Displaying 3 results from an estimated 3 matches for "testacc77003".
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testacc77000
2014 Jul 31
1
Subscription-State always active ?
...32273]: chan_sip.c:26194 sip_poke_noanswer:
Peer 'testacc77000' is now UNREACHABLE! Last qualify: 49//
//[Jul 31 11:56:58] Really destroying SIP dialog
'78b0d1701d3694b1494a0c4b55344d57 at ip-sip-server:5060' Method: OPTIONS//
//[Jul 31 11:56:58] set_destination: Parsing
<sip:testacc77003 at 192.168.1.109:1024> for address/port to send to//
//[Jul 31 11:56:58] set_destination: set destination to 192.168.1.109:1024//
//[Jul 31 11:56:58] Reliably Transmitting (NAT) to my-public-ip:1024://
//NOTIFY sip:testacc77003 at 192.168.1.109:1024 SIP/2.0//
//Via: SIP/2.0/UDP ip-sip-server:506...
2014 Oct 07
1
Grandstream GXP2160 + SRTP
Hello,
I am trying to setup a Grandstream GXP2160 IP-phone with secure calling
(SRTP).
Secure signaling SSIP for registration is working great !
I follow this guide :
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
But when I try to make a call with SRTP, I get stuck. There is an
initial INVITE which is anwered with a 401. There should follow a new
INVITE with a nonce,
2014 Aug 11
0
NotifyCID to see who is calling for call pickup
...how can I see the external number that is calling in ?
I would expect to see : 10 --> 3221234567
3221234567 being the external number I would like to know about.
This is what Asterisk sends to my Snom IP-phone :
[Aug 11 16:37:56] Reliably Transmitting (NAT) to my.pub.lic.ip:1024:
NOTIFY sip:testacc77003 at 192.168.1.109:1024 SIP/2.0
Via: SIP/2.0/UDP ip.ast.ser.ver:5060;branch=z9hG4bK5b999cd4;rport
Max-Forwards: 70
From: <sip:10 at ip.ast.ser.ver;user=phone>;tag=as6b302bda
To: <sip:testacc77003 at ip.ast.ser.ver>;tag=ashydm1he5
Contact: <sip:10 at ip.ast.ser.ver:5060>
Call-ID: 3c2...