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The Wonderfull World Of SIP!!! 1

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jamie77

Programmer
Oct 15, 2004
4,523
GB
Is this how it is supposed to work or not???

Dial out over SIP. IPO picks up a VCM. When ringing starts the IPO picks up a second VCM. When the call is answered it drops back to 1 VCM.

I/C calls only ever use 1 VCM.

We have 4 SIP trunks and only 4 VCM's so this is causing a major problem.

Does this mean to guarantee everyone can dial out at any time we need twice as many VCM's as SIP trunks???

IP500 4.0.5.

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office


Fooball is not a matter of life and death-It is far more important!!!!
 
This looks to me like a codec thing? Are you dialling out from an IP Handset by any chance?



 
No from DS phones. Codec set to G729 as requested by the ITSP. I was wondering if it was to do with the RTP routing.

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office


Fooball is not a matter of life and death-It is far more important!!!!
 
The use of 2 VCM Channels during setup, would make sense. Initially you're using one VCM to provide the audio to the handset calling out, then you're using a second VCM channel to provide the ringback tone to the station, another TDM-IP Connection, basically a conference between TDM (Station)<-> IP (SIP Trunk) <-> TDM (ringback).

then when the call is answered the "ringback" drops from the VCM and you have direct TDM <-> IP.

>mike


ACA ; ACS
 
No the way yours is working is not correct.

IP500 v 4.0.5 with a sip trunk to Gamma.

Uses just one vcm channel when i make a call.

I have just tried a 5610 in te same scenario and that does indeedy use two channels and the drops back to one.

Are you 100% you (or your customer who usually talks parp) are not using an IP handset?

Forgive my low opinion of customers ability to accurately describe a situation.
 
This is on our own IP500. My phone is a 5410 and I have SA running on my PC while I do this testing.

I am setting up my test 4602 to try it from an IP set.

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office


Fooball is not a matter of life and death-It is far more important!!!!
 
What SIP provider are you using?

I have been trying some different "non supported" SIP providers and have been finding various issues, not like this i have to say, but wierd issues where by the SIP provider dos not support the same RFC that the IPO uses.

We have one sip provider that works but you cannot pass dtmf down the trunk from an ip handset but it works from a ds. Just weird stuff that doesnt make much sense.
 
***UPDATE***

If I dial out from an IP phone (4602) O/G calls DO NOT use a VCM at any time. I/C calls use a VCM for setup and then it is dropped on connection!!!!!!! The IP phone is set to G711 and the lines are G729!!!!

I/C calls to a DS set (or Sets) only ever use 1 VCM.

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office


Fooball is not a matter of life and death-It is far more important!!!!
 
That does not make any sense.

If your ip handset is using one codec and your sip trunk is using another there needs to be qa codec negotiation that is performed by the vcm.
 
That is what I thought!! But that is what is happening.

It's all a bit wierd!!!

I am waiting for a call back from Tech Support.

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office


Fooball is not a matter of life and death-It is far more important!!!!
 
Get this then.

5610 set to g711 and sip trunk set with same codec, this is a supported codec by my sip provider.

Seeing as the IP extn is set to direct media i would fully expect all vcm channels to be dropped once the call is established.

Nah. Two channels used.
 
Good news for you Jamie. Well kind of!!!!!!!!!!!!

There is a bug here that is not on the caveats website yet.

Despite RTP relay being introduced in v4, it is broken.

In my scenario above i have a 5610 with direct media on and g711 alaw. A sip trunk set to g711 that the sip provider supports.

If i make a call on my ip phone i take a vcm channel and then i take another when i start dialling over the sip trunk.

I would expect both channels to drop when the call is established because i am using the same codec throughout.

Not so, it uses two. RTP relay should handle the voice but doesnt due to the fact that it is nadgered. Its fixed in the next release, June 15th.
 
Cool.

I will get Tech Support to quiz Avaya if that is part of the issue that I am having.

Good man.

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office


Fooball is not a matter of life and death-It is far more important!!!!
 
New word of the day for everyone

NADGERED

You get a star for quality use of the English language!!!!

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office


Fooball is not a matter of life and death-It is far more important!!!!
 
MTV Telecom

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office


Fooball is not a matter of life and death-It is far more important!!!!
 
How do they work? Do you have to log it and they phone you back? How quickly do they respond etc?
 
This got logged and got a call back. Normally I ring them and hold in the queue. Normally pretty good and the guys there are generally pretty good.

We also get some support from Crane. Log a fault and they phone back.

I prefer to hold for an engineer as you have no idea how long they are going to be before you get a call back and sometimes time is something there is little of!!!!!

Who does your support??

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office


Fooball is not a matter of life and death-It is far more important!!!!
 
Nobody. I work for a distributor, not a reseller.

I give, not recieve, so to speak.


 
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