Allworx Technical Support Forum

Hello all! My name is Ben, I work for a small telecom company that support Allworx as well as other system vendors. Last week, my boss tasked me with getting a few generic SIP phones hooked up and talking to our in-house Allworx system. I got out Yealink phone to hook up NO PROBLEM, but both AASTRA phone I've tried are giving me headache and keep saying "No Service". I've attached a couple screenshots to help you guys see if I did anything incorrectly. I searched for similar threads before posting this one, found one, tried to replicate it into my setup and I'm still not having luck.

Any help would be greatly apprectiated. Thanks!

Ben

Remove both proxy server settings. The only settings that should be needed under the SIP section on the phone are the registrar ip (your allworx ip) and port 5060 plus the authentication ID (Login ID on allworx and password) in the top section.

I have to mess with one of these brand of phones later this week and can advise after I have done so.

The above advice has worked for me before with conference room ip phones and other specialized generic SIP devices.

Look under "Reports / System events" to see what the error is. That will provide a clue. You can also capture a SIP log under maintenance.

Thanks for the reply!

So I tried removing the Proxy Addresses and having just the registrar address/port and the login id/password, but the phone is still saying "No Service." When I look in the logs, the only things i see over the last 96 hours are my logins to the administration console and messages pertaining to our SIP trunk provider. I don't see anything about the Aastra phone trying to register. What's strange is I was able to get my Yealink phones to register no problem using the same procedure.

Are there any other sections of the Aastra configuration that need touched on or is SIP the only one?


I'm fresh out of ideas

Specific model would help. Is it a wired or wireless phone?

Aastra 6731i wired phone hooked into an 8-port PoE switch connected to our main switch. The Allworx system is the 6x running 8.0.9.5

I wanted to make sure it was not a DECT (wireless, which requires it to be registered to the base)

Since its not, here is how I add a generic SIP to allworx, Home > Phone System > Handsets > Add Generic SIP Handset.
Enter the extension and Description then for the login ID enter 1111 and of course in this version your forced to use the password they want.

Then go back to handsets and see what UserID was assigned to the phone, modify it and change the login ID to match what was shown for the user ID.

Use the new login ID (User ID) on the phone along with the god awful password it assigned.

Are there any AASTRA-specific configurations that need done? I tried following your posts and I'm still getting "No Service" on all AASTRA phones I've tried. Other 3rd party vendors work, it seems my problem is localized to this specific vendor.

Here are a few screenshots. Since this isn't for a customer but rather for in-house testing, I have no problem giving specific configuration if need be. The login ID assigned is "5107" and i assigned a password of "123456" for ease of testing.


SIP CONFIG PAGE (AASTRA)


LINE CONFIG PAGE (AASTRA - wasn't sure if I needed to configure this page but did it anyhow)


ALLWORX SIP HANDSET (login: 5107, password: 123456)


ALLWORX USER PAGE (login: sip2, password: 123456)



I've tried using both credentials for "5107" as well as "sip2" when setting up the phone. Neither seem to be working. I only modified settings in the SIP registration and Line 1 pages of the AASTRA phone. Do I need to go into anywhere else? What am I missing that prevents this from working?

only the global SIP needs filled out. The other are optional as they allow buttons to pick different systems. Delete them under the line configs.

go to maintenance tools and turn on sip log. Then reboot the phone several times. View the SIP log and you should see the failure response back from the allworx. And Im sorry but it was a yealink phone I setup last week not an aastra. Look to see if the phone has a log where you can also get a hint of where it is failing. Also set the registration period to something like 10 mins instead of zero.

I can't get syslog reports to get sent to our syslog server from the phone, but I was able to do a diagnostic capture to see what is going on in the Allworx itself. Below is the section pertaining to the AASTRA phone:



(seqNum[457] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: === RCVD UDP 172.16.16.71:5060 [647 bytes] ===
(seqNum[458] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: REGISTER sip:172.16.16.141:5060 SIP/2.0
(seqNum[459] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: Via: SIP/2.0/UDP 172.16.16.71:5060;branch=z9hG4bK4bc6389ee5eaea77b.9c61c7772b9443936
(seqNum[460] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: Max-Forwards: 70
(seqNum[461] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: From: <sip:No%20User@172.16.16.141:5060>;tag=d1583dd78e
(seqNum[462] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: To: <sip:No%20User@172.16.16.141:5060>
(seqNum[463] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: Call-ID: 9cd3060f51f81053
(seqNum[464] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: CSeq: 16263 REGISTER
(seqNum[465] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
(seqNum[466] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: Allow-Events: talk, hold, conference, LocalModeStatus
(seqNum[467] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: Contact: "No User" <sip:No%20User@172.16.16.71:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D22BA55>"
(seqNum[468] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: Supported: gruu, path
(seqNum[469] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: User-Agent: Aastra 6731i/2.5.3.18
(seqNum[470] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: Content-Length: 0
(seqNum[471] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE:
(seqNum[472] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: ===========================
(seqNum[473] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: === SENT UDP 172.16.16.71:5060 [403 bytes] ===
(seqNum[474] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: SIP/2.0 404 Not Found
(seqNum[475] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: Via: SIP/2.0/UDP 172.16.16.71:5060;branch=z9hG4bK4bc6389ee5eaea77b.9c61c7772b9443936
(seqNum[476] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: To: <sip:No%20User@172.16.16.141:5060>;tag=2484a-950-5e328
(seqNum[477] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: From: <sip:No%20User@172.16.16.141:5060>;tag=d1583dd78e
(seqNum[478] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: CSeq: 16263 REGISTER
(seqNum[479] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: Call-ID: 9cd3060f51f81053
(seqNum[480] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: Date: Fri, 11 Sep 2015 17:03:12 GMT
(seqNum[481] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: Contact: <sip:No%20User@172.16.16.141:5060>
(seqNum[482] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: Server: IST-OIS-8.0.3.6
(seqNum[483] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: Content-Length: 0
(seqNum[484] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE:
(seqNum[485] - 7:09/11/2015 01:03:11pm) tSip: SIP_CAPTURE: ===========================


For whatever reason, it's saying "no user" when trying to register SIP. Now I've tried both the "sip2" login and the "5107" login and both are saying the same thing. I set the registration period to "360" and still the same thing. Now what I did notice when booting up the phone is it'll come up and say "Network Connected", then about 5-10 seconds later it goes to "No Service".