BCM50 Avaya VOIP 1220 US->Germany 2

[GrimPapaBear]

Good Afternoon Tek-Tips,

I have run into a strange issue which I am hoping is a quick fix.
It was reported this morning when a employee was attempting to dial from US->Germany for a conference call, it would not recognize the employee inputting the 6 digit conference code. After "No code was entered" they were transferred to the operator. The employee was able to hear the operator, yet the operator could not hear the employee.

I have duplicated the issue on other 1220 VOIP phones throughout the office.

I have successfully connected through the handful of T7316E & T7208/M7208 phones still in production.

What would cause the pot's phones to function internationally yet not allow the VOIP phones to do the same?

Warm Regards & I hope this is an simple fix.

 

[Firebird Scrambler]

Is your BCM 50 fully patched up to date with the latest firmware for the IP sets?.

All the best

Firebird Scrambler
Meridian 1 / Succession and BCM / Norstar Programmer in the UK

http://www.linkedin.com/in/davidbromley

If it's working, then leave it alone!.

 

[GrimPapaBear]

I have not updated it since I started with the company earlier this year, I will look into these updates.

Thank you for your quick reply.

So there is no programming difference for VoiP Lines vs POTs lines?

 

[cook1082]

There is a lot of difference. VOIP routing and gateway routing codecs etc...
We need more info.
How are the voip going out the network H323/sip?

 

[GrimPapaBear]

Thus far I have looked into update's for the BCM50 and related VoIP 1220's yet without a ongoing service contract I am unable to acquire said updates.

Cook1082 "
There is a lot of difference. VOIP routing and gateway routing codecs etc...
We need more info.
How are the voip going out the network H323/sip? "

Where in the BCM would I be able to find the routing information? and respectfully the method used h323/sip? if i am not mistaken SIP is the configured method.

 

[GrimPapaBear]

Sorry for my ignorance on this matter, I have located in Resources -> Telephony Resources -> IP Trunks. The H323/Sip settings, most of it looks defaulted. I guess easy fix went out the window with this one.

 

[GrimPapaBear]

Routing Table: Blank
IP Trunk Settings: Remote Capability MWI Send Name Display
H323 Settings: Enabled-All MCDN Protocol None Normal Route Fallback to: None Call signaling direct tunnelling off rasport 0 regttl 60 gatettl 0 status H323 is running in direct mode
H323 Media Param: preferred codecs G.729->G.723->G711-uLaw->G711-aLaw enable voice activity detection off jitterbuff auto 729 30ms 723 30ms 711 30ms incremental payload off t.38 fax on 711 for 3.1k audo off
Sip Settings: fallback to circuit-switch enabled all dynamic payload 120 sip settings local domain blank call signaling port 4digits status gateway is running
SIP proxy table blank mcdn proto cse
SIP media param 729-?>723->711 ulaw & alaw enable voice activity detect on jitter buffer auto 20ms 30 ms 20 ms fax t.38 3.1k audo off in-band ring off

let me know if this helps, or if more clarification is needed.

 

[GrimPapaBear]

ALSO NOTE VOIP CALLS TO OTHER INTERNATIONAL NUMBERS FUNCTION CORRECTLY IN ALL REGARDS

 

[GrimPapaBear]

http://www.tek-tips.com/viewthread.cfm?qid=1659600&page=1

Solved due to call compression, noted on the bottom of this page.

Putting preference on G711-ulaw -> G711-alow -> G723 -> G729 has corrected the issue