similar to: sip_chan - how to use value of the SIP 'To:' header field for extension logic

Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? None of the below Dial() command give the correct result : exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz) exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user762 at

Hi, I was trying to compiles addons 1.4 and res_config_mysql doesn't compile. is res_config_mysql still supported and is it still posible to use mysql with asterisk RealTime?? Bests Tomasz

I can smile now. I made my * work with my Cisco. Finally. First problem was Ethernet on my Linux. After installing * on a different machine, I got rid of that "icmp udp unreachable" error. My next problem was the call stays on on Cisco gateway, but the ATA drops it. I figured out it was my mistake on dialplan in extensions.conf --- (it took me a day to notice it.. damn!). my config

Hello a call goes out and is answered : [Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b is making progress passing it to SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b answered SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel SIP/myprovider-0000010b joined

Hi, Is it possible to filter the calling user with the usage of mysql realtime the same as it is done in extensions.conf file: exten => some_exten/calling-user.... is there some flag which activates this extra check?? Cheers Tomasz

Hi, I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is "cut". Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? my extension.conf for meetme: ;switch =>

In the following example exten => _1NXXNXXXXXX,1,Set(GROUP(outbound)=myprovider) exten => _1NXXNXXXXXX,n,Set(COUNT=${GROUP_COUNT(myprovider at outbound)}) exten => _1NXXNXXXXXX,n,NoOp(There are ${COUNT} calls for myprovider) exten => _1NXXNXXXXXX,n,GotoIf($[ ${COUNT} > 2 ]?denied : continue) exten => _1NXXNXXXXXX,n(denied),NoOp(There are too many calls up) exten =>

Le 15/01/2020 à 19:50, C.Maj a écrit : > On 2020-01-15 11:24, Administrator wrote: > > 8<'s > >> One of the provider took a pcap and told us that expiration was set to 0 >> that's why they don't accept the registration. We took a pcap on our >> side when SIP packet goes out of our server and we see that the >> expiration parameter is setted to

Hi, If I have [myprovider] section with context=something. When I do an outgoing call by using Dial(SIP/myprovider/464646)", does context=... affect anything? As I understand it, it only affects incoming calls, but I might be wrong. Another thing, the setting of context=... on [default] section will affect all [provider] sections without context=..., right? What if I don't specify any

Le 19/01/2020 à 00:31, Joshua C. Colp a écrit : > On Sat, Jan 18, 2020 at 1:14 PM Administrator <admin at tootai.net > <mailto:admin at tootai.net>> wrote: > > > 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

hi to all and i hope, someone can help me. i did not found an solution, here is my problem/issue. i setup samba 4.9.5+dfsg-5+deb10u1 (debian10) with vfs recycle. now: - i create an file (maybe simple txt), edit, save -> ok - i open the file again an save -> copy of the file goes to recycle repository - if i use recycle:versions = Yes, i got Copy # of ... everytime i save the file -

Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is dialled I'd like processing to return to the attendant so the user can place a subsequent call. I have setup features.conf to include: [featuremap] disconnect => ** My Dial command looks like this:

Hi all, I have a small system of two hardware boxes (residential gateways) running Linux with Asterisk on them. Each RG has some FXS ports to which analog telephones can be connected. I already had a working system including an external SIP provider, where both RGs would register to that provider with a telephone number and they could call each other via that telephone number. Each RG had a line

p 176 has exten => 1NXXNXXXXXXX,1,Dial(SIP/${EXTEN}@myprovider) how is "exten" distinct from "EXTEN"? What is this line of code doing? https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables says that EXTEN is the current extension. In ruby, you this: H = Hash["a" => 100, "b" => 200] The => is a mapping, or at least

> On Mon, 6 Feb 2017, Tech Support wrote: > > We were able to develop a feature to send the call to voicemail about 90% of the time. That way, an end user could (1) not be bothered by having to answer the call, (2) > delete the message without listening to it, or (3) listen to the message when it was most convenient for them. That way, they were in control and things were

? Hi, I am working on a small inbound call center solution that uses an ACD system. I might add an IVR system later on. I only have 2 extensions set up (extensions 1000 and 1001), I want the system to put new calls in a queue if both extensions are busy. I am currently subscribed with a SIP trunk provider and can successfully recieve calls. I want?to design a system where customers?can call my

Hi, using EAGI variables like agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode I get a lot of data about a call, but I need to obtain P-Asserted-Identity value from a SIP call. Are tehe any eagi variable to get that? Or have you any solution?? Thanks!!! -------------- next part

Hi all, I'm implementing call files and everything works nicely except that the variable that I set in the call file does not seem to get populated. Channel: SIP/MyProvider/9105555555 MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: myCallFileContext Extension: s Priority: 1 Set: myVar=MyNewValue and... [myCallFileContext] exten=>s,1,NoOp(${myVar}) ; <====== myVar is empty

After so many rings when the party does not answer, my SIP phone says Call Failed. Why doesn't it just keep ringing? Here's the dial plan rule: exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sip.myprovider.com,,r) exten => _NXXXXXXXXX,n,Hangup()

Hi, I'm new to Asterisk and I've been trying to configure it to talk to several SIP providers (such as FWD). I found that, although there are some "recipes" on how to do it, there are few documents that really explain *why* the settings are used, and overall I found very little documentation on sip.conf. I've been using this page as a reference: