Home » Forums » Vendor Forums - Certified » Aastra endpoints

Aastra 6739i Problem - Calls ring and then proceed right to voicemail

SupaF1y
Posts: 6
 Member Since:
2010-01-06
Submitted by SupaF1y on Wed, 01/06/2010 - 12:39pm.

Hi all,

I am not sure where to post this information but this seems to be the correct forum. Hopefully you guys can help.

Problem:
When dialing the new 6739i, the phone will ring once and push the caller straight to voice mail. Even if you were to pickup on the first ring, it would still push to voice mail. The 6739i is able to dial outside and dial extensions, but it can not be dialed.

What I have tried:
6739i has been setup by running aastra-setup command and since there were no templates on the endpoint manager, I chose the 57i Aastra. Phone pulls image fine from TFTP and seems to be configured just happily other than the fact that incoming calls drop straight to voice mail after the first ring.

The configuration is the same as my Cisco 7940 which is right next to the 6739i and can receive calls just fine.

I have tried to change extension numbers to no avail, also have tried to remove and reboot trixbox, no go. Also rebooted the phone to no avail. The trixbox configuration also shows the phone as online. DND was checked and is off, but still no go.

Any ideas? I am very lost at this point. This is our first Aastra phone and it's not exciting to fight with something so expensive brand new out of the box.

‹ Firmware 2.3 published today Application not found on Aastra 55i... ›

&nbsp

Gasmanz
Posts: 235
 Member Since:
2007-09-04
Debug
Wed, 01/06/2010 - 1:32pm

The quickest way to see why this is happening is to do a SIP debug on the Aastra extension or even just watch the asterisk CLI to see why the call has stopped ringing and sent the caller to voice mail. Compare the logs with a normal working extension and you should be able to pick out what is different, then figuring out a solution should be easier.

Is the Aastra phone actually ringing or are you just hearing a ringing tone through the handset of the phone you are making the call from?

--

Graham Campbell

FtOCC Tech Certified
Hi-Tech Solutions Ltd.
Auckland
New Zealand


&nbsp

SupaF1y
Posts: 6
 Member Since:
2010-01-06
Thanks for your response.
Wed, 01/06/2010 - 4:13pm

Thanks for your response. I'll give that a try and see how it goes.

The Aastra actually rings once and displays the caller ID information, when I pick up, it's just dead, no tone, no nothing. One the caller side the phone just rings about 4 times and goes to voice mail... as if the Aastra never even picked up.


&nbsp

SupaF1y
Posts: 6
 Member Since:
2010-01-06
Ok, I ran the CLI log and I
Thu, 01/07/2010 - 1:23pm

Ok, I ran the CLI log and I can't find any differences between the two phones. The only difference with the Aastra it says that no one is available to answer the phone and continues on to voice mail. All the other checks and metrics are the same.

Also, I call the Aastra extension and just let it ring, it will ring several times before the voice mail picks up... but the moment I pickup the handset, it stops ringing on the Aastra but continues to ring on through until it pushes into voice mail.

I did also an SIP debug and I simply don't see anything that jumps out at me.


&nbsp

SupaF1y
Posts: 6
 Member Since:
2010-01-06
Does anyone have ANY
Fri, 01/08/2010 - 12:45pm

Does anyone have ANY suggestions or ideas? Even if it means dancing around the phone 5 times with a glass of milk on my head... I'm willing to try it :)

As I mentioned earlier, I can call from the phone but I just can't pickup when being called, it just continues straight to voice mail. I can also view caller history and just dial back without any problems.


&nbsp

Gasmanz
Posts: 235
 Member Since:
2007-09-04
Not sure whats wrong
Fri, 01/08/2010 - 8:47pm

We had something similar happen with some ATAs but the phone never rang even though the asterisk CLI said it was trying for the regular 20000ms before defaulting to the voice mail. The problem was caused by an update from Fonality on our trixbox pro systems that added a couple of lines to the SIP header that the ATAs did not know how to respond to.

the lines of code were

exten => s,1,SIPAddHeader(Alert-Info: null)
exten => s,2,SIPAddHeader(Call-Info: null)

Which show as this in the debug SIP invite from the trixbox server

Call-Info: null
Alert-Info: null

Not sure if this is the same issue you are having.

--

Graham Campbell

FtOCC Tech Certified
Hi-Tech Solutions Ltd.
Auckland
New Zealand


&nbsp

mag
Posts: 135
 Member Since:
2006-05-31
Remove or rename the
Sat, 01/09/2010 - 2:04am

Remove or rename the aastra.cfg file from /tftpboot

Reset the phone to factory defaults and start again.

Instead of using the config file, just program the basics (server, sip account details) manually via the web gui and give that a try.

--

MaG
www.eGuest.ca


&nbsp

SupaF1y
Posts: 6
 Member Since:
2010-01-06
Gasmanz - what did you end
Mon, 01/11/2010 - 7:41pm

Gasmanz - what did you end up doing to fix it? I see you mention the problem, what did you change or delete or add to make it work for you, perhaps we have a similar issue

mag - thanks for the advice, we will give this a shot. Usually the config system worked fine in the past for us, but if this is the working way then so be it :)

Will update with a report if these work or don't work, thanks guys.


&nbsp

Gasmanz
Posts: 235
 Member Since:
2007-09-04
This was on trixbox pro
Thu, 01/14/2010 - 1:54am

There is a file called "macro-stdextn.conf" which has a check in it to see if the extension being called has FMFM enabled. These lines are sent to the phone in the SIP header at the very beginning but that's about all I know.

It really must be something in the SIP signaling that is being lost or misinterpreted because the phone should be telling the trixbox server that the call has been answered so it can start to send the RTP packets with the audio. Have you tried another phone or a softphone with the same registration details to eliminate the trixbox server config?

If you post this info here http://trixbox.org/forums/aastra-endpoints in the Aastra endpoints section of the forum you could get an Aastra engineer to help you or send him a private message here http://trixbox.org/user/aastra2

Good luck.

--

Graham Campbell

FtOCC Tech Certified
Hi-Tech Solutions Ltd.
Auckland
New Zealand


&nbsp

Atcom Alberta
Posts: 219
 Member Since:
2008-07-14
We are experiencing the same
Sat, 01/30/2010 - 8:34am

We are experiencing the same issue on Aastra 9480i's - all the phones that are on the same network (192.168.100.0) as the trixbox server work just fine. However, the phones that are on a different network (192.168.5.0) intermittently ring once then go to voicemail.

--


TDM & VoIP; Trixbox, Aastra (IP PBX) telephone equipment in Calgary.
atcomsystems.ca


&nbsp

Atcom Alberta
Posts: 219
 Member Since:
2008-07-14
We've also noticed that the
Sat, 01/30/2010 - 8:50am

We've also noticed that the asterisk CLI shows the following message for any calls that fail (just before they're sent to voicemail):

Everyone is busy/congested at this time (1:0/1/0)

--


TDM & VoIP; Trixbox, Aastra (IP PBX) telephone equipment in Calgary.
atcomsystems.ca


&nbsp

Atcom Alberta
Posts: 219
 Member Since:
2008-07-14
We tested our scenario with
Mon, 02/01/2010 - 7:30am

We tested our scenario with a Linksys SPA962 phone; the issue did not appear. In our case it seems isolated to Aastra phones on a different subnet.

--


TDM & VoIP; Trixbox, Aastra (IP PBX) telephone equipment in Calgary.
atcomsystems.ca


&nbsp

bilby
Posts: 11
 Member Since:
2007-04-26
sip_nat.conf
Wed, 02/03/2010 - 1:02am

Maybe it would help to define your subnets in sip_nat.conf
... i just seem to remember that you get no incoming connection whatsoever if that is the problem....but give it a try all the same :-)

Bilby


&nbsp

Atcom Alberta
Posts: 219
 Member Since:
2008-07-14
Thanks, but all the local
Wed, 02/03/2010 - 7:46am

Thanks, but all the local networks are defined in sip_nat.conf. This issue is isolated to the Aastra phones - could be something in the config, I just don't know what...

--


TDM & VoIP; Trixbox, Aastra (IP PBX) telephone equipment in Calgary.
atcomsystems.ca


&nbsp

nhue
Posts: 31
 Member Since:
2007-06-07
I would try and compare the
Wed, 02/03/2010 - 7:56am

I would try and compare the SIP messages flow of a local subnet call versus the ones that don't work by enabling SIP debugging and analysing the logs.

I'v had that sort of problem with an Adtran gateway on a remote subnet, and if i remember correctly it was an error with rtpdirect setup and reinvites on the trunk.. But analysing the SIP messages would get you further then the unmeaningful "everyone is busy/congested" message.


&nbsp

SupaF1y
Posts: 6
 Member Since:
2010-01-06
Sorry that I did not update
Wed, 05/05/2010 - 6:32am

Sorry that I did not update as I promised I would. The problem turned out to be a firewall issue, the stupid firewall allowed other phones through but for some reason this phone did not want to go through. What we did (small office) was put the phone in-front of the firewall and it works great now.


&nbsp

Comment viewing options

Select your preferred way to display the comments and click "Save settings" to activate your changes.