similar to: Asterisk on smartphones ?

Hello Some SOHO prospects only have a cellphone and I was wondering if someone had investigate running Asterisk on a smartphone, to perform tasks such as IVR, CID rewriting, voice-mail, notifications through e-mails, etc.? Thank you.

+1 spending money to get that many fxs ports is going to negate any savings of reusing analog phones instead of buying 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

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

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 > > Context: $CONTEXT >

Thanks AJ and David, We were actually using GSM gateways by setting busy forward number on the SIMs and just giving busy signal on every incoming call, 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.

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

On Wednesday 17 Feb 2016, imperium broadcast wrote: > I kinda have it working with chan_sip. > > Dial(SIP/+${EXTEN}\;phone-context=+44 at 10.10.10.10;user=phone) > But it doesn't include the user=phone at the end when dialling out. > > "To: <sip:+4499999999999;phone-context=+44 at 10.10.10.10>". > > even adding > usereqphone=yes > to the

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 mobile phone, GSM will be > used anyway, at

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 this setting? Thanks Luca Bertoncello (lucabert

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 Context: $CONTEXT Extension: $OUTSIDE_NUMBER Priority: 1 CallerId: $INSIDE_NUMBER in /var/spool/asterisk/outgoing/ . It works very well. However, it would be nice to be able to attach an additional

Hey Don, How are you? I may be heading your way in the next month or so. Have to meet with a guy in Eden Prairie, and stop off at my brother/sisterm-in-law's as well. Got a question for you - with TBCT, who pays for the call once it is transferred? Still me as the owner of the trunk? Lets say I take a call that was dialled locally (caller believes this is "free"), and I do a

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 > On 2/02/2017, at 9:52 pm, A J Stiles <asterisk_list at earthshod.co.uk> wrote: > > <snip> > > but in simple solidarity with everyone who has ever > > been pissed off by a machine-initiated spam marketing phone call at an > > inappropriate moment, I am not going to tell you how to do it. > >

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

On a BT line, how do I determine whether the number on an incoming call has been deliberately withheld (by dialling 141) or is merely unavailable (e.g. because it originated from overseas or passed through some ancient switching equipment) ? In the first case, I want the caller to be played a message to the effect that we are not at home to anonymous cowards but if their business is

Hello Greg If money isn't a problem Intel has an optimized compiler for eVC and XScale processors http://www.intel.com/software/products/compilers/techtopics/PCA_Optimization_WP.pdf If you have any luck getting the eVC compiler closer to realtime I'd really like to know. I'm still far from realtime when using Speex 1.1.3 on a HP iPAQ (Intel pxa255). Best regards Bjoern D.

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

I'm having a problem with a new SIP trunk. Calls within the UK to fixed lines are fine, but calls to mobiles have noticeably poorer audio quality. I thought it might have been a codec issue; we have used G.726 for internal and external calls (over primary ISDN and GSM). So I tried allowing "alaw", (G.711 A-law) which is the native codec used within the PSTN in this country,

Thanks Joshua and everyone, Joshua's solution seems a lot simpler and works well. Only one thing now - The reason I named the classes as I did, was so that I could select the class based on callerID plus extension. Unless I've misread it, I'm limited to 9 switchable classes via the "digit=#" option, is that correct? Or is there a clever hack around this? extensions.conf