SIP Trunks

[Jockey]

Hi, Mitel 3300 MXE III, MCD 5.
The problem is we have installed SIP Trunks for our customer and have had considerable problems to get to the stage where I can say that we can let the customers traffic on it.
The latest and hopfully the last is that outgoing dialled calls from an IP phone are OK, but fail if you use the redial, no speech, from an Analogue phone no problems.
Any ideas appreciated.

Regards Dave

[martyn]

It's not quite clear on what you are asking, so can you clarify if the it is only when you redial that you get no audio, or is there no audio all the time?

I would assume that the audio problem is all the time. If so is it one way or none at all? Most likely this is a Layer 3 routing problem.

If the redial is a seperate problem, then can you elaborate more on what happens when you try to redial? Does it just not put the prefix code at the front, or does it fail in another way?

[Jockey]

Hi, Thanks for your reply.
To explain.
An outgoing dialled call from an IP phone speech bothway good call.
An outgoing call via redial key on the IP phone called party phone rings no speech either way, display on IP phone shows call answered and timer starts on both the caller and external called party, also noticed a lower level of ring tone.
An outgoing dialled call from an analogue phone speech bothways good call.
An outgoing call via redial key on the analogue phone speech bothways good call.
Regards
Dave

[ralph]

Silly question I know, but what is a redial key on an analog phone?  What model of phone is it?
I can't see how the 3300 would know you pushed a redial button an an analog phone vs just dialing.

Ralph

[martyn]

There has to be more to this than that, as like Ralph said it doesn't make sense.

From the maintenance command run:
SIP TCPDUMP ON
Test a normal call and a redial call seperately. Go back to the maintenance command and enter:
SIP TCPDUMP OFF

log in to the controller via FTP and browse to the vmail folder. Grab the packet capture from there and either have a look at it in wireshark, or attach it to a reply post here.

[Jockey]

Thanks Guys we will try the maitenance commands as given.
Regarding your remark about the redial key/button on an analogue phone, I have been involved with PABX Systems for approx 50 years and to my knowledge there has always been a redial button on analogue phones that we have installed on business systems here in NZ, may not apply in other countries.

[Jockey]

Hello Ralph
The analogue phones in question are the teledex L2S+ installed in a large Hotel.

[matthew]

I have no idea if this is relevant. I'm just musing away my last few minutes of the day..

If you haven't put the new trunks into production, I'm assuming you haven't put them in a route? Are you directly dialling the trunk number to access the SIP trunk? I wonder if that is fooling the redial function on the system phones some how?

Again, apologies if I'm wasting your time.

[petr.necas]

An outgoing call via redial key on the IP phone called party phone rings no speech either way

I would try compare INVITE packets when calling out via redial key and when calling by typing the number on the phone's keypad. To me it looks like SDP is not part of the INVITE packet when using redial. If this is the case try to enable "Force sending SDP in initial Invite message" in SIP Peer Profile.