DID problem

thekrazytaco · ‎03-07-2012

Hello there,

I've got a uc320 with a esw 24 port poe switch, set up with a sip service provider for 1 trunk. We have 20 numbers available to set up for direct inward dialing but none of the numbers seem to work other then the first one on the list. All incoming and outgoing calls work perfectly, I just can't seem to assign other DID numbers correctly. When I do assign them and call in they just give a busy signal. I've made seperate call routing rules and have tried "1" and "+1" before the DID numbers.

I called my sip service provider and they stated that all the 20 numbers are up and running and registered to my sip log in credentials. Is there some other setting I'm missing, or do I need to ask my sip provider to enable a certain setting?

Thanks for any help.

Christopher Edgeworth · ‎03-07-2012

Hello,

First thing you should do is turn on your SIP trunk logs. To do this log into the Configuration Utility and go to Status -> Support Tools -> Log. Enable syslog and then check the box next to your SIP trunk.

I would suggest clearly the logs for easier readability and then making a test call into the device to one of the secondary configured SIP trunk numbers. After the call is complete (recieved busy signal), look in the logs for the incoming SIP INVITE message. It will looks something like this:

Jan 30 14:27:11 UC320W user.debug voice: INVITE sip:4089629001@10.1.1.1:5084 SIP/2.0

Via: SIP/2.0/UDP 10.1.1.117:5060;branch=z9hG4bK-f3027c7a

From: "Main Reception 2" <401>;tag=50e150f532d93042o0

To: <>4089629001@10.1.1.1>

Remote-Party-ID: "Main Reception 2" <401>;screen=yes;party=calling

Call-ID: 3e116931-99df9536@10.1.1.117

CSeq: 103 INVITE

Max-Forwards: 70

Contact: "Main Reception 2" <401>

Expires: 240

User-Agent: Cisco/SPA509G-7.4.7v26

P-Mailbox: 5401

Content-Length: 391

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE

Allow-Events: dialog

Supported: replaces

Content-Type: application/sdp

v=0

o=- 828346 828346 IN IP4 10.1.1.117

s=-

c=IN IP4 10.1.1.117

t=0 0

m=audio 16418 RTP/AVP 0 2 8 9 18 96 97 98 101

a=rtpmap:0 PCMU/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:18 G729a/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

You want to make sure that the number used in your route matches the phone number (bolded) in the TO: line.

Hope this helps

Chris

NOTE: Phone numbers in this invite were modified from the original capture.

thekrazytaco · ‎03-08-2012

Thanks for the reply Chris. I looked at the logs and in the TO: field it just shows

To: for DID calls and only give busy signals. The X's being my sip providers proxy IP address. This is the case whenever I call any number that is DID under my main line number. When I call the main line it shows

To: <>zzz-zzz-zzzz@xxx.xx.xxx.xx> The Z's being my main line number. So the DID numbers are absent in the To: field if it is a DID number. I'm using a uc320 with an esw switch and spa504 phones. I plugged a phone directly into the uc320 to see if that made a difference and it didn't, while the rest are still behind the switch. Would it mean my sip provider hasn't activated the DID lines if they are absent in the To: field? They assured me that they were set up, but like I said I still only get calls on the main line and busy signal on any DID ones. I don't believe there are any commands to assign DID's with a uc320 since its a gui system? Other then what I set up with the inbound call routing rules.

Christopher Edgeworth · ‎03-09-2012

Hi,

OK that helps a little. If you will notice in my post above the phone number was bolded in two places. If you still have the invite or can make another call what do you see in the position of the first occurance of the phone number in the invite?

Thanks,

Chris