I have a test SIP account set up with WorldCom and I have been trying to
have Asterisk register to the WorldCom server with no luck. It appears
that the SIP headers are different coming from Asterisk. I have included
a packet capture from a successful login with a Windows Messenger client
for reference. I have also copied in the SIP packet I captured with sip
debug turned on. In my sip.conf file, I have the following line under
the general context:
Register=bobscheller:password@166.60.255.41
Any pointers or assistance is appreciated.
Thanks,
Bob
******ASTERISK SIP PACKET ****************
XXX Need to handle Retransmitting XXX:
REGISTER sip:166.60.255.41 SIP/2.0
Via: SIP/2.0/UDP 68.86.118.111:5060;branch=12720924
From: <sip:bobscheller@166.60.255.41>;tag=08e71f4b
To: <sip:bobscheller@166.60.255.41>;tag=08e71f4b
Contact: <sip:s@68.86.118.111:5060;transport=udp>
Call-ID: 4a4af9d005d5b69523478ccd5b01bd8e@68.86.118.111
CSeq: 113 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Event: registration
(no NAT) to 166.60.255.41:5060
********************************************
******SUCCESSFUL REGISTRATION WITH MS-MESSENGER****
Frame 3 (702 bytes on wire, 702 bytes captured)
Ethernet II, Src: 44:45:53:54:42:00, Dst: 00:00:66:61:6b:65
Internet Protocol, Src Addr: 166.50.129.159 (166.50.129.159), Dst Addr:
166.60.255.41 (166.60.255.41)
User Datagram Protocol, Src Port: 1221 (1221), Dst Port: 5060 (5060)
Session Initiation Protocol
Request line: REGISTER sip:bobscheller SIP/2.0
Method: REGISTER
Message Header
Via: SIP/2.0/UDP 166.50.129.159:15329
From: <sip:bobscheller>;tag=d09262e2-fe8f-4350-92ab-74500ad74218
To: <sip:bobscheller>
Call-ID: 55323746-7eaa-4b37-9ca4-d9328f2e158e@166.50.129.159
CSeq: 2 REGISTER
Contact: <sip:166.50.129.159:15329>;methods="INVITE, MESSAGE,
INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
User-Agent: Windows RTC/1.0
Expires: 1200
Event: registration
Allow-Events: presence
Proxy-Authorization: Digest username="",
realm="WCOM",
algorithm="md5", uri="sip:bobscheller",
nonce="2a26142dab7d57ee87a2a7af113d9ca8.1047441261",
response="2210be3b6c2b51959e3d544bcc80f601"
Content-Length: 0
Frame 4 (419 bytes on wire, 419 bytes captured)
Ethernet II, Src: 00:00:66:61:6b:65, Dst: 44:45:53:54:42:00
Internet Protocol, Src Addr: 166.60.255.41 (166.60.255.41), Dst Addr:
166.50.129.159 (166.50.129.159)
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 15329 (15329)
Session Initiation Protocol
Status line: SIP/2.0 407 Proxy Authentication Required
Status-Code: 407
Message Header
v: SIP/2.0/UDP 166.50.129.159:15329;received=166.50.129.159
f: <sip:bobscheller>;tag=d09262e2-fe8f-4350-92ab-74500ad74218
t: <sip:bobscheller>;tag=382ebd3e
i: 55323746-7eaa-4b37-9ca4-d9328f2e158e@166.50.129.159
CSeq: 2 REGISTER
l: 0
Proxy-Authenticate: DIGEST
realm="WCOM",nonce="2a26142dab7d57ee87a2a7af113d9ca8.1047441261"
Frame 7 (713 bytes on wire, 713 bytes captured)
Ethernet II, Src: 44:45:53:54:42:00, Dst: 00:00:66:61:6b:65
Internet Protocol, Src Addr: 166.50.129.159 (166.50.129.159), Dst Addr:
166.60.255.41 (166.60.255.41)
User Datagram Protocol, Src Port: 1222 (1222), Dst Port: 5060 (5060)
Session Initiation Protocol
Request line: REGISTER sip:bobscheller SIP/2.0
Method: REGISTER
Message Header
Via: SIP/2.0/UDP 166.50.129.159:15329
From: <sip:bobscheller>;tag=0fa3a037-aa1e-4583-a193-53daefdab75f
To: <sip:bobscheller>
Call-ID: 900227e0-eaad-4221-a570-921e0e62f7b1@166.50.129.159
CSeq: 2 REGISTER
Contact: <sip:166.50.129.159:15329>;methods="INVITE, MESSAGE,
INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
User-Agent: Windows RTC/1.0
Expires: 1200
Event: registration
Allow-Events: presence
Proxy-Authorization: Digest username="bobscheller",
realm="WCOM", algorithm="md5",
uri="sip:bobscheller",
nonce="b7700c89f79057028b90322d86796384.1047441266",
response="6f3b8678d269830521af0ae47ed542be"
Content-Length: 0
Frame 8 (290 bytes on wire, 290 bytes captured)
Ethernet II, Src: 00:00:66:61:6b:65, Dst: 44:45:53:54:42:00
Internet Protocol, Src Addr: 166.60.255.41 (166.60.255.41), Dst Addr:
166.50.129.159 (166.50.129.159)
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 15329 (15329)
Session Initiation Protocol
Status line: SIP/2.0 100 Trying
Status-Code: 100
Message Header
v: SIP/2.0/UDP 166.50.129.159:15329;received=166.50.129.159
f: <sip:bobscheller>;tag=0fa3a037-aa1e-4583-a193-53daefdab75f
t: <sip:bobscheller>
i: 900227e0-eaad-4221-a570-921e0e62f7b1@166.50.129.159
CSeq: 2 REGISTER
l: 0
Frame 9 (429 bytes on wire, 429 bytes captured)
Ethernet II, Src: 00:00:66:61:6b:65, Dst: 44:45:53:54:42:00
Internet Protocol, Src Addr: 166.60.255.41 (166.60.255.41), Dst Addr:
166.50.129.159 (166.50.129.159)
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 15329 (15329)
Session Initiation Protocol
Status line: SIP/2.0 200 OK
Status-Code: 200
Message Header
v: SIP/2.0/UDP 166.50.129.159:15329;received=166.50.129.159
f: <sip:bobscheller>;tag=0fa3a037-aa1e-4583-a193-53daefdab75f
t: <sip:bobscheller>;tag=3d4d7cc5
i: 900227e0-eaad-4221-a570-921e0e62f7b1@166.50.129.159
CSeq: 2 REGISTER
l: 0
m:
<sip:166.50.129.159:15329>;q=0.500;expires=1200;methods="INVITE,
MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
> ******ASTERISK SIP PACKET **************** > > XXX Need to handle Retransmitting XXX: > REGISTER sip:166.60.255.41 SIP/2.0 > Via: SIP/2.0/UDP 68.86.118.111:5060;branch=12720924 > From: <sip:bobscheller@166.60.255.41>;tag=08e71f4b > To: <sip:bobscheller@166.60.255.41>;tag=08e71f4b > Contact: <sip:s@68.86.118.111:5060;transport=udp> > Call-ID: 4a4af9d005d5b69523478ccd5b01bd8e@68.86.118.111 > CSeq: 113 REGISTER > User-Agent: Asterisk PBX > Expires: 120 > Event: registration > (no NAT) to 166.60.255.41:5060 > > ********************************************Do we not receive anything back at all? Mark
That is basically what I see as well. I do not see any response coming back with the SIP debug. Could it be a problem with the first header line XXX Need to handle Retransmitting XXX: (or is that something generated by asterisk). Just a thought. I am not a protocol expert so I am just curious. I will try to find out what the registration server is in the morning and that may help as well. Bob>From: Masakazu Nakano <n-mack@md.neweb.ne.jp> >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] SIP registration >Date: Thu, 13 Mar 2003 09:21:21 +0900 > > >version is 'Asterisk CVS-03/11/03-09:57:33' > >we can regist to wcom in two ways. > >first. >register => masakazu:password@siptest.wcom.com > >* send REGISTER, but no response from wcom. > > >second. quit * and change the way with number. like this. >register => 9706052:password@siptest.wcom.com > >and REGISTER again. > >in this time,get this result following. > >mack*CLI> >Interface is eth0 >IP Address is 210.194.204.16 >XXX Need to handle Retransmitting XXX: >REGISTER sip:0.0.0.0 SIP/2.0 >Via: SIP/2.0/UDP 210.194.204.16:5060;branch=2d864abf > >From: <sip:@0.0.0.0>;tag=2f12f9af >To: <sip:@0.0.0.0>;tag=2f12f9af >Contact: <sip:@210.194.204.16:5060;transport=udp> >Call-ID: 3ad5682a41c995f064d140150a6b6444@210.194.204.16 >CSeq: 102 REGISTER >User-Agent: Asterisk PBX >Expires: 120 >Event: registration > (no NAT) to 0.0.0.0:0 >XXX Need to handle Retransmitting XXX: >REGISTER sip:166.60.255.41 SIP/2.0 >Via: SIP/2.0/UDP 210.194.204.16:5060;branch=5cf6a49f > >From: <sip:masakazu@166.60.255.41>;tag=2d864abf >To: <sip:masakazu@166.60.255.41>;tag=2d864abf >Contact: <sip:s@210.194.204.16:5060;transport=udp> >Call-ID: 3ad5682a41c995f064d140150a6b6444@210.194.204.16 >CSeq: 103 REGISTER >User-Agent: Asterisk PBX >Expires: 120 >Event: registration > (no NAT) to 166.60.255.41:5060 >XXX Need to handle Retransmitting XXX: >REGISTER sip:0.0.0.0 SIP/2.0 >Via: SIP/2.0/UDP 210.194.204.16:5060;branch=2d864abf > >From: <sip:@0.0.0.0>;tag=2f12f9af >To: <sip:@0.0.0.0>;tag=2f12f9af >Contact: <sip:@210.194.204.16:5060;transport=udp> >Call-ID: 3ad5682a41c995f064d140150a6b6444@210.194.204.16 >CSeq: 103 REGISTER >User-Agent: Asterisk PBX >Expires: 120 >Event: registration > (no NAT) to 0.0.0.0:0 >XXX Need to handle Retransmitting XXX: >REGISTER sip:0.0.0.0 SIP/2.0 >Via: SIP/2.0/UDP 210.194.204.16:5060;branch=2d864abf > >From: <sip:@0.0.0.0>;tag=2f12f9af >To: <sip:@0.0.0.0>;tag=2f12f9af >Contact: <sip:@210.194.204.16:5060;transport=udp> >Call-ID: 3ad5682a41c995f064d140150a6b6444@210.194.204.16 >CSeq: 104 REGISTER >User-Agent: Asterisk PBX >Expires: 120 >Event: registration > (no NAT) to 0.0.0.0:0 >XXX Need to handle Retransmitting XXX: >REGISTER sip:166.60.255.41 SIP/2.0 >Via: SIP/2.0/UDP 210.194.204.16:5060;branch=5cf6a49f > >From: <sip:masakazu@166.60.255.41>;tag=2d864abf >To: <sip:masakazu@166.60.255.41>;tag=2d864abf >Contact: <sip:s@210.194.204.16:5060;transport=udp> >Call-ID: 3ad5682a41c995f064d140150a6b6444@210.194.204.16 >CSeq: 104 REGISTER >User-Agent: Asterisk PBX >Expires: 120 >Event: registration > (no NAT) to 166.60.255.41:5060 > >and keep this trying. >I sometime meets segfault.maybe that cause... > >--- >Masakazu Nakano > >On Tue, 11 Mar 2003 22:34:30 -0600 (CST) >Mark Spencer <markster@digium.com> wrote: > > >> ******ASTERISK SIP PACKET **************** > >> > >> XXX Need to handle Retransmitting XXX: > >> REGISTER sip:166.60.255.41 SIP/2.0 > >> Via: SIP/2.0/UDP 68.86.118.111:5060;branch=12720924 > >> From: <sip:bobscheller@166.60.255.41>;tag=08e71f4b > >> To: <sip:bobscheller@166.60.255.41>;tag=08e71f4b > >> Contact: <sip:s@68.86.118.111:5060;transport=udp> > >> Call-ID: 4a4af9d005d5b69523478ccd5b01bd8e@68.86.118.111 > >> CSeq: 113 REGISTER > >> User-Agent: Asterisk PBX > >> Expires: 120 > >> Event: registration > >> (no NAT) to 166.60.255.41:5060 > >> > >> ******************************************** > > > >Do we not receive anything back at all? > > > >Mark > > > >_______________________________________________ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users_________________________________________________________________ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail
I use Windows Messenger ( I duck as to let the hurled penguins barely miss my head :-) ) and I am able to place and receive calls. So what is the problem you ask??? If I specify a password in the password field of WM I get a Proxy Authentication Error during SIP debug and I am not able to connect until I remove the secret=blah and do not specify a password. Has anyone had this problem before??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030707/64b57ac4/attachment.htm
Not 100% sure here but its probably somthing to do with the fact that MS doesn't support MD5 and I think * makes use of md5 password hashing during authentication.. Maybe you can try adding auth=plaintext to that account in the sip.conf I know this option works in the iax.conf.. Later..> I use Windows Messenger ( I duck as to let the hurled penguins barely > miss my head :-) ) and I am able to place and receive calls. So what is > the problem you ask??? If I specify a password in the password field of > WM I get a Proxy Authentication Error during SIP debug and I am not able > to connect until I remove the secret=blah and do not specify a password. > Has anyone had this problem before??? > > > > >-- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze
I am trying to get SIP registrations to work within Asterisk. From my
snom 200 phone (and on my SJPhone soft client) I can dial via extension.
Example:
To Dial extension 1110 on my asterisk1 server:
I can simply enter SIP:1110@asterisk1 and the call goes through just
like it should.
As I understand it (and I probably don't), once my SIP device has
established communication with the asterisk server, it registers the
device name (in the sip registry) and thus I can dial the phone by
entering:
SIP:snom1@asterisk1
(providing of course snom1 is the context for my sip phone in sip.conf)
In fact I do see the following on the sip console when I make a call
from snom1:
asterisk1*CLI>
-- Registered SIP 'snom1' at 172.16.14.11 port 5060 expires 3600
-- Executing Macro("SIP/snom1-a17d", "oneline|Zap/4") in
new stack
-- Executing Dial("SIP/snom1-a17d", "Zap/4|20") in new
stack
-- Called 4
-- Zap/4-1 is ringing
-- Zap/4-1 is ringing
-- Zap/4-1 is ringing
-- Zap/4-1 is ringing
I haven't found much documentation on sip registration in asterisk, but
I kind of assumed that entering "sip show registry" on the console
would
show me the registrations, but only the following is returned by this
command:
asterisk1*CLI> sip show registry
Host Username Refresh State
Anyone have any ideas?
--
Steve Woolley
ADS Telecom, Inc.
59 Skyline Drive
Suite 1250
Lake Mary, FL 32746
(407)682-6226 x1110
http://www.adstelecom.com
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: Sip read: REGISTER sip:s.s.s.s;transport=UDP SIP/2.0 User-Agent: ATI-RG613/1-1-0_8 From: atrg613test <sip:1234@s.s.s.s>;tag=AABcMQAMRhB0AAxx To: atrg613test <sip:1234@s.s.s.s> Call-ID: AAA4Mwxx@c.c.c.c CSeq: 94 REGISTER Contact: <sip:1234@c.c.c.c> Max-Forwards: 70 Via: SIP/2.0/UDP c.c.c.c;branch=z9hG4bKAQA4Mwxx Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to c.c.c.c : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP c.c.c.c;branch=z9hG4bKAQA4Mwxx From: atrg613test <sip:1234@s.s.s.s>;tag=AABcMQAMRhB0AAxx To: atrg613test <sip:1234@s.s.s.s>;tag=as1966d2fc Call-ID: AAA4Mwxx@c.c.c.c CSeq: 94 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:1234@s.s.s.s> Content-Length: 0 to c.c.c.c:5060 ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/adS12TEAILET3McRAjnlAJ9HE+zxry1+qp2/Y7fqJFh8ea4MFACbB2/E YOLGiZTXMKqBtGCtZqBryD4=3qTK -----END PGP SIGNATURE-----
Hi! I want to accept all the incoming calls (SIP) and redirect them to the good extensions. How do I do that? (Asterisk is acting as a SIP server then... isn't it? Thanks. Best regards, Mireia ___________________________________________________ Yahoo! Messenger - Nueva versi?n GRATIS Super Webcam, voz, caritas animadas, y m?s... http://messenger.yahoo.es
Hello all, I am very new to asterisk I have been using it for sometime but now I want to maintain it myself I have built my own server and am trying to get my cisco ata 186 to register I am having a problem I get this un 17 17:34:54 NOTICE[1116941120]: chan_sip.c:6715 handle_request: Registration from 'sip:dstech3@69.39.68.99' When my 186 logs in I edited sip.conf And have this entry in there. [cisco1] type=context username=dstech4 secret=mysecret qualify=200 ; Qualify peer is no more than 200ms away nat=yes ; This phone may be natted host=dynamic canreinvite=no defaultip=192.168.0.4 disallow=all allow=ulaw allow=alaw allow=g729 Please help me thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040617/2757e02c/attachment.htm