Hi All,
I added the following two lines in ser.cfg.
strip(0);
prefix("99");
The purpose is to strip the leading '0' first then add the prefix '99'. But it just added '99' and didn't strip the '0'. Does the command 'strip' still work?
Thanks,
Bob
Here's the scenario:
As of now, I can only get users to register with my
SER if I don't use database authentication. These
users must also have a static IP address to register.
I am now trying to get a client to register with SER
from behind a NAT. I am using a STUN server address
to help with the NAT situation. When I do this I get
a 483 message.
>From looking at similar posts, the problem is usually
caused by setting uri=="some.domain.com"
I don't have this configuration; in ser.cfg I have the
following:
alias="my.domain.com"
if(uri==myself)
Also, I am running ser-0.8.12 if that makes any
difference.
Does anybody know why I am getting 483 error messages?
Any help is always appreciated.
__________________________________
Do you Yahoo!?
New Yahoo! Photos - easier uploading and sharing.
http://photos.yahoo.com/
The server seems to be slow. It took me a few days of repeated trials to
successfully download the server.
-girija
-----Original Message-----
From: Jan Janak [mailto:jan@iptel.org]
Sent: Thursday, December 18, 2003 5:10 PM
To: Meril Vasantha Fernandopulle
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] request
http://iptel.org/ser/download/
Jan.
On 18-12 00:17, Meril Vasantha Fernandopulle wrote:
> dear sir
>
> I am interesting about SIP Express Router and I try to
> download it several times but I could not download it.
> So please advice me a good way to do it.
> or send me by mail.
> looking forward your reply.
>
> regards
>
> vasantha
>
> __________________________________
> Do you Yahoo!?
> New Yahoo! Photos - easier uploading and sharing.
> http://photos.yahoo.com/
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Hi,
I saw messages complaining about no email addresses (even if there was
one). I saw some postings about this topic and multi-domain, but I
only have one domain, beside this message appear after a while.
I'm going to upgrade to 0.8.12 and redo more tests.
But the main issue in my opinion is the lack of IVR to retrieve the
messages without using a computer.
I'm trying to see if asterisk can support acting as a media server
only. I want to have ser the sip proxy server where all the phones
register with and asterisk being only the voicemail where calls get
forwarded to from ser when there is failure or the user is busy.
I don't know if it's feasable, according to what I see so far it
doesn't look like it support this kind of config. Can you confirm this?
So I'm left with not much option here if I want to have IVR support
for voicemail:
* Try to implement it in sems, what is your estimate on the complexity
of this task, do you have any suggestions on tacking this task ?
Thanks.
Samy.
----- Original Message -----
From: Jan Janak <jan(a)iptel.org>
Date: Thursday, December 18, 2003 3:31 pm
Subject: Re: [Serusers] Using asterisk as a voicemail server with SER
> On 16-12 09:31, Samy Touati wrote:
> > Hi,
> >
> > I installed ser 0.8.11.rc1 along with the sems tied to the same
> version.> Everything installed fine and was running on the same
> host with different
> > port numbers.
> > The instability comes from the fact that the ser configured to
> receive voice
> > calls will just stop answering them after some time.
> > I looked through ethereal and no abnormal messages were generated:
> > I see the invite going to ser-vm on port 5071, if the user is
> busy or not
> > answering, I see the trying/ringing and the caller hears the
> ringing. After
> > a timeout the ringing stops and I get a fast busy with a 408.
> > Now restarting ser and ser-vm puts everything back on track, but
> after a
> > couple of hours I see this behavior again.
>
> Sounds strange. Is there any message in the logs of ser ? Ser should
> log some error messages if it is not working. Could you check
> that ?
>
> Jan.
>
SIP/2.0 200 OK
To: <sip:8646783182@63.86.212.154>;tag=bc5a80577e3d5642
From: <sip:8644679887@63.86.212.157>;tag=322BCA98-25A3
Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)63.86.212.157
CSeq: 101 INVITE
Via: SIP/2.0/UDP 63.86.212.154;branch=z9hG4bKb2f4.035.0
Via: SIP/2.0/UDP 63.86.212.157:5060
Record-Route: <sip:8646783182@63.86.212.154;ftag=322BCA98-25A3;lr>
Timestamp: 1071766157
Contact: John Walter <sip:8646783182@63.86.212.155:5060>
Server: Sipura/SPA2000-1.0.20
Content-Length: 150
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel
Content-Type: application/sdp
v=0
o=- 88599 88599 IN IP4 192.168.0.101
s=-
c=IN IP4 63.86.212.155
t=0 0
m=audio 16404 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
?Úá?ÉF�ó��ó�����������ªûì�
^N�Eå"8@�þÁ?VÔsÀ¨�eÄÄÑBEACK
sip:8646783182@63.86.212.155:5060 SIP/2.0
Record-Route: <sip:8646783182@63.86.212.154;ftag=322BCA98-25A3;lr>
Via: SIP/2.0/UDP 63.86.212.154;branch=0
Via: SIP/2.0/UDP 63.86.212.157:5060
From: <sip:8644679887@63.86.212.157>;tag=322BCA98-25A3
To: <sip:8646783182@63.86.212.154>;tag=bc5a80577e3d5642
Date: Thu, 18 Dec 2003 16:49:17 GMT
Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)63.86.212.157
Max-Forwards: 5
Content-Length: 0
CSeq: 101 ACK
À¨�e?VÔsÄÄ·á}BYE sip:8644679887@63.86.212.157:5060 SIP/2.0
Via: SIP/2.0/UDP 63.86.212.155:5060;branch=z9hG4bK-b76fe29f
From: <sip:8646783182@63.86.212.154>;tag=bc5a80577e3d5642
To: <sip:8644679887@63.86.212.157>;tag=322BCA98-25A3
Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)63.86.212.157
CSeq: 101 BYE
Max-Forwards: 70
Route: <sip:8646783182@63.86.212.154;ftag=322BCA98-25A3;lr>
User-Agent: Sipura/SPA2000-1.0.20
Content-Length: 0
Úá?��������������ªûì�
^N�E":@�þ%?VÔsÀ¨�eÄÄkè¯SIP/2.0 481 Call/Transaction Does
Not Exist
Via: SIP/2.0/UDP 63.86.212.155:5060;branch=z9hG4bK-b76fe29f
From: <sip:8646783182@63.86.212.154>;tag=bc5a80577e3d5642
To: <sip:8644679887@63.86.212.157>;tag=322BCA98-25A3
Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)63.86.212.157
CSeq: 101 BYE
Contact: <sip:8644679887@63.86.212.148>
Server: Sip EXpress router (0.8.12 (i386/linux))
Content-Length: 0
Warning: 392 63.86.212.148:5060 "Noisy feedback tells: pid=20033
req_src_ip=63.86.212.154 req_src_port=5060
in_uri=sip:8644679887@63.86.212.148:5060
out_uri=sip:8644679887@63.86.212.148:5060 via_cnt==0"
-----Original Message-----
From: Andres [ mailto:andres@telesip.net <mailto:andres@telesip.net> ]
Sent: Thursday, December 18, 2003 3:25 PM
To: Darren Sessions; serusers(a)lists.iptel.org
Subject: Re: [Serusers] BYE Message Problem
On Thursday 18 December 2003 15:11, Darren Sessions wrote:
> Sipura Firmware is 1.0.20
>
> We are already in talks with Sipura about this issue as we found a bug for
> them yesterday. It appears though based on the Etheral traces and ngrep
> that the Sipura is sending the BYE messages - but SER is not forwarding
> them along.
Can you post all the messages here(INVITE...to ...BYE)? Otherwise it is
hard
to guess where the problem might be. I'm sure Jiri or Jan can spot the
error
right away.
>
> We are using Ser 8.12 on a Sun Ultra 60 - and as I said before, all other
> instances work flawlessly.
>
> Thanks,
>
> - Darren
>
> -----Original Message-----
> From: Andres [ mailto:andres@telesip.net <mailto:andres@telesip.net> ]
> Sent: Thursday, December 18, 2003 2:58 PM
> To: Darren Sessions; serusers(a)lists.iptel.org
> Subject: Re: [Serusers] BYE Message Problem
>
> On Thursday 18 December 2003 14:50, Darren Sessions wrote:
> > We also have a problem with BYE's not being sent from an on-net call
> > (sipura to sipura).
>
> What firmware version are you using? We have been testing the SPA2000 for
> about 3 weeks now and have not seen this issue.
>
> > -----Original Message-----
> > From: Darren Sessions
> > Sent: Thursday, December 18, 2003 2:32 PM
> > To: 'serusers(a)lists.iptel.org'
> > Subject: RE: Serusers Digest, Vol 8, Issue 16
> > Importance: High
> >
> >
> > Situation:
> >
> > Endpoint is called Party (Sipura SPA2000)
> >
> > Calling Party is routed through from a TDM call and VoIP'd by Cisco 3640
> > router then sent to SER.
> >
> > Endpoint receives call - everything works. When the Sipura hangs up the
> > call, it sends a BYE to ser - but ser does not send the BYE to the
router
> > to disconnect the TDM channel.
> >
> > All other situations result in a completed call with channels released
on
> > the router just fine.
> >
> > After further inspection - we noticed that when the Sipura hangs up and
> > send the BYE to SER, SER responds with this:
> >
> >
> > SIP/2.0 481 Call/Transaction Does Not Exist
> > Via: SIP/2.0/UDP 11.11.111.155:5060;branch=z9hG4bK-b76fe29f
> > From: <sip:8646783182@22.22.222.154>;tag=bc5a80577e3d5642
> > To: <sip:8644679887@33.33.333.157>;tag=322BCA98-25A3
> > Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)33.33.333.157
> > CSeq: 101 BYE
> > Contact: <sip:8644679887@44.44.444.148>
> > Server: Sip EXpress router (0.8.12 (i386/linux))
> > Content-Length: 0
> > Warning: 392 44.44.444.148:5060 "Noisy feedback tells: pid=20033
> > req_src_ip=22.22.222.154 req_src_port=5060
> > in_uri=sip:8644679887@44.44.444.148:5060
> > out_uri=sip:8644679887@44.44.444.148:5060 via_cnt==0"
> >
> >
> > Any ideas?
> >
> > Thanks,
> >
> > - Darren
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
<http://lists.iptel.org/mailman/listinfo/serusers>
Hi All,
Just me, still trying to get authentication working... I have found by
placing a high debug value and starting ser I received some errors I never
saw before as I had no ata box plugged in during the startup. I'm not sure
if this points to why I can't authenticate. When the output says there is
no routing headers, is that the fault of my ATA?
Dec 11 16:07:19 sip ser: 8(4583) SIP Request:
Dec 11 16:07:19 sip ser: 8(4583) method: <REGISTER>
Dec 11 16:07:19 sip ser: 8(4583) uri: <sip:sip.coptalk.com>
Dec 11 16:07:19 sip ser: 8(4583) version: <SIP/2.0>
Dec 11 16:07:19 sip ser: 8(4583) parse_headers: flags=1
Dec 11 16:07:19 sip ser: 8(4583) end of header reached, state=5
Dec 11 16:07:19 sip ser: 8(4583) parse_headers: Via found, flags=1
Dec 11 16:07:19 sip ser: 8(4583) parse_headers: this is the first via
Dec 11 16:07:19 sip ser: 8(4583) After parse_msg...
Dec 11 16:07:19 sip ser: 8(4583) preparing to run routing scripts...
Dec 11 16:07:19 sip ser: 8(4583) entering main route 8(4583) DEBUG :
is_maxfwd_present: searching for max_forwards header
Dec 11 16:07:19 sip ser: 8(4583) parse_headers: flags=128
Dec 11 16:07:19 sip ser: 8(4583) end of header reached, state=8
Dec 11 16:07:19 sip ser: 8(4583) DEBUG: get_hdr_field: <To> [27];
uri=[sip:Rick@sip.coptalk.com]
Dec 11 16:07:19 sip ser:
Dec 11 16:07:19 sip ser: ]
Dec 11 16:07:19 sip ser: 8(4583) get_hdr_field: cseq <CSeq>: <7> <REGISTER>
Dec 11 16:07:19 sip ser: 8(4583) DEBUG: get_hdr_body : content_length=0
Dec 11 16:07:19 sip ser: 8(4583) found end of header
Dec 11 16:07:19 sip ser: 8(4583) DEBUG: is_maxfwd_present: max_forwards
header not found!
Dec 11 16:07:19 sip ser: 8(4583) DEBUG: add_param: tag=1804086518
Dec 11 16:07:19 sip ser: 8(4583) end of header reached, state=29
Dec 11 16:07:19 sip ser: 8(4583) parse_headers: flags=256
Dec 11 16:07:19 sip ser: 8(4583) find_first_route(): No Route headers found
Dec 11 16:07:19 sip ser: 8(4583) loose_route(): There is no Route HF
Dec 11 16:07:19 sip ser: 8(4583) check_self - checking if host==us: 16==9
&& [sip.coptalk.com] == [127.0.0.1]
Dec 11 16:07:19 sip ser: 8(4583) check_self - checking if port 5060
matches port 5060
Dec 11 16:07:19 sip ser: 8(4583) check_self - checking if host==us: 16==13
&& [sip.coptalk.com] == [64.189.165.205]
Dec 11 16:07:19 sip ser: 8(4583) check_self - checking if port 5060
matches port 5060
Dec 11 16:07:19 sip ser: 8(4583) request for registration 8(4583)
check_nonce(): comparing [3fd907c2aa2d6c72cd666f9109e45155055b2d36] and
[3fd907c23a57f75133a55828041f2c9ee7b081c2]
Dec 11 16:07:19 sip ser: 8(4583) pre_auth(): Invalid nonce value received,
very suspicious !
Dec 11 16:07:19 sip ser:
Dec 11 16:07:19 sip ser: '
Dec 11 16:07:19 sip ser: 8(4583) parse_headers: flags=-1
Dec 11 16:07:19 sip ser: 8(4583) check_via_address(64.189.165.206,
64.189.165.206, 0)
Dec 11 16:07:19 sip ser: 8(4583) receive_msg: cleaning up
Dec 11 16:07:19 sip ser: 6(4581) SIP Request:
Dec 11 16:07:19 sip ser: 6(4581) method: <REGISTER>
Dec 11 16:07:19 sip ser: 6(4581) uri: <sip:sip.coptalk.com>
Dec 11 16:07:19 sip ser: 6(4581) version: <SIP/2.0>
Dec 11 16:07:19 sip ser: 6(4581) parse_headers: flags=1
Dec 11 16:07:19 sip ser: 6(4581) end of header reached, state=5
Dec 11 16:07:19 sip ser: 6(4581) parse_headers: Via found, flags=1
Dec 11 16:07:19 sip ser: 6(4581) parse_headers: this is the first via
Dec 11 16:07:19 sip ser: 6(4581) After parse_msg...
Dec 11 16:07:19 sip ser: 6(4581) preparing to run routing scripts...
Dec 11 16:07:19 sip ser: 6(4581) entering main route 6(4581) DEBUG :
is_maxfwd_present: searching for max_forwards header
Dec 11 16:07:19 sip ser: 6(4581) parse_headers: flags=128
Dec 11 16:07:19 sip ser: 6(4581) end of header reached, state=8
Dec 11 16:07:19 sip ser: 6(4581) DEBUG: get_hdr_field: <To> [27];
uri=[sip:Rick@sip.coptalk.com]
Dec 11 16:07:19 sip ser:
Dec 11 16:07:19 sip ser: ]
Dec 11 16:07:19 sip ser: 6(4581) get_hdr_field: cseq <CSeq>: <8> <REGISTER>
Dec 11 16:07:19 sip ser: 6(4581) DEBUG: get_hdr_body : content_length=0
Dec 11 16:07:19 sip ser: 6(4581) found end of header
Dec 11 16:07:19 sip ser: 6(4581) DEBUG: is_maxfwd_present: max_forwards
header not found!
Dec 11 16:07:19 sip ser: 6(4581) DEBUG: add_param: tag=1804086518
Dec 11 16:07:19 sip ser: 6(4581) end of header reached, state=29
Dec 11 16:07:19 sip ser: 6(4581) parse_headers: flags=256
Dec 11 16:07:19 sip ser: 6(4581) find_first_route(): No Route headers found
Dec 11 16:07:19 sip ser: 6(4581) loose_route(): There is no Route HF
Dec 11 16:07:19 sip ser: 6(4581) check_self - checking if host==us: 16==9
&& [sip.coptalk.com] == [127.0.0.1]
Dec 11 16:07:19 sip ser: 6(4581) check_self - checking if port 5060
matches port 5060
Dec 11 16:07:19 sip ser: 6(4581) check_self - checking if host==us: 16==13
&& [sip.coptalk.com] == [64.189.165.205]
Dec 11 16:07:19 sip ser: 6(4581) check_self - checking if port 5060
matches port 5060
Dec 11 16:07:20 sip ser: 6(4581) request for registration 6(4581)
check_nonce(): comparing [3fd907e3929aea9bad25fbd47b3ea062d9215574] and
[3fd907e3929aea9bad25fbd47b3ea062d9215574]
Dec 11 16:07:20 sip ser: 6(4581) get_ha1(): no result for user
'Rick(a)coptalk.com'
Dec 11 16:07:20 sip ser:
Dec 11 16:07:20 sip ser: '
Dec 11 16:07:20 sip ser: 6(4581) parse_headers: flags=-1
Dec 11 16:07:20 sip ser: 6(4581) check_via_address(64.189.165.206,
64.189.165.206, 0)
Dec 11 16:07:20 sip ser: 6(4581) receive_msg: cleaning up
HI all
thanks for all replyes on my question about ser and asterisk..
Now i have that problem when i try to register a user in serweb:
Warning: fopen("/tmp/ser_fifo", "w") - Permission denied in /var/www/html/functions.php on line 206
sipserver.com.br Gerenciamento de usuários
sorry -- cannot open write fifo
We regret but your sipserver.com.br confirmation attempt failed.
Please contact info(a)ipfone.com.br for further assistance.
What can be the problem?
regards
Miklos
----- Original Message -----
From: listas iPfone
To: serusers(a)lists.iptel.org
Sent: Wednesday, December 17, 2003 6:01 PM
Subject: SER AND ASTERISK
Hi All!
I´m trying to run ser as my sip server and asterisk as my pbx in the same machine with acess to pstn.
My goal is to have external users registering in ser ( with my dyndns - sipserver.com.br) and redirect the calls to my estensions in asterisk.
How to do it?
When i run ser, asterisk don´t find my phones anymore.
I have two nic in this machine 192.168.0.31/37:
[root@localhost root]# ser
Listening on
127.0.0.1 [127.0.0.1]:5060
192.168.0.37 [192.168.0.37]:5060
192.168.0.31 [192.168.0.31]:5060
Aliases: localhost:5060 localhost.localdomain:5060
i think that i need to make ser listen to one ip and asterisk to other.
If that is rigth, how can i make ser listen only on 192.168.0.31:5060?
Thanks for all
Miklos
dear sir
I am interesting about SIP Express Router and I try to
download it several times but I could not download it.
So please advice me a good way to do it.
or send me by mail.
looking forward your reply.
regards
vasantha
__________________________________
Do you Yahoo!?
New Yahoo! Photos - easier uploading and sharing.
http://photos.yahoo.com/
##I did try to post this question before, but it
didn't get through.##
Yet another question.
I'm trying to get Windows Messenger to register with
my ser, but I can't get it to work.
Here are my configurations:
export SIP_DOMAIN="mydomain.com"
[Selected portions of ser.cfg]
alias="demoSIP.mydomain.com"
#---module loading---
if (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest
authentication
if
(!www_authorize("demoSIP.mydomain.com", "subscriber"))
{
www_challenge("demoSIP.mydomain.com", "0");
break;
};
save("location");
break;
};
# native SIP destinations are handled
using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not
Found");
break;
};
};
When I run ngrep -n 5060 -d eth0 myAccount I get the
following:
U myMessengerIPaddress:1082 -> mySERipAddress:5060
REGISTER sip:demoSIP.mydomain.com SIP/2.0..Via:
SIP/2.0/UDP myMessengerIPaddress:6946..From:
<sip:myAccount@demoSIP.mydomain.com>;tag=06d7c189-
0a01-40b6-8b60-cdb0163da022..To:
<sip:myAccount@demoSIP.mydomain.com>..Call-ID:
50fab216-4377-4432-962d-e324ad46b035@myMessengerIPaddress..CSeq
: 1 REGISTER..Contact:
<sip:myMessengerIPaddress:6946>;methods="INVITE,
MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL,
NOTIFY, ACK"..User-Agen
t: Windows RTC/1.0..Expires: 1200..Event:
registration..Allow-Events: presence..Content-Length:
0....
#
U mySERipAddress:5060 -> myMessengerIPaddress:6946
SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
myMessengerIPaddress:6946..From:
<sip:myAccount@demoSIP.mydomain.com>;tag=06d7c189-0a01-40b6-8b60-cd
b0163da022..To:
<sip:myAccount@demoSIP.mydomain.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.c1f1..Call-ID:
50fab216-4377-4432-962d-e324ad4
6b035@myMessengerIPaddress..CSeq: 1
REGISTER..WWW-Authenticate: Digest
realm="demoSIP.mydomain.com",
nonce="3fe0cac641974960b96106d8e436ce1c0a3
713c8"..Server: Sip EXpress router (0.8.12
(i386/linux))..Content-Length: 0..Warning: 392
mySERipAddress:5060 "Noisy feedback tells: pid
=3922 req_src_ip=myMessengerIPaddress
req_src_port=1082 in_uri=sip:demoSIP.mydomain.comout_uri=sip:demoSIP.mydomain.com via_cnt==1"....
###
U myMessengerIPaddress:1082 -> mySERipAddress:5060
REGISTER sip:demoSIP.mydomain.com SIP/2.0..Via:
SIP/2.0/UDP myMessengerIPaddress:6946..From:
<sip:myAccount@demoSIP.mydomain.com>;tag=06d7c189-
0a01-40b6-8b60-cdb0163da022..To:
<sip:myAccount@demoSIP.mydomain.com>..Call-ID:
50fab216-4377-4432-962d-e324ad46b035@myMessengerIPaddress..CSeq
: 2 REGISTER..Contact:
<sip:myMessengerIPaddress:6946>;methods="INVITE,
MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL,
NOTIFY, ACK"..User-Agen
t: Windows RTC/1.0..Expires: 1200..Event:
registration..Allow-Events: presence..Authorization:
Digest username="myAccount",
realm="demoSIP.mydomain.com", algorithm="md5",
uri="sip:demoSIP.mydomain.com",
nonce="3fe0cac641974960b96106d8e436ce1c0a3713c8",
response="a03d6b206
d71f54674b5a48497b3e8aa"..Content-Length: 0....
###
U mySERipAddress:5060 -> myMessengerIPaddress:6946
SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
myMessengerIPaddress:6946..From:
<sip:myAccount@demoSIP.mydomain.com>;tag=06d7c189-0a01-40b6-8b60-cd
b0163da022..To:
<sip:myAccount@demoSIP.mydomain.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.c1f1..Call-ID:
50fab216-4377-4432-962d-e324ad4
6b035@myMessengerIPaddress..CSeq: 2
REGISTER..WWW-Authenticate: Digest
realm="demoSIP.mydomain.com",
nonce="3fe0cac641974960b96106d8e436ce1c0a3
713c8"..Server: Sip EXpress router (0.8.12
(i386/linux))..Content-Length: 0..Warning: 392
mySERipAddress:5060 "Noisy feedback tells: pid
=3932 req_src_ip=myMessengerIPaddress
req_src_port=1082 in_uri=sip:demoSIP.mydomain.comout_uri=sip:demoSIP.mydomain.com via_cnt==1"....
I followed the documentation in seruser.pdf regarding
the messenger bug.
Any help is appreciated.
Thank you for your time.
__________________________________
Do you Yahoo!?
New Yahoo! Photos - easier uploading and sharing.
http://photos.yahoo.com/
Yet another question.
I'm trying to get Windows Messenger to register with
my ser, but I can't get it to work.
Here are my configurations:
export SIP_DOMAIN="mydomain.com"
[Selected portions of ser.cfg]
alias="demoSIP.mydomain.com"
#---module loading---
if (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest
authentication
if
(!www_authorize("mydomain.com", "subscriber")) {
www_challenge("mydomain.com", "0");
break;
};
save("location");
break;
};
# native SIP destinations are handled
using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not
Found");
break;
};
};
When I run ngrep -n 5060 -d eth0 myAccount I get the
following:
U WindowsMessengerIP:1061 -> mySERip:5060
REGISTER sip:demoSIP.mydomain.com SIP/2.0..Via:
SIP/2.0/UDP mySERip:6946..
From:
<sip:myAccount@demoSIP.mydomain.com>;tag=57bd73f9-1d75-4da2-a0d4
-d3ed0acb4da3..To:
<sip:myAccount@demoSIP.mydomain.com>..Call-ID:
ec1d143e-
26a9-4144-9b6f-58af38b522f7@67.70.231.221..CSeq: 1
REGISTER..Contact: <sip:
67.70.231.221:6946>;methods="INVITE, MESSAGE, INFO,
SUBSCRIBE, OPTIONS, BYE
, CANCEL, NOTIFY, ACK"..User-Agent: Windows
RTC/1.0..Expires: 1200..Event:
registration..Allow-Events:
presence..Content-Length: 0....
#
U mySERip:5060 -> WindowsMessengerIP:6946
SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
WindowsMessengerIP:6946..
From:<sip:myAccount@demoSIP.mydomain.com>;tag=57bd73f9-1d75-4da2-a0d4-d3ed0acb4da3
To:
<sip:myAccount@demoSIP.mydomain.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58
.c1f1..Call-ID:
ec1d143e-26a9-4144-9b6f-58af38b522f7@67.70.231.221..CSeq:
1
REGISTER..WWW-Authenticate: Digest
realm="mydomain.com", nonce="3fe0b9858a
bc49c954578b998d4b1f709a6160f6"..Server: Sip EXpress
router (0.8.12 (i386/l
inux))..Content-Length: 0..Warning: 392 mySERip:5060
"Noisy feedback
tells: pid=3495 req_src_ip=67.70.231.221
req_src_port=1061
in_uri=sip:demoSIP.mydomain.comout_uri=sip:demoSIP.mydomain.com via_cnt==1"....
Does anybody know why the communication stops after
the first 401?
Any help is appreciated.
Thank you for your time.
=====
Asterisk is my lover, and IAX2 is our scented lubricant
__________________________________
Do you Yahoo!?
New Yahoo! Photos - easier uploading and sharing.
http://photos.yahoo.com/
