Re: [asterisk-users] Paging for Praying
How many people do you plan to page? because if numbers are high (or variable) you may have an easier life by using some sort of dialer if numbers are not very high and two lines are enough, our WombatDialer is free to use. l. 2012/12/29 bilal ghayyad bilmar...@yahoo.com 2) Praying time need to be obtained from text (or database). So, it is not always the same time. What actually is needed to be obtained from the text file or the database is the time of the pray for each date (for example, if today is 28 of December so the query will be for this date and then it is required to check if the time is same as the current time to page the wave file on the Phones). -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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] new user help required to build voice recorder with asterisk
With just one PRI card this should not be an issue, but for larger systems you may consider using something like Oreka to offload the I/O from the Asterisk server l. 2012/12/31 Vinod Nadiadwala thinw...@gmail.com Hi, I am new to asterisk, i want to know that is it possible to use asterisk for build voice recording system. Scenario : ISDN PRI line (30 line) I want every incoming outgoing call has to recorded, but without manual action. system has to auto receive the call. Please suggest, how should i start and with which hardware / cards it is possible. -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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] Catching hold in dialplan
Steve Murphy submitted a patch a while ago to track MOH on queues, you can find it at https://issues.asterisk.org/jira/browse/ASTERISK-20742 - it could be a good starting point to work on as it is quite short. Too bad it is still in limbo :-( l. 2012/12/19 Andrew White and...@computersforall.com.au Hey all, I’ve built a custom application for our call center and am having one problem. Unfortunately certain things happen whilst the agent has the customer on hold which I’d like to work around. But I can’t work out how to catch the actual hold event so I can do something about it. From the console with verbosity on 12, all I can see is: -- Started music on hold, class 'default', on SIP/trunk-9546 -- Started music on hold, class 'default', on SIP/100-9547 I’m happy to try and catch this AGI or via manager if needed, however a dialplan based solution would be best. Thanks all! Andrew -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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] Users list email totals by year .
So where has every body else gone? :) l. 2012/12/30 Mr. James W. Laferriere bab...@baby-dragons.com 2003, 24471 2004, 48608 2005, 59116 2006, 41215 2007, 26414 2008, 20746 2009, 18304 2010, 14948 2011, 11588 2012, 7542 -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 3237 Holden Road | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99709 | only on AXP | +--+ -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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] new user help required to build voice recorder with asterisk
Top post for the New Year. Yes, if you might scale up to 60 or more simultaneous calls, definitely look at OrecX or RTPTap because you will run into I/O issues. Not sure what current hardware can accommodate but it is best not to find out. Considering the very low cost of hardware these days compared with the cost of possible downtime, poor audio, lost recordings or whatever else you can assign a monetary value, I always suggest a separate machine for Passive recording when dealing with more than a handful of simultaneous calls. Thanks, Steve Totaro On Wed, Jan 2, 2013 at 6:18 AM, Lenz Emilitri lenz.lo...@gmail.com wrote: With just one PRI card this should not be an issue, but for larger systems you may consider using something like Oreka to offload the I/O from the Asterisk server l. 2012/12/31 Vinod Nadiadwala thinw...@gmail.com Hi, I am new to asterisk, i want to know that is it possible to use asterisk for build voice recording system. Scenario : ISDN PRI line (30 line) I want every incoming outgoing call has to recorded, but without manual action. system has to auto receive the call. Please suggest, how should i start and with which hardware / cards it is possible. -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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] new user help required to build voice recorder with asterisk
I don't know how many I/O can be achieved on a modern hardware, but I don't think 60 concurrent calls will be a problem. 60 calls are just 4 Mbit/s of data. However can be a good idea to start loading a server and be prepared to share the load on another server. Leandro 2013/1/2 Steve Totaro stot...@asteriskhelpdesk.com Top post for the New Year. Yes, if you might scale up to 60 or more simultaneous calls, definitely look at OrecX or RTPTap because you will run into I/O issues. Not sure what current hardware can accommodate but it is best not to find out. Considering the very low cost of hardware these days compared with the cost of possible downtime, poor audio, lost recordings or whatever else you can assign a monetary value, I always suggest a separate machine for Passive recording when dealing with more than a handful of simultaneous calls. Thanks, Steve Totaro On Wed, Jan 2, 2013 at 6:18 AM, Lenz Emilitri lenz.lo...@gmail.com wrote: With just one PRI card this should not be an issue, but for larger systems you may consider using something like Oreka to offload the I/O from the Asterisk server l. 2012/12/31 Vinod Nadiadwala thinw...@gmail.com Hi, I am new to asterisk, i want to know that is it possible to use asterisk for build voice recording system. Scenario : ISDN PRI line (30 line) I want every incoming outgoing call has to recorded, but without manual action. system has to auto receive the call. Please suggest, how should i start and with which hardware / cards it is possible. -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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 -- _ -- 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] Users list email totals by year .
So where has every body else gone? Still here, but mature working systems, still running 1.4.x Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging for Praying
Doesn’t the OP wish to page all phones? So it’s not an issue of dumping dozens of call files all at once. Does paging work? http://www.voip-info.org/wiki/view/Asterisk+cmd+Page http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom Overhead paging might also be something to consider, requiring just one call to page “everyone.” --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri Sent: Wednesday, January 02, 2013 5:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Paging for Praying How many people do you plan to page? because if numbers are high (or variable) you may have an easier life by using some sort of dialer if numbers are not very high and two lines are enough, our WombatDialer is free to use. l. 2012/12/29 bilal ghayyad bilmar...@yahoo.com 2) Praying time need to be obtained from text (or database). So, it is not always the same time. What actually is needed to be obtained from the text file or the database is the time of the pray for each date (for example, if today is 28 of December so the query will be for this date and then it is required to check if the time is same as the current time to page the wave file on the Phones). -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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] Dialing out and recording
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten = _X.,1,Dial(SIP/${EXTEN},60,…)
exten = _X.,n,Agi(agi://localhost/aj.agi?action=……..)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have AGI
control of it. Is there a way to do this or do I have to use G() and connect
the both ends to AGI separately and then bridging them before recording the
call?
Thanks for help.
Regards,
Henrik
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Dialing out and recording
Put the AGI call in a macro context and add M(macro) to your Dial string.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dialing out and recording
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten = _X.,1,Dial(SIP/${EXTEN},60,.)
exten = _X.,n,Agi(agi://localhost/aj.agi?action=)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have
AGI control of it. Is there a way to do this or do I have to use G() and
connect the both ends to AGI separately and then bridging them before
recording the call?
Thanks for help.
Regards,
Henrik
--
_
-- 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] Users list email totals by year .
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Wednesday, January 02, 2013 7:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Users list email totals by year . So where has every body else gone? Still here, but mature working systems, still running 1.4.x Doug As the thread said earlier (I think it was Shaun), the response mechanism has moved a good bit into the forums. The users list still is functional for folks who want to contribute but don’t keep a browser window open to monitor the forums. P.S. since the world has now turned twice, Happy New Year to anyone reading. -- _ -- 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 for Razberry Pi
Has anyone ported Asterisk to the Razzberry Pi? If so could you point me to info on doing so? Bob 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
Re: [asterisk-users] new user help required to build voice recorder with asterisk
It depends on what you do with them. Years ago, 60 calls would start to crap out audio on live calls and I learned that the hard way on a production call center. There was the I/O of just SLIN, then converting to MP3, then transferring to a not too forgiving SAMBA share. Scheduling things for a slower times and moving the MP3 conversion to the mass storage significantly helped while scrambling to find the permanent solution. People could increase those numbers with RAMDisk and other tricks but just moving it off the Phone System makes more sense. Why not engineer something to scale and last without knowing that you will have to revisit it and quite possibly at the most inopportune time, like when you just spent a good deal of money on an advertising spot? Thanks, Steve T On Wed, Jan 2, 2013 at 7:35 AM, Leandro Dardini ldard...@gmail.com wrote: I don't know how many I/O can be achieved on a modern hardware, but I don't think 60 concurrent calls will be a problem. 60 calls are just 4 Mbit/s of data. However can be a good idea to start loading a server and be prepared to share the load on another server. Leandro 2013/1/2 Steve Totaro stot...@asteriskhelpdesk.com Top post for the New Year. Yes, if you might scale up to 60 or more simultaneous calls, definitely look at OrecX or RTPTap because you will run into I/O issues. Not sure what current hardware can accommodate but it is best not to find out. Considering the very low cost of hardware these days compared with the cost of possible downtime, poor audio, lost recordings or whatever else you can assign a monetary value, I always suggest a separate machine for Passive recording when dealing with more than a handful of simultaneous calls. Thanks, Steve Totaro On Wed, Jan 2, 2013 at 6:18 AM, Lenz Emilitri lenz.lo...@gmail.com wrote: With just one PRI card this should not be an issue, but for larger systems you may consider using something like Oreka to offload the I/O from the Asterisk server l. 2012/12/31 Vinod Nadiadwala thinw...@gmail.com Hi, I am new to asterisk, i want to know that is it possible to use asterisk for build voice recording system. Scenario : ISDN PRI line (30 line) I want every incoming outgoing call has to recorded, but without manual action. system has to auto receive the call. Please suggest, how should i start and with which hardware / cards it is possible. -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk for Razberry Pi
On 13-01-02 10:55 AM, Robert Rawlinson wrote: Has anyone ported Asterisk to the Razzberry Pi? If so could you point me to info on doing so? Bob R If the Pi is running debian or a variant thereof, won't # apt-get install asterisk work? -- Looking for (employment|contract) work in the Internet industry, preferrably working remotely. Building / Supporting the net since 2400 baud was the hot thing. Ask for a resume! ispbuil...@gmail.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] new user help required to build voice recorder with asterisk
Mixmonitor also muxes the two sides of the conversation after hangup. That is quite a bit of I/O for 60 simultaneous calls lasting an average of 5-15mins On Wed, Jan 2, 2013 at 9:59 AM, Steve Totaro stot...@totarotechnologies.com wrote: It depends on what you do with them. Years ago, 60 calls would start to crap out audio on live calls and I learned that the hard way on a production call center. There was the I/O of just SLIN, then converting to MP3, then transferring to a not too forgiving SAMBA share. Scheduling things for a slower times and moving the MP3 conversion to the mass storage significantly helped while scrambling to find the permanent solution. People could increase those numbers with RAMDisk and other tricks but just moving it off the Phone System makes more sense. Why not engineer something to scale and last without knowing that you will have to revisit it and quite possibly at the most inopportune time, like when you just spent a good deal of money on an advertising spot? Thanks, Steve T On Wed, Jan 2, 2013 at 7:35 AM, Leandro Dardini ldard...@gmail.com wrote: I don't know how many I/O can be achieved on a modern hardware, but I don't think 60 concurrent calls will be a problem. 60 calls are just 4 Mbit/s of data. However can be a good idea to start loading a server and be prepared to share the load on another server. Leandro 2013/1/2 Steve Totaro stot...@asteriskhelpdesk.com Top post for the New Year. Yes, if you might scale up to 60 or more simultaneous calls, definitely look at OrecX or RTPTap because you will run into I/O issues. Not sure what current hardware can accommodate but it is best not to find out. Considering the very low cost of hardware these days compared with the cost of possible downtime, poor audio, lost recordings or whatever else you can assign a monetary value, I always suggest a separate machine for Passive recording when dealing with more than a handful of simultaneous calls. Thanks, Steve Totaro On Wed, Jan 2, 2013 at 6:18 AM, Lenz Emilitri lenz.lo...@gmail.com wrote: With just one PRI card this should not be an issue, but for larger systems you may consider using something like Oreka to offload the I/O from the Asterisk server l. 2012/12/31 Vinod Nadiadwala thinw...@gmail.com Hi, I am new to asterisk, i want to know that is it possible to use asterisk for build voice recording system. Scenario : ISDN PRI line (30 line) I want every incoming outgoing call has to recorded, but without manual action. system has to auto receive the call. Please suggest, how should i start and with which hardware / cards it is possible. -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk for Razberry Pi
On Wed, Jan 02, 2013 at 09:55:44AM -0500, Robert Rawlinson wrote: Has anyone ported Asterisk to the Razzberry Pi? If so could you point me to info on doing so? apt-get install asterisk -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 for Razberry Pi
On Wed, 2 Jan 2013, Robert Rawlinson wrote: Has anyone ported Asterisk to the Razzberry Pi? If so could you point me to info on doing so? Maybe you will find this interesting: http://nerdvittles.com/?p=3880 Regards, -- Tom m...@tdiehl.org Spamtrap address me...@tdiehl.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] Dialing out and recording
I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination and we
announce the call before the transfer is complete. The carrier requires a
diversion header if the ANI is not one of our DIDs. Does someone have
experience with this working?
(Top-posting 'cause the last guy did)
--Don
Danny Nicholas
Sent: Wednesday, January 02, 2013 8:18 AM
Put the AGI call in a macro context and add M(macro) to your Dial string.
Henrik Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten = _X.,1,Dial(SIP/${EXTEN},60,.)
exten = _X.,n,Agi(agi://localhost/aj.agi?action=)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have
AGI control of it. Is there a way to do this or do I have to use G() and
connect the both ends to AGI separately and then bridging them before
recording the call?
Thanks for help.
Regards,
Henrik
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialing out and recording
Henrik Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten = _X.,1,Dial(SIP/${EXTEN},60,.)
exten = _X.,n,Agi(agi://localhost/aj.agi?action=)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have
AGI control of it. Is there a way to do this or do I have to use G() and
connect the both ends to AGI separately and then bridging them before
recording the call?
Thanks for help.
Regards,
Henrik
Danny Nicholas
Sent: Wednesday, January 02, 2013 8:18 AM
Put the AGI call in a macro context and add M(macro) to your Dial string.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Wednesday, January 02, 2013 9:32 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Dialing out and recording
I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination and we
announce the call before the transfer is complete. The carrier requires a
diversion header if the ANI is not one of our DIDs. Does someone have
experience with this working?
--
Two suggestions for you, Don. #1 if the Dial is Private the
announcement is taken care of. #2 I'm supposing that you could do a SIP
Header command before the Dial to resolve the diversion header issue.
--
_
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Re: [asterisk-users] Top Posting
On Sun, Dec 30, 2012 at 2:54 PM, Benny Amorsen benny+use...@amorsen.dkwrote: Gergo Csibra csi...@gmail.com writes: Complaining about top posting on a list where's no moderation, no sanction if somebody top posting is pointless. There is a sanction. People like me will score top posters lower and soon not see their posts at all. Let me point out that there are users of this mailing list (such as myself) who are completely unaware of what you're talking about. It is often a quick way to see if it is worth responding to someone. If they top post, nothing of value is likely to come out of the conversation. That kind of attitude is unlikely to yield dividends in the long term. So by all means, everybody who wants to, keep top posting. I probably will, from time to time. Not always, but as Gmail evolves as a service, they seem to be making this style of conversation more and more difficult. Inline replies and bottom-posting are nearly impossible to do nicely on an iPhone. Outlook – as mentioned elsewhere in this thread – isn't helping here either. But a thinly veiled I'll take my ball and go home reaction isn't productive for either you or the communities you participate in. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Top Posting
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Harrington Sent: Wednesday, January 02, 2013 10:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting On Sun, Dec 30, 2012 at 2:54 PM, Benny Amorsen benny+use...@amorsen.dk wrote: Gergo Csibra csi...@gmail.com writes: Complaining about top posting on a list where's no moderation, no sanction if somebody top posting is pointless. There is a sanction. People like me will score top posters lower and soon not see their posts at all. I'm the opposite. I'm likely not to scroll down 10 pages to see the comments at the end. -- _ -- 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] Top Posting
On 2 Jan 2013, at 15:54, Eric Wieling wrote: On Sun, Dec 30, 2012 at 2:54 PM, Benny Amorsen benny+use...@amorsen.dk wrote: There is a sanction. People like me will score top posters lower and soon not see their posts at all. I'm the opposite. I'm likely not to scroll down 10 pages to see the comments at the end. Wouldn't need to if people trimmed their posts properly. 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] Top Posting
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, January 02, 2013 9:54 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Harrington Sent: Wednesday, January 02, 2013 10:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting On Sun, Dec 30, 2012 at 2:54 PM, Benny Amorsen benny+use...@amorsen.dk wrote: Gergo Csibra csi...@gmail.com writes: Complaining about top posting on a list where's no moderation, no sanction if somebody top posting is pointless. There is a sanction. People like me will score top posters lower and soon not see their posts at all. I'm the opposite. I'm likely not to scroll down 10 pages to see the comments at the end. -- I personally don't give a rat's rear whether it's at the top or the bottom; If it's relevant, I'll read it and if not it goes to file-13. Quit picking on Outlook and Blackberry users (no, keep it up, the list need the volume?) -- _ -- 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] Top Posting
I'm the opposite. I'm likely not to scroll down 10 pages to see the comments at the end. Wouldn't need to if people trimmed their posts properly. Precisely (e.g., see above)! Indeed, my sense is that top-posting *discourages* properly trimming email and that's my main reason against it. If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. -- _ -- 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] Top Posting
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner Sent: Wednesday, January 02, 2013 10:00 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Top Posting I'm the opposite. I'm likely not to scroll down 10 pages to see the comments at the end. Wouldn't need to if people trimmed their posts properly. Precisely (e.g., see above)! Indeed, my sense is that top-posting *discourages* properly trimming email and that's my main reason against it. If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. -- Good point. I've found myself having to edit and trim replies to poorly constructed conversations in the past because we got to the N'th iteration using either or both formats. -- _ -- 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 for Razberry Pi
On Wed, Jan 2, 2013 at 9:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Wed, Jan 02, 2013 at 09:55:44AM -0500, Robert Rawlinson wrote: Has anyone ported Asterisk to the Razzberry Pi? If so could you point me to info on doing so? apt-get install asterisk Does anyone know of any asterisk 11 packages for the Pi? I ended up compiling it myself this weekend. Took a while. -- Chris -- _ -- 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 for Razberry Pi
On 2 January 2013 16:16, Chris Gentle gent...@gmail.com wrote: Does anyone know of any asterisk 11 packages for the Pi? I ended up compiling it myself this weekend. Took a while. Take a look at http://www.raspberry-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] Top Posting
I'm the opposite. I'm likely not to scroll down 10 pages to see the comments at the end. Wouldn't need to if people trimmed their posts properly. Precisely (e.g., see above)! Indeed, my sense is that top-posting *discourages* properly trimming email and that's my main reason against it. If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. Good point. I've found myself having to edit and trim replies to poorly constructed conversations in the past because we got to the N'th iteration using either or both formats. In this properly trimmed example, there's no record of who said what. I like top posting--with trimming that removes no value and a resultant message that has the entire history so I can delete the older messages. As Outlook doesn't support the usenet approach to mail lists, what do people recommend for a good way of managing this type of list? --Don -- _ -- 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] Top Posting
On Wed, Jan 2, 2013 at 11:00 AM, Richard Kenner ken...@gnat.com wrote: I'm the opposite. I'm likely not to scroll down 10 pages to see the comments at the end. Wouldn't need to if people trimmed their posts properly. Precisely (e.g., see above)! Indeed, my sense is that top-posting *discourages* properly trimming email and that's my main reason against it. If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. Not really true often times when people do the right thing and post debug and conf files often required to get meaningful help. Thanks, Steve T -- _ -- 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 for Razberry Pi
On Wed, Jan 2, 2013 at 10:19 AM, Dan Jenkins dan.jenk...@holidayextras.comwrote: On 2 January 2013 16:16, Chris Gentle gent...@gmail.com wrote: Does anyone know of any asterisk 11 packages for the Pi? I ended up compiling it myself this weekend. Took a while. Take a look at http://www.raspberry-asterisk.org/ :) OK, thanks. Looks like that is a modified raspbian image. I'll give it a try. -- Chris -- _ -- 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] Top Posting
In this properly trimmed example, there's no record of who said what. When it's relevant, I trim in such a way that that information is preserved. But I would *never* leave in a header, just the identification of the person who typed that part. Most mailers, when you include text from another email, put someting like XYZ wrote: before the included text. So usually it's just a matter of preservating that and adding any that are needed that aren't there. Yes, it takes a few minutes longer, but given that there are probably hundreds of people reading my email, that's an investment that I find *well* worth it. -- _ -- 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] Top Posting
If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. Not really true often times when people do the right thing and post debug and conf files often required to get meaningful help. Yes, but if you put those at the end, where they belong, people reading the email can follow the thread quite easily and can ignore those if they don't need them. Certainly only a tiny part of such, if any at all, should be included in a reply. -- _ -- 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 seg fault 1.4.43
I finally got it to happen again. #0 0x00296f96 in __memcpy_ia32 () from /lib/libc.so.6 #1 0x0002 in ?? () #2 0x4d44fa0e in snd_pcm_area_copy () from /usr/lib/libasound.so.2 #3 0x4d44ff09 in snd_pcm_areas_copy () from /usr/lib/libasound.so.2 #4 0x4d4620f4 in snd_pcm_mmap_read_areas () from /usr/lib/libasound.so.2 #5 0x4d454bd0 in snd1_pcm_read_areas () from /usr/lib/libasound.so.2 #6 0x4d4624e4 in snd_pcm_mmap_readi () from /usr/lib/libasound.so.2 #7 0x4d44bbe5 in _snd_pcm_readi () from /usr/lib/libasound.so.2 #8 0x4d44d2d3 in snd_pcm_readi () from /usr/lib/libasound.so.2 #9 0xb7496575 in alsa_read (chan=0x830ac00) at chan_alsa.c:711 #10 0x0808b658 in __ast_read (chan=0x830ac00, dropaudio=0) at channel.c:2411 #11 0x0808d325 in ast_read (c0=0xb750eb68, c1=0x830ac00, config=0xb6f2acdc, fo=0xb6f29dac, rc=0xb6f29da8) at channel.c:2720 #12 ast_generic_bridge (c0=0xb750eb68, c1=0x830ac00, config=0xb6f2acdc, fo=0xb6f29dac, rc=0xb6f29da8) at channel.c:4647 #13 ast_channel_bridge (c0=0xb750eb68, c1=0x830ac00, config=0xb6f2acdc, fo=0xb6f29dac, rc=0xb6f29da8) at channel.c:4989 #14 0xb74f2fad in ast_bridge_call (chan=0xb750eb68, peer=0x830ac00, config=0xb6f2acdc) at res_features.c:2281 #15 0xb6f6df63 in dial_exec_full (chan=0xb750eb68, data=value optimized out, peerflags=0xb6f2ae4c, continue_exec=0x0) at app_dial.c:1894 #16 0xb6f703c6 in dial_exec (chan=0xb750eb68, data=0xb6f2cebc) at app_dial.c:1942 #17 0x080d2d9b in pbx_exec (c=0xb750eb68, con=value optimized out, context=0xb750ece8 smvoice-pa, exten=0xb750ed38 s, priority=8, label=0x0, callerid=0xb7510b30 501, action=E_SPAWN) at pbx.c:550 #18 pbx_extension_helper (c=0xb750eb68, con=value optimized out, context=0xb750ece8 smvoice-pa, exten=0xb750ed38 s, priority=8, label=0x0, callerid=0xb7510b30 501, action=E_SPAWN) at pbx.c:1893 #19 0x080d432f in ast_spawn_extension (c=0xb750eb68) at pbx.c:2367 #20 __ast_pbx_run (c=0xb750eb68) at pbx.c:2461 #21 0x080d5e3e in pbx_thread (data=0xb750eb68) at pbx.c:2688 #22 0x08107e6b in dummy_start (data=0xb750f4a8) at utils.c:856 #23 0x003c1a49 in start_thread () from /lib/libpthread.so.0 #24 0x002fe63e in clone () from /lib/libc.so.6 This is from the gdb where command. I am just calling into the box and using the ALSA channel for audio. This is VERY hard to re-create but it does happen. jerry -- _ -- 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] Top Posting
On 1/2/2013 11:30 AM, Richard Kenner wrote: If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. Not really true often times when people do the right thing and post debug and conf files often required to get meaningful help. Yes, but if you put those at the end, where they belong, people reading the email can follow the thread quite easily and can ignore those if they don't need them. Certainly only a tiny part of such, if any at all, should be included in a reply. Ok folks, could not stop myself any longer. This pissing and moaning is foolish to say the least. There was a post a while ago in the original hijacked thread by Steve Edwards that gave a link to the rules of the list at: http://www.asterisk.org/community/discuss/ GO READ THEM! Directly before the list of Rules is: Show consideration. It's important to read the rules before posting on a mailing list. Sage advice if you ask me, and yes I know nobody actually asked me. It is not hard to follow the rules . If the nice folks at Digium took the time to post rules we should at least TRY to follow them. If you do not like the rules you can always petition Digium to change them but, taking up bandwidth on the list in this all to frequent pissing match is a futile waste of time. Grow up, follow the rules, have a good day. JohnM -- _ -- 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 seg fault 1.4.43
What version of ALSA do you have installed? 1.0.26 is current ( http://alsa-project.org/main/index.php/Main_Page ) and it looks like the crash is in there. On Wed, Jan 2, 2013 at 10:32 AM, Jerry Geis ge...@pagestation.com wrote: I finally got it to happen again. #0 0x00296f96 in __memcpy_ia32 () from /lib/libc.so.6 #1 0x0002 in ?? () #2 0x4d44fa0e in snd_pcm_area_copy () from /usr/lib/libasound.so.2 #3 0x4d44ff09 in snd_pcm_areas_copy () from /usr/lib/libasound.so.2 #4 0x4d4620f4 in snd_pcm_mmap_read_areas () from /usr/lib/libasound.so.2 #5 0x4d454bd0 in snd1_pcm_read_areas () from /usr/lib/libasound.so.2 #6 0x4d4624e4 in snd_pcm_mmap_readi () from /usr/lib/libasound.so.2 #7 0x4d44bbe5 in _snd_pcm_readi () from /usr/lib/libasound.so.2 #8 0x4d44d2d3 in snd_pcm_readi () from /usr/lib/libasound.so.2 #9 0xb7496575 in alsa_read (chan=0x830ac00) at chan_alsa.c:711 #10 0x0808b658 in __ast_read (chan=0x830ac00, dropaudio=0) at channel.c:2411 #11 0x0808d325 in ast_read (c0=0xb750eb68, c1=0x830ac00, config=0xb6f2acdc, fo=0xb6f29dac, rc=0xb6f29da8) at channel.c:2720 #12 ast_generic_bridge (c0=0xb750eb68, c1=0x830ac00, config=0xb6f2acdc, fo=0xb6f29dac, rc=0xb6f29da8) at channel.c:4647 #13 ast_channel_bridge (c0=0xb750eb68, c1=0x830ac00, config=0xb6f2acdc, fo=0xb6f29dac, rc=0xb6f29da8) at channel.c:4989 #14 0xb74f2fad in ast_bridge_call (chan=0xb750eb68, peer=0x830ac00, config=0xb6f2acdc) at res_features.c:2281 #15 0xb6f6df63 in dial_exec_full (chan=0xb750eb68, data=value optimized out, peerflags=0xb6f2ae4c, continue_exec=0x0) at app_dial.c:1894 #16 0xb6f703c6 in dial_exec (chan=0xb750eb68, data=0xb6f2cebc) at app_dial.c:1942 #17 0x080d2d9b in pbx_exec (c=0xb750eb68, con=value optimized out, context=0xb750ece8 smvoice-pa, exten=0xb750ed38 s, priority=8, label=0x0, callerid=0xb7510b30 501, action=E_SPAWN) at pbx.c:550 #18 pbx_extension_helper (c=0xb750eb68, con=value optimized out, context=0xb750ece8 smvoice-pa, exten=0xb750ed38 s, priority=8, label=0x0, callerid=0xb7510b30 501, action=E_SPAWN) at pbx.c:1893 #19 0x080d432f in ast_spawn_extension (c=0xb750eb68) at pbx.c:2367 #20 __ast_pbx_run (c=0xb750eb68) at pbx.c:2461 #21 0x080d5e3e in pbx_thread (data=0xb750eb68) at pbx.c:2688 #22 0x08107e6b in dummy_start (data=0xb750f4a8) at utils.c:856 #23 0x003c1a49 in start_thread () from /lib/libpthread.so.0 #24 0x002fe63e in clone () from /lib/libc.so.6 This is from the gdb where command. I am just calling into the box and using the ALSA channel for audio. This is VERY hard to re-create but it does happen. jerry -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Top Posting
On 1/2/2013 12:00 PM, j...@millican.us wrote: On 1/2/2013 11:30 AM, Richard Kenner wrote: If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. Not really true often times when people do the right thing and post debug and conf files often required to get meaningful help. Yes, but if you put those at the end, where they belong, people reading the email can follow the thread quite easily and can ignore those if they don't need them. Certainly only a tiny part of such, if any at all, should be included in a reply. Ok folks, could not stop myself any longer. This pissing and moaning is foolish to say the least. There was a post a while ago in the original hijacked thread by Steve Edwards that gave a link to the rules of the list at: http://www.asterisk.org/community/discuss/ GO READ THEM! Directly before the list of Rules is: Show consideration. It's important to read the rules before posting on a mailing list. Sage advice if you ask me, and yes I know nobody actually asked me. It is not hard to follow the rules . If the nice folks at Digium took the time to post rules we should at least TRY to follow them. If you do not like the rules you can always petition Digium to change them but, taking up bandwidth on the list in this all to frequent pissing match is a futile waste of time. Grow up, follow the rules, have a good day. JohnM PS. Did not intend to imply that it was Steve that hijacked the thread, in case anyone read my comment that way JohnM -- _ -- 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] Top Posting
Grow up, follow the rules, have a good day. JohnM PS. Did not intend to imply that it was Steve that hijacked the thread, in case anyone read my comment that way JohnM Steve has waded through enough of these that he should be a hijacker. -- _ -- 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 seg fault 1.4.43
What version of ALSA do you have installed? 1.0.26 is current ( http://alsa-project.org/main/index.php/Main_Page ) and it looks like the crash is in there. I am using 1.0.25 Jerry -- _ -- 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] Top Posting
On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us wrote: On 1/2/2013 11:30 AM, Richard Kenner wrote: If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. Not really true often times when people do the right thing and post debug and conf files often required to get meaningful help. Yes, but if you put those at the end, where they belong, people reading the email can follow the thread quite easily and can ignore those if they don't need them. Certainly only a tiny part of such, if any at all, should be included in a reply. Ok folks, could not stop myself any longer. This pissing and moaning is foolish to say the least. There was a post a while ago in the original hijacked thread by Steve Edwards that gave a link to the rules of the list at: http://www.asterisk.org/community/discuss/ GO READ THEM! Directly before the list of Rules is: Show consideration. It's important to read the rules before posting on a mailing list. Sage advice if you ask me, and yes I know nobody actually asked me. It is not hard to follow the rules . If the nice folks at Digium took the time to post rules we should at least TRY to follow them. If you do not like the rules you can always petition Digium to change them but, taking up bandwidth on the list in this all to frequent pissing match is a futile waste of time. Grow up, follow the rules, have a good day. JohnM I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Thanks, Steve Totaro -- _ -- 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] Top Posting
On 1/2/2013 12:20 PM, Steve Totaro wrote: On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us wrote: On 1/2/2013 11:30 AM, Richard Kenner wrote: If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. Not really true often times when people do the right thing and post debug and conf files often required to get meaningful help. Yes, but if you put those at the end, where they belong, people reading the email can follow the thread quite easily and can ignore those if they don't need them. Certainly only a tiny part of such, if any at all, should be included in a reply. Ok folks, could not stop myself any longer. This pissing and moaning is foolish to say the least. There was a post a while ago in the original hijacked thread by Steve Edwards that gave a link to the rules of the list at: http://www.asterisk.org/community/discuss/ GO READ THEM! Directly before the list of Rules is: Show consideration. It's important to read the rules before posting on a mailing list. Sage advice if you ask me, and yes I know nobody actually asked me. It is not hard to follow the rules . If the nice folks at Digium took the time to post rules we should at least TRY to follow them. If you do not like the rules you can always petition Digium to change them but, taking up bandwidth on the list in this all to frequent pissing match is a futile waste of time. Grow up, follow the rules, have a good day. JohnM I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Thanks, Steve Totaro So Steve, can I steal this and send it to the IRS? The ATF? Local Police Department? G Wouldn't that be nice! Sorry couldn't resist. JohnM -- _ -- 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] Top Posting
On 01/02/2013 12:20 PM, Steve Totaro wrote: good one - me too ! On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us wrote: On 1/2/2013 11:30 AM, Richard Kenner wrote: If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. Not really true often times when people do the right thing and post debug and conf files often required to get meaningful help. Yes, but if you put those at the end, where they belong, people reading the email can follow the thread quite easily and can ignore those if they don't need them. Certainly only a tiny part of such, if any at all, should be included in a reply. Ok folks, could not stop myself any longer. This pissing and moaning is foolish to say the least. There was a post a while ago in the original hijacked thread by Steve Edwards that gave a link to the rules of the list at: http://www.asterisk.org/community/discuss/ GO READ THEM! Directly before the list of Rules is: Show consideration. It's important to read the rules before posting on a mailing list. Sage advice if you ask me, and yes I know nobody actually asked me. It is not hard to follow the rules . If the nice folks at Digium took the time to post rules we should at least TRY to follow them. If you do not like the rules you can always petition Digium to change them but, taking up bandwidth on the list in this all to frequent pissing match is a futile waste of time. Grow up, follow the rules, have a good day. JohnM I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Thanks, Steve Totaro -- _ -- 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] Top Posting
At 08:22 AM 1/2/2013, you wrote: Wouldn't need to if people trimmed their posts properly. Precisely (e.g., see above)! Indeed, my sense is that top-posting *discourages* properly trimming email and that's my main reason against it. If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. I would think that if people properly trimmed their posts it would hardly matter at all and that life would be a lot better if we griped about that instead of where to post. And I find the people who top post somewhat insulting, I usually top post because my email program has no clue what a thread is. I always trim my messages and unless the topic is really interesting I won't scroll more than 1 or 2 pages to see an answer. And I started communicating with a 2400 baud modem so trimming was a necessity and a requirement of friendship. I think the Will Asterisk run on a Rasberry Pi thread the perfect example of why this list is dying. I don't have a Pi but I did spend an hour one day researching one and I know I came across all the answers in that thread. Sadly, for me, the Pi is the perfect example of why there needs to be $25 USB to POTs and USB to analog phone adapters. Ira -- _ -- 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] Top Posting
It is not hard to follow the rules . If the nice folks at Digium took the time to post rules we should at least TRY to follow them. If you do not like the rules you can always petition Digium to change them but, taking up bandwidth on the list in this all to frequent pissing match is a futile waste of time. Grow up, follow the rules, have a good day. JohnM I'm 69, not too likely to do much more growing up, and I do follow the rules, unless the thread is already top-posted. I'm young enough, though, that I don't have a problem discussing change, and I thought I had started a new thread with the Top Posting subject so you wouldn't need to waste your time looking at it. If there were change, I'd think it would be better to come from the list users rather than from Digium. If you'd like to add real value to this discussion, you might respond to my request for information on what product/procedure/whatever would enable me to follow and participate in bottom-posted discussions as it doesn't appear that Outlook or gmail are very effective. --Don -- _ -- 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] Top Posting
On Wed, Jan 2, 2013 at 11:02 AM, Ira i...@extrasensory.com wrote: And I started communicating with a 2400 baud modem so trimming was a necessity and a requirement of friendship. Bah, spoiled kids. Mine was a 110 baud acoustic. I think the Will Asterisk run on a Rasberry Pi thread the perfect example of why this list is dying. The number of questions posted here that are easily answered with a search or which are far too basic and open (how do I make Asterisk work) is very high these days, and that does kill a list. A lot of us are interested in helping people who help themselves, and solving complex problems. I've seen many tech lists die off when people stop trying to help themselves and ask intelligent questions. As to top-posting, another example of when I think it's generally acceptable is people using tablets. I have found no way on either my iOS or Android tablets to quickly/easily post in the traditional manner. If I'm faced with spending a few minutes carefully trimming a useful reply or just not posting it at all, I'm likely to choose the latter if I'm on a list that says absolutely never top post. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Top Posting
On 1/2/2013 1:10 PM, Don Kelly wrote: It is not hard to follow the rules . If the nice folks at Digium took the time to post rules we should at least TRY to follow them. If you do not like the rules you can always petition Digium to change them but, taking up bandwidth on the list in this all to frequent pissing match is a futile waste of time. Grow up, follow the rules, have a good day. JohnM I'm 69, not too likely to do much more growing up, and I do follow the rules, unless the thread is already top-posted. I'm young enough, though, that I don't have a problem discussing change, and I thought I had started a new thread with the Top Posting subject so you wouldn't need to waste your time looking at it. If there were change, I'd think it would be better to come from the list users rather than from Digium. If you'd like to add real value to this discussion, you might respond to my request for information on what product/procedure/whatever would enable me to follow and participate in bottom-posted discussions as it doesn't appear that Outlook or gmail are very effective. --Don Umm, what about positioning the cursor below the previous post before writing your reply in outlook, I used to do it all the time when forced to use outlook by company policy or such. Click on scroll bar drag - to bottom of reply - click in message body, about a half seconds time, maybe a full second if you choose to move slowly. Admittedly though it has been a few versions since I have been forced to use Outlook, I currently use Thunderbird for mail and can set it to start my reply on top or at the bottom. JohnM -- _ -- 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] Top Posting
On 02/01/2013 1:11 PM, Carlos Alvarez wrote: On Wed, Jan 2, 2013 at 11:02 AM, Ira i...@extrasensory.com mailto:i...@extrasensory.com wrote: And I started communicating with a 2400 baud modem so trimming was a necessity and a requirement of friendship. Bah, spoiled kids. Mine was a 110 baud acoustic. I think the Will Asterisk run on a Rasberry Pi thread the perfect example of why this list is dying. The number of questions posted here that are easily answered with a search or which are far too basic and open (how do I make Asterisk work) is very high these days, and that does kill a list. A lot of us are interested in helping people who help themselves, and solving complex problems. I've seen many tech lists die off when people stop trying to help themselves and ask intelligent questions. As to top-posting, another example of when I think it's generally acceptable is people using tablets. I have found no way on either my iOS or Android tablets to quickly/easily post in the traditional manner. If I'm faced with spending a few minutes carefully trimming a useful reply or just not posting it at all, I'm likely to choose the latter if I'm on a list that says absolutely never top post. -- Carlos Alvarez TelEvolve 602-889-3003 If you are answering one of my questions, please feel free to top post, bottom post or post in the middle. I would rather have an answer than nothing - no matter how nicely formatted. Part of the problem is the way that Asterisk is delivered. The configuration files are way too complex and handle a lot of obscure situations rather than being minimal working configurations. I am not sure that all of the defaults actually make sense - I just had to go in and turn on tos in SIP. The default is none which is not what the docs that I found, recommend. SIP login comes with defaults that are not recommended for security reasons. The documentation is hard to use. At the same time, there is an expectation in the public that a competent system administrator can install an Asterisk PBX. This being said, given the number of Asterisk installations being installed each day by first-time administrators, the traffic here seems pretty reasonable both in volume and in level of difficulty. Ron -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Top Posting
On 01/02/2013 07:11 PM, Carlos Alvarez wrote: The number of questions posted here that are easily answered with a search or which are far too basic and open (how do I make Asterisk work) is very high these days, and that does kill a list. A lot of us are interested in helping people who help themselves, and solving complex problems. I've seen many tech lists die off when people stop trying to help themselves and ask intelligent questions. Good point Carlos and I share your feeling. On the Postfix mailing list, when someone asks a basic how do I ... question, inevitably the response is one or more links to a section in the documentation. And that works really well. The interesting problems discussed on that ML outnumber the questions from those who can't be bothered to try to help themselves by spending a couple of minutes reading the docs. I would welcome similar responses on this mailing list to improve the S/N ratio. As to top-posting, another example of when I think it's generally acceptable is people using tablets. I have found no way on either my iOS or Android tablets to quickly/easily post in the traditional manner. If I'm faced with spending a few minutes carefully trimming a useful reply or just not posting it at all, I'm likely to choose the latter if I'm on a list that says absolutely never top post. I only use Thunderbird to post but I now have seen several arguments that MUAs like Outlook and iOS/Android clients are simply not capable of bottom posting trimming. Perhaps the list admins could take that into account when appropriate. Regards, Patrick -- _ -- 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] Top Posting
On 1/2/2013 Don Kelly wrote: ... what product/procedure/whatever would enable me to follow and participate in bottom-posted discussions as it doesn't appear that Outlook or gmail are very effective. Umm, what about positioning the cursor below the previous post before writing your reply in outlook, I used to do it all the time when forced to use outlook by company policy or such. Click on scroll bar drag - to bottom of reply - click in message body, about a half seconds time, maybe a full second if you choose to move slowly. Admittedly though it has been a few versions since I have been forced to use Outlook, I currently use Thunderbird for mail and can set it to start my reply on top or at the bottom. JohnM I don't have any problem getting my reply to the bottom of the email, but Outlook doesn't do any indenting or or anything (Makes in-line comments really hard to work with). When people following the rules trim everything, I end up seeing Works for me Me too with no way of following the thread to see what they're talking about (especially if the subject is Merry Christmas and they're talking about Razberry Pi). I don't think Outlook does what I'd like, so I'm not limiting my options. I can use different email to keep track of the Asterisk lists. --Don -- _ -- 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] Users list email totals by year .
It would be nice (for me anyway) if the mailing list and forum were combined. Google Groups does this nicely I believe. Mitch On 01/02/2013 08:53 AM, Eric Wieling wrote: I don't use forums as my web browser can't automatically filter the messages for me like my e-mail program can. I stopped participating in the mailing list when it became clear most of the questions were about FreePBX. That seems to have died down a little in recent years. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, January 02, 2013 9:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Users list email totals by year . From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Wednesday, January 02, 2013 7:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Users list email totals by year . So where has every body else gone? Still here, but mature working systems, still running 1.4.x Doug As the thread said earlier (I think it was Shaun), the response mechanism has moved a good bit into the forums. The users list still is functional for folks who want to contribute but don’t keep a browser window open to monitor the forums. P.S. since the world has now turned twice, Happy New Year to anyone reading. -- _ -- 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] Top Posting
On 01/02/2013 06:20 PM, Steve Totaro wrote: I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Just looked it up. I see my first post back in April 2003, yours in September 2003 and Jon in March 2003. Wow you find something fun to play with and suddenly a decade has passed :-) Regards, Patrick -- _ -- 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 as answering machine
I have connected a PSTN line to a Digium FXO card. There is also an ordinary analogue phone attached to the same line. The Asterisk answers the line on the first ring. I would like it to wait for a few seconds so that someone can answer the PSTN line with an analogue phone. This would allow a person to directly pick up the line if they wanted to or if not, let it go to the Asterisk where it would be dispatched through the normal process. Currently, as soon as the analogue phone rings, the Asterisk PBX has already answered the call and starts the You have reached. Dial and tries to dispatch the call. This makes it hard to carry on a conversation. Ron -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Top Posting
On Wed, Jan 2, 2013 at 12:25 PM, j...@millican.us j...@millican.us wrote: On 1/2/2013 12:20 PM, Steve Totaro wrote: On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us wrote: On 1/2/2013 11:30 AM, Richard Kenner wrote: If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. Not really true often times when people do the right thing and post debug and conf files often required to get meaningful help. Yes, but if you put those at the end, where they belong, people reading the email can follow the thread quite easily and can ignore those if they don't need them. Certainly only a tiny part of such, if any at all, should be included in a reply. Ok folks, could not stop myself any longer. This pissing and moaning is foolish to say the least. There was a post a while ago in the original hijacked thread by Steve Edwards that gave a link to the rules of the list at: http://www.asterisk.org/community/discuss/ GO READ THEM! Directly before the list of Rules is: Show consideration. It's important to read the rules before posting on a mailing list. Sage advice if you ask me, and yes I know nobody actually asked me. It is not hard to follow the rules . If the nice folks at Digium took the time to post rules we should at least TRY to follow them. If you do not like the rules you can always petition Digium to change them but, taking up bandwidth on the list in this all to frequent pissing match is a futile waste of time. Grow up, follow the rules, have a good day. JohnM I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Thanks, Steve Totaro So Steve, can I steal this and send it to the IRS? The ATF? Local Police Department? G Wouldn't that be nice! Sorry couldn't resist. JohnM What the hell are you implying? The local police love me, I am in good standing with the ATF, FBI, DoD, DoS, USAID, DoE, DoL, and NSA. IRS wants some money in April but don't they always? LOL. Thanks, Steve T -- _ -- 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 as answering machine
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler Sent: Wednesday, January 02, 2013 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk as answering machine I have connected a PSTN line to a Digium FXO card. There is also an ordinary analogue phone attached to the same line. The Asterisk answers the line on the first ring. I would like it to wait for a few seconds so that someone can answer the PSTN line with an analogue phone. This would allow a person to directly pick up the line if they wanted to or if not, let it go to the Asterisk where it would be dispatched through the normal process. Currently, as soon as the analogue phone rings, the Asterisk PBX has already answered the call and starts the You have reached. Dial and tries to dispatch the call. This makes it hard to carry on a conversation. Ron In your dialplan, put Wait(10) in front of Answer(). This will give the human 4 rings to pick up before Asterisk does. -- _ -- 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 as answering machine
Problematic at best. Just make a phone an extension and allow that to ring in a hunt group. Thanks, Steve Totaro On Wed, Jan 2, 2013 at 3:30 PM, Ron Wheeler rwhee...@artifact-software.com wrote: I have connected a PSTN line to a Digium FXO card. There is also an ordinary analogue phone attached to the same line. The Asterisk answers the line on the first ring. I would like it to wait for a few seconds so that someone can answer the PSTN line with an analogue phone. This would allow a person to directly pick up the line if they wanted to or if not, let it go to the Asterisk where it would be dispatched through the normal process. Currently, as soon as the analogue phone rings, the Asterisk PBX has already answered the call and starts the You have reached. Dial and tries to dispatch the call. This makes it hard to carry on a conversation. Ron -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as answering machine
I recommend using WaitForRing instead of Wait. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, January 02, 2013 3:33 PM To: rwhee...@artifact-software.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk as answering machine -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler Sent: Wednesday, January 02, 2013 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk as answering machine I have connected a PSTN line to a Digium FXO card. There is also an ordinary analogue phone attached to the same line. The Asterisk answers the line on the first ring. I would like it to wait for a few seconds so that someone can answer the PSTN line with an analogue phone. This would allow a person to directly pick up the line if they wanted to or if not, let it go to the Asterisk where it would be dispatched through the normal process. Currently, as soon as the analogue phone rings, the Asterisk PBX has already answered the call and starts the You have reached. Dial and tries to dispatch the call. This makes it hard to carry on a conversation. Ron In your dialplan, put Wait(10) in front of Answer(). This will give the human 4 rings to pick up before Asterisk does. -- _ -- 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] Top Posting
Hi, one more hint... (trying to translate the commands to english) in Thunderbird open - Extras - Filter.. - Filter-Name: enter Top Posting Subject - Contains: enter Top Posting Action: Delete Markus Am 02.01.2013 21:31, schrieb Steve Totaro: On Wed, Jan 2, 2013 at 12:25 PM, j...@millican.us j...@millican.us wrote: On 1/2/2013 12:20 PM, Steve Totaro wrote: On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us wrote: On 1/2/2013 11:30 AM, Richard Kenner wrote: If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. Not really true often times when people do the right thing and post debug and conf files often required to get meaningful help. Yes, but if you put those at the end, where they belong, people reading the email can follow the thread quite easily and can ignore those if they don't need them. Certainly only a tiny part of such, if any at all, should be included in a reply. Ok folks, could not stop myself any longer. This pissing and moaning is foolish to say the least. There was a post a while ago in the original hijacked thread by Steve Edwards that gave a link to the rules of the list at: http://www.asterisk.org/community/discuss/ GO READ THEM! Directly before the list of Rules is: Show consideration. It's important to read the rules before posting on a mailing list. Sage advice if you ask me, and yes I know nobody actually asked me. It is not hard to follow the rules . If the nice folks at Digium took the time to post rules we should at least TRY to follow them. If you do not like the rules you can always petition Digium to change them but, taking up bandwidth on the list in this all to frequent pissing match is a futile waste of time. Grow up, follow the rules, have a good day. JohnM I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Thanks, Steve Totaro So Steve, can I steal this and send it to the IRS? The ATF? Local Police Department? G Wouldn't that be nice! Sorry couldn't resist. JohnM What the hell are you implying? The local police love me, I am in good standing with the ATF, FBI, DoD, DoS, USAID, DoE, DoL, and NSA. IRS wants some money in April but don't they always? LOL. Thanks, Steve T -- _ -- 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] Top Posting
On 01/02/2013 12:16 PM, Don Kelly wrote: I don't think Outlook does what I'd like, so I'm not limiting my options. I can use different email to keep track of the Asterisk lists. Thunderbird (by default) bottom posts. And it does the nice indenting and allows you to turn off that HTML crap... :) Anybody have any suggestions on a good email client for an Andriod device. A client that actually lets me set BCC or allows me to edit the original message when I replying? The built in client sucks!!! -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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] Top Posting
On 01/02/2013 03:22 PM, Patrick Lists wrote: On 01/02/2013 06:20 PM, Steve Totaro wrote: I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Just looked it up. I see my first post back in April 2003, yours in September 2003 and Jon in March 2003. Wow you find something fun to play with and suddenly a decade has passed :-) Are you sure about that ? I know I was doing stuff with asterisk back in the LSS days and that was around 2001 Regards, Patrick -- _ -- 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] Top Posting
On Wed, Jan 2, 2013 at 3:46 PM, jon pounder j...@inline.net wrote: On 01/02/2013 03:22 PM, Patrick Lists wrote: On 01/02/2013 06:20 PM, Steve Totaro wrote: I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Just looked it up. I see my first post back in April 2003, yours in September 2003 and Jon in March 2003. Wow you find something fun to play with and suddenly a decade has passed :-) Are you sure about that ? I know I was doing stuff with asterisk back in the LSS days and that was around 2001 The archives are a bit sketchy before Feb of 2003. I would guess my first dabble was circa 2001 and started making money from it in 2002. Right around the debut of the 3COM NBX 100. Thanks, Steve T -- _ -- 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] Top Posting
On 01/02/2013 03:35 PM, Jim Lucas wrote: On 01/02/2013 12:16 PM, Don Kelly wrote: I don't think Outlook does what I'd like, so I'm not limiting my options. I can use different email to keep track of the Asterisk lists. Thunderbird (by default) bottom posts. And it does the nice indenting and allows you to turn off that HTML crap... :) Anybody have any suggestions on a good email client for an Andriod device. A client that actually lets me set BCC or allows me to edit the original message when I replying? The built in client sucks!!! maildroid has a lot of features but kills your battery FAST. I only start it when I am expecting an important email and task kill it afterwards or it stays running. -- _ -- 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 as answering machine
On 02/01/2013 3:33 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler Sent: Wednesday, January 02, 2013 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk as answering machine I have connected a PSTN line to a Digium FXO card. There is also an ordinary analogue phone attached to the same line. The Asterisk answers the line on the first ring. I would like it to wait for a few seconds so that someone can answer the PSTN line with an analogue phone. This would allow a person to directly pick up the line if they wanted to or if not, let it go to the Asterisk where it would be dispatched through the normal process. Currently, as soon as the analogue phone rings, the Asterisk PBX has already answered the call and starts the You have reached. Dial and tries to dispatch the call. This makes it hard to carry on a conversation. Ron In your dialplan, put Wait(10) in front of Answer(). This will give the human 4 rings to pick up before Asterisk does. I feel pretty silly. It worked. I saw this in a few Google responses but thought that I had already tested this. Now I have just dumped myself back to newbee status. Thanks Ron -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Dialing out and recording
Thanks Danny I will try this.
/Henrik
Message: 12
Date: Wed, 2 Jan 2013 08:17:59 -0600
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] Dialing out and recording
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: 001501cde8f3$f7d2b290$e77817b0$@debsinc.com
Content-Type: text/plain; charset=us-ascii
Put the AGI call in a macro context and add M(macro) to your Dial string.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dialing out and recording
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten = _X.,1,Dial(SIP/${EXTEN},60,.)
exten = _X.,n,Agi(agi://localhost/aj.agi?action=)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and
have
AGI control of it. Is there a way to do this or do I have to use G() and
connect the both ends to AGI separately and then bridging them before
recording the call?
Thanks for help.
Regards,
Henrik
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Re: [asterisk-users] Dialing out and recording
#2 works for me on Asterisk 1.8.12 when setting the header like this:
exten = _S,n,SipSetHeader(Diversion: ${CALLERID(rdnis)})
I haven't been able to make it work on 1.6 yet though, has anyone else?
/Henrik
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Wednesday, January 02, 2013 9:32 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Dialing out and recording
I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination and
we
announce the call before the transfer is complete. The carrier requires
a
diversion header if the ANI is not one of our DIDs. Does someone have
experience with this working?
--
Two suggestions for you, Don. #1 if the Dial is Private the
announcement is taken care of. #2 I'm supposing that you could do a SIP
Header command before the Dial to resolve the diversion header issue.
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[asterisk-users] AST-2012-014: Crashes due to large stack allocations when using TCP
Asterisk Project Security Advisory - AST-2012-014 ProductAsterisk SummaryCrashes due to large stack allocations when using TCP Nature of Advisory Stack Overflow SusceptibilityRemote Unauthenticated Sessions (SIP) Remote Authenticated Sessions (XMPP, HTTP) Severity Critical Exploits KnownNo Reported On 7 November, 2012 Reported By Walter Doekes Posted On 2 January, 2013 Last Updated OnJanuary 2, 2013 Advisory Contact Mark Michelson mmichelson AT digium DOT com CVE Name CVE-2012-5976 Description Asterisk has several places where messages received over various network transports may be copied in a single stack allocation. In the case of TCP, since multiple packets in a stream may be concatenated together, this can lead to large allocations that overflow the stack. In the case of SIP, it is possible to do this before a session is established. Keep in mind that SIP over UDP is not affected by this vulnerability. With HTTP and XMPP, a session must first be established before the vulnerability may be exploited. The XMPP vulnerability exists both in the res_jabber.so module in Asterisk 1.8, 10, and 11 as well as the res_xmpp.so module in Asterisk 11. Resolution Stack allocations when using TCP have either been eliminated in favor of heap allocations or have had an upper bound placed on them to ensure that the stack will not overflow. For SIP, the allocation now has an upper limit. For HTTP, the allocation is now a heap allocation instead of a stack allocation. For XMPP, the allocation has been eliminated since it was unnecessary. Affected Versions Product Release Series Asterisk Open Source 1.8.xAll versions Asterisk Open Source 10.x All versions Asterisk Open Source 11.x All versions Certified Asterisk 1.8.11SIP: unaffected HTTP and XMPP: All versions Asterisk Digiumphones 10.x-digiumphones All versions Corrected In Product Release Asterisk Open Source 1.8.19.1, 10.11.1, 11.1.1 Certified Asterisk 1.8.11-cert10 Asterisk Digiumphones10.11.1-digiumphones Patches SVN URL Revision http://downloads.asterisk.org/pub/security/AST-2012-014-1.8.diff Asterisk 1.8 http://downloads.asterisk.org/pub/security/AST-2012-014-10.diff Asterisk 10 http://downloads.asterisk.org/pub/security/AST-2012-014-11.diff Asterisk 11 Links
[asterisk-users] AST-2012-015: Denial of Service Through Exploitation of Device State Caching
Asterisk Project Security Advisory - AST-2012-015 ProductAsterisk SummaryDenial of Service Through Exploitation of Device State Caching Nature of Advisory Denial of Service SusceptibilityRemote Unauthenticated Sessions Severity Critical Exploits KnownNone Reported On 26 July, 2012 Reported By Russell Bryant Posted On 2 January, 2013 Last Updated OnJanuary 2, 2013 Advisory Contact Matt Jordan mjordan AT digium DOT com CVE Name CVE-2012-5977 Description Asterisk maintains an internal cache for devices. The device state cache holds the state of each device known to Asterisk, such that consumers of device state information can query for the last known state for a particular device, even if it is not part of an active call. The concept of a device in Asterisk can include things that do not have a physical representation. One way that this currently occurs is when anonymous calls are allowed in Asterisk. A device is automatically created and stored in the cache for each anonymous call that occurs; this is possible in the SIP and IAX2 channel drivers and through channel drivers that utilize the res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). Attackers exploiting this vulnerability can attack an Asterisk system configured to allow anonymous calls by varying the source of the anonymous call, continually adding devices to the device state cache and consuming a system's resources. Resolution Channels that are not associated with a physical device are no longer stored in the device state cache. This affects Local, DAHDI, SIP and IAX2 channels, and any channel drivers built on the res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). Affected Versions Product Release Series Asterisk Open Source 1.8.xAll Versions Asterisk Open Source 10.x All Versions Asterisk Open Source 11.x All Versions Certified Asterisk 1.8.11All Versions Asterisk Digiumphones 10.x-digiumphones All Versions Corrected In Product Release Asterisk Open Source 1.8.19.1, 10.11.1, 11.1.1 Certified Asterisk 1.8.11-cert10 Asterisk Digiumphones10.11.1-digiumphones Patches SVN URL Revision http://downloads.asterisk.org/pub/security/AST-2012-015-1.8.diff Asterisk 1.8 http://downloads.asterisk.org/pub/security/AST-2012-015-10.diff Asterisk 10 http://downloads.asterisk.org/pub/security/AST-2012-015-11.diff Asterisk 11 Links https://issues.asterisk.org/jira/browse/ASTERISK-20175 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-2012-015.pdf and http://downloads.digium.com/pub/security/AST-2012-015.html Revision
Re: [asterisk-users] Dialing out and recording
1.6.2 is a deader soldier than 1.4.X.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 3:20 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dialing out and recording
#2 works for me on Asterisk 1.8.12 when setting the header like this:
exten = _S,n,SipSetHeader(Diversion: ${CALLERID(rdnis)})
I haven't been able to make it work on 1.6 yet though, has anyone else?
/Henrik
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Wednesday, January 02, 2013 9:32 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Dialing out and recording
I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination
and we announce the call before the transfer is complete. The carrier
requires a diversion header if the ANI is not one of our DIDs. Does
someone have experience with this working?
--
Two suggestions for you, Don. #1 if the Dial is Private the
announcement is taken care of. #2 I'm supposing that you could do a
SIP Header command before the Dial to resolve the diversion header issue.
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[asterisk-users] Asterisk 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones, 11.1.1 Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.11 and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones, and 11.1.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these versions resolve the following two issues: * Stack overflows that occur in some portions of Asterisk that manage a TCP connection. In SIP, this is exploitable via a remote unauthenticated session; in XMPP and HTTP connections, this is exploitable via remote authenticated sessions. * A denial of service vulnerability through exploitation of the device state cache. Anonymous calls had the capability to create devices in Asterisk that would never be disposed of. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2012-014 and AST-2012-015, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert10 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.19.1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.11.1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.11.1-digiumphones http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2012-014.pdf * http://downloads.asterisk.org/pub/security/AST-2012-015.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
Re: [asterisk-users] Top Posting
Gmail has just updated some stuff and I've been fiddling with the gmail ap on Android (Ice Cream Sandwich). I can select inline reply, delete superfluous stuff and go to the bottom for my post. After a few messages back and forth, the thread is displayed with a Show quoted text link for each post and the current post at the bottom. In gmail in my Chrome browser, the message is displayed with the subject at the top and each of the posts (without quoting--even though it's in the message) all nicely stacked up below. I haven't found the bottom post button in the browser. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder Sent: Wednesday, January 02, 2013 2:57 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Top Posting On 01/02/2013 03:35 PM, Jim Lucas wrote: On 01/02/2013 12:16 PM, Don Kelly wrote: I don't think Outlook does what I'd like, so I'm not limiting my options. I can use different email to keep track of the Asterisk lists. Thunderbird (by default) bottom posts. And it does the nice indenting and allows you to turn off that HTML crap... :) Anybody have any suggestions on a good email client for an Andriod device. A client that actually lets me set BCC or allows me to edit the original message when I replying? The built in client sucks!!! maildroid has a lot of features but kills your battery FAST. I only start it when I am expecting an important email and task kill it afterwards or it stays running. -- _ -- 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] Top Posting
On 01/02/2013 09:46 PM, jon pounder wrote: On 01/02/2013 03:22 PM, Patrick Lists wrote: On 01/02/2013 06:20 PM, Steve Totaro wrote: I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Just looked it up. I see my first post back in April 2003, yours in September 2003 and Jon in March 2003. Wow you find something fun to play with and suddenly a decade has passed :-) Are you sure about that ? I know I was doing stuff with asterisk back in the LSS days and that was around 2001 I only looked at the list archives. LSS definitely predates anything else so it's safe to say you are dinosaured in :-) http://lists.digium.com/mailman/listinfo/asterisk-users Here's to another decade of fun! Regards, Patrick -- _ -- 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] Speaking opportunities at Digium Asterisk World/IT Expo - Miami Beach - 1/31 and 2/1
Happy New Year. Digium Asterisk World is a set of conference sessions that run at IT Expo in Miami Beach, FL at the end of this month. See http://www.tmcnet.com/voip/conference/digium-asterisk-world/default.htm for more details. The conference runs 1/30-2/1, and we have a small number (2 or 3 ) of speaking slot available, over 1/31 and 2/1. This presents an opportunity for you to talk about your contribution(s) to the Asterisk project or describe your interesting Asterisk implementations - identifying you as a thought-leader in your field. Digium Asterisk World is more commercially focussed than AstriCon, and a proportion of the audience will be looking for solutions, rather than specifically wanting to get directly involved in the implementation itself - so this may provide a way for you to meet customers, or to show the community what you have been up to with Asterisk. The sessions must be educational and non-commercial. No pitches. Each session is 45 mins in duration, including time for Q and A. As a thank you for your successful speaking submission, we will give you an all-access pass to IT Expo, which includes all conference sessions (for all tracks at IT Expo) and conference meals. [You will need to make your own arrangements to get to the conference and for local accommodation, if needed - we will not fund or contribute to this]. Please get in touch with me directly if you would like to speak - sending a session title and description and a few lines about you, as a speaker. All the best, David Digium logo David Duffett Digium, Inc. · Director, Worldwide Asterisk Community 6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX · UK direct/fax: +1 256 428 6119 · mobile: +44 7722 442236 twitter: dduffett · linkedin: www.linkedin.com/in/davidduffett Check us out at: http://digium.com · http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto ban IP addresses
Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100sip:100@108.161.145.18;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? Thank you -- _ -- 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] Auto ban IP addresses
On Wed, Jan 2, 2013 at 3:49 PM, Frank fr...@efirehouse.com wrote: Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100sip:100@108.161.145.18;** tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? http://www.fail2ban.org/wiki/index.php/Asterisk -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Auto ban IP addresses
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Wednesday, January 02, 2013 4:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Auto ban IP addresses On Wed, Jan 2, 2013 at 3:49 PM, Frank fr...@efirehouse.com wrote: Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100sip:100@108.161.145.18 mailto:sip%3A100@108.161.145.18 ;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? http://www.fail2ban.org/wiki/index.php/Asterisk Fail2ban is a nice program, but deny=108.161.145.18 in sip.conf should satisfy OP's request. -- _ -- 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] RES: Auto ban IP addresses
Hi, Fail2ban http://en.gentoo-wiki.com/wiki/HOWTO_fail2ban -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Frank Enviada em: quarta-feira, 2 de janeiro de 2013 20:50 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: [asterisk-users] Auto ban IP addresses Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100sip:100@108.161.145.18;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? Thank you -- _ -- 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] RES: Auto ban IP addresses
Howto fail2ban in asterisk http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Frank Enviada em: quarta-feira, 2 de janeiro de 2013 20:50 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: [asterisk-users] Auto ban IP addresses Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100sip:100@108.161.145.18;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? Thank you -- _ -- 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] Telecom Best Practices
OK. I'm getting out the fireproof suit because it's coming and my hackles have been raised by a number of comments on the list of late. Disclaimer: No disrespect intended to the individuals of any *specific* thread. I'm a little frustrated over energy wasted on pedantic top/bottom posting crap rather than understanding the technology and industry best-practices which have been built upon for years. I'm not against change - far from it. I'm against throwing out good work and history done by an entire industry to make telecom one of the most complex and yet stable computing environments (class 4/5 entrants from Nortel, Lucent, etc.) We should learn from and extend best practices where they do not address circumstances which weren't available 20 years ago (or more) but not to ignore proven practices simply because the transport mechanism is now a packet instead of a circuit. I'm not alone. Here's the deal with Asterisk as an Answering Machine - industry best practices. - don't put phones in parallel with the pbx except for the single, emergency phone next to the PBX. - PBX's are directors of calls. For it to direct, it must have control. For it to have control, you can't answer some calls in parallel. - even if it is a home/1-phone-office, the PBX accepts and directs the call to phones *behind* it. The phone rings, if you don't answer it goes to voicemail. If you don't follow this practice you will have: - timing issues with the answering of analogue phones - rings are not always consistent. - people will pick up just in time and will have to compete with voicemail. - you won't get accurate CDR's which means you can do proper billing reconcilliation, chargebacks or help you understand your call paths and volumes to help troubleshoot down the road. (You may not care about bill reconcilliation or chargebacks but remember - this is a PBX (aka business phone system) and that's what business does so that's the business model that is supported by most practices. Just to prove I'm not too old for change and acceptance of new technology... - If you get charged by the connect from your provider, route by DID but don't answer it in an IVR. That way you don't get billed. - Once you are looking to route to a phone behind the PBX, hey - check your jabber status. Is your desktop in IDLE, you're not there - send it to your cell phone. Oh, BTW - change the CLID on the way back out to append H- to the caller so you know it came redirected from the house. This helps you decide context of the caller and decide if you want to answer or *how* you will answer. There is no reason to not have all phones behind the PBX. There is nothing mandating you to dial a 9, or similar to get an outside line. Be creative. Use internal extensions that don't conflict with your local calling area exchanges. Then you write dialplans for the phones that will dial right away and not make you wait to timeout on the 10-digit+ dial. There are *way* too many cool things we can do with Asterisk that worry about top/bottom posting. Let's get back to reading docs - asterisk industry practices. Fireproof suit on and buttoned up. I'm ready. -dbc -- _ -- 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] DAHDI: How to know since when it is used? How to shutdown after max time?
Hi; How can I know the duration that the DAHDI channel is still used? I need to know its status and since when it is in this status, how? Also, is it possible to hangup the channel if it has been openned more than 90 minute? Other than using the timeout in the Dial command (because this I know it). What is happening with me that from time to time, I find some DAHDI channels are stayed connected (stuck) for long time. I know how to write the extensions.conf in a way to handle the hangup properly, also I send the incoming calls to the voicemail to be sure it is hanged up properly. One more thing, I set the rtptimeout in case there is any problem in the sip phone and the network .. But, still after sometime, I am surprised that some channels are stuck and stayed connected and then I have to reset it manually !! This is happening only in the analoge channels. What other than the rtptimeout, the hangup in the extensions.conf, the voicemail? Is there anything I have to take care for it that might cause this stuck and keeping the channel openned? By the way, for such cases, what should I place the value of the rtpkeepalive as currently it is 0? What other things I have to take care for it? Regards Bilal -- _ -- 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 for Razberry Pi
I've created some images. I currently don't have a free Raspberry Pi so I have not updated any images for a little while. A how to on building your own. www.klaverstyn.com.au/david/wiki/index.php?title=Asterisk_for_Raspberry_Pi A how to on writing a pre-compiled image http://www.klaverstyn.com.au/david/wiki/index.php?title=Asterisk_for_Raspber ry_Pi_Image A list of existing images for the Raspberry Pi http://www.klaverstyn.com.au/david/asterisk/rpi/ This may help some people. Regards David. -- _ -- 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] Paging for Praying
Thanks for the help. As I see that the call file is used to generate calls, can I use this technique to page the Phones? It is one wave file only that need to be Paged for all the Phones connected on the Asterisk PBX. When I say Paging, I mean that they are going to hear the sound from the speaker (without pickup the handset). By using AMI, then I can build PHP script that will use the AMI to do the Page? Thanks and Regards Bilal A call file is a text file that you create. The format is very specific. On Tue, 1 Jan 2013, bilal ghayyad wrote: * How can I know this format? Because I need to know what should I place in this file so it will execute Paging for this group of Phones? This may help: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out How many customers will be receiving these reminders? * It is required that all the employers at the company to hear this on their IP Phones. In my experience, you can't just dump xxx call files into the outgoing directory. If you expect more than a dozen or so, you'll have to move them in blocks as they are processed. Another good reason to use AMI. You can 'schedule' a call file to be processed in the future by setting the file's 'mtime.' * Can you explain for me please? Create a file named fajr containing: application: playback channel: sip/bilal data: fajr-in-10-minutes Copy the file to a directory we assume is on the same file system as /var/spool/asterisk/outgoing/: cp\ fajr\ /var/spool/asterisk/tmp/ Set the file's 'mtime' touch\ --date='now + 2 minutes'\ --time=mtime\ /var/spool/asterisk/tmp/fajr Move it to the outgoing directory: mv\ /var/spool/asterisk/tmp/fajr\ /var/spool/asterisk/outgoing/ Your phone should ring in about 2 minutes. You may want to look into setting 'auto-answer' or some sort of 'overhead paging' with a very discreet sound file like a short, single beep. Please consider AMI if you are looking for a robust service. -- _ -- 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] Paging for Praying
Please trim cruft irrelevant to the current questions.
On Wed, 2 Jan 2013, bilal ghayyad wrote:
As I see that the call file is used to generate calls, can I use this
technique to page the Phones?
Yes. The call file would look something like:
application:page
data: sip/bilalsip/steve
channel:local/fajr@prayer-reminder
and a snippet of extensions.conf would look something like:
[prayer-reminder]
exten = fajr,1, verbose(1,[${EXTEN}@${CONTEXT}])
exten = fajr,n, playback(time-for-fajr)
exten = fajr,n, hangup()
When I say Paging, I mean that they are going to hear the sound from the
speaker (without pickup the handset).
Then you will have to learn how to set the 'auto-answer' SIP header for
each of your phone types.
The page at:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Page
should help with the specifics of setting auto-answer.
This also means the 'data' line in the call file will change to something
like:
data: local/bilal@pagelocal/steve@page
By using AMI, then I can build PHP script that will use the AMI to do
the Page?
I'm sorry. I thought we had a 'failure to communicate' in your use of the
word page in your previous emails. I though you meant playback.
Since you are only placing a single call, the call file approach should be
fine. But, to answer your question, you should be able to use AMI as well.
--
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] User busy issue in A400P 4 FXO card
Hi, I setup PBX with A400P 4 x FXo board. There are one analog line plugged into port 1. Internal extension cane make calls to PSTN without any issue. When I make inbound call, caller get busy tone user busy' message right away. Asterisk log shows following log and internal extension (200) rings for that call and hangup (log below). I tested the system with some other service provider and it worked fine for IB and OB calls. i would like to get your feedback to resolve the issue and will appreciate your feedback. Thanks Selva [PBX1.localdomain dahdi]# lsdahdi ### Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 FXOFXSKS (In use) (SWEC: MG2) 2 FXOFXSKS (In use) (SWEC: MG2) RED 3 FXOFXSKS (In use) (SWEC: MG2) RED 4 FXOFXSKS (In use) (SWEC: MG2) RED PBX1*CLI dahdi show regdump 1 Direct registers: 0. 001. 002. 033. 004. e95. 086. 007. 00 8. 009. 00 10. 00 11. 35 12. 40 13. 58 14. 00 15. 00 16. 01 17. 02 18. 00 19. 02 20. 00 21. 00 22. 96 23. 2d 24. 19 25. 08 26. c0 27. 05 28. 00 29. 34 30. 00 31. a3 32. 07 33. 28 34. 18 35. 00 36. 18 37. 00 38. 10 39. 10 40. 10 41. 10 42. 00 43. 00 44. 00 45. 00 46. 00 47. 00 48. 00 49. 00 50. 00 51. 00 52. 00 53. 00 54. 00 55. 00 56. 0f 57. 00 58. 00 59. 00 TIP: 52. Volts RING: 52. Volts VBAT: 52. Volts PBX1*CLI dahdi show regdump 2 Direct registers: 0. 001. 002. 033. 004. 295. 086. 007. 00 8. 009. 00 10. 00 11. 35 12. 40 13. 58 14. 00 15. 00 16. 01 17. 00 18. 00 19. 02 20. 00 21. 00 22. 96 23. 2d 24. 19 25. 08 26. c0 27. 07 28. 00 29. 01 30. 00 31. a3 32. 07 33. 28 34. 10 35. 00 36. 10 37. 00 38. 10 39. 10 40. 10 41. 10 42. 00 43. 00 44. 00 45. 00 46. 00 47. 00 48. 00 49. 00 50. 00 51. 00 52. 00 53. 00 54. 00 55. 00 56. 0f 57. 00 58. 00 59. 00 TIP: 1. Volts RING: 1. Volts VBAT: 1. Volts - -- Starting simple switch on 'DAHDI/1-1' -- Executing [s@from-pstn:1] Set(DAHDI/1-1, __FROM_DID=s) in new stack -- Executing [s@from-pstn:2] Gosub(DAHDI/1-1, app-blacklist-check,s,1) i n new stack -- Executing [s@app-blacklist-check:1] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack -- Executing [s@app-blacklist-check:2] Return(DAHDI/1-1, ) in new stack -- Executing [s@from-pstn:3] ExecIf(DAHDI/1-1, 1 ?Set(CALLERID(name)=)) in new stack -- Executing [s@from-pstn:4] Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_ screened) in new stack -- Executing [s@from-pstn:5] Set(DAHDI/1-1, CALLERPRES()=allowed_not_scre ened) in new stack -- Executing [s@from-pstn:6] Goto(DAHDI/1-1, from-did-direct,200,1) in n ew stack -- Goto (from-did-direct,200,1) -- Executing [200@from-did-direct:1] Macro(DAHDI/1-1, exten-vm,200,200) in new stack -- Executing [s@macro-exten-vm:1] Macro(DAHDI/1-1, user-callerid) in new stack -- Executing [s@macro-user-callerid:1] Set(DAHDI/1-1, AMPUSER=) in new s tack -- Executing [s@macro-user-callerid:2] GotoIf(DAHDI/1-1, 0?report) in ne w stack -- Executing [s@macro-user-callerid:3] ExecIf(DAHDI/1-1, 1?Set(REALCALLER IDNUM=)) in new stack -- Executing [s@macro-user-callerid:4] Set(DAHDI/1-1, AMPUSER=) in new s tack -- Executing [s@macro-user-callerid:5] Set(DAHDI/1-1, AMPUSERCIDNAME=) i n new stack -- Executing [s@macro-user-callerid:6] GotoIf(DAHDI/1-1, 1?report) in ne w stack -- Goto (macro-user-callerid,s,10) -- Executing [s@macro-user-callerid:10] GotoIf(DAHDI/1-1, 0?continue) in new stack -- Executing [s@macro-user-callerid:11] Set(DAHDI/1-1, __TTL=64) in new stack -- Executing [s@macro-user-callerid:12] GotoIf(DAHDI/1-1, 1?continue) in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] NoOp(DAHDI/1-1, Using CallerID ) in new stack -- Executing [s@macro-exten-vm:2] Set(DAHDI/1-1, RingGroupMethod=none) i n new stack -- Executing [s@macro-exten-vm:3] Set(DAHDI/1-1, VMBOX=200) in new stack -- Executing [s@macro-exten-vm:4] Set(DAHDI/1-1, EXTTOCALL=200) in new s tack -- Executing [s@macro-exten-vm:5] Set(DAHDI/1-1, CFUEXT=) in new stack -- Executing [s@macro-exten-vm:6] Set(DAHDI/1-1, CFBEXT=) in new stack -- Executing [s@macro-exten-vm:7] Set(DAHDI/1-1, RT=15) in new stack -- Executing [s@macro-exten-vm:8] Macro(DAHDI/1-1, record-enable,200,IN) in new stack -- Executing [s@macro-record-enable:1] GotoIf(DAHDI/1-1, 1?check) in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] AGI(DAHDI/1-1, recordingcheck,2013 0103-002934,1357190966.4) in new stack --
