SPA-3102 - Sipgate + VoIpDiscount - Page 2

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 · ‎06-27-2013

Ok, now it's clear.

But when I set dialpla 1 to: (<#9,:>xx.|xx.<:@gw1>) no call will be routed via voipdiscount when I press 9 before number.

Moreover, making calls via sipgate are also not possible when I enter this dialplan.

After dialing the line is quiet and I can't hear ringing.

If I remove the dialplan sipgate-calls work.

Why there are no outoing calls possible via gateway1?

(<#9,:>xx.<:@gw1>|xx.) tries to make call via sipgate and not via voipdiscount (error message from sipgate)

(<#9,:>xx.|xx.<:@gw1>) no ringing

dial plan 1: (<#9,:>xx.|xx.<:@gw1>)

dial plan 2: (S0<:3>)

update - configuration:

PSTN: sipgate account for incoming and outgoing calls

dial plan 1 on PSTN: (<#9,:>xx.|xx.<:@gw1>) make outgoing calls via gateway 1 (voipdiscount)

dial plan 2 on PSTN: (S0<:3>) forwards incoming calls from sipgate to analogue extension 3

Line 1: line disabled but

Gateway 1: userid@sip.voipdiscount.com

NAT: yes and no tried

GW1 Auth: userid

GW1 password: voipdiscount password

testing:

without prefix --> 0800......7@gw1 no ringing

with #9: --> 0800......8 sipgate

I can see the @gw1 at VoIP Peer Number while calling.

When I call a number without prefix the SPA tries to route via gateway1 (voipdiscount but I can'T hear ringing)

When no prefix is dialed the call should go via sipgate, shouldn't?

When I dial #9 dial plan will be executed but then the call goes via sipgate!

Should this not be reversed?

no prefix --> call via sipgate

with #9 --> call via voipdiscount

Ah: with (<#9,:>xx.<:@gw1>|xx.) it's reversed :-)

But making calls via gatway 1 is not possible. SPA seems to make it correct: by pressing #9 he wants to dial number@gw1 - so he should use the voipdiscount settings but the line is quiet and no ringig - also no data will be transfered. But it's not a firewall problem!

Under User Line 1 setting I found the option: Cfwd All Dest

Why PSTN-User has no option to forward all calls to any extension? Since last changes the forwarding to an incoming PSTN call via sipgate will also not be forwarded to extension 3 via dialplan 3. :-(

Now I also don't receive incoming calls via sipgate on PSTN Line. Outgoing calls works but when I try to call my SIP with mobile it seems that the number is not available. :-(

Howard Wittenberg · ‎06-27-2013

Ah: with (<#9,:>xx.<:@gw1>|xx.) it's reversed :-)

But making calls via gatway 1 is not possible. SPA seems to make it correct: by pressing #9 he wants to dial number@gw1 - so he should use the voipdiscount settings but the line is quiet and no ringig - also no data will be transfered. But it's not a firewall problem!

Your posted attachment showing some of the INFO Tab settings of the SPA3102 during a call show the Last Called VoIP Number as 9. It should show the complete number you are dialing. The VoIP Peer Number perhaps does show the number. This would suggest to me that you may have a mistake in the dial plan. The 9 is supposed to be stripped from the dialing digits. I would try typing the dial plan in again.

Voipdiscount is a Betamax company. It was always my understanding that that group of companies always required dialed numbers to start with 00. It doesn't look like you are doing that. You are dialing 0800xxxxxx

The additional troubleshooting tool available to you is a sip debug trace. The trace would show the sip signalling sent and received by the SPA3102. To run a sip debug trace you download and install a Syslog program on your local pc. You put your pc's local ip address on the SPA3102 System Tab under Debug Server, you set the Debug Level to 3 on the System Tab and on the Line 1 and on the PSTN Line Tab you set the Sip Debug Option to FULL. You can download a simple DOS pc program from Cisco here:

https://supportforums.cisco.com/docs/DOC-9862

The format of your posted INFO Tab is different from my format suggesting that you are not running the latest firmware on the SPA3102 which is v5.2.13. I would suggest you do run the latest firmware. You can download the firmware here:

http://software.cisco.com/download/release.html?mdfid=282414112&softwareid=282463187&release=5.2.13

This afternoon I ran an extensive series of tests trying to duplicate the trouble you are experiencing. I could not duplicate your problem. My calls all worked with the dial plan I suggested. The #9 calls were all sent via @gw1 and the calls without #9 were sent via the configuration I had setup on the PSTN Line Tab.

Under User Line 1 setting I found the option: Cfwd All Dest

Why PSTN-User has no option to forward all calls to any extension?

The PSTN-User tab has no option to forward all calls to any extension. Forwarding is done via the dial plan.

Now I also don't receive incoming calls via sipgate on PSTN Line. Outgoing calls works but when I try to call my SIP with mobile it seems that the number is not available. :-(

If the SPA3102 INFO Tab shows the PSTN Line Tab Configuration (SipGate) is REGISTERED then it should receive incoming calls if you have properly configured the voip-to-pstn gateway and the PSTN Line is available for calling. The only use of incoming calls to the PSTN Line Tab configuration is to bridge an outgoing call on the attached PSTN Line.

onlineuser · ‎06-28-2013

Ok, I upgraded from 3.3.6 to 5.2.13.

When I try to call my SIP via mobile, should I see at Info-website that a external call is ringing? I only see idle.

The SPA3102 must forward an incoming call to extension 3 - this means the SPA must dial a 3 that the analogue phone is ringing - is there the dialplan 2 with (S0<:3>) enough? Moreover, I should hear the relais when a VoIP call will be forwarded to extension 3, shouldn't.

I will try to log all syslog entries from SPA-3102.

I set the registration expire from 3600 to 600 seconds but when I try to call my SIP after warmboot the SIP-messages counter will be raised - When I try the same after 2 minutes the counter won't get raised. But the status registered is ok and outgoing calls also work fine.

New firmware can also export the configuration.

Is there a way to reset the voice part of router to defaults?

I only found this:

You can reset the SPA3102 to factory default using the Interactive Voice Response (IVR) and following these steps:

- Plug a phone into the Phone 1 port

- Press “****”

- Enter “73738,” then “#”

- Press “1” to confirm

But this resets all, also LAN settings.

When I hear the dialtone from SPA and I dial a number 0800... it will be routed via sipgate. 0800... represents a german free number.

When I hear the dialtone from SPA and I dial #9 I hear another dialtone and then the call should be routed via voipdiscount. the #9 will be cut and in VoIP Peer number field I see 0800...@gw1 but the line is quiet.

Here is my syslog when I try to make a call via voipdiscount ( also SPA dialtone --> #9 --> 0800......

syslog server(port:514) started on Fri Jun 28 21:21:00 2013

FXO:Start CNDD

AUD:Stop PSTN Tone

FXO:Off Hook

FXO:Stop CNDD

AUD:Play PSTN Tone 1

FXO:Digit=#

AUD:Stop PSTN Tone

FXO:Digit=9

[1]PlayOutDial 1

AUD:Play PSTN Tone 3

FXO:Digit=0

AUD:Stop PSTN Tone

FXO:Digit=8

AUD:Stop PSTN Tone

FXO:Digit=0

AUD:Stop PSTN Tone

FXO:Digit=0

.

FXO:Digit=1

AUD:Stop PSTN Tone

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

[1:0]AUD ALLOC CALL (port=16450)

[1:0]RTP Rx Up

[1]->77.72.169.131:5060(1118)

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

Via: SIP/2.0/UDP 192.168.11.6:5060;branch=z9hG4bK-bb2cfdb0;rport

From: 49sipgatenumber....... <>voipuserid@sip.voipdiscount.com>;tag=5bc47c7515a77068o1

To: <0800....>

Remote-Party-ID: 49sipgatenumber <>voipuserid@sip.voipdiscount.com>;screen=yes;party=calling

Call-ID: b69e7339-4064874@192.168.11.6

CSeq: 101 INVITE

Max-Forwards: 70

Contact: 49sipgatenumber

Expires: 240

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

Content-Length: 444

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

Supported: x-sipura, replaces

Content-Type: application/sdp

v=0

o=- 1054682 1054682 IN IP4 192.168.11.6

s=-

c=IN IP4 192.168.11.6

t=0 0

m=audio 16450 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

this block above will be repeated 10 or 20 times

.

RSE_DEBUG: getting alternate from domain:sip.voipdiscount.com

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

[1:0]AUD Rel Call

CC:Failed w/ Calling

AUD:Play PSTN Tone 8

Sess Terminated

AUD:Stop PSTN Tone

FXO:On Hook

AUD:Stop PSTN Tone

Why does the SPA send my sipgate number - why voipdiscount should know my SIP number from PSTN account?

The IP where the SIP request will be send is ok but I can't see the password for the voipdiscount account in the syslog data!?

Does it mean that the SPA doesn't send the password for voipdiscount!?

Howard Wittenberg · ‎06-28-2013

When I try to call my SIP via mobile, should I see at Info-website that a external call is ringing? I only see idle.

The SPA3102 must forward an incoming call to extension 3 - this means the SPA must dial a 3 that the analogue phone is ringing - is there the dialplan 2 with (S0<:3>) enough? Moreover, I should hear the relais when a VoIP call will be forwarded to extension 3, shouldn't.

The dialplan with (S0<:3>) would dial 3 as you say. I don't understand. Where do you want to dial the 3? To SipGate which you have configured on the PSTN Line Tab or you want to dial 3 on the PSTN Line? Or are you doing something with the phone attached to the SPA3102 which is using the configuration on the Line 1 Tab?

There is only one other type of reset called User Factory defaults which has to do with not erasing something setup on the unit by a voip provider (provisioning). The Factory Reset you mentioned is what you would want to use and you then have to reenter everything.

onlineuser · ‎06-28-2013

In meantime I tried it with sicall instead of sipgate - same problems - no incoming calls. :-(

It must be any configuration because when I connect my TA612V it works fine with sipgate und sipcall.

No, when a call comes in via sipgate the PSTN line must dial a 3 that the extension 3 rings for incoming sipgate call.

The SPA hangs on analogue system on extension 2. When extension 3 dials 2 I hear the dialtone from SPA (for outgoing calls).

When a sipgate call comes in it can't ring anywhere - so the SPA must forward the call to analogue system to extension 3.

No, on Line 1 there is no phone connected!

here is the log when I try to call my sipgate number via mobile - service unavailable

SIP/2.0 503 Service Unavailable

I removed all the other log because this was the relevant line.

onlineuser · ‎06-28-2013

Now I tried to enter the sipgate or sipcall account on Line 1 (no phone is connected) and there I can see in the info tab the incoming call ringing!

Also the PSTN line has a configuration problem.

But the VoIP to PSTN configuraion seems to be ok.

But the forwarding to extension 3 doesn't work proper.

The SPA is connected with extension 2 from analogue system. When I connect a phone instead of the SPA (to simulate) I must pick up and dial a 3 that extension 3 rings.

And so also must do it by the SPA. When a VoIP call comes in he must forward to extension 3.

Howard Wittenberg · ‎06-28-2013

The SPA3102 PSTN Line responds to the incoming call:

[1]->212.117.203.32:5060(521)

SIP/2.0 503 Service Unavailable

You would get this because either the Voip-to-PSTN Dial plan was not correct or because the SPA3102 believes the PSTN Line is currently being used. To check for PSTN Line in use the SPA3102 looks at the voltage presented on the FXO port. If the voltage is lower that the "Line-In-Use Voltage:" it determines that the line is in use, it will not take the line off hook, and it returns 503 Service Unavailable.

You can read the on-hook voltage of the analog line connected to the SPA3102 when the line is idle. The reading is on the SPA3102 INFO Tab.

The default setting for Line In Use is 30. If the on-hook voltage of the line attached to the SPA3102 is less than 30v you need to adjust the Line In Use Voltage Setting. Usually it is set about half way between the on-hook and the off-hook voltage reading.

onlineuser · ‎06-28-2013

The idle voltage is 24-25V.

When extension 3 makes a call via sipgate the line voltage only has 6-7 volts.

The Line-In-Use Voltage on PSTN tab is set to 30

Not bad, now I set it to 20 volts and incoming calls will be forwarded to extension 3 :-)

Is it also possible to change the ringing but now it rings in same kind like when extension 4 calls extension 3. And the analogoue system doesn't show the number in display because it does not support CLIP. But maybe the SPA can change the ringing-behaviour or is it not possible because the ringing is made by analogue system.

Now there is only the problem why outgoing calls via voipdiscount not work!?

Howard Wittenberg · ‎06-28-2013

But maybe the SPA can change the ringing-behaviour or is it not possible because the ringing is made by analogue system.

If I understand correctly, all the SPA3102 is doing is dialing a number. The SPA3102 doesn't have anything to do with ringing.

Now there is only the problem why outgoing calls via voipdiscount not work!?

Looking at the sip trace you posted of a call to voipdiscount, you show the sip INVITE and then say it is repeated a number of times. In other words there is no response from voipdiscount.

I did validate that you can send a 0800xxxx German toll free number to a Betamax account without 00+country code, so that is not the problem.

The Sip INVITE does not show your external ip address in the Contact line of the header or the c= line of the message body. It shows your local network ip address. I believe that is the problem. Voipdiscount is responding to the wrong address and you do not receive the response.

I would set NAT Mapping Enable: Yes on the PSTN Line Tab and the GW1 NAT Mapping Enable: Yes on the Line 1 Tab

If that by itself doesn't solve the problem, I would keep that setting and add on the Sip Tab:

Handle VIA received: YES

Handle VIA rport: YES

Insert VIA received: YES

Insert VIA rport: YES

Substitute VIA Addr: YES

Send Resp To Src Port: YES

STUN Enable: YES

STUN Server: stun.voipdiscount.com

The Sip INVITE you questioned does not show the authentication. That comes in the followup second sip INVITE after it receives the 401 Unauthorized Response.

onlineuser · ‎06-28-2013

Ok, I tried it with NAT and Line 1 is disabled.

Although there the Line 1 display-name was sip but Line 1 is disabled.

When I try to make a call via voipdiscount it sends the display from Line 1 - strange - why will be the display-name from Line 1 used for gateway callings?

AUD:Stop PSTN Tone

FXO:Digit=0

AUD:Stop PSTN Tone

FXO:Digit=0

AUD:Stop PSTN Tone

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

[1:0]AUD ALLOC CALL (port=16454)

[1:0]RTP Rx Up

[1]->77.72.169.134:5060(1109)

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

Via: SIP/2.0/UDPmypublicIP:1624;branch=z9hG4bK-bc80c6ae;rport

From: sipcall <>voipdiscountid@sip.voipdiscount.com>;tag=a94788244fc52feco1

To: <0800......>

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

Call-ID: 5b0c3684-aa71550c@192.168.11.6

CSeq: 101 INVITE

Max-Forwards: 70

Contact: sipcall

Expires: 240

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

Content-Length: 444

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

Supported: x-sipura, replaces

Content-Type: application/sdp

v=0

o=- 34625 34625 IN IP4 mypublicIP

s=-

c=IN IP4 mypublicIP

t=0 0

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

The caller ID is always the internal IP - but this did the TA612V in same way. Now the voipdiscount server knows to answer to my public IP. But it also not work. :-(

Howard Wittenberg · ‎06-29-2013

The caller ID is always the internal IP - but this did the TA612V in same way. Now the voipdiscount server knows to answer to my public IP. But it also not work. :-(

I would get something working on the SPA3102 with voipdiscount. If you get no response to the Sip INVITE then they are not sending a response or something is blocking the response.

Can you configure VoipDiscount on the Line 1 Tab to register, attach a phone to the SPA3102 and make a successful call? Of course you need to enable Line 1, configure Line 1 including NAT Mapping Enable, use the default dial plan or a simple dial plan like (xx.).

Edit: Of course to get the full trace you need to also set Sip Debug Option: FULL on the Line 1 Tab. This test will show that your voipdiscount account will basically work on your SPA3102.

If that works can you change the dial plan to use Gateway 1 and make a successful call? A dial plan to use Gateway 1 from the phone attached to the SPA3102 would be something like (xx.<:@gw1>)

A successful call sip trace sip signalling would look something like this:

--> Invite

<-- 401 Unauthorized

--> Invite (this invite has the authentication)

<-- 100 Trying

<-- 183 Session progress

Voice Stream Starts Up

<-- 200 OK

--> ACK

--> BYE

<-- 200 OK

onlineuser · ‎06-30-2013

Ok, I will try it.

One more problem.

When the SPA is rebooted incoming and outgoing calls works fine.

Outgoing calls always work but after one hour idle-status no incoming calls are possible.

Now I enabled NAT on PSTN and keep alive feature.

Now the SPA send the keep alive packet every half minute but theSIP answer is 480 - Temporarily Unavailable

Is this correct?

[1]->212.117.203.32:5060(435)

NOTIFY sip:free3.voipgateway.org SIP/2.0

Via: SIP/2.0/UDP myPublicIP:4761;branch=z9hG4bK-20c0958e;rport

From: sipcall <>49mySipNumber@free3.voipgateway.org>;tag=e8e4e38f54878df9o1

To:

Call-ID: 8541d697-273f691@192.168.11.6

CSeq: 57 NOTIFY

Max-Forwards: 70

Contact: sipcall <49MYSIPNUMBER>

Event: keep-alive

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

Content-Length: 0

[1]<<212.117.203.32:5060(313)

SIP/2.0 480 Temporarily Unavailable

Via: SIP/2.0/UDP myPublicIP:4761;branch=z9hG4bK-20c0958e;rport=4761

To: ;tag=64814e37

From: "sipcall"<>49mySipNumber@free3.voipgateway.org>;tag=e8e4e38f54878df9o1

Call-ID: 8541d697-273f691@192.168.11.6

CSeq: 57 NOTIFY

Content-Length: 0

Now it seems to work - I hope it also works in 2 hours without problems.

Moreover, when extension 1,2 or 4 is busy via landline and I make a incoming call via SIP I get a busy-tone but on mobile this call costs money. Why this?

Moreover, when I make a call from mobile to ma SIP-number the ringing also costs money? Why this?

Howard Wittenberg · ‎06-30-2013

The purpose of the Keep-Alive messages is to keep the sip port in your router's firewall open to incoming packets. Otherwise your router's firewall could discard them as unsolicited. The response you get from the distant server is fine.

If you get billed by your mobile it is probably because the SPA3102 had to answer the phone to bridge the call.

onlineuser · ‎06-30-2013

But the status 480 Temporarily Unavailable from server is strange - there someone could believe that the keep-alive was not successful (a more positive statement would be finer).

If you get billed by your mobile it is probably because the SPA3102 had to answer the phone to bridge the call.

It seems! But I thought the SPA only forwards the call.

I can't hear the relais inside so I think the SPA doesn't hook-off the call.

I tried to setup the sipgate account on Line 1. When I call the SIP number I hear any other ringing tone and it won't be billed. So the ringtone while forwarding from PSTN to analogue system will be generated by the SPA. So when anyone lets ringing for 2 minutes he must pay for 2 minutes. :-(

Is there a solution available for forwarding without that the SPA hook-of the incoming VoIP-call?

The next thing I hate that a call from SIP to SIP (2 different providers) are not for free. The providers cash money for publishing the SIP-address: sip:NUMBER@free3.voipgateway.org :-(

This should be changed worldwide. ;-)

I tried to set my PSTN VoIP Answer Delay: from 0 to 10. At incoming call via mobile I hear for 10 seconds ringing (free) and then sthe SPA forwards the call and extension 3 begings ringing - from this moment the ringtone changes and this costs money.

Why does the SPA send it's firmware histroy to server?

[0]RegOK. NextReg in 86 (1)

IDBG: st-0

fs:11919:11994:65536

fls:af:1:0:0

fbr:1:3000:3000:14f02:0004:0005:5.2.13(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)

So the server knows that the device was upgraded from 3.3.6 to 5.2.13. ;-)

Ok, back to my voipdiscount problem for outgoing calls.

I tried to set you suggested NAT options but same problem. Entering the STUN server from voipdiscount also not helped.

Now, I have one more question. When there is configured the gateway 1, how knows the SPA whick proxy should be used.

In syslog I see that he sends the request to voipdiscount server, he also sends the server my public IP.

sip:voipdiscountID@myPublicIP:6999

.

c=IN IP4 myPublicIP

So the dialplans can't be wrong when the SPA tries to send the call-request to the right server!?

I tried to configure the voipdiscount-account on Line 1 - there outgoing calls work fine!

Here I read something interessting!

http://forum.voxilla.com/threads/spa3102-voip-to-pstn-gateway-voip-answer-delay-affecting-outbound-inbound.15344/

If you have outgoing calls (PSTN to voip) (using the pstn-to-voip gateway) then you need the credentials on the PSTN tab, but your provider needs to allow outgoing calls without registration (Make Call Without Reg).

This could be my problem - the SPA also tries to call without registration at voipdiscount.

Why the SPA-3102 not support a dialplan for use Line 1 settings for outgoing calls - then I could use Line 1 for voipdiscount and PSTN for sipgate/sipcall. And the dialplan #9 uses Line 1 instead of gw1. Is there a way for this idea?

Is it possible to forward a Line 1 incoming call with Line 1 VoIP Caller DP in PSTN settings to extension 3?

Then the reverse way should also be configurable - outgoing call with dialplan 1 to Line 1 instead to gw1?

Why does SPA uses for gw 1 calling the same display-name from PSTN options?

From: sipcall <>voipdiscountID@sip.voipdiscount.com>;tag=74544cb0825649ddo1

To: <0800.....>

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

calling

Call-ID: 4399276c-2d4ace51@192.168.11.6

CSeq: 101 INVITE

Max-Forwards: 70

Contact: sipcall

Expires: 240

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

Content-Length: 443

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

Supported: x-sipura, replaces

Content-Type: application/sdp

The display-name will be taken from PSTN options - I also tried to enter here my sipcall SIP number - also not helped.

Howard Wittenberg · ‎06-30-2013

So the dialplans can't be wrong when the SPA tries to send the call-request to the right server!?

The dial plan either sends the call to the server configured on the PSTN Line Tab or it sends the call to the server configured on @gw1 depending on the dial plan. When @gw1 is used it also uses some settings from whatever is configured on the PSTN Line Tab or the Line 1 Tab depending on which dial plan is originating the call.

I tried to configure the voipdiscount-account on Line 1 - there outgoing calls work fine!

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.

Is there a solution available for forwarding without that the SPA hook-of the incoming VoIP-call?

I don't think so, they didn't design it that way. Anyway there is no way to tell that the analog side of the call is answered. There is no positive signal, no "answer supervision".

The next thing I hate that a call from SIP to SIP (2 different providers) are not for free.

You can make a (free) sip uri call to a voip account if the provider accepts incoming sip uri calls. I believe voipdiscount accepts incoming sip uri calls to a registered account. I have no idea whether or not sipgate accepts incoming sip uri calls. To make a sip uri call with the SPA3102 the best way to do it is with a speed dial.