Hi,i'm new to SIP. I want to integrate my existing Instant Messenger to SIP server. I mean i have to write some thing so that our IM users can chat with the other users using SIP. But i don't know SIP. Now i'm going through SIP docs. I'm getting the architecture, but i'm unable to understand what to write for the above problem. Whether we need to write one SIP enabled server or we have to use third party server. So please let me know what actually i need to do to get out from this problem. I'll be very thankful to you if you show me a path. thanks in advance. cheersg.sreedhar reddy
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Hello Jiri,
Of course, I meant the non-paranoic meaning, that is forcing the subsequent
requests to go through the proxy.
The question is related to the FCP module I'm trying to complete. In order to
close the previously opened ports, the module needs to see BYE and CANCEL
messages. When adding record-routing header from a natted proxy, this address
must be changed to a public one whenever the message is for the public Internet.
Is my understanding correct?
My other concern with Record Routing is whether this translation for the request
involves translating the RR field back to the one of the natted proxy, when the
response hits the proxy.
Regards,
Jaime
Jiri Kuthan <jiri(a)iptel.org> on 21/03/2003 17:28:48
To: Jaime GILL/EN/HTLUK@HTLUK
serusers(a)lists.iptel.org
cc:
Subject: Re: [Serusers] Routing all SIP traffic through the proxy
At 12:29 PM 3/21/2003, jaime.gill(a)orange.co.uk wrote:
>Hi,
>
>Not sure if someone has asked this before. Is there any way to route all SIP
>messages through SER without using record_route()?
>
>Jaime
I'm not sure what it means "all SIP messages". In a paranoid understanding, that
would include interception of iptel.org messages by someone else.
In general, there are two options: setting an outbound proxy in phones and
forcing subsequent requests hit a proxy through rr-ing.
-Jiri
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Do any body knows if there is a push to talk solution to implement in my web
in orther to call to my ATA186 when somebody push the talk .... I read there
is a solution ????
Esteban
Hi,
I want all users who have accounts on host A to have access to ser which
runs on
host B, (without having to add them again to the mysql database on host B.)
I learned that Radius Digest Authentication cannot be used because it
needs the
Radius server to know the cleartext password of the user. Is there any
other
method that I can use in this situation?
Nimal R.
Hi,
I would like to create more than one virtual line, like FWD,
deltathree, iptel all to a single ATA box.
Can anyone here please guide how can I do with ser.
Regards,
Kannaiyan
Hi,
I have built ser and modules from the tar.gz file available on CVS.
Authentication using auth_db works fine, but auth_radius results in
a segmentation fault when I try to register.
The debug log (with debug=9) says:
Mar 22 16:21:32 voip ser: SIP Request:
Mar 22 16:21:32 voip ser: method: <REGISTER>
Mar 22 16:21:32 voip ser: uri: <sip:voip.pdn.ac.lk>
Mar 22 16:21:32 voip ser: version: <SIP/2.0>
Mar 22 16:21:32 voip ser: parse_headers: flags=1
Mar 22 16:21:32 voip ser: Found param type 232, <branch> =
<z9hG4bK3378053857>; state=16
Mar 22 16:21:32 voip ser: end of header reached, state=5
Mar 22 16:21:32 voip ser: parse_headers: Via found, flags=1
Mar 22 16:21:32 voip ser: parse_headers: this is the first via
Mar 22 16:21:32 voip ser: After parse_msg...
Mar 22 16:21:32 voip ser: preparing to run routing scripts...
Mar 22 16:21:32 voip ser: DEBUG : is_maxfwd_present: searching for
max_forwards header
Mar 22 16:21:32 voip ser: parse_headers: flags=128
Mar 22 16:21:32 voip ser: DEBUG: add_param: tag=2520454554
Mar 22 16:21:32 voip ser: end of header reached, state=29
Mar 22 16:21:32 voip ser: DEBUG: get_hdr_field: <To> [44];
uri=[sip:nimalr@voip.pdn.ac.lk]
Mar 22 16:21:32 voip ser: DEBUG: to body [<sip:nimalr@voip.pdn.ac.lk>]
Mar 22 16:21:32 voip ser: get_hdr_field: cseq <CSeq>: <0> <REGISTER>
Mar 22 16:21:32 voip ser: DEBUG: is_maxfwd_present: value = 10
Mar 22 16:21:32 voip ser: parse_headers: flags=4096
Mar 22 16:21:32 voip ser: DEBUG: get_hdr_body : content_length=0
Mar 22 16:21:32 voip ser: found end of header
Mar 22 16:21:32 voip ser: pre_auth(): Credentials with given realm not
found
Mar 22 16:21:32 voip ser: ERROR: fifo_server fgets failed: Illegal seek
Mar 22 16:21:33 voip last message repeated 3 times
Mar 22 16:21:33 voip ser: INFO: signal 15 received
Mar 22 16:21:33 voip ser: INFO: signal 15 received
My configuration is :
.........
.........
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/auth_radius.so"
..........
..........
if (uri=~"voip.pdn.ac.lk") {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!radius_www_authorize("voip.pdn.ac.lk")) {
# if (!www_authorize("voip.pdn.ac.lk","subscriber")) {
www_challenge("voip.pdn.ac.lk", "0");
break;
};
save("location");
break;
};
........
........
Any help for solving the problem would be appreciated. Thanks.
Nimal R.
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Folks,
We found that in some cases it is necessary to have a function which
will save contact without sending reply to the UA. For example, it is
useful when there is a need to do stateless forwarding of the REGISTER
request, but at the same time save it in the local registration
database (sending reply in this case will disable any retransmissions
that might be necessary to ensure that the request actually reached the
destination). Attached patch provides necessary functionality, it would
be nice to have it included into the next release.
Thanks!
-Maxim
Hi,
When we have a situation in which a PC changes IP addresses (like on
dialup), we end up with multiple registrations (see below). Is it possible
to configure SER so that it only has the latest registration?
[root@maui ser]# serctl ul show rvilla
<sip:rvilla@200.58.193.72:5060;transport=udp>;q=0.00;expires=2773
<sip:rvilla@200.58.203.236:5060;transport=udp>;q=0.00;expires=3194
<sip:rvilla@200.58.203.61:5060;transport=udp>;q=0.00;expires=3544
On a related note, the registrar README says:
"Name: lookup
Params: table - Name of table that should be used for the lookup
Desc: The functions extracts username from Request-URI and tries to find all
contacts
for the username in usrloc. If there are no such contacts, -1 will be
returned.
If there are such contacts, Request-URI will be overwritten with the contact
that
has the highest q value and optionally the rest will be appended to the
message
(depending on append_branches parameter value)."
....but as you can see all 3 contacts have the same "q value" (whatever that
stands for).
How does one resolve such an issue?
Thanks,
Ricardo