ty.roach@acecomm.com
2004-Nov-05 13:54 UTC
[Asterisk-Users] Questions from an Asterisk newbie - follow-up question.
Thanks, I'm getting closer. My phones now register successfully, however,
when I call phone x2000 calls phone x2001, I immediately get a fast busy
signal. I've included results from an ethereal trace, the asterisk console
window as well as my sip.conf and extensions.conf.
My apologies for asking goofie beginner questions, but the quick and
insightful responses here have been very helpful.
936.185203 172.20.23.211 -> 172.20.23.201 SIP/SDP Request: INVITE
sip:2001@172.20.23.201, with session description
936.186147 172.20.23.201 -> 172.20.23.211 SIP Status: 100 Trying
936.186937 172.20.23.201 -> 172.20.23.212 SIP/SDP Request: INVITE
sip:2001@172.20.23.212:5060, with session description
936.187604 172.20.23.201 -> 172.20.23.211 SIP Status: 180 Ringing
936.211385 172.20.23.212 -> 172.20.23.201 SIP Status: 400 Bad Request
936.211718 172.20.23.201 -> 172.20.23.212 SIP Request: ACK
sip:2001@172.20.23.212:5060
936.212364 172.20.23.201 -> 172.20.23.211 SIP/SDP Status: 200 OK, with
session description
936.233966 172.20.23.211 -> 172.20.23.201 SIP Request: BYE
sip:2001@172.20.23.201:5060
936.234312 172.20.23.201 -> 172.20.23.211 SIP Status: 200 OK
937.212643 172.20.23.201 -> 172.20.23.211 SIP/SDP Status: 200 OK, with
session description
938.213493 172.20.23.201 -> 172.20.23.211 SIP/SDP Status: 200 OK, with
session description
939.213350 172.20.23.201 -> 172.20.23.211 SIP/SDP Status: 200 OK, with
session description
940.213192 172.20.23.201 -> 172.20.23.211 SIP/SDP Status: 200 OK, with
session description
941.214025 172.20.23.201 -> 172.20.23.211 SIP/SDP Status: 200 OK, with
session description
The asterisk CLI shows this message:
*CLI>
*CLI>
*CLI> Nov 5 15:45:53 WARNING[-159011920]: chan_sip.c:683 retrans_pkt:
Maximum retries exceeded on call
ce30300-17dcd5-1f5ae-2e323731@172.20.23.211 for seqno 101 (Non-critical
Response)
FYI, here's my asterisk configuration files:
sip.conf -->
;************** Protocol definitions ***************
[general]
;----------- general setup
port = 5060
bindaddr = 0.0.0.0
tos = none
;----------- codecs setup
allow = all
;----------- other options
;context = default
context = bogon-calls
;----------- register to peers
;********************* Users ***********************
[2000]
type=friend
username=2000
;secret=9overthruster7
host=dynamic
;host=172.20.23.211
context=from-sip
mailbox=100
[2001]
type=friend
username=2001
;secret=11bbanzai9
host=dynamic
;host=172.20.23.212
context=from-sip
mailbox=101
extensions.conf -->
;***************** General options *****************
[general]
static=yes
writeprotect=yes
;******************* Globals values ******************
[globals]
;******************** DIAL PLAN ********************
[bogon-calls]
exten => _.,1,Congestion
[from-sip]
exten => 2000,1,Dial(SIP/2000,200,tr)
exten => 2000,2,Voicemail(u2000)
exten => 2000,102,Voicemail(b2000)
exten => 2000,103,Hangup
exten => 2001,1,Dial(SIP/2001,200,tr)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup
exten => 2999,1,VoicemailMain(${CALLERIDNUM})
Ben Greear
<greearb@candelatech.com> To:
Asterisk Users Mailing List - Non-Commercial Discussion
Sent by:
<asterisk-users@lists.digium.com>
asterisk-users-bounces@lists cc:
.digium.com Subject: Re:
[Asterisk-Users] Questions from an Asterisk newbie
11/05/04 02:33 PM
Please respond to Asterisk
Users Mailing List -
Non-Commercial Discussion
ty.roach@acecomm.com wrote:> 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 with two 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 just did something similar. I added these lines to
/etc/asterisk/sip.conf:
; Grandstream
[1001]
type=friend
host=dynamic
; cisco phone
[1002]
type=friend
host=dynamic
Then I added these lines to /etc/asterisk/extensions.conf
exten => 1001,1,Dial(SIP/1001,200,tr)
exten => 1002,1,Dial(SIP/1002,200,tr)
My phones register as phone numbers 1001 and 1002. There may be
a better way to do it, but with this config I was able to make
calls...
Ben
--
Ben Greear <greearb@candelatech.com>
Candela Technologies Inc http://www.candelatech.com
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