[asterisk-users] One-side Audio on Bridged Calls With Both Legs Crossing NAT Device
Hi, Let me provide the details first: * Asterisk 1.8.32 on CentOS behind the NAT firewall * Two (2) SIP trunks with "canreinvite=no" and "directmedia=no" If a call comes from either trunk and is bridged to a local extension there is never a problem with audio. The same is true for outbound calls on either trunk. If an incoming call from Trunk A is forwarded to Trunk B there is a large percentage of the one-side audio calls. Has anybody run into this kind of a situation? Thank you, Vladimir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP Timestamp rewind
Hi folks. I have a couple of questions regarding RTP. The background of my inquiry is that I have packet captures of SIP and RTP traffic on an Asterisk and Broadworks SIP trunk and the RTP many times has a time stamp that rewinds by 480 using g.711u. The Sequence number continues to increment appropriately, but the timestamp just rewinds. It doesn't happen on every call, but it's frequent enough to make me want to understand it better. My questions are: Is there ever a circumstance where it would be normal or logical to see the RTP timestamp go backwards during the RTP stream? Consistently by 480, 3 voice frames? Will Asterisk just drop the packets that compromise the rewind? Thanks Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What version of Linux?
Hello; Have you run the script that's included in the Asterisk distribution that lists and installs the needed dependencies? It's called "install_prereq" and it's in the contrib/scripts directory. Hope this helps. Regards; John V. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira Sent: Monday, August 28, 2017 03:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] What version of Linux? Hello Asterisk, I've been running CentOS since 2006 or so and support for the 32 bit version recently ended. CentOS no longer offers a 32 bit version so I thought I'd try Fedora 26 as they have 32 bit and support. Got it installed, then downloaded Asterisk 14.6.0 but can't seem to get it built. The configure script fails with some error about CPP not working correctly? I did discover that kernel-devel was not installed so I fixed that but I'm still stuck. Is the latest Fedora a good choice for an Asterisk box or should I try something else. The machine is an Intel Atom board with a Digium PCI analog board for my one last analog line. I believe the board is limited to a 32 bit OS. So two questions, is Fedora a good choice and if not, what should I use for a machine running only Asterisk and Samba? Is there a list of dependencies I need to install before Asterisk will compile? Thanks, Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UNIQUEID not unique in different channels
On Tue, Aug 29, 2017, at 10:57 AM, Thomas wrote: > Hello, > > since al long time I have used UNIQUEID for identify calls in my > dialplan, > statistics... > > Now I have had an problem, after I have checked log file I found out > following: > > calls same time ( hours:seconds) came in. > > CallID, DID, channel name (3cf9 to 3cfa) are different. > > Only UNIQUEID is identical > > Is there any known bugs or ist UNIQUEID not unique and can not be used > for > identify calls? There's no known bugs and it is absolutely supposed to be unique. I'd suggest filing an issue[1] with logs and details. Also ensure you are using a currently supported version of Asterisk. [1] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UNIQUEID not unique in different channels
Hello,
since al long time I have used UNIQUEID for identify calls in my dialplan,
statistics...
Now I have had an problem, after I have checked log file I found out following:
calls same time ( hours:seconds) came in.
CallID, DID, channel name (3cf9 to 3cfa) are different.
Only UNIQUEID is identical
Is there any known bugs or ist UNIQUEID not unique and can not be used for
identify calls?
[Aug 28 10:14:11] VERBOSE[6413][C-1f1e] pbx.c: -- Executing
[XX@patton4970in:1] NoOp("SIP/outbound.patton-3cf9", "INFO:
eingehender Anruf Context patton4970in von
SIP/outbound.patton-3cf9 "" zu XX") in new stack
UNIQUEID: 1503908051.54095
[Aug 28 10:14:11] VERBOSE[6418][C-1f1f] pbx.c: -- Executing
[XX@patton4970in:1] NoOp("SIP/outbound.patton-3cfa", "INFO:
eingehender Anruf Context patton4970in von
SIP/outbound.patton-3cfa "" zu XX") in new stack
UNIQUEID: 1503908051.54095
thanks..
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
