similar to: Dialing multiple endpoints and CallerID presentation

hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can

I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in

lvqcl wrote: > Oops. It seems that vsnprintf_s isn't always available on MinGW platform: > MinGW declares this function only if MINGW_HAS_SECURE_API macro is defined. > That's because WinXP version of msvcrt.dll doesn't contain secure functions > like vsnprintf_s. > > Maybe it's better to revert vsnprintf_s to vsprintf or to use vnsprintf? Ok, we need to drop

On Tue, 30 Jun 2015, Damien Miller wrote: | On Mon, 29 Jun 2015, Tim Rice wrote: | | > On Tue, 30 Jun 2015, Damien Miller wrote: | > | > | I think we should just disable the test if the host doesn't support IPv6. | > | | > | diff --git a/regress/cfgparse.sh b/regress/cfgparse.sh | > | index 7f377d8..e19b4d0 100644 | > | --- a/regress/cfgparse.sh | > | +++

Thanks for the mighty quick response, Joshua! I am using Zoiper on Linux softclient: REGISTER sip:<ipAddr>;transport=TCP SIP/2.0 Changed the port back to 5060. On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > > <snip> > > > *CLI> pjsip set logger on >> PJSIP Logging enabled >> [Feb 15

Hello, Groups in asterisk are summarized here ( http://www.voip-info.org/wiki/view/Channels+and+Groups). Is there any difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789 (as I've been advised in another thread, to switch from one notation to the other and I can't see the reason behind that) ? Regards -------------- next part -------------- An HTML attachment was scrubbed...

I R-users, I am trying to find the way to manipulate a character string to select a 4 digit number after some specific word/s. Example: mytext <- "I do not want the first number 1234, but the second number 5678" Is there any function that allows you to select a certain number of digits (in this case 5678) after a particular word/s (e.g., second number) Thank you for your help

Hi, as stated in the documentation, it's allowed to set FAXOPT(faxdetect)=yes/no to allow fax detection. It's done (see below) but still fax detection :-( Extension 300 is hylafax with iaxmodem. On the upper Asterisk gw it's the same, despite the faxdetect set to no we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile phone calling the 0123456789 PSTN number.

hello list, i use chanspy with the code below [app-chanspy] exten => _007.,1,Macro(user-callerid,) exten => _007.,n,Answer exten => _007.,n,Authenticate(1111) exten => _007.,n,ChanSpy(SIP/${EXTEN:3},dqs) exten => _007.,n,Hangup i have a question related to chanspy i have created extension from 100 to 300 and i will give the permission with group of extension i want to use

Hi all, I’m trying to rewrite Diversion header when call forwarding is done on the phone. The phone sends "302 Moved Temporarily" response and sets Diversion header to a local number, but before Asterisk sends this call towards TSP provider I need to change Diversion header to a full PSTN number. I am using PJSIP_HEADER in a pre-dial handler (configuration is below). On the same

I solved the issue by not answering the call as I assume others have done. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Wednesday, February 18, 2015 12:50 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Res_fax - FAXOPT(faxdetect) Hello Le

hi at all, i'm trying to migrate a vm between two host but fails, this is what I did: virsh # start win2008 Domain win2008 started virsh # list Id Name State ---------------------------------- 1 win2008 running virsh # migrate --live win2008 qemu+ssh://host2/system error: internal error Attempt to migrate guest to the same host

Hello, Did someone have a solution for a line fax detection for outgoing call For exemple I call number 0123456789 - if it is a fax then redirect to extension A - if it is a line then redirect to exention B whats ia want its somthing like AMD application that i use for the answering machine . http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD search in the wiki give this application :

Hi All, I am running openssh (OpenSSH_2.9p1, SSH protocols 1.5/2.0). The scenario is that I have got three machines (A, B and C). The sshd on host A is listensing on port 1234, and the sshd on host B is listensing on port 5678. How can I set up a scp from a third host C so as to copy a file from host A to host B? scp -P 1234 myname at A:/var/tmp/file1 -P 5678 myname at B:/var/tmp/file1 does

thank you so much it work you must add 1 like below [app-chanspy] exten => _0071XX,*1,*Macro(chanspy,1234) exten => _0072XX,*1,*Macro(chanspy,5678) exten => _0073XX,*1,*Macro(chanspy,8910) best regards. 2015-03-11 19:48 GMT+00:00 Carlos Chavez <cursor at telecomabmex.com>: > On 3/11/15 12:48 PM, Salaheddine Elharit wrote: > >> hello list, >> >> i use

Hi, I would like to originate my first call from CLI. As I'm new to this, I'm wondering if it's possible. When I type "originate" from CLI, I've got this : " There are two ways to use this command. A call can be originated between a channel and a specific application, or between a channel and an extension in the dialplan. This is similar to call files or the

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.

Hi, I am trying to dial a registered user via his IP:Port mechanism, but problem is that the audio data is not reaching to dialed user. here is the scenario. caller and callee both are registered at asterisk server. asterisk server is on public ip so no port forwarding and natting necessary there. however caller and callee both are behind router and there is port forwarding enabled and nat=yes,

> On Fri, 10 Jul 2020 at 09:11, Anton Korobeynikov via llvm-dev > <llvm-dev at lists.llvm.org> wrote: > > 3. Wipe existing issues and pull requests > Does this really wipes the "auto-increment" IDs used by PRs and issues > and starts from zero again? I will need to clarify whether we will be able to reset the counter or not > > 4. Migrate all issues from

Hello, I'm trying to get started with asterisk/SIP so I was trying the demo that is provided in the extensions config file, but the call isn't "answered" by my server when I try calling the number that I registered at my SIP provider. I've registered with register => John.Doe:MyPass:MyUser@my-sip-provider/1000 in sip.conf and if I use "sip debug" I can see the