Re: [asterisk-users] Need more meetme users -- hitting some limit

On Mon, 31 Mar 2014, Shaun Ruffell wrote: If you're looking to reduce the CPU overhead of processing meetme conferences, this email from awhile ago may be of some help: http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/51750/focus=51777 Thanks for the clue. I can hit my target o

Re: [asterisk-users] Unable to build DAHDI-Linux in mock chroot

On Sunday, March 30, 2014 02:24:35 PM Anthony Messina wrote: > On Sunday, March 30, 2014 07:07:47 PM Tzafrir Cohen wrote: > > On Fri, Mar 28, 2014 at 07:57:54PM -0500, Anthony Messina wrote: > > > On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote: > > > > Unfortunately, after > > > > > >

Re: [asterisk-users] Need more meetme users -- hitting some limit

On Fri, Mar 21, 2014 at 11:26:22AM -0700, Steve Edwards wrote: > On Fri, 21 Mar 2014, Steve Totaro wrote: > > >I found below here: > > http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe > > > >If you have too many conferences, one CPU may not be able to mix all the > >audio and you will have a

[asterisk-users] DAHDI-Linux v2.9.1.1 Now Available

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.1.1 dahdi-linux-complete-2.9.1.1+2.9.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete Fix f

Re: [asterisk-users] CLI command to see if SRTP is active?

On Mon, Mar 31, 2014 at 1:26 PM, Rusty Newton wrote: > On Fri, Mar 28, 2014 at 7:39 PM, Patrick Laimbock > wrote: >> Hi, >> >> I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI >> command to see if SRTP is active on a channel/call. I went through sip show >> ... and core show

Re: [asterisk-users] CLI command to see if SRTP is active?

On Fri, Mar 28, 2014 at 7:39 PM, Patrick Laimbock wrote: > Hi, > > I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI > command to see if SRTP is active on a channel/call. I went through sip show > ... and core show channel... and did not see any mentioning of SRTP while > there

Re: [asterisk-users] Function REGEX

All working fine. Thank you for your help. Att, *Rafael dos Santos Saraiva* 2014-03-31 12:29 GMT-03:00 Eric Wieling : > Here is an example from one of my production dialplans > > same => > n,ExecIf(${REGEX("^1205|^1256|^1850|^1718|^1212|^1

Re: [asterisk-users] Function REGEX

Here is an example from one of my production dialplans same => n,ExecIf(${REGEX("^1205|^1256|^1850|^1718|^1212|^1917|^1347|^1646|^1929" ${CALLERID(num)})}]?Hangup) Assuming you meant 0-9 and not the literal X (which means nothing special in regular expressions): same => n,ExecIf(${REGEX("^[0-

[asterisk-users] Function REGEX

Hi I need help to use the function REGEX. My question is if is possible test a expression as [X123 == 5123] ( If an extension corresponding to a previously defined regular expression). I saw various examples about this function, but nothing as the my needs. I do not understanding exactly how to w

Re: [asterisk-users] Video calls using Cisco phones are 176x144(QCIF) and 15FPS both ways

Matt Rabbitt wrote: What would need to be changed in the source code to accommodate this? Can the imageattr attribute be hard coded into h264_format_attr_sdp_generate() in res_format_attr_h264.c? A lot. Yes, you could hard code it. -- Joshua Colp Digium, Inc. | Senior Software Developer 445

Re: [asterisk-users] Video calls using Cisco phones are 176x144(QCIF) and 15FPS both ways

What would need to be changed in the source code to accommodate this? Can the imageattr attribute be hard coded into h264_format_attr_sdp_generate() in res_format_attr_h264.c? On Mon, Mar 31, 2014 at 9:07 AM, Joshua Colp wrote: > Matt Rabbitt wrote: > >> We are experiencing an issue with our C

Re: [asterisk-users] Video calls using Cisco phones are 176x144(QCIF) and 15FPS both ways

Matt Rabbitt wrote: We are experiencing an issue with our Cisco 9971 and 8945 phones where H264 video calls are connecting at 176x144 resolution instead of 640x480. Soft clients can connect at higher resolutions and the 9971 can even receive video at a higher resolution (although it still sends

[asterisk-users] Video calls using Cisco phones are 176x144(QCIF) and 15FPS both ways

We are experiencing an issue with our Cisco 9971 and 8945 phones where H264 video calls are connecting at 176x144 resolution instead of 640x480. Soft clients can connect at higher resolutions and the 9971 can even receive video at a higher resolution (although it still sends 176x144). I contacted

Re: [asterisk-users] IAXModem or T38Modem?

> "j" == joakimsen writes: j> I wouldn't mind if someone posted on the list a known working provider j> with the proper configuration to use T.38. In my case I don't consider j> it an issue with the provider because they sent the proper T.38 j> Invite, but Asterisk IMO does not know how to h

Re: [asterisk-users] Duplicate incoming channel into two outgoing channels

On 27.03.2014 10:39, jg wrote: Wouldn't it make more sense to handle this by just monitoring the calls and doing everything else with normal data processing? Basically yes, but the whole idea is a workaround to fix issues in legacy systems. klaus -- ___