[bulimia]

Ive been playing with an Asterisk setup and an SPA3k device and am very impressed on how easy it's been to get them working.

Eventually I found Voxilla's configuration tool and that make it even easier. Well ... the problem is always on the details.

I have some X-ten SIP phones connected to * and the SPA3K phone can dial out via PSTN, all other extensions can dial out to PSTN and among themselves.

Incoming calls via PSTN worked and were routed to the proper extension/context within * until I decided to enable the "PSTN CID->VoIP CID" option. Now I get a reorder tone when dialing from outside (after the device answer) and my * console shows:

Dec 26 21:39:05 NOTICE[9049]: Failed to authenticate user PSTN-No ID <sip:799XXXX@192.168.102.250:5060>;tag=b266b0533d7f4bb2o1

Versions: SPA3k FW 2.0.11(GWg), *version 1.0.3

At this point I have the SPA3k configured using the Voxilla configurator but with PSTN CID->VoIP CID =yes. The * configuration is pretty much the suggested on the configurator as well (contexts my be a bit different)

I even included the caller ID as an extension for the caller ID but it did no help:

exten => 799XXXX,1,Dial(SIP/150)
exten => 799XXXX,2,Voicemail(u150)

I know it's very likely a problem with my * config but I hope someone on this list with a working config could provide me with some solution (or working config files: SIP.conf and extensions.conf including contexts) so I can get my setup to work with CID.

Thanks!

[nabeelj]

In sip.conf, under the username entry that the SPA3000 PSTN-to-VoIP gateway uses, add insecure=very. It should work, but I don't know how "secure" it might be.

[bulimia]

Originally Posted by nabeelj

In sip.conf, under the username entry that the SPA3000 PSTN-to-VoIP gateway uses, add insecure=very. ....

I have already tried that, I tried again just in case but the same behaviour persists.

Thanks for your suggestion anyways.

[bulimia]

Got it working by including the "insecure=very" on the section [pstn-spa3k] which I believe is used for "Outgoing" calls TO the PSTN. I'm really puzzled, can someone explain? I could understand if it would have worked for the extention used to register the SPA (PSTN user) but that did not help.

Perhaps I will take this question to the Asterisk group.

Cheers!

[ichilton]

Hi,

Did you get an answer to your question? - i'm having the same problem.

Thanks

--ian

[thedriver]

This also solved my problem. but I too don't understand why added the insecure line to the sip entry for the outbound calls has allowd inbound from the other entry to work. Anyone with more skills have an idea (PhoneBoy)?

Thanks,

Chris

[mnaffar]

I am in India and the telco does pass callerid
pstn<->sip3000<->asterisk<->x-lite
I am not gettin caller id transferred to my x-lite phone nor on my asterisk
my config is like this on the sipura3000

PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable : yes ( i have tries no also)
PSTN CID For VoIP CID: yes (i have tried no also)
VoIP Answer Delay: 6
PSTN Answer Delay: 10
PSTN Ring Timeout: 5
PSTN Ring Thru Delay: 6

[mnaffar]

i solved it
i put an dtmf to fsk converter

[rajat]

Originally Posted by mnaffar

i solved it
i put an dtmf to fsk converter

Hey mnaffar,
I'm looking for the same thing in Delhi. Any ideas where I can get it from ?
Thanks in advance.