[Asterisk-Users] Siupra-2002 with astersik

Fri Sep 9 04:16:41 MST 2005

Ahh wow.. that dial plan is seriously messed up... Try the default
one... it will work alot better and give you less lag time between
dialing a number and actually going through.

On 9/8/05, Joseph <syscon at interbaun.com> wrote:
> On Thu, 2005-09-08 at 23:29 +0200, Sander wrote:
> >  What is your problem with asterisk ans sipura ? Config files ?? Settings
> > Give some more info on the problems
> 
> Sipura-2002 CAN NOT dial out, incoming call works OK.
> I just got a new Sipura-2002 to my collection (I have few Sipura-3000
> units that work OK).
> I setup the unit, Sipura-2002 to register with Asterisk and it registers
> OK.
> The unit will accept the call but I can not make a call out.
> 
> My sip.conf entry:
> [SPA-2] ; incoming/outgoing calls on FXS Sipura-2002-Line1 ext.711
> type=friend
> secret=711
> username=711
> mailbox=711
> host=dynamic
> port=5068 ; port on FXS line
> dtmfmode=rfc2833
> nat=no
> context=incoming
> callgroup=1
> pickupgroup=1
> 
> Dial Plan on Sipura-2002:
> (xx.|*xx.|#xx.) (this dial plan works OK on Sipura-3000)
> 
> I tried to compare the setup of 2002 unit to 3000 but I can not find
> anything that would be blocking outgoing calls.
> The firmware on Sipura-2002: Software Version:3.1.5
> 
> When I try to make a call out the asterisk is not registering anything
> on the command line from the unit. When I turn the SIP Debugging:
> SIP Debugging Enabled for IP: 10.0.0.155:5068
> ----------- debug output ---------------
> Sip read:
> INVITE sip:321 at 10.0.0.103 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.155:5068;branch=z9hG4bK-ceffa3af
> From: <sip:SPA-2 at 10.0.0.103>;tag=e96b9f56902aab60o0
> To: <sip:321 at 10.0.0.103>
> Call-ID: 53bc6f0e-d4d5f08 at 10.0.0.155
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: <sip:SPA-2 at 10.0.0.155:5068>
> Expires: 240
> User-Agent: Sipura/SPA2002-3.1.5
> Content-Length: 420
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> Content-Type: application/sdp
> 
> v=0
> o=- 1015871 1015871 IN IP4 10.0.0.155
> s=-
> c=IN IP4 10.0.0.155
> t=0 0
> m=audio 16434 RTP/AVP 0 2 4 8 18 96 97 98 100 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/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:100 NSE/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
> 
> 14 headers, 19 lines
> Using latest request as basis request
> Sending to 10.0.0.155 : 5068 (non-NAT)
> Reliably Transmitting (no NAT):
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 10.0.0.155:5068;branch=z9hG4bK-ceffa3af
> From: <sip:SPA-2 at 10.0.0.103>;tag=e96b9f56902aab60o0
> To: <sip:321 at 10.0.0.103>;tag=as3395f791
> Call-ID: 53bc6f0e-d4d5f08 at 10.0.0.155
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:321 at 10.0.0.103>
> Proxy-Authenticate: Digest realm="asterisk", nonce="05664a87"
> Content-Length: 0
> 
> 
>  to 10.0.0.155:5068
> Scheduling destruction of call '53bc6f0e-d4d5f08 at 10.0.0.155' in 15000 ms
> Found user 'SPA-2'
> syscon2*CLI>
> 
> Sip read:
> ACK sip:321 at 10.0.0.103 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.155:5068;branch=z9hG4bK-ceffa3af
> From: <sip:SPA-2 at 10.0.0.103>;tag=e96b9f56902aab60o0
> To: <sip:321 at 10.0.0.103>;tag=as3395f791
> Call-ID: 53bc6f0e-d4d5f08 at 10.0.0.155
> CSeq: 101 ACK
> Max-Forwards: 70
> Contact: <sip:SPA-2 at 10.0.0.155:5068>
> User-Agent: Sipura/SPA2002-3.1.5
> Content-Length: 0
> 
> 
> 10 headers, 0 lines
> syscon2*CLI>
> 
> Sip read:
> INVITE sip:321 at 10.0.0.103 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.155:5068;branch=z9hG4bK-93cd1d87
> From: <sip:SPA-2 at 10.0.0.103>;tag=e96b9f56902aab60o0
> To: <sip:321 at 10.0.0.103>
> Call-ID: 53bc6f0e-d4d5f08 at 10.0.0.155
> CSeq: 102 INVITE
> Max-Forwards: 70
> Proxy-Authorization: Digest
> username="SPA-2",realm="asterisk",nonce="05664a87",uri="sip:321 at 10.0.0.103",algorithm=MD5,response="da6bd6dd8a890f2e37a88ff339ec0419"
> Contact: <sip:SPA-2 at 10.0.0.155:5068>
> Expires: 240
> User-Agent: Sipura/SPA2002-3.1.5
> Content-Length: 420
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> Content-Type: application/sdp
> 
> v=0
> o=- 1015871 1015871 IN IP4 10.0.0.155
> s=-
> c=IN IP4 10.0.0.155
> t=0 0
> m=audio 16434 RTP/AVP 0 2 4 8 18 96 97 98 100 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/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:100 NSE/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
> 
> 15 headers, 19 lines
> Using latest request as basis request
> Sending to 10.0.0.155 : 5068 (non-NAT)
> Found user 'SPA-2'
> Found RTP audio format 0
> Found RTP audio format 2
> Found RTP audio format 4
> Found RTP audio format 8
> Found RTP audio format 18
> Found RTP audio format 96
> Found RTP audio format 97
> Found RTP audio format 98
> Found RTP audio format 100
> Found RTP audio format 101
> Peer audio RTP is at port 10.0.0.155:16434
> Found description format PCMU
> Found description format G726-32
> Found description format G723
> Found description format PCMA
> Found description format G729a
> Found description format G726-40
> Found description format G726-24
> Found description format G726-16
> Found description format NSE
> Found description format telephone-event
> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51d
> (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc
> (ulaw|alaw)
> Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
> 0x1 (g723)
> Looking for 321 in incoming
> Reliably Transmitting (no NAT):
> SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 10.0.0.155:5068;branch=z9hG4bK-93cd1d87
> From: <sip:SPA-2 at 10.0.0.103>;tag=e96b9f56902aab60o0
> To: <sip:321 at 10.0.0.103>;tag=as3395f791
> Call-ID: 53bc6f0e-d4d5f08 at 10.0.0.155
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:321 at 10.0.0.103>
> Content-Length: 0
> 
> 
>  to 10.0.0.155:5068
> syscon2*CLI>
> 
> Sip read:
> ACK sip:321 at 10.0.0.103 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.155:5068;branch=z9hG4bK-93cd1d87
> From: <sip:SPA-2 at 10.0.0.103>;tag=e96b9f56902aab60o0
> To: <sip:321 at 10.0.0.103>;tag=as3395f791
> Call-ID: 53bc6f0e-d4d5f08 at 10.0.0.155
> CSeq: 102 ACK
> Max-Forwards: 70
> Proxy-Authorization: Digest
> username="SPA-2",realm="asterisk",nonce="05664a87",uri="sip:321 at 10.0.0.103",algorithm=MD5,response="2659e6c5135e18723ec0eb769fc7db49"
> Contact: <sip:SPA-2 at 10.0.0.155:5068>
> User-Agent: Sipura/SPA2002-3.1.5
> Content-Length: 0
> 
> 
> 11 headers, 0 lines
> Destroying call '53bc6f0e-d4d5f08 at 10.0.0.155'
> ------- end debug output ---------------
> 
> --
> #Joseph
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