receive updates across external protocols and pass them along, it handles call redirection, and various other features. -- 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 -- _ -- 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] Capturing call Reject/Decline events on a PRI line
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Ishwar Sridharan Sent: Friday, July 29, 2011 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line -- Called G11/yy -- DAHDI/i1/y-137 is proceeding passing it to DAHDI/i1/a-136 -- DAHDI/i1/y-137 is ringing # At this point, y rejected the call. Asterisk doesn't recognise this, and continues to dial for 30s(the default) before hanging up. -- DAHDI/i1/y-137 is making progress passing it to DAHDI/i1/a-136 -- Nobody picked up in 3 ms Exactly *how* is y rejecting the call? What is y? An ISDN Phone? A POTS phone? A PSTN telephone number? I am assuming that y is a PSTN TN. It is starting to sound like that is not the case. -- _ -- 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] Accept the dtmf input in call patch
Yep. Look the dtails of option of Dial command and features.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vinod Dharashive Sent: Friday, July 29, 2011 8:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Accept the dtmf input in call patch Hi team, Is it possible to capture dtmf input once call is patched between a-party and b-party? Also on dtmf input issue hangup request to b-party with out disconnecting A-party. How is this scenario implemented in dialplan? Thanks Vinod Dharashive Sent from BlackBerryR on Airtel -- _ -- 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] Capturing call Reject/Decline events on a PRI line
HI Eric, is a mobile number in India, and the call id rejected by ending the call from the mobile. BTW, why is the mail going to asterisk-users-bounces? -- Thanks, Ishwar. On Fri, Jul 29, 2011 at 9:34 PM, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Ishwar Sridharan Sent: Friday, July 29, 2011 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line -- Called G11/yy -- DAHDI/i1/y-137 is proceeding passing it to DAHDI/i1/a-136 -- DAHDI/i1/y-137 is ringing # At this point, y rejected the call. Asterisk doesn't recognise this, and continues to dial for 30s(the default) before hanging up. -- DAHDI/i1/y-137 is making progress passing it to DAHDI/i1/a-136 -- Nobody picked up in 3 ms Exactly *how* is y rejecting the call? What is y? An ISDN Phone? A POTS phone? A PSTN telephone number? I am assuming that y is a PSTN TN. It is starting to sound like that is not the case. -- _ -- 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] Capturing call Reject/Decline events on a PRI line
Hello, We enable pri intense debug with the standard asterisk PRI dialplan, collected the logs and you can find the logs attached to the mail. After the call was made, the called party cut the call, and asterisk doesn't seem to recognise the event. I can't make much sense of the logs given my non-existent background in telephony. Would somebody here help me figure why the event wasn't captured? -- Thanks, Ishwar. On Fri, Jul 29, 2011 at 11:54 PM, Ishwar Sridharan ish...@exotel.in wrote: HI Eric, is a mobile number in India, and the call id rejected by ending the call from the mobile. BTW, why is the mail going to asterisk-users-bounces? -- Thanks, Ishwar. On Fri, Jul 29, 2011 at 9:34 PM, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Ishwar Sridharan Sent: Friday, July 29, 2011 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line -- Called G11/yy -- DAHDI/i1/y-137 is proceeding passing it to DAHDI/i1/a-136 -- DAHDI/i1/y-137 is ringing # At this point, y rejected the call. Asterisk doesn't recognise this, and continues to dial for 30s(the default) before hanging up. -- DAHDI/i1/y-137 is making progress passing it to DAHDI/i1/a-136 -- Nobody picked up in 3 ms Exactly *how* is y rejecting the call? What is y? An ISDN Phone? A POTS phone? A PSTN telephone number? I am assuming that y is a PSTN TN. It is starting to sound like that is not the case. -- _ -- 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 1 Done handling message for SAPI/TEI=0/0 Span: 1 Processing event: PRI_EVENT_RINGING -- DAHDI/i1/09880847047-22c is ringing 1 TEI: 0 State 7(Multi-frame established) 1 V(A)=96, V(S)=96, V(R)=34 1 K=7, RC=1, l3initiated=0, reject_except=0, ack_pend=0 1 T200_id=0, N200=3, T203_id=1 1 [ 02 01 44 c0 08 02 81 1a 03 1e 02 8a 81 1e 02 8a 88 ] 1 Informational frame: 1 SAPI: 00 C/R: 1 EA: 0 1 TEI: 000EA: 1 1 N(S): 034 0: 0 1 N(R): 096 P: 0 1 13 bytes of data 1 Protocol Discriminator: Q.931 (8) len=13 1 TEI=0 Call Ref: len= 2 (reference 282/0x11A) (Sent to originator) 1 Message Type: PROGRESS (3) 1 [1e 02 8a 81] 1 Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Network beyond the interworking point (10) 1Ext: 1 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ] 1 [1e 02 8a 88] 1 Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Network beyond the interworking point (10) 1Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] 1 -- Got ACK for N(S)=96 to (but not including) N(S)=96 1 -- T200 requested to stop when not started 1 T203 requested to start without stopping first 1 -- Starting T203 timer 1 Received message for call 0xd6885d0 on 0x2c5c0530 TEI/SAPI 0/0, call-pri is 0x2c5c0530 TEI/SAPI 0/0 1 -- Processing IE 30 (cs0, Progress Indicator) 1 -- Processing IE 30 (cs0, Progress Indicator) 1 1 TEI: 0 State 7(Multi-frame established) 1 V(A)=96, V(S)=96, V(R)=35 1 K=7, RC=1, l3initiated=0, reject_except=0, ack_pend=0 1 T200_id=0, N200=3, T203_id=1 1 [ 02 01 01 46 ] 1 Supervisory frame: 1 SAPI: 00 C/R: 1 EA: 0 1 TEI: 000EA: 1 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] 1 N(R): 035 P/F: 0 1 0 bytes of data 1 Done handling message for SAPI/TEI=0/0 Span: 1 Processing event: PRI_EVENT_PROGRESS -- DAHDI/i1/09880847047-22c is making progress passing it to DAHDI/i1/8088919888-22b 1 TEI: 0 State 7(Multi-frame established) 1 V(A)=96, V(S)=96, V(R)=35 1 K=7, RC=1, l3initiated=0, reject_except=0, ack_pend=0 1 T200_id=0, N200=3, T203_id=0 1 [ 00 01 01 47 ] 1 Supervisory frame: 1 SAPI: 00 C/R: 0 EA: 0 1 TEI: 000EA: 1 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] 1 N(R): 035 P/F: 1 1 0 bytes of data 1 -- Starting T200 timer 1 1 TEI: 0 State 8(Timer recovery) 1 V(A)=96, V(S)=96, V(R)=35 1 K=7, RC=0, l3initiated=0, reject_except=0, ack_pend=0 1 T200_id=1, N200=3, T203_id=0 1 [ 02 01 01 c1 ] 1 Supervisory frame: 1 SAPI: 00 C/R: 1 EA: 0 1 TEI: 000EA: 1 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] 1 N(R): 096 P/F: 1 1 0 bytes of data 1 1 TEI: 0 State 8(Timer recovery) 1 V(A)=96, V(S)=96,
Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line
We enable pri intense debug with the standard asterisk PRI dialplan, collected the logs and you can find the logs attached to the mail. After the call was made, the called party cut the call, and asterisk doesn't seem to recognise the event. I can't make much sense of the logs given my non-existent background in telephony. Would somebody here help me figure why the event wasn't captured? There is no event for Asterisk to recognize. The PROGRESS message just says that there is an audio message available for the caller to listen to. Asterisk just passes the indication to the peer channel and opens the audio path. It is the caller who must recognize any audio message that their call has been dropped. As far as ISDN is concerned, the call has not been answered yet so Asterisk must keep waiting. Richard -- _ -- 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] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
Hi everyone, Asterisk 1.6.2.19 has a bug per: https://issues.asterisk.org/jira/browse/ASTERISK-18103 What is the general time to fix this? I think a similar thing is also noted in 1.8x install. Is it not going to be taken care of because it's 1.6x ? 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
Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
On 11-07-29 06:12 PM, Bruce B wrote: Hi everyone, Asterisk 1.6.2.19 has a bug per: https://issues.asterisk.org/jira/browse/ASTERISK-18103 What is the general time to fix this? I think a similar thing is also noted in 1.8x install. Is it not going to be taken care of because it's 1.6x ? 1.6.2.19 was to be the last release of the 1.6.2 branch, so I'm not sure if another build is expected. However the issue does reference 1.6.2.19.1 so it is possible. However, you can see what changed between 1.6.2.18 and 1.6.2.19 in an attempted to narrow down the bug. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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
Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
On 07/29/2011 06:20 PM, Paul Belanger wrote: On 11-07-29 06:12 PM, Bruce B wrote: Hi everyone, Asterisk 1.6.2.19 has a bug per: https://issues.asterisk.org/jira/browse/ASTERISK-18103 What is the general time to fix this? I think a similar thing is also noted in 1.8x install. Is it not going to be taken care of because it's 1.6x ? 1.6.2.19 was to be the last release of the 1.6.2 branch, so I'm not sure if another build is expected. However the issue does reference 1.6.2.19.1 so it is possible. However, you can see what changed between 1.6.2.18 and 1.6.2.19 in an attempted to narrow down the bug. If it was a regression from 1.6.2.18 to 1.6.2.19, then it will be fixed. -- 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] Tutorial on the Asterisk Manager Interface
I've used the manager interface to make calls successfully, now I'd like a look at some of he other ways it can be used. I've seen references to its use to perform call cut off and rate CDRs. Is anyone aware of a reference or tutorial I could look at? Bruce Ferrell -- _ -- 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] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
I think this should be a quick fix since it's rendering the latest stable version useless and making the impression that it was released just to break things and force people onto 1.8x. Just a thought...no blame game. But really something like this should be tackled quickly. No point to break things so badly on the last stable version. Regards, On Fri, Jul 29, 2011 at 6:23 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 07/29/2011 06:20 PM, Paul Belanger wrote: On 11-07-29 06:12 PM, Bruce B wrote: Hi everyone, Asterisk 1.6.2.19 has a bug per: https://issues.asterisk.org/**jira/browse/ASTERISK-18103https://issues.asterisk.org/jira/browse/ASTERISK-18103 What is the general time to fix this? I think a similar thing is also noted in 1.8x install. Is it not going to be taken care of because it's 1.6x ? 1.6.2.19 was to be the last release of the 1.6.2 branch, so I'm not sure if another build is expected. However the issue does reference 1.6.2.19.1 so it is possible. However, you can see what changed between 1.6.2.18 and 1.6.2.19 in an attempted to narrow down the bug. If it was a regression from 1.6.2.18 to 1.6.2.19, then it will be fixed. -- 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-**usershttp://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
