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HI,
First you should start by editing two different .conf
files: sip.conf and extensions.conf in
/etc/asterisk
The best way is to let asterisk create the samples
command make samples in
/sr/src/asterisk
Sample .conf files will be created in /etc/asterisk
A step by step info can be found in http://www.voip-info.org/wiki-Asterisk+installation+tips
go to the part of SIP configuration. There is also some info
about your Cisco phones.
Good Luck
Paulo
Francisco Paulo Adriano
WaveLIS LDA Mobile +351 91 870 87 98 Office + 351 21 989 83 34 Fax +351 21 989 83 35 E-mail : [EMAIL PROTECTED] >>> [EMAIL PROTECTED] 05-11-2004 18:45:46 >>> I have just installed asterisk in the hopes of operating a very simple
VoIP
demo. The demo environment is as follows: Asterisk 1.0.2 installed on a Fedora 2 Linux laptop. The laptop is connected to a hub along wittwo Cisco 7960 IP phones (SIP enabled). I've manually configured the phones setting the IP address of the phones, phone names (extensions), the IP address of the SIP proxy (Asterisk server?). I have not made any modifications to any of the asterisk configuration files. I run asterisk ('asterisk -cv') from the command line just to see what happens. Essentially, I get messages from both SIP phones indicating that registration is failing (I guess not such as surprise since I haven't configured anything). For starters, I was hoping that some of the experts on this board could give me some tips on what I need to do to allow one phone to successfully call the other phone. I did a similar thing several years ago using a SIP proxy server (from Dynamicsoft, albeit, with help from their support group). Any advise would be greatly appreciated. Thanks and advance. Ty P.S. I've included command line output from my asterisk console below... *CLI> sip debug SIP Debugging Enabled *CLI> *CLI> *CLI> *CLI> Sip read: REGISTER sip:172.20.23.201 SIP/2.0 Via: SIP/2.0/UDP 172.20.23.211:5060 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER Contact: <sip:[EMAIL PROTECTED]:5060> Expires: 3600 Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 172.20.23.211 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 172.20.23.211:5060 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as106566ef Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 172.20.23.211:5060 Nov 5 13:37:01 NOTICE[-159417424]: chan_sip.c:7571 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '172.20.23.211' Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Destroying call '[EMAIL PROTECTED]' Sip read: REGISTER sip:172.20.23.201 SIP/2.0 Via: SIP/2.0/UDP 172.20.23.212:5060 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Fri, 05 Nov 2004 18:38:54 GMT CSeq: 101 REGISTER Contact: <sip:[EMAIL PROTECTED]:5060> Expires: 3600 Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 172.20.23.212 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 172.20.23.212:5060 From: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as66d562fd Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 172.20.23.212:5060 Nov 5 13:37:36 NOTICE[-159417424]: chan_sip.c:7571 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '172.20.23.212' Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms _______________________________________________ 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 |
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