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We want to record inbound channels at 16kHz, but send only 8kHz to our peers. I've set our default profile in sip.conf to disallow all but g722, and the peers disallow all but ulaw. We have a proxy in front of Asterisk that is configured to disallow all but G722 also. My test calls show inbound to the proxy is recorded at 16kHz, inbound in Asterisk is only 8kHz, and the peers receive 8kHz. So

Response below... On Fri, Dec 27, 2019 at 12:02 PM David P <davidswalkabout at gmail.com> wrote: > > > > > I'm looking for a way of detecting in my dialplan when a peer becomes > > non-responsive after answering. [deleted] Is there a way to configure > > a handler for this state? > > > > We use v14.7.6 and we dial the peer this...

We tried to use the CURL fn to POST json, but it's sent as form data and there seems no support for changing the Content-Type header. We switched to invoking curl in the shell. All the documentation I could find says there is just one parameter for the url and an optional second for POST body. Is there an undocumented way to set Content-Type? -------------- next part -------------- An HTML

This has been super-helpful, Eric. However, the handleHangupByPeer priorities below are still not run when the peer hangs-up. The last line in the cli when the peer hangs-up is still: Strict RTP learning complete - Locking on source address (Although sometimes there is also: Retransmission timeout reached on transmission) same =>

I'm looking for a way of detecting in my dialplan when a peer becomes non-responsive after answering. It seems that Asterisk knows when the peer becomes non-responsive because it logs "Remote UNIX connection disconnected" around the same time, and it seems that if there is no following "Remote UNIX connection" within a short time, then the peer can be considered

Please point me to samples of popping and wiping hangup handlers. I don't need to use the values returned; I just need to clear any handlers before I push a new one. It's not clear at https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers+Specification how to provide vars on the right-hand side. Cheers, David -------------- next part -------------- An HTML attachment was scrubbed...

It might work for you to branch on ${DIALSTRING} just after your Dial command, if you want to handle a BUSY, NOANSWER, or other result. But if the peer of that Dial hungup, then based on what Joshua said, it seems there's no recovery. -------------- next part -------------- An HTML attachment was scrubbed... URL:

We use Asterisk 14 to proxy calls between two servers, 10.0.0.192 to 10.0.0.228. But sometimes another of our servers becomes listed as a SIP agent, even though the server's IP address isn't part of our sip.conf, extensions.conf, nor any other config I know of. For example in the log snippet below, the source server experienced an SDP renegotiation in the middle of a call, and seemingly as

We're using Asterisk 14.7.6 and I have a dialplan that ends like this: same => n,Dial(SIP/${EXTEN:0:4}@peer1) same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH}) same => n,Hangup() When peer1 hangsup, the priorities after the Dial are executed fine. But when the caller hangsup during the Dial, the cleanup steps aren't done. Why? I did read "Note that on a successful

Thanks, Anthony. I added both 'g' and 'F' options. Now, when the caller hangs-up, my cleanup code is run by both the caller channel and the peer channel, but I only want the caller channel to do that. Also, when the peer hangs-up, there is no execution of the priorities following the Dial. Finally, is there a way to reset all globals, maybe as a variant of "dialplan

Thanks, Eric. I just tried a hangup handler, but it's showing a similar problem: When the peer hangs-up, the hangup handler is not invoked and the caller channel remains open. same => n(callPeer),Set(GLOBAL(Peer${IndexIntoPeers}CurrentCallsCount)=$[${PeerCurrentCallsCount} + 1]) same => n,Set(CHANNEL(hangup_handler_push)=handleHangupByCallerOrPeer,doesntMatter,1(args)) same =>