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...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 ports, each li...

...nt, it sets the dialstatus to BUSY. Where should I be looking, if I want to reproduce the "old-fashioned" behaviour and return an engaged signal straight away? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk .

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 destination is a mo...

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 can I change t...

...are 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 service provider...

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 line in pjsi...

...n is used to denote a comment (on account of the comment mark being a valid DTMF digit). So you will have to insert a backslash before the semicolon before user=phone . -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk .

Hello all, Can someone recommend what hardware to use for a 1000 analogue line capacity asterisk PABX? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/2bcd322f/attachment.html>

...ine 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 possible with DAHDI...

...t;Codec was alaw", but there was no improvement to the sound quality. Is there something I am doing wrong, or do I need to get in touch with our SIP trunk provider? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk .

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

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 arranging the "s...

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 like >...

Hi! Is possible to use X100p TDM400P, Tdm410p, Tdm400, A400p, Ax400p or any others digium card FXO for use Fax modem? Thanks.

>>>>> "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 -- James Cloos <c...

...Maildir/.lists.hylafax-201210 doesn't exist yet, using > default permissions > Oct 01 10:57:09 lda(archive): Debug: Namespace : Using permissions > from /home/archive/Maildir: mode=0755 gid=-1 > Oct 01 10:57:09 lda(archive): Info: > msgid=<201210010954.02813.hylafax_resp at earthshod.co.uk>: save failed > to open mailbox lists.hylafax-201210: Mailbox doesn't exist: > lists.hylafax-201210 > Oct 01 10:57:09 lda(archive): Info: > msgid=<201210010954.02813.hylafax_resp at earthshod.co.uk>: saved mail to > INBOX Can anyone tell me what goes wrong her...

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 => _NXXXXXX...

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

Dear List wanna configure click2call in such a manner that my asterisk box call two mobile numbers and connect both numbers to talk. I have configured voip gateway, my internal and external calls are working fine. please help , abhi -------------- next part -------------- An HTML attachment was scrubbed... URL: