Creating extension for PSTN number with extension 25.02.2013 08:52

Forum: Feature Requests

I'd like to set-up an extension to dial my work. To do so, however, the system would have to call a PSTN and at the phone tree, then dial my extension. Is there a way to build-in an option to dial a PSTN, pause for a set number of seconds, then dial a 4-digit extension? (For example, program PSTN extension for 212-555-1212PPPP1234.) Thanks for your help!

making second call WITHOUT hold 27.07.2012 01:33

Forum: Miscellaneous

Back in the day, I remember that telephone operators were able to call a second number and conference the call while they were still talking to the first WITHOUT putting the first person on hold. Is something like this possible with pbxes?

Google Voice not working? 27.04.2011 23:30

Forum: Bugs

For about four days now, Google Voice peering seems to be an "on-and-off" service. For example, I've been unable to make outgoing calls over my GV trunk.

Is anyone else having problems?!

Problems with Broadvoice 30.09.2010 18:38

Forum: Providers

Hi,

I've been using Broadvoice on my PBXes since May and everything has been working fine. On Tuesday, however, my Broadvoice service stopped functioning. I contact BV and they had me setup the trunk on X-Lite, and it worked. For that reason, they said something must be wrong on PBXes end. He mentioned something about me needing to enter my proxy server setting into the system, but I don't see this as an option in the Trunk screen.

Could something help me with some troubleshooting here? Thanks in advance!

RE: all extensions dead 20.07.2010 16:23

Forum: Bugs

Why did the moderators remove my last post? This was an extremely valid comment!

Once again: How can I switch to a different server manually if I can't access pbxes.org from my browser? When www3. goes down, I can't login!

Please do not delete this post...

RE: all extensions dead 20.07.2010 16:08

Forum: Bugs

iptel...it's nice that you give us the possibility to switch manually as an option. BUT I can't even access pbxes.org from my browser, as the server's down. So how do you expect me to switch to a different server?

I agree with bobmats...the problems over the last year are becoming somewhat unbearable. At a minimum, you should change your commitment listed on the store page of 99% uptime for premium account! That's just ridiculous!

RE: all extensions dead 20.07.2010 13:16

Forum: Bugs

Well, it was up for 20 minutes, but now it's down again!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!

Outgoing down after switching to 722! 24.04.2010 11:27

Forum: Bugs

Last night I switched to newest system software and then I couldn't make outgoing calls. So I switched back to Stable, and now I still cannot make outgoing calls.

Please help...completely disabled!

RE: www5 down again 27.01.2010 10:47

Forum: Bugs

My lines are down, too. This is incredible. I thought paid accounts were supposed have a backup or something!!

RE: Xfer to PSTN extensions not working 09.11.2009 04:37

Forum: Bugs

• A SIP User Agent (UA) can be found in many forms, but it has a SIP stack and allows you to dial or receive a SIP based phone call. SIP is a protocol which allows SIP UAs and Proxies to communicate between them.

I use both Grandstream and Aastra phones.

• The vocoder related questions can be determined easily, via the SIP UA used to make the call, or in a more complicated process, via the System Log.

Yes, both phones accept G729.

***Please be aware, I've had pbxes for about 2 years now with the same phones, and this problem has just started recently with the server crashes. Before that it always worked.
***Plus, it's not the phones, because the disconnects also occur when calls should be transferred from the digital receptionist directly (see first post).

That means, two separate (but related?) problems:

1) no transfers to PSTNs from digital receptionist
2) no blind OR attended transfers possible on INCOMING calls (Tfers possible on outbound calls)

PS: These problems exhibit themselves regardless of trunk combinations (i.e., various SIP providers) tried.

I really think it's a PBXes problem...IS ANYONE ELSE EXPERIENCING THE SAME THING?

Thanks for the feedback

RE: Xfer to PSTN extensions not working 07.11.2009 15:35

Forum: Bugs

I'm sorry for the misunderstanding:

• Do transfers to other extensions complete properly?

NO, neither blind transfer NOR attended transfer works to other extensions.

• What kind of SIP UAs are you using to initiate the transfers?

I don't understand what you mean with SIP UA?

• When the direct call to the PSTN number is successful, which vocoder is used for it?

How do I determine this?

• Do the trunks of the ITSPs you selected to handle the PSTN call, support the G.729 or another vocoder, except the G.711a?

I use sipgate.de. I don't know what vocoders they use.


Please understand, I'm just a layman and not an expert. I don't understand all this stuff...Sorry!

Danke trotzdem fuer Ihre Hilfe!

RE: Xfer to PSTN extensions not working 06.11.2009 13:53

Forum: Bugs

Hi,

A new discovery which might explain a bit. When I RECEIVE phone calls, I can make neither attended nor blind transfers. Pressing *2 or ## just sends the digits as tones.

When I make a phone call, though, I get the voice "transfer".

This seems like a programming problem. Please advise.

Thank you

Xfer to PSTN extensions not working 15.10.2009 16:30

Forum: Bugs

T-fers to PSTN extensions disconnect call immediately. For example, via digital receptionist:

Oct 15 16:20:36 VERBOSE[5840] logger.c: We're at 188.40.65.148 port 39948
Oct 15 16:20:36 VERBOSE[5840] logger.c: Video is at 188.40.65.148 port 39336
Oct 15 16:20:36 VERBOSE[5840] logger.c: Adding codec 0x8 (alaw) to SDP
Oct 15 16:20:36 VERBOSE[5840] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Oct 15 16:20:37 VERBOSE[5840] logger.c: -- Playing 'custom/DK_general_phone_tree' (language 'en')
Oct 15 16:20:41 VERBOSE[5840] chan_sip.c: Hangup call SIP/UNO127536-3f48, SIP callid 726d6d9d210094ca47a664ab6df7c320@94.231.97.6
Via direct t-fer:

Oct 15 16:23:57 VERBOSE[21236] chan_sip.c: SIP call transfer received for call 6c808bc441c411a25969634218926f6d@188.40.65.148 (REFER)!
Oct 15 16:23:57 VERBOSE[21236] logger.c: Transfer to 900 in from-internal
Oct 15 16:23:57 VERBOSE[21236] logger.c: Transfer from 703220-801 in from-internal
Oct 15 16:23:57 VERBOSE[21236] logger.c: -- Stopped music on hold on SIP/UNO127536-4d41
Oct 15 16:23:57 VERBOSE[9168] chan_sip.c: Hangup call SIP/703220-801-edae, SIP callid 6c808bc441c411a25969634218926f6d@188.40.65.148
Oct 15 16:23:58 VERBOSE[9168] chan_sip.c: Hangup call SIP/UNO127536-4d41, SIP callid 4d7fc9060a03215303277187171ca366@94.231.97.6
When I dial PSTN extensions directly, it works fine. T-fers to ANY PSTN do not work, not just to 900...tried 908 as well (different country with different trunk!).

Please advise.

PBXes now bordering on the obsurd 12.10.2009 10:13

Forum: Bugs

Clearly there are problems, and I fully understand that uncontrollable incidents can occur.

BUT, you advertise premium service with a 99.9% average reliability which has obviously NOT been the case over the last two weeks. And then you provide no telephone support at all???? EINE KULANZLÖSUNG WÄRE EINE SCHÖNE IDEE, NICHT WAHR?!

My extension 800 is down again and I can't get it up. PLEASE advise ASAP...

Thank you

not receiving v/m e-mails 08.10.2009 21:07

Forum: Bugs

and email address is correct!! Only know I received a message because of call monitor!
Please advise!

Aastra 57i CT outbound problems 06.09.2009 15:16

Forum: Terminal Equipment

I have set up an Aastra 57i CT and itworks only partially. When I try to dial out using a PSTN phone number, it says "Call Failed" on the phone. When I dial an extension, however, which is then routed to the same PSTN number, it goes through. FYI, I have no problems with incoming calls.

Can anyone help me here?

Thanks!

Barge In 18.08.2009 23:30

Forum: Bugs

Using the status screen, I can barge in on other extensions as the function suggest. When I do this, though, the call is muted. That's fine, but the problem is that pressing *1 does NOT unmute the call. Could this be a bug?

If you change the last line of
/etc/asterisk/extensions_hud.conf to: exten => _2X.,1,MeetMe(20000000${EXTEN:1},aqs,271721), would it work?

Thanks!

PRO and Premium account PROBLEM 25.07.2009 20:36

Forum: PBXes PRO

I accidentally signed up for PRO and now it's limiting my adding new extensions. I cancelled PRO and now I still seem not to have access to my Premium account. It's SOOOOO confusing. Could you check my account please?

Thanks!

Cancelled Account 30.05.2009 02:13

Forum: Miscellaneous

Because a credit card I had cancelled, I had to change my Paypal account to reflect the new CC information. After that, I received an email saying my PBXes account had been cancelled. As I just recently purchased a ONE YEAR subscription, I don't want to renew. But I also don't want my account to expire. COULD SOMEONE PLEASE HELP ME!!!

Thank you, Michael

RE: Interoperability 07.05.2009 03:52

Forum: Providers

With my SIP provider (les.net), I must be able to leave fromuser= blank in order formy caller id to work. Is there any way to do this?