SOHO account overage 26.07.2011 13:17

Forum: Miscellaneous

What happens when a SOHO account goes past 2000 minutes?

RE: Omnivoice call through problem 12.05.2011 13:06

Forum: Providers

This is with the old server. I will test the new one and get back to you.

Thanks!

Omnivoice call through problem 06.01.2011 17:32

Forum: Providers

Hi,
I have setup an Omnivoice trunk (GR DID) and registered it with PBXes. Calling in and out worked fine.

I have then tested call thru with a pin and my mobile number Caller ID as the trigger.
When I complete a call with call through, I then cannot make another one. It seems that PBXes returns a busy tone to Omnivoice which directs me towards voicemail.

The call registers to PBXes as:
Destination: s/6972xxxxxxx
IP: host3.de.o­mnivoice.eu
Trunk:omnivoice.eu
Context:ext-did
App:Goto
Duration: 00:00:00

If the call through connects, I get
Context: Callthru
App: WaitExten

Not sure whether this is PBXes or Omnivoice, but as far as I can see, Omnivoice just gets a busy tone from the pbx.

Many thanks
Dimitris.


UPDATE: This goes away when I reboot PBXes but then comes back after a successful call. I am not sure, but I think the PBXes returns a "busy" tone to Omnivoice because the call is not properly hung up.

While on a call, is it possible to instruct PBXes to hang up with a DTMF tone?

RE: SPA941 problem 07.09.2010 20:08

Forum: Terminal Equipment

Hi Diafora,
it was actually a bad connection which wasn't noticed in other software.
When I used a VPN for something else, the connection kept dropping, which explains the bad performance of the SPA941.
I now use On Telecoms and that works wonders.

btw, do you know if any mobile operators are blocking VoIP in Greece?
Just FYI, I can't get sipdroid to register and make a call on Vodafone 3G which is strange.

RE: SPA941 problem 15.06.2010 16:00

Forum: Terminal Equipment

1) www1
2) Two ISPs: On Telecoms and Tellas

RE: SPA941 problem 14.06.2010 08:55

Forum: Terminal Equipment

Hi Diafora,
thanks for the settings.

FYI, I think that the problem is broader than the SPA941.
I have also a SPA962 and a SPA3102 which sporadically show the same problem.
Rebooting sets things straight but that's hardly an efficient solution.

This never appeared before a month or two btw.

Dimitris.

RE: SPA941 problem 07.06.2010 11:00

Forum: Terminal Equipment

What are these setting.
Definitely interested, thanks for the update.

RE: SPA941 problem 30.05.2010 20:21

Forum: Terminal Equipment

Thanks for the interest.

I did a restart, a reboot and a factory reset.
Same problem persists after all of these.

I will check again and let you know.

Σε ευχαριστώ για το ενδιαφέρον

EDIT: It seems to be working now. No more long DTMF tones for the # key. Sometimes an outgoing call times out but redial works fine.
Incoming calls work fine too.

RE: SPA941 problem 26.05.2010 10:44

Forum: Terminal Equipment

I think it has something to do with the actual hardware.

When I press the # key at the end of a number to dial, I hear a very long tone (5 sec) and then it does nothing.

However, this is not a problem with other providers.

I also have a SPA962 registered and that works fine, with the same settings actually.

Don't have access to the router unfortunately so can't port-forward but can send you a saved page of SPA941 settings.

Σε ευχαριστω!

SPA941 problem 25.05.2010 10:52

Forum: Terminal Equipment

Hi,
I am using a Linksys SPA941 with the latest firmware and have a problem.
As soon as I hangup a call (inbound or outbound) the phone cannot make or receive any calls. I have to reboot it for it to work with PBXes again - but other providers work fine.

I have rebooted PBXes, changed extension settings but nothing works.
Any ideas?

Thanks,
Dimitris.

change paypal account 01.02.2010 12:13

Forum: Miscellaneous

Hi,
how can I change paypal account for payments?
I have changed the email address and tried to do a SOHO payment using the new address, but that was immediately refunded.
Should I wait for the new paypal account to get activated?
Thanks,
Dimitris.

RE: "Congestion" in call log on anonymous calls 02.12.2009 12:05

Forum: Terminal Equipment

Hi
it was actually a Linksys 941 and a 962 that had the
"Block ANC serv = Yes" as default.

Sorry to add to your workload. Please close this thread.
Dimitris.

RE: Sipgate - Congestion 19.11.2009 14:53

Forum: Terminal Equipment

Any news regarding this? Thanks for the effort.

[EDIT] any news?

RE: Sipgate - Congestion 10.11.2009 14:04

Forum: Terminal Equipment

UPDATE: Sipgate has provided feedback regarding this.

your PBX rejects the calls with an 503 error.

"Nov 8 19:36:13 ser04 /usr/sbin/openser[5818]: reply: 503 Service Unavailable -
F=sip:anonymous@sipgate.co.uk T=sip:00442088199250@sipgate.co.uk
SRCIP=188.40.65.148:34982"

Unfortunately, we are not able to tell from this side why your PBX is behaving this way.

RE: Sipgate - Congestion 09.11.2009 16:36

Forum: Terminal Equipment

Ευχαριστώ Diafora,
but the problem seems to be specific to PBXes.
The problem dissolves when I use my phone to register directly with sipgate.

I have tried everything, setting up incoming routes, privacy manager etc but nothing works. Sipgate forwards the call to PBXes which issues a congestion message and the calling user is blocked.

RE: Sipgate - Congestion 08.11.2009 20:18

Forum: Terminal Equipment

the problem remains.
I have named the trunk the same as the DID number sipgate has provided, but still I get a Congestion message if dialling without caller id.

Is there anything that can be done?

RE: Congestion 04.11.2009 21:01

Forum: Terminal Equipment

Apparently, there is a problem with Sipgate and Asterisk when the
"Allow Anonymous inbound SIP calls" is set to "no".

This happens when the sipgate trunk is called from another Cisco PBX which does not send a caller-id. The inbound route works fine from any other phone.

Is it possible to fix this? It is vital for me and very annoying.

[EDIT] Diafora - Sipgate says:
Enclosed the necessary settings to configure to Asterisk for sipgate. (Very important to have the SIP ID at the end of the register command)

So can you add my SIP ID in the register command? Instead of
register=xxxxxxx:xxxxxxx@sipgate.co.uk/sipgate
can you change to
register=xxxxxxx:xxxxxxx@sipgate.co.uk/SIP ID?

Many thanks,
Dimitris.

Congestion 04.11.2009 14:12

Forum: Terminal Equipment

When people call me through a sipgate.co.uk trunk they don't get a ringtone and I see a congestion message in the 'Call Monitor'.
Why is that? Is there something I can do to resolve this?

RE: asterisk / skype 01.11.2009 19:08

Forum: Feature Requests

No, it's not working for me.
As bobmats has written, as soon as I press the red bar, the "@skype" postfix is removed from the username in the PSTN extension.

RE: asterisk / skype 30.10.2009 14:28

Forum: Feature Requests

any update regarding this? Calling skype usernames could lead to some very interesting synergies.