similar to: Call token is ip$localhost

Hello, I am looking for information on setting up digium FXO card for use as a PSTN Gateway (H323-PSTN) to work with GNUGk. I am basically looking for the setup and it would be great if anyone can share his experiences with the same. Also, if there are any limitations in going for such a setup and problems that may arise/things that I should keep in consideration. Thanks & Regards, Deepak

I tried to use it to create a 'trunk' to Cisco's call manager. The 0.7.1 code worked up to a point. The call would be established, but audio was one-way from the Call Manager. Asterisk with Chan_h323 would not setup the sending rtp stream. The debug results showed the sending stream as using ip:0.0.0.0 I have not checked for a CVS update to see if it is fixed, or if that

Hello ! I am experiencing some problems with Asterisk trunking, this is the scenario: There are 3 servers, a DID server provider (VOIP provider) which delegates us a bunch of DID numbers to our asterisk server number one (I will call it AA), from which I route the calls to Asterisk server number 2 (I will call it BB), which then terminate on phone handsets. The trouble is, that I probably

Hello Nhadie, I had a very similar situation. My solution, even tough might not look very wise, solved my problem the way I needed. I repeated the GotoIftime command in the next line in my extensions.conf . Like this: GotoIfTime(22:00-23:59|*|30|jul?test,s,1) GotoIfTime(00:00-02:00|*|31|jul?test,s,1) Rgs, Marco Cordeiro -----Mensagem original----- De: asterisk-users-bounces at

Hello! Tried to compile, but : [CC] chan_h323.c -> chan_h323.o chan_h323.c: In function '__oh323_update_info': chan_h323.c:349: error: dereferencing pointer to incomplete type chan_h323.c:350: error: dereferencing pointer to incomplete type chan_h323.c: In function 'oh323_rtp_read': chan_h323.c:790: error: dereferencing pointer to incomplete type chan_h323.c:791: error:

Hello all, I have problem. I can't makel asterisk addon, asterisk-ooh323. I use Asterisk and addons svn version. OS:redhat EL4 Linux 2.6.9-5.EL #1 Wed Jan 5 19:22:18 EST 2005 i686 i686 i386 GNU/Linux Please help me . [root@asterisk asterisk-ooh323c]# make make all-am make[1]: Entering directory `/usr/local/src/asterisk-addons/asterisk-ooh323c' source='src/chan_h323.c'

If I have different opening hours on different days, can I accomodate that in a single gotoiftime, or will I need to filter them out one by one ? For example, our hours are Mon-Fri 9:00-17:00 and Sat 09:00-13:00 can this be done something like GotoIfTime([9:00-17:00|mon-fri][9:00-13:00|sat]|*|*?open,s,1) or something like that, or do I have to do:

Running a fairly recent subversion release of Asterisk, I'm running into a problem using labels (as opposed to priorities) with this application. Here is the dialplan segment: ; isolate gotoiftime bug with labels ;exten => 806,1,GotoIfTime(8:00-20:00,*,*,*?2:4) exten => 806,1,GotoIfTime(8:00-20:00,*,*,*?light:dark) exten => 806,n(light),noop(light) exten => 806,n,hangup exten

In my * box I've configured two queues and incoming number and whenever any one calls those number call comes to my *box and it sends call to my agents in queue. but if no agent is available it still answer the call. Is there any why when my agents are not available I don't want call to get answered. Here is my dialplan: exten => xxxx,1,GotoIfTime(*|*|20|dec?ccagents,xxxx,6) exten

Heya, I changed the Makefile of the h323-channel-makefile (I downloaded cvs of a couple of hours ago) so that I don't have to rebuild pwlib and openh323, but can use the precompiled versions. I'm using pwlib 1.8.3 and openh323 1.15.2. There aren't many changes. I replaced OPENH323DIR with OPENH323INC ,which points to /usr/include/openh323 for me and OPENH323LIB, which points to

Hi, all! when my h323 phone dial in Asterisk system, i can hear nothing. and the following is the log slice i picked from /var/log/asterisk/full. ps: i am using red hat AS5 kernel 2.6.18-53.el5,Asterisk-1.4.24.1, pwlib_v1_11_0, openh323_v1_19_0_1. Best Regards! 81948 [May 6 10:07:34] VERBOSE[11579] logger.c: -- Remote UNIX connection 81949 [May 6 10:07:51] VERBOSE[29627] logger.c:

Does anyone know if gotoiftime can take any subset of 7 days for the days of the week or only a contiguous range? I want to use gotoiftime to change dialplan behavior on Monday, wedneday and Friday -- Executing GotoIfTime("Zap/8-1", "09:00-20:00|MON WED FRI|?21") in new stack Aug 27 19:27:25 WARNING[2676]: pbx.c:3729 get_dow: Invalid day 'MON WED FRI',

Hello, I have a strange h323 issue. After executing command "Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18 22:32:23. Meanwile I have sniffing traffic on port 1720. The call was established just at Oct 18 22:33:03 (New H.323 Connection created.) and also packet sniffer grabs the h323 invites at this time also. So my question is what

Hello all, I got the following error compiling h323 support in the latest cvs. Below the error is a diff to the file that I got to make it work. I took an example out of sip as far as the syntax for ast_rtp_new. Not sure if it is correct or not, but it seems to work. Please correct me if I am wrong in the additional 2 arguements. Regards, Scott cc -g -pg -c -o chan_h323.o -march=i686

I am trying to use the Gotoiftime CMD to not allow calls to be placed between the hours of 12am-5am, except if you know the PIN number to dial out and if the call is for 911. What is the best way to implement this solutions? I have the gotoiftime like so: exten => s,1,GotoIfTime(5:00-11:59|mon-fri|*|*?custom-blacklist,s,1) and using Read for the PIN like so: exten => s,3,Read(Secret,,3)

Hello list, why is it that GoToIfTime thinks a date of **|*|29-*|jun *is not valid ?? [Jun 29 14:06:34] -- Executing [s at macro-vac:10] *GotoIfTime*("SIP/testcorp-00000036", "**|*|29-*|jun*?onvac") in new stack [Jun 29 14:06:34] WARNING[3076]: pbx.c:4127 get_range: Invalid end day '*', assuming none [Jun 29 14:06:34] -- Executing [s at macro-vac:11]

I am trying to use the GotoIfTime function and get a busy signal. What I am trying to accomplish is to have the system tell callers that we are closed after 5:00pm. Here is the code below. ; If we're open, then go to the open context ; We're open from 9am to 6pm Monday through Friday exten => 3200,1,GotoIfTime(09:00-17:59,mon-fri,*,*?open,3200,1) ; ; We're also late on Tuesday and

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

Hi, I've an PRI coming to my asterisk ,calls are coming fine and my agents are able to answer no prob. but I've an agreement with my telco with some incoming no if the no of calls on these no are more then 3 then send to another no. they use busy signal to divert call on another number so I'm sending the call to Congestion() if no of calls in this group are more then 3. But my

I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version 1.400 and I am simply trying to configure into the "Extensions.conf" script an entry that will add to the "Auto-Attendant" a line that will allow a "Caller" to enter a "0" (Zero) will then ring the extension(s) of the "Operator" to speak directly with the "OPERATOR"