Nove funkcije #14728
Zatvorenasterisk.bring.out.ba <-> gizmo registration problem
0%
Opis
vanjska adresa asterisk.bring.out.ba
koja ide na ifold
Povezani tiketi 1 (0 otvoreno — 1 zatvoren)
Izmjenjeno od Ernad Husremović prije skoro 18 godina
imam ovaj problem kod registracije sipphone-a
[Jul 2 10:27:07] NOTICE[336]: chan_sip.c:7492 sip_reg_timeout: -- Registration for '17473375695@proxy01.sipphone.com' timed out, trying again (Attempt #8) [Jul 2 10:27:07] WARNING[336]: chan_sip.c:7570 transmit_register: Probably a DNS error for registration to 17473375695@proxy01.sipphone.com, trying REGISTER again (after 20 seconds)
hm kao da mu neki dns treba
podesio na router-wan-sa-1 (refresh_ip config) internet.sigma-com stavku:
[root@ernadh ~]# ping internet.sigma-com.net PING officesa.sigma-com.net (89.146.169.194) 56(84) bytes of data. 64 bytes from 89.146.169.194: icmp_seq=0 ttl=51 time=128 ms
Izmjenjeno od Ernad Husremović prije skoro 18 godina
na kraju problem riješio tako što sam stavio qualify=no
[proxy01.sipphone.com] type=peer disallow=all allow=ulaw allow=ilbc dtfmode=rfc2833 host=proxy01.sipphone.com fromdomain=proxy01.sipphone.com insecure=very qualify=no <<<<<<<<<<<<<<<<<<<<<<<<<<<< bilo yes fromuser=17473375695 authuser=17473375695 username=17473375695 secret=ultra_Super_Tajni_PassWord canreinvite=no
iako mi sistem i dalje govori:
[Jul 2 10:47:11] NOTICE[336]: chan_sip.c:7492 sip_reg_timeout: -- Registration for '17473375695@proxy01.sipphone.com' timed out, trying again (Attempt #19)
ja mogu raditi pozive preko gizmo-a
Izmjenjeno od Ernad Husremović prije skoro 18 godina
hm ne mogu, ja sam testirao kucajući nepostojeći broj, pa mi se teta javi, ali registracija je očigledno neophodna da ostvarim konekciju sa drugom stranom
Izmjenjeno od Ernad Husremović prije skoro 18 godina
- Naslov promijenjeno iz asterisk.bring.out.ba u asterisk.bring.out.ba <-> gizmo
http://www.internettablettalk.com/forums/showthread.php?t=10344
[n800] type=friend username=n800 secret=12345678 canreinvite=no dtmf=rfc2833 host=dynamic mailbox=2100@default context=provider
Don't worry about the 405 "Method Not Allowed" messages,
I think those have to do with the mailbox=2100@default line. As far as I can tell the N800 Rtcom stuff doesn't support voicemail info
(#messages waiting, new message, etc).
Izmjenjeno od Ernad Husremović prije skoro 18 godina
ovo mi takođe kaže, ako kažem qualify=yes
[Jul 2 11:13:40] NOTICE[336]: chan_sip.c:15766 sip_poke_noanswer: Peer 'proxy01.sipphone.com' is now UNREACHABLE! Last qualify: 0!!
Izmjenjeno od Ernad Husremović prije skoro 18 godina
proxy01.sipphone.com/1747 198.65.166.131 5060 UNREACHABLE
Izmjenjeno od Ernad Husremović prije skoro 18 godina
Retransmitting #3 (no NAT) to 198.65.166.131:5060: OPTIONS sip:proxy01.sipphone.com SIP/2.0 Via: SIP/2.0/UDP 89.146.169.194:5060;branch=z9hG4bK5dfbeeaa;rport From: "asterisk" <sip:asterisk@89.146.169.194>;tag=as1d2a7385 To: <sip:proxy01.sipphone.com> Contact: <sip:asterisk@89.146.169.194> Call-ID: 64eb46bc2d4b215c78e73427655a92f2@89.146.169.194 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 02 Jul 2008 09:17:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0
stavio u peer nat=yes
Retransmitting #4 (NAT) to 198.65.166.131:5060: OPTIONS sip:proxy01.sipphone.com SIP/2.0 Via: SIP/2.0/UDP 89.146.169.194:5060;branch=z9hG4bK52965f7d;rport From: "asterisk" <sip:asterisk@89.146.169.194>;tag=as74b8a3fc To: <sip:proxy01.sipphone.com> Contact: <sip:asterisk@89.146.169.194> Call-ID: 00f4336a2e7070b74ea92b005546ccdf@89.146.169.194 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 02 Jul 2008 09:20:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0
Izmjenjeno od Ernad Husremović prije skoro 18 godina
interesantno je ovo
nakon 10:19 nije bilo uopšte saobraćaja sa ovog servera:
root@monitor:~# tail /var/log/syslog --lines=100000 | grep 198.65.166.131
Jul 2 09:43:40 192.168.45.254 kernel: P_BIHNET_TR IN=ppp0 OUT=br-lan SRC=198.65.166.131 DST=192.168.45.4 LEN=70 TOS=0x00 PREC=0x00 TTL=47 ID=1 DF PROTO=UDP SPT=46670 DPT=10174 LEN=50 ... Jul 2 09:43:40 192.168.45.254 kernel: P_BIHNET_TR IN=ppp0 OUT=br-lan SRC=198.65.166.131 DST=192.168.45.4 LEN=70 TOS=0x00 PREC=0x00 TTL=47 ID=5 DF PROTO=UDP SPT=46670 DPT=10174 LEN=50 ... P_BIHNET_TR IN=ppp0 OUT=br-lan SRC=198.65.166.131 DST=192.168.45.4 LEN=200 TOS=0x00 PREC=0x00 TTL=47 ID=12 DF PROTO=UDP SPT=47322 DPT=8616 LEN=180 Jul 2 10:19:35 192.168.45.254 kernel: P_BIHNET_TR IN=ppp0 OUT=br-lan SRC=198.65.166.131 DST=192.168.45.4 LEN=116 TOS=0x00 PREC=0x00 TTL=47 ID=1 DF PROTO=UDP SPT=47323 DPT=8617 LEN=96
Izmjenjeno od Ernad Husremović prije skoro 18 godina
https://my.sipphone.com/mysip/app
podesio vremensku zonu
Izmjenjeno od Ernad Husremović prije skoro 18 godina
ponovo sam stavio qalify=no (jer izgleda da mi OPTION message koja se koristi za pinganje remote peer-a javlja da je nedozvoljena
sada imam ovu sip message:
Retransmitting #3 (no NAT) to 198.65.166.131:5060: REGISTER sip:proxy01.sipphone.com SIP/2.0 Via: SIP/2.0/UDP 89.146.169.194:5060;branch=z9hG4bK5d1f514c;rport From: <sip:17473375695@proxy01.sipphone.com>;tag=as5a7dd7de To: <sip:17473375695@proxy01.sipphone.com> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< ? Call-ID: 202691e20c5d7bc8564b778a5cf4f2c5@proxy01.sipphone.com CSeq: 106 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:s@89.146.169.194> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< ? Event: registration Content-Length: 0
Izmjenjeno od Ernad Husremović prije skoro 18 godina
evo šta kaže lokal 51 kada njega debugujem
ifold*CLI> sip set debug o peer 51
<--- SIP read from 77.239.10.223:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 89.146.169.194:5060;branch=z9hG4bK202c7e4b;rport Call-ID: 130915ad3e8bd6cb506f19550383120b@89.146.169.194 CSeq: 102 OPTIONS From: "asterisk" <sip:asterisk@89.146.169.194>;tag=as0deb2327 To: <sip:51@77.239.10.223:5060>;tag=UEgJh9QlEoFOwgYm <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< Contact: <sip:51@77.239.10.223:5060> Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Accept: application/sdp, message/sipfrag, application/dtmf-relay Supported: replacesdebug o peer 51 Content-Length: 0 Content-Type: application/sdp Content-Length: 268
Izmjenjeno od Ernad Husremović prije skoro 18 godina
evo kako se registruje aastra: sip 20
REGISTER sip:asterisk.bring.out.ba:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.45.151:5060;branch=z9hG4bK3b80ae266a3e0834e Max-Forwards: 70 From: <sip:20@asterisk.bring.out.ba:5060>;tag=015f686e14 To: <sip:20@asterisk.bring.out.ba:5060> Call-ID: 5f1600951bd31e4d CSeq: 29073 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: 20 <sip:20@192.168.45.151:5060;transport=udp>;expires=10 User-Agent: Aastra 53i/2.2.1.25 Content-Length: 0
Izmjenjeno od Ernad Husremović prije skoro 18 godina
Izmjenjeno od Ernad Husremović prije skoro 18 godina
ovo je da poludiš ... jutros je radilo i sada više ne radi
http://users.757.org/~joat/wiki/index.php/Asterisk_and_Gizmo
Izmjenjeno od Ernad Husremović prije skoro 18 godina
ifold*CLI> sip show registry
Host Username Refresh State Reg.Time proxy01.sipphone.com:5060 17473375695 120 Request Sent
Izmjenjeno od Ernad Husremović prije skoro 18 godina
sigmacom officesa (1-747-337-5695)
Izmjenjeno od Ernad Husremović prije skoro 18 godina
root@ifold:~# nmap -p 5060 -sU proxy01.sipphone.com
Starting Nmap 4.20 ( http://insecure.org ) at 2008-07-02 13:31 CEST Interesting ports on northamerica.sipphone.com (198.65.166.131): PORT STATE SERVICE 5060/udp open|filtered sip <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< interesantno open|filtered ?!
Nmap finished: 1 IP address (1 host up) scanned in 3.317 seconds
root@ifold:~# nmap -p 5060 proxy01.sipphone.com
Starting Nmap 4.20 ( http://insecure.org ) at 2008-07-02 13:31 CEST Interesting ports on northamerica.sipphone.com (198.65.166.131): PORT STATE SERVICE 5060/tcp open sip Nmap finished: 1 IP address (1 host up) scanned in 1.334 seconds
Izmjenjeno od Ernad Husremović prije skoro 18 godina
http://www.voipuser.org/forum_topic_9521.html
sličan problem
Izmjenjeno od Ernad Husremović prije skoro 18 godina
stavio ovo podešenje
[general] context=demo ;allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) canreinvite=no srvlookup=yes language=bs realm=asterisk.bring.out.ba domain=asterisk.bring.out.ba localnet=192.168.45.0/255.255.255.0 externhost=internet.sigma-com.net ;Specify how often (in seconds) a hostname DNS lookup should be performed for the value entered in 'externhost'. Default 10 seconds externrefresh=10 nat=yes register => 17473375695:xxxxxxxxxxxxx@proxy01.sipphone.com/80
vidi sad contact polja:
SIP/2.0 200 OK Via: SIP/2.0/UDP 89.146.169.194:5060;branch=z9hG4bK433810c3;rport=1035 From: <sip:17473375695@proxy01.sipphone.com>;tag=as6927dbb1 To: <sip:17473375695@proxy01.sipphone.com>;tag=21a483426c2cd5d9b85bffe6bba40a2e.55d5 Call-ID: 0af466286da9c3d8147a54137449bd81@127.0.1.1 CSeq: 103 REGISTER Contact: <sip:80@89.146.169.194>;expires=120 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< Content-Length: 0
Retransmitting #1 (NAT) to 198.65.166.131:1035: OPTIONS sip:17473375695@proxy01.sipphone.com:5060 SIP/2.0 Via: SIP/2.0/UDP 89.146.169.194:5060;branch=z9hG4bK33e0bb8a;rport From: "asterisk" <sip:asterisk@89.146.169.194>;tag=as67d636ce To: <sip:17473375695@proxy01.sipphone.com:5060> Contact: <sip:asterisk@89.146.169.194> Call-ID: 0c8384534e379b9f069ac4e81b8cf232@89.146.169.194 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 02 Jul 2008 13:05:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0
Izmjenjeno od Ernad Husremović prije skoro 18 godina
- Naslov promijenjeno iz asterisk.bring.out.ba <-> gizmo u asterisk.bring.out.ba <-> gizmo registration problem
Izmjenjeno od Ernad Husremović prije više od 17 godina
- Status promijenjeno iz Dodijeljeno u Zatvoreno