Voiceflex SIP trunk to trunk transfer

[mtechsteve]

Can anyone help.. I have 5 SIP trunks from Voiceflex, I am trying to set up a blind transfer from an incomming call on a SIP trunk to an external number using another SIP trunk (For example an uncon' forward to a mobile from an extention)

The call is presented to the mobile but there is no audio once the call is answered.

I have asked Voiceflex for help but they think its an Avaya issue?

Any suggestions?


Steve

 

[kholladay]

Do you have reinvite enabled? If so and you are behind a firewall I would turn it off.

Kyle Holladay
ACA-I, ACA Call Center, ACS-I, ACS-M, TIA-CTP, MCP/MCTS Exchange 2007
ACE Implement: IP Office

"If it worked the way it should you wouldn't need me

 

[mtechsteve]

I have removed the tick from the "Reinvite" box and rebooted. Still no difference....

Also.... The IP Office drops the call about 15 seconds after I answer it.

????

 

[Bas1234]

You could try to set the "Binding Refresh Time" on 60msec.
System > Lan Tab > Network Topology.

Greetzzz...Bas

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________

 

[kholladay]

58389 Call Forwarding Unconditional provides NO SPEECH when a call on SIP is forwarded Off-switch down the same SIP provider MT_RELEASE_2Q08_4.1


Kyle Holladay
ACA-I, ACA Call Center, ACS-I, ACS-M, TIA-CTP, MCP/MCTS Exchange 2007
ACE Implement: IP Office

"If it worked the way it should you wouldn't need me

 

[mtechsteve]

bas134 - That made no difference....

Still no audio and the IPO hung the call up after 15secons!


kholladay - Is that a tech bullitin? I cant find it.


Dam Im getting desperate.... Anyone got any ISDN??

 

[kholladay]

It's a super secret hidden CQ

Kyle Holladay
ACA-I, ACA Call Center, ACS-I, ACS-M, TIA-CTP, MCP/MCTS Exchange 2007
ACE Implement: IP Office

"If it worked the way it should you wouldn't need me

 

[mtechsteve]

any chance of unhiding it?

s

 

[DavidCT]

I've been plagued with the same problem since at least 4.0.10 and if my memory serves me correctly, the problem started when we went to SIP Registration rather than IP.

We have the problem with two different SIP providers.

That most odd part though is this problem rears it's ugly head 9 out of 10 times. Once, every so often, the call is fine.

 

[mtechsteve]

so guys... how do we fix it??


steve

 

[Bas1234]

Is STUN activated?
If your provider doesn't support STUN then leave it blank 0.0.0.0 but tick "Run STUN on Startup"

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________

 

[kholladay]

The issue seems to be with the swap from VCM to RTP Relay. I could swear this is broken somehow although I have no formal confirmation. I can say that when sending the call out another SIP trunk (i.e. 2nd provider) this isn't an issue for me. I only experienced the issue when I was behind NAT and I sent the call down the same provider’s trunk. I have since put the IP500's WAN port on the Internet and haven't had any issues.

Kyle Holladay
ACA-I, ACA Call Center, ACS-I, ACS-M, TIA-CTP, MCP/MCTS Exchange 2007
ACE Implement: IP Office

"If it worked the way it should you wouldn't need me

 

[IPOfficeIreland]

i use it on the same sip provider
stun is open internet although this has nothing to do with it
binding at default 0
registration is turned off
provider has it set as trusted ip
reinvite turned off - this caused a problem with the call disconnecting as soon as it was answered by the external party
codecs set at G711alaw

i found problems with the same config when the provider tried switching it to G729 codec on the trunk. we found no speech and disconnect problems like yours.

ask provider to set all incoming and outgoing to g711alaw and match the same in ip office.

 

[Bas1234]

The binding refresh time is needed sometimes when you do a sip trunk to sip trunk transfer.

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________

 

[mtechsteve]

I have had the following responce back from Avaya 4th Teir support....

"I’m just waiting to hear from them. It’s been raised as a bug - CQ58389 – but has been initially rejected, but Tier 4 can reproduce the same problem so they have re-raised it and are waiting to hear from them again."

Good news.... Watch this space!!!!

 

[mtechsteve]

This was the official responce from Avaya....

Avaya have advised that this is a issue with certain routers using sip. The routers need to be manually configured for port redirection for NAT. Basically as SIP is coming in and going back out RTP is using 2 different ports, these need to be mapped to each other. Old routers do not have this option but newer ones do. Avaya can not specifically show up where to configure this, they have asked if you can take a look and see if there are settings you can change?

Any comments?? Im gonna give it a go this weekend.