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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!!!!
 
I can take a good guess at who that is. Are you checking up on the competition???????

One of our guys did his IPO Implement course there last summer.



Jamie Green

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


Fooball is not a matter of life and death-It is far more important!!!!
 
Wrong side of the Pennines by the way!!!!! ;-)

Jamie Green

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


Fooball is not a matter of life and death-It is far more important!!!!
 
Yes i am just seeing how the dark side works.

There is no right side of the pennines.

In my experience it rains on both sides.

If the truth be known we are all the wrong side of the english channel.

 
Both sides and even more in the middle!!!!!!!!!

Not a fan of the other side of the Channel. French phone systems have always caused me more issues than even the IPO (Alcatel Omni)!!!!!!!!!!!!!1

Jamie Green

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


Fooball is not a matter of life and death-It is far more important!!!!
 
I think the issue my be to do with the SIP message being passed back and forth.

From my Ethereal traces

OG call DS to SIP:
INVITE SDP - IPO>ITSP
100 Trying - ITSP>IPO
180 Ringing SDP (g729 telephone event) - ITSP>IPO
RTP - ITSP>IPO (Open for 7.721s)
RTP - IPO>ITSP (Open for 1.939s)
200 OK SDP (g729 telephone event) - ITSP>IPO
ACK - IPO>ITSP
RTP - IPO>ITSP (Open for 5.719s)
BYE - ITSP>IPO
200 Ok - IPO ITSP


I/C call SIP to DS:
INVITE SDP - ITSP>IPO
100 Trying - IPO>ITSP
180 Ringing - IPO>ITSP
200 Ok SDP (g729 telephone event) - IPO>ITSP
RTP - IPO>ITSP (Open for 6.340s)
ACK - ITSP>IPO
RTP - ITSP>IPO (Open for 6.281s)
BYE - ITSP>IPO
200 Ok - IPO>ITSP

I am wondering if when they send the 180 ringing to us with SDP that starts an RTP session running for the ringback signals.

When dialling in the IPO just sends 180 ringing (No SDP).

I love new technology!!!

This ITSP is not on the Avaya Interop Tested list. Does anyone have any experience with VoiceFlex/Gamma etc... and will they supply a test trunk for a short time (free???) to see if it just this ITSP is doing something that the IPO doesn't like.

Jamie Green

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


Fooball is not a matter of life and death-It is far more important!!!!
 
We have a gamma account here. It works as sweet as a nut.

We also have an account for another sip provider who isnt supported and doesnt work.

I think you will find a sip provider will give you test accounts on the basis that you may push them also as long as you dont phone your grandma in delhi.
 
I have also found that I cannot pick SIP calls on an IP phone.

I have had the issue of 2 VCMs in use and the IP Phone pickup problems confirmed as bugs!!!!

I am still waiting to hear why an IP Phone forces the SIP trunk to use the codec that the IP Phone is set to ignoring the settings in the SIP line config.

Jamie Green

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


Fooball is not a matter of life and death-It is far more important!!!!
 
Good info all through there guys... I completely agree about the OMNI. those dark days are behind me now. and you're both on the wrong end of the planet, let alone the english channel.. sorry. :)

Go Kiwi!! Team New Zealand just about the beat the Italians in the yachting.

Newcastle Brown isn't a bad drop though..!

Cheers,

Chris

Auckland
New Zealand
 
All my issues and testing have gone through Tier 4 and are now with engineering!!!!

I have got some trunks with Voiceflex and they only use 1 VCM to dial out for the first call but the second uses two!!!!

Voiceflex trunks should be configured.

I also found that when I played with the settings a little, the sencond call killed my IPO!!!!!

Jamie Green

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


Fooball is not a matter of life and death-It is far more important!!!!
 
has anyone got experience with sip.aql.com. the problem I'm having is getting a speech path down the trunk. why does the Manager program 6.0.5.0 refuse a stun server thats not on the same IP Address as the ITSP?

existing SIP trunks.. the bane of my last three days.
 
What I reported about VoiceFlex trunks is wrong. Calls only use 1 VCM but it will use a second if a call anywhere on the SIP trunks is put on hold.

I think Avaya need to carify how SIP trunks should work as we are all guessing.

It's always good to play with these things but if you don't know what you are supposed to end up with, how are you supposed to get there??

It is like a jigsaw with no picture!!!!!

Jamie Green

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


Fooball is not a matter of life and death-It is far more important!!!!
 
i have sip trunks from priority andalso uses 2 vcm channels
this is with g711 for the trunk and g729 for the ip phones
if i set the phone to g911 then it won't work
hopefully it will get better in the feature because i don't like this kind of working
i don't mind to test but there isn't a rightway at the moment
i did not test this with 4.0.7 by the way

ACA - Implement IP Office
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!
 
On my trunks the IPO forces the SIP trunk to the codec of the IP Phone so it doesn't use any!!!!!!

this is with 4.0.5 and 4.0.7

Jamie Green

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


Fooball is not a matter of life and death-It is far more important!!!!
 
Any news about the RTP relay bug from your Tech support?

Still seeing two VCM's consumed with Direct media on with 4.0.7. Doesn't make a difference which codecs are used on either the handset or trunk.
 
Just to add another odd problem to the list, a parked call from a SIP trunk occupies two VCM's as well with internal MoH (dunno whether internal/external makes a difference)
 
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