SPA-3102 - Sipgate + VoIpDiscount - Page 3

onlineuser · ‎05-30-2013

Hi,

I have connected the SPA-3102 to extension 2 of my telephone-system.

Now when I dial the extension from the SPA-3102, I can make phonecalls via SIP.

Now I want to use sipgate (inbound and outbound) and voipdiscount (only outbound) on my SPA-3102. But all other analogue extension should be able to make phonecalls via SPA.

So, when any extension rings the SPA, it picks up the call and I hear the dialtone from SIP.

How is it possible that the user can choose between sipgate and voipdiscount for outgoing calls?

When a inbound-call via sipgate arrives the SPA the call should be forwarded to extension 3 on the pstn-line from the telephone-system. How can SPA-3102 manage this configuration?

Regards

onlineuser · ‎06-30-2013

You tried the dial plan that sends the call to gw1 instead of the Line 1 configuration, the (xx.<:@gw1) dial plan? Yes or No!

If yes then try it with Line 1 set to Register: No, Make Call Without Reg: Yes. Do a test to see if voipdiscount requires registration. Before you make the call wait for the Line 1 registration to time out and expire.

My tests to my voipbuster account, which is one of the Betamax sisters, works fine every time. I can even change the proxy to sip.voipdiscount.com and it works fine.

No. I only tried to setup Line 1 with voipdiscount and I tried to make a call via the phone I connected to Line port of SPA-3102. There I didn't need a dialplan. Because I could make a outgoing call via Line 1 phone I asked for the dialplan for forwarding the PSTN line to Line 1 (also analogue system call forwarding to Line 1).

Before I surely used the dialplan (<#9,:>xx.<:@gw1>|xx.) and I also tried (xx.<:@gw1>) to send all outgoing calls via voipdiscount.

I also noticed that the resolved domains (voipdiscount, voipbuster, 12voip) points to the same IPs. So all or most betamax calls goes through the same servers.

Also, when Line 1 is registered to voipdiscount (phone on Line 1 works fine) the forwarding from PSTN to gw1 not works. I also tried the same with disabled registration for Line 1. Now I have tried all combinations :-)

This log comes from enabled Line 1 but registration disabled - same comes when registration is enabled.

[0]Off Hook

Calling:0800....@sip.voipdiscount.com:0

[0:0]AUD ALLOC CALL (port=16470)

[0:0]RTP Rx Up

SDP RTPMAP 101 --> 136

[0:0]ENC INIT 0

[0:0]RTP Tx Up (pt=0->4d48a81c:41590)

[0:0]RTCP Tx Up

CC:CallProgress

[0]adp line session start

[0:0]RTP Rx 1st PKT @16470(2)

[0:0]DEC INIT 0

SDP RTPMAP 101 --> 136

[0:0]RTP Tx Dn

[0]adp line session stop

--------------------------

[0] duration:2 s

path:nb_in, did:50, start at 0 s

path:nb_out, did:51, start at 0 s

path:full, did:52, start at 0 s

--------------------------

[0:0]ENC INIT 0

[0:0]RTP Tx Up (pt=0->4d48a81c:41590)

CC:Remote Resume

CC:Connected

[0]adp line session start

There no authentication will be send! I don't understand this! How can voipdiscount server decide if the call will be accepted - or is it because it is a free call!? Ok I also tried to call a non-free number - same log - not authentication.

This is really strange - although registration for voipdiscount I can't see any entry in logging.

You tried it with voipbuster on gw1 and you have used the same dialplan for outgoing calls (<#9,:>xx.<:@gw1>|xx.)?

I also could register a voipbuster account - but I think it will be the same problem. I tried it with voipbuster as gw1 - same problem! :-(

Do you have any other idea?

Howard Wittenberg · ‎06-30-2013

With the log you posted on the Line 1 Tab you did not set Sip Debug Option FULL which I mentioned in the Jun 29, 2013 10:58 AM posting. If you had made that setting you would have seen the sip INVITE detail.

There no authentication will be send! I don't understand this! How can voipdiscount server decide if the call will be accepted - or is it because it is a free call!? Ok I also tried to call a non-free number - same log - not authentication.

The authentication comes in the 2d sip INVITE that is sent after receiving the 401 Unauthorized response to the first sip INVITE.

Do you have any other idea?

If you can make a voipdiscount call from the phone attached to Line 1, as you reported, with Line 1 registered to voipdiscount but you cannot make a call when you change the Line 1 Dial plan to (xx.<:@gw1>) then you have something incorrect in the Gateway 1 configuration entries.

onlineuser · ‎06-30-2013

Gateway 1: voipdiscountUsername@sip.voipdiscount.com

GW1 NAT: yes

GW1 Auth ID: voipdiscountUsername

GW1 Password: password from voipdiscount

All the other parameters from Line 1 are the same like while testing it with attached phone on Line1 port.

I noticed follwing. When make a outgoing call via sipgate and hang up the analogue extension phone it will take a while before I can redial because the SPA waits for checking the disconnect tone that the SPA can also hang up. This takes 2 or 3 seconds.

When I try to make a outgoing call via gw1 and I hang up the phone I can immediately make a new call. Can it be that the connection will be disconnected before the voipdiscount call is connected?

In Info-tab the status is "Calling" while the SPA sends the Invite packets.

Howard Wittenberg · ‎06-30-2013

Gateway 1: voipdiscountUsername@sip.voipdiscount.com

GW1 NAT: yes

GW1 Auth ID: voipdiscountUsername

GW1 Password: password from voipdiscount

That is the way you do it.

I want to see a sip debug trace of a call from a phone attached to the SPA3102 (Line 1) plus a trace of a call from the same phone attached to the SPA3102 dialing the call using a line 1 dial plan (xx.<:@gw1>) where the call is sent using the configuration you have on Gateway 1.

The trace with the Debug Level set to 3 on the System Tab and Sip Debug Option set to FULL on the Line 1 Tab.

I believe you said the latter call did not work.

I noticed follwing. When make a outgoing call via sipgate and hang up the analogue extension phone it will take a while before I can redial because the SPA waits for checking the disconnect tone that the SPA can also hang up. This takes 2 or 3 seconds.

When I try to make a outgoing call via gw1 and I hang up the phone I can immediately make a new call. Can it be that the connection will be disconnected before the voipdiscount call is connected?

In Info-tab the status is "Calling" while the SPA sends the Invite packets.

I am not sure of the significance of the difference. I would think the disconnect should be the same one way or the other. Probably is different because one call connected and the other one did not.

onlineuser · ‎07-01-2013

Ok, here is the sylog of making a voipdiscount call from phone connected to Line 1 with enabled voipdiscount-proxy. It also work without proxy.

SYSCFG:WARMBOOT 24 83

[0]->77.72.169.134:5060(516)

REGISTER sip:sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP 192.168.11.6:5061;branch=z9hG4bK-5103e995;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=cb95b2b62f61b45o0

To: <>voipdiscountID@sip.voipdiscount.com>

Call-ID: 659879d5-d3647af4@192.168.11.6

CSeq: 54399 REGISTER

Max-Forwards: 70

Contact: ;expires=3600

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

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

Supported: x-sipura, replaces

[0]<<77.72.169.134:5060(515)

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.11.6:5061;branch=z9hG4bK-5103e995;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=cb95b2b62f61b45o0

To: <>voipdiscountID@sip.voipdiscount.com>

Contact: sip:77.72.169.134:5060

Call-ID: 659879d5-d3647af4@192.168.11.6

CSeq: 54399 REGISTER

Server: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

WWW-Authenticate: Digest realm="sip.voipdiscount.com",nonce="1842827359",algorithm=MD5

Content-Length: 0

[0]->77.72.169.134:5060(698)

REGISTER sip:sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP 192.168.11.6:5061;branch=z9hG4bK-ffa10baa;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=cb95b2b62f61b45o0

To: <>voipdiscountID@sip.voipdiscount.com>

Call-ID: 659879d5-d3647af4@192.168.11.6

CSeq: 54400 REGISTER

Max-Forwards: 70

Authorization: Digest username="voipdiscountID",realm="sip.voipdiscount.com",nonce="1842827359",uri="sip:sip.voipdiscount.com",algorithm=MD5,response="d7628c63131e8d872fa3612a7a6551ae"

Contact: ;expires=3600

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

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

Supported: x-sipura, replaces

[0]<<77.72.169.134:5060(421)

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.11.6:5061;branch=z9hG4bK-ffa10baa;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=cb95b2b62f61b45o0

To: <>voipdiscountID@sip.voipdiscount.com>

Contact: sip:77.72.169.134:5060

Call-ID: 659879d5-d3647af4@192.168.11.6

CSeq: 54400 REGISTER

Server: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Content-Length: 0

[0]<<77.72.169.134:5060(442)

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.11.6:5061;branch=z9hG4bK-ffa10baa;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=cb95b2b62f61b45o0

To: <>voipdiscountID@sip.voipdiscount.com>

Contact: ;expires=3600

Call-ID: 659879d5-d3647af4@192.168.11.6

CSeq: 54400 REGISTER

Server: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Content-Length: 0

[0]RegOK. NextReg in 3570 (1)

[1]RegOK. NextReg in 3570 (1)

AUD:Stop PSTN Tone

IDBG: st-0

fs:11892:11965:65536

fls:af:1:0:0

fbr:1:3000:3000:1667b:0004:0005:5.2.13(GW002)

fhs:01:0:0001:upg:app:0:3.3.6(GW)

fhs:02:0:0002:upg:app:1:3.3.6(GW)

fhs:03:0:0003:upg:app:2:3.3.6(GW)

fhs:04:0:0004:upg:app:0:5.2.13(GW002)

fhs:05:0:0005:upg:app:1:5.2.13(GW002)

fhs:06:0:0006:upg:app:2:5.2.13(GW002)

PLKUP: 2048, 768, 11, 1.5

fu:1:66a2, 0003 0001

[0]Off Hook

[0]Reg Addr Change(0) 0:0->4d48a986:5060

[0]->77.72.169.134:5060(413)

NOTIFY sip:sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-584396fa;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=cb95b2b62f61b45o0

To:

Call-ID: 132fe343-9fc7720d@192.168.11.6

CSeq: 1 NOTIFY

Max-Forwards: 70

Contact:

Event: keep-alive

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[0]<<77.72.169.134:5060(463)

SIP/2.0 200 Ok

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-584396fa;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=cb95b2b62f61b45o0

To:

Contact: sip:77.72.169.134:5060

Call-ID: 132fe343-9fc7720d@192.168.11.6

CSeq: 1 NOTIFY

Supported:

User-Agent: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Accept: application/sdp

Accept-Encoding:

Accept-Language:

Calling:0800....@sip.voipdiscount.com:0

[0:0]AUD ALLOC CALL (port=16410)

[0:0]RTP Rx Up

[0]->77.72.169.134:5060(1083)

INVITE sip:0800....@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-52e8ee16;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=252efeb65cf05f5eo0

To: <0800....>

Remote-Party-ID: <>voipdiscountID@sip.voipdiscount.com>;screen=yes;party=calling

Call-ID: fa701396-bea74ffe@192.168.11.6

CSeq: 101 INVITE

Max-Forwards: 70

Contact:

Expires: 240

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 442

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

Supported: x-sipura, replaces

Content-Type: application/sdp

v=0

o=- 3101 3101 IN IP4 mypublicIP

s=-

c=IN IP4 mypublicIP

t=0 0

m=audio 16410 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=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

[0]<<77.72.169.134:5060(525)

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-52e8ee16;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=252efeb65cf05f5eo0

To: <0800....>

Contact: sip:0800....@77.72.169.134:5060

Call-ID: fa701396-bea74ffe@192.168.11.6

CSeq: 101 INVITE

Server: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

WWW-Authenticate: Digest realm="sip.voipdiscount.com",nonce="1793830031",algorithm=MD5

Content-Length: 0

[0]->77.72.169.134:5060(413)

ACK sip:0800....@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-52e8ee16;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=252efeb65cf05f5eo0

To: <0800.....>

Call-ID: fa701396-bea74ffe@192.168.11.6

CSeq: 101 ACK

Max-Forwards: 70

Contact:

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[0]->77.72.169.134:5060(1276)

INVITE sip:0800....@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-573934fa;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=252efeb65cf05f5eo0

To: <0800....>

Remote-Party-ID: <>voipdiscountID@sip.voipdiscount.com>;screen=yes;party=calling

Call-ID: fa701396-bea74ffe@192.168.11.6

CSeq: 102 INVITE

Max-Forwards: 70

Authorization: Digest username="voipdiscountID",realm="sip.voipdiscount.com",nonce="1793830031",uri="sip:0800....0@sip.voipdiscount.com",algorithm=MD5,response="b8d4d65139c6ff4377b8385b87431d1e"

Contact:

Expires: 240

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 442

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

Supported: x-sipura, replaces

Content-Type: application/sdp

v=0

o=- 3101 3101 IN IP4 mypublicIP

s=-

c=IN IP4 mypublicIP

t=0 0

m=audio 16410 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=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

[0]<<77.72.169.134:5060(431)

SIP/2.0 100 Trying

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-573934fa;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=252efeb65cf05f5eo0

To: <0800....>

Contact: sip:0800....@77.72.169.134:5060

Call-ID: fa701396-bea74ffe@192.168.11.6

CSeq: 102 INVITE

Server: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Content-Length: 0

[0]<<77.72.169.134:5060(431)

SIP/2.0 100 Trying

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-573934fa;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=252efeb65cf05f5eo0

To: <0800....>

Contact: sip:0800....@77.72.169.134:5060

Call-ID: fa701396-bea74ffe@192.168.11.6

CSeq: 102 INVITE

Server: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Content-Length: 0

SDP RTPMAP 101 --> 136

[0]<<77.72.169.134:5060(701)

SIP/2.0 183 Session progress

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-573934fa;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=252efeb65cf05f5eo0

To: <0800....>;tag=720313ac51b8481f98c19

Contact: sip:0800....@77.72.169.134:5060

Call-ID: fa701396-bea74ffe@192.168.11.6

CSeq: 102 INVITE

Server: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Content-Type: application/sdp

Content-Length: 201

v=0

o=voipdiscountID 1372703499 1372703499 IN IP4 77.72.168.67

s=SIP Call

c=IN IP4 77.72.168.67

t=0 0

m=audio 24472 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=ptime:20

[0:0]ENC INIT 0

[0:0]RTP Tx Up (pt=0->4d48a843:24472)

[0:0]RTCP Tx Up

CC:CallProgress

[0:0]RTP Rx 1st PKT @16410(2)

[0:0]RxBigGapSeqNo:9436 21245

[0:0]DEC INIT 0

SDP RTPMAP 101 --> 136

[0]<<77.72.169.134:5060(687)

SIP/2.0 200 Ok

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-573934fa;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=252efeb65cf05f5eo0

To: <0800....>;tag=720313ac51b8481f98c19

Contact: sip:0800.....@77.72.169.134:5060

Call-ID: fa701396-bea74ffe@192.168.11.6

CSeq: 102 INVITE

Server: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Content-Type: application/sdp

Content-Length: 201

v=0

o=voipdiscountID 1372703502 1372703502 IN IP4 77.72.168.67

s=SIP Call

c=IN IP4 77.72.168.67

t=0 0

m=audio 24472 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=ptime:20

[0:0]RTP Tx Dn

[0:0]ENC INIT 0

[0:0]RTP Tx Up (pt=0->4d48a843:24472)

CC:Remote Resume

CC:Connected

[0]->77.72.169.134:5060(630)

ACK sip:0800....@77.72.169.134:5060 SIP/2.0

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-ee0fff47;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=252efeb65cf05f5eo0

To: <0800....>;tag=720313ac51b8481f98c19

Call-ID: fa701396-bea74ffe@192.168.11.6

CSeq: 102 ACK

Max-Forwards: 70

Authorization: Digest username="voipdiscountID",realm="sip.voipdiscount.com",nonce="1793830031",uri="sip:0800....@sip.voipdiscount.com",algorithm=MD5,response="b8d4d65139c6ff4377b8385b87431d1e"

Contact:

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[0]->77.72.169.134:5060(413)

NOTIFY sip:sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-55b47195;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=cb95b2b62f61b45o0

To:

Call-ID: 132fe343-9fc7720d@192.168.11.6

CSeq: 2 NOTIFY

Max-Forwards: 70

Contact:

Event: keep-alive

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[0]<<77.72.169.134:5060(463)

SIP/2.0 200 Ok

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-55b47195;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=cb95b2b62f61b45o0

To:

Contact: sip:77.72.169.134:5060

Call-ID: 132fe343-9fc7720d@192.168.11.6

CSeq: 2 NOTIFY

Supported:

User-Agent: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Accept: application/sdp

Accept-Encoding:

Accept-Language:

[0]On Hook

[0]FM Alert Stop RxTx (c=0024e22c;a=0)

[0:0]AUD Rel Call

[0]->77.72.169.134:5060(581)

BYE sip:0800....@77.72.169.134:5060 SIP/2.0

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-bd817d5c;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=252efeb65cf05f5eo0

To: <0800....>;tag=720313ac51b8481f98c19

Call-ID: fa701396-bea74ffe@192.168.11.6

CSeq: 103 BYE

Max-Forwards: 70

Authorization: Digest username="voipdiscountID",realm="sip.voipdiscount.com",nonce="1793830031",uri="sip:0800....@77.72.169.134:5060",algorithm=MD5,response="f4bf66e708c1fcd94d835e840cb3ff8c"

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[0]adp line session stop

--------------------------

[0] duration:15 s

path:nb_in, did:50, start at 0 s

path:nb_out, did:51, start at 0 s

path:full, did:52, start at 0 s

--------------------------

[0]<<77.72.169.134:5060(450)

SIP/2.0 200 Ok

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-bd817d5c;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=252efeb65cf05f5eo0

To: <0800....>;tag=720313ac51b8481f98c19

Contact: sip:0800....@77.72.169.134:5060

Call-ID: fa701396-bea74ffe@192.168.11.6

CSeq: 103 BYE

Server: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Content-Length: 0

DLG Terminated 2e1b54

[0]Off Hook

[0]->77.72.169.134:5060(413)

NOTIFY sip:sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-26d48863;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=cb95b2b62f61b45o0

To:

Call-ID: 132fe343-9fc7720d@192.168.11.6

CSeq: 3 NOTIFY

Max-Forwards: 70

Contact:

Event: keep-alive

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[0]<<77.72.169.134:5060(463)

SIP/2.0 200 Ok

Via: SIP/2.0/UDP mypublicIP:5061;branch=z9hG4bK-26d48863;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=cb95b2b62f61b45o0

To:

Contact: sip:77.72.169.134:5060

Call-ID: 132fe343-9fc7720d@192.168.11.6

CSeq: 3 NOTIFY

Supported:

User-Agent: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Accept: application/sdp

Accept-Encoding:

Accept-Language:

I tried to enter the dialplan on Line 1 (default is:(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) ).

Making a call works but the call won't be send through gw1.

[0]Off Hook

IDBG: st-0

fs:11831:11905:65536

fls:af:1:0:0

fbr:1:3000:3000:167f3:0004:0005:5.2.13(GW002)

fhs:01:0:0001:upg:app:0:3.3.6(GW)

fhs:02:0:0002:upg:app:1:3.3.6(GW)

fhs:03:0:0003:upg:app:2:3.3.6(GW)

fhs:04:0:0004:upg:app:0:5.2.13(GW002)

fhs:05:0:0005:upg:app:1:5.2.13(GW002)

fhs:06:0:0006:upg:app:2:5.2.13(GW002)

PLKUP: 2048, 768, 11, 1.5

fu:1:681a, 0003 0001

[0]On Hook

[0]Off Hook

[0]Reg Addr Change(0) 0:0->4d48a986:5060

[0]->77.72.169.134:5060(414)

NOTIFY sip:sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5061;branch=z9hG4bK-4e132e21;rport

From: <>voipdiscountUserID@sip.voipdiscount.com>;tag=33fddf49acf735c2o0

To:

Call-ID: 247900a5-eef92416@192.168.11.6

CSeq: 1 NOTIFY

Max-Forwards: 70

Contact:

Event: keep-alive

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[0]<<77.72.169.134:5060(464)

SIP/2.0 200 Ok

Via: SIP/2.0/UDP myPublicIP:5061;branch=z9hG4bK-4e132e21;rport

From: <>voipdiscountUserID@sip.voipdiscount.com>;tag=33fddf49acf735c2o0

To:

Contact: sip:77.72.169.134:5060

Call-ID: 247900a5-eef92416@192.168.11.6

CSeq: 1 NOTIFY

Supported:

User-Agent: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Accept: application/sdp

Accept-Encoding:

Accept-Language:

Calling:0800.....@voipdiscountUserID@sip.voipdiscount.com

[0:0]AUD ALLOC CALL (port=16396)

[0:0]RTP Rx Up

[0]->77.72.169.134:5060(1085)

INVITE sip:0800.....@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5061;branch=z9hG4bK-8810d655;rport

From: <>voipdiscountUserID@sip.voipdiscount.com>;tag=53140ae9469e059eo0

To: <0800.....>

Remote-Party-ID: <>voipdiscountUserID@sip.voipdiscount.com>;screen=yes;party=calling

Call-ID: 14819105-6462d632@192.168.11.6

CSeq: 101 INVITE

Max-Forwards: 70

Contact:

Expires: 240

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 442

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

Supported: x-sipura, replaces

Content-Type: application/sdp

v=0

o=- 2292 2292 IN IP4 myPublicIP

s=-

c=IN IP4 myPublicIP

t=0 0

m=audio 16396 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=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

[0]<<77.72.169.134:5060(527)

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP myPublicIP:5061;branch=z9hG4bK-8810d655;rport

From: <>voipdiscountUserID@sip.voipdiscount.com>;tag=53140ae9469e059eo0

To: <0800.....>

Contact: sip:0800.....@77.72.169.134:5060

Call-ID: 14819105-6462d632@192.168.11.6

CSeq: 101 INVITE

Server: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

WWW-Authenticate: Digest realm="sip.voipdiscount.com",nonce="1792065000",algorithm=MD5

Content-Length: 0

[0]->77.72.169.134:5060(415)

ACK sip:0800.....@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5061;branch=z9hG4bK-8810d655;rport

From: <>voipdiscountUserID@sip.voipdiscount.com>;tag=53140ae9469e059eo0

To: <0800.....>

Call-ID: 14819105-6462d632@192.168.11.6

CSeq: 101 ACK

Max-Forwards: 70

Contact:

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[0]->77.72.169.134:5060(1279)

INVITE sip:0800.....@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5061;branch=z9hG4bK-3df42c89;rport

From: <>voipdiscountUserID@sip.voipdiscount.com>;tag=53140ae9469e059eo0

To: <0800.....>

Remote-Party-ID: <>voipdiscountUserID@sip.voipdiscount.com>;screen=yes;party=calling

Call-ID: 14819105-6462d632@192.168.11.6

CSeq: 102 INVITE

Max-Forwards: 70

Authorization: Digest username="voipdiscountUserID",realm="sip.voipdiscount.com",nonce="1792065000",uri="sip:0800.....@sip.voipdiscount.com",algorithm=MD5,response="f896e2e57b3b20d5ddfb01efdf0e5394"

Contact:

Expires: 240

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 442

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

Supported: x-sipura, replaces

Content-Type: application/sdp

v=0

o=- 2292 2292 IN IP4 myPublicIP

s=-

c=IN IP4 myPublicIP

t=0 0

m=audio 16396 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=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

[0]<<77.72.169.134:5060(433)

SIP/2.0 100 Trying

Via: SIP/2.0/UDP myPublicIP:5061;branch=z9hG4bK-3df42c89;rport

From: <>voipdiscountUserID@sip.voipdiscount.com>;tag=53140ae9469e059eo0

To: <0800.....>

Contact: sip:0800.....@77.72.169.134:5060

Call-ID: 14819105-6462d632@192.168.11.6

CSeq: 102 INVITE

Server: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Content-Length: 0

Here the log from PSTN line with dialplan 1 via gw1 (not successful).

FXO:Start CNDD

AUD:Stop PSTN Tone

FXO:Off Hook

FXO:Stop CNDD

AUD:Play PSTN Tone 1

FXO:Digit=0

AUD:Stop PSTN Tone

FXO:Digit=8

AUD:Stop PSTN Tone

FXO:Digit=0

AUD:Stop PSTN Tone

FXO:Digit=0

.

AUD:Stop PSTN Tone

[0]->77.72.169.134:5060(414)

NOTIFY sip:sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5061;branch=z9hG4bK-c478db7d;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=f8f6fbfddd97a09eo0

To:

Call-ID: f27336b9-64bc4512@192.168.11.6

CSeq: 3 NOTIFY

Max-Forwards: 70

Contact:

Event: keep-alive

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[0]<<77.72.169.134:5060(464)

SIP/2.0 200 Ok

Via: SIP/2.0/UDP myPublicIP:5061;branch=z9hG4bK-c478db7d;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=f8f6fbfddd97a09eo0

To:

Contact: sip:77.72.169.134:5060

Call-ID: f27336b9-64bc4512@192.168.11.6

CSeq: 3 NOTIFY

Supported:

User-Agent: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Accept: application/sdp

Accept-Encoding:

Accept-Language:

AUD:Stop PSTN Tone

Calling:0800.......@myVoipdiscountID@sip.voipdiscount.com

[1:0]AUD ALLOC CALL (port=16440)

[1:0]RTP Rx Up

[1]->77.72.169.134:5060(1085)

INVITE sip:0800.......@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5060;branch=z9hG4bK-ffc8175c;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=ff577530e0b8d335o1

To: <0800.......>

Remote-Party-ID: <>myVoipdiscountID@sip.voipdiscount.com>;screen=yes;party=calling

Call-ID: e7d8fd0c-4ec38d09@192.168.11.6

CSeq: 101 INVITE

Max-Forwards: 70

Contact:

Expires: 240

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 442

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

Supported: x-sipura, replaces

Content-Type: application/sdp

v=0

o=- 5158 5158 IN IP4 myPublicIP

s=-

c=IN IP4 myPublicIP

t=0 0

m=audio 16440 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=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

[1]->77.72.169.134:5060(1085)

INVITE sip:0800.......@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5060;branch=z9hG4bK-ffc8175c;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=ff577530e0b8d335o1

To: <0800.......>

Remote-Party-ID: <>myVoipdiscountID@sip.voipdiscount.com>;screen=yes;party=calling

Call-ID: e7d8fd0c-4ec38d09@192.168.11.6

CSeq: 101 INVITE

Max-Forwards: 70

Contact:

Expires: 240

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 442

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

Supported: x-sipura, replaces

Content-Type: application/sdp

v=0

o=- 5158 5158 IN IP4 myPublicIP

s=-

c=IN IP4 myPublicIP

t=0 0

m=audio 16440 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=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

[1]->77.72.169.134:5060(1085)

INVITE sip:0800.......@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5060;branch=z9hG4bK-ffc8175c;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=ff577530e0b8d335o1

To: <0800.......>

Remote-Party-ID: <>myVoipdiscountID@sip.voipdiscount.com>;screen=yes;party=calling

Call-ID: e7d8fd0c-4ec38d09@192.168.11.6

CSeq: 101 INVITE

Max-Forwards: 70

Contact:

Expires: 240

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 442

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

Supported: x-sipura, replaces

Content-Type: application/sdp

v=0

o=- 5158 5158 IN IP4 myPublicIP

s=-

c=IN IP4 myPublicIP

t=0 0

m=audio 16440 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=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

[1]->77.72.169.134:5060(1085)

INVITE sip:0800.......@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5060;branch=z9hG4bK-ffc8175c;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=ff577530e0b8d335o1

To: <0800.......>

Remote-Party-ID: <>myVoipdiscountID@sip.voipdiscount.com>;screen=yes;party=calling

Call-ID: e7d8fd0c-4ec38d09@192.168.11.6

CSeq: 101 INVITE

Max-Forwards: 70

Contact:

Expires: 240

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 442

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

Supported: x-sipura, replaces

Content-Type: application/sdp

v=0

o=- 5158 5158 IN IP4 myPublicIP

s=-

c=IN IP4 myPublicIP

t=0 0

m=audio 16440 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=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

[1]->77.72.169.134:5060(1085)

INVITE sip:0800.......@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5060;branch=z9hG4bK-ffc8175c;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=ff577530e0b8d335o1

To: <0800.......>

Remote-Party-ID: <>myVoipdiscountID@sip.voipdiscount.com>;screen=yes;party=calling

Call-ID: e7d8fd0c-4ec38d09@192.168.11.6

CSeq: 101 INVITE

Max-Forwards: 70

Contact:

Expires: 240

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 442

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

Supported: x-sipura, replaces

Content-Type: application/sdp

v=0

o=- 5158 5158 IN IP4 myPublicIP

s=-

c=IN IP4 myPublicIP

t=0 0

m=audio 16440 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=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

[0]->77.72.169.134:5060(414)

NOTIFY sip:sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5061;branch=z9hG4bK-92aebc48;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=f8f6fbfddd97a09eo0

To:

Call-ID: f27336b9-64bc4512@192.168.11.6

CSeq: 4 NOTIFY

Max-Forwards: 70

Contact:

Event: keep-alive

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[0]<<77.72.169.134:5060(464)

SIP/2.0 200 Ok

Via: SIP/2.0/UDP myPublicIP:5061;branch=z9hG4bK-92aebc48;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=f8f6fbfddd97a09eo0

To:

Contact: sip:77.72.169.134:5060

Call-ID: f27336b9-64bc4512@192.168.11.6

CSeq: 4 NOTIFY

Supported:

User-Agent: (Very nice Sip Registrar/Proxy Server)

Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Accept: application/sdp

Accept-Encoding:

Accept-Language:

FXO:PSTN Disconnect Tone

AUD:Stop PSTN Tone

FXO:On Hook

AUD:Stop PSTN Tone

[0]FM Alert Stop RxTx (c=0025422c;a=0)

[1:0]AUD Rel Call

[1]->77.72.169.134:5060(374)

CANCEL sip:0800.......@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5060;branch=z9hG4bK-ffc8175c;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=ff577530e0b8d335o1

To: <0800.......>

Call-ID: e7d8fd0c-4ec38d09@192.168.11.6

CSeq: 101 CANCEL

Max-Forwards: 70

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[1]->77.72.169.134:5060(374)

CANCEL sip:0800.......@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5060;branch=z9hG4bK-ffc8175c;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=ff577530e0b8d335o1

To: <0800.......>

Call-ID: e7d8fd0c-4ec38d09@192.168.11.6

CSeq: 101 CANCEL

Max-Forwards: 70

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[1]->77.72.169.134:5060(1085)

INVITE sip:0800.......@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5060;branch=z9hG4bK-ffc8175c;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=ff577530e0b8d335o1

To: <0800.......>

Remote-Party-ID: <>myVoipdiscountID@sip.voipdiscount.com>;screen=yes;party=calling

Call-ID: e7d8fd0c-4ec38d09@192.168.11.6

CSeq: 101 INVITE

Max-Forwards: 70

Contact:

Expires: 240

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 442

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

Supported: x-sipura, replaces

Content-Type: application/sdp

v=0

o=- 5158 5158 IN IP4 myPublicIP

s=-

c=IN IP4 myPublicIP

t=0 0

m=audio 16440 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=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

[1]->77.72.169.134:5060(374)

CANCEL sip:0800.......@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5060;branch=z9hG4bK-ffc8175c;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=ff577530e0b8d335o1

To: <0800.......>

Call-ID: e7d8fd0c-4ec38d09@192.168.11.6

CSeq: 101 CANCEL

Max-Forwards: 70

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[1]->77.72.169.134:5060(374)

CANCEL sip:0800.......@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5060;branch=z9hG4bK-ffc8175c;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=ff577530e0b8d335o1

To: <0800.......>

Call-ID: e7d8fd0c-4ec38d09@192.168.11.6

CSeq: 101 CANCEL

Max-Forwards: 70

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[1]->77.72.169.134:5060(374)

CANCEL sip:0800.......@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:5060;branch=z9hG4bK-ffc8175c;rport

From: <>myVoipdiscountID@sip.voipdiscount.com>;tag=ff577530e0b8d335o1

To: <0800.......>

Call-ID: e7d8fd0c-4ec38d09@192.168.11.6

CSeq: 101 CANCEL

Max-Forwards: 70

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

Does the PSTN line close the connection too early?

AUD:Stop PSTN Tone

Howard Wittenberg · ‎07-01-2013

I have studied the traces and I have no idea why you receive no response on the call. Assuming the INVITE is sent (the trace says it is) there must be a response coming back and somehow it is getting discarded.

An actual packet sniffer like WireShark would more positively verify the packet sending and receiving, however I don't believe you are setup to provide that.

"I tried to enter the dialplan on Line 1 (default is:(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) ).

Making a call works but the call won't be send through gw1."

I'm not sure what you mean here. You made the call with the dial plan sends the call thru gw1? YES/NO?? How do you know "the call won't be send thru gw1". I'm not sure the trace would look any different, or at least I don't know how it would look different.

If the call thru gw1 did work it would validate that you have the Gateway 1 parameters setup correctly.

Does the PSTN line close the connection too early?

With the trace of the PSTN Line call thru Gw1, the adapter sent the sip INVITE 5 times before it received the PSTN Disconnect Tone. It didn't receive any response to any of the 5 sip INVITES. The disconnect tone must be the result of some sort of PBX ringing timeout to no answer. On incoming FXO port calls the SPA3102 doesn't take the line off hook before the distant bridged call answers, at least that is the default setting for a PSTN-to-VoIP Gateway setting so the PBX thinks it is just ringing and it finally stops the ringing.

Bridging an inbound FXO port call out thru the gateway 1 does work on my SPA3102, and I have no idea of any settings that might cause my SPA3102 to respond like yours.

onlineuser · ‎07-01-2013

I'm not sure what you mean here. You made the call with the dial plan sends the call thru gw1? YES/NO?? How do you know "the call won't be send thru gw1". I'm not sure the trace would look any different, or at least I don't know how it would look different.

If the call thru gw1 did work it would validate that you have the Gateway 1 parameters setup correctly.

YES. Firstly I thought that I can see it in log like I see it on Info-tab but this won't be logged.

The gw1 settings are correctly - maybe the format is wrong - but the voipdiscount sip-uri I found on google, too. And the syslog is has also the same format mentioned to the generated sip-uri.

I have two more ideas.

- I will try the SPA-3102 with same settings on a simple ADSL modem

- If this works I will try Wireshark, if not there must be a problem with the settings.

But when calling via Line 1 works then it normally can't be a firewall problem when I try to make a call via gw1 instead of Line 1.

Ah I have a idea how I can proof if the call via dialplan 1 goes through gw1. I will set the gw1 settings NAT for gw1: no

Then the packets must have the internal IP instead of the right external IP.

Ok, I tried a call via PSTN with gw1 and then the INVITE packet had the internal IP: c=IN IP4 192.168.11.6

If your SPA with firmware 5.2.13 can this an my SPA not then there must be any settings wrong!?

The INVITE packets from gw 1 and Line 1 are ident but ony Line 1 requests I get an answer and on gw request there doesn't come any answer.

My PSTN account uses port 5060. For calls via gw1 also port 5060 will be used. The keepalive for sipcall account tells the server that the SPA is reachable via port 7739. How can it be that the voipdiscount server also must answer to the same port? In the INVITE request the SPA tells the server that he is also reachable via port 7739. Can this work in reality?

[1]->77.72.169.131:5060(1093)

INVITE sip:0800......@sip.voipdiscount.com SIP/2.0

Via: SIP/2.0/UDP myPublicIP:7739;branch=z9hG4bK-7886a120;rport

From: <>voipdiscountID@sip.voipdiscount.com>;tag=42cd383b4910ac4eo1

To: <0800......>

Remote-Party-ID: <>voipdiscountID@sip.voipdiscount.com>;screen=yes;party=calling

Call-ID: 999f947f-34965cda@192.168.11.6

CSeq: 101 INVITE

Max-Forwards: 70

Contact:

Expires: 240

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 448

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

Supported: x-sipura, replaces

Content-Type: application/sdp

v=0

o=- 2837710 2837710 IN IP4 myPublicIP

s=-

c=IN IP4 myPublicIP

t=0 0

m=audio 16462 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=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

[1]->212.117.203.32:5060(422)

NOTIFY sip:free3.voipgateway.org SIP/2.0

Via: SIP/2.0/UDP myPublicIP:7739;branch=z9hG4bK-a16cf899;rport

From: <49......>;tag=645d286678e8b757o1

To:

Call-ID: 18ac8d1a-d4e77a33@192.168.11.6

CSeq: 1889 NOTIFY

Max-Forwards: 70

Contact: <49.......>

Event: keep-alive

User-Agent: Linksys/SPA3102-5.2.13(GW002)

Content-Length: 0

[1]<<212.117.203.32:5060(307)

SIP/2.0 480 Temporarily Unavailable

Via: SIP/2.0/UDP myPublicIP:7739;branch=z9hG4bK-a16cf899;rport=7739

To: ;tag=92bd0327

From: <49......>;tag=645d286678e8b757o1

Call-ID: 18ac8d1a-d4e77a33@192.168.11.6

CSeq: 1889 NOTIFY

Content-Length: 0

onlineuser · ‎07-03-2013

I tried the SPA behind a simple ADSL modem without any other security devices. Same problem.

Do you think we can exchange out SPA http://SPA/admin/config.xml files - maybe there is a parameter which makes troubles!?

I will try Wireshark but I need a hub instead my switch first.

Do you have any other idea?

Howard Wittenberg · ‎07-05-2013

Sorry to take so long to respond. I was away from home without my computer.

The best thing to try at this point would be the factory reset and re-entry of the parameters, the reset that you previously mentioned. You already updated the firmware to the latest version. You said you tried a different adsl modem.

onlineuser · ‎07-07-2013

Strange thing!

Now I made a factory reset 73738#....

Now it works - the only difference is that I let the Line 1 SIP port on 5060 and the PSTN SIP port on 5061 (default). Before reset my PSTN port had 5060 and the Line 1 5061.

I will test if this was the problem.

No, this was not making the troubles - it also works fine!

Now I will compare my old configuration-exportfile with the new one.

Maybe the device just needed a factory reset after firmware upgrade!?

I would have one more new SPA-3102 - when the first one will be defect I can test it with second one if it has the same behavior. ;-)

Thanks for your help!

I would have two more questions.

1) Now, Line 1 is not set to any VoIP account - So when Line 1 is connected with phone it makes a fallback to gw0 which goes to the analogue system. How can I disable this fallback to gw0 (The line to which the SPA is connected is not allowd to use the landline via 0 but I want to disable all the fallback features to gw 0)?

VoIP Fallback To PSTN

Auto PSTN Fallback: no

This disables the fallback but when a sip account would be set on Line 1 it would work, wouldn't?

2) How must I set the dialplan if I want to use voipbuster on gw2?

(<#9,:>xx.<:@gw1>|xx.)

3) How can I make a free sip uri call? How must I configure the speed dial?

Where can I find out the sip uri from any other SIP-phone-number?

The people from who I know the data are no problem but there are also a lot of SIP-numbers from which I don't know it's provider.

The differences before factoryreset and now:

before: Off_Hook_While_Calling_VoIP_2_>Yes

now: Off_Hook_While_Calling_VoIP_2_>No

I can't remember if I have changed this config but I think it should not be responsible for my problems before factory reset!?

Do you know where I can find this setting in SPA-3102? Because of "_2" it seems that it must be findable in PSTN settings!?

Ah, it must be "Off Hook While Calling VoIP" in PSTN tab. But I can't find it in manual what this flag does!

before: Yes

now: No

Outgoing calls also work if this flag is not set. Now I set it to Yes and it also works fine for voipdiscount and sipgate calls.

Howard Wittenberg · ‎07-07-2013

Franz Toeffert

It is good to know that the factory reset solved your problem

I want to disable all the fallback features to gw 0)?

VoIP Fallback To PSTN

Auto PSTN Fallback: no

This disables the fallback but when a sip account would be set on Line 1 it would work, wouldn't?

That is the setting to enable or disable the auto fallback to gw0 if Line 1 is not registered or the link is down. If you set it to NO then I believe the only fallback would occur if the SPA3102 lost power. There is also the setting Make Call Without Reg: Yes and would call if Line 1 is not registered and if you have that set you would set the auto fallback to NO.

How must I set the dialplan if I want to use voipbuster on gw2?

(<#9,:>xx.<:@gw1>|xx.)

Change gw1 to gw2

How can I make a free sip uri call? How must I configure the speed dial?

Where can I find out the sip uri from any other SIP-phone-number?

The people from who I know the data are no problem but there are also a lot of SIP-numbers from which I don't know it's provider.

First you need to know if the voip provider accepts incoming sip uri calls. Some providers do accept them, but not all voip providers accept them. You also need to know the format. Usually the format is userid@proxy or userid@proxy:port (if the port is not 5060). Sometimes it is userid@special_proxy like CallCentric is userid@in.callcentric.com.

The SPA3102 has 8 speed dials. You just put the sip uri in the speed dial and to dial the call you dial 3# (if you want to dial speed dial number 3.

Many/most voip providers will allow you to call the PSTN DID number of another user of the same voip provider without charge.

There is a service, SipBroker, that can be very useful in making free calls to voip numbers. You need to know the voip provider though. You need to setup a special dial plan element to use SipBroker effectively.

Off Hook While Calling VoIP/ Make Call Without Reg

Off Hook While Calling VoIP is a setting under the PSTN-to-VoIP gateway. That setting causes the SPA3102 to answer the incoming PSTN Line call before placing the bridged outgoing call, instead of making the bridged call and waiting to connect to the PSTN Line call when the bridged call answers.

Make Call Without Reg: Yes/No is also a setting on the PSTN Line Tab to not allow outgoing calls if the configuration on the PSTN Line Tab is not registered.

I don't believe these settings were causing the problem you encountered. In your case I believe some undocumented setting was messed up and the factory reset fixed the problem.

onlineuser · ‎07-08-2013

I don't believe these settings were causing the problem you encountered. In your case I believe some undocumented setting was messed up and the factory reset fixed the problem.

I also think so.

I would need a dialplan that the user can choose between more betamax accounts.

Per default the PSTN account should be used, with 9# the SPA uses gw1 and with 8# the gw2 will be used.

Is this possible?

Howard Wittenberg · ‎07-08-2013

Per default the PSTN account should be used, with 9# the SPA uses gw1 and with 8# the gw2 will be used.

Is this possible?

Sure. Just add an additional element to the dial plan in question |<#8,:>xx.<:@gw2>|

Note that the dial plans were using #9 and #8, not the 9# and 8# that you mentioned. The speed dials use 2# thru 8# so you don't want a conflict with the speed dials. As you know the dial plan on the Line 1 tab is for the phone attached to the SPA3102, the dial plan on the PSTN Line Tab is for bridging an incoming PSTN Line or Voip Line call with an outgoing call.

onlineuser · ‎07-08-2013

Ok, here are my dialplans.

DP1: (<#11,:>xx.|xx.<:@gw1>)

DP2: (<#12,:>xx.|xx.<:@gw2>)

DP8: (S0<:1>)

DP8 works, DP1 also works but when I dial #12 I don't get a dialtone from SPA.

On gw2 I want to use my voipbuster-account. The settings on Line 1 tab for gw2 are correct.

Howard Wittenberg · ‎07-08-2013

I am not entirely sure what you are trying to accomplish. In all cases there will be a single dial plan that will be used for dialing a call, whether the call is from the phone attached to the SPA3102 (Line 1 Tab), an incoming PSTN Line Tab call to be bridged to an outgoing voip call (PSTN Caller Default DP:), or an incoming voip call to the PSTN Line Tab to be bridged to an outgoing PSTN (FXO) port call (VoIP Caller Default DP:).

In the case of the incoming PSTN (FXO) port call this will be the dial plan number referenced by

PSTN Caller Default DP:

The comma in the dial plan (for example after #11,) gives a second dial tone.

when I dial #12 I don't get a dialtone from SPA

You don't get a second dial tone because with the first dial plan

(<#11,:>xx.|xx.<:@gw1>)

you either dial #11 or everything else goes to gw1.

If you wish to sometimes dial #11, other times dial #12, and other times do not preface the call with #11 or #12 you will need to put it all in one dial plan ... (<#11,>xx.<:@gw1>|<#12,>xx.<:@gw2>|xx.}

You can't have #11 and #12 both doing the same thing.