[Asterisk-Users] Don't be upset !!! Architecture is need !!!
Ahmed Boreau
ahmed.boreau at esmt.sn
Mon Apr 7 02:36:26 MST 2003
asterisk-users-request at lists.digium.com wrote:
>Send Asterisk-Users mailing list submissions to
> asterisk-users at 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 at lists.digium.com
>
>You can reach the person managing the list at
> asterisk-users-admin at 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 at junghanns.net>
>To: asterisk-users at lists.digium.com
>Date: 06 Apr 2003 21:30:13 +0200
>Reply-To: asterisk-users at 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 at 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 at digium.com>
>To: <asterisk-users at lists.digium.com>
>Subject: Re: [Asterisk-Users] FS: Cisco DT-24+, Dialogic D/240SC/T1, Natural
> Microsystem AG-T1
>Reply-To: asterisk-users at 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 at 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 at daviesfam.org>
>To: asterisk-users at lists.digium.com
>Subject: [Asterisk-Users] Bug? * not correctly honouring tag on To?
>Reply-To: asterisk-users at 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 at 81.96.69.210:5062>;tag=14925711
>To: <sip:18478974611 at 4.42.235.170>;tag=t2907cab0911c8g
>Call-ID: 232752ec518d398f25ce03f45a8940f3 at 81.96.69.210
>CSeq: 102 INVITE
>Allow: INVITE, CANCEL, REFER, BYE, ACK
>Contact: <sip:0403242876 at 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 at 4.42.235.170 SIP/2.0
>Via: SIP/2.0/UDP 81.96.69.210:5062;branch=7b858c09
>Route: <sip:0403242876 at 195.217.255.36:5061>
>From: "steve-ata186" <sip:asterisk at 81.96.69.210:5062>;tag=14925711
>To: <sip:18478974611 at 4.42.235.170>;tag=14925711
>Contact: <sip:asterisk at 81.96.69.210:5062>
>Call-ID: 232752ec518d398f25ce03f45a8940f3 at 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 at rongage.org>
>To: asterisk-users at lists.digium.com
>Date: 06 Apr 2003 17:16:46 -0400
>Subject: [Asterisk-Users] TDM400 question
>Reply-To: asterisk-users at 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.
More information about the asterisk-users
mailing list