Re: [asterisk-users] How to know how many calls are into hold byasterisk command
hi, Hint will work all VoIP hardware or specific hardware device ? I am planing to using CISCO 79XX series so please suggest me.. And What about softphone ? On Wed, Apr 20, 2011 at 8:57 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati *Sent:* Wednesday, April 20, 2011 2:19 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to know how many calls are into hold byasterisk command Hi All, I never use hint in asterisk and I don't know how to use Hints and why we use Hints. If you give some details with example then it will be helpful for me. On Tue, Apr 19, 2011 at 10:52 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati *Sent:* Tuesday, April 19, 2011 12:10 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] How to know how many calls are into hold byasterisk command Hi All, Is it possible o know how many call are into hold ? who are on hold ? By whom these extension are on hold ? And after 60 sec asterisk will call them automatically as Call Parking do? I wan to make this concept to my PBX system... Thanks in advance -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer *[Danny Nicholas] * *#1, 2 and 3 can be accomplished using hints and/or AMI. I doubt #4 is possible, but hey I’ve been wrong plenty of times before.* -- _ -- 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 -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer *[Danny Nicholas] * *I personally use hints to allow my Polycom phones to more effectively communicate with my Asterisk server. Each SIP extension and each DAHDI line has a hint, so I can keep track of them using the “buddies” feature on the polycom phone. I also have a “roll-your-own” Apache interface that utilizes these hints.* -- _ -- 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 -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk 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] allowguest=yes, how?
Op 20-04-11 21:47, Danny Nicholas schreef:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Paul van der Vlis
Sent: Wednesday, April 20, 2011 2:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] allowguest=yes, how?
Hello,
I want that people from other servers like ekiga.net can make calls to
my users. When I do an allowguest=no then people from other domains
cannot call me. So I think I need allowguest=yes.
cut setup
Is this a good setup?
[Danny Nicholas]
Talking a little out of school, but the safest method is going to be to
set up one context with allowguest=yes that has ABSOLUTELY NO DIALING
PRIVILEDGES, otherwise you will open Pandora's box.
Maybe you mean something like this?
extensions.conf:
-
[default]
include = users
[dialout]
include = users
exten=_0.,1,Dial(SIP/trunk/0${EXTEN:1},30,tT)
[users]
exten=6001,1,Dial(SIP/paul,20)
exten=6002,1,Dial(SIP/ann,20)
(...)
sip.conf:
---
[general]
context=default
allowguest=no
(...)
[guests]
context=default
allowguest=yes
[trunk]
context=dialout
(...)
[phone-paul]
context=dialout
(...)
[phone-ann]
context=dialout
(...)
---
Thanks for your help!
With regards,
Paul van der Vlis.
--
http://www.vandervlis.nl/
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Re: [asterisk-users] issue with installtion asterisk
Hi Jose thank you for your help after this command yum install asterisk16-configs* all is ok now i can install asterisk without problem Kind Regards Salah. 2011/4/20 Jose P. Espinal j...@slackware-es.com but when i finished i found a file in etc asterisk but is empty You found a file in etc asterisk?, what file? My English is not very good (sorry for that), but if you meant that you found '/etc/asterisk' directory, but it was empty, then you are missing the configuration files, do this: yum install asterisk16-configs* For a list of available/installed packages regarding to Asterisk, do this: yum list asterisk16* The outer right column shows installed/available (asterisk-current/digium-current) packages. Regards, -- Jose P. Espinal http://www.eSlackware.com http://www.eslackware.com/ IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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] Meetme Time Limit?
Hi, You can use Meetme(1234,dL(1800)) where 1800 = 6 hours after 6 hours channel is hanf up regards Dhaval On Mon, Apr 18, 2011 at 9:31 PM, Bryant Zimmerman brya...@zktech.comwrote: Is there a way to place a hangup time on a dynamic Meetme conference. I am using Page() with a Meetme conf and I have had a few instances where someone from a wifi voip phone looses ip while doing a page and the page never hangs up. I have to kill it. I need to somehow limit the page so after a worse case 2Min timeout it hangs up. Thanks Bryant -- _ -- 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] Voicemail forward issue
Hi all, I have that when voice mail is forward to any guys then some text will include into recorded file like this is a forward voice mail from extension or Mr. Virendra the actual file will play Is it possible ? if yes then how ? -- - 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
[asterisk-users] [asterisk-user] Can't get hostname on asterisk dialplan by ENV()
Hi Friend,
Can't get hostname environment variable on asterisk dialplan.
Help me about how to get hostname environment variable on asterisk dialplan.
I have written export HOSTNAME in /root/.bash_profile and when i execute
echo $HOSTNAME then get right hostname but not success through asterisk
dialplan.
Get environment variable path right value through below statement.
exten = XXX,n,NoOp(--- ${ENV(PATH)})
*I have tried like this:*
exten = XXX,n,Set(CDR(hostname)=${System(echo $HOSTNAME)})
exten = XXX,n,Set(CDR(hostname)=${ENV(HOSTNAME)})
Thanks in advance.
--
Best Regards,
Rajnikant Vanza
Call : +91-9737456583
Software Engineer
---
Working On Linux,C/C++,Asterisk Technology
Gandhinagar - Gujarat
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Re: [asterisk-users] Meetme Time Limit?
On Thu, Apr 21, 2011 at 4:03 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.comwrote: Hi, You can use Meetme(1234,dL(1800)) where 1800 = 6 hours after 6 hours channel is hanf up regards Dhaval On Mon, Apr 18, 2011 at 9:31 PM, Bryant Zimmerman brya...@zktech.comwrote: Is there a way to place a hangup time on a dynamic Meetme conference. I am using Page() with a Meetme conf and I have had a few instances where someone from a wifi voip phone looses ip while doing a page and the page never hangs up. I have to kill it. I need to somehow limit the page so after a worse case 2Min timeout it hangs up. Thanks Bryant -- Dhaval's reply works for when you're running a MeetMe conference directly, which does not help Bryant (the question was phrased a little oddly, which caused the confusion I think) Regarding how to limit how long the Paging call can be, use the TIMEOUT(absolute) function. Here's an AEL example: [paging] exten = _92XX,1,Noop(Making sure the call only lasts 60 seconds or less) same = n,Set(TIMEOUT(absolute)=60); same = n,Page(insert page targets and options) Let me know if that works out for you! Regarding MeetMe time limiting in general, I'd like to add an alternative to Dhaval's solution, just to get it back out there in the intertubes so people can find it in the future. As of Asterisk 1.6 you can schedule RealTime MeetMe conferences. I've attached a structure dump of a table called conferences, just direct your extconfig.conf to use it for meetme, set schedule=yes in meetme.conf, and then set the start and end times in the table when creating a scheduled conference. Cheers all! Sherwood McGowan Coming soonSamuPBX scheduled_conferences.sql Description: Binary data -- _ -- 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] VoiceMail to text mail
On Wed, Apr 20, 2011 at 4:10 PM, Mark Deneen mden...@gmail.com wrote: On Wed, Apr 20, 2011 at 4:35 PM, satish patel satish...@hotmail.com wrote: Hey Thanks for that reply after add following option it works but the text output is totally different.. what its totally different is this dictionary problem ? -hmm /var/lib/asterisk/communicator -samprate 8000 In audio file its just: Hello satish this is test message 0: i started is it see no oil you did to less this tonight How many years have you spoken gibberish without knowing? Seriously, though, do you have a bit of an accent (compared to the pocketsphinx developers)? That's most likely the issue, I've seen weird stuff come out of STT apps that were running a British english dictionary and an American was speaking. Additionally, even the difference between accents in America (like between, for instance Maine/Vermont and Ohio, or even Boston and New York) can cause errors...Hell, I even have trouble sometimes deciphering what some people are saying here, and I've lived in almost every major accent area of the US ;-) -- _ -- 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 Export Fax from Wave file
On Thursday 21 Apr 2011, Khaled W. Chehab wrote: Dears, I configured an account on my asterisk pbx to record the outgoing calls. When the asterisk pbx user make a call and send a fax the call recorded to wave file format. I searched the internet and found a software that can play the recorded wave file and export from it the tiff fax document sent. Is there a way that asterisk can play the wav file and export the tiff document ??? You *might* be able to recover the document, *if and only if* the recording quality is high enough. Easiest way to try it is to call up a fax machine (either an actual real one, or a copy of Hylafax) from Asterisk and play the wav file down the line to it. Hylafax might even be able to do this directly; so if you don't have a real fax machine, start looking there. -- 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] Asterisk Export Fax from Wave file
On 21 Apr 2011, at 13:46, A J Stiles wrote: You *might* be able to recover the document, *if and only if* the recording quality is high enough. Easiest way to try it is to call up a fax machine (either an actual real one, or a copy of Hylafax) from Asterisk and play the wav file down the line to it. But fax requires two-way negotiation right?.. S -- _ -- 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] allowguest=yes, how?
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Paul van der Vlis
Sent: Thursday, April 21, 2011 3:09 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] allowguest=yes, how?
Op 20-04-11 21:47, Danny Nicholas schreef:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Paul van der Vlis
Sent: Wednesday, April 20, 2011 2:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] allowguest=yes, how?
Hello,
I want that people from other servers like ekiga.net can make calls to
my users. When I do an allowguest=no then people from other domains
cannot call me. So I think I need allowguest=yes.
cut setup
Is this a good setup?
[Danny Nicholas]
Talking a little out of school, but the safest method is going to be
to
set up one context with allowguest=yes that has ABSOLUTELY NO DIALING
PRIVILEDGES, otherwise you will open Pandora's box.
Maybe you mean something like this?
extensions.conf:
-
[default]
include = users
[dialout]
include = users
exten=_0.,1,Dial(SIP/trunk/0${EXTEN:1},30,tT)
[users]
exten=6001,1,Dial(SIP/paul,20)
exten=6002,1,Dial(SIP/ann,20)
(...)
sip.conf:
---
[general]
context=default
allowguest=no
(...)
[guests]
context=default
allowguest=yes
[trunk]
context=dialout
(...)
[phone-paul]
context=dialout
(...)
[phone-ann]
context=dialout
(...)
---
Thanks for your help!
With regards,
Paul van der Vlis.
--
http://www.vandervlis.nl/
--
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[Danny Nicholas]
I think that's the general idea - perhaps a higher food chain poster can
offer better insight.
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Re: [asterisk-users] Asterisk Export Fax from Wave file
On 04/21/2011 08:12 PM, Khaled W. Chehab wrote: Dears, I configured an account on my asterisk pbx to record the outgoing calls. When the asterisk pbx user make a call and send a fax the call recorded to wave file format. I searched the internet and found a software that can play the recorded wave file and export from it the tiff fax document sent. Is there a way that asterisk can play the wav file and export the tiff document ??? If you have found software to do this, what are you looking for now? Steve -- _ -- 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] Transcode ulaw/g722 problem
We are getting the following on 1.8.3 and 1.8.4-rc2, HELP! Why is Asterisk unable to transcode to/from ulaw and g722? [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:35] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:35] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) -- _ -- 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] Transcode ulaw/g722 problem
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Thursday, April 21, 2011 8:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Transcode ulaw/g722 problem We are getting the following on 1.8.3 and 1.8.4-rc2, HELP! Why is Asterisk unable to transcode to/from ulaw and g722? [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:35] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:35] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) -- _ -- 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 [Danny Nicholas] Because you didn't ask nicely? -- _ -- 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] Transcode ulaw/g722 problem
Might be both end codace are different. That's why this message is come. On Thu, Apr 21, 2011 at 7:26 PM, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Thursday, April 21, 2011 8:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Transcode ulaw/g722 problem We are getting the following on 1.8.3 and 1.8.4-rc2, HELP! Why is Asterisk unable to transcode to/from ulaw and g722? [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:35] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:35] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) -- _ -- 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 [Danny Nicholas] Because you didn't ask nicely? -- _ -- 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 -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk 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
[asterisk-users] IAX2 codec selection and video
Hi, Can anyone let me know how I can enable video (h.263) on SIP, but if a video call is passed over IAX, it will remove the video and pass the audio only. What I tried was: SIP - videosupport=yes - disallow=all - allow=alaw - allow=h263 IAX - disallow=all - allow=alaw What appears to occur is that the SIP call negotiates h263 video, and when passed over IAX, the h263 frames are passed, and are also accepted at the far end which also does not have a video codec allowed. Should that be happening? This is with 1.6.2.18-rc1. Am I missing a setting somewhere? Thanks, Steve -- _ -- 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] missed call notification
Hi,
I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call
notification but i am having issue. following is my dialplan
[macro-stdexten]
exten = s,1,Dial(${ARG2})
exten = s,2,Goto(s-${DIALSTATUS},1); Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send
to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #,
return to start
exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to
voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press
#, return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press
*, send the user into VoicemailMain
exten = h,1,System(/var/lib/asterisk/agi-bin/processcallemail.sh ${ARG3}
${CALLERID(num)} ${CALLERID(name)} ${DIALSTATUS} ${VMSTATUS} ${EXTEN})
[from-sip]
exten = _7[0123]XX,1,macro(stdexten,${EXTEN},sip/${EXTEN})
Following CLI output look like its not executing h extension in macro-stdexten.
But if i add h extension in [from-sip] it works! do you know why ?
-- Executing [7207@from-sip:1] Macro(SIP/7101-000a,
stdexten,7207,sip/7207) in new stack
-- Executing [s@macro-stdexten:1] Dial(SIP/7101-000a, sip/7207) in
new stack
== Using SIP RTP CoS mark 5
-- Called 7207
-- SIP/7207-000b is ringing
== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/7101-000a' in macro 'stdexten'
== Spawn extension (from-sip, 7207, 1) exited non-zero on 'SIP/7101-000a'
-- Executing [h@from-sip:1] Hangup(SIP/7101-000a, ) in new stack
== Spawn extension (from-sip, h, 1) exited non-zero on 'SIP/7101-000a'
--
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Re: [asterisk-users] missed call notification
On Thu, Apr 21, 2011 at 12:26 PM, satish patel satish...@hotmail.comwrote:
Hi,
I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed
call notification but i am having issue. following is my dialplan
[macro-stdexten]
exten = s,1,Dial(${ARG2})
exten = s,2,Goto(s-${DIALSTATUS},1); Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable,
send to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #,
return to start
exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to
voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they
press #, return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they
press *, send the user into VoicemailMain
exten = h,1,System(/var/lib/asterisk/agi-bin/processcallemail.sh ${ARG3}
${CALLERID(num)} ${CALLERID(name)} ${DIALSTATUS} ${VMSTATUS}
${EXTEN})
[from-sip]
exten = _7[0123]XX,1,macro(stdexten,${EXTEN},sip/${EXTEN})
Following CLI output look like its not executing h extension in
macro-stdexten. But if i add h extension in [from-sip] it works! do you know
why ?
-- Executing [7207@from-sip:1] Macro(SIP/7101-000a,
stdexten,7207,sip/7207) in new stack
-- Executing [s@macro-stdexten:1] Dial(SIP/7101-000a,
sip/7207) in new stack
== Using SIP RTP CoS mark 5
-- Called 7207
-- SIP/7207-000b is ringing
== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/7101-000a' in macro 'stdexten'
== Spawn extension (from-sip, 7207, 1) exited non-zero on
'SIP/7101-000a'
-- Executing [h@from-sip:1] Hangup(SIP/7101-000a, ) in new
stack
== Spawn extension (from-sip, h, 1) exited non-zero on
'SIP/7101-000a'
... google
http://www.voip-info.org/wiki/view/Asterisk+cmd+Macro
The Useful info was only a few lines from the beginning:
*'h' extension:* *If a macro executes a Dial() and the called party hangs
up, then the control passes to the 'h' extension of the calling context.
However, the 'h' extension is still needed inside the Macro context in case
of a command, application, or extension exiting non-zero - i.e. the user
hangs up in the middle of a Record() - in this case the 'h' extension of the
Macro context is used, not the 'h' extension of the calling context.)
Tilghman, May 2010: So Macro returns upon hangup to execute the h
extension in the original calling context, though even that is conditional,
based upon it having been broken for a long time.*
--
Sherwood McGowan
Telecommunications and VOIP Consultant
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Re: [asterisk-users] missed call notification
I am always googleing before putting anything here.. I was confused that's why
i came across to you guys! Still i am confused :(
-S
Date: Thu, 21 Apr 2011 13:01:52 -0500
From: sherwood.mcgo...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] missed call notification
On Thu, Apr 21, 2011 at 12:26 PM, satish patel satish...@hotmail.com wrote:
Hi,
I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call
notification but i am having issue. following is my dialplan
[macro-stdexten]
exten = s,1,Dial(${ARG2})
exten = s,2,Goto(s-${DIALSTATUS},1); Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send
to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #,
return to start
exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to
voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press
#, return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press
*, send the user into VoicemailMain
exten = h,1,System(/var/lib/asterisk/agi-bin/processcallemail.sh ${ARG3}
${CALLERID(num)} ${CALLERID(name)} ${DIALSTATUS} ${VMSTATUS} ${EXTEN})
[from-sip]
exten = _7[0123]XX,1,macro(stdexten,${EXTEN},sip/${EXTEN})
Following CLI output look like its not executing h extension in macro-stdexten.
But if i add h extension in [from-sip] it works! do you know why ?
-- Executing [7207@from-sip:1] Macro(SIP/7101-000a,
stdexten,7207,sip/7207) in new stack
-- Executing [s@macro-stdexten:1] Dial(SIP/7101-000a, sip/7207) in
new stack
== Using SIP RTP CoS mark 5
-- Called 7207
-- SIP/7207-000b is ringing
== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/7101-000a' in macro 'stdexten'
== Spawn extension (from-sip, 7207, 1) exited non-zero on 'SIP/7101-000a'
-- Executing [h@from-sip:1] Hangup(SIP/7101-000a, ) in new stack
== Spawn extension (from-sip, h, 1) exited non-zero on 'SIP/7101-000a'
... google
http://www.voip-info.org/wiki/view/Asterisk+cmd+Macro
The Useful info was only a few lines from the beginning:
'h' extension: If a macro executes a Dial() and the
called party hangs up, then the control passes to the 'h' extension of
the calling context.
However, the 'h' extension is still needed inside the Macro context
in case of a command, application, or extension exiting non-zero - i.e.
the user hangs up in the middle of a Record() - in this case the 'h' extension
of the Macro context is used, not the 'h' extension of the calling context.)
Tilghman, May 2010: So Macro returns upon hangup to execute
the h extension in the original calling context, though even that is
conditional, based upon it having been broken for a long time.
--
Sherwood McGowan
Telecommunications and VOIP Consultant
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Re: [asterisk-users] [asterisk-user] Can't get hostname on asterisk dialplan by ENV()
Rajnikant,
This surely depends on how you start asterisk. How are you starting
the asterisk process?
-M
On Thu, Apr 21, 2011 at 7:20 AM, RAJNIKANT VANZA rajniva...@gmail.com wrote:
Hi Friend,
Can't get hostname environment variable on asterisk dialplan.
Help me about how to get hostname environment variable on asterisk dialplan.
I have written export HOSTNAME in /root/.bash_profile and when i execute
echo $HOSTNAME then get right hostname but not success through asterisk
dialplan.
Get environment variable path right value through below statement.
exten = XXX,n,NoOp(--- ${ENV(PATH)})
I have tried like this:
exten = XXX,n,Set(CDR(hostname)=${System(echo $HOSTNAME)})
exten = XXX,n,Set(CDR(hostname)=${ENV(HOSTNAME)})
Thanks in advance.
--
Best Regards,
Rajnikant Vanza
Call : +91-9737456583
Software Engineer
---
Working On Linux,C/C++,Asterisk Technology
Gandhinagar - Gujarat
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[asterisk-users] Nationalprefix chan_dahdi option
Asterisk 1.8.4-rc2 (and 1.8.3) DAHDI Version: 2.4.1.2 libpri version: 1.4.12-beta3 We are having a problem with getting the nationalprefix option of chan_dahdi.conf to work. National calls do not have a 1 added to them when nationalprefix=1. The PRI debug shows the call coming in as a National Call, but the dialplan sees the call without a 1. chan_dahdi.conf: snip switchtype=national internationalprefix = 011 nationalprefix = 1 context=pbxmax-incoming-xo-pri group=1 signalling=pri_cpe channel =1-23 snip PRI Debug: 1 1 Protocol Discriminator: Q.931 (8) len=69 1 TEI=0 Call Ref: len= 2 (reference 457/0x1C9) (Sent from originator) 1 Message Type: SETUP (5) 1 [04 03 80 90 a2] 1 Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 User information layer 1: u-Law (34) 1 [18 03 a9 83 85] 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0 1ChanSel: As indicated in following octets 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 1Ext: 1 Channel: 5 Type: CPE] 1 [1c 15 9f 8b 01 00 a1 0f 02 01 01 06 07 2a 86 48 ce 15 00 04 0a 01 00] 1 Facility (len=23, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x0F, 0x02, 0x01, 0x01, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x04, 0x0A, 0x01, 0x00 ] 1 [1e 02 82 83] 1 Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) 1Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] 1 [6c 0c 21 83 32 35 36 34 32 35 37 38 31 34] 1 Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation allowed of network provided number (3) '2564257814' ] 1 [70 0b a1 33 34 37 32 37 33 31 32 31 33] 1 Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '3472731213' ] 1 -- Making new call for cref 457 1 Received message for call 0xb6e7a148 on link 0x89f1060 TEI/SAPI 0/0 1 -- Processing Q.931 Call Setup 1 -- Processing IE 4 (cs0, Bearer Capability) 1 -- Processing IE 24 (cs0, Channel Identification) 1 -- Processing IE 28 (cs0, Facility) 1 -- Processing IE 30 (cs0, Progress Indicator) 1 -- Processing IE 108 (cs0, Calling Party Number) 1 -- Processing IE 112 (cs0, Called Party Number) 1 -- Delayed processing IE 28 (cs0, Facility) 1 ASN.1 dump 1 Context Specific [11 0x0B] 8B Len:1 01 1 00 - ~ 1 Context Specific/C [1 0x01] A1 Len:15 0F 1 Integer(2 0x02) 02 Len:1 01 1 01 - ~ 1 OID(6 0x06) 06 Len:7 07 1 2A 86 48 CE 15 00 04 - *~H 1 Enumerated(10 0x0A) 0A Len:1 01 1 00 - ~ 1 ASN.1 end 1 interpretation Context Specific [11 0x0B] = 0 0x 1 INVOKE Component Context Specific/C [1 0x01] 1 invokeId Integer(2 0x02) = 1 0x0001 1 operationValue OID(6 0x06) = 42.840.10005.0.4 1 operationValue = ROSE_NI2_InformationFollowing 1 unknown Enumerated(10 0x0A) = 0 0x 1 !! ROSE invoke operation not handled! ROSE_NI2_InformationFollowing 1 q931.c:7587 post_handle_q931_message: Call 457 enters state 6 (Call Present). Hold state: Idle Span: 1 Processing event: PRI_EVENT_RING 1 q931.c:4906 q931_call_proceeding: Call 457 enters state 9 (Incoming Call Proceeding). Hold state: Idle 1 1 DL-DATA request 1 Protocol Discriminator: Q.931 (8) len=10 1 TEI=0 Call Ref: len= 2 (reference 457/0x1C9) (Sent to originator) 1 Message Type: CALL PROCEEDING (2) 1 TEI=0 Transmitting N(S)=93, window is open V(A)=93 K=7 1 1 Protocol Discriminator: Q.931 (8) len=10 1 TEI=0 Call Ref: len= 2 (reference 457/0x1C9) (Sent to originator) 1 Message Type: CALL PROCEEDING (2) 1 [18 03 a9 83 85] 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0 1ChanSel: As indicated in following octets 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 1Ext: 1 Channel: 5 Type: CPE] -- Accepting call from '12564257814' to '3472731213' on channel 0/5, span 1 -- Executing [3472731213@pbxmax-incoming-xo-pri:1] Goto(DAHDI/i1/12564257814-57, 13472731213,1) in new stack -- _ -- 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-user] Can't get hostname on asterisk dialplan by ENV()
On Thu, 21 Apr 2011, RAJNIKANT VANZA wrote:
Can't get hostname environment variable on asterisk dialplan.
1) Is HOSTNAME in the Asterisk process's environment? What does executing:
tr '\000' '\n' /proc/$(cat /var/run/asterisk.pid)/environ
show on the shell console?
2) What does executing:
exten = *,n,verbose(1,${ENV(HOSTNAME)})
show on the Asterisk console?
I start my Asterisk with a minimal environment using the following
snippet:
nice --adjustment=-20\
env --ignore-environment\
HOSTNAME=${HOSTNAME}\
PATH=${PATH}\
$ASTERISK $START_OPTIONS
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] [asterisk-user] Can't get hostname on asterisk dialplan by ENV()
On Thu, Apr 21, 2011 at 3:23 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 21 Apr 2011, RAJNIKANT VANZA wrote: Can't get hostname environment variable on asterisk dialplan. 1) Is HOSTNAME in the Asterisk process's environment? What does executing: tr '\000' '\n' /proc/$(cat /var/run/asterisk.pid)/environ This is /var/run/asterisk/asterisk.pid on my system. I use runit to manage the asterisk process, and the chpst program allows fine control over environment and other limits. -M -- _ -- 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-user] Can't get hostname on asterisk dialplan by ENV()
On Thu, 21 Apr 2011, Mark Deneen wrote: I use runit to manage the asterisk process, and the chpst program allows fine control over environment and other limits. runit is intended to be a sysvinit (/sbin/init) replacement and is not installed (by default) on CentOS or Ubuntu distributions. Can chpst be used by itself? It seems a useful program except that you need to explicitly name each environment variable you want 'ignored' and it is part of a larger package that may have far reaching implications -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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-user] Can't get hostname on asterisk dialplan by ENV()
On Thu, Apr 21, 2011 at 4:30 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 21 Apr 2011, Mark Deneen wrote: I use runit to manage the asterisk process, and the chpst program allows fine control over environment and other limits. runit is intended to be a sysvinit (/sbin/init) replacement and is not installed (by default) on CentOS or Ubuntu distributions. Can chpst be used by itself? It seems a useful program except that you need to explicitly name each environment variable you want 'ignored' and it is part of a larger package that may have far reaching implications Steve, runit is actually very unobtrusive. It is capable to replacing init, but I don't think many people actually use it that way. http://smarden.org/runit/useinit.html documents how to use it with init. If I wanted to clear the environment first, I'd just use env and have that call chpst. I like runit because it manages the process without the typical pid-file tracking that most init scripts use. If the process dies, for whatever reason, it is automatically restarted. stdout is captured and redirected to an optional log process which can roll logs, removing the need for logrotate and figuring out what special signal to send the process to tell it that you've truncated the log file. There is a catch, though. Your process has to run in the foreground, and runsv keeps it in the background. So, for programs which auto-detach and background themselves, you have to run them with a switch that says not to run as a daemon. It's not everyone's cup of tea, but I find it to be perfect for my needs, and a very well written utility. -M -- _ -- 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] AST-2011-005: File Descriptor Resource Exhaustion
Asterisk Project Security Advisory - AST-2011-005 Product Asterisk Summary File Descriptor Resource Exhaustion Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated TCP Based Sessions (TCP SIP, Skinny, Asterisk Manager Interface, and HTTP sessions) Severity Moderate Exploits Known Yes Reported On March 18, 2011 Reported By Tzafrir Cohen tzafrir.cohen AT xorcom DOT com Posted On April 21, 2011 Last Updated On April 21, 2011 Advisory Contact Matthew Nicholson mnichol...@digium.com CVE Name CVE-2011-1507 Description On systems that have the Asterisk Manager Interface, Skinny, SIP over TCP, or the built in HTTP server enabled, it is possible for an attacker to open as many connections to asterisk as he wishes. This will cause Asterisk to run out of available file descriptors and stop processing any new calls. Additionally, disk space can be exhausted as Asterisk logs failures to open new file descriptors. Resolution Asterisk can now limit the number of unauthenticated connections to each vulnerable interface and can also limit the time unauthenticated clients will remain connected for some interfaces. This will prevent vulnerable interfaces from using up all available file descriptors. Care should be taken when setting the connection limits so that the combined total of allowed unauthenticated sessions from each service is not more than the file descriptor limit for the Asterisk process. The file descriptor limit can be checked (and set) using the ulimit -n command for the process' limit and the /proc/sys/fs/file-max file (on Linux) for the system's limit. It will still be possible for an attacker to deny service to each of the vulnerable services individually. To mitigate this risk, vulnerable services should be run behind a firewall that can detect and prevent DoS attacks. In addition to using a firewall to filter traffic, vulnerable systems can be protected by disabling the vulnerable services in their respective configuration files. Affected Versions Product Release Series Asterisk Open Source1.4.x All versions Asterisk Open Source 1.6.1.x All versions Asterisk Open Source 1.6.2.x All versions Asterisk Open Source1.8.x All versions Asterisk Business Edition C.x.x All versions Corrected In Product Release Asterisk Open Source1.4.40.1, 1.6.1.25, 1.6.2.17.3, 1.8.3.3 Asterisk Business Edition C.3.6.4 Patches URL Branch http://downloads.asterisk.org/pub/security/AST-2011-005-1.4.diff1.4 http://downloads.asterisk.org/pub/security/AST-2011-005-1.6.1.diff 1.6.1 http://downloads.asterisk.org/pub/security/AST-2011-005-1.6.2.diff 1.6.2 http://downloads.asterisk.org/pub/security/AST-2011-005-1.8.diff1.8 Asterisk Project Security Advisories are posted at http://www.asterisk.org/security This document may be superseded by later versions; if so, the latest version will be posted at http://downloads.digium.com/pub/security/AST-2011-005.pdf and
[asterisk-users] AST-2011-006: Asterisk Manager User Shell Access
Asterisk Project Security Advisory - AST-2011-006 ProductAsterisk SummaryAsterisk Manager User Shell Access Nature of Advisory Permission Escalation SusceptibilityRemote Authenticated Sessions Severity Minor Exploits KnownYes Reported On February 10, 2011 Reported By Mark Murawski markm AT intellasoft DOT net Posted On April 21, 2011 Last Updated OnApril 21, 2011 Advisory Contact Matthew Nicholson mnichol...@digium.com CVE Name Description It is possible for a user of the Asterisk Manager Interface to bypass a security check and execute shell commands when they should not have that ability. Sending the Async header with the Application header during an Originate action, allows authenticated manager users to execute shell commands. Only users with the system privilege should be able to do this. Resolution Asterisk now performs the proper access check where appropriate during the originate manager action. Affected Versions Product Release Series Asterisk Open Source1.4.x All versions Asterisk Open Source 1.6.1.x All versions Asterisk Open Source 1.6.2.x All versions Asterisk Open Source1.8.x All versions Asterisk Business Edition C.x.x All versions Corrected In Product Release Asterisk Open Source1.4.40.1, 1.6.1.25, 1.6.2.17.3, 1.8.3.3 Asterisk Business Edition C.3.6.4 Patches URL Branch http://downloads.asterisk.org/pub/security/AST-2011-006-1.4.diff1.4 http://downloads.asterisk.org/pub/security/AST-2011-006-1.6.1.diff 1.6.1 http://downloads.asterisk.org/pub/security/AST-2011-006-1.6.2.diff 1.6.2 http://downloads.asterisk.org/pub/security/AST-2011-006-1.8.diff1.8 Links Asterisk Project Security Advisories are posted at http://www.asterisk.org/security This document may be superseded by later versions; if so, the latest version will be posted at http://downloads.digium.com/pub/security/AST-2011-006.pdf and http://downloads.digium.com/pub/security/AST-2011-006.html Revision History Date Editor Revisions Made 4/21/11Matthew NicholsonInitial version Asterisk Project Security Advisory - AST-2011-006 Copyright (c) 2011 Digium, Inc. All Rights Reserved. Permission is hereby granted to distribute and publish this advisory in its original, unaltered form. -- _ -- 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 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 Now Available (Security Releases)
The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 resolve two issues: * File Descriptor Resource Exhaustion (AST-2011-005) * Asterisk Manager User Shell Access (AST-2011-006) The issues and resolutions are described in the AST-2011-005 and AST-2011-006 security advisories. For more information about the details of these vulnerabilities, please read the security advisories AST-2011-005 and AST-2011-006, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.40.1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.25 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.3 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.3 Security advisory AST-2011-005 and AST-2011-006 are available at: http://downloads.asterisk.org/pub/security/AST-2011-005.pdf http://downloads.asterisk.org/pub/security/AST-2011-006.pdf 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] Help Required---Problem in Installation without dahdi
Hi, Installation of dahdi requires kernel source that is not available with my remote virtual machine. Therefore I installed Asterisk without installing dahdi but when I start Asterisk it crashes while loading chan_agent.so (noload is also not useful in this case). Any suggestions or hints to overcome this issue? Regards -- _ -- 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
