Re: [asterisk-users] Pass Sound files as Argument to Macro Asterisk 1.8

It should be. I'd write something like the below: [macro-test] exten => s,1,NoOp exten => s,n,GotoIf($[${STAT(e,/var/lib/asterisk/sounds/${ARG1}.ulaw)} = 0]?NOPROMPT:PLAYBACK) exten => s,n(NOPROMPT),Background(nothing-recorded&for&pm-prompt-number) exten => s,n,SayPhonetic(${ARG1}) exten => s,n,G

Re: [asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint

Thanks Mathew, That would be great - just to validate the status of the AOR before you send the INVITE. Great mailing list. On Tue, Mar 11, 2014 at 5:38 PM, Matthew Jordan wrote: > On Tue, Mar 11, 2014 at 8:45 AM, Yaron Nachum > wrote: > > > > Thanks for the response anyway. > > > > I think

Re: [asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833

Mathew, Thanks Mathew. It's good to know the limitations :-) Is there any plan to add it? On Tue, Mar 11, 2014 at 6:38 PM, Matthew Jordan wrote: > On Tue, Mar 11, 2014 at 11:23 AM, Yaron Nachum > wrote: > > Hi Mathew, > > The regular sip stack has 'auto' dtmfmode which behaved as I said - if

Re: [asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833

On Tue, Mar 11, 2014 at 11:23 AM, Yaron Nachum wrote: > Hi Mathew, > The regular sip stack has 'auto' dtmfmode which behaved as I said - if the > remote replied with telephony event it used RFC2833 otherwise it used > inband. > Correct. There is no setting for dtmf_mode that is analogous to the c

Re: [asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833

Hi Mathew, The regular sip stack has 'auto' dtmfmode which behaved as I said - if the remote replied with telephony event it used RFC2833 otherwise it used inband. On Tue, Mar 11, 2014 at 5:43 PM, Matthew Jordan wrote: > On Tue, Mar 11, 2014 at 8:23 AM, Yaron Nachum > wrote: > > Hello, > > I

Re: [asterisk-users] Asterisk Authentication

actually alwaysauthreject was the problem, making it yes was a solution. thanks everyone On Tue, Mar 11, 2014 at 8:17 PM, Eric Wieling wrote: > Try setting the sip.conf entry to friend, not peer and not user. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailt

Re: [asterisk-users] Oddity with FFA

Steve, I appreciate you elaborating on my problem. I don't suppose this is as easy as putting a wait(3) in my dial plan before hangup.? (Didn't think so.) Aside from checking (and hoping) for a newer version of FFA that fixes this issue, I guess there's not much I can do, then. Thanks agai

Re: [asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833

On Tue, Mar 11, 2014 at 8:23 AM, Yaron Nachum wrote: > Hello, > I have installed the latest version 12 that has been released (12.1.0.rc3). > > I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint > that doesn't support it (no telephony event in the rtpmap) the asterisk > r

Re: [asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint

On Tue, Mar 11, 2014 at 8:45 AM, Yaron Nachum wrote: > > Thanks for the response anyway. > > I think that it would be great if someone would make it happen. It seems to > me trivial that once you enable to setup multiple AORs you would use them :-) > > Yaron. > > > On Tue, Mar 11, 2014 at 3:38 PM

Re: [asterisk-users] Asterisk Authentication

Try setting the sip.conf entry to friend, not peer and not user. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Boykin Sent: Tuesday, March 11, 2014 10:34 AM To: Asterisk Users Mailing List - Non-Commercia

Re: [asterisk-users] Asterisk Authentication

I am not sure if you understood the problem. Asterisk introduced "match_auth_username" option for what I exactly want but it doesn't seem to be work. On Tue, Mar 11, 2014 at 5:23 PM, Ishfaq Malik wrote: > > > On 11 March 2014 11:39, Jim Boykin wrote: > >> Hi, >> >> I am trying to setup aster

Re: [asterisk-users] Linux call router

On 11-03-14 12:15, binary dreamer wrote: hello there, I am facing an issue with misd/misdnuser/lcr in the system I am running debian 7 and I managed to install from git misdn/misdnuser but in lcr I am getting: chan_lcr.c: In function 'load_module': chan_lcr.c:3520:24: warning: assignment makes po

Re: [asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint

Thanks for the response anyway. I think that it would be great if someone would make it happen. It seems to me trivial that once you enable to setup multiple AORs you would use them :-) Yaron. On Tue, Mar 11, 2014 at 3:38 PM, Joshua Colp wrote: > Yaron Nachum wrote: > >> Thanks Joshua, >> I t

Re: [asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint

Yaron Nachum wrote: Thanks Joshua, I tried it already. That would generate a call to both AORs which is not what I was looking for. Isn't there a way to retrieve the AOR status from the dialplan? Not currently. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - H

Re: [asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint

Thanks Joshua, I tried it already. That would generate a call to both AORs which is not what I was looking for. Isn't there a way to retrieve the AOR status from the dialplan? On Tue, Mar 11, 2014 at 3:27 PM, Joshua Colp wrote: > Yaron Nachum wrote: > >> Hello everyone, >> I have started testi

Re: [asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint

Yaron Nachum wrote: Hello everyone, I have started testing the PJSIP stack. I saw that it is possible to setup statically multiple AOR contacts, setup qualify_timeout and attach it to an endpoint, and then dial using this endpoint. When I setup the configuration I used the cli in order to see t

[asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint

Hello everyone, I have started testing the PJSIP stack. I saw that it is possible to setup statically multiple AOR contacts, setup qualify_timeout and attach it to an endpoint, and then dial using this endpoint. When I setup the configuration I used the cli in order to see the status of the conta

[asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833

Hello, I have installed the latest version 12 that has been released (12.1.0.rc3). I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesn't support it (no telephony event in the rtpmap) the asterisk responds OK in the signalling but DTMF is not working. Is it a kn

Re: [asterisk-users] Asterisk Authentication

On 11 March 2014 11:39, Jim Boykin wrote: > Hi, > > I am trying to setup asterisk so that anyone from any IP can call using > any callerid as long they have an account - also no registration is > required. > > However, it seems like asterisk tries to find peer based on either the IP > address or

[asterisk-users] Asterisk Authentication

Hi, I am trying to setup asterisk so that anyone from any IP can call using any callerid as long they have an account - also no registration is required. However, it seems like asterisk tries to find peer based on either the IP address or from header. What I really want is asterisk to find acco

[asterisk-users] Linux call router

hello there, I am facing an issue with misd/misdnuser/lcr in the system I am running debian 7 and I managed to install from git misdn/misdnuser but in lcr I am getting: chan_lcr.c: In function 'load_module': chan_lcr.c:3520:24: warning: assignment makes pointer from integer without a cast [enabl