similar to: Ignoring the called number in the INVITE message

Hello, I am trying to setup a2billing system for asterisk. I have installed it corectly, but I have not found any users manual. I do not understand the whole structure. How do the parts like calling cards and sip friends cooperate together? I simply need to know how to make a call through it. With all the features like CDR's, etc. Can anybody help me with this? Thanks in advance.

Hello, I have such a problem. I have an * configured as a peer connected to the gateway to PSTN. While calling to the switched off cell phone, the gateway sends to the * the SIP message 180 with the SDP part, and also a lot of rtp packets containing the operator's in band info. But * forwards the 180 to the UAC without the sdp part and also without the rtp stream. Is there any way, how

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 the INVITE as the extension in the dialplan. The INVITE R-URI looks like: INVITE sip:+19135041291;rn=+19136630000;npdi at 12.4.240.200:5060;user=phone;transport=udp SIP/2.0

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 >

Trouble getting chan_sip2 to compile below is what I have done -download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software cp /root/software/chan_sip2s.c /usr/src/asterisk/channels cd /usr/src/asterisk/ patch -p0 acl.c /root/software/acl.c.patch cd /usr/src/asterisk/include/asterisk patch -p0 acl.h /root/software/acl.h.patch - added the follow to /usr/src/asterisk/channels/Makefile

Hi, I would like to configure asterisk to allow INVITE for hold to pass through it and not provide music on hold by itself. Can anyone help me out here? Regards, Mayur -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080221/dee4c2d7/attachment.htm

I'm tying to use park and announce for call park on Asterisk 1.4.2 but I'm having trouble getting it to work properly. This use to work with an older version of Asterisk. A telephone on the PSTN calls an IP phone. The IP phone is assigned extension 3-8396. 3-8396 answers the call and attempts to perform a blind transfer to x700, the parking lot number. The transfer gets to Asterisk,

Dear list members, I need to see the content of the functions in package geepack. It uses some C functions in the computation they are not available in the package directory. How can I see the content of C functions? Kind Regards, Thank you, Aysun Cetinyurek

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

In the below code fragment, print(arg) always prints the last element of rekeningen$rekening. Is this because of lazy evaluation? I.e. arg is evaluated at the time the button is pressed? And, if so, how can I avoid this? I tried function() {force(arg); print(arg)} but that didn't work either. Thanks, Jeebee. for(rek in seq(1,nrow(rekeningen))) { arg <- rekeningen$rekening[rek]

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 remember seeing something somewhere that described how I could get SIP header information with Asterisk. It was a command or a variable. Anyone know what it is? Thanks. Doug.

Hi, I would like to take advantage of new features in Dovecot 2.2 on my servers. But I'm having difficulties to build packages for Ubuntu-12.04-LTS. Does anyone know a repository that has new stable versions for Dovecot and Pigeonhole ? Thanks ! --&nbsp; Thiago Henrique adminlinux.com.br

Hello when setting a local forward (in this case to extension 23) on a SIP phone, I see the following on the Asterisk CLI : [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from 11.22.33.44:40670 [Aug 31 14:59:34] -- Now forwarding Local/myaccount184 at CallFromQueue-000007f4;2 to 'Local/23 at from-internal' (thanks to SIP/myaccount184-00003729)

Dear all I have a dataset that looks like: fr.wt site 1 4400 glen 2 235 glen 3 225 glen ' ' ' ' ' ' ' ' ' 82 550 glen 83 550 kom 84 550 kom ' ' ' ' ' ' ' ' ' 191 820 kom 192 2000 soet ' ' ' ' ' ' I need to do a series of histograms for each of the codes, levels or factors in

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, 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

We're doing some SIP development and have a question on "additional parameters" supplied to the register (in this case maddr= and the non-standard clport= in our example below). What we're experiencing is the INVITE doesn't included these parameters and they get dropped when the INVITE is sent to the 10.1.1.97 address. Ideas? Supported? SIP Bug? REGISTER

Hello, I would like to verify if a specific SIP header exists, and if yes, extract the partial content from another header. 1. Is there a way to verify if a specific header exists? 2. How do I extract data that is between the first : and the following @? Specifically, The data looks like <sip:1234567890 at 10.0.0.1:5060> and I would like to get only the 1234567890 I tried to use the CUT()

Hi, I am trying to capture "Server" header in a 200 OK reply message. My idea was to use Dail(SIP/user at domain,30,M(GetOtherPartyInfo)), and inside of GetOtherPartyInfo macro use SIP_HEADER function. For example: [default] exten => _X.,1,Dial(SIP/user at domain,30,M(GetOtherPartyInfo)) exten => _X.,n,Hangup() [macro-GetOtherPartyInfo] exten => s,1,NoOp(SIP Server: