Ash,
You can set up the SIP Proxies to use the same nonce.
Adrian
>>>>>>>>>>
We are having a separate issue now with our ATA - the ATA (a cheap
generic brand) we use as our CPE, won't issue two digest authentication
response unlike X-Lite. Seems like we will be forced to use
authentication at B2BUA level only after all.
Anyone knows any ATA positively supports sending two separate
authentication digests for two separate nonce in same INVITE?
Ovidiu Sas wrote:
> I don't think it is reported.
>
> On Fri, Apr 25, 2008 at 6:36 AM, Iñaki Baz Castillo <ibc at
in.ilimit.es> wrote:
>
>> El Friday 25 April 2008 09:57:09 Ovidiu Sas escribió:
>>
>>
>>> Yes, it should work but it doesn't.
>>>
>> > I remember trying this long time ago (using openser 1.0 and
chaining
>> > several openser proxies) and I hit the same issue: openser is
not able
>> > to deal properly with multiple Proxy-Authorization headers.
>>
>> So this is a bug (IMHO an important bug), isn't? is it reported?
>>
>>
>> --
>>
>>
Hi Gaoqiang,
MSRPRelay is not dependent on OpenSER. You can set it up separately
based on the instructions available at http://msrprelay.org.
You only need a client with support for the MSRP relay extension, the
SIP session can be established by any version of OpenSER.
I believe you refer to the MSRP chat server when you say "session
switch". MSRP chat server is work in progress, no software has been
released yet, we plan to release a first version this summer.
Regards,
Adrian
>>>
Hi list,
Could anyone here tell me is the openser 1.3.x version support
MSRP(e.g. MSRPRelay)? Is there any document describe how to configure
the openser work with MSRP(MSRPRelay) as a message session switch.
Does anyone have successful experience configure openser to work with
the MSRP Relay.
Thanks a lot.
--
Gaoqiang Qian
Automation System and Technology
Helsinki University Of Technology
GSM: +358415058138
Hi,
How can I force OpenSER to send all SIP messages to source port of the UA?
That is, the same port as userloc lists in
MySQL/openser/location/received column?
For some reason it only sends the Invite message to that port, but
then all Info,Ack,Ok messages go to the port found in VIA...which the
UA doesn't receive because it's behind NAT.
I've tried force_rport(); without any success, but I must admit that
I'm not sure I used it in the right route.
My config is previously posted here:
http://www.voipuser.org/forum_topic_12946.html
Any ideas?
> Hi all,
>
> Is there an easy way to disable RTPProxy for certain calls eg. based
> on called number?
> Am using the ser.cfg from
> http://www.iptel.org/ser/howtos/optimizing_the_use_of_rtp_proxy.
>
> For whatever reason all calls are passing the RTPProxy.
> - I would like to disable the RTPProxy for calls where I know the
> destination is able to handle NAT.
> - For calls where I know both UAT are behind the same NAT.
> As soon as I update the realm of those clients with whatever
> value beside the ser IP adress, those useres cannot login anymore.
>
> Is this sufficient information - do you need the ser.cfg?
>
> Thanks
> Patrick
>
>
>
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Hello everyone,
I have setup the openser server with the very basic default
configuration. Now I have to configure my openser server to connect
with a xmpp envoironment on another host with a purpose of piloting
the SIP SIMPLE - XMPP IOP.
So does anyone here have successful experience of configure openser to
connect with XMPP on different hosts? Or could someone provide me some
instruction how to configure it.
Thanks a lot in advance
--
Gaoqiang Qian
Automation System and Technology
Helsinki University Of Technology
GSM: +358415058138
Hello,
I am trying to make a design like below to work.
X-Lite ----- OpenSER ----- Asterisk ----->(PSTN Calls)
X-Lite registers with OpenSer and PSTN calls are routed through Asterisk
from OpenSER. When a call is sent to Asterisk, Asterisk tries to
authenticate the user on X-Lite. I maintain same username and password
for both OpenSER and Asterisk.
Now when an INVITE from X-Lite hits OpenSER, it goes through the
following script and is asked for Proxy Authorization:
if (!proxy_authorize("","subscriber")) {
proxy_challenge("","0");
exit;
}
When I dial a PSTN number from X-Lite, X-Lite at some point, ends up
sending two Digests (one for OpenSER and one for Atserisk) in same
INVITE but gets stuck with Proxy Authorization failure (from OpenSER).
If I take off the above proxy_authorize section from OpenSER script,
everything works fine.
Can anyone suggest a solution to this.
Thanks in advance.
U 2008/04/23 13:28:42.314669 110.110.110.110:26986 -> 120.120.120.120:5060
INVITE sip:6048484848484@sip.dummydomain.com SIP/2.0.
Via: SIP/2.0/UDP
172.16.40.14:26986;branch=z9hG4bK-d87543-886860777744b40e-1--d87543-;rport.
Max-Forwards: 70.
Contact: <sip:1274229212@110.110.110.110:26986>.
To: "6048484848484"<sip:6048484848484@sip.dummydomain.com>.
From: "1274229212"<sip:1274229212@sip.dummydomain.com>;tag=7d74b26b.
Call-ID: ZjIyNDQzOWIxZTM2MWJjMTgzNmE1YWE3ZDY1M2RjZWE..
CSeq: 3 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO.
Content-Type: application/sdp.
Proxy-Authorization: Digest
username="1274229212",realm="asterisk",nonce="01d3972c",uri="sip:6048484848484@sip.dummydomain.com",response="ff9058f8ea89c55d0b110d4eccf27e9c",algorithm=MD5.
Proxy-Authorization: Digest
username="1274229212",realm="sip.dummydomain.com",nonce="480ee655da312e1c8f977cae40a747d26f7e9c5f",uri="sip:6048484848484@sip.dummydomain.com",response="361700cce632c00ff70ede5e5126c6ac",algo
rithm=MD5.
User-Agent: X-Lite release 1011s stamp 41150.
Content-Length: 333.
.
v=0.
o=- 9 2 IN IP4 172.16.40.14.
s=CounterPath X-Lite 3.0.
c=IN IP4 172.16.40.14.
t=0 0.
m=audio 45136 RTP/AVP 0 101.
a=alt:1 3 : gpvy8HMY JXNZYRF+ 172.16.40.14 45136.
a=alt:2 2 : 8S3XPC3M 6q9Z76Pq 192.168.38.1 45136.
a=alt:3 1 : rISpUdBc PRYZ7B/8 192.168.23.1 45136.
a=fmtp:101 0-15.
a=rtpmap:101 telephone-event/8000.
a=sendrecv.
U 2008/04/23 13:28:42.314910 120.120.120.120:5060 -> 110.110.110.110:26986
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
172.16.40.14:26986;branch=z9hG4bK-d87543-886860777744b40e-1--d87543-;rport=26986;received=110.110.110.110.
To:
"6048484848484"<sip:6048484848484@sip.dummydomain.com>;tag=058e81974577b8ca6a831d36c0f6fe25.d85d.
From: "1274229212"<sip:1274229212@sip.dummydomain.com>;tag=7d74b26b.
Call-ID: ZjIyNDQzOWIxZTM2MWJjMTgzNmE1YWE3ZDY1M2RjZWE..
CSeq: 3 INVITE.
Proxy-Authenticate: Digest realm="sip.dummydomain.com",
nonce="480ee6560e7141c28e990448575d0918ce86a82d".
Server: OpenSER (1.3.1-notls (i386/linux)).
Content-Length: 0.
Hi list,
Could anyone here tell me is the openser 1.3.x version support
MSRP(e.g. MSRPRelay)? Is there any document describe how to configure
the openser work with MSRP(MSRPRelay) as a message session switch.
Does anyone have successful experience configure openser to work with
the MSRP Relay.
Thanks a lot.
--
Gaoqiang Qian
Automation System and Technology
Helsinki University Of Technology
GSM: +358415058138
Hello:
I have a case where multiple INVITEs crash Polycom phones. I'm wondering if anyone else has seen this and if so how you addressed the problem. We are using SER0.9.7pre3. We have Polycom 550s and Cisco 7960s running SIP 2.2.2.0084 and 7.3 respectively. These phones are configured for simultaneous/parallel ringing which we call a ring group. In a ring group the same 5-digit extension appears on each phone. So in this case the AOR has three contact addresses in the location table.
When a call arrives for the AOR all three phones ring and if you answer one of them at least one, possible all, Polycom phones will reboot. Ngrep traces on the proxy show multiple invites being sent to each phone each with a different branch tag in the Via header. I'm not sure what is happening so any insight anyone can share will be appreciated.
Polycom admits the reboot is a bug but believes the multiple invites is a configuration problem in our proxy. My questions is where and why does this only cause a problem for Polycom phones?
Thanks,steve
Senior Network Engineer,
Information Systems and Computing
Networking and Telecommunications , Suite 221A /6228
University of Pennsylvania
Voice:215-573-8396
FAX:215-898-9348