similar to: Park/VoiceMail on DAHDI congestion

Asterisk 1.2.16 I have an extension "102" with a Polycom 430 I am trying to protect against forwarding loops If I set the phone to forward the line to itself, extension 102 I get the following -- Got SIP response 302 "Moved Temporarily" back from 206.83.240.18 -- Now forwarding Local/102@mycontext-b2ee,2 to 'Local/102@mycontext' (thanks to

I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis. I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter. [mycontext] exten =>

Hi, I'm running Asterisk SVN-trunk-r226890 and whenever I restart asterisk and try to make a call I get the following error message: ======================================================================================== -- Executing [6781948 at default:1] Dial("IAX2/iaxy-7477", "DAHDI/g1/96781948") in new stack [Apr 25 13:00:10] WARNING[3772]: app_dial.c:1806

hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06xxxxxxxx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten => s,1,Ringing() exten => s,n,Playback(hello-world) exten => s,n,Dial(SIP/105) exten => s,n,Hangup() it works with one number how can i do in order to create a

Using CVS from 7/12/04 and trying to get the voicemail broadcast feature to work. Voicemail.conf has [mycontext] 3722 => 1234,BroadCast Test,,,cc=*@mycontext . then many other voicemail boxes. ----- whenever I leave voicemail at box 3722, only box 3722 gets the voicemail. It is not expanding it to other voicemail boxes in the [mycontext] context. Even if I replace the cc= line with

On machine A I have something like the following in extensions.conf: [iax-extensions] exten => _9.,1,Dial(IAX2/machineB/${EXTEN:1}@mycontext) exten => _9.,2,NoOp(DIALSTATUS=${DIALSTATUS}) exten => _9.,3,Hangup On machineB I have something like this: [mycontext] exten => 2002,1,Dial(SIP/2002,60) exten => 2002,2,NoOp(DIALSTATUS=${DIALSTATUS}) exten => 2002,3,Hangup If I use a

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to

Hi folks Is it possible to turn off the native bridging on Asterisk? I've been hacking about app_disa.c to support account & pin numbers, that tag the calls depending on who logs in..... It all works fine, then dials the destination number they requested. My setup is as follows [ENDPOINT] <IAX1> [MYASTERISKBOX] < IAX1 > [TELCOBOX]<>(PSTN) If i dial

Hi all, In order to do loadbalancing between my two *, i wanted to stock all things concerning voicemail on a NFS partition... I see that the voicemail system put his files onto two differents directories : /var/spool/asterisk/voicemail/mycontext etc. and /var/lib/asterisk/voicemail/mycontext etc. I've two questions : Why ? and how can i do to centralize the destination of the messages AND

Hello all, I would like to set a different caller-id for the second leg of a call when doing an originate. For example: Action: Originate Channel: sip/1234 Context: mycontext Exten: 1 Priority: 1 Callerid: "123 <123>" Async: true This sets the caller-id correctly when dialing sip/1234, but I would like to set the caller-id for the second leg of the call (the one that goes to 1 at

Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

My SIP-provider sends my a SIP-invite like this : INVITE sip:329298yyy6 at 80.XX.XX.69:5060 SIP/2.0 Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c Max-Forwards: 70 From: <sip:321445xxx6 at 80.XX.XX.69>;tag=f395877e02bf8eb2fd8f5a0e To: <sip:329298yyy6 at 80.XX.XX.69> Call-ID: f395877e02187250fd8f5a0f at 80.XX.XX.68 CSeq: 1 INVITE User-Agent: SysMaster VoIP

I can't seem to retrieve a parked call. Here is what I do: Call from phoneA to phoneB Answer phoneB Press Flash/Callwait on phoneB Press 700 to park the call A voice says that the call is parked at 701 When I try to dial 701, I don't get connected to the parked call Below is the asterisk output when I tried to park

Sorry to resurrect an old thread, but I finally got around to testing this approach (round tripping through bitcode in memory) and it works beautifully - and isn't that much slower than cloning. I have noticed however that the copy process isn't thread-safe. The problem is that in Function, there is lazy initialization code for arguments: void CheckLazyArguments() const { if

I have two servers connected together. Server 1: RHEL 3 running Asterisk SVN-branch-1.2-r8735 Server 2: LinksysWRT54GS/Whiterussian RC4/Asterisk 1.0.7 Trunk call between Server 1 -> Server 2 phone rings, recipient picks up, no audio. Message in logs on Server 1: chan_iax2.c: Peer did not understand our iax command '24' There is no message in the logs on the WRT54GS as most debug

In the example I posted previous, there is an obvious gaping security hole, it would be trivial for someone to read the querystring and exploit it to make free phone calls, spoof caller ID (if you allow the CallerID to be set with a QueryString value), etc. You want to make damn sure that the URL is not publicly accessible or somehow obsfucate the querystring, or use POST. In my case, I

Currently using PJSIP. First, they want me to get this working with the existing PJSIP configuration, but then setup a second box using chan_sip performing similar work. For PJSIP... I currently have an endpoint configured to a system using IP based authentication. It is configured with a match setting in the endpoint section. All channels coming from that IP address go to this endpoint. They

Per the wiki: Logging off 1. call the extension for AgentCallbackLogin 2. enter your password followed by # 3. when asked for the extension number just press # But if your exten=> is this: exten => 2010,1,AgentCallbackLogin(3333|3044@mycontext) How do they logoff per the wiki's directions? If you use ACBL as above, it never asks you for the extension number because you have

I was running some sample code from a help page tonight and wished to redirect the sample output to my Desktop under Linux (Mint-Debian 64 if it makes a different). I was surprised to find that file name expansion using the ubuquitious "~/" was not recognized, in fact it caused R to crash. Is this expected behavior? See my output below and sessionInfo(). Also, not shown, but

I am using AMI Originate to perform a new outbound call. The SIP Provider we send the call to wants us to pass the caller id of the person we are calling for in the Contact header. For the AMI Originate, I pass the caller id information data in the CallerID field. However, this is never being passed through the PJSIP INVITE header Action: Originate ActionID: S598 Channel: PJSIP/133 at 1002