I'm learning Asterisk, and am going through O'Reilly's "Asterisk: the
definitive guide," but it's a bit of a slog. Where I'm hung up on is
connecting a softphone. Because I'm having difficulty getting a
softphone to connect, I've turned to sipsak and now SIPp.
Can I use SIPp as a CLI based softphone? At this point, I just want
output from SIPp showing a connection to the local Asterisk box.
My setup in Asterisk:
tleilax*CLI>
tleilax*CLI> sip show users
Username Secret Accountcode Def.Context
ACL Forcerport
101 password 101 default No Yes
gs102 password gs102 default No Yes
tleilax*CLI>
tleilax*CLI> sip show user 101
* Name : 101
Secret : <Set>
MD5Secret : <Not set>
Context : default
Language : en
Accountcode : 101
AMA flags : Unknown
Netborder CPD: No
Transfer mode: open
MaxCallBR : 384 kbps
CallingPres : Presentation Allowed, Not Screened
Call limit : 0
Callgroup :
Pickupgroup :
Callerid : "" <101>
ACL : No
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Sess-Min-SE : 90 secs
RTP Engine : asterisk
Codec Order : (ulaw:20,gsm:20)
Auto-Framing: No
tleilax*CLI>
I think I have a good connection with asterisk:
thufir@doge:~$
thufir@doge:~$ sudo sipsak -vv -s sip:345@tleilax -m "hi"
No SRV record: _sip._tcp.tleilax
No SRV record: _sip._udp.tleilax
using A record: tleilax
Max-Forwards set to 0
message received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
127.0.1.1:60506;branch=z9hG4bK.2702c76c;alias;received=192.168.1.3;rport=60506
From: sip:sipsak@127.0.1.1:60506;tag=70433c99
To: sip:345@tleilax;tag=as6e76406d
Call-ID: 1883454617@127.0.1.1
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.29.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.1.2:5060>
Accept: application/sdp
Content-Length: 0
** reply received after 0.751 ms **
SIP/2.0 200 OK
final received
thufir@doge:~$
thufir@doge:~$ sudo sipsak -vv -s sip:thu...@ekiga.net -m "hi"
No SRV record: _sip._tcp.ekiga.net
No SRV record: _sip._udp.ekiga.net
using A record: ekiga.net
Max-Forwards set to 0
message received:
SIP/2.0 483 Too Many Hops
Via: SIP/2.0/UDP
192.168.1.3:40598;branch=z9hG4bK.343b9a15;rport=40598;alias;received=96.48.128.162
From: sip:sipsak@127.0.1.1:40598;tag=4bc6f498
To: sip:thu...@ekiga.net;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.3c66
Call-ID: 1271329944@127.0.1.1
CSeq: 1 OPTIONS
Server: Kamailio (1.5.3-notls (i386/linux))
Content-Length: 0
** reply received after 170.903 ms **
SIP/2.0 483 Too Many Hops
final received
thufir@doge:~$
Obviously, I can't send anything to Ekiga due to the poor connection,
but tleilax and doge have a good connection.
http://unix.stackexchange.com/questions/186041/
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