PSTN gateway unavailable for SIP2SIP

Added by Adrian Georgescu about 4 years ago

It seems like our PSTN gateway provider went out of business and we were left holding the bag (GRN VoIP). We are working on finding another PSTN termination partner. The balances stored in SIP2SIP platform will remain unchanged, no money was lost.

I apologise for this and keep you posted.

News: September 4th: We have a new PSTN carrier now. Please provide feedback about quality.

Adrian Georgescu


Comments

Added by Adrian Georgescu about 4 years ago

A new outgoing trunk has been setup. We are investigating interoperability issues.

Added by Yonghan Ching about 4 years ago

I still cannot call to a PSTN number... Error 403 PSTN calls forbidden is the error I got from CsipSimple.

Added by Tijmen de Mes about 4 years ago

We still have interoperability issues with the PSTN trunk and it is currently not working.

Added by Alex Rath about 4 years ago

I had trouble getting code 408 and strange delays calling PSTN until tomorrow morning, when I deactivated all audio codecs except from G722 and G711a in the settings of my phone (G711u, G726 and G729 are now switched off – when I activated all codecs again in order to verify it, the problems were back, thus it must be the codecs making trouble). G722 and G711a only works perfectly now.

Added by Alex Rath about 4 years ago

…and I did a lot of testing today, calling also numbers in the US and Japan (the G711u countries) and it works. (As the users of G711u are obliged to offer G711a for international calls, G711a seems to be in fact the only mandatory codec, but thats another topic.)

Added by Alex Rath about 4 years ago

Does work:

m=audio 8976 RTP/AVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Makes trouble:

m=audio 11236 RTP/AVP 9 0 8 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Added by Alex Rath about 4 years ago

OMG – Why did you change again the SIP broker?
sip.voippro.com worked perfectly for me now (and has even G.729), the new one (sip.clearcom.mx) returns server error 500 when calling to Austria.

Added by Adrian Georgescu about 4 years ago

We tried two different providers but both exhibit various kind of problems. We are working to solve the problem but finding a partner that just works if like finding the nidle in the haystack.

Voippro was not providing any support at all, they did no even understand the question we were raising. Clearcom has per destination issue but they do work on solve the issue and understand what is going on.

Added by Alex Rath about 4 years ago

I see. I would like to propose a Solution for +43800 that will work quite reliably and it's even free: Please redirect all +43800 numbers directly to (for example 0043800700700 -> ). It's a VoIP friendly provider from Austria who provides a free high quality gateway for Austrian freephone numbers.

Added by Alex Rath about 4 years ago

…this worked over e164.org until now but e164.org seems to be dead, thus a direct redirect to selfnet.at would be a perfect replacement.

Added by Adrian Georgescu about 4 years ago

We have a new PSTN carrier now. Can you try again?

Added by Alex Rath about 4 years ago

The codec(s) G.711u and/or G.729 and/or GSM have/has to be supported by the client and activated to make it work. The new carrier does NOT support G.711a.

Added by Alex Rath about 4 years ago

Ok, there are still some issues:

When calling to Slovenia the code "486 Busy Here" is returned. I tried to call mobile phones of 2 differend providers.

When calling to Austria, there are sometimes caller ID issues: A phone I called displayed the prefix +01143720/… instead of +43720/…, thus the North American international prefix AND the plus got added. Maybe thats also a reason, why some calls don't work, as the called network does not accept the incorrect prefix +01143720/…

In genaral the "500 Server Internal Error" is returned in some cases, but when calling again it works.

Added by sean farrell almost 4 years ago

I added credit to my account in an effort to support sip2sip and to make inexpensive outbound calls but it seems I receive message stating PSTN calls are forbidden. Will it take a little time to have the credits activated on my account? I see the $20 payment visible at the credit tab.

Thank you for continuing to develop sip2sip!

Added by Tudor Popovici over 3 years ago

same problem, i added credit and when i want to call a real number says:PSTN calls are forbidden.any help please?