similar to: Asterisk SIP Trunk between two Asterisk Servers

Hi all, I need to test the following scenario: +-----------+ +-----------+ | asterisk 1| | asterisk 2| +-----------+ +-----------+ | | | | _______|__________________|___________ | | | | | | +-------+ +-------+ | ATA 1 |

Hi all, new to the list and this is probably a basic question and couldn't find anything clear googling around but I don't know how to handle calls to sip extensions not defined on sip.conf while using pattern matching. On my example I have sip extensions 10, 11, 12, and 13 on sip.conf. On a basic extension.conf I set up a pattern starting with "1" and a second digit should dial

I need help translating extensions.conf to AEL: [default] exten => _X.,1,Set(DID=${EXTEN:6}) exten => _X.,n,Goto(continue,1) exten => _1X.,1,Set(DID=${EXTEN:7}) exten => _1X.,n,Goto(continue,1) exten => continue,1,Noop(${DID}) exten => continue,n,Set(GROUP(IAX)=incoming) exten => continue,n,GotoIf($[${MATH(${GROUP_COUNT(incoming at IAX)}+${GROUP_COUNT(outgoing at

I've been banging my head on this for several hours, and I have no idea what's going on. Maybe there is a very simple result, and I've been looking too hard at this this evening. This is a brand new system, and I'm wondering if there have been SIP bugs introduced in the latest CVS that are preventing from working what should be a stupendously simple test. - Cisco 7960

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Le 18/02/2016 11:03, Richard Mudgett a ?crit : > I've been using Grandstream phones for more than 10 years, but onl y > yesterday tried to use Early Dial... and I failed. What is needed on the > Asterisk side to reply 484 to INVITE? Phones are talking to chan_p jsip > on Asterisk-13.7.1. > > > Look into the

I have to design a dialplan for mulitple contexts (multiple companies) and I'm not sure how to go about it and I thought someone may offer help. Here is some background. There are three separate companies, let's say A, B and C. Each has their own context and each has their own set of numbers (these are just examples, not the actual config): [ContextA] exten =>

First, here's what I want to do / what I have: X100P and a Quicknet PhoneJack. I want to be able to pick up the analog phone (connected to the phonejack) and dial another computer (with the same hardware) or just make a regular phone call which will be decided by asterisk depending on the phone number dialed. I know that this won't be taking full advantage of asterisk, but I'm

Hi All! I have been a happy and contented user of samba for several years now, when I got asked several questions I didn't know the answer to. The questions were: 1) Is there a way to create a network appliance type unix box, where usernames and groupnames are pulled from an existing NT4 or W2K domain? (without maintaining a local /etc/passwd file) What is the best way to do this?

Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy, then caller hears busy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Bryant, Thanks for your reply. It didn't work immediately, I had to create a second context, or else it was looping between the second and first line. This seems to work: [earlydial] ; Test Early Dial exten => _.,1,Set(l_Extension=${EXTEN}) exten => _.,n,Goto(earlydial2,${l_Extension},1) [earlydial2] exten => _.,n,Goto(noMatch,1)

Hi, I'm currently using Asterisk 1.4.23.1, and I have a problem (also on previous version). Sometimes, when I try to do an attended transfer to another internal with default feature *2, Asterisk doesn't make it (it doesn't play 'pbx-transfer'). Sometimes on second time, Asterisk make transfer correctly. I have this problem on variuos type of SIP phones (GrandStream, Aastra,

Dear, the goto function to the iax dialing, makes bill duration and call duration wrong, in cdr.they are equal to ringing time. the cdr will be produced and saved into the dbase, when the callee picks up the phone. is any way to have real duration time ? [main] exten => _1X.,1,GOTO(LOPL,${EXTEN},1) .... [LOPL] exten => _X.,1,Dial(IAX2/MAIN/${EXTEN},60)

Scenario: Both boxes are behind firewall, port udp 4569 is open. If I don't want the username and password in dialing string do I have to use register statement in IAX.CONF. Can anybody post some working samples; I have a hard time making it to work with the samples posted on wiki. -- #Joseph

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I've been using Grandstream phones for more than 10 years, but only yesterday tried to use Early Dial... and I failed. What is needed on the Asterisk side to reply 484 to INVITE? Phones are talking to chan_pjsip on Asterisk-13.7.1. Thanks, - -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran?aise

Hi all, I'm trying to setup a simple IVR menu in a context in extensions.conf. So far, I have: extension s for playing back the menu # to repeat it * for directory 0 for operator 1 which goes to another context: exten => 1,1,GoTo(option_1,s,1) Here is what I have in extensions.conf: [incoming] ; main greeintg exten => s, 1, Ringing exten => s, 2, Wait(10) exten => s, 3, NoOp()

Can multiple FWD accounts be registered? I have the following output in my sip.conf file: register=74928:xxx@fwd.pulver.com/74928 register=75160:xxx@fwd.pulver.com/75160 register=74573:xxx@fwd.pulver.com/74573 [fwd-74928] type=friend secret=xxx username=74928 host=fwd.pulver.com [fwd-75160] type=friend secret=xxx username=75160 host=fwd.pulver.com [fwd-74573] type=friend secret=xxx

http://bugs.freedesktop.org/show_bug.cgi?id=16266 Summary: Youtube video doesn't seek. Product: swfdec Version: git Platform: Other URL: http://youtube.com/watch?v=uz93LVg6oXk OS/Version: All Status: NEW Severity: enhancement Priority: medium Component: library AssignedTo: swfdec

This is an automated email from the git hooks/post-receive script. arrfab pushed a change to branch main in repository centos/centos.org. from 92ccbf4 Update distro-faq.md new 8542f8a Removing hostdime from active sponsors new 411cbb1 Removing estruxture from active sponsors The 2 revisions listed above as "new" are entirely new to this repository and will be

i finally got web services to work between two apps that i am currently working on. the application that serves the API stores the user id in a session when the user logs in. but when i use the client application and then go back to the server app, my session is erased and i have a message something like "Couldn''t find User without an ID " any ideas why this is happening?

This is an automated email from the git hooks/post-receive script. arrfab pushed a commit to branch main in repository centos/centos.org. commit 411cbb1912b8a61c406f2a705a2c3d77a21933e9 Author: Fabian Arrotin <arrfab at centos.org> AuthorDate: Tue Jul 11 08:49:21 2023 +0200 Removing estruxture from active sponsors Signed-off-by: Fabian Arrotin <arrfab at centos.org> ---