Re: [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny
On Mon, 30 Aug 2010, J. Oquendo wrote: Gordon Henderson wrote: On Mon, 30 Aug 2010, J. Oquendo wrote: I also posted a very effective iptables script some weeks ago if you care to search the archives. It works and is extremely effective in blocking these types of attacks - however, it will not stop a broken sipvicious from continuing to send data to your server, and that's the issue I have at present. Alright, so I'm slightly confused maybe I'm reading this wrong... Someone using an older version of sipvicious was blocked and the blocking of the traffic still carried a load? Yes. It's UDP, they just keep on sending. If so then you should have logged into your router and simply sinkholed him. There is nothing you can do against a flood whether or not its sipvicious or any other program. It's the golf ball through the water hose effect. Did you try: 1) sinkholing from your router Yes. works fine until they can send faster than the router/incoming line can handle the load. With a good VPS host you can trivially max-out a typical UK ADSL line. 2) Contacting your upstream to inform them of the DoS to see if they'd sinkhole it Yes. My (ADSL) upstream will not block inbound floods like this. They have a financial incentive not to - they get paid for the data the allow into their network and through to you. I only know of one UK broadband ISP that will actively block inbound traffic for you and they're technically superb, but that comes with a price which is more than your average small business is wiling to pay. None of the others I know and have used will block an inbound flood of anything for you. My main hosting upstream will only block such attacks when it has a detrimental effect on their network (and then they're very good at it) - last time my hosted servers got hit, they soaked up just over 30GB from a single VPS site in France in a 12-hour period. 3) Contact the UPSTREAM of the attacking host? Yes. No reply. And in the few times I've tried, I've only ever had a reply from Amazon - some 18 hours after the flood started and then it took another 12 hours for them to stop it (well documented here in the archives by myself and others) The reality is that most bulk VPS providers just don't care, or you've got to go through layes of their own (semi-automated) protocol to get anywhere (cf. Amazon) Basically if you have to pay for inbound traffic in any shape or form (monthly cap, daily limit, etc.) then you're fucked when this happens. That's why the author of Sipvicious added svcrash.py to his set of scripts. Gordon -- _ -- 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] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny
On Tue, Aug 31, 2010 at 8:30 AM, Gordon Henderson gordon+aster...@drogon.net wrote: 3) Contact the UPSTREAM of the attacking host? Yes. No reply. And in the few times I've tried, I've only ever had a reply from Amazon - some 18 hours after the flood started and then it took another 12 hours for them to stop it (well documented here in the archives by myself and others) Amazon did something about it? I don't remember seeing that, Gordon, it's a new record. The average response has been zero. /r -- _ -- 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] No audio on call forward after upgrade from Asterisk 1.4 to 1.6
Hi everyone, This is my first post to the list, although I am a long term user of Asterisk. I have recently found a problem that I just can't seem to solve. I have a client that has an Ubuntu x64 based Asterisk server with and ISDN Dahdi interface and about 25 SIP handsets. Everything was working fine in Asterisk 1.4 and now after migrating the config to Asterisk 1.6.2.5 I have one single issue that I can't explain. I have an extension that if you call it, it will play a sound file and hangup. Pretty simple stuff. Below is the extensions.conf entry for this extension. exten = 849,1,Playback(custom/ceh-meetingmsg) exten = 849,n,Hangup The following happens if I dial it from a SIP handset == Using SIP RTP CoS mark 5 -- Executing [...@smallanimals:1] Playback(SIP/812-0074, custom/ceh-meetingmsg) in new stack -- SIP/812-0074 Playing 'custom/ceh-meetingmsg.gsm' (language 'en') -- Executing [...@smallanimals:2] Hangup(SIP/812-0074, ) in new stack == Spawn extension (smallanimals, 849, 2) exited non-zero on 'SIP/812-0074' The scenario is during the day, if my client has a staff meeting, they simply turn on call forwarding on the reception phone to this extension. In the past, the audio would start as soon as the caller dials in. After upgrading to Asterisk 1.6, we simply get no audio until the dialplan finishes. On the Asterisk console, I can see that the sound file is indeed playing, but we can't hear it. This happens if I am dialing the from a SIP extension on the phone system, or if I dial in from the public phone system. == Using SIP RTP CoS mark 5 -- Executing [...@smallanimals:1] Dial(SIP/811-0046, SIP/812,60) in new stack == Using SIP RTP CoS mark 5 -- Called 812 -- Got SIP response 302 Moved Temporarily back from 192.168.1.148 -- Now forwarding SIP/811-0046 to 'Local/8...@smallanimals' (thanks to SIP/812-0047) -- Executing [...@smallanimals:1] Playback(Local/8...@smallanimals-b5dd;2, custom/ceh-meetingmsg) in new stack -- Local/8...@smallanimals-b5dd;2 Playing 'custom/ceh-meetingmsg.gsm' (language 'en') I have tried so many things that I have lost count, and I humbly ask the collective intelligence of the Asterisk community for assistance. Many thanks aF -- _ -- 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] No audio on call forward after upgrade from Asterisk 1.4 to 1.6
Hi! After upgrading to Asterisk 1.6, we simply get no audio until the dialplan finishes. On the Asterisk console, I can see that the sound file is indeed playing, but we can't hear it. [...] I have tried so many things that I have lost count, and I humbly ask the collective intelligence of the Asterisk community for assistance. For a start: * upgarde to the current release of 1.6.2.x * does that message play when you call it without a forward (302) on your admin phone? * convert the .gsm prompt to a .wav or .alaw or .ulaw prompt and see if that improves matters * do a RTP debug to see if there is any RTP being sent at all * consider ChanSpy for listening in (although I doubt that'll help you) Philipp -- _ -- 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] No audio on call forward after upgrade from Asterisk 1.4 to 1.6
Hi Alex, I'm new to this list, but I had this problem too, and I solved it looking at the codecs the sip handsets use, and then I converted the voice prompts to that codec just like Philipp said.. Ondrej On Tue, Aug 31, 2010 at 10:04 AM, Alex Ferrara a...@receptiveit.com.auwrote: Hi everyone, This is my first post to the list, although I am a long term user of Asterisk. I have recently found a problem that I just can't seem to solve. I have a client that has an Ubuntu x64 based Asterisk server with and ISDN Dahdi interface and about 25 SIP handsets. Everything was working fine in Asterisk 1.4 and now after migrating the config to Asterisk 1.6.2.5 I have one single issue that I can't explain. I have an extension that if you call it, it will play a sound file and hangup. Pretty simple stuff. Below is the extensions.conf entry for this extension. exten = 849,1,Playback(custom/ceh-meetingmsg) exten = 849,n,Hangup The following happens if I dial it from a SIP handset == Using SIP RTP CoS mark 5 -- Executing [...@smallanimals:1] Playback(SIP/812-0074, custom/ceh-meetingmsg) in new stack -- SIP/812-0074 Playing 'custom/ceh-meetingmsg.gsm' (language 'en') -- Executing [...@smallanimals:2] Hangup(SIP/812-0074, ) in new stack == Spawn extension (smallanimals, 849, 2) exited non-zero on 'SIP/812-0074' The scenario is during the day, if my client has a staff meeting, they simply turn on call forwarding on the reception phone to this extension. In the past, the audio would start as soon as the caller dials in. After upgrading to Asterisk 1.6, we simply get no audio until the dialplan finishes. On the Asterisk console, I can see that the sound file is indeed playing, but we can't hear it. This happens if I am dialing the from a SIP extension on the phone system, or if I dial in from the public phone system. == Using SIP RTP CoS mark 5 -- Executing [...@smallanimals:1] Dial(SIP/811-0046, SIP/812,60) in new stack == Using SIP RTP CoS mark 5 -- Called 812 -- Got SIP response 302 Moved Temporarily back from 192.168.1.148 -- Now forwarding SIP/811-0046 to 'Local/8...@smallanimals' (thanks to SIP/812-0047) -- Executing [...@smallanimals:1] Playback(Local/8...@smallanimals-b5dd;2, custom/ceh-meetingmsg) in new stack -- Local/8...@smallanimals-b5dd;2 Playing 'custom/ceh-meetingmsg.gsm' (language 'en') I have tried so many things that I have lost count, and I humbly ask the collective intelligence of the Asterisk community for assistance. Many thanks aF -- _ -- 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 -- -- Ondrej Škopek -- _ -- 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 core dump
Hi all, my asterisk will coredump in runing about ten days one time, and the following is bt infor: #0 0x00aac410 in __kernel_vsyscall () (gdb) bt #0 0x00aac410 in __kernel_vsyscall () #1 0x00bead80 in raise () from /lib/libc.so.6 #2 0x00bec691 in abort () from /lib/libc.so.6 #3 0x00c2324b in __libc_message () from /lib/libc.so.6 #4 0x00c2b883 in _int_malloc () from /lib/libc.so.6 #5 0x00c2d0be in calloc () from /lib/libc.so.6 #6 0x00215828 in statechange_queue (dev=0xb6e05a1c SIP/jcc, state=2, ign=0x0) at /root/asterisk-2010/include/asterisk/utils.h:360 #7 0x08096bd4 in do_state_change (device=0xb6e05a1c SIP/jcc) at devicestate.c:291 #8 0x08096c89 in do_devstate_changes (data=0x0) at devicestate.c:346 #9 0x080fe7ab in dummy_start (data=0x93f36a8) at utils.c:895 #10 0x00d1349b in start_thread () from /lib/libpthread.so.0 #11 0x00c9342e in clone () from /lib/libc.so.6 any help will be appreciated. -- Best regards! jordan pan Location:Shenzhen China Company:www.justcall.cn -- _ -- 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] billsec exceeds duration on some calls
On 8/20/10 1:24 PM, A J Stiles wrote: On Wednesday 11 Aug 2010, Tilghman Lesher wrote: On Wednesday 11 August 2010 03:59:28 A J Stiles wrote: I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr addon. With some calls, the value in the `billsec` field in the CDR is exceeding the value in the `duration` field. I'd love to know what circumstance caused that. I agree that this should not occur. I've done some more digging about. I was getting calls in the monitor folder where the outgoing and incoming halves were different lengths; so I temporarily disabled removing them after combining them into a single file, and let them build up for a few days. There doesn't seem to be any correlation between this phenomenon and billsec being duration, though. Can anyone else with a similar setup try running a query such as SELECT COUNT(*) FROM cdr WHERE calldate LIKE 2010-08-20% AND billsecduration ; and seeing if they have any calls like this? Any chance this has something to do with your system time? Are you running ntpd, or setting time at regular intervals via a central system clock and a cron job? Again... also just stabbing in the dark. N. -- _ -- 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] Maximum Wait Time queue option
Take a look here: http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue Queue(queuename[|options][|URL][|announceoverride][|*timeout*][|AGI]) Hope it helps! 2010/8/30 Tino t...@sparksupport.com Hello, Is there any option to set the maximum number of seconds a caller can wait in a queue before being pulled out ? 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 -- Saludos Danny Dias -- _ -- 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] No audio on call forward after upgrade from Asterisk 1.4 to 1.6
On Tue, Aug 31, 2010 at 4:04 AM, Alex Ferrara a...@receptiveit.com.au wrote: exten = 849,1,Playback(custom/ceh-meetingmsg) exten = 849,n,Hangup exten = 849,1,Progress() exten = 849,n,Playback(custom/ceh-meetingmsg) exten = 849,n,Hangup -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 core dump
On Tue, Aug 31, 2010 at 5:57 AM, jordan pan pylon...@gmail.com wrote: my asterisk will coredump in runing about ten days one time, and the following is bt infor: Open an issue on https://issues.asterisk.org, besure to follow doc/backtrace.txt and post all relevant information. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] help with dialplan
Here's the updated debug log. http:/www.computerworkx.net/client/Document.txt On 8/30/2010 2:55 PM, Paul Belanger wrote: On Mon, Aug 30, 2010 at 2:48 PM, Todd Reesetrees...@gmail.com wrote: Thanks for pointing out the misspelling. I've corrected that and still no luck. Create a new debug log with your recent changes, re-attach it the list. -- _ -- 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] help with dialplan
On Tue, Aug 31, 2010 at 9:16 AM, Todd Reese trees...@gmail.com wrote: Here's the updated debug log. [Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from '150' to extension '6789542133' rejected because extension not found in context 'remote'. So, again, a context problem. You can confirm by entering: *CLI dialplan show 6789542...@remote -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] help with dialplan
asterisk*CLI dialplan show 6789542...@remote There is no existence of 'remote' context Command 'dialplan show 6789542...@remote' failed. asterisk*CLI On 8/31/2010 9:58 AM, Paul Belanger wrote: dialplan show 6789542...@remote -- _ -- 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] help with dialplan
From extensions.conf
[remote]
include = from-internal
include = dialout1
include = dialout2
include = dialout3
include = intercom
exten = 150,1,Macro(oneline,${EXTERNPHONE0})
[dialout1]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[dialout2]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[dialout3]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6783747049/${EXTEN},40,Ttr)
On 8/31/2010 9:58 AM, Paul Belanger wrote:
On Tue, Aug 31, 2010 at 9:16 AM, Todd Reesetrees...@gmail.com wrote:
Here's the updated debug log.
[Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from '150' to
extension '6789542133' rejected because extension not found in context
'remote'.
So, again, a context problem. You can confirm by entering:
*CLI dialplan show 6789542...@remote
--
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Re: [asterisk-users] help with dialplan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese
Subject: Re: [asterisk-users] help with dialplan
asterisk*CLI dialplan show 6789542...@remote
There is no existence of 'remote' context
Command 'dialplan show 6789542...@remote' failed.
asterisk*CLI
On 8/31/2010 9:58 AM, Paul Belanger wrote:
dialplan show 6789542...@remote
Ok. I'm a late joiner to this thread. Reading the original post I see
that you are trying to do an external SIP dial to 678-954-2133. These
questions:
1. Does the dial string need to be 1xxx instead of xxx (presumably local 10
digit dialing)? If yes, change
exten = _NXXNXX,n,Dial(SIP/v6781234567/${EXTEN},40,Ttr)
to
exten = _NXXNXX,n,Dial(SIP/v6781234567/1${EXTEN},40,Ttr)
2. voipdialACA and v6781234567 are registered trunks with credentials?
Hope this helps.
--
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Re: [asterisk-users] help with dialplan
Todd--
There is probably some nifty anti-infinite-recursion code in the
extensions.conf parser,
to keep asterisk from going into infinite loops trying to descend into the
right context.
In your dialplan, [remote] includes dialout1, dialout2, dialout3, and each
of those
include remote.
Straighten out that mess and maybe things might work. Just a guess, but
worth a try!
murf
On Tue, Aug 31, 2010 at 8:25 AM, Todd Reese trees...@gmail.com wrote:
From extensions.conf
[remote]
include = from-internal
include = dialout1
include = dialout2
include = dialout3
include = intercom
exten = 150,1,Macro(oneline,${EXTERNPHONE0})
[dialout1]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[dialout2]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[dialout3]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6783747049/${EXTEN},40,Ttr)
On 8/31/2010 9:58 AM, Paul Belanger wrote:
On Tue, Aug 31, 2010 at 9:16 AM, Todd Reesetrees...@gmail.com wrote:
Here's the updated debug log.
[Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from '150' to
extension '6789542133' rejected because extension not found in context
'remote'.
So, again, a context problem. You can confirm by entering:
*CLI dialplan show 6789542...@remote
--
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--
Steve Murphy
ParseTree Corp
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[asterisk-users] Running System() after call completion, not in 'h'?
Greetings all-
I have some dialplan code on an Asterisk 1.2.x box that basically dials a call,
then after call completion, runs a command via System(). However, I'm finding
that roughly 5% of the time, the System() command never executes and seems to
be on specific destinations. Simplified/paraphrased example:
exten = 1,1,Set(VARIABLE=SOMEVALUE)
exten = 1,n,Dial(SIP/somepeer/1234567980)
exten = 1,n,System(/bin/bash /root/bin/somescript.sh ${VARIABLE}
It almost seems to be related to how fast the destination 'hangs up' and
whether or not the dialplan has time to run the System command before jumping
to 'h'.
I have to believe there is a better way to do this, possibly using DeadAGI?
All suggestions welcome. Thank you!
--Tim
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Re: [asterisk-users] help with dialplan
I had already check on this. Thanks for the info, though.
On 8/31/2010 10:36 AM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese
Subject: Re: [asterisk-users] help with dialplan
asterisk*CLI dialplan show 6789542...@remote
There is no existence of 'remote' context
Command 'dialplan show 6789542...@remote' failed.
asterisk*CLI
On 8/31/2010 9:58 AM, Paul Belanger wrote:
dialplan show 6789542...@remote
Ok. I'm a late joiner to this thread. Reading the original post I see
that you are trying to do an external SIP dial to 678-954-2133. These
questions:
1. Does the dial string need to be 1xxx instead of xxx (presumably local 10
digit dialing)? If yes, change
exten = _NXXNXX,n,Dial(SIP/v6781234567/${EXTEN},40,Ttr)
to
exten = _NXXNXX,n,Dial(SIP/v6781234567/1${EXTEN},40,Ttr)
2. voipdialACA and v6781234567 are registered trunks with credentials?
Hope this helps.
--
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Re: [asterisk-users] Running System() after call completion, not in 'h'?
On Tue, 31 Aug 2010, Tim Nelson wrote:
I have some dialplan code on an Asterisk 1.2.x box that basically dials
a call, then after call completion, runs a command via System().
However, I'm finding that roughly 5% of the time, the System() command
never executes and seems to be on specific destinations.
Simplified/paraphrased example:
exten = 1,1,Set(VARIABLE=SOMEVALUE)
exten = 1,n,Dial(SIP/somepeer/1234567980)
exten = 1,n,System(/bin/bash /root/bin/somescript.sh ${VARIABLE}
It almost seems to be related to how fast the destination 'hangs up' and
whether or not the dialplan has time to run the System command before
jumping to 'h'.
I have to believe there is a better way to do this, possibly using
DeadAGI?
I think you're on the right track -- moving from system() to [dead]agi().
I think you should be using agi() instead of deadagi() since the call
isn't dead yet. IMO, trapping HUP is a prerequisite for a well written
AGI.
--
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] Running System() after call completion, not in 'h'?
- Steve Edwards asterisk@sedwards.com wrote:
On Tue, 31 Aug 2010, Tim Nelson wrote:
I have some dialplan code on an Asterisk 1.2.x box that basically
dials
a call, then after call completion, runs a command via System().
However, I'm finding that roughly 5% of the time, the System()
command
never executes and seems to be on specific destinations.
Simplified/paraphrased example:
exten = 1,1,Set(VARIABLE=SOMEVALUE)
exten = 1,n,Dial(SIP/somepeer/1234567980)
exten = 1,n,System(/bin/bash /root/bin/somescript.sh ${VARIABLE}
It almost seems to be related to how fast the destination 'hangs up'
and
whether or not the dialplan has time to run the System command
before
jumping to 'h'.
I have to believe there is a better way to do this, possibly using
DeadAGI?
I think you're on the right track -- moving from system() to
[dead]agi().
I think you should be using agi() instead of deadagi() since the call
isn't dead yet. IMO, trapping HUP is a prerequisite for a well
written
AGI.
The problem is that the call is hitting the 'h' extension too quickly. And,
unfortunately, in 'h', the variables I set earlier are already gone so I cannot
simply run the System() from 'h'.
Is there any situation where 'h' would not be called? I could certainly do the
work in DeadAGI() via 'h' knowing it would be called regardless of timing.
--Tim
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Re: [asterisk-users] help with dialplan
Interesting things going on herel.
After your suggestions, Steve. I reran the dialplan show
16789542...@remote command with the below results.
Phone calls are geting the 404 error and the NOTICE on the console.
[Aug 31 11:06:46] NOTICE[11884]: chan_sip.c:20161 handle_request_invite:
Call from '150' to extension '16789542133' rejected because extension
not found in context 'remote'.
asterisk*CLI dialplan show 16789542...@remote
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout2' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout3' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout2' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout3' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr)
[pbx_config]
-= 7 extensions (7 priorities) in 7 contexts. =-
[Aug 31 11:03:38] WARNING[11903]: pbx.c:5741 show_dialplan_helper:
Avoiding circular include of from-internal within remote
On 8/31/2010 10:49 AM, Steve Murphy wrote:
Todd--
There is probably some nifty anti-infinite-recursion code in the
extensions.conf parser,
to keep asterisk from going into infinite loops trying to descend into
the right context.
In your dialplan, [remote] includes dialout1, dialout2, dialout3, and
each of those
include remote.
Straighten out that mess and maybe things might work. Just a guess,
but worth a try!
murf
On Tue, Aug 31, 2010 at 8:25 AM, Todd Reese trees...@gmail.com
mailto:trees...@gmail.com wrote:
From extensions.conf
[remote]
include = from-internal
include = dialout1
include = dialout2
include = dialout3
include = intercom
exten = 150,1,Macro(oneline,${EXTERNPHONE0})
[dialout1]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[dialout2]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[dialout3]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6783747049/${EXTEN},40,Ttr)
On 8/31/2010 9:58 AM, Paul Belanger wrote:
On Tue, Aug 31, 2010 at 9:16 AM, Todd Reesetrees...@gmail.com
mailto:trees...@gmail.com wrote:
Here's the updated debug log.
[Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from '150' to
extension '6789542133' rejected because extension not found in
context
'remote'.
So, again, a context problem. You can confirm by entering:
*CLI dialplan show 6789542...@remote
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[asterisk-users] Yes it is a dimensioning question! Atom CPU
I am looking for pros and cons on the Intel Atom cpu. Has anybody been using these in production? I am looking at an Atom D510 (dual core 1.6GHz, 1M cache) to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high as 7 simultaneous calls), g729 all the way through except voicemail will be wav format for email purposes(requirement). I will be tying 3 of these together to route calls between different offices. What are the gotchas and things to think about that you may have seen in your deployments. Havent decided between latest 1.4 or 1.6 yet, probably 1.6. Thank you for any words of wisdom you may wish to share. JohnM __ This E-Mail was sent with WebMail Client. -- _ -- 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] help with dialplan
On Tue, Aug 31, 2010 at 9:14 AM, Todd Reese trees...@gmail.com wrote:
Interesting things going on herel.
After your suggestions, Steve. I reran the dialplan show
16789542...@remote command with the below results.
Phone calls are geting the 404 error and the NOTICE on the console.
[Aug 31 11:06:46] NOTICE[11884]: chan_sip.c:20161 handle_request_invite:
Call from '150' to extension '16789542133' rejected because extension not
found in context 'remote'.
-- This one is easy to solve, Add the extension 16789542133 to the remote
context and have it do what you need to be done for an incoming call.
and also, see below:
asterisk*CLI dialplan show 16789542...@remote
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout2' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout3' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout2' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout3' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr)
[pbx_config]
-= 7 extensions (7 priorities) in 7 contexts. =-
[Aug 31 11:03:38] WARNING[11903]: pbx.c:5741 show_dialplan_helper: Avoiding
circular include of from-internal within remote
See the Avoiding circular include? Get rid of that by removing one of the
includes that make the cycle; make your
inclusions hierarchical. One context to include them all, and in the
darkness bind them! (sorry, too much Tolkien)
murf
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Re: [asterisk-users] help with dialplan
Why not just copy the _1NXXNXX line into the remote context? -- _ -- 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] Yes it is a dimensioning question! Atom CPU
On 31 Aug 2010, at 16:37, jmilli...@sentinelcommunications.com wrote: I am looking at an Atom D510 (dual core 1.6GHz, 1M cache) to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high as 7 simultaneous calls), g729 all the way through Sounds fine to me. Reckon you could do that on a toaster ;) 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] Yes it is a dimensioning question! Atom CPU
On 31 Aug 2010, at 17:58, jmilli...@sentinelcommunications.com wrote: On 31 Aug 2010, at 16:37, jmilli...@sentinelcommunications.com wrote: I am looking at an Atom D510 (dual core 1.6GHz, 1M cache) to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high as 7 simultaneous calls), g729 all the way through Sounds fine to me. Reckon you could do that on a toaster ;) That is what I was thinking. I have an eeebox at home that does fine with a single core Atom(1 or 2 simultaneous calls) but I do not have any real world testing/experiance for these so I thought I would get some expert opionions. Please don't reply to me directly, reply to the list address (contained within the Reply-To field). Thanks. 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
[asterisk-users] Asterisk with Blockhosts
Just in case anyone is using Blockhosts
(http://www.aczoom.com/blockhosts/) with their Linux servers and
Asterisk here are the rules necessary to block invalid users:
asterisk-NoPeer:
r'Registration from .* failed for \'{HOST_IP}\' - No matching peer
found',
asterisk-NoAuth:
r'Registration from .* failed for \'{HOST_IP}\' - Username/auth name
mismatch',
asterisk-NoPass:
r'Registration from .* failed for \'{HOST_IP}\' - Wrong password',
Just add these rules to your /etc/blockhosts.conf file.
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Re: [asterisk-users] Yes it is a dimensioning question! Atom CPU
On Tue, 31 Aug 2010, jmilli...@sentinelcommunications.com wrote: I am looking for pros and cons on the Intel Atom cpu. Has anybody been using these in production? I am looking at an Atom D510 (dual core 1.6GHz, 1M cache) to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high as 7 simultaneous calls), g729 all the way through except voicemail will be wav format for email purposes(requirement). I will be tying 3 of these together to route calls between different offices. What are the gotcha?s and ?things to think about? that you may have seen in your deployments. Haven?t decided between latest 1.4 or 1.6 yet, probably 1.6. Thank you for any words of wisdom you may wish to share. JohnM I can do that on my mobile phone, so you will have no issues doing that on an Atom. I use Atoms for VoIP/Asterisk servers. I have some with dozens times more than you're trying to achieve. (very highly tuned and not an off the shelft type installation, but that's just me) Get fanless board if you can. 1GB of RAM is fine if you're not running a GUI on it. Gordon -- _ -- 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] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny
On Tue, 31 Aug 2010, Randy R wrote: On Tue, Aug 31, 2010 at 8:30 AM, Gordon Henderson gordon+aster...@drogon.net wrote: 3) Contact the UPSTREAM of the attacking host? Yes. No reply. And in the few times I've tried, I've only ever had a reply from Amazon - some 18 hours after the flood started and then it took another 12 hours for them to stop it (well documented here in the archives by myself and others) Amazon did something about it? I don't remember seeing that, Gordon, it's a new record. The average response has been zero. Well... I don't know if they actually did anything )-: However, after going through their automated submission widget the attack last time did stop... Some 12 hours later. I suspect the poor sod who'd EC2 got hacked ran out of credit or something... Their whole system is designed as a device to waste the time effort of those trying to submit reports, etc. to them. Gordon -- _ -- 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] Yes it is a dimensioning question! Atom CPU
Sounds fine to me. Reckon you could do that on a toaster ;) S Thanks, I needed to clean this keyboard anyway -- _ -- 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 core dump
On Tuesday 31 August 2010 07:49:19 Paul Belanger wrote: On Tue, Aug 31, 2010 at 5:57 AM, jordan pan pylon...@gmail.com wrote: my asterisk will coredump in runing about ten days one time, and the following is bt infor: Open an issue on https://issues.asterisk.org, besure to follow doc/backtrace.txt and post all relevant information. You can save some time and follow doc/valgrind.txt. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) 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] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny
On Tue, Aug 31, 2010 at 7:09 PM, Gordon Henderson gordon+aster...@drogon.net wrote: Their whole system is designed as a device to waste the time effort of those trying to submit reports, etc. to them. This is not the right list for the following comment, but vested interests always ruin life. Ego conflicts, often found in true open-source projects, are far less damaging. Email spammers got away for the longest time with doing that because the providers didn't want to throw out paying customers. Eventually, it because clear that they would be forced to do something. I believe eventually, Amazon will be put in that position. Until then, they have done very little, and we have stopped using their cloud services as much as possible. /r -- _ -- 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] Pickup parcked call from Aastra 9480i ct cordless
Here, we have 2 aastra 9133i and on 9480i ct cordless. We use often the park call feature of asterisk to transfer calls to one another. But the 9480i ct cordless cannot pickup a parked call. When manually entering 701 (parked call extention), the phone display Call failed (appel écoué in french). Nothing is displayed on the asterisk console. When doing it from other aastra phones (same config), or other make phones, it works. Any hints on possible causes ? -- _ -- 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] Pickup parcked call from Aastra 9480i ct cordless
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nicolas Ross Subject: [asterisk-users] Pickup parcked call from Aastra 9480i ct cordless Here, we have 2 aastra 9133i and on 9480i ct cordless. We use often the park call feature of asterisk to transfer calls to one another. But the 9480i ct cordless cannot pickup a parked call. When manually entering 701 (parked call extention), the phone display Call failed (appel écoué in french). Nothing is displayed on the asterisk console. When doing it from other aastra phones (same config), or other make phones, it works. Any hints on possible causes ? Make sure your verbosity is set to at least 5 and try to see the CLI output on failure again. Are you sure the call is parked on 701 (not 702-720 as defined in features.conf)? Other calls/extension dials work from this phone? -- _ -- 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] Pickup parcked call from Aastra 9480i ctcordless
Make sure your verbosity is set to at least 5 and try to see the CLI output on failure again. Are you sure the call is parked on 701 (not 702-720 as defined in features.conf)? Yes, after I can pick it up from my phone (9133i), and it works. I had verbosity at 6 at the moment of testing. When he enters 701, only his phones displays Failed, nothing in asterisk. I can pickup after that on mine. Other calls/extension dials work from this phone? Yes, our extensions are 3 digits, and all other calls to / from this (problematic) phone works. Regads, -- _ -- 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] Pickup parcked call from Aastra 9480ictcordless
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nicolas Ross Subject: Re: [asterisk-users] Pickup parcked call from Aastra 9480ictcordless Make sure your verbosity is set to at least 5 and try to see the CLI output on failure again. Are you sure the call is parked on 701 (not 702-720 as defined in features.conf)? Yes, after I can pick it up from my phone (9133i), and it works. I had verbosity at 6 at the moment of testing. When he enters 701, only his phones displays Failed, nothing in asterisk. I can pickup after that on mine. Other calls/extension dials work from this phone? Yes, our extensions are 3 digits, and all other calls to / from this (problematic) phone works. Is the phone defined as a SIP extension/peer? If so, try sip set debug peer xxx and try the call/pickup again. -- _ -- 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] Pickup parcked call from Aastra 9480i ctcordless
Yes, after I can pick it up from my phone (9133i), and it works. I had verbosity at 6 at the moment of testing. When he enters 701, only his phones displays Failed, nothing in asterisk. I can pickup after that on mine. Is there a dial plan on the phone that you need to alter? -- _ -- 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] Yes it is a dimensioning question! Atom CPU
On 31 Aug 2010, at 18:10, Andrew Latham wrote: Sounds fine to me. Reckon you could do that on a toaster ;) Thanks, I needed to clean this keyboard anyway Hehe. It's true though. I was amazed what our atom boards would do. We even chucked transcoding/conferencing at them and they worked amazingly. I almost didn't believe it when I measured their power usage too... 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] Pickup parcked call from Aastra 9480i ct cordless
Your (local phone) dialplan is not getting pushed out to the handset. Increase the version number in your config to force it out to the handset... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Nicolas Ross [rossnick-li...@cybercat.ca] Sent: Tuesday, August 31, 2010 2:22 PM To: Asterisk Users List Subject: [asterisk-users] Pickup parcked call from Aastra 9480i ct cordless Here, we have 2 aastra 9133i and on 9480i ct cordless. We use often the park call feature of asterisk to transfer calls to one another. But the 9480i ct cordless cannot pickup a parked call. When manually entering 701 (parked call extention), the phone display Call failed (appel écoué in french). Nothing is displayed on the asterisk console. When doing it from other aastra phones (same config), or other make phones, it works. Any hints on possible causes ? -- _ -- 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] Pickup parcked call from Aastra9480i ctcordless
Is the phone defined as a SIP extension/peer? If so, try sip set debug peer xxx and try the call/pickup again. Yes, and doing so, the phone could no longer dial out, bizare. Yes, after I can pick it up from my phone (9133i), and it works. I had verbosity at 6 at the moment of testing. When he enters 701, only his phones displays Failed, nothing in asterisk. I can pickup after that on mine. Is there a dial plan on the phone that you need to alter? Yes : x+#|xx+*, same as mine. After the problem with dial-out stated on top, we rebooted the phone another time, and now everithing works... That was strange... -- _ -- 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] Yes it is a dimensioning question! Atom CPU
On Tue, Aug 31, 2010 at 3:27 PM, Steve Howes steve-li...@geekinter.net wrote: On 31 Aug 2010, at 18:10, Andrew Latham wrote: Sounds fine to me. Reckon you could do that on a toaster ;) Thanks, I needed to clean this keyboard anyway Hehe. It's true though. I was amazed what our atom boards would do. We even chucked transcoding/conferencing at them and they worked amazingly. I almost didn't believe it when I measured their power usage too... S I migrated to the Atom platform after I used a kill-a-watt measuring device, the power bill, and a calculator. Headphones are also a good migration point for power savings. This email typed from a very nice MSI AP1900 -- _ -- 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] Yes it is a dimensioning question! Atom CPU
On Tue, Aug 31, 2010 at 11:37 AM, jmilli...@sentinelcommunications.com wrote: snip simultaneous calls), g729 all the way through except voicemail will be wav format for email purposes(requirement). /snip This will take up most of your CPU cycles, be sure to keep transcoding to a minimum. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] No audio on call forward after upgrade from Asterisk 1.4 to 1.6
Hi Paul, I tried adding Progress() to no avail. I still get no audio and below is what comes up in the console. -- Accepting call from '403xx' to '0812' on channel 0/10, span 1 -- Executing [0...@isdn-incoming:1] Dial(DAHDI/10-1, SIP/812,60) in new stack == Using SIP RTP CoS mark 5 -- Called 812 -- Got SIP response 302 Moved Temporarily back from 192.168.1.148 -- Now forwarding DAHDI/10-1 to 'Local/8...@smallanimals' (thanks to SIP/812-0016) -- Executing [...@smallanimals:1] Progress(Local/8...@smallanimals-21bd;2, ) in new stack -- Executing [...@smallanimals:2] Playback(Local/8...@smallanimals-21bd;2, custom/ceh-meetingmsg) in new stack -- Local/8...@smallanimals-21bd;2 Playing 'custom/ceh-meetingmsg.gsm' (language 'en') -- Channel 0/10, span 1 got hangup request, cause 16 == Spawn extension (isdn-incoming, 0812, 1) exited non-zero on 'DAHDI/10-1' -- Hungup 'DAHDI/10-1' == Spawn extension (smallanimals, 849, 2) exited non-zero on 'Local/8...@smallanimals-21bd;2' The notion brought up earlier of a codec mismatch and Asterisk not transcoding feels like the right answer, but I won't know until I get on site. Thanks for the reply. aF On 31/08/2010, at 10:47 PM, Paul Belanger wrote: On Tue, Aug 31, 2010 at 4:04 AM, Alex Ferrara a...@receptiveit.com.au wrote: exten = 849,1,Playback(custom/ceh-meetingmsg) exten = 849,n,Hangup exten = 849,1,Progress() exten = 849,n,Playback(custom/ceh-meetingmsg) exten = 849,n,Hangup -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] No audio on call forward after upgrade fromAsterisk 1.4 to 1.6
You're probably not going to buy this, but if custom/ceh-meetingmsg is less than 7 seconds long, it could be playing before the connection is established. -- _ -- 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] No audio on call forward after upgrade from Asterisk 1.4 to 1.6
On Tue, Aug 31, 2010 at 5:50 PM, Alex Ferrara a...@receptiveit.com.au wrote: Hi Paul, I tried adding Progress() to no avail. I still get no audio and below is what comes up in the console. Try moving Progress() before the Dial(). If you Answer() the channel, do you have the same problem? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] STUN
Hello, i want to instal STUN Server in my Asterisk-PC. is it possible ? if yes, how kann i do it ? where can i find STUN Program? Thanks for your help. -- _ -- 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] Logging the CID from the Privacy Manager
Hi folks,
My v1.6 Asterisk system logs all Call Detail Records to a PostgreSQL
database, including those handled by the Privacy Manager.
Unfortunately, even though I can use the CLI to see the information
being submitted by anonymous callers to satisfy the demands of the the
Privacy Manager, that information is not recorded in the database.
Instead, all that is written to it:
clid: Privacy Manager anonymous
src:anonymous
Can the number submitted to the Privacy Manager somehow be recorded in
the database, instead of anonymous?
Thanks,
Jaap
PS -- Currently, the configuration I'm using in the dialplan for the
Privacy Manager looks like this:
exten = jw,1,Verbose(-- CID is ${CALLERID(num)})
exten = jw,n,GotoIf($[${CALLERID(num)}=anonymous]?true:false)
exten = jw,n(true),Set(CALLERID(num)=)
exten = jw,n(false),NoOp()
exten = jw,n,Verbose(-- CID is ${CALLERID(num)})
exten = jw,n,PrivacyManager(3,10)
exten = jw,n,GotoIf($[${PRIVACYMGRSTATUS}=FAILED]?bad)
exten = jw,n,Verbose(-- CID is ${CALLERID(num)})
exten = jw,n,Dial(SIP/1000,60,w)
exten = jw,n(bad),Playback(im-sorry)
exten = jw,n,Playback(vm-goodbye)
exten = jw,n,Hangup()
This message was sent using IMP, the Internet Messaging Program.
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Re: [asterisk-users] Logging the CID from the Privacy Manager
On 1/09/10 11:27 AM, Jaap Winius wrote:
exten = jw,1,Verbose(-- CID is${CALLERID(num)})
exten = jw,n,GotoIf($[${CALLERID(num)}=anonymous]?true:false)
exten = jw,n(true),Set(CALLERID(num)=)
exten = jw,n(false),NoOp()
exten = jw,n,Verbose(-- CID is${CALLERID(num)})
exten = jw,n,PrivacyManager(3,10)
exten = jw,n,GotoIf($[${PRIVACYMGRSTATUS}=FAILED]?bad)
exten = jw,n,Verbose(-- CID is${CALLERID(num)})
exten = jw,n,Dial(SIP/1000,60,w)
Maybe you could do:
Set(CDR(userfield)=${CALLERID(num)})
Before dialing SIP/1000
--
Cheers,
Matt Riddell
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http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
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Re: [asterisk-users] Mobile answer machine cut off
On 25/08/10 7:35 PM, Julian Lyndon-Smith wrote: Hey Matt, thanks for the response. I know it sounds impossible. Hell, I sound like a user :) But it *is* happening. And only on the cisco phones. We're trying to lab it up right now. What should I be looking for in the sip debug ? Just something happening when the call gets cut off. Is there any DTMF being transmitted, why was the call disconnected etc. Or just take a snippet and put it up on pastebin/post here -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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-1.8 problem with one-way audio with no nat
On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote: Hi. I have a soft phone -- expresstalk-- on a computer in my network and I use the internal ip address of the asterisk box to register the phone. But using asterisk-1.8 between revisions 281912 and 281982 it breaks -- after a few seconds of the call, I lose audio from the asterisk box to my soft phone, but not the other way around. This looks like one commit, but obviously I would like to know what's going on here? What's in the commit? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] Early media and IAX2
On 28/08/10 10:18 AM, Russ Dill wrote: My IAX2 trunk provider, Teliax, seems to be forcing early media. Early media is cool and all, but my Asterisk install doesn't seem to be fully supporting it. My initial setting was using Dial() to call all of my dahdi (TDM400P) extensions. The results were that incoming calls would not hear any ringing tones and the call would be ended by Teliax after 21 seconds. You could just answer the call before dialling your internal extensions. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] Digest Username/auth name mismatch
On 30/08/10 2:48 PM, kawanobe tomohito wrote: Hi I want to know how to solve below an error case. Uac cant's change username of from and digest header. I tried to put a...@192.168.0.1 on username of sip.conf.but same error returned. You don't need to have the @192.168.0.1 in there - just make sure the username and password are correct in the user's device. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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
