[rvtango]

"asterisk" is not an extension it is a user name that Asterisk uses to authenticate itself to SPA. (using HTTP Auth method)

[larryalk]

Thanks for your reply rvtango.

When I start Asterisk CLI, I tee to a log file because the messages come so fast.

I have many entries like this:

Via: SIP/2.0/UDP 10.22.44.22:5060;branch=z9hG4bK19b07740;rport
From: "asterisk" <sip:asterisk@10.22.44.22>;tag=as00c1ca40
To: <sip:424@10.22.44.124:5060>


Are the "asterisk" and sip:asterisk@ the
"user name that Asterisk uses to authenticate itself to SPA. (using HTTP Auth method)" ?

Can I change this name in the PSTN Line tab to something more unique? Something like ast-auth?

Larryalk

[SME]

Originally Posted by larryalk

Can I change this name in the PSTN Line tab to something more unique? Something like ast-auth?

Larryalk

Yes you can, but if you do, you'll also have to change the right entry in sip.conf, at the very least.

Good luck...

[larryalk]

Thanks SME. I'll just leave the "asterisk" user alone.

The Wizard states that
"The PSTN Line will also be connected to Asterisk as a different extension for the purposes of forwarding inbound PSTN calls to Asterisk;"

Is the above reference to the "asterisk" user or extension?

What gets (or should get) the inbound PSTN calls that are forwarded to Asterisk?

My Asterisk / Sipura system works perfectly except it will not receive inbound PSTN calls after several months of work.

I'd be very grateful if someone would give me a simple explanation of how to direct inbound PSTN calls to my SIP or POTS phones?

My system as setup by the Sipura Wizard has extensions in sip.conf for sipurfxs1 (Line 1), sipurfxo1 (PSTN line) and pstn-spa3k (also PSTN line) all going to context [home] and a dial plan 8 entry of <S0stn-inbound>.

Larryalk

[SME]

Originally Posted by larryalk

Thanks SME. I'll just leave the "asterisk" user alone.

The Wizard states that
"The PSTN Line will also be connected to Asterisk as a different extension for the purposes of forwarding inbound PSTN calls to Asterisk;"

Is the above reference to the "asterisk" user or extension?

What gets (or should get) the inbound PSTN calls that are forwarded to Asterisk?

My Asterisk / Sipura system works perfectly except it will not receive inbound PSTN calls after several months of work.

I'd be very grateful if someone would give me a simple explanation of how to direct inbound PSTN calls to my SIP or POTS phones?

My system as setup by the Sipura Wizard has extensions in sip.conf for sipurfxs1 (Line 1), sipurfxo1 (PSTN line) and pstn-spa3k (also PSTN line) all going to context [home] and a dial plan 8 entry of <S0stn-inbound>.

Larryalk

I found that the wizard ended up giving me a bad dialplan (my fault somehow I'm sure) so I had to change it to a valid extension. For example, in the wizard I used 4000, 3000 and 2000 so it suggested a PSTN dialplan of (<S0:2000>) but my FXS port and analog phone were on 3000 so I just changed it and inbound PSTN started working.

Good luck...

[larryalk]

Originally Posted by SME

I found that the wizard ended up giving me a bad dialplan (my fault somehow I'm sure) so I had to change it to a valid extension. For example, in the wizard I used 4000, 3000 and 2000 so it suggested a PSTN dialplan of (<S0:2000>) but my FXS port and analog phone were on 3000 so I just changed it and inbound PSTN started working.

SME thank you very much. I think you triggered the solution to my problem.

In PSTN tab I changed Dial Plan 8 to be:
(S0<:sipurafxs1>)
where sipurafxs1 is exactly the same UserID in the Line 1 tab. Suddenly my POTS phone connected to Sipura rings and works!

Who knew it was that simple! Shouldn't that have been made clear in the Wizard?

Is your extension 2000 exactly the same as your UserID in the Line 1 tab? If not, could you post both extension 2000 and the UserID for Line 1?

One final point. I have been told that the correct syntax for Dial Plan 8 is to put the < bracket just before the : of the extension name, not right after the opening ( parenthesis mark. I asked Eric about that but got no response. In other words,
(<S0:2000>) is not correct but
(S0<:2000>) is correct.

If I'm wrong on this I'd appreciate a correction from someone.

Larryalk

[SME]

Originally Posted by larryalk

SME thank you very much. I think you triggered the solution to my problem.

In PSTN tab I changed Dial Plan 8 to be:
(S0<:sipurafxs1>)
where sipurafxs1 is exactly the same UserID in the Line 1 tab. Suddenly my POTS phone connected to Sipura rings and works!

Who knew it was that simple! Shouldn't that have been made clear in the Wizard?

Is your extension 2000 exactly the same as your UserID in the Line 1 tab? If not, could you post both extension 2000 and the UserID for Line 1?

One final point. I have been told that the correct syntax for Dial Plan 8 is to put the < bracket just before the : of the extension name, not right after the opening ( parenthesis mark. I asked Eric about that but got no response. In other words,
(<S0:2000>) is not correct but
(S0<:2000>) is correct.

If I'm wrong on this I'd appreciate a correction from someone.

Larryalk

I had to change mine to 3000, which IS the uid in the spa line1 tab. The wizard gave me 2000, which was not a valid extension. I'm not ready to blame the wizard, maybe I just didn't understand the directions clearly and gave it 2000 when I shouldn't have. As for my conf files, I had already posted (edited versions) of them here: http://forum.voxilla.com/asterisk-us...oad-19836.html

As for the "<", I saw a post suggesting it was wrong from the wizard but my fxs started working so I haven't even tried it the other way, perhaps it'd be better but I haven't gotten there yet. I'm still working on getting outbound pstn working right.

Glad I could help...

[maurda00]

I am new to this forum and have read thru the thread on SPA3000 for Asterick. I'm confused (nothing new), I currently used an internal Digium TDM400P with 2 FXO ports. If I move to the SPA what about the zapata.conf file? Where do I set signalling and channels on the Asterisk Server or do I need to worry?

I ask because I spent a good week trying to get a Audio Code MP114 to work without success. The Digium works great but I really want an external gateway so I can use a laptop for customer demos.

[eric]

You configure SIP gateways in the sip.conf file.

[SPA3000]

Question for Phoneboy:

Just configured SPA3102 with wizard for Asterisk with FWD.
Asterisk runs on Mac. Calls to 500, 600 work fine, in comming FWD works fine,...BUT outgoing FWD calls (initialized with default **393 prefix) terminats into busy signal after one (1) ring which is also not received at th recipiants end. I suspect dial plan! - dial plan is configured as per wizard: ([2-79]11<:@gw0>|xx.|*xx.|**xx.|<#1,:>xx.<:@gw1>|<#9,:>xx .<:@gw0>|<#9,:>*xx<:@gw0>)

What do you think???? I appreciate any help, since I can,t get this to work for day (and nights) Thank you.