[triwhdxk]

Yes, saw that the syslog is missing some piece of information so used a different syslog server. attached is the zip file

PSTN to VOIP
called the PSTN and than tried to dial gizmo test number 17474743246
can't see the numbers going through. it gave me this message on info tab Last PSTN Disconnect Reason: VoIP Dialing Timeout

VOIP to PSTN
called the VOIP number from SIPBroker access number and than on PSTN tried to dial 07944550927 my mobile number. I think some digits are getting missed.
it gave me this message on info tab
Last PSTN Disconnect Reason: PSTN Disconnect Tone

[hwittenb]

Originally Posted by triwhdxk

PSTN to VOIP
called the PSTN and than tried to dial gizmo test number 17474743246
can't see the numbers going through. it gave me this message on info tab Last PSTN Disconnect Reason: VoIP Dialing Timeout

After the ring thru completes, the pstn-to-voip gateway takes the FXO port off hook and gives the dial tone (AUD:Play PSTN Tone 1). Ten seconds later the adapter gives the fast busy tone (AUD:Play PSTN Tone 8), the adapter does not show any digits dialed during this period which is consistent with the VoIP Dialing Timeout.

If you are calling your PSTN number via a voip provider the assumption would be they aren't passing thru the dtmf digits.

VOIP to PSTN
called the VOIP number from SIPBroker access number and than on PSTN tried to dial 07944550927 my mobile number. I think some digits are getting missed.
it gave me this message on info tab
Last PSTN Disconnect Reason: PSTN Disconnect Tone

The voip-to-pstn trace shows the entered digits 09445XX0927 where XX is TP Parser error 34. This does not match any element of the dial plan you posted in your last configuration posted on 8/23 which was (07[789]x.|08[4-7]x.|020[2-9]x.|01[2-9]x.).

The configuration says you have an interdigit long timer setting of 10 seconds. In other words the adapter will consider the dialing completed after 10-seconds. In your case you didn't have enough digits entered to match a dial plan so there was a disconnect after 10-seconds.

If you entered more digits than this and they don't show up in the adapter trace the assumption is you are having a problem with GizmoProject either receiving the digits initially or in transmitting the digits thru their proxy to your adapter.

For the voip-to-pstn gateway there is a way to make a call from a softphone, another ata, or an asterisk system using direct ip dialing, bypassing your voip provider, that sends the number in a sip invite instead of by dtmf digits. This technique is a more complicated configuration that uses http digest authentication.

[triwhdxk]

for PSTN-to-VOIP I called my PSTN home number from my office desk (which works through pbx). the syslog file was using this config.

Now I have tried calling my PSTN home number from mobile but sometimes it works and sometimes it doesnt. most of the time it doesnt works.

IS there any settings that I can change.


For VOIP-to-PSTN should I change the interdigit long timer setting and see. VOIP to PSTN works fine using a softphone. I have tried this and everytime I am successful.

i am new to all this so not sure whats are the next steps. i am willing to go and reload the firmware and use the default settings if that works out of the box.

regards,
kashif

[hwittenb]

Originally Posted by triwhdxk

for PSTN-to-VOIP I called my PSTN home number from my office desk (which works through pbx). the syslog file was using this config. IS there any settings that I can change.

On the PSTN tab under International Control there are some gain settings you could try adjusting up or down. I believe the default is 0, and you are currently running with some positive gain. Try first setting it to zero and see if the results are any different. There is also an impedance setting that should be correct, however I believe you have the correct setting for the U.K. British Telephone.

For VOIP-to-PSTN should I change the interdigit long timer setting and see. VOIP to PSTN works fine using a softphone. I have tried this and everytime I am successful.

Doubling the value of the interdigit long timer will give you more time to enter the digits, but if they aren't transmitted thru the network it doesn't help. Was the softphone with the same provider? The softphone probably sends the dtmf via rfc2833 which is called AVT on the Sipura adapters. I would see if you can determine the softphone settings to send dtmf and match that with the 3102.
[/quote]

[triwhdxk]

for the softphone I used Xlite

[hwittenb]

Originally Posted by triwhdxk

for the softphone I used Xlite

I ran some tests on my SPA with GizmoProject. For me, it appears to pass dtmf thru their proxy usiing AVT and either G711(u or a), or G729.

Their softphone prefers to use the iLBC codec which isn't supported by the SPA. It passes dtmf using AVT (rfc2833).