Hi Everybody
I'm working on openser + freeradius. Recently Installed both .
I had a issue below onw, actual what this means...
By Typing '" radtest 444 444 192.168.2.55 1812 radiustest " its shows below
one.
Sending Access-Request of id 236 to 192.168.2.55 port 1812
User-Name = "444"
User-Password = "444"
NAS-IP-Address = 255.255.255.255
Re-sending Access-Request of id 236 to 192.168.2.55 port 1812
User-Name = "444"
User-Password = "444"
NAS-IP-Address = 255.255.255.255
--
Thanks and Regards with cheers
Sunkara Ravi Prakash (Voip Developer)
Hyperion Technology
www.hyperion-tech.com
Hi, I compile SER 0.9.6 from source on a computer running Fedora Core 4.
I suppose make all, make install all go well as I can't see any warning or
error.
When I start ser, it seems normal but I can't find the pid with ps.
[root@localhost ~]# serctl start
Starting SER : started pid(18726)
[root@localhost ~]# ser
Listening on
udp: 127.0.0.1 [127.0.0.1]:5060
udp: 192.168.1.4 [192.168.1.4]:5060
udp: 192.168.1.17 [192.168.1.17]:5060
tcp: 127.0.0.1 [127.0.0.1]:5060
tcp: 192.168.1.4 [192.168.1.4]:5060
tcp: 192.168.1.17 [192.168.1.17]:5060
Aliases:
tcp: localhost:5060
tcp: localhost.localdomain:5060
udp: localhost:5060
udp: localhost.localdomain:5060
[root@localhost ~]# serctl stop
Stopping SER : /usr/local/sbin/serctl: line 813: kill: (18726) - No such
processstopped
[root@localhost ~]#
How can I fix this so ser run normally?
Thank you.
--
Best regards,
Linh Pham
I was making some fax tests using a Cisco AS5300, OpenSER 1.0.1 and a
Grandstream HandyTone 486. Everything results fine when I previously
make the same test directly from the HandyTone to the AS5300, however
when I put OpenSER in the middle the fax connection could not be
established. Checking captures at both ends, I noticed the HandyTone was
not receiving the OK response to its RE-INVITE request, but the AS5300
was sending it. Looking a capture in the OpenSER, I suppose it didn't
send the OK response from the AS5300 to the HandyTone because the format
of the Via Header used by the AS5300. Look at the captures below, the
AS5300 sends the Via header in a comma-separated format instead using a
line for each one. Is there a way to workaround this?
I think my best option is to modify the OpenSER code to support both
formats. Is this something that somebody already does it? If the answer
it is not, where can I start?
I'll appreciate any help.
Dioris Moreno
Cisco AS5300 IP Address: AAA.BBB.CCC.DDD
OpenSER IP Address: WWW.XXX.YYY.ZZZ
Grandstream HT486 IP Address: FFF.GGG.HHH.III
Calling Party Number (PSTN): UserA
Called Party Number (VoIP): UserB
Message From OpenSER to Cisco AS5300
Session Initiation Protocol
Request-Line: INVITE sip:UserA@AAA.BBB.CCC.DDD:5060 SIP/2.0
Method: INVITE
Resent Packet: False
Message Header
Record-Route: <sip:WWW.XXX.YYY.ZZZ;ftag=266f5f157860b9ca;lr=on>
Via: SIP/2.0/UDP WWW.XXX.YYY.ZZZ;branch=z9hG4bK6a82.677fb8e1.0
Via: SIP/2.0/UDP
FFF.GGG.HHH.III:50841;branch=z9hG4bK90436408698e6881
From: <sip:UserB@WWW.XXX.YYY.ZZZ>;tag=266f5f157860b9ca
SIP from address: sip:UserB@WWW.XXX.YYY.ZZZ
SIP tag: 266f5f157860b9ca
To: <sip:UserA@AAA.BBB.CCC.DDD>;tag=D33F86B0-94F
SIP to address: sip:UserA@AAA.BBB.CCC.DDD
SIP tag: D33F86B0-94F
Message From Cisco AS5300 to OpenSER
Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Status-Code: 200
Resent Packet: False
Message Header
Via: SIP/2.0/UDP
WWW.XXX.YYY.ZZZ;branch=z9hG4bK6a82.677fb8e1.0,SIP/2.0/UDP
FFF.GGG.HHH.III:50841;branch=z9hG4bK90436408698e6881
From: <sip:UserB@WWW.XXX.YYY.ZZZ>;tag=266f5f157860b9ca
SIP from address: sip:UserB@WWW.XXX.YYY.ZZZ
SIP tag: 266f5f157860b9ca
To: <sip:UserA@AAA.BBB.CCC.DDD>;tag=D33F86B0-94F
SIP to address: sip:UserA@AAA.BBB.CCC.DDD
SIP tag: D33F86B0-94F
I got a big problem here that I don't know how to fix.
An OpenSER system forwards calls from phones to Asterisk boxes. Up until now we had been using the dispatcher to evenly distribute calls to the Asterisk boxes. However, when someone transfers a call, using the dispatcher, the call can go to a different Asterisk box, which breaks the transfer.
I need to find some way to make OpenSER route transfers to the same Asterisk system. I'm not sure how to do this. If there was some way to, upon an INVITE, check and see if there was already a call in progress for that user and then send it to the same Asterisk system. Would stateful processing help in this way? Can stateful processing 'remember' that there's a call already in progress like this, or does the stateful simply relate to stateful processing of a single call from INVITE to OK/ACK?
Doug.
Hi all,
i have 2 questions for the forum:
1) What can I do to simulate a failure message to understand if
openser store it into database, through acc_db_request?
2) Have I any chance to get information from reply messages, like
storing some in db or something else? I know I can't use acc and avp
module in on_reply route block, so any suggestion?
Thanks in advance
Davide
This is not openser specific, but can I just check my understanding of
the SIP RFC? Given an INVITE where the Request-URI is different from the
To field, a UAS should route the INVITE based on the Request-URI,
correct? The reason I ask it that I am having problems integrating
openser with a proprietry SIP proxy as the proprietry proxy routes
INVITES based on the To field.
Jeff
Hi folks,
I have implemented the openser.cfg with forwarding and voicemail features completely. Only problem is,
if the user is forwarding the call on a pstn number how should he be billed, as the one who has forwarded is accountable and not the one who called.
I have implemented forwarding using avps serial forking method. I've read about multiple leg accounting and was not successful in understanding it properly. The docs only mention to set the parametrs and after I set the params, I get a n/a in src_leg and dst_leg column. Can someone pls give me examples of successfully using multiple leg accounting in openser.cfg.
Is it like, the call leg accounting will only take place if there is a 3XX response from the server. I am asking this as my cfg does not produce any 302 response, but creates a brand new Invite request for the forwarded call. Can someone pls tell me more about multi-leg accounting.
Also I read about the diversion module which can be used in the forwarding scenarios.
But I added the diversion header and forwarded the call to Cisco gateway, it gave a 400 Bad Request reponse saying 'Malformed CC-Diversion/Diversion/CC-Redirect Header'. Can someone explain what does this mean.
It will help me a lot as this is the only thing I am left with.
Thanks a lot in advance.
w/regards,
Jayesh
---------------------------------
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Has anyone seen any good OpenSER documentation, or know of any training courses in North America that relate directly to OpenSER? I've been involved with OpenSER for about 6 months now and the available documentation has remained about the same - REALLY BAD.
Thanks,
Doug
Hello.
Sorry for this stupid question.
Can someone explain to me how the RPID parameter is used?. Can i
use this parameter for other purposes ?
Hope thank someone can help me.
Thanks in advance.
Regards,
Ricardo.-
