[samarthk]

Folks,

I have a SPA3102 that has been configured with sipnumber.com and stanaphone.com. I have gotten US-based DID numbers from these providers.

My aim is to configure the SPA to accept VoIP calls on Line 1. This goal has been achieved.

The second goal was to have the SPA be configured in such a manner so as to allow the phone connected to it to use the PSTN for making the calls. For this, I put (xx.<:@gw0>) in the dial plan for Line 1. However what this does is dials the number and then within a couple of seconds gives me the dialtone for the PSTN line instead of dialing the number through the PSTN line. I thought that the way I crafted the dial plan the dialed numbers would automatically go through the PSTN line.

I also tried (xx.S0<:@gw0>) , (<#9,:><:@gw0) in the dial plan but to no avail. All of the above demonstrate the same behavior.

I want to do this so that the users who use the phone connected to the FXS port on the SPA3102 do not have to dial some additional characters to get the PSTN dialtone.

The third and final goal is to have the SPA accept VoIP calls and then either allow the phone connected to the FXS port to ring or present the VoIP caller with a dial-tone for the PSTN line. The dial-tone would only be presented after the user was authenticated.

I am trying to learn by taking baby steps however the second problem has me baffled.

I am hoping that somebody can point me in the right direction as to what I am missing to get all this done.

Thanks,
Samarth

[hwittenb]

Originally Posted by samarthk

The second goal was to have the SPA be configured in such a manner so as to allow the phone connected to it to use the PSTN for making the calls. For this, I put (xx.<:@gw0>) in the dial plan for Line 1.

The dial plan (xx.<:@gw0>) should dial all calls out on the attached pstn line. My guess is that your problem is that on the PSTN tab you have "one-stage-dialing" set to NO in the Voip-to-PSTN Gateway.

On the PSTN tab you should have
Line Enable YES
Preferred Codec G711u (or at least allow it)
Voip-to-PSTN Gateway Enable YES
Voip Caller Default DP 1
Dial Plan 1 (xx.)
One Stage Dialing YES

The SPA processes the dialed numbers then takes the PSTN line off hook and then dials the number entered.

[rizsher]

You may also find you need to put some sort of SIP credentials (even dummy ones) in the PSTN Tab to get the PSTN Gateway to work.

[samarthk]

Originally Posted by hwittenb

The dial plan (xx.<:@gw0>) should dial all calls out on the attached pstn line. My guess is that your problem is that on the PSTN tab you have "one-stage-dialing" set to NO in the Voip-to-PSTN Gateway.

On the PSTN tab you should have
Line Enable YES
Preferred Codec G711u (or at least allow it)
Voip-to-PSTN Gateway Enable YES
Voip Caller Default DP 1
Dial Plan 1 (xx.)
One Stage Dialing YES

The SPA processes the dialed numbers then takes the PSTN line off hook and then dials the number entered.

I tried with the changes that you mentioned but to no avail. The behavior is still the same, the numbers get dialed immediately and then a couple of rings and then the PSTN line goes Off Hook.

My settings presently on the 3102 in the Line 1 tab are as given below:

Audio Configuration
Preferred Codec G729a
Use Pref Codec Only: No
DTMF Process INFO: Yes
DTMF Process AVT: Yes
DTMF Tx Method: Auto

VoIP Fallback To PSTN:
Auto PSTN Fallback: No

Dial Plan
Dial Plan: (xx.<:@gw0>)

Enable IP Dialing: No


PSTN Tab Settings:

Line Enable: Yes
Preferred Codec: G711u
DTMF Process INFO: Yes
DTMF Process AVT: Yes
DTMF Tx Mode: Strict

Dial Plan 1: (xx.)

VoIP To PSTN Gateway Setup
Enable: Yes
Caller Auth Method: None
One Stage Dialing: Yes
Line 1 VoIP Caller DP: None
VoIP Caller DP: 1
Line 1 Fallback DP: None
VoIP Caller ID Pattern: xx.

[hwittenb]

Originally Posted by samarthk

The behavior is still the same, the numbers get dialed immediately and then a couple of rings and then the PSTN line goes Off Hook.

I looked at your settings and they seem OK. Depending on the setting of VOIP Answer Delay you will hear "rings" before the voip-to-pstn gateway "answers" the internal voip call and the adapter dials the numbers on the pstn. If you set VOIP Answer Delay to 0 you probably won't hear any "rings".

If your configuration still doesn't work, with your web browser save the configuration to your hard disk. Using WinXP use the send to option to create a zip file of the configuration and then attach the configuration to a posting. Passwords are not saved.

[samarthk]

Originally Posted by hwittenb

I looked at your settings and they seem OK. Depending on the setting of VOIP Answer Delay you will hear "rings" before the voip-to-pstn gateway "answers" the internal voip call and the adapter dials the numbers on the pstn. If you set VOIP Answer Delay to 0 you probably won't hear any "rings".

If your configuration still doesn't work, with your web browser save the configuration to your hard disk. Using WinXP use the send to option to create a zip file of the configuration and then attach the configuration to a posting. Passwords are not saved.

hwittenb,

Here is my setup saved as html and attached to this message. I hope you can help me in solving this puzzle.

Thanks,
Samarth

[hwittenb]

The Line 1 dial plan in your attached configuration is:
(<xx.<:@gw0>)
Change it to the following:
(xx.<:@gw0>)

I ran a test with your dial plan in your configuration on my SPA3000 and the results were consistent with the symptoms that you described.

[samarthk]

Originally Posted by hwittenb

The Line 1 dial plan in your attached configuration is:
(<xx.<:@gw0>)
Change it to the following:
(xx.<:@gw0>)

I ran a test with your dial plan in your configuration on my SPA3000 and the results were consistent with the symptoms that you described.

I did change the dial plan to what you mentioned and the numbers do get dialed. I have a feeling that some additional information is being passed through because now I get a message from the PSTN service that the call could not be completed. If I try to call up using a different phone that is not connected to the SIPURA then the call goes through.

I'll probably have to turn up the logging and then see what is being passed through to the PSTN line.


Here is the log for a call to the number 8008000800

syslog server(port:514) started on Thu Jan 11 08:22:15 2007
CC:Clean Up
--- OBJ POOL STAT ---
OP:RTPRXB = 96 ( 96 192) OP:RTPREB = 40 ( 40 48)
OP:RTPTXB = 64 ( 64 108) OP:TIMEOU = 107 (120 52)
OP:SIPCOR = 0 ( 1 28) OP:SIPCTS = 31 ( 32 1136)
OP:SIPSTS = 32 ( 32 4056) OP:SIPAUS = 7 ( 8 588)
OP:SIPDLG = 10 ( 10 148) OP:SIPSES = 12 ( 12 9352)
OP:SIPREG = 2 ( 4 296) OP:SIPLIN = 0 ( 2 140)
OP:SUBDLG = 2 ( 2 6436) OP:STUNTS = 16 ( 16 68)
OP:XMNODE = 1024 (1024 112)
[0]Off Hook
2. Report digit 8 (1)(40 ms)
2. Report digit 0 (1)(40 ms)
2. Report digit 0 (1)(40 ms)
2. Report digit 8 (1)(40 ms)
2. Report digit 0 (1)(40 ms)
2. Report digit 0 (1)(40 ms)
2. Report digit 0 (1)(40 ms)
2. Report digit 8 (1)(40 ms)
2. Report digit 0 (1)(40 ms)
2. Report digit 0 (1)(40 ms)
Calling:8008000800@127.0.0.1:5061
[0:0]AUD ALLOC CALL (port=16454)
[0:0]RTP Rx Up
[1:0]AUD ALLOC CALL (port=16456)
[1:0]RTP Rx Up
CC:Ringback
[0:0]RTP Rx Dn
AUD:Stop PSTN Tone
[0:0]ENC INIT 0
[0:0]RTP Tx Up (pt=0->c0a80164:16456)
[0:0]RTCP Tx Up
CC:Remote Resume
CC:Connected
[0:0]RTP Rx Up
CC:Connected
AUD:Stop PSTN Tone
[1:0]ENC INIT 0
[1:0]RTP Tx Up (pt=0->c0a80164:16454)
[1:0]RTCP Tx Up
FXO:Off Hook
FXO:Stop CNDD
[0:0]RTP Rx 1st PKT @16454(3)
[1:0]RTP Rx 1st PKT @16456(3)
[0:0]DEC INIT 0
[1:0]DEC INIT 0
[1:0]RTP Dst Change:7f000001:16454
[0:0]RTP Dst Change:7f000001:16456
[0]On Hook
[0]FM Alert Stop RxTx (c=0024900c;a=0)
[0:0]AUD Rel Call
CC:Ended
AUD:Stop PSTN Tone
[0]FM Alert Stop RxTx (c=0024eaa4;a=0)
[1:0]AUD Rel Call
AUD:Stop PSTN Tone
FXO:On Hook
AUD:Stop PSTN Tone
FXO:Stop CNDD
AUD:Stop PSTN Tone
DLG Terminated 286080
Sess Terminated
DLG Terminated 285fec
Sess Terminated
[5060]STUN trying 0
[5060]STUN trying 1
[0]REG: STUN c0a80164->97c99fad, 5060->5060
[0]RegOK. NextReg in 53 (1)

[hwittenb]

As you say, it looks like you are dialing the number 8008000800 and it is being sent to the FXO port. You could see more detail in the log if you set Sip Debug Option Full in both the Line 1 and PSTN tabs. I'm not sure if more detail would pinpoint your problem.

What country are you located? Maybe your problem is something to do with your pstn interface.

[samarthk]

Originally Posted by hwittenb

As you say, it looks like you are dialing the number 8008000800 and it is being sent to the FXO port. You could see more detail in the log if you set Sip Debug Option Full in both the Line 1 and PSTN tabs. I'm not sure if more detail would pinpoint your problem.

What country are you located? Maybe your problem is something to do with your pstn interface.

I am located in the US in Pennsylvania. The debug option is set to the maximum level, that I believe is 3. I think there is only one place to set the debug level in and that is in the System tab. Whether this setting captures both the VoIP and PSTN interfaces is something that I do not know.