NEW SERVER INFORMATION IN ALL SETUP GUIDES

tjardick,


My tests showed that calls to PSTN numbers failed and calls to SIP (FWD echo test) went through using Cisco 7940 and Linksys PAP2. Using X-Lite both types were successful. It looks like the difference is that X-Lite allows configuration of domain realm, but the others only the SIP proxy. Using the numerical IP address gets round this problem at the moment as it is the same for both proxies.

HTH

alistair

Thanks Alistair,

I think Tj has this one worked out, but might take a day or so to fix up, as some changes need to be made on the PSTN switches.

We might rope you in for another test when that's done if that's ok.



Dean

Quote:

We might rope you in for another test when that's done if that's ok.

No problem, glad to help.

alistair

HI alistar (and others),

I have 2 possible causes for the problem. I made the first change, can you let me know if you are able to dial PSTN now or not?

If not i guess i have to do the 2nd change too, but trying to pinpoint the problem so doing it step by step.

Thanks alot!

Regards,

Tj

tjardick :

I have 2 possible causes for the problem. I made the first change, can you let me know if you are able to dial PSTN now or not?

Hi tjardick,

I have just set my Cisco phone back to using sip.voipuser.org and sorry to say the result is still the same. The invites go to number [!at] sip.voipuser.org (replace the [!at] with a @) and get no reply so eventually time out. Calls to FWD still work

alistair

Ok made some more changes, any chance you could try again ?

Thanks,

Tj

Hi Tj - Same problem for me too. I was hoping that the server changes would resolve this for me - I can make SIP calls but PSTN still doesn't work for me from Voipuser

Hi tjardick,

Been out for a while - market day here. I tried again and it is still not working on PSTN.

alistair

Linksys PAP2 cannot dial a PSTN service on either server

voipuser.org or sip.voipuser.org

can dial another voipuser using his 0844 number, works fine

voipuser.org was working now you get a failed call tone instantly

using sip.voipuser.org get nothing for about 10 seconds then the faild call tone

Any idea?

Roger

Ps: incoming calls seem to work, well the phone rings anyway!

Latest report is that neither X-Lite nor Cisco can make calls to PSTN using either proxy.

alistair

My linksys PAP2 is now working on sip.voipuser.org

can dial out PSTN

Roger

I reverted back the old changes and made some new changes, i'm hoping that alistar is now also able to use his cisco on the sip.voipuser.org domain?

Regards,

Tj

Hi Tj,

Cisco now can call PSTN using sip.voipuser.org. Will test the other clients and let you know if there are any problems. I am only testing the call setup, not audio. Looks like you are having success.

alistair

It looks like it works indeed. I can dial PSTN numbers and get a tone using sip.voipuser.org (proxy field) on my Sipura 1001. Congrats.

Good to see it all works now.

This was an important step to more redundancy and performance with the system in the near future.

Thanks everyone for testing.

regards,

Tj