hi. I just migrate from astlinux distro to PIAF,
I have set extensions and trunks. however the extension from a polycom 601 and pap2 na do not register themselves.
I have tried a lot of time researching but without success
any help would be welcome.
the network is working in eth1.
14 internal extensions have been set in polycom and one in pap2 the last show
Can’t connect to login server
so no one device is registering extensions. they were working nicely at astlinux distro. but not in PIAf
here some logs. the trunks register just fine but the extensions none
[2009-10-24 02:18:08] VERBOSE[5366] logger.c:
<— SIP read from 198.65.166.131:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.67:5060;branch=z9hG4bK7b793e36;rport=5060
From: sip:[email protected];tag=as0511b9f9
To: sip:[email protected];tag=92390300a369f0d75803e369c733575e.ab80
Call-ID: [email protected]
CSeq: 103 REGISTER
Contact: sip:[email protected]:5060;expires=96, sip:[email protected];expires=120
Content-Length: 0
<------------->
[2009-10-24 02:18:08] VERBOSE[5366] logger.c: — (8 headers 0 lines) —
[2009-10-24 02:18:08] VERBOSE[5366] logger.c: Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)
[2009-10-24 02:18:08] NOTICE[5366] chan_sip.c: Outbound Registration: Expiry for proxy01.sipphone.com is 120 sec (Scheduling reregistration in 105 s)
[2009-10-24 02:18:08] VERBOSE[5366] logger.c:
<— SIP read from 192.168.2.108:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.109:5060;branch=z9hG4bK5449517f;rport
From: “Unknown” sip:[email protected];tag=as4c109691
To: sip:192.168.2.108;tag=6D319E7F-99ABD314
CSeq: 102 OPTIONS
Call-ID: [email protected]
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.1.0137
Accept-Language: en
Content-Length: 0
<------------->
[2009-10-24 02:18:08] VERBOSE[5366] logger.c: — (11 headers 0 lines) —
[2009-10-24 02:18:08] VERBOSE[5366] logger.c: Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
[2009-10-24 02:18:08] VERBOSE[5366] logger.c: Reliably Transmitting (NAT) to 192.168.2.108:5060:
OPTIONS sip:192.168.2.108 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.109:5060;branch=z9hG4bK6bd6a6e6;rport
From: “Unknown” sip:[email protected];tag=as04dffefc
To: sip:192.168.2.108
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 24 Oct 2009 07:18:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
[2009-10-24 02:18:09] VERBOSE[5366] logger.c:
<— SIP read from 192.168.2.108:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.109:5060;branch=z9hG4bK6bd6a6e6;rport
From: “Unknown” sip:[email protected];tag=as04dffefc
To: sip:192.168.2.108;tag=FA9F31D8-EA739891
CSeq: 102 OPTIONS
Call-ID: [email protected]
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.1.0137
Accept-Language: en
Content-Length: 0
<------------->
[2009-10-24 02:18:09] VERBOSE[5366] logger.c: — (11 headers 0 lines) —
[2009-10-24 02:18:09] VERBOSE[5366] logger.c: Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
[2009-10-24 02:18:09] VERBOSE[5366] logger.c: Reliably Transmitting (NAT) to 192.168.2.108:5060:
OPTIONS sip:192.168.2.108 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.109:5060;branch=z9hG4bK53b33458;rport
From: “Unknown” sip:[email protected];tag=as4749e14d
To: sip:192.168.2.108
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 24 Oct 2009 07:18:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
and the following is in the pap2 na
[2009-10-24 02:18:25] VERBOSE[5366] logger.c:
<— SIP read from 192.168.2.103:5060 —>
NOTIFY sip:192.168.2.109 SIP/2.0
v: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-f05a2b30;rport
f: 2222 sip:[email protected];tag=cdb562997b0d0966o1
t: sip:192.168.2.109
i: [email protected]
CSeq: 81 NOTIFY
Max-Forwards: 70
o: keep-alive
User-Agent: Linksys/PAP2-3.1.22(LS)
Proxy-Require: Yes
l: 0
<------------->
[2009-10-24 02:18:25] VERBOSE[5366] logger.c: — (11 headers 0 lines) —
[2009-10-24 02:18:25] VERBOSE[5366] logger.c: Sending to 192.168.2.103 : 5060 (NAT)
[2009-10-24 02:18:25] VERBOSE[5366] logger.c:
<— Transmitting (NAT) to 192.168.2.103:5060 —>
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-f05a2b30;received=192.168.2.103;rport=5060
From: 2222 sip:[email protected];tag=cdb562997b0d0966o1
To: sip:192.168.2.109;tag=as2242954c
Call-ID: [email protected]
CSeq: 81 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Content-Length: 0
here are another issue the from: to: show the same ip
From: “3121 Casa.” sip:[email protected];tag=C5F6EFE0-9F0664B9
the 3121 casa is in the polycom phone which has 192.168.2.108 ip but show 192.168.2.109 (the pbx ip)
and in to:
To: sip:[email protected] show the same ip
so the from: to: has de same ip instead of
from 192.168.2.108
to:192.168.2.109
[2009-10-24 10:32:26] VERBOSE[5366] logger.c: Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)
[2009-10-24 10:32:27] VERBOSE[5366] logger.c:
<— SIP read from 192.168.2.108:5060 —>
REGISTER sip:192.168.2.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.108;branch=z9hG4bK722d2212EDC5A965
From: “3121 Casa.” sip:[email protected];tag=C5F6EFE0-9F0664B9
To: sip:[email protected]
CSeq: 1 REGISTER
Call-ID: [email protected]
Contact: sip:[email protected];methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.1.0137
Accept-Language: en
Max-Forwards: 70
Expires: 3600
Content-Length: 0
<------------->
[2009-10-24 10:32:27] VERBOSE[5366] logger.c: — (12 headers 0 lines) —
[2009-10-24 10:32:27] VERBOSE[5366] logger.c: Using latest REGISTER request as basis request
[2009-10-24 10:32:27] VERBOSE[5366] logger.c: Sending to 192.168.2.108 : 5060 (no NAT)
[2009-10-24 10:32:27] ERROR[5366] chan_sip.c: Peer ‘3121’ is trying to register, but not configured as host=dynamic
[2009-10-24 10:32:27] VERBOSE[5366] logger.c:
<— Reliably Transmitting (no NAT) to 192.168.2.108:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.108;branch=z9hG4bK722d2212EDC5A965;received=192.168.2.108
From: “3121 Casa.” sip:[email protected];tag=C5F6EFE0-9F0664B9
To: sip:[email protected];tag=as6bea5c52
Call-ID: [email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="57045639"
Content-Length: 0