similar to: Problem with incoming calls.

Trying now to set up the final part of my * switch. I must admit I've had great fun over the last week or so playing with it, and would like to thank you guys for all the assistance (past and present, since I've been trawling a lot of old posts!!!). Scenario - using voiptalk.org to supply the incoming gateway, tied to an 0845 number for convenience in testing. Internal 7960 -> 7960

Now this is really frustrating. Everything was working fine, and now it isn't ... I don't think I've changed anything that would affect this, but I guess you never can be too sure. My setup is as follows: SIP softphone (SJphone) connected to Asterisk running my Linux NAT firewall box. This is all on the internal network. Asterisk then dialing out through various means - SIP to

There was some discussion on this list recently about the voiptalk silver service. I've just had an e-mail from them saying that the price has been reduced to 2.99 per month. However, they still only provide an 0870 number whereas pipecall provide a local call rate 0845 number in the fee. Chris

Hello I have contacted my line provider who is saying that in order to get my 0845 or 0870 number to id as the incoming number on a landline that i may call i need the following. User must provide - NPI set to E.163/E.164 User must provide - TON = "national or international I have had a good search around and can't seem to find a good answer to this. Does anyone have any idea where i

I've installed successfully freePBX with Asterisk, and got various internal extensions working, however. recently my internet facing IP address has been removed by my ISP (for various reason) and I'm not going to be able to get it back for a few weeks. Is there anyway in which I can successfully receive incoming calls from my Voip-Talk.org numbers (an 0845 number) without the static

I've been using NuFone with Asterisk for a while, but I've started seeing this error with incoming calls: NOTICE[114696]: File chan_iax2.c, Line 4581 (socket_read): Rejected connect attempt from 216.234.116.189, requested/capability 0x4/0x4 incompatible with our capability 0xff03. Outgoing works just fine, but I can't get incoming to work at all. Any ideas? I googled for the

Hi All, I was able to install Asterisk and make outgoing calls. Recently I purchased two DID's and I am facing a problem configuring them to my Asterisk, I hope with the help I get from this list I will be able to configure successfully. Mu errors are Feb 28 08:31:58 NOTICE[19133]: pbx.c:1331 pbx_extension_helper: Cannot find extension context 'context_mantra2' Feb 28 08:31:58

Hi Guys, I have a problem with IE6. Using any version of prototype and any version of scriptaculous IE loses the ability to select text by dragging the text cursor over it. You can double/treble click text, or use shift and the arrow keys but you cannot use the mouse. This applies to text in both input and fixed elements. Has anyone else got this, or knows how to fix it? Cheers Paul Shannon

Hi, Is it possible to configure NTLM HTTP proxy authentication using the winbind/squid "ntlm_auth" helper, to authenticate users against a Samba 3 server? I already have the NTLM auth working against a Windows 2003 Active Directory, but I also have a completely separate Samba 3 server that I would also like to configure NTLM proxy authentication against. Please advise, as I can't

Hi all, I spent most of the last weekend testing and trying to diagnose some mostly incoming call issues. I'll start with the easy one in the hopes it might have a positive impact on the others. First- I have an account with IAXTel. I can place calls to other IAXTel subscribers and also through IAXTel to landline toll free numbers and all works great. iax2 show registry shows I am

Hi Everyone, I've got a weird problem with both Firefly & iaxLite (both IAX softphones). They don't seem to stop ringing when an incoming call is make to them. If the call is answered the conversation starts both ways but the ringing sound still keeps going and the softphones keep displaying that a call is coming in (but they do not display that the call is answered). I read

I have no analog line interfaces on my asterisk system, but I do have two UK 0870 numbers routed to two separate VoIP accounts (one with FWD, one with gossiptel). Asterisk is configured to register with these accounts. I get voice calls through just fine this way. I thought I could get one of these 0870 numbers to route through to rxfax, thus allowing folks to fax me directly. I've set up

Hi, when I make a call, chan_capi always uses controller 2, and never uses controller 1 (so I have 4 lines for incoming calls, but only 2 lines instead of 4 for outgoing calls). this is with 2 AVM Fritz cards PCI. -- _______________________________ Simone Vasoli BK s.r.l. - Brain and Knowledge e-mail: simone.vasoli[at]b-k.it cell: +39 348 0830539 tel: 0187 1874200

Hello, I am attempting to use validation.js from http://tetlaw.id.au/view/blog/really-easy-field-validation-with-prototyp e/ with form_remote_tag. The rails form helper binds the needed Ajax.Updater to the onSubmit action of the form. The validation.js also attempts to bind to the onSubmit of the form. The logic that is needed is if( valid.validate() ) { ...Ajax.Updater... } where valid

If anyone can help me with this I'll be soooo grateful :) We have a isdn30 line, with a DDI range. We also have 2 business units that have separate 0870 numbers that are mapped onto 2 DDI numbers. I would like to be able to present these 0870 numbers from the business units so that the correct number is displayed on a callerid, or when 1471 is dialled. BT claim that I can only have a

Hello All, Can anyone help me solve this problem. Am trying to extract hash-tagged words from tweets downloaded from twitteR. I can extract hash-tagged words from single tweet using (stringr) str_extract_all(tweets, "#[a-z//A-Z//0-9]+")  but cannot with more than one tweet at a time except I manually remove all regular expressions and tweets numbers such as [[1]] and [1.] I want to

We're in the process of testing some equipment and configurations and to do this we have setup a UK PSTN Gateway to Free World Dialup. Simply dial 0845 004 5566 (UK local rate call) and at the prompt enter the FWD subscriber number - within a couple of seconds you should be connected. We can also terminate UK 0800/0808 numbers for SIP/IAX -> PSTN calls, at the moment we don't have an

All Probably teaching you all to suck eggs but. My provider is Kingston Comms ( UK ) and I have had a bit of a struggle to get the * system setup on their cct's. Just thought I would let you all know what I had to do. Firstly order ISDN 110 they will try to provide ISDN 85 as standard. Make sure these settings are correctly defined. [zapata.conf] signalling=pri_cpe switchtype=euroisdn

We've been using SIP with Asterisk for a couple of years now, and it's generally worked fine. However we're now trying to use a more complicated codec setup, and I've hit a problem with how codecs are selected that I can't get around. For a simple configuration: XLite > GSM > Asterisk where GSM is the _only_ codec selected on XLite, and in sip.conf we have:

On Friday 17 November 2017 18:41:17 Ilia Mirkin wrote: > On Fri, Nov 17, 2017 at 12:33 PM, Ondrej Zary > > <linux at rainbow-software.org> wrote: > > @@ -483,8 +483,8 @@ > > nouveau 0000:02:00.0: disp: 0860: 00000000 -> 00000500 > > nouveau 0000:02:00.0: disp: 0864: 00000000 > > nouveau 0000:02:00.0: disp: 0868: 00000000 -> 04000500 >