web statisticsweb stats Business Phone Systems Tech Talk Forum - VOIP & Cloud Phone Help

Business Phone Systems

Previous Thread
Next Thread
Print Thread
Rate Thread
Joined: Nov 2008
Posts: 299
Member
OP Offline
Member
Joined: Nov 2008
Posts: 299
Does anyone here have any experience connecting to the Digium SIP trunk service? I've got a CIX40 with the most recent software as well as a current GIPU. I followed the instructions in the manual on SIP trunking as well as I could, but given that Digium isn't one of their listed providers, I'm guessing at some things.

Short story is that the trunks are working on a Grandstream gateway connected to an old Avaya system, but won't seem to work with the CIX.

Symptoms:

Calls into the DID go to busy signal. They should be routing to a PhDN.

For calls out, I dial the access code I created in Flexible Access Codes, hear dial tone, and dial an outside number. This service requires 10 digit dialing (no 1 or 0) but after 6 digits I get an "All Lines Busy" message on the display and hear a busy tone.

Clues? Thanks for reading smile




Dylan. SATUBAW (Some American techs use BIX as well)!
Atcom VoIP Phones
VoIP Demo

Best VoIP Phones Canada


Visit Atcom to get started with your new business VoIP phone system ASAP
Turn up is quick, painless, and can often be done same day.
Let us show you how to do VoIP right, resulting in crystal clear call quality and easy-to-use features that make everyone happy!
Proudly serving Canada from coast to coast.

Joined: Jan 2010
Posts: 937
Likes: 2
Moderator-Toshiba
*****
Offline
Moderator-Toshiba
*****
Joined: Jan 2010
Posts: 937
Likes: 2
Wireshark the incoming and outgoing calls to see what is happening or the responses to your SIP invite.

Are the SIP services actually registering with Digium?


Regards
Carl
Joined: Jun 2005
Posts: 2,702
Likes: 7
Member
Offline
Member
Joined: Jun 2005
Posts: 2,702
Likes: 7
As Carl said, Wireshark will be vital, especially when trying to connect to SIP provider that hasn't been tested with Toshiba. You may not understand all the SIP communication, but it could give you a place to start.

For Incoming calls the phone number must exactly match the URI table in the CIX or the call will get rejected by the PBX. Some SIP providers send the DID number in the E.164 format (such as Twillio SIP trunks). The phone number has a +1 before the area code and number. The Toshiba does not support this format. It doesn't understand the + sign.

If the outgoing call does not have the +1 before the number, the carrier may reject the call. The All Lines Busy may have more to do with SIP trunk registration.

I could not get information able to get info about Digium SIP trunks, so I don't know if that is the problem, but the Wireshark will be important in either case.

To run Wireshark you need to setup a mirrored port on the data switch port connected to the GIPU.

Joined: Nov 2008
Posts: 299
Member
OP Offline
Member
Joined: Nov 2008
Posts: 299
You've certsinly given me something to think about. I will have to investigate WireShark. I've not used it before, but I have heard of it.

I'm wondering if DID length is the problem on incoming calls. I believe Digium is sending a 10 digit number, but the most I can set the CIX to handle is 7. We're going to explore that today, too.

Thank you for the insight, I'll update you with how it goes.


Dylan. SATUBAW (Some American techs use BIX as well)!
Joined: Jan 2010
Posts: 937
Likes: 2
Moderator-Toshiba
*****
Offline
Moderator-Toshiba
*****
Joined: Jan 2010
Posts: 937
Likes: 2
The SIP URI from Digium needs to be configured in P329 as the same 10 digits

The DID length can be anything from 1-7 digits determined in the ILG FB11, this will match the last 1-7 digits of the URI number.

If you incorrectly program the DID then there is always the DID Intercept P318 that can be configured as a "catch all" if you have not created the DID or the destination is incorrect.
Set the type to "not determined on your SIP ILG and send the audio day 12 and night to "dialling digits" and a working extension or MCG or GCO whatever you have configured on the system.


Regards
Carl
Joined: Jun 2005
Posts: 2,702
Likes: 7
Member
Offline
Member
Joined: Jun 2005
Posts: 2,702
Likes: 7
The nice thing about SIP trunks in the CIX is that you get to type in all the phone numbers twice.

Not to keep parroting Carl, but the DID length doesn't matter with SIP, but the URI table must have the full phone number. If they are sending 10 digits then your SIP URI must have the full 10 digit number in the table. Then you can set the number of DID digits to 4 and then enter in all the last 4 digits of the numbers in the DID table. I usually do 4 digits in the DID table.

Remember for Wireshark to work, you must setup port mirroring. I have received many useless Wireshark captures from IT running Wireshark on a server without port mirroring.

Once you get the data captured you will have a ton of data on the screen. You will want to filter it to only show SIP info.

Joined: Jan 2010
Posts: 937
Likes: 2
Moderator-Toshiba
*****
Offline
Moderator-Toshiba
*****
Joined: Jan 2010
Posts: 937
Likes: 2
Or you can do a SIP capture filter to keep the file sizes down.

But we also need to find out if the VOIP service has actually registered with the SIP server first again this is done in Wireshark.


Regards
Carl
Joined: Nov 2008
Posts: 299
Member
OP Offline
Member
Joined: Nov 2008
Posts: 299
Gentlemen, thanks for the input. I've been away at a convention this past week, but I'm back in town now so I'll start digging into this one again. I'll keep you posted on y progress.

Thanks!
Dylan


Dylan. SATUBAW (Some American techs use BIX as well)!

Moderated by  Carlos#1, phonemeister 

Link Copied to Clipboard
Forum Statistics
Forums84
Topics94,262
Posts638,697
Members49,757
Most Online5,661
May 23rd, 2018
Popular Topics(Views)
211,098 Shoretel
187,712 CTX100 install
186,799 1a2 system
Newest Members
BPopilek, Rich F, LewisR, TDKs79, Buttinset
49,757 Registered Users
Top Posters(30 Days)
dexman 18
Toner 11
TDKs79 8
pvj 4
jc2it 4
Who's Online Now
0 members (), 163 guests, and 244 robots.
Key: Admin, Global Mod, Mod
Contact Us | Sponsored by Atcom: One of the best VoIP Phone Canada Suppliers for your business telephone system!| Terms of Service

Sundance Communications is not affiliated with any of the above manufacturers. Sundance Phone System Forums - VOIP & Cloud Phone Help
©Copyright Sundance Communications 1998-2024
Powered by UBB.threads™ PHP Forum Software 7.7.5