system toolbox » voip

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

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

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

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!!

proxy01.sipphone.com/1747  198.65.166.131              5060     UNREACHABLE

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

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 

https://my.sipphone.com/mysip/app

podesio vremensku zonu

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

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

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

http://karlsbakk.net/asterisk/gizmo-project.php

ovo je da poludiš ... jutros je radilo i sada više ne radi

http://users.757.org/~joat/wiki/index.php/Asterisk_and_Gizmo

ifold*CLI> sip show registry

Host                            Username       Refresh State                Reg.Time                 
proxy01.sipphone.com:5060       17473375695        120 Request Sent                   

sigmacom officesa (1-747-337-5695)

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

http://www.voipuser.org/forum_topic_9521.html

sličan problem

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