I was wondering what the best practice is to convert a user from an analog extension to a H323 extension / VOIP phone.
Currently I'm having an issue where I have to completely remove the user for the old extension to delete so I can switch it over to a H323 extension which would be fine, but...
Hello All,
I have setup a hunt group and enabled the voicemail. We currently have voicemails in it but can't get to them. Even when dialing *17 and entering in the extension and password. Do these store somewhere else?
What I'm trying to do:
This hunt group has 2 users in it and they will need...
Any ideas?
What I think may be happening is the first URI is picking up all the traffic and it's not routing the 2nd URI. Does it matter what order I have them in? Should I put in a URI for all of the DID's?
I was told there are just sending down the trunk. That's why I was wondering how to setup as I'm used to assigning to a DID.
The sip log attached was when I called the TF#.
Hello All,
We have a SIP trunk with about 4 DID's and 2 TF coming over it. We would like all but one TF to answer to our main IVR application. I'm trying to route one of the TF# 855-405-9160 but to no luck; it goes into the IVR with the other lines. I've tried to setup a SIP URI with this...
Okay so I got this to work after fiddling around with it.
I went back to the *DCP but changed it from 0,0,2,1,0 to the following 0,0,2,2,0 and now it's working.
Thanks for your help!!
Okay so I added the button under button programming on that user.
Label: AA
Action: Internal Auto-Answer
Action Data: FF
This is for an Avaya 9611G phone.
When pushing the button it turns green on the phone but doesn't auto-answer. Did I set something wrong?
Right now I have Action 'Internal Auto-Answer' with data of FF. Is this correct? I pushed the button and still didn't auto answer.
Do I need to remove the *DCP short code for internal auto-answer?
-D
amriddle01,
Do you know of a way to turn on auto-answer so they can continue testing without user interaction? I have internal auto-answer turned on and it works when dialing from an internal extension, but doesn't work when they are calling from the SIP line / server.
Thanks,
D
amriddle01 I think this fixed it....I knew I missed some little thing and it was the blank with a '.' destination. Now the call calls the user and then once they pick it up it dials the outside number.
Thank you!
Below is another log. Is it possible it's dialing to the external SIP provider but with the wrong information? I see the Nextiva (external) voip getting dialed in the below but a call never completes.
UNICODE-UTF8
enu
1266038mS CMCallEvt: 0.1226.0 -1 BaseEP: NEW CMEndpoint f4c0d228 TOTAL...
I have attached a wireshark log from the server getting the '404 Not found' error and the log from the IPO. I don't think these are at the same time but they have the same result. What is trying to happen is:
1. Call inbound user (ext@ip)
2. Once answered, call external customer (ld#@ip)
3...
Hello All,
We are currently running Avaya IPO r8.1 and we have a vendor trying to connect to us using some SIP lines but running into issues. There is two steps to this process one is connecting to internal extensions and the other is dialing external numbers.
Our setup is a SIP line to an...
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