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Calls occasionally cutting out on 2820n
- simondo
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25 Nov 2010 15:30 #65020
by simondo
Replied by simondo on topic Calls occasionally cutting out on 2820n
I'm not too sure how to read it, but for the record, we're moving on from gradwell and bringing the server in house; asterisk by way of piaf and freepbx has been painless to set up, full of options, and we've noticed an increase in call quality...
The call costs with a sip trunk look very reasonable too.
The call costs with a sip trunk look very reasonable too.
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- faris
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25 Nov 2010 19:32 #65021
by faris
Replied by faris on topic Calls occasionally cutting out on 2820n
Thanks Simon,
That's what we do with Gradwell already.
We have an incoming trunk from them into our co-located Asterisk box for incoming calls, and have another trunk for outgoing.
The annoying thing is that we have had no noticable problems with another VoIP trunk supplier (I can't say who -- we are one of their wholesale distributors so it would be wrong to do so).
You might think that points the finger firmly at Gradwell, but I don't think they are at fault at all. Rebooting the router solves the problem for a while, for example, and when listening to recordings of the calls made on our Asterisk server, both sides of the conversation can clearly be heard (mostly consisting of "hello? Hello? Can you hear me?/Yes I can hear you but you can't seem to hear me" sort of thing).
So that kind of points to the SIP phone to Asterisk side, except that the one-way audio can be in either direction, so both sides reach the server. So that kind of points to the Asterisk server itself, except that when using the other VoIP trunk supplier we never have any problems at all, as previously mentioned, and we are back full circle.
And on top of all that, in all of the above cases there's a potential additional "but.." or "except..." that could implicate something else as the possible culprit.
Bah! Maybe our problem is just a particularly unfortunate combination of phones, router, trunk supplier and Asterisk version which happens not to work in harmony for some reason.
At least it happens very infrequently at the moment, especially as long as I remember to do weekly router reboots.
Faris.
That's what we do with Gradwell already.
We have an incoming trunk from them into our co-located Asterisk box for incoming calls, and have another trunk for outgoing.
The annoying thing is that we have had no noticable problems with another VoIP trunk supplier (I can't say who -- we are one of their wholesale distributors so it would be wrong to do so).
You might think that points the finger firmly at Gradwell, but I don't think they are at fault at all. Rebooting the router solves the problem for a while, for example, and when listening to recordings of the calls made on our Asterisk server, both sides of the conversation can clearly be heard (mostly consisting of "hello? Hello? Can you hear me?/Yes I can hear you but you can't seem to hear me" sort of thing).
So that kind of points to the SIP phone to Asterisk side, except that the one-way audio can be in either direction, so both sides reach the server. So that kind of points to the Asterisk server itself, except that when using the other VoIP trunk supplier we never have any problems at all, as previously mentioned, and we are back full circle.
And on top of all that, in all of the above cases there's a potential additional "but.." or "except..." that could implicate something else as the possible culprit.
Bah! Maybe our problem is just a particularly unfortunate combination of phones, router, trunk supplier and Asterisk version which happens not to work in harmony for some reason.
At least it happens very infrequently at the moment, especially as long as I remember to do weekly router reboots.
Faris.
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