[Asterisk-Users] Siupra-2002 with astersik

Fri Sep 9 09:50:20 MST 2005

The original dial plan was:
(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

So I change it to:
(xx.|*xx.|#xx.)

I don't think it is complicated, beside it works with Sipura-3000, and I
don't see a reason why shouldn't it work with Sipura-2002.

I contact Sipura technical support but they didn't solve my problem yet.

-- 
#Joseph

On Fri, 2005-09-09 at 07:16 -0400, Matt wrote:
> 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
> sers