Hi golfdoctor,
1- we are talking about 2000 phones;
2- We start to notice this when we are migrating more users from analog/digital phones to Avaya 1140/1120 sets.
3- The 1120/1140's software version is SIP 4.3 (I'm currently testing 4.4 sp2)
4- All media cards (MC32S) are configured with G711...
Hi Tacosoup
At one point yes we've loaded all latest patches that were included in the 7.5 service pack 24 and another delta patch that was suggested by Avaya;
Did you see any similar case with one of your customers?
Thank you.
Hi Everyone,
Hi Everyone,
We are facing a weird issue: When a call is originated from a SIP set 1140/1120 to external phone set via PRI/PSTN randmonly call get crosstalk/jumpped to another call;
Or sometimes during the call, the phone goes on hold without pressing any key.
The calling party can...
Hi everyone,
If I have 2 PRI connected to CS1000M:
PRI #1 Carrier _1
PRI #2 Carrier _2
If I dial destination 1(NPA)-NXX-#### and the routing table is set to terminate on Carrier_ 1;
For troubleshooting purposes , how can I force a call to carrier_2 without changing the routing table.
Thank you.
Hi everyone,
1 - Do you have any idea how to make an Avaya 1140e (SIP frimware 4.03.012) automatically login to Sip server without prompting for the User login credentials?I've tried couple of options without any success.
2 - How can I disable LLDP (not manually on the phone) from the...
Allenmac : Sounds good I'll try that and I'll keep you posted.
Trvlr1: I use to manage VOIP Pbx like Asterisk, Trixbox, shortel,... Adding multiple keys to the same Extension does not require a lot of configuration so this is why I'm trying to understand the concept from Avaya's expert users...
Hi Everyone,
1- Can someone explain me how to add on the same SIP Nortel 1140e device that is currently connected to a CS1000 ver 7.5 another extension (DN).
2- Also, on the same device, how can have the same extension (DN) with more than one key (<2 lines).
---
Q1: SIP 1140e key #1...
Hi Everyone,
Is there a way (command line ?) to debug SIP protocol (SIP Messages exchange) on CS1000 (Rel 7.5) for peer connection and UA as it exists on cisco (ccsip debug all) or asterisk (sip set debug ip x.x.x.x).
I'm looking for the same debugging trace feature for UNISTIM as well
Thank you.
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