When I call 450-371-xxxx, it rings about 10-12 times then it hangs up without any reason!

Hi community,

When I call the number 450-371-xxxx, it rings about 10-12 times then the line hang up without any reason.
It happens ONLY with this number and I can reproduce the issue from many telephones in the office.
If I call from my cellular, it works perfectly.
I searched a lot but I can’t find the right answer.
You can see some logs here: http://privatepaste.com/e59e277e19/freepbxmtl
Anyone can explain me whats happens ?

I use:
TelUS BRI > Cisco IAD > FreePBX.
FreePBX 12.0.33 fully updated.
Asterisk 13.
Cisco 7960 phones with latest firmware.

If I miss some details, please tell me and I will add them.
Any answers will be very appreciated.

Thanks a lot and have a good week.

Sébastien B.

That woulld seem to be problem with your Cisco device

[2015-01-23 08:32:45] VERBOSE[1345][C-00001208] app_dial.c: SIP/CiscoIADMtl-00002441 is making progress passing it to SIP/254-00002440
[2015-01-23 08:33:05] VERBOSE[1345][C-00001208] app_dial.c: SIP/CiscoIADMtl-00002441 is making progress passing it to SIP/254-00002440
[2015-01-23 08:33:35] VERBOSE[6794][C-00001208] chan_sip.c: Got SIP response 500 “Internal Server Error” back from 192.168.7.3:5060
[2015-01-23 08:33:35] VERBOSE[1345][C-00001208] app_dial.c: SIP/CiscoIADMtl-00002441 is circuit-busy
[2015-01-23 08:33:35] VERBOSE[1345][C-00001208] app_dial.c: Everyone is busy/congested at this time (1:0/1/0)

wow, Thanks for your fast answer.
how can I diagnos it then fix it ?

Thanks again!

You would have to talk to Cisco, it was their error.

Can you explain why this issue happens ONLY when I call this number ?
I never got this problem before…

Its why I thought there was a misconfiguration with my FreePBX server.

What do you think ?

Anything I would have say would be a guess, there is a lot involved with ISDN and how particular hardware does call routing successfully. Does the phone ring?

I’m not a fan of the Cisco SIP load - it’s touchy like this, and you lose most of the power of the phone when you run it as a SIP device. I recommend you reset the phone to factory and reload the software. You can also try downgrading to either an older version of the SIP load and see if it still does it. I would rather use a SIP Softphone than use a 7960 or 7940 with SIP. Cisco provided the SIP software as a bridge for some of their customers, so it’s not very well tested or written. Most of the people that I’ve talked to don’t like the 7960 for SIP; your mileage may, of course, vary. Personally, it seems like a lot of the features of the phone were crippled in the SIP image.

The CISCO Skinny firmware works a lot better with these phones and allows for things like programmable buttons and softbuttons that make sense. Unfortunately, the Skinny software that comes with Asterisk isn’t very well documented. I used Chan-SCCP-B for my Cisco phones and have written up a tutorial on their website for installing the software.

Whether you pursue SIP or SKINNY with these phones, the one you are working with now sounds (to me) like it needs to be factory reset and reflashed.

@dicko: Yes I can hear the phone ringing.
Can you help me understand why you belive that the CIsco IAD is involved ?
@cynjut: Thanks I will try.

Because it sent an internal error back to asterisk while in a SIP session ?

@cynjut’s suggestion probably wont help you as your device is an IAD not a phone :wink:

@dicko: Yes it sent an internal error back to asterisk while in a SIP session.
I also think the issue is with my Cisco IAD and not a phone.

Hi,
[TK]D-Fender asked me some debugs.
I produced the debugs then I pasted them here.
https://privatepaste.com/464f3755fa/linehangup

We did a test with the extension 254 (IP 192.168.7.247) to the 4503716444.
I can hear ring 10-15 times then the line hang up!

Anyone can tell me what is the cause ?

Thanks you VERY much.

Sébastien

To answer your question directly the hangup "cause " was 38 (network out of order) but as I said before you will have to debug at your IAD level as it converts the underlying ISDN call to sip and is apparently unable to properly handle the Q931/Q850 sequence that caused those calls to be rejected.