Re: [asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?

2019-12-27 Thread Greg Troxel
"Joshua C. Colp" writes: >> I am curious if the "reuse registration TCP connection" is required by >> standards or if it is merely obviously good practice. >> >> I have had this problem too with asterisk 16.5.0 >> >> This is not the first recommendation I have seen to use kamailio as a >> proxy

Re: [asterisk-users] AGI: "Get variable" returns variable VALUE vs "Get full variable" returns variable NAME - bug or my misunderstanding?

2019-12-27 Thread Sean Bright
On 12/27/2019 2:56 PM, Jonathan H wrote: OK, that works - looks like a documentation bug? (Also very confusing!) Should I report it on the page at https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+AGICommand_get+full+variable or on the main tracker? These wiki pages are automatically

Re: [asterisk-users] AGI: "Get variable" returns variable VALUE vs "Get full variable" returns variable NAME - bug or my misunderstanding?

2019-12-27 Thread Jonathan H
OK, that works - looks like a documentation bug? (Also very confusing!) Should I report it on the page at https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+AGICommand_get+full+variable or on the main tracker? Many thanks - couldn't figure out what was wrong! On Fri, 27 Dec 2019 at 19:36,

Re: [asterisk-users] AGI: "Get variable" returns variable VALUE vs "Get full variable" returns variable NAME - bug or my misunderstanding?

2019-12-27 Thread Sean Bright
On 12/27/2019 2:24 PM, Jonathan H wrote: AGI Rx << SET VARIABLE myVar "Hello World!!!" AGI Tx >> 200 result=1 AGI Rx << GET FULL VARIABLE myVar AGI Tx >> 200 result=1 (myVar) Is this a bug, poor documentation, or my poor understanding of them? I believe the syntax you are looking for is:    

[asterisk-users] AGI: "Get variable" returns variable VALUE vs "Get full variable" returns variable NAME - bug or my misunderstanding?

2019-12-27 Thread Jonathan H
Just trying out a node agi package (https://github.com/sergey12313/ts-agi/ , and it wasn't behaving as I expected, but when turning on agi debug, it looks like it might be Asterisk (using 17.1.0) This works as expected AGI Rx << SET VARIABLE myVar "Hello World!!!" AGI Tx >> 200 result=1 AGI Rx

Re: [asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?

2019-12-27 Thread Joshua C. Colp
On Fri, Dec 27, 2019 at 2:00 PM Greg Troxel wrote: > Dovid Bender writes: > > > So long as the tcp socket is open your SBC should send the call back over > > the same socket. Now it can be that your SBC is seeing the socket as > > timing out. If you are using Kamailio you can have it send tcp

Re: [asterisk-users] Handling a non-responsive peer after it answers

2019-12-27 Thread Joshua C. Colp
On Fri, Dec 27, 2019 at 12:02 PM David P wrote: > > 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

Re: [asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?

2019-12-27 Thread Greg Troxel
Dovid Bender writes: > So long as the tcp socket is open your SBC should send the call back over > the same socket. Now it can be that your SBC is seeing the socket as > timing out. If you are using Kamailio you can have it send tcp keep alives > every so often so that the socket stays up. SBC?

Re: [asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?

2019-12-27 Thread Dovid Bender
So long as the tcp socket is open your SBC should send the call back over the same socket. Now it can be that your SBC is seeing the socket as timing out. If you are using Kamailio you can have it send tcp keep alives every so often so that the socket stays up. On Fri, Dec 27, 2019 at 10:41 AM

[asterisk-users] Handling a non-responsive peer after it answers

2019-12-27 Thread David P
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

[asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?

2019-12-27 Thread Benoit Panizzon
Hi List I wonder how SIP via TCP is supposed to work. Not realy Asterisk related, but I hope you experts might be able to help out :-) One of our customers has a SIP device registering via a complex NAT. To benefit from TCP Connection Tracking, he choose TCP instead of UDP. So he expected, that