search for: mycontext

...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 SIP/exten-mycontext-102-094c2c08) -- Executing Dial("Local/102@mycontext-e1e0,2", "SIP/exten-mycontext-102") in new stack -- Called exten-mycontext-102 -- Got SIP response 302 "Moved Temporarily" back fr...

...ouble 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 => 1000,1,NoOp() same => n,Stasis(hello-world) same => n,GotoIf($[${STASISSTATUS}=FAILED]?IS_hello_world2:stasis_done) same => n(IS_hello_world2),Stasis(hello-world2) same => n(stasis_done),Hangup() For testing, I am using curl curl -v -u asterisk:asterisk -X POST http://aste...

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 callfiles with a lot of numbers i try to create a callfiles.cal...

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 cc=xxx, the vmail b...

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 of annonces in a unique directorie ? thx in advance.

...n one message will be played, and if machine another will be played theoretically after the answering machine/voicemail is done playing. By the way, I'd like to mention that this is not at all for spamming, or telemarketing. This is an appointment reminder service. from extensions.conf: [mycontext] exten => 899,1,Answer exten => 899,2,Wait(2) exten => 899,3,AMD exten => 899,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach) exten => 899,n(mach),WaitForSilence(2500) exten => 899,n,Playback(were-sorry) exten => 899,n,Hangup exten => 899,n(humn),WaitForSilence(500) exten => 89...

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 at 8000)->(slin at 192000) ReadTranscode: No When I do the same fr...

...X] but then gets transferred directly to TELCOBOX and I loose call duration etc :-( Logs from MYASTERISKBOX show the following: -- Executing Dial("IAX[<ENDPOINTUSER>@<ENDPOINTIP>:5036]/55", "IAX/<MYUSER>:<MYPASS>@<TELCOBOXIP>/<DESTNUMBER>@<MYCONTEXT>") in new stack -- Calling using options 'exten=<DSTNO>;callerid=<SRCNO>;language=en;context=<MYCONTEXT>;username=<MYUSER>;formats= 2;capability=2;version=1;adsicpe=0' -- Called <MYUSER>:<MYPASS>@<TELCOBOXIP>/<DSTNO>@<MYCONT...

Hi Guys, We have just set up Asterisk (CVS Head) for a realtime enviorment using MySQL & Asterisk Addons. I have populated the "sip_buddies" table with the same information that is came from our sip.conf, however registration seems to fail for the softphone we have set up. Does anyone have any idea as to what I should be looking for here? I'm not getting any error messages

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 phone on machineA to dial 92002, then of cour...

...CEL,OPTIONS,BYE,INFO,REFER,SUBSCRIBE Content-Type: application/sdp Content-Length: 261 This is my sip.conf : [outgoing] type=peer host=sip.XXX.tld username=329298xxx6 secret=my-secret fromuser=329298xxx6 disallow=all allow=gsm allow=alaw ; incoming [329298yyy6] type=user host=sip.XXX.tld context=mycontext disallow=all allow=gsm allow=alaw The call does not come into the context "mycontext" but into the default context... How can I authenticate this call so that it does not go into the default context ??? Jonas. -------------- next part -------------- An HTML attachment was scrubbed... U...

...ess go to this endpoint. They want me to keep this endpoint, but add a new endpoint where we register with them. Existing... [transport1] type = transport bind = 0.0.0.0 protocol = udp [1002] type = aor remove_existing = yes contact = sip:1002 at xxx.xxx.xxx.xxx [1002] type = endpoint context = mycontext transport = transport1 accountcode = 6 dtmf_mode = inband device_state_busy_at = 48 force_rport = no identify_by = username from_user = 1002 disallow = all allow = ulaw acl = acl1 [identify112] type = identify endpoint = 1002 match = 1002 at xxx.xxx.xxx.xxx I setup the registration and the endpoi...

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 already supplied it. Ideas? Thanks, Matthew

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 mycontext) to something different. How do I do that? Would it be enough to...

...tures that need to be solidified before threading? Should I be playing it safe and put a thread lock around the entire copy procedure? In case you are interested, here is the algorithm I'm using to copy a Module to a same (or different) LLVMContext: if (&context == &other.myContext) { // If the context is shared, we can use cloning ValueToValueMapTy map; myModule = CloneModule(other.myModule, map); } else { // Otherwise, round trip the module to a stream and then back // int...

Hello List, I work for an IP communication provider in upstate NY as the engineer assisting our technical support team. We provide a number of different Telco systems to residential subscribers; and in an effort to more effectively trouble shoot termination problems I came up with the idea of creating a click to call system that will allow our agents to effortlessly place test calls. On a

...d 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 Exten: createcall Context: MyContext Priority: 1 Timeout: 60000 CallerID: CustomerName <########## > Variable: CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=5,OriginateCallId=396 Async: true Is there a setting that's required on the PJSIP endpoint to allow overwriting the INVITE packet's Contact header? Is th...

...d be used with multiple LLVMContexts. Making this a class you need an instance of would likely also allow typing to be reduced a bit because said instance can be given a shorter name than 'ConstantBuilder'. For example: 'cb.GetStruct(...)' instead of 'ConstantBuilder::GetStruct(MyContext, ...)'.

On Wed, 22 Nov 2017, Dmitriy Serov wrote: > ?same => n,System(printf "Action: Originate\nActionID: 1\nChannel: Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" > > /var/spool/asterisk/outgoing/${number}-${confnum}) I get: Unknown keyword 'Action' at line 1 of /var/spool/asterisk/outgoing/... Unknown keyword 'ActionID' at line 2 of /var/spool/asterisk/outgoing/... These are 'AM...

Good day all I've tried my iax conf and I'm struggling.So I want to know If someone else got this working and if they can pleas send my their configs I have to asterisk server,in different tows,both offices connected wit a direct line so both servers are on the same network running SIP.Each town got different extension register to each sever.Town A=100+ town B=200+ How do I get town A