similar to: set(CALLERID(name) not working

I'm having trouble setting callerid with teliax. I use a simple dial-out subroutine to set the callerid depending on the calling extension, and then dial out. Teliax is saying they're not seeing any callerid info. [DialOut] ; subroutine for dialing out. exten => s,1,NoOp(Context: DialOut called with outgoing number ${ARG1} ) exten => s,n,NoOp(${CALLERID(num)}XXXX) exten =>

I've have a simple caller id lookup on incoming: [teliax-in] .......... exten =>s,n,GoSub(set-callerid-name,0${CALLERID(num)},1) ................ [set-callerid-name] exten => 0,1,NoOp( no CALLERID num set) exten => 02135590993,1,Set(CALLERID(name)=Matthew ) ............................................... exten => _0!,n,NoOp(CALLERID: ${CALLERID(name)}) exten => _0!,n,Return()

I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten

On 9/12/18 1:22 PM, Joshua Colp wrote: > On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: >> I understand that HangUp() hangs up the calling channel. I want to >> hangup the called channel. >> >> SIP/mycall-xxxxx calls and bridges with DAHDI/1-1. >> >> I send SIP/.... to listen to a long, very long, file. > > Define "send". How are you

My internal lan is small, 100mb, all wired. aastra phones. sip show peers ....... 142/... 10.10.10.42 D A 5060 OK (136 ms) 144/... 10.10.10.44 D A 5060 OK (138 ms) 145/... 10.10.10.45 D A 5060 OK (133 ms) But pings are < 1ms: ping 10.10.10.42 ........ rtt min/avg/max/mdev = 0.479/0.483/0.497/0.021 ms Why are the sip latencies so

For our internal fax machines, I'm checking if the faxes are going to branch offices. If they are, I want to capture and email them to the branches. I've set up extension 8447 to test this. A fax machines is connected via an SPA 2102 on 173. Any calls from 173 are sent to: [outbound-fax] exten => 8447,1,Answer() exten => 8447,n,GoSub(Capture-Fax,s,1) exten

I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-xxxxx calls and bridges with DAHDI/1-1. I send SIP/.... to listen to a long, very long, file. GoSub(play-long-file,s,1) [play-long-file] exten=s,1, ;;; Here I want to hangup DAHDI/1-1, the called channel same=n,Playback(very-long-file) same=n,Hangup() How do I hangup the called channel,

On Sat, 4 Jun 2016, Frank Vanoni wrote: > Another possible approach to blacklist two (or more) specific callers > (098765432 and 012345678 as example) > > exten => _+x.,1,Gosub(blacklist,s,1) > exten => _+x.,n,.... > exten => black,1,playback(tt-monkeys) > > In blacklist.conf > > exten => s/098765432,1,Goto(black,1) > exten =>

Has any one used this? U(x[^arg[^...]]): x - Name of the subroutine to execute via Gosub arg - Arguments for the Gosub routine Execute via Gosub the routine <x> for the *called* channel before connecting to the calling channel. Arguments can be specified to the Gosub using '^' as a delimiter. The Gosub routine can set the variable ${GO

Hi, Is there some more thorough documentation of this change that has happened in 1.6? The upgrade.txt and changes.txt files mention it, but I have already seen details of this change that do not appear to be documented except in conversations on the mailing list... 1) It appears that it is no longer legal to have: [macro-contaxtA] ...stuff... [contextA] ...stuff... Is this true? Or have I

Hi, Let's say I do: Set(data=xxx,yyy) Gosub(my-sub,s,1(${data})) My subroutine will only receive "xxx" for ARG1. How can I pass a literal with a comma to a single argument in a subroutine? (The point is: when calling the subroutine I do not know if the variable has a comma or not.) Thanks, Daniel -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello. I'm having trouble with callerid on outgoing calls. The recipient of the call only sees "unknown" rather than the number I'm specifying. If I set callerid info when calling an internal extension then I see the callerid name and number when I call that extension. I did that thusly: exten => 101,1,Set(CALLERID(number)=1112223333) exten =>

Suppose I have a subroutine (called by Gosub) S that's called from a macro M and there's a goto to an illegal extension in S. That does go to 'i' in S but seems to pop the macro stack so that when there's a later fallthrough in M, the calls hangs up rather than returning to the caller of M. Is this a bug or a feature?

Hey all! I'm trying to do a bit of logic here so that a user only has to dial one code to pause/unpause in a queue (e.g. *0 will (un)pause depending on the users's state). My logic looks fine to me but every time ${PQMSTATUS} shows up empty. Here's the extensions.conf part.... exten => *0,1,NoOp(${PQMSTATUS}) exten => *0,n,Macro(user-callerid,SKIPTTL,) exten =>

Greetings list, Over the last few days I've been gearing up to replace a couple of our servers with 1.6 as something of a testbed, but I'm encountering a few problems, and wondering if anyone can help... In extensions.conf, there are a number of contexts defined for each group of users, along the lines of: [groupa] [groupb] etc. In each of those, there's a command include =>

Is anyone successfully passing Outbound CallerID to Teliax? If so can you please tell me how. Thanks in advance! Dan

I've just got Uverse installed. I had dsl, but ATT insisted I couldn't keep my old dsl, but had to switch to Uverse internet - vdsl. My setup: linux box as router : 10.10.11.252 asterisk box: 10.10.11.180 10.10.11.252 is multihomed and connected to the Uverse Residential Gateway. I've set it up as DMZplus, and it shows the public ip address as eth1. I can ssh into the

I have a 100MB internal lan. aastra's are wired. asterisk box is wired next to the switch. But look: sip show peers ........ 142/142 10.10.10.42 D A 5060 OK (137 ms) 144/144 10.10.10.44 D A 5060 OK (136 ms) 145/145 10.10.10.45 D A 5060 OK (168 ms) 150/150 10.10.10.50

With asterisk 1.8.7.0 has been running ok for months. Now, this morning, it's crashing. I can restart it, but it crashes after 10+ minutes. It dies like this -- Executing [s at macro-stdexten:2] Dial("SIP/teliax-00000019", "SIP/176,18,rtT") in new stack == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP

Hello There, I would like to play a busy tone (ie BEEP BEEP BEEP) when the underline carrier sends back 486 Busy Here. Looking at Dial parameters ( http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial), it mentioned something about the r parameter as not being very professional or something like that... Then there was: U(x): Executes, via gosub, routine x on the called channel. This is similar