similar to: Losing CALLERID{dnid}

I run CentOS at home, not just at work... Anyway, I've got a friend in Chicago who recently mentioned that they'd like to do videocalling. Now, I've heard of skype, but a quick google says there's some problems on Linux. I also see ekiga, and aMSN. Anyone here run such a beast, and have any recommendations or comments? Obviously, must work on CentOS, not Ubuntu, or Fedora, or

Hi Srs., Do you know if it's possible make a videocall from asterisk to UMTS mobile phone?. Both technologies use H.263 like videocodec. Any idea? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050930/ea37cdf9/attachment.htm

I am trying to set add a SIP Header to a call before adding it to the Queue. The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs

Hi , i worked with h324m gateway for 3g video calling .It? configured successfully . my code in extensions.conf is [from-zaptel] exten => _X.,1,h324m_gw(0 at mainmenu) exten=>_X.,n,Hangup [mainmenu] exten => 0,1,h324m_gw_answer() exten => 0,2,mp4play(/tmp/menu/menu.mp4,'n(1)') when i make a video call (either sip or through pri) , asterisk cli shows the following error --

Hi, We encounter a problem with the variable CALLERID(dnid) We use E1 lines where we can make an inbound call or an outbound call on the same channel (not at the same time) If the CALLERID(dnid) is not used, than the CALLERID(dnid) will be the CALLERID(dnid) of the previous call For example: - First we get a inbound call on channel DAHDI/11-1 with CALLERID(dnid) = '655871460' We read

Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system to work with *? I'm not getting inbound or outbound calls to work. (Inbound caller gets line busy tone.) SETTINGS FROM MITEL: I built a Crossover cable and connected it like this: PSTN--T1--ASTRISK--T1--OLD MITEL -Crossover Cable Pin-out: 1 - 4 2 - 5

I have both the PJSIP add and the chan_sip way of adding SIP headers in there. The Verbose is showing the variable value is there. The INVITE to PJSIP/Agent1 does not include either X-My-DNID or X-My-DNID2 headers. exten => 1234,1,Verbose(X-My-DNID:${MY_DNID}) same => n,Set(X-My-DNID=${MY_DNID}) same => n,Set(PJSIP_HEADER(add,X-My-DNID2)=${MY_DNID}) same => n,Dial(PJSIP/Agent1)

Hello, I've been looking at some SIP packet dumps captured with tcpdump on the PBX itself, and analyzed with Wireshark 0.99.4. I'm noticing something strange, at least to me. All of the SIP packets going out from our Asterisk PBX to either of our 2 VoIP providers are consistently 50% out of order. In addition, if I use Wireshark's voip call player, the outgoing side of the call

Hi, In my application I jump to different extensions For example: [begin] exten => s,1,Goto(starts,s,1) [start] exten => s,1,Play(welkom) ..... exten => h,1,Goto(end,s,1) [end] exten => s,1,Macro(end_call) exten => s,n, Hangup When I look at my CDR record I see three different CDR's in my record. Is there a way to use one CDR on every call, and not

Thanks Scott. I was able to get the basic concept to run. However, it seems PJSIP INVITE for the Dial also does not support added headers. The Local channel dial plan did have the channel variable values. I added them as SIP headers, then Dial(PJSIP/Agent). The INVITE for the Dial on PJSIP continues to not include the SIP Headers I added. For chan_sip, I have no problem with this. Even the

hello everyone, I have a trixbox server with an E1 card(not digium).It connects to an AVAYA pbx use E1. It works fine.But when i change the E1 card to digium TE220B,there is a problem. When sip extension A(on trixbox) call PSTN extension B(on avaya),A must wait longtime before B start to ring.From the log I find there are two times call. I don't know why the first request be rejected

Hi, Does anybody know if RAMI (Ruby Ami) is still functional? And is this still compatible with asterisk 1.8 Best Regards, Arjan Kroon Mobillion BV

Hello, I'm using X-web lite in a webpage to connect to one of our asterisk server. But now I have a problem, when you are connected to a voice script the voice will not be heard after a couple of seconds. When you press or say something that the voice will come back for a couple of seconds. When I thy X-Lite (stand-alone version) I had the same problem, but when I turned off the

Hi, I have an Asterisk 1.2.9.1 with chan_misdn 0.3.1-rc23 and a beronet octoBRI on a Debian box I have to set up instead of an old legacy PBX. My problem is I get only the base DNID and not the extensions (the last two digits) in Asterisk but the old PBX got all the DNID number so I think it is the card. Is there anybody experiencing a problem like this? How can I solve it? Any ideas? TIA

Hi, Does anyone has a *working* patch to set the dnid from extension when it is now supplied, cos this plugin doesn't working without it. I tried and it passes empty string. THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/aae0a7f4/attachment.htm

I haven't been able to find much support for the RDNIS or DNID variables online. I am trying to prove a concept of call routing before we move towards development of a production system. I need to have calls routed coming into a call center based on DNIS. What type of syntax is needed in the extensions.conf file and how can I test it with a softphone (ie: can I emulate the DNIS from xlite)?

hello i am using ast-rad-acc.pl from portaone connected with asterisk manager. my (%cdr) = @_; $cdr{'CALLERID'}, $cdr{'DNID'}, these are empty why these two variables are not working on hangup any comments thanks Kamran Ahamd __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around

Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 11111 -> ext. 1 913 - 22222 -> ext. 2 913-11111 & 913-22222 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the

Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 11111 -> ext. 1 913 - 22222 -> ext. 2 913-11111 & 913-22222 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the

Hi, I installed asterisk 1.2.1 on a debian distro with 2 tdm400p cards. I'm receiving the DNID instead of callerid...this is very strange. I found many errors and warnings inside asterisk log like these: Feb 28 15:28:12 ERROR[18218] callerid.c: fsk_serie made mylen < 0 (-14) Feb 28 15:28:12 WARNING[18218] chan_zap.c: CallerID feed failed: Success Feb 28 15:28:12 WARNING[18218] chan_zap.c: