similar to: Capture "Server" header in SIP reply.

Hi, Just noticed Asterisk is not playing 'ring' tone as defined in indications.conf when Dial command is used with 'r' option. For example: [test] exten => 123,1,PlayTones(ring) exten => 123,n,Wait(5) exten => 123,n,Playback(demo-congrats) exten => 123,n,Hangup() exten => 321,1,Dail(LOCAL/123 at test/n,60,r) When I now dial with a SIP phone - 123 I can hear nice

Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: ---- sip.conf: ---- [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk

Hi, I am trying to use ConfBridge application, but it throws "Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw)" error. Please see console output below. -- Executing [501 at services:9] ConfBridge("SIP/OpenSER-00000005", "1001") in new stack [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:404 join_conference_bridge: Trying to find conference

Hi, I am trying to send "404 Not found" reply, without any luck with the following: exten => 555,1,Playback(you-dialed-wrong-number,noanswer) exten => 555,n,Playback(check-number-dial-again,noanswer) exten => 555,n,Congestion() However the above results in "500 Service Unavailable" being send out. What would be the correct application/function to generate "404

Hi, Is there any way to tell Asterisk not to generate additional headers like: X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 I can't find any relevant option in sip.conf file :-( Thanks for help. Chris

Hello, I'm trying to use SIP_HEADER function on my dialplan but I receive this message (on the console): pbx.c:3367 ast_func_read: Function SIP_Header not registered Why? Thank's - Bakko

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Ext No Problems Panasonic Ext -> SIP Ext No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1] Dial("SIP/610-097aee60",

Hi * users, I would like to extract the information present in the SIP REFER message that comes to asterisk. Would SIP_HEADER() allow me to do that ? I have used SIP_HEADER() for extracting the to and from SIP headers previously. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

Asterisk 1.4.12 I wish to extract some custom headers from a SIP REFER message but am unable to do so. However I can extract them from an INVITE. The code is: exten => _.,n,Set(custom-id=${SIP_HEADER(custom-id)}) ; exten => _.,n,Set(custom-valid=${SIP_HEADER(custom-valid)}) ; Examples of the INVITE (works) and REFER (doesn't) messages are below. U 147.202.001.001:5060 ->

Hi, I just started with setting up a new asterisk system, that will operate on a sip trunk, but I wonder, how to transfer the calls to different extensions, because all calls appear as being send to the base number of the trunk. E.g. given the trunk range of 1234567800-12345678099, a call to 1234567800 is matched by the same pattern as a call to 12345678099. ; matches 12345678099, too exten

Hi all, is there anyway i can set SIP_HEADER(To) to the value i like? -- Regards Rizwan Hisham Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070409/528077f9/attachment.htm

I'm using "s" extension in my dialplan: [gv-voice] exten => s,1,Verbose(callerid is "${CALLERID(all)}" or "${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)}) ; PJSIP_HEADER(read,To) same=>n,.... But when a call comes in to the gv-voice context, "s" doesn't match the extension: res_pjsip_session.c:2991 new_invite: Call from

On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <admin at tootai.net> wrote: > Le 18/03/2016 16:20, Trey Hilyard a ?crit : > > I am trying to set up my Asterisk server so that it will recognize an > > incoming call to the Asterisk's own Location Routing Number (LRN), > > validating the "rn" in the INVITE and then using the Called Number from >

Hi, Just to be sure: Is there a dialplan function in Asterisk that parses custom "name-addr"-style SIP headers for me? If I wanted to do it right the syntax name-addr *(SEMI generic-param) is quite complex to parse in the dialplan using nothing but CUT(). It's so easy to make false assumtions about angle brackets (< >), whitespace (LWS), quotes (") around the

Readin : https://wiki.samba.org/index.php/Samba_AD_schema_extensions Is it an option to make an ldiff for the msNPAllowDialin and others on that Dail-in Tab. Im looking at the automount example. Hints tips? Greetz, Louis > -----Oorspronkelijk bericht----- > Van: samba [mailto:samba-bounces at lists.samba.org] Namens > L.P.H. van Belle via samba > Verzonden: donderdag 14

I am not proxying the media, but never the less I am forced to store the source media IP in my CDR, for regulatory reasons. Asterisk gets that information when the reinvite comes, but how do I store it? If I don't figure this out my next email will be from Federal Prison. Kindly help me stay away from those guys. Eventually we all need to save that information or we shall not be able to stay

I have a home network with my dial up connection on the windows 98 machine. Since I have installed SAMBA and configured the computers to see each other, I can no longer pull up a website on the windows 98 machine. The problem is that I have assigned a permanent IP address for the Intranet. I have to use DHCP for the dail up connection. Therefore the IP address that I assign to Windows machine

Hi Dmitriy and others and thanks for your help so far. The option "match_auth_username=yes" seems to have had no effect. From my reading, this option will try to match the username of the incoming SIP account to a section heading. If that is how it must work then i can see a big problem. I'm trying to present the receptionist with a nice display of which line the call came in on.

Quick question for those who know the innards of chan_sip: Does chan_sip use the "To:" header of an incoming INVITE request, for anything other than setting SIP_HEADER(TO) ? As far as I can tell so far, the target extension is taken from the request URI, i.e. sip:extension at domain, and the target context is taken from the section in sip.conf that matches the request's source IP

hi is it possible to set up in the dialplan (on in sip.conf, or something else) the hostname of the outgoing uri call? This is my scenario: - CCM integrated with Asterisk via h323 - SIP user registerd to Asterisk - Asterisk is behind NAT - Asterisk ip is 10.10.10.2 - SIP user view Asterisk as 10.10.15.2 (Asterisk is behind NAT) When the CCM calls the SIP user the call works perfectly. The