similar to: Hint and state

Hi, when sending IMs from endpoint to endpoint with the MessageSend() application, I can check the MESSAGE_SEND_STATUS and send another message to the sender of the message to notify them that their message was not sent when the status indicates it. This works fine with chan_sip. With chan_pjsip, this works differently in that MESSAGE_SEND_STATUS is "SUCCESS" after sending the

Last question for today, I promise! The problem: In order to disconnect calls after x minutes, I need to do this: [setup] exten => setup,1,Answer() same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes) same => n,Set(LIMIT_WARNING_FILE=/var/lib/asterisk/sounds/en_GB_TNS/time_limit_reached) same => n,Dial(Local/s at root/n,3,L(3540000:60000)) same => n,Hangup() [root] exten

Hello, We have a provider which is using Kamailio as front end. Our asterisk 13/chan_sip server has no problem to register and pass/receive calls form this provider. Now we want to move to asterisk 16/pjsip and face problem. Registration is OK but when we pass a call our INVITE never receive answer from the provider. We opened a ticket to their support but in the mean time we want to know

Le 27/11/2018 à 12:13, Joshua C. Colp a écrit : > On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...] >> >> [TOOTAiAudio] >> ; >> ; Call our gateway >> >> exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1}) >>  same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T) >>  same = n,Return >> >> exten = h,1,NoOp()

Hello I'm trying to use ChanIsAvail() to check when the landline is back to idle after a call, but for some reason, Asterisk doesn't detect that the callee has hung up after listening to MoH for a few seconds: ========== extensions.conf ;Play MoH for a few seconds, hang up, and ;check ChanIsAvail() able to detect when line idle again exten => 8888,1,Answer() exten =>

Le 17/01/2020 à 11:54, Administrator a écrit : > > Le 15/01/2020 à 19:24, Administrator a écrit : >> Hi all, >> >> we face a strange behavior while connecting an Asterisk16 instance >> with PJSIP to 2 providers: we receive error 401 Unauthorized, both of >> them having Kamailio as front-end. With other providers -we don't >> know if they run

Le 30/09/2019 à 11:45, Joshua C. Colp a écrit : > On Fri, Sep 27, 2019, at 11:31 AM, Administrator TOOTAI wrote: >> Hi list, >> >> I would like to now what is the sense of such type of entry in security.log >> >> [2019-09-27 15:12:24] SECURITY[26964] res_security_log.c: >>

Hi list, to manage an external queue agent the only solution I found is to connect a local account and redirect calls to this account using forward features from the phone (SNOM). The problem I face is that before calling the agent I would like to set extra header. Dialplan to call external agent is this one with (Gosub): [TOOTAiAudio] ; ; Call our gateway exten =

Hi I am trying to use ChanIsAvail to decide if a particular extension is available in the sip channel I am using MySQL to hold my SIP friends. and wy cvs version shows Asterisk CVS-08/02/04 my intention is, that if the extension is not available in Sip channel, I will send the call somewhere else my extensions file contains the following: exten => _[123]XX,1,ChanIsAvail(sip/${EXTEN}) exten

Hi list, I would like to now what is the sense of such type of entry in security.log [2019-09-27 15:12:24] SECURITY[26964] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="2019-09-27T15:12:24.181+0200",Severity="Informational",Servic e="PJSIP",EventVersion="1",AccountID="<unknown>",

Hi all, I am having a problem with ChanIsAvail. It always returns the same result, regardless of whether an extension is available or not. It always returns 0 Unknown Status. This is my dialplan. exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s) exten => _2XX,2,Verbose(0, ${AVAILSTATUS}) exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5) exten =>

I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten => 911,1,ChanIsAvail(Zap/1) exten => 911,2,Dial(Zap/1/911) exten => 911,3,Hangup() exten => 911,102,ChanIsAvail(Zap/4) exten => 911,103,Dial(Zap/4/911) exten => 911,104,Hangup() exten => 911,203,ChanIsAvail(Zap/5) exten =>

I'm trying to add an emergency dial to my context. However, when I try to dial it, I get caught in an endless loop. For debugging, I have pared out nearly all the control flow and just have ChanIsAvail() and Dial() called. Using two different extensions to call teh same number, I get two different actions by *. Here is the vvverbose output: -- Starting simple switch on

Guys. Anybody doing ChanisAvail on IAX2 channels? Im trying to do this: exten => s,7,ChanIsAvail(IAX2/anton:intrudercom@armando-gw) But I get that the chan is unavailable eventhough I can make calls to that channel. Is there any chatch? The channels is defined as peer and Ialso tried doing a register on iax.conf for that channel. Everything is registering ok and I CAN make the call. Any

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

Here's my situation. I have two receptionists that answer incoming lines. Each has a 7960G with 5 incoming lines each. I'm trying to set this up so each line on each phone doesn't utilize call waiting. My problem seems to be that ChanisAvail(Sip/cisco1&Sip/cisco2&Sip/cisco3&Sip/cisco4&Sip/cisco5) always returns cisco1. Here are the sip.conf entries: (mind you,

Hi All, We are using asterisk 16.5 and having an issue with the first re-invite after the call has been established. We can see the call gets up and you see in the logs the bridge type has changed and after that a re-invite is triggered. Is there any possibility to deactivate this kind of reInvite? We have some race conditions while have multiple asterisk in the call flow and the different

Hi, So I've done my research on Chanisavail, read the wiki, checked the archive but can't seem to find anything to suit my scenario. I've played around with it a lot, but I'm still scratching my head on what I need to do. What I need is to be able to accept a call by SIP and ring all telephones that are not in use (which just so happen to be on Zap interfaces, but might be SIP

Hi list: I'm having the next problem, it appear that the application ChanIsAvail is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS. I add my dialplan and the output to the cli. THanks. In the example i'm dialing from extension SIP/112 My DialPlan Secction: [macro-callonlyiffree] exten => s,1,ChanIsAvail(${ARG1}|s) exten => s,n,NoOp(${AVAILCHAN}) exten

Hi All, As Net2Phone don't permit more than one session per account, I configured about 10 sip trunks and configure multiple trunk routing but once the first trunk is used I cannot make additional calls, I also cofigure my dial plan in other way using the chanisavail command but still not working. The chanisavail command configuration is correct as I can make calls using other trunk than