Re: [Asterisk-Users] Wo uses H323-phones with asterisk?
On Sat, Jul 17, 2004 at 10:35:58AM +0200, Christian Ekhart wrote: Hi, we successfully use innovaphone IP200 H.323 hardware phones with OH323/Asterisk. Calling/talking is OK, but call transfer does not work. Does anyone of you use H323-phones with asterisk AND IS ABLE to perform CALL TRANFERS?! Hello! A few months ago I tried to get an IP-200 to work with *. I had to use GnuGk where the IP-200 and * could register to. When using the R button on the phone to dial another call * would crash. So I am curious what versions of * and OH323 are you using? I am also interested in the configs. Maybe I have overlooked something back then... -Walter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi: sending incoming calls to different contexts
Hello, I am using chan_capi and would like * to behave differently depending on the MSN the caller dials. Is there a way to route incoming ISDN calls to different contexts based on the MSN dailled? I have tried something like msn=1234 incomingmsn=1234 context=msn1 msn=4567 incomingmsn=4567 context=msn2 in capi.conf but with no results. Thanks for any hints. -Walter -- Walter Doerr =*= [EMAIL PROTECTED] =*= FAX: +49 2421 962001 The poor folks who only have 100MBytes of RAM five years from now may not be able to buffer a 16MB packet, but that's their tough luck. (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?
On Fri, Jan 30, 2004 at 08:20:48AM -, Kannaiyan Natesan wrote: You can use, ;sip.conf register = username:[EMAIL PROTECTED]/extension to make asterisk as a SIP client. [...] Can Asterisk act like a normal Sip phone and e.g. connect to another sip-gateway? Background: There is a new german company at: http://www.sipgate.de (sorry German only page) I set up an account with sipgate yesterday evening and tried to use the above mentioned register in sip.conf * to login to sipgate. No luck so far. They use SER and I get 483 too many hops replies back from them. Any help is greatly appreciated. -Walter -- Walter Doerr =*= [EMAIL PROTECTED] =*= FAX: +49 2421 962001 The poor folks who only have 100MBytes of RAM five years from now may not be able to buffer a 16MB packet, but that's their tough luck. (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?
On Fri, Jan 30, 2004 at 09:51:44AM -, Kannaiyan Natesan wrote: I think there is a loopback. Or, their SER forwards packets to itself. Hard to tell without knowing their config. Did you debug that with sip debug in console and look at SIP Messages what is doing ? Yes. -Walter -- Walter Doerr =*= [EMAIL PROTECTED] =*= FAX: +49 2421 962001 The poor folks who only have 100MBytes of RAM five years from now may not be able to buffer a 16MB packet, but that's their tough luck. (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?
On Fri, Jan 30, 2004 at 12:06:47PM +0100, Klaus-Peter Junghanns wrote: hi, just signed up and it works like a charm. :-) They even support g711 :) and multiple channels :) make sure you have in sip.conf: register = :[EMAIL PROTECTED]/extension in your context I believe that I have that entry in sip.conf. Maybe not the extension. Still no luck. you will get the too many hops if you try to register with their proxy (proxy.de.sipgate.net). I tried both. Besides, both names resolve to the same IP. Here is what I just received: Sip read: SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP 212.102.234.130:5060;branch=z9hG4bK7ba3fc35 From: asterisk sip:[EMAIL PROTECTED];tag=as2f4213d0 To: sip:217.10.79.9;tag=b11cb9bb270104b49a99a995b8c68544.566f Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 217.10.79.9:5060 Noisy feedback tells: pid=30305 req_src_ip=217.10.79.9 req_src_port=5060 in_uri=sip:217.10.79.9 out_uri=sip:217.10.79.9 via_cnt==22 -Walter -- Walter Doerr =*= [EMAIL PROTECTED] =*= FAX: +49 2421 962001 The poor folks who only have 100MBytes of RAM five years from now may not be able to buffer a 16MB packet, but that's their tough luck. (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?
On Fri, Jan 30, 2004 at 12:22:18PM +0100, Walter Doerr wrote: On Fri, Jan 30, 2004 at 12:06:47PM +0100, Klaus-Peter Junghanns wrote: hi, just signed up and it works like a charm. :-) They even support g711 :) and multiple channels :) make sure you have in sip.conf: register = :[EMAIL PROTECTED]/extension in your context I believe that I have that entry in sip.conf. Maybe not the extension. Still no luck. Following up to my own message: * is working with sipgate now (should be no surprise as they are using * too). Apparently I have no idea how to setup a sip.conf file. I have the above mentioned register command and in addition a [sipgate] section in sip.conf. After removing the [sipgate] section everything works fine. -Walter -- Walter Doerr =*= [EMAIL PROTECTED] =*= FAX: +49 2421 962001 The poor folks who only have 100MBytes of RAM five years from now may not be able to buffer a 16MB packet, but that's their tough luck. (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for SIP
On Thu, Jan 29, 2004 at 05:04:22PM +0100, Cristian Manoni wrote: Hi All i have continuos error: Unable to handle DTMF tone 'f' for 'SIP on the asterisk console. after this the call hang up. Look at softdtmf in capi.conf. Setting the parameter to 0 solved the problem for me. -Walter -- Walter Doerr =*= [EMAIL PROTECTED] =*= FAX: +49 2421 962001 The poor folks who only have 100MBytes of RAM five years from now may not be able to buffer a 16MB packet, but that's their tough luck. (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maillinglist as newsgroup ?
On Fri, Jan 23, 2004 at 07:00:49AM +, Hans-Henrik Andresen wrote: Hi, I was thinking if it was possible to get this list as news ? http://www.gmane.org offers many mailinglists as a newsfeed. Even VoIP stuff such as * and SER. You can also read/search the mailinglists via the web interface. It would be much easier that 'hotmail-account' No hotmail please. -Walter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP: Register that isn't a register?
On Mon, Jan 19, 2004 at 12:33:41PM +0100, Philipp von Klitzing wrote: Ok, here comes part two of the log quiz, this time SIP not MGCP: WARNING[8201]: chan_sip.c:4821 handle_response: Got 200 OK on REGISTER that isn't a register This is most probably cause by registration of * with FWD. I am seeing this with iptel.org -Walter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN CAPI and anonymous callers
Hello, I am trying to use * to handle anonymous ISDN callers. Something like exten = 5150/0,1,Congestion should work but doesn't. Apparently because the ISDN CAPI doesn't use 0 for callers who don't send their number. Is there a way to make * identify ISDN callers who use CLIR? -Walter -- Walter Doerr =*= [EMAIL PROTECTED] =*= FAX: +49 2421 962001 The poor folks who only have 100MBytes of RAM five years from now may not be able to buffer a 16MB packet, but that's their tough luck. (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
