similar to: setting outbound caller ID

Thanks very much for all the responses. I now have a few more things to try. I should have noted that I am using IAX2 rather than SIP to connect to my provider. I do have some internal phones that use SIP to connect to my asterisk box, as well as some corded phones connected through a Digium DAHDI-driven card. I am certain that the old number that is showing up as my caller ID is not present in

On Thu, 18 Jun 2015 11:27:08 -0600 Greg Woods <greg at gregandeva.net> wrote: > [macro-callout] > exten => s,1,Set(CALLERID(num)=##########) > exten => s,2,Set(CALLERID(name)=##########) > exten => s,3,Set(CALLERID(all)="##########" <##########>) > exten => s,4,Dial(${PROVIDER}/"1${ARG1}",30,tr) > exten => s,n,Hangup() CALLERID is a

> CALLERID is a read only variable. That's not correct. I set it all over the place in my dialplan.

Set(CALLERID(number)=XXXXXXXXXX) works here. Also check with your VoIP provider what format they want for the number. (I believe) most accept a 10-digit number, but I seem to remember reading about the odd provider that wanted a leading "1". On Thu, Jun 18, 2015 at 11:47 AM, D'Arcy J.M. Cain <darcy at vex.net> wrote: > On Thu, 18 Jun 2015 13:45:10 EDT > kenner at

hi, all asterisk 1.4.24 , zaptel 1.4.10.1 , E1 Manager API Action : Action: Originate Channel: ZAP/G1/8888888 Callerid: 12345678 Context: callout Exten: s Priority: 1 extensions.conf [callout] exten => s,1,Answer() exten => s,n,Wait(10) exten => s,n,Hangup() when the phone 8888888 pick up , it will come to callout context, after hangup, one cdr generate, but the

I have this in my sip.conf: exten => *98,1,Verbose(0,CALLERID number is "${CALLERID(num)}") same => n,VoicemailMain(${CALLERID(num)}@LocalSets,s) same => n,Hangup However, my extensions are set up so that they always show the external number, not the extension: [foobar2](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxxxxx callerid=Candace <5555551212>

Hi All, I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan: [callback-outbound] exten => _00.,1,Macro(callout|${EXTEN}) [macro-callout] exten => s,1,AGI(getchannel.php|${ARG1}) exten => s,2,Dial(Local/${OUTBOUND}@from-internal/nj||tr) exten => s,3,Hangup() but for some reason i am not receiving the argument: Executing [s at macro-callout:2]

On Thu, Jun 6, 2019 at 12:17 PM Malcolm Davenport <malcolmd at sangoma.com> wrote: > Howdy, > > There's a dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.tar.gz. > > Try that. > I noticed that was there, but I didn't try it originally because it's obviously a beta version. However, I did download it and try it. It does compile, but doesn't work correctly. For one

Seems like I post about this about once a year, when it's time to upgrade Fedora. I first got this error trying to compile a patched version of dahdi-linux-2.11.1; I noticed that there is now a dahdi-linux-complete-3.0.0+3.0.0, so I tried that one with the same result. If I compile it while running kernel-4.16.8-300.fc28.x86_64, it compiles fine, but when I try to compile it while running

Hi All. Does anybody know if Google/Android has an API I can sign up for that will allow us to query the caller ID and find out if it is spam or a robocaller? I ask because we've had increase in spam calls and I'd like to simply play dead air or something really annoying. Thanks all, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Thu, Jun 18, 2015 at 1:26 PM, Matt Riddell <lists at venturevoip.com> wrote: > Did you buy the number from your carrier? Maybe it?s set on their side > for the trunk. > That's what I think too, but they are denying this. I think what's happening is they have a customer service guy interpreting logs (probably incorrectly). When I had a Century Link POTS line, I had a

My last problem was nicely solved through this mailing list so hopefully this new problem will have the same happy outcome. My situation is that I have many extensions. Here is a sample: [client-phone](!) type=friend host=dynamic secret=XXXXXXXXXX dtmfmode=auto disallow=all allow=ulaw allow=gsm allow=g723 allow=ilbc subscribemwi=no [4165555555](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxx

On Thu, 18 Jun 2015 13:45:10 EDT kenner at gnat.com (Richard Kenner) wrote: > > CALLERID is a read only variable. > > That's not correct. I set it all over the place in my dialplan. Then someone needs to fix this page. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/

HI I am trying to use the outcall going by the wiki.( http://www.voip-info.org/wiki-Asterisk+auto-dial+out) But I keep getting the errors below. Here is a sample of a callout file. What am I doing wrong? ////Begin Outgoing.call//// Channel: sip/2075 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: managers Extension: 2184 Priority: 1 ////End outgoing.call//// Nov 9 20:32:02

I have an extension that looks like this: exten => 5555551111,1,Verbose(Door buzzer calling) same => n,Dial(SIP/user1&SIP/user2&SIP/user3) The idea is that any of the three users can answer the phone to let someone in. The problem is that if, say, user2 unplugs his phone then the call immediately goes to his voice mail and the other two do not have the ability to open the door.

> On 18Jun, 2015, at 13:44, Greg Woods <greg at gregandeva.net> wrote: > > I am certain that the old number that is showing up as my caller ID is not present in any of my config files (that includes sip.conf and iax.conf, everything in the /etc/asterisk directory has been checked). Did you buy the number from your carrier? Maybe it?s set on their side for the trunk. -- Cheers,

On 30 July 2016 at 19:32, D'Arcy J.M. Cain <darcy at vex.net> wrote: > > > Not playing the prompt changes nothing. If someone presses '*' while > listening to your answer message then they are in your mailbox. You > better have a password that they need to enter to continue. I have now tested the 'Unavailable' message by pressing "*" while

In a dual server configuration one of the two servers fail with: WARNING[3705]: res_crypto.c:335 __ast_sign_bin: RSA Signature (key sintel-voip) failed NOTICE[3705]: chan_iax2.c:5187 authenticate: Unable to sign challenge withy key WARNING[3705]: chan_iax2.c:7191 socket_read: I don't know how to authenticate sintel-user to 10.27.33.1 We use the rsa authenticatio: in the server A

This just started after upgrading to 11.11.0. After a call is completed (both ends hang up) the call still shows as active. # asterisk -x "core show channels" Channel Location State Application(Data) SIP/thinktel-0000000 (None) Up AppDial((Outgoing Line)) SIP/4164251212-00000 4165555555 at LocalSets Up Dial(SIP/thinktel/4165559999) 2 active

I tried your extension definition as suggested: exten => *98,1,Verbose(0,${CHANNEL(peername)} calling voicemail) same => n,VoicemailMain(${CHANNEL(peername)}@VoiceMail,s) same => n,Hangup But there was no change in the prompts asked, ie. the voice first asked for 'mailbox', and then 'password' as before. The prompts are not removed. Please clarify what you mean by the