search for: xlite1

hello, Im tryin to make Calls from MS Netmeeting(h323) to Xlite(SIP) it rings, but as soon as i answered it dissconnects!!!! This is what i get from the Asterisk console: -- Executing Dial("OH323/R27469", "SIP/xlite1|10") in new stack Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1265 create_addr: Setting NAT on RTP to 0 Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1500 sip_call: Outgoing Call for xlite1 Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1633 update_user_counter: Call from user 'xlite1' is 1 ou...

Hello, These are error messages I get when I try to call a number over CAPI channel. -- Executing SetCallerID("SIP/xlite1-3b80", "0") in new stack -- Executing Dial("SIP/xlite1-3b80", "CAPI/hfcpci/b17") in new stack > data = hfcpci/b17 > capi request for interface 'hfcpci' == hfcpci: Call CAPI/hfcpci/b17-1 (pres=0x00, ton=0x00) -- hfcpci: recei...

...l 500) and I just get the following messages. I am behind a NAT server and did NOT change anything in any of the sample config files from CVS. Could this be the problem? BTW - I'm using the Xlite soft phone running on the same box as the asterisk server. -- Executing Playback("SIP/xlite1-625c", "demo-abouttotry") in new stack -- Playing 'demo-abouttotry' (language 'en') -- Executing Dial("SIP/xlite1-625c", "IAX2/guest@misery.digium.com/s@default") in new stack -- Called guest@misery.digium.com/s@default -- IAX2/...

...on me and point me in the right direction. Needless to say, Asterisk pukes on my dialplan when I try and startup. . (zapata.conf) context=analog signalling=fxs_ks language=en channel => 1 (sip.conf) [sip_proxy] For incoming calls only. Example: FWD (Free World Dialup) type=user context=sip [xlite1] "Transmit Silence"=YES type=friend regexten=1234 ; When they register, create extension 1234 username=xlite1 callerid="Jane Smith" <5678> host=dynamic allow=ulaw allow=alaw (extensions.conf) [general] static=yes writeprotect=no [analog] include=>test...

...--> PBX (Alcatel OmniPCX) Grand idea is to use the micronet's POTS interfaces to connect SIP phones to the PBX and to the PSTN. I think i even managed my way in the arcane and cryptic management interface of that appliance, but I am stuck against theese messages: -- Executing Dial("SIP/xlite1-7a03", "H323/120/smallbox") in new stack --- h323_request - data 120/smallbox format 0x4 (ulaw) --- find_peer +++ find_peer +++ h323_request --- h323_call- 120/smallbox +++ h323_call -- Called 120/smallbox --- onNewCallCreated ooh323c_1 --- find_call +++ find_call...

...at I did not search the wiki first. I only have email access at the moment. I am having trouble setting both variables and global variables thru an extension. I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4 with an Xlite softphone. I have two xlite phones on diffent computers. One logs in as xlite1 and the other as SNOM. My dial plan is as follows Exten => 200,1,Dial(${OnCall},10) Exten => 201,1,Set(OnCall=SIP/SNOM) Exten => 202,1,Set(OnCall=SIP/xlite1) (I have tried Set and SetGlobalVar). When I use Set I get the following in the CLI -- Executing Set("SIP/snom-a645", &...

...to register correctly. Here is a snippet from the "sip debug" output. Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.x.x:5060;branch=z9hG4bK51bd9fa5 From: "asterisk" <sip:asterisk@192.168.x.x>;tag=as6a4689e3 To: <sip:192.168.2.50>;tag=1713780919 Contact: <sip:xlite1@192.168.2.50:5060> Call-ID: 2edd9eef1e40bad20f48302e4a1d673a@192.168.x.x Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1103m Content-Length: 0 Any reasons why I can't place a call. Thanks, Geoff

...2sp/30vip as skinny/sccp client on Asterisk (so, my conclusion is that problem is in Asterisk). The biggest problem there is that some numbers have free voice messages (for example: "number is changed, call xxx-xxxx") and I do NOT hear that messages. *CLI> -- Executing Dial("SIP/xlite1-f896", "SIP/xxxxxxxxxx@sipgate|50|tr") in new stack -- Called xxxxxxxxxx@sipgate -- SIP/sipgate-ce45 is making progress passing it to SIP/xlite1-f896

Hi guys, I do have some weird situation. I do have an * box, and I want to connect to that box from my Windows box by SIP via X-Ten Lite. I made configuration of that soft phone as it was suggested by lots of tutorials I found by Google. But... it doesn't work! I don't know what is wrong there, but I have unobstructed access to my asterisk box, created user in sip.conf, enabled

...onald [mailto:asterisk107@gmail.com] > Sent: 26 May 2005 10:47 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Little Php question > > > Hi > I'm trying to make a call from a local webpagee through my > xlite softphone > (xlite1) > BTW when I'm trying to do it through telnet it works > > For this I'm using a php script I found > > <PRE> > <?php > $socket = fsockopen("192.168.1.1","5038"); > $uname = "test"; > $secret = "test"; > #$exte...

...@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Ronald Sendt: 26 May 2005 11:47 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] Little Php question Hi I'm trying to make a call from a local webpagee through my xlite softphone (xlite1) BTW when I'm trying to do it through telnet it works For this I'm using a php script I found <PRE> <?php $socket = fsockopen("192.168.1.1","5038"); $uname = "test"; $secret = "test"; #$exten = 12345678; fputs($socket, "Action: Login\r...

The configuration for kphone in sip.conf: [177204] ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend ;regexten=1234 ; When they register, create extension 1234 ;username=xlite1 ;callerid="Jane Smith" <5678> host=dynamic ;nat=yes ; X-Lite is behind a NAT router ;canreinvite=no ; Typically set to NO if behind NAT disallow=all ;allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw ;allow=alaw...

...ex allow=ilbc allow=slinear [general] nat=yes register => 2129999999:<password>:2129999999@147.135.8.128:5060 register => 2129999999:<password>:2129999999@147.135.0.128:5060 externip = 208.59.47.2 localnet=192.168.1.0/255.255.0.0 [sip_proxy] type=user context=from-broadvoice [xlite1] type=friend regexten=101 username=xlite1 secret=<password> callerid="Stephen's Laptop" <101> host=dynamic nat=no canreinite=yes disallow=all allow=gsm allow=ulaw allow=alaw dtmfmode=inband qualify=yes [xlite2] type=friend regexten=103 context=sip username=103 secret=<...

...ead of a 'reload' or 'sip reload' as it hangs and stops processing calls or responding on the CLI. I tracked this down to something dealing with the peers I have in sip.conf. However, I removed all peers and just placed a simple friend in sip.conf (right from the sample file): [xlite1] ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend regexten=1234 ; When they register, create extension 1234 username=xlite1 callerid="Jane Smith" <5678&g...

Hello configuration as follows, dial-out works: capi.conf: [hfcpci] ;;PointToPoint (55512-0) isdnmode=MSN incomingmsn=* ;msn=61 controller=1 devices=2 context=incoming extensions.conf: [incoming] exten => _XX,1,Playback(demo-abouttotry) exten => _XX,n,Dial,SIP/xlite1 exten => _XX,n,HangUp When call is placed, the following debug info is shown, after the last line, it stalls until caller gives up: INFO_IND ID=001 #0x040a LEN=0016 Controller/PLCI/NCCI = 0x101 InfoNumber = 0x7e InfoElement = <04&g...

....1.0-RC2k) handles a single PCI HFC-S based card. I own a Cisco 7940 Sip phone (fw 7.1) and a pc running X-Lite. Zaptel.conf and zapata.conf are taken directly from zaphfc samples. Extension.conf contains the following lines: [from-ISDN1] exten=>s,1,Wait(1) exten=>s,2,Dial(Sip/cisco1&Sip/xlite1,30,tTr) exten=>s,3,HangUp The problem is that when I receive a call, I can't see the CallerID neither on the Cisco 7940 nor on the X-Lite client. Exactly the cisco phone tells me: "From asterisk / asterisk" x-lite: "Call incoming on line & asterisk" The strange thi...

...ons. Something like this... [foo-incoming] exten => 2122222222, 1, Goto(ACorp|1000|1) exten => 2123333333, 1, Goto(BCorp|1000|1) . . . exten => _NXXNXXXXXX, 1, Goto(GenericCorp|1000|1) [GenericCorp] exten => 1000, 1, SetCallerID("Generic Corp") exten => 1000, 2, Dial(SIP/xlite1&SIP/kphone1&SIP/xlite2,30,tr) exten => 1000, 3, Voicemail(u10000) [ACorp] exten => 1000, 1, SetCallerID("A Corp") exten => 1000, 2, Dial(SIP/xlite1&SIP/kphone1&SIP/xlite2,30,tr) exten => 1000, 3, Voicemail(u10000) [BCorp] exten => 1000, 1, SetCallerID(&quo...

...to do in order to get this working. I assume this goes in my SIP.conf file but I am unclear (a) if/how I need to register the client (b) what other code might be required to set this up (c) how to properly configure the settings for my SIP client now that just about every configuration changes. [XLITE1] ;www.xten.com type=friend username=ChooseAUsername secret=ChooseAPassword context=outbound ; match with the outgoing context in extensions.conf host=dynamic ; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=yes ; Typically set to NO if behind NAT allow=all ; code...

...to do in order to get this working. I assume this goes in my SIP.conf file but I am unclear (a) if/how I need to register the client (b) what other code might be required to set this up (c) how to properly configure the settings for my SIP client now that just about every configuration changes. [XLITE1] ;www.xten.com type=friend username=ChooseAUsername secret=ChooseAPassword context=outbound ; match with the outgoing context in extensions.conf host=dynamic ; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=yes ; Typically set to NO if behind NAT allow=all ; codec...