Avaya 96xx/Cisco SIP phone - Stutter tone

[coertvc]

All,

I have installed the 3CX server v12 and have connected a series of Avaya 9650 IP/SIP phones to it. of course the firmware is the latest, the credentials correct and the 3CX server is been seen and, yes, SIP Trunk is up and running

I can receive calls in the group with out issues, but dialing out is a going strange; every-time I initiate a call with phone number x I get a response as if the line is not there, I do not see any lines in the event log (logging is set to verbose) extension status of the phones are registered, idle and available in the 3CX console

if I immediately dial the same phone number x again, the call works without any issues and I get connected and can talk. (there is no difference in internal or external call behavior)

any thoughts on what this can be ? (on the previous asterisk server the phones worked quite fine)

regards,

Coert

 

[leejor]

Re: Avaya 96xx SIP phone

If there is no 3CX log of a call, then you have to assume that (all server settings being correct), the call is not leaving the set. Being an intermittent issue make it more difficult to troubleshoot.

If the set has the ability to have a connection to a Syslog server, you may want to make use of that. If the set has an internal dialplan to pre-screen dialled numbers, you may want to review that.

Does the issue arise when dialling any number, or certain numbers. If certain digit combinations, then that might be a clue to leading to the solution.

 

[coertvc]

Re: Avaya 96xx SIP phone

Thank you Leeror for your response,

the behavior is consistant, so every time you first dail a number (no matter which) it comes with a fast series of beeps until you hang up again. then when you redail; no issues

it feels like there is some initiation with the server that may be delayed and is not fully active, and at second attempt it is. then when you do not touch the Phone for a couple of minutes, it may be back in 'sleep' again and hence the same behavior. it can be that I'm overlooking a kind of 'keep alive' setting.

anyway, I will check out the syslogging

thanks,

Coert

 

[pauljm]

Re: Avaya 96xx SIP phone

Are you downloading the 46xxsettings.txt file to the phone on bootup or are you entering the settings manually?
The settings file contains a lot more configuration detail that needs customising than the manual method so I would recommend using this as a means of setting up the phones. I am using the Abyss Web Server as the HTTP source for the upgrade & settings files.
I have a 9640 working fine, just need to play with the dial plan in the settings file as I still need to enter # when dialling to indicate end of string or wait for the entry time out & having issues with Contacts list.

PS My proper job is as an Avaya Specialist.

 

[coertvc]

Re: Avaya 96xx SIP phone

Thanks Pauljm

the config was done manually, and i saw the 46xx files and they are massive, so I took the short route first

I activated the logs and this is what I got;

First attempt to dail failed
ALSIP - - - ALSIP: +02:00 2014 000 1 .TEL | 0 UA: CLineSet:rocessChallenge: for lineID <737> - authorization failed
NETMGR - - - NETMGR: +02:00 2014 000 1 .TEL | 0 Failed to Remove Filter rule Custom Rx Filter-2 in the Network Manager.
SIGADAP - - - SIGADAP: +02:00 2014 000 1 .TEL | 0 EndSession() failed. Error Line Manager Error: Invalid line number..
NETMGR - - - NETMGR: +02:00 2014 000 1 .TEL | 0 Failed to Remove Filter rule Custom Rx Filter-2 in the Network Manager.

Second attempt 20 seconds later that was successful
SESSION - - - SESSION: +02:00 2014 000 1 .TEL | 0 free mem = 5884048 #011 max free block = 5570544
AUDIO - - - AUDIO: +02:00 2014 000 1 .TEL | 0 CAudioStream::CreateStream: could not update the stream with SDES parameters.
NETMGR - - - NETMGR: +02:00 2014 000 1 .TEL | 0 Failed to Remove Filter rule Custom Rx Filter-3 in the Network Manager.

and btw; there was no mistake in the number that I entered

this is what i have in the logs as well non stop
PPMDATA - - - PPMDATA: +02:00 2014 000 1 .TEL | 0 getInitialEndpointConfiguration: Encountered HTTP SOAP error. Error= 7
HTTP - - - HTTP: +02:00 2014 000 1 .TEL | 0 RequestURL():curl_easy_perform error rc=7 0x81240990#012URL=https://192.168.200.9:443/axis/services/PPM#012ERROR=couldn't connect to host

While I now dive into the log file, any suggestion on the above is appreciated, I also double checked again on against the astrisk server and the behavior is there the same as well, this may have been since the latest firmware upgrade I did...

Thanks again and regards,

Coert

 

[coertvc]

Re: Avaya 96xx SIP phone

I have now updated the phones to the latest and greatest Binaries, also I have used the 46xx to config the phones, unfortunatly no luck at the moment. (I tested quickly a Yealink t22, worked without issues within seconds, but its NOT an Avaya :S )

Any suggestions are welcome !

 

[crombiecrunch]

Re: Avaya 96xx SIP phone

Would you be able to post your 46xx file and the 3CX phone template you used? I have a 9611G thats causing me some headaches.

 

[pauljm]

Re: Avaya 96xx SIP phone

The Avaya 46xxxsettings.txt file does look rather large but it needs to be able to configure virtually the whole range of Avaya IP phones (96xx, 46xx, 16xx & IP conference phones), the main areas to concentrate on are the settings that apply to 96xx SIP & some general settings. It takes a bit of scrolling through but is worth it.
Particular settings --- Use the KISS protocol (Keep It Simple, Stupid).
SIP Registrar -- 3CX IP address.
SIP port -- stick with 5060
SIP Signalling -- stick with UDP
Avaya Environment = 0 (Avaya Environment Disabled)
SIP Domain -- as on your 3CX server

Just don't try to be too clever, you can play around with the other settings once the basics are working.

As I said I have a 9640 working OK (just problems with the phones Contact list).
Good luck.

 

[coertvc]

Re: Avaya 96xx SIP phone

So, I could not get it working on an avaya and bought my self a few Cisco 525G (supported by 3CX) Same issue; pick up the phone, piep piep piep piep piep piep and after so many seconds a dail tone that sometimes would give you the option to dail. ALso changed the Netgear GS724TP PoE switch (which is crap btw) to unmanaged switch and PoE adapters; no change. moved the #cx server to a new OS, no change (from 2003 to 2008R2 by backup and restore)

for some reason I came by accident into the voicemail system and thought why on earth I was having these 5 voicemails still in the system, GUESS; after I deleted the 5 voice mails every started to work; I was now able to pick up the phone and immediately start to dail a number and actually get a person on the other side of the line..

to make sure i was not stupid; I took an earlier copy of the vm I had, had the issue, deleted the 5 voicemails and all started to work as expected.

case closed, this is a very strange bug and would appreciate response from 3CX

 

[coertvc]

Re: Avaya 96xx SIP phone

On avaya it self on 3CX;
I have an extra http port 80 server that provides the files and updates, the 46xx file was easy to update (long read, I admit but all fine) from there the latest SIP 2.6.5 firmware, settings and menu config files. I did not try to make/post a config file in the 3CX (still somewhat Newby to 3CX)

comment in general;
I'm very interested in how I should translate signals of a phone. we all know the "occupied" and "ready to dail" signal, but there are quiet a bit more that if I would know how to name them, I would be better off telling what the issue was.

on this particular issue the 3CX did not get ANY info in their logs, the avaya showed the info as stated earlier and the Cisco only said: "dtmf 48" and that was all

 

[lneblett]

Re: Avaya 96xx SIP phone

On a couple of your comments:

I have used the Netgear switch with excellent results for several years. I have also used Cisco and a few others, but for the most part, Netgear has been an excellent choice given its price to feature ratio. I have yet to have one fail.

One the second issue of the beep, beep, beep, you were hearing. It is oftentimes called a "stutter-tone" and is used to let those who might be sight impaired and unable to see the visual voice mail indications know that upon hearing the stutter before getting to a dial tone that they have a voice mail. There is no bug as this is common in the PSTN world and in most PBX systems. When the vmails have been heard and/or deleted, the stutter-tone will no longer present itself as an immediate dial tone will be heard. If the vmails are not handled, you merely need to wait until the stutter is done and then you should be able to dial.

When you dial out - try dialing an internal extension or *777 (echo test) and see how this reacts. I was not sure when you mentioned dialing out if this meant "outside of the LAN" or merely dialing any number.

 

[coertvc]

Re: Avaya 96xx SIP phone

Thanks Ineblett,

first on Netgear; the issue is that a browser that is able to do more that two (data)sessions (mozilla, IE, Firefox)from a GS724TP (latest firmware) will not have the advanced options that actually make you to really manage a switch (setting LAG ports as an example) Netgear in NL have already responded last week to me that this is a known issue and that if you see this you need to use Opera or Firefox portable. downside is , because it is these two are so slow and clunky it actually able to FULLY lock up your switch. it does not support NTP fqdn properly and if you point it to a ip instead; it locks up as well.

it seems that finally someone is able to give the tone a name. Funny thing is though why my Yealink t22 did not have the issue and the avaya and Cisco had the issue... (before the wipe of the Voicem)

stutter-tone..... seem to take 7 seconds and then you can work... I took the red light as an error as the voicemails are actually forwarded to a different extension than the extension that is calling. Group ring, if not present mail to VM box X, mail voice mail to mail address and delete VM.

 

[lneblett]

Re: Avaya 96xx SIP phone

Well I will agree that I wish the web GUI response was more robust and it does seem to like some browsers more than others, but I have not had the lock -up problems you describe. I use an IP for NTP with great success. In any event......

On the stutter tone, the Yealink has an option to enable or disable in the web interface. Most of the phones seem to have this option but I cannot say how each is defaulted or how 3CX may handle in their various templates. The Yealink admin manual has some more detail on the feature.

 

[leejor]

Most PSTN lines and analogue PBX's (and Cisco/Linksys ATA's) that have this feature implemented, do not require you to wait until the stutter dialtone had changed to a steady tone, before dialling a number. The situation may be different on your sets. Most people that are familiar with this tone are not used to waiting.

Many sets/ATA's have the ability to select visual MWI. audible MWI or both. Stutter dialtone is usually reserved for use on sets that do not have a message waiting light (standard analogue set with ATA), or (as mentioned) a set used by a person with a visual disability. In most cases, visual message waiting is preferred, and the only option required