similar to: Starting things off without a dial tone

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 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

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

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

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.

+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

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 >

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 "steps" closer > together near the zero

Is anybody else using the following combination: * a TE410P card (wct4xxp driver) * a BT ISDN connection * DAHDI 2.3.0.1 * Asterisk 1.6.2.9 I'm trying to configure a new box to replace a legacy system (same hardware; some old version of Asterisk with Zaptel; works lovely but hopelessly out-of-date) and not having much joy. Specifically, I couldn't get it to see a D-channel on

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

Hi list! I noticed that when the phone of my wife calls the gsm codec will be used, but if someone calls the phone, alaw will be used: 00493511111111 calls 00493512222222: OpenWrt*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 192.168.200.11 00493512222222 5305ad0e07977dd 0x4 (ulaw) No

Hi, Long story short - I have an ancient Britsh Telecom phone attached to my Asterisk PBX via Dahdi. It works beautifully, receiving calls, and the call quality is excellent. However, dialling out is impossible, as Asterisk consistently mis-reads the number of pulses the dial sends (it could be a squiffy dial, I'm not sure). Not to mention the fact that, in today's modern "want

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

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 > MOBILE=EXTRA/r1 > for an

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

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

Hi, I need dialplan to set INCOMING call forwarding during lunch break to my secretary. I want that I can set call forwarding by dialing an extension number to turn it ON or OFF. I am using asterisk 11. Thanks Abdullah Faheem -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi. I just want to make sure I understand this before doing something that might break things spectacularly for our users and customers :) We are using Asterisk 1.6.2.9 and my programming language of choice is Perl. I want, when a call comes in on someone's DDI number (which the person who dialled it can only possibly have obtained by dialling 1471 after we called them), to be able to