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Sip phones down on UCX50 1

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atdjr123

atdjr123

Technical User
Mar 12, 2012
559
US
I have a ucx50 that is my demo but also doubling as my office system. The system has been up and running for the most part successfully for the past year. HI have a license for 10 users and only have 5 connected or at least I did until now. I have 2 nortel phones and 3 sip phones. Just out of the blue my sip phones all de-registered at the same time. However, my nortel phones are still up and running with no issues. I haven't changed anything in configuration and all my sip trunks work just fine. These are three different phones, yealink, polycom and snom. Unable to dial internally or externally. Any thoughts anyone????

Voice Connect Plus, Inc.
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can you ping yahoo.com or other sites from the command prompt to see if name resolution is working
PBX>TOOLS>JAVA SSH
on some asterisk systems if DNS is not working sip phones won't register
 
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Hey dutchie, I did type that and this is what came back below
No such command 'sip set debug' (type 'core show help sip set debug' for other possible commands

So I typed in core show help sip set debug and execute- response below
sip set debug {on|off|ip|peer} Enable/Disable SIP debugging------------->
So i typed sip set debug on and tried to register set. navigated to PBX and support tab, then refresh and below is a portion of what it spitted out
[
User-Agent: voip.ms
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
[Jan 17 09:43:26] VERBOSE[3240] chan_sip.c: --- (10 headers 0 lines) ---
[Jan 17 09:43:26] VERBOSE[3240] chan_sip.c:
<--- SIP read from UDP:74.54.54.178:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK642d7326;received=69.136.23.217
From: <sip:148336_0930@74.54.54.178:5060>;tag=as44c22779
To: <sip:148336_0930@74.54.54.178:5060>;tag=as6f126235
Call-ID: 65acdcb006ba7b6a37a819f803a7f056@127.0.0.1
CSeq: 954 REGISTER
User-Agent: voip.ms
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Expires: 120
Contact: <sip:s@192.168.1.199:5060>;expires=120
Date: Fri, 17 Jan 2014 14:43:26 GMT
Content-Length: 0

<------------->
[Jan 17 09:43:26] VERBOSE[3240] chan_sip.c: --- (13 headers 0 lines) ---
[Jan 17 09:43:26] VERBOSE[3240] chan_sip.c: Scheduling destruction of SIP dialog '65acdcb006ba7b6a37a819f803a7f056@127.0.0.1' in 32000 ms (Method: REGISTER)
[Jan 17 09:43:26] NOTICE[3240] chan_sip.c: Outbound Registration: Expiry for 74.54.54.178 is 120 sec (Scheduling reregistration in 105 s)
[Jan 17 09:43:41] VERBOSE[3240] chan_sip.c: Reliably Transmitting (no NAT) to 74.54.54.178:5060:
OPTIONS sip:74.54.54.178 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK53ea72b7
Max-Forwards: 70
From: "Unknown" <sip:148336_0930@192.168.1.199>;tag=as3923d9c1
To: <sip:74.54.54.178>
Contact: <sip:148336_0930@192.168.1.199:5060>
Call-ID: 121212a85a21e45935472c607788a846@192.168.1.199:5060
CSeq: 102 OPTIONS
User-Agent: UCX-3.0(1.8.24.0)
Date: Fri, 17 Jan 2014 14:43:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Jan 17 09:43:41] VERBOSE[3240] chan_sip.c:
<--- SIP read from UDP:74.54.54.178:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK53ea72b7;received=69.136.23.217
From: "Unknown" <sip:148336_0930@192.168.17.2:5060>;tag=as3923d9c1
To: <sip:74.54.54.178:5060>;tag=as12dc0c30
Call-ID: 121212a85a21e45935472c607788a846@192.168.1.199:5060
CSeq: 102 OPTIONS
User-Agent: voip.ms
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:74.54.54.178:5060>
Accept: application/sdp
Content-Length: 0

<------------->
[Jan 17 09:43:41] VERBOSE[3240] chan_sip.c: --- (12 headers 0 lines) ---
[Jan 17 09:43:41] VERBOSE[3240] chan_sip.c: Really destroying SIP dialog '121212a85a21e45935472c607788a846@192.168.1.199:5060' Method: OPTIONS
[Jan 17 09:43:58] VERBOSE[3240] chan_sip.c: Really destroying SIP dialog '2cedf6cf53267f872a0027ec7c92eb96@74.54.54.178' Method: OPTIONS
[Jan 17 09:43:58] VERBOSE[3240] chan_sip.c: Really destroying SIP dialog '3dd1a1a77106bb2f4c8f410775d47ca9@127.0.0.1' Method: REGISTER
[Jan 17 09:43:58] VERBOSE[3240] chan_sip.c: Really destroying SIP dialog '65acdcb006ba7b6a37a819f803a7f056@127.0.0.1' Method: REGISTER
[Jan 17 09:44:02] VERBOSE[3240] chan_sip.c: Reliably Transmitting (no NAT) to 74.54.54.178:5060:
OPTIONS sip:dallas.voip.ms SIP/2.0
Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK36bce6e5
Max-Forwards: 70
From: "Unknown" <sip:148336@192.168.1.199>;tag=as1b4b38f5
To: <sip:dallas.voip.ms>
Contact: <sip:148336@192.168.1.199:5060>
Call-ID: 60bab4434040e980695da1c457382955@192.168.1.199:5060
CSeq: 102 OPTIONS
User-Agent: UCX-3.0(1.8.24.0)
Date: Fri, 17 Jan 2014 14:44:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

I can't read this but I'm concerned when I see Really destroying sip dialog

Voice Connect Plus, Inc.
have a great day and a better tomorrow
 
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Are your SIP phones configured to register with your SIP provider or with UCx? If you still have them configured to register with the SIP provider, there is no point to look at logs on UCx. If you changed them to register with UCx, what are the credentials (user name, authentication user name, server host name or IP address, server port) that you used?

BTW - if you look closely, dutchie correctly stated that you should use the command "sip set debug on" (not "sip set debug") and also recommended to use "sip set debug off" when you're done. He also indicated that 100 lines is not enough - you should look at the last maybe 5000 lines and search of the IP address of your SIP phone(s). If you don't find any log entries with IP addresses of the phones, that would be an indication that the phones are not configured to communicate with UCx or that there is no network connectivity from your phones to the UCx.
 
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Hey UCXGUY, Yeah I did notice that he said that. I misread it. The yealink is configured register name and user name 205 - password is same as secret in sip extension on pbx and sip server is ip address of ucx50.

Voice Connect Plus, Inc.
have a great day and a better tomorrow
 
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This is for anyone...I just ran a filter of ext. 205 in CDR reports and it clearly shows activity on that extension up until the 14th of Jan. Is there anyway to input that time frame into the ucx to try and isolate when the problem occurred????

Voice Connect Plus, Inc.
have a great day and a better tomorrow
 
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I noticed that sip domain is not enabled when i request sip domains in CLI. Would this be an issue and hinder registration??

Voice Connect Plus, Inc.
have a great day and a better tomorrow
 
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My guess is your problem is the configuration of your phones - you do not have them configured to register with UCx. You didn't reply to questions that I asked, so it's somewhat difficult to help you.

My suggestion for you is to install a SIP soft phone (X-Lite or Zoiper) on your PC, configure it to register with UCx and try if the soft phone will work properly. Instructions how to do that are included in the E-MetroTel documentation. Once you have the soft phone working, you can configure your SIP phones the same way.

You could also watch the following video to see what is needed to set up a SIP phone on UCx:
 
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Hey ucxguy, I did reply to your question.Check above....I also copied this from Global Settings. It appears that there might be a few parameters that should be changed.. Any thoughts


Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: UCX-3.0(1.8.24.0)
SDP Session Name: Asterisk PBX 1.8.24.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Legacy userfield parse: No
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Voice Connect Plus, Inc.
have a great day and a better tomorrow
 
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In my opinion, you should be looking at the configuration of your phones - not at the configuration of UCx. Do you have them configured to register with UCx? Did you find their IP addresses in the log (after you execute "sip set debug on")? Did you try a SIP soft phone?
 
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The yealink is configured register name and user name 205 - password is same as secret in sip extension on pbx and sip server is ip address of ucx50 port 5060 also I don't see anything in documentation relating to setting up soft phone.

Voice Connect Plus, Inc.
have a great day and a better tomorrow
 
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I already had the zoiper on my pc. So I configured it and to register to the ucx ip address as domain and added the sip extension in pbx and nothing!!!! I can see it attempting registration but that's just about it.. Thanks I appreciate your help man..

Voice Connect Plus, Inc.
have a great day and a better tomorrow
 
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Did you review the log file to determine if UCx can see packets from the soft phone?

Just wondering - did you by any chance enable firewall on UCx without a rule to allow SIP signaling packets? If you have firewall active, deactivate it (Security - Firewall - Firewall Rules) and try Zoiper or your phones again.
 
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Hey Ucxguy, that was exactly the issue. I had a feeling something was blocking it but I thought maybe it was the firewall of my router. Just as I was getting ready to check that, I saw your email. All the phones are up and running. I was working with Chris from emetrotel this morning and didn't know if we should deactivate that or not. However, the firewall is deactivated but a rule needs to be created to allow sip signaling packets and then re activate firewall.. Many, many thanks. Definitely know another place to look in the future. You wouldn't know how to create rule, wouldya??:)

Voice Connect Plus, Inc.
have a great day and a better tomorrow
 
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To create a new rule, activate the firewall, click on the New Rule button, select protocol UDP, source port Any, destination port SIP (all other fields leave at defaults), press Save and move the created rule up so it is used (I have a SIP rule at position 3).
 
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Done brother and it works like a charm. New rule created and firewall activated...It's always a challenge when when moving from one phase to another. Stepping up and meeting that challenge is what sets us apart from the rest and allows us to grow!!! We both embraced that and it demanded both time and patience.. I thank you for all of your help ucxguy and the rest of the tek-tip people who offered their input. Shout out to Chris at emetrotel for rising up early on a Saturday morning and giving of his time when their are other issues on his plate...Man it has been a long week!!!!!
TO WHOM MUCH IS GIVEN, MUCH IS EXPECTED

Voice Connect Plus, Inc.
have a great day and a better tomorrow
 
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Also, I took the liberty of reconfiguring the Yealink back to it's original state for verification purposes only. I set it up as follows
register name and user name 205 - password is same as secret in sip extension on pbx and sip server is ip address of provider port 5060 and outbound proxy is ucx ip address port 5060. I did this for both yealink and snom phones. Both registered and online with ucx. Now I don't know why it does but obviously there are two different config's that work.....Something for the guys at emetrotel to research!!!!

Voice Connect Plus, Inc.
have a great day and a better tomorrow
 
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