Ahmed Boreau
2003-Apr-07 02:36 UTC
[Asterisk-Users] Don't be upset !!! Architecture is need !!!
asterisk-users-request@lists.digium.com wrote:>Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can reach the person managing the list at > asterisk-users-admin@lists.digium.com > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Asterisk-Users digest..." > > >Today's Topics: > > 1. Re: FS: Cisco DT-24+, Dialogic D/240SC/T1, > Natural Microsystem AG-T1 (Klaus-Peter Junghanns) > 2. Re: FS: Cisco DT-24+, Dialogic D/240SC/T1, Natural > Microsystem AG-T1 (Mark Spencer) > 3. Bug? * not correctly honouring tag on To? (Stephen Davies) > 4. TDM400 question (Ron Gage) > 5. Re: Priority usage: absolute sequential vs. sequential (Steven Critchfield) > 6. Re: Bug? * not correctly honouring tag on To? (Mark Spencer) > 7. Re: Priority usage: absolute sequential vs. sequential (John Harragin) > 8. Re: Priority usage: absolute sequential vs. sequential (John Harragin) > 9. Quesiton about SIP and MSN (it) > 10. Re: Priority usage: absolute sequential vs. > sequential (John Todd) > 11. Re: TDM400 question (Michael Bielicki) > 12. Re: Priority usage: absolute sequential vs.sequential (John Harragin) > 13. SIP Testing (Mark Spencer) > 14. Re: Call completion/error codes and extensions.conf > call flow (Tilghman Lesher) > >--__--__-- > >Message: 1 >Subject: Re: [Asterisk-Users] FS: Cisco DT-24+, Dialogic D/240SC/T1, > Natural Microsystem AG-T1 >From: Klaus-Peter Junghanns <kpj@junghanns.net> >To: asterisk-users@lists.digium.com >Date: 06 Apr 2003 21:30:13 +0200 >Reply-To: asterisk-users@lists.digium.com > >hmmm....did you stop selling FXS-FXO adapters? and started >selling various telephony things? ;-) > >regards, >kapejod > >p.s. i have nothing for sale .... > >Am Son, 2003-04-06 um 22.12 schrieb info@aislecom.com: > > >>I have the following for sale: >>1) Cisco DT-24+ >>2) Dialogic D/240SC/T1 >>3) 2 Natural Microsystem AG-T1 >> >>Please contact me directly if you are interested. >> >>Dave >> >> >> >> > > >--__--__-- > >Message: 2 >Date: Sun, 6 Apr 2003 15:38:22 -0500 (CDT) >From: Mark Spencer <markster@digium.com> >To: <asterisk-users@lists.digium.com> >Subject: Re: [Asterisk-Users] FS: Cisco DT-24+, Dialogic D/240SC/T1, Natural > Microsystem AG-T1 >Reply-To: asterisk-users@lists.digium.com > >Do not post ads on the Asterisk mailing list. If there is demand on hte >list for ads such as these, we can setup a special list for that purpose. > >Mark > >On Sun, 6 Apr 2003 info@aislecom.com wrote: > > > >>I have the following for sale: >>1) Cisco DT-24+ >>2) Dialogic D/240SC/T1 >>3) 2 Natural Microsystem AG-T1 >> >>Please contact me directly if you are interested. >> >>Dave >> >> >> >> >> > > >--__--__-- > >Message: 3 >Date: Sun, 6 Apr 2003 22:16:45 +0100 (BST) >From: Stephen Davies <steve@daviesfam.org> >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] Bug? * not correctly honouring tag on To? >Reply-To: asterisk-users@lists.digium.com > >Hi Mark, > >Current CVS, * isn't correctly remembering the tag added to the To header >by a server. > >For instance: > >Sip read: >SIP/2.0 200 OK >Via: SIP/2.0/UDP 81.96.69.210:5062;branch=7b858c09 >From: steve-ata186 <sip:asterisk@81.96.69.210:5062>;tag=14925711 >To: <sip:18478974611@4.42.235.170>;tag=t2907cab0911c8g >Call-ID: 232752ec518d398f25ce03f45a8940f3@81.96.69.210 >CSeq: 102 INVITE >Allow: INVITE, CANCEL, REFER, BYE, ACK >Contact: <sip:0403242876@195.217.255.36:5061> >Content-Type: application/sdp >Record-Route: <sip:4.42.235.170:5060;lr> >Server: DTA SIP/0.11.7 NNOS/VR30 >Content-Length: 144 > >v=0 >o=0403242876 0 2 IN IP4 195.217.255.36 >s=- >c=IN IP4 4.42.235.170 >t=0 0 >m=audio 16082 RTP/AVP 8 101 >a=rtpmap:101 telephone-event/8000 > > >Transmitting: >ACK sip:18478974611@4.42.235.170 SIP/2.0 >Via: SIP/2.0/UDP 81.96.69.210:5062;branch=7b858c09 >Route: <sip:0403242876@195.217.255.36:5061> >From: "steve-ata186" <sip:asterisk@81.96.69.210:5062>;tag=14925711 >To: <sip:18478974611@4.42.235.170>;tag=14925711 >Contact: <sip:asterisk@81.96.69.210:5062> >Call-ID: 232752ec518d398f25ce03f45a8940f3@81.96.69.210 >CSeq: 102 ACK >User-Agent: Asterisk PBX >Content-Length: 0 > > >Notice that the tag in the ACK's To doesn't match that set by the server >in the 200 OK. > >Steve > > >--__--__-- > >Message: 4 >From: Ron Gage <ron@rongage.org> >To: asterisk-users@lists.digium.com >Date: 06 Apr 2003 17:16:46 -0400 >Subject: [Asterisk-Users] TDM400 question >Reply-To: asterisk-users@lists.digium.com > >Hi folks: > >Does the TDM400 card from Digium only support FXS, or is FXO >functionality available or planned? > > >Hi, I finally resolved my problem by using debian. I wanted to know exactly what devices I need to experience my asterisk servers ? Is my sound card enough or do I need additional devices ? thx for Ur help.
Hello Everyone, My * is behind a NAT connection, for which the IAX protocol works perfectly btw, but i was looking into getting sip to work. I forwarded the 5060 port to * which seems to work for connecting and outgoing data streams, but offcourse the incomming voice isn't working. Is there a way to supply * wich a port range and ip/hostname of the external host to be used for SIP ? so for example: rtp_port_start=8766 rtp_IP=xxx.xxx.xxx.xxx and maybe even a range for which not to use these ports rtp_local=192.168.0.0/24 Just some ideas, maybe some of them are allready implemented or on the todo list but i couldn't find anything about it on the mailinglist archive myself. Greetings, Tjardick