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Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > On Monday 06 Jul 2015, Luca Bertoncello wrote: >> Well, but for voice quality, which codec is better? >> alaw or gsm? > > A-law is better for voice quality (sorry, thought my original > explanation was > obvious). But note that if the destin...

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > Yes. You should definitely be using A-law for calls to the Outside World. Well, I wanted to change these settings, but I'm not sure, where I have to do that... I think in the users.conf, but I think, the "allow" keywords is for the network... How c...

...uying ip phones 1000 analog ports sounds like hell and if it was me I would be embarrassed to have a setup like that tied to my name if I was a consultant etc. Someone will come in after you and ask who set it up and the customer will say you :) On Feb 17, 2016 4:23 AM, "A J Stiles" <asterisk_list at earthshod.co.uk> wrote: > On Wednesday 17 Feb 2016, Goke Aruna wrote: > > Hello all, > > Can someone recommend what hardware to use for a 1000 analogue line > > capacity asterisk PABX? > > > > Regards > > A PCI express card with four primary rate ISDN p...

>>>>> "AS" == A J Stiles <asterisk_list at earthshod.co.uk> writes: AS> If you are paying for a business-grade Internet connection, you AS> should get a static IP address -- or a block of them -- as AS> standard. Maybe you need to change your ISP? In some places (including here) static ip is not affordable. -JimC -- Jame...

I'm not sure if this is an Asterisk thing, a handset thing or a telco thing, so please be gentle with me if this is not the right place to ask ..... When placing a call over a SIP channel to a mobile phone, if the phone is engaged, it does not return a BUSY status straightaway. Rather, I get a ringing-out tone for the timeout duration specified in the Dial() statement; *then* I get

Hi, I've upgraded my asterisk 1.4 to the version 1.8.11. After making some adjustments to the configuration files to port it to the new version, calls between registered phones in asterisk, work fine, but inbound calls coming from the SIP trunk I have with a telco to asterisk, don't work anymore. I don't know why!... This is the SDP portion that comes in the INVITE messages of calls

Hi, I need to make calls to a list of numbers one at a time and once the user pick the phone connects to an IVR where I can get few data, after a call finishes the 2nd number get called and so forth. I'm familiar with Asterisk/Elastix but the Campaign feature on Elastix does not seem to fill this need. I'm now looking GoAutodial & AsterCC. Anyone with an idea to solve this issue I

...warding and the line was free within seconds. Now > we need to scale up the setup but GSM gateways a very very expensive > if we want to scale upto a 1000 DIDs, which means thousand SIMs and a > gateway/gateways big enough. > > On Wed, Mar 18, 2015 at 3:43 PM, A J Stiles > <asterisk_list at earthshod.co.uk <mailto:asterisk_list at earthshod.co.uk>> > wrote: > > On Wednesday 18 Mar 2015, Rizwan H Qureshi wrote: > > Hi All, > > I have to forward incoming call on PRI back out to PRI but I need the > > original Callerid to passthrough. Is it possibl...

...all, telco took care of the forwarding and the line was free within seconds. Now we need to scale up the setup but GSM gateways a very very expensive if we want to scale upto a 1000 DIDs, which means thousand SIMs and a gateway/gateways big enough. On Wed, Mar 18, 2015 at 3:43 PM, A J Stiles <asterisk_list at earthshod.co.uk> wrote: > On Wednesday 18 Mar 2015, Rizwan H Qureshi wrote: > > Hi All, > > I have to forward incoming call on PRI back out to PRI but I need the > > original Callerid to passthrough. Is it possible with DAHDI PRI cards > > without involving the ser...

...d_php installed ? > > Which distro are you using ? > >> -----Original Message----- >> From: asterisk-users-bounces@lists.digium.com >> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >> Julius Kidubuka >> Sent: 03 March 2005 11:45 AM >> To: asterisk_list@burntwires.com >> Cc: asterisk-users@lists.digium.com >> Subject: [Asterisk-Users] Getting phpconfig to work? >> >> Hi, >> >> I have just tried to get phpconfig to work but to no avail. >> In my browser I type; http://ip-of-machine/phpconfig/ and >> t...

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: Hi, > GSM is the native codec used for calls to mobile phones; it uses lossy > compression to achieve a low bit rate. > > A-law is the native codec used by physical exchanges on the land line network > (PSTN and ISDN). It is non-lossy. It works by arrangin...

We apparently have a spam bot subscribed to the list and replying *directly* to anyone who posts on the list. The E-mails, generally use the name Alyssa or a Katie in the mail and have images attached. They come from a variety of addresses that so far don't appear subscribed to the list. However spammers don't typically subscribe to lists at the addresses they send from or appear to send

2015-02-26 10:45 GMT-06:00 A J Stiles <asterisk_list at earthshod.co.uk>: > > You just need to use call groups. > > In your chan_extra.conf (if it's an OpenVox) or chan_dahdi.conf, add > something like > group=1 > to the definition for each span. > > Now in the [globals] section of your dialplah, have something...

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > On Thursday 11 Jun 2015, Luca Bertoncello wrote: >> Now my problem is to check in my dialplan if the peer, that originate >> the call, is reachable, and if not, to give an error... >> >> Is there any function to know if the peer is reachable? &g...

2015-02-27 10:25 GMT-06:00 A J Stiles <asterisk_list at earthshod.co.uk>: > O.K. So what does your existing Dial() statement in extensions.conf look > like? > apology, put the gateway was sangoma but is a openvox , all my outgoing calls out for this context: [my-mobile-out] exten => _NXXXXXXX,n,Dial(SIP/1003/${EXTEN},55,rT) exten =...

Yes, it is enabled on port 5060. I do receive a TCP ACK back from the server, so I know the TCP segment is received at the server hosting the Asterisk build. On Wed, Feb 17, 2016 at 8:36 AM, A J Stiles <asterisk_list at earthshod.co.uk> wrote: > On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote: > > OK. Let me ask this. Is anything else necessary, except choosing TCP as > the > > preferred protocol on the client, to make TCP w Asterisk work? At the > > moment, I have only changed one...

On Monday 20 Dec 2010, Olivier wrote: > 2010/12/20 A J Stiles <asterisk_list at earthshod.co.uk> > > > Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application > > (written by someone else before me) which sets up calls by creating > > files of > > the general form > > > > Channel: SIP/$INSIDE_NUMBER > > Contex...

Dear All, Is there anyone who is having FastAGI support for C & C++? We do have FastAGI working for the JAVA and rest of the language / script. But I am unable to find FastAGI for C/C++. Please let us know how to write FastAGI using C/C++. Thanks in Advance, Kashyap -------------- next part -------------- An HTML attachment was scrubbed... URL:

...free within seconds. Now >> we need to scale up the setup but GSM gateways a very very expensive >> if we want to scale upto a 1000 DIDs, which means thousand SIMs and a >> gateway/gateways big enough. >> >> On Wed, Mar 18, 2015 at 3:43 PM, A J Stiles >> <asterisk_list at earthshod.co.uk >> <mailto:asterisk_list at earthshod.co.uk>> wrote: >> >> On Wednesday 18 Mar 2015, Rizwan H Qureshi wrote: >> > Hi All, >> > I have to forward incoming call on PRI back out to PRI but I need the >> > original Callerid to pa...