internal extension calling remote staff extension

areid

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#1
This is using elastix 1.6.2-x. When I make outside calls, works. but when I call from my extension to another extension (remote), both of us cut in and out. And only can hear parts what the other person is saying.

How do I solve this?
 

trymes

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#2
We're going to need more information here.

What sort of phones and extensions are you using? Dahdi? SIP? Softphone or hardphone?

What sort of trunk are you dialing out on when it works well? Dahdi, SIP, IAX?

Tom
 

areid

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#3
both extensions are sip

remote extension is a softphone

internal extensions, hardphone and softphone. both softphones are using same software and version.

dialing out sip
 

dicko

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#4
That sounds like a network problem,

try

rtp debug

at the asterisk command line, if the the rtp (audio) packets are not rigorously Got, Sent, got, sent without change during a call then you need to investigate the network.
 

areid

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#5
ok. I will try that but I did another test.

outside call to the remote staff extension. It was doing the same thing cutting in and out.

when outside calls come to internal extensions, it works fine.

any suggestions?
 

dicko

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#6
I can but reiterate:-

That STILL sounds like a network problem,

try

rtp debug

at the asterisk command line, if the the rtp (audio) packets are not rigorously Got, Sent, got, sent without change during a call then you need to investigate the network.

check that nobody at the remote locations is downloading porn flicks :)
 

areid

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#7
I did asterisk -r and rtp debug and got this

Got our ip
Send remote staff ip
Send 10.0.0.200 (not sure what this is)

more like "got send send". the first call was "got send got sent" and had a few "sent" in a row.

Is there anything we need to do for our router here? our remote staff is looking at his network.
 

dicko

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#8
got got . . suggests your router is having problems or your asterisk box is over stressed, or you are merely catching buffered but un-lost packets, check the sequence and time-stamp fields for that decision.

send send . . the most often encountered scenario suggests the far end, but again check seq and ts for diagnosis.

but either way, more than the occasional two in a row, of either, will spoil your whole day, sorry, you almost certainly have a network based problem, not an Elastix/Asterisk one.

As to router configuration, an over extended network connection without QOS/TOS properly deployed will never be satisfactory. . . . EVER, and even then you will have to "do your sums"

dicko

As a comment , each rtp packet is conventionally 20 milliseconds of audio, psych-acoustic studies suggests that temporal disjunction of more than 100 ms in that channel will start to cause user stress, and that THAT user stress will increases exponentially by the millisecond. At about 300 msecs, the conversation has deteriorated into 'overspeaking', and then shouting (which of course won't help) and "I can't hear you, WTF I'll call you from a LandLine", when you get to that point, your VOIP solution should be considered a failure :)

p.s. and from a re-read, you should rtp debug ONLY if one call is going on to start with, rtp debug ip <remote-ip> will isolate the one call (unless you have two calls to that router :), either way the debug info remains in /var/log/asterisk/full(*) and can be grepped for better post-mortem forensics.
 

areid

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#9
I called our router manufacturer to confirm configuation with QOS and port priority. They said what I did is correct. I did some more tests, internal extensions to landlines and see "got send send" and "got send got send".

Our router manufacturer mentioned that on our pbx system it is codex that needs to be configured. Any suggestions? They also said I should downlonad wireshark.

Our internal network is not a demanding one has alot of bandwidth.
 

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