similar to: Subscribe to Local channel status

Hi, There's a quite complex dialplan scenario and I found out that CDR of main channel is flushed right after hangup on Local channel. I will try to simplify my scenario: [incoming] exten => 555,1,Noop(do something before using local channel, fill some variables, play IVR menus and so on) same => n,Dial(Local/555 at office/n,,g) same => n,Noop(Notice the option "/n" and

Hello, I'm running Asterisk 1.0.1 (the same was with 0.9, 1.0). When it receives inbound H.323 call it makes connection and uses local 127.0.0.1 address to send audio stream: remoteIpAddress: 127.0.0.1 When making outbound calls from Asterisk it makes correct connection to send audio stream. Is it a bug in h.323? Is there some more settings to make in .conf files? See detailed debug below:

Hi, I need to get type of called number (TON), which is displayed in pri debug messages: Called Party Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'xxxxxxxxxx' ] Does anyone know how to do it? According to documentation it is only possible for calling number. But I need to make decision in dialplan upon the value of type

Hi, I'm trying to connect Asterisk 1.4 and Cisco CallManager 5 using SIP Trunk without MTP (media termination point). Howerver, Cisco 79xx phones do not support RFC2833, they always notify CCM5 via SKINNY channel no matter where they send RTP to. For non-MTP trunk there's Out-of-band DTMF support in CCM5 called "kpml". I wonder if Asterisk can support it. I found an

Hello, I'm using digium E1 cards and serving SIP users at Asterisk. After the following call (see below) CDR shows two records. First looks as outbound call, but the second - as inbound call. Is it a bug or intended behavior? Call flow: SIP (ext: 100) -> ZAP (national number) SIP (ext: 100) transfers to SIP (ext: 200) SIP (ext: 200) -> ZAP (national number). In CDR it looks like

Hello all, I am trying to set up a dynamic channel to be used as an Agent dialer for a queue - you know, trying to replace AgentCallBackLogin for an Asterisk 1.6. I would like to do something like: [myagents] exten => XXX,1,Set(realchan=${DB(myagent/${EXTEN})}) exten => XXX,n,Dial(${realchan},tT,60) This basically fetches the actual channel to be used for dialling and dials it. What I

Hello all, I have a number of Polycom phones 601's and 430's and I'm seeing: Got SUBSCRIBE for extensions without hint. Please add hint to s to context local-hints in the CLI over and over. I have: [local-hints] exten => 110,hint,SIP/110 exten => 111,hint,SIP/111 exten => 112,hint,SIP/112 exten => 113,hint,SIP/113 exten => 114,hint,SIP/114 The hints seem to be

Asterisk 12.5 I'm using AMI to initiate a "call me now" feature from the web site. The AMI looks like: Action: Originate Channel: Local/s at callmenow Context: dial-to-customer Exten: s Priority: 1 Async: true Variable: CHANNEL_TO_CUSTOMER=SIP/voipms/1112223333 Timeout: 999999 Dial Plan: [callmenow] exten => s,1,NoOp(callmenow: Queue without answer) same

Hello all I asked similar question some time ago but didn't get answer... Maybe this should asked in asterisk-dev or bugs.digium.com? For example, I have 3 sip phones defined in sip.conf - 101, 103, 109 and this simple dialplan: [local-ext] exten => 101,1,Dial(SIP/101,,t) exten => 107,1,Dial(SIP/107,,t) exten => 109,1,Dial(SIP/109,,t) exten =>

Hello Everyone, I've been playing with an agi script for paging sip phones. page.agi will take all available sip extensions and assign them to the global variable PAGE_GROUP. Allowing the phones to be paged from the dialplan with the new Page cmd. Extensions to be excluded are presented as arguments to the agi. Each time a page is made this agi refreshes the global variable. This works with

Hi all, i am playing around with several optipoint4x0 and run into trouble trying to get hint functionality to work. I notice that there is no status notifications. But afaik this should be implemented via the SUBSCRIBE/NOTIFY mechanism. I can see INVITE, TRYING, RINGING, ACK, BYE but no SUBSBCRIBE in my sip debug traces. I have problem to understand how hint priority works. I follow the

Hello List I am in the progress of migrating from chan_sip to pjsip. I fear I have missed something on how hints need to be specified for pjsip. For chan_sip I have configured sip.conf subscribecontext = localuser and in the dialplan I set: [localuser] exten => 11,hint,SIP/11 Now if a phone subscribes to '11' this works. Now I try to get the same working for pjsip. I understood

I have an app which handles a Mitel's command port to change the MWI lights. It detects dial tone, plays some DTMF digits, listens for dialtone-or-busy, does a manager event on what it finds, and returns. Since the Mitel command port does not give answer supervision (looks like it's ringing), and since I run this app via a AMI "originate" command, I set up an extension in

2013/3/26 Richard Mudgett <rmudgett at digium.com> > > On 03/25/2013 05:17 PM, Olivier wrote: > > > Hello, > > > > > > I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup. > > > My plan is to use this handler to update my CDRs with values such > > > as > > > Asterish and Tech cause (see function HANGUP_CAUSE). >

Hello, I'm trying to use presence with PJSIP and I have a "issue". I created correctly hint priorities like: exten => 1000,hint,PJSIP/1000 exten => 1001,hint,PJSIP/1001 Extension 1000 can subscribe extension 1001 y vice-versa. The problem is when the extension 1000 make or receive a call. In the softphone where the extension is present on buddy list, the extension appear

Hi All, I am in the process of migrating from 1.4.20 to 1.6.2.x and have stumbled across a number of "differences" between the 2 versions that are forcing me to use local channels in my dialplan (mainly centered around the different behavior of CDR fields in the 2 versions) . Previously, I would place a call via an AMI Originate action similar to: action:.Originate..

Hi Joshua > The chan_pjsip module doesn't prevent that. You'd need to provide the > full SUBSCRIBE now that it is actually finding the endpoint and coming > in. Ok, let's see if we can solve the mystery.. pjsip.conf [endpt-home](!) type=endpoint disallow=all allow=g722 allow=alaw allow=gsm ice_support=yes context=from-home allow_subscribe=yes

Hi All, I am using asterisk 1.4 branch on server. Here is a my dialplan. i have set the incoming route to incoming context, and then i have set dial with local channel, The call comes to my server and the call is routed to matched case, so my phone 1001 ring for 30 seconds. If i got the NOANSWER then the channel is not passing to next priority. I need to pass that channel to the next priority of

Hi, I have been having a problem with asterisk crashing when using local channels and realtime on asterisk 1.8.3-rc2. The example given here is I think the easiest way to reproduce this problem. In extensions.conf I have: [internal] switch => Realtime/extensions/p exten => 301,1,Answer() exten => 301,2,Dial(Local/501 at internal) exten => 301,3,Hangup() exten => 501,1,Answer()

Hello, I don't really understand how channel Local works. I need that asterisk initiate a call and get some data (DTMF). So to do that I've created this dialplan : ; extensions.conf - the Asterisk dial plan ; [general] static=yes writeprotect=no clearglobalvars=no [dtmf] exten => 1,1,Verbose(Get User ID) exten => 1,n,Dial(dahdi/1/99999999,120,G(read^1^1)) exten