[Asterisk-Users] Questions from an Asterisk newbie - follow-up question.

Fri Nov 5 13:54:30 MST 2004

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 at 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 at 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 at 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 at 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 at 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 at candelatech.com>           To:       Asterisk Users Mailing List - Non-Commercial Discussion                    
                      Sent by:                             <asterisk-users at lists.digium.com>                                                   
                      asterisk-users-bounces at 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 at 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 at candelatech.com>
Candela Technologies Inc  http://www.candelatech.com

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