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Topic # 107464 12-Aug-2012 18:53 Send private message

Testing setup of a UC320 device. Using latest available firmwares etc. All settings are pretty much standard. I cannot dial 111.
If I dial 1-111 (i.e. 1 for outside line) I get does not exist anywhere
If I dial 111 (i.e. ignore the 1 for outside line) I get service unavailable

Also I can't dial 082210 - VFX portal = service unavailable
All other numbers I have tested work fine.

Any ideas?


Logs:
Aug 12 18:44:13 UC320W user.debug voice: INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.7.13:5060;branch=z9hG4bK-f8e02158

From: "name" <sip:[email protected]>;tag=a2accf20261c9446o5;ref=200

To: <sip:[email protected]>

Remote-Party-ID: "name" <sip:[email protected]>;screen=yes;party=calling

Call-ID: [email protected]

CSeq: 101 INVITE

Max-Forwards: 70

Contact: "name" <sip:[email protected]:5060;ref=7nnnnnnn>

Expires: 240

User-Agent: Cisco/UC320W-2.2.2(1)

Allow-Events: talk, hold, conference, x-spa-cti

Content-Length: 395

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

Supported: x-sipura, replaces

Content-Type: application/sdp

 

v=0

o=- 187653 187653 IN IP4 192.168.7.13

s=-

c=IN IP4 192.168.7.13

t=0 0

m=audio 16456 RTP/AVP 8 0 2 9 18 96 97 98 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:2 G726-32/8000

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

Aug 12 18:44:13 UC320W user.debug voice: SIP/2.0 100 Trying

From: "Name" <sip:[email protected]>;tag=a2accf20261c9446o5;ref=200

To: <sip:[email protected]>

Call-ID: [email protected]

CSeq: 101 INVITE

Via: SIP/2.0/UDP 192.168.7.13:5060;branch=z9hG4bK-f8e02158;rport=38244;received=58.28.nnn.nn

Content-Length: 0

Aug 12 18:44:13 UC320W user.debug voice: SIP/2.0 604 Does not exist anywhere

From: "name" <sip:[email protected]>;tag=a2accf20261c9446o5;ref=200

To: <sip:[email protected]>;tag=3201400a-13c4-502750bd-3dc471e5-2ec4ba72

Call-ID: [email protected]

CSeq: 101 INVITE

Via: SIP/2.0/UDP 192.168.7.13:5060;branch=z9hG4bK-f8e02158;rport=38244;received=58.28.nnn.nn

Content-Length: 0

Aug 12 18:44:13 UC320W user.debug voice: ACK sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.7.13:5060;branch=z9hG4bK-f8e02158

From: "name" <sip:[email protected]>;tag=a2accf20261c9446o5;ref=200

To: <sip:[email protected]>;tag=3201400a-13c4-502750bd-3dc471e5-2ec4ba72

Call-ID: [email protected]

CSeq: 101 ACK

Max-Forwards: 70

Contact: "name" <sip:[email protected]:5060;ref=7nnnnnnn>

User-Agent: Cisco/UC320W-2.2.2(1)

Allow-Events: talk, hold, conference, x-spa-cti

Content-Length: 0




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219 posts

Master Geek
+1 received by user: 21


  Reply # 720378 20-Nov-2012 21:20 Send private message

As an extra bit of information, whenever you try to dial out Fuzzerbear, it adds +64 in front of your number that you dial, so if you find a way to take that prefix out it should be ok.

Ok a google search yields :

http://www.cisco.com/en/US/docs/voice_ip_comm/unified_communications/csbuc300/html/uc320w_ag_78-19371en.htm

under sip/bri trunks and then sip provider settings look for this option :

"Prefix dialed numbers with: Check the box if this feature is required by your service provider. When this feature is enabled, the system automatically inserts the appropriate dialing prefix to an external number (such as +1 in the US or +44 in the UK, based on the specified Regional Settings)."

Yours is dialling +64, if you can turn this off, it should fix your issue.

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