- sr-users - kamailio.org

[Users] SER SIP-T
by Piccari Gianluca 21 Nov '05

21 Nov '05

Hi all, i'd like to know if SER supports SIP - INFO method and then SIP-T (I haven't found Doc. about this matter). Can u help me to find any Documens about the objec above mentionated? Thanks Gianluca

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[Users] Pseudo vars in exec_msg() or exec_dset()
by Patrick Jordan-Smith 21 Nov '05

21 Nov '05

Hi, Is there any way in which I could use pseudo variables in one of the exec functions? It would be incredibly useful! Thanks, Patrick -- Patrick Jordan-Smith Network Operations Mercury Telecommunications Ltd / Quicksilver Internet voice +64-9-916-0300 / +64-27-5900595 fax +64-9-916-0301

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[Serusers] Contact field in SIP HF between asterisk + ser
by harry gaillac 21 Nov '05

21 Nov '05

___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] how to restrict registration
by Francesco Fondelli 21 Nov '05

21 Nov '05

Hi all, I would like to restrict registration based on IP in Contact, something like: foo(a)bar.com --> allowed contact IP are 192.168.1.10 or 192.168.1.15 boo(a)bar.com --> allowed contact IP are 192.168.1.20 or 192.168.1.30 I tried with permissions module: if (!allow_register("register.allow", "register.deny")) { sl_send_reply("403", "Permission denied"); break; }; and # content of register.allow "^sip:foo@.*$" : "^sip:.*192\.168\.1\.… [View More]

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Re: [Users] MySQL 2-way replication with SER
by Marcello Lupo 21 Nov '05

21 Nov '05

Hi, what you mean for 2-way replication in mysql?? The replica as A-->B-->A or the clustering of mysql 5?? I see limitation in both situation: In A-->B-->A you have trouble in case of failure of one server to get again a working environment being sure that all is syncrhonized well. If you can suggest a good way to accomplish it will be welcome. I'm trying to do the same thing. As i know (may be i'm wrong and i will be happy to be) In clustering you do not have support for … [View More]

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[Serusers] ser.cfg's interpretation of return codes from modules
by Ramin Nikaeen 21 Nov '05

21 Nov '05

Valued Colleagues, Can someone tell me how ser.cfg interprets the integer code returned by module functions? I want to be able to distinguish, in ser.cfg, between success and failure of a module function's execution when invoked from ser.cfg. As long as I return "1" from the module function to signalize success, things work fine but when I try returning "0" or "-1" to indicate failure of the function, ser can not complete the call and in a way appears to stop processing. My UA … [View More]

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[Users] Voicemail Failure Routes
by Daryl Sanders 21 Nov '05

21 Nov '05

What is best way to make SER/OpenSER route to a different feature server if the first one doesn't respond? Should this be done with an additional failure route? Can either SER or OpenSER do SRV queries? I am basically trying to get a little extra redundancy in place for my VM server. They will be replicating between each other, but if one goes down I need to make sure SER can still forward calls to a VM server that is still alive. My current ser.cfg is based off of the onsip call-forwarding … [View More]

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[Users] OpenSER Capability query
by Rahul Iyer 21 Nov '05

21 Nov '05

Hello Group, I am a new user to this software. I have a couple of questions, and would really appreciate if someone can take time to clarify. 1. Is OpenSER the same as SER? If no, is there a huge difference? 2. Does any of the above have any media processing capabilities? Eg, say there are 2 SIP endpoints one can talk only G711 and the other only G729. Can SER/OpenSER act as a Gateway for these 2? I would assume no, but are there any modules that can enable this? Thanks

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[Users] Openser + sems
by Loide Mara 21 Nov '05

21 Nov '05

Hi, could anybody tell me how to use the function load_avp(). I'm trying to use openser with sems... Currently I am using as "load_avp("ruri", "email",0)" but the proxy gives an error. I am using the openser-0.9.4. But I think that this function doesn't anymore exist in this version... Could I use other function? Best regards []'s -------------------------------------------------- Loide Mara Verdes de Jesus 6º período - Ciência da Computação / UFAM Laboratório VoIP - UFAM

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[Serusers] ser-0.9.4 radius accounting
by han yibing 21 Nov '05

21 Nov '05

Hi all: i install the SER-0.9.4 with radius accounting ,but it is can't work ,there are error log information in /var/log/messages as follow: rc-avpair_new:unknow attribute 4 ERROR:acc_rad_request:radius-ing failed. i had configured the right radius server at /etc/radiusclient-ng/radiusclient.conf and servers file. please help me . thanks ___________________________________________________________ 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 http://cn.mail.yahoo.com

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Re: [Serusers] Get alias for user
by Voipers Portugal 20 Nov '05

20 Nov '05

But you can rewrite the user then.. if it comes user(a)ser.com and you rewrite to user_number(a)asterisk.com doesn't it work? Jose Simoes 2005/11/20, Pol <kuroki(a)gmail.com>: > Of course i do rewritehost to forward to asterisk but, voicemail.conf > is configured to accept only numbers not users like pnovell@domain, so > before rewritehost i should overwrite my TO header ( pnovell ) for ( > user alias which is always numerical 17940 ). > understood? > Thx Users! > &… [View More]

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[Serusers] Urgent NAT question: mediaproxy,rtp or STUN ?
by Adrian Georgescu 20 Nov '05

20 Nov '05

Antonello, There is no better solution for all possible cases. The solutions have differences explained in the readme files for example RTP proxy can take more load per server, Media Proxy can distribute load on multiple boxes based on DNS SRV records and STUN can help your Stun enabled clients to traverse the NAT in majority of cases but not all. So a combination between STUN plus one of the relay solutions will cover all cases. Configuring per session when to use Stun or a media relay … [View More]

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[Serusers] Multiple entries in Location Table with Same Call-Id.
by Ramu Yadav 20 Nov '05

20 Nov '05

Hi All, With Some User-Agents we are facing multiple entries in "location" Table. The main problem is each and every time UA sending register packet with different port number in the contact uri. I am here with attaching the location table values as well as traces. Thanks -- Ramu Yadav

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[Users] How Effective is STUN?
by Tavis P 20 Nov '05

20 Nov '05

I'm trying to find some statistics as to what the ratio of Cone vs Symmetric NAT solutions deployed in the world are, has anyone done some research into this? I'm curious what percentage of users in certain demographics (broadband clients, for example) i can expect to be serviced using STUN alone, so i can come up with some figure to help me build out my network Even just some anecdotal information of peoples experiences would be very useful Tavis

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Re: [Users] Problem with the registration of X-PRO client with Openser
by Lokesh Kumar 20 Nov '05

20 Nov '05

----- Original Message ----- From: "Lokesh Kumar" <lokesh(a)interacesso.pt> To: <daniel(a)voice-system.ro> Sent: Friday, November 18, 2005 5:35 PM Subject: Re: [Users] Problem with the registration of X-PRO client with Openser > Hello Daniel, > > Thanks for your reply. > > Here is the ouput I got wheb i did ngrep on openser > > "----------snip---------------## > > U 192.168.0.47:5060 -> 192.168.0.60:5060 > > REGISTER sip:192.168.0.60 SIP/2.0..… [View More]Via: SIP/2.0/UDP > 192.168.0.47:5060;rport;branch=z9hG4bKA4 > > E96E15E966428C88FB407711BD924E..From: Alexandre Alves > <sip:3501@192.168.0.60>..To: Alexandre > > Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" > <sip:3501@192.168.0.47:5060>..Cal > > l-ID: CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq: 26879 > REGISTER..Expires: 1800..Ma > > x-Forwards: 70..User-Agent: X-PRO build 1101..Content-Length: 0.... > > # > > I 192.168.0.60 -> 192.168.0.47 3:3 > > ....E.........1D.../...<......"MREGISTER sip:192.168.0.60 SIP/2.0..Via: > SIP/2.0/UDP 192.168. > > 0.47:5060;rport;branch=z9hG4bKA4E96E15E966428C88FB407711BD924E..From: > Alexandre Alves <sip:3 > > 501(a)192.168.0.60>..To: Alexandre Alves <sip:3501@192.168.0.60>..Contact: > "Alexandre Alves" < > > sip:3501@192.168.0.47:5060>..Call-ID: > CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq: 2 > > 6879 REGISTER..Expires: 1800..Max-Forwards: 70..User-Agent: X-PRO build > 1101..Content-Length > > : 0....------------------snip-------------------------" > > > > BELOW IS THE OUTPUT I GOT WHEN I DO NGREP 5060...I AM UNABLE TO FIGURE OUT > ,WHATS WRONG WITH IT. PLEASE HELP. > > "------------------snip-------------- > > U 192.168.0.47:5060 -> 192.168.0.60:5060 > > REGISTER sip:192.168.0.60 SIP/2.0..Via: SIP/2.0/UDP > 192.168.0.47:5060;rport;branch=z9hG4bK7E > > 650EAFB35F40FEB93391F06D534AC5..From: Alexandre Alves > <sip:3501@192.168.0.60>..To: Alexandre > > Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" > <sip:3501@192.168.0.47:5060>..Cal > > l-ID: CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq: 26883 > REGISTER..Expires: 1800..Ma > > x-Forwards: 70..User-Agent: X-PRO build 1101..Content-Length: 0.... > > # > > I 192.168.0.60 -> 192.168.0.47 3:3 > > ....E.........0..../...<.......>REGISTER sip:192.168.0.60 SIP/2.0..Via: > SIP/2.0/UDP 192.168. > > 0.47:5060;rport;branch=z9hG4bK7E650EAFB35F40FEB93391F06D534AC5..From: > Alexandre Alves <sip:3 > > 501(a)192.168.0.60>..To: Alexandre Alves <sip:3501@192.168.0.60>..Contact: > "Alexandre Alves" < > > sip:3501@192.168.0.47:5060>..Call-ID: > CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq: 2 > > 6883 REGISTER..Expires: 1800..Max-Forwards: 70..User-Agent: X-PRO build > 1101..Content-Length > > : 0....------------------snip-----------------------" > > ----- Original Message ----- > From: "Daniel-Constantin Mierla" <daniel(a)voice-system.ro> > To: "Lokesh Kumar" <lokesh(a)interacesso.pt> > Cc: <users(a)openser.org> > Sent: Friday, November 18, 2005 4:49 PM > Subject: Re: [Users] Problem with the registration of X-PRO client with > Openser > > > > It seems that the REGISTER message was not received by openser at all, > > otherwise a reply should be received. Do the ngrep on openser machine to > > see if the sip messages reach the system. Also, check if you have any > > firewall on port 5060 in place on openser server. > > > > Cheers, > > Daniel > > > > On 11/18/05 18:20, Lokesh Kumar wrote: > > > Hello Friends, > > > > > > I am having trouble with registration of X-PRO softphone with openser, > > > where my softphone is on 192.168.0.47 and openser is on 192.168.0.60. > > > My softphone says "Login Timed out , Contact Network admin". > > > > > > Here I am posting ngrep output also.Openser is having default configs, > > > but I had added this softphone as a user also. > > > > > > Can anyone help me please. > > > > > > Best Regards > > > Lokesh > > > > > > > > > [root@virtualmachine configs]# ngrep 192.168.0.47 > > > interface: eth0 (192.168.0.0/255.255.255.0) > > > match: 192.168.0.47 > > > ##################################################################### > > > U 192.168.0.47:5060 -> 192.168.0.60:5060 > > > REGISTER sip:192.168.0.60 SIP/2.0..Via: SIP/2.0/UDP > > > 192.168.0.47:5060;rport;branch=z9hG4bK96 > > > DBF3C437034F28A50968410DEB564C..From: Alexandre Alves > > > <sip:3501@192.168.0.60>..To: Alexandre > > > Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" > > > <sip:3501@192.168.0.47:5060>..Cal > > > l-ID: CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq > > > <mailto:CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq>: 26821 > > > REGISTER..Expires: 1800..Ma > > > x-Forwards: 70..User-Agent: X-PRO build 1101..Content-Length: 0.... > > > # > > > I 192.168.0.60 -> 192.168.0.47 3:3 > > > ....E...o\....H..../...<......<PREGISTER sip:192.168.0.60 > > > SIP/2.0..Via: SIP/2.0/UDP 192.168. > > > > > > 0.47:5060;rport;branch=z9hG4bK96DBF3C437034F28A50968410DEB564C..From: > > > Alexandre Alves <sip:3 > > > 501(a)192.168.0.60>..To <mailto:501@192.168.0.60%3E..To>: Alexandre > > > Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" < > > > sip:3501@192.168.0.47:5060>..Call-ID: > > > CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq > > > <mailto:CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq>: 2 > > > 6821 REGISTER..Expires: 1800..Max-Forwards: 70..User-Agent: X-PRO > > > build 1101..Content-Length > > > : 0.... > > > ##### > > > U 192.168.0.47:5060 -> 192.168.0.60:5060 > > > REGISTER sip:192.168.0.60 SIP/2.0..Via: SIP/2.0/UDP > > > 192.168.0.47:5060;rport;branch=z9hG4bK96 > > > DBF3C437034F28A50968410DEB564C..From: Alexandre Alves > > > <sip:3501@192.168.0.60>..To: Alexandre > > > Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" > > > <sip:3501@192.168.0.47:5060>..Cal > > > l-ID: CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq > > > <mailto:CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq>: 26821 > > > REGISTER..Expires: 1800..Ma > > > x-Forwards: 70..User-Agent: X-PRO build 1101..Content-Length: 0.... > > > # > > > I 192.168.0.60 -> 192.168.0.47 3:3 > > > ....E...o]....H..../...<......<PREGISTER sip:192.168.0.60 > > > SIP/2.0..Via: SIP/2.0/UDP 192.168. > > > > > > 0.47:5060;rport;branch=z9hG4bK96DBF3C437034F28A50968410DEB564C..From: > > > Alexandre Alves <sip:3 > > > 501(a)192.168.0.60>..To <mailto:501@192.168.0.60%3E..To>: Alexandre > > > Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" < > > > sip:3501@192.168.0.47:5060>..Call-ID: > > > CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq > > > <mailto:CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq>: 2 > > > 6821 REGISTER..Expires: 1800..Max-Forwards: 70..User-Agent: X-PRO > > > build 1101..Content-Length > > > : 0.... > > > ##exit > > > 78 received, 0 dropped > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > [View Less]

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Fwd: [Serusers] mediaproxy module: RTP packet mapping after reINVITE
by Adrian Georgescu 20 Nov '05

20 Nov '05

Hi Joel, MediaProxy development goes further. I will take a look at your observations and let you know if there is any duplicate effort. Thanks Adrian >>>>>>>>>> Anyone have a comment on the problem that I observe in which the mediaproxy receives a request for a session with a media description that has a new (rather than additional) connection address, but does not act upon it? Is there continuing work going on for this module? Here is my latest code which … [View More]

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[Serusers] Mediaproxy on Multiprocessor System
by Adrian Georgescu 20 Nov '05

20 Nov '05

Your RTP generator process is also using CPU, this is why you see both loaded during the test. MediaProxy uses only one CPU. Adrian >>>>>> Hi All, I run mediaproxy on a 2 CPUs machine (Fedora Code 3). When I stress the mediaproxy with rtpgenerator locally, i.e., just rtpgenerator.py --count=50, I noticed that the load is distributed evenly between the 2 CPUs. But when I stress remotely, using a seperate PC, i.e., rtpgenerator.py --ip=x.x.x.x --proxy=y.y.y.y --count=… [View More]

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Fwd: [Serusers] Get alias for user
by Voipers Portugal 20 Nov '05

20 Nov '05

Why don't you use rewritehost to do that? Jose Simoes 2005/11/19, Pol <kuroki(a)gmail.com>: > I Jan, I tryied your solution( creating a reversealiases table and do > lookup() function before rewrite host, but it didn't success. > > I have no idea on how to acomplish that!!! it seems to be easy but, remains > kicking me hard, and I'm stopped at this point, for several weeks. > I just want to forward incoming calls that are for users not On-line to > asterisk by … [View More]replacing TO header for his/her alias. > > > INVITE sip:pnovell@domain.com ---> SER ( USER NOT ONLINE ) > | > | > lookup("aliases") pnovell=17940 > | > > ----> FORWARD sip:17940@IP_OF_ASTERISK > Any help? > > -- > Pol Novell > UPF- Universitat Pompeu Fabra. > > > On 10/24/05, Jan Janak <jan(a)iptel.org> wrote: > > > > On 24-10-2005 14:12, Damien Sandras wrote: > > > Le lundi 24 octobre 2005 ? 14:06 +0200, Samuel Osorio Calvo a écrit : > > > > lookup("aliases"); > > > > > > > > will do the job for you if aliases is the name of the table containing > the alias bindings (by default it will be). > > > > > > > > > > I had understood this, but the problem here is that I need the reverse: > > > the original request uri contains the username and I need to rewrite it > > > into the user alias. > > > > > > Example, SER receives INVITE dsandras @ gnomemeeting.net, and I want > > > to use rewriteuri to rewrite the INVITE into INVITE 600001 @ > > > gnomemeeting.net. > > > > > > Is that possible? > > > > Just setup reverse aliases. You can either use aliases table or create > > your own, let's say revaliases. > > > > Revaliases should have exactly same structure as aliases table and you > > should als add a row in version table for it. revaliases could contain > > usernames and map them to numbers. That is username would be dsandras > > and contact sip:600001@gnomemeeting.net . You can then call > > lookup("realiases") in the script. > > > > Jan. > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > _____________________________________________________________________ > > Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es > > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > [View Less]

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[Users] Re: Users Digest, Vol 6, Issue 50
by Michael Kahlke 19 Nov '05

19 Nov '05

Tavis, Check out http://www.brynosaurus.com/pub/net/p2pnat/, it's not quite what you're asking for but it does include a table of data gathered from volunteers, detailing the percentage of NATs that support udp/tcp hole-punching and hairpin (loopback) translation. I'm concerned about the latter, it appears that not even ICE can resolve that issue. Mike > Date: Fri, 18 Nov 2005 15:36:35 -0800 > From: Tavis P <tavis.lists(a)galaxytelecom.net> > Subject: [Users] How … [View More]

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Re: [Serusers] Get alias for user
by Damien Sandras 19 Nov '05

19 Nov '05

Le lundi 24 octobre 2005 à 14:06 +0200, Samuel Osorio Calvo a écrit : > lookup("aliases"); > > will do the job for you if aliases is the name of the table containing the alias bindings (by default it will be). > I had understood this, but the problem here is that I need the reverse: the original request uri contains the username and I need to rewrite it into the user alias. Example, SER receives INVITE dsandras @ gnomemeeting.net, and I want to use rewriteuri to rewrite the … [View More]

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[Serusers] Contact field in SIP HF between asterisk + ser
by harry gaillac 19 Nov '05

19 Nov '05

Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and … [View More]

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[Serusers] Version upgrade
by Jan Henkins 18 Nov '05

18 Nov '05

Hello all, Newbie question (apologies if this has been asked already): I have been asked to upgrade an existing SER 0.8.14 to 0.9.4, or at least to investigate a possible upgrade path. This particular installation has quite a large number of users (about 550) in MySQL, so naturally an upgrade needs to be planned rather carefully. Is there any issues that I need to be aware of (like MySQL schema changes or similar major challenges) in order to affect an upgrade where no membership details are … [View More]

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Fwd: [Serusers] mediaproxy module: RTP packet mapping after reINVITE with new SDP
by Joel Rosenfield 18 Nov '05

18 Nov '05

Anyone have a comment on the problem that I observe in which the mediaproxy receives a request for a session with a media description that has a new (rather than additional) connection address, but does not act upon it? Is there continuing work going on for this module? Here is my latest code which seems to solve the problem of a simple 1-audio-session call, while attempting to preserve the existing video add-on capability. Please excuse the clumsiness of the code as I am new to python. … [View More]While it doesn't solve the general SDP management case, it does solve the simple and most common case of 1 audio stream. If the project leadership thinks this is useful, I can come up with a more complete solution and submit the code. However, if there is already ongoing work in this area, I don't want to duplicate the effort. Thanks, - Joel # class RTPSession def updateStreams(self, sessionData): print "updateStreams" streams = [s.split(':') for s in sessionData[2].split(',')] streamCount = len(self.mediaStreams) if streamCount < len(streams): # Add one media session. print "Adding Streams" visibleIP = sessionData[3] asymmetric = self.callerInfo.get('asymmetric', 0) for sinfo in streams[streamCount:]: try: mediatype = sinfo[2] except IndexError: mediatype = 'Audio' # remove at a later time print("warning: SER is using an old mediaproxy.so module. " "Please upgrade it to get correct media type information.") try: stream = MediaStream(self, mediatype) except MediaStreamError: # check this. also at session creation #for stream in self.mediaStreams: #stream.close() raise RTPSessionError, "cannot find enough free ports for the media streams" stream.setCaller(sinfo, visibleIP, asymmetric) self.mediaStreams.append(stream) self.oldCount = streamCount self.needStreamUpdate = 1 sinfo = streams[0] print "updateStreams checking for same media" # if sinfo[0] == self.mediaStreams[0].rtpStream.caller.__v.contactIP # if sinfo[1] == self.mediaStreams[0]rtpStream.caller.__v.rtpPort # # No change to media # return # Media changed. print "Media IP:port is %s:%s" % (sinfo[0], sinfo[1]) visibleIP = sessionData[3] asymmetric = self.callerInfo.get('asymmetric', 0) self.mediaStreams[0].rtpStream.caller = None self.mediaStreams[0].setCaller(sinfo, visibleIP, asymmetric) print "updateStreams returning" --- Joel Rosenfield <joelrosenfield(a)yahoo.com> wrote: > Date: Tue, 15 Nov 2005 08:31:00 -0800 (PST) > From: Joel Rosenfield <joelrosenfield(a)yahoo.com> > To: serusers(a)lists.iptel.org > Subject: [Serusers] mediaproxy module: RTP packet mapping after reINVITE > with new SDP > > As I am new to this list, please tell me if there is a more appropriate list > to > which I should post this question. > > Does the mediaproxy handle changes in the media description within the same > call dialog? If not, are there plans to do so? > > > Here is what I have done: > > I am using the SER mediaproxy module v1.4.2 and a co-located proxydispatcher > and mediaproxy to accomplish NAT traversal between a 3PCC and an remote ATA. > All basic call scenarios work fine. However, when the module receives a > reINVITE to reconnect the ATA to a different party, the mediaserver fails to > forward the RTP packets between the ATA and party described in the new SDP. > > I saw in the mediaproxy logs that RTP packets had been received from both > sides, but the signIn method was being invoked as "called" for both parties. > > I explored the rtphandler.py and saw that the "catchall" clause at the end of > RTPStream.handle_read was being invoked as you can see by the debug statement > that I added to the code. It appears that this is invoked because both > parties > had already signed in based upon the previous media description, so none of > the > other mapping/forwarding clauses are invoked. > > It seems that there needs to be a mechanism to clear the mappings when the > mediaproxy receives a "request" command for the same session (based upon the > SIP Call-ID) but a new SDP. I got it work by adding code to the beginning of > RTPSession.updateStreams() that clears the list "mediaStreams" and letting > the > rest of the code in that method recreate the mappings from scratch. See the > attached file containing the code, a mediaproxy log before the change, and a > log from after the change. > > Although this works, the solution is a bit incomplete since it can cause > brief > interruptions in the RTP stream when a reINVITE is received that does not > include an SDP change, etc. > > Please let me know if there is something I am missing with respect to how I > should using the mediaproxy. > > Thanks, > - Joel Rosenfield > > > > __________________________________ > Yahoo! FareChase: Search multiple travel sites in one click. > http://farechase.yahoo.com> _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > __________________________________ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers [View Less]

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[Users] dbtext again?
by Florian Overkamp 18 Nov '05

18 Nov '05

Hi guys, I'm only just getting started with OpenSer 1.0.0 (from the tarball) and I'd like to use the dbtext option. However, it seems the sc.dbtext doesnt really generate good files. OpenSER reports: 0(0) AUTH module - initializing 0(0) AUTH_DB module - initializing Content of [version] table_name(str) table_version(int) subscriber:3 location:6 aliases:6 Content of result table_version(int) 3 Content of result table_version(int) 6 0(0) register_udomain(): Invalid table version (… [View More]

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[Serusers] mysql.so not found
by Colin Jordan 18 Nov '05

18 Nov '05

Hi there I have installed SER 9.4 with MYSQL 4.0.25. I am having trouble with mysql.so. As soon as I activate it in the ser.cfg ser will not start. I have noted that it is not instaledl in the modules directory. I have also run the following to try and include it in making them: make modules include_modules="mysql" make install include_modules="mysql" As per the documentation I need to get ser-mysql but the rpm versions do not find the ser module. What can I do to get mysql.so on the … [View More]

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[Serusers] ser listening on multiple signalling ports?
by Ramin Nikaeen 18 Nov '05

18 Nov '05

Dear friends, I am trying to get two SIP User Agents behind a single router device to be able to call each other. But the signaling doesn't work because they both contend for the same signaling port 5060 on the router device. Is it possible to have ser listen on multiple signaling ports such as 5060, 5061, 5062, etc rather than the single port 5060? Thanks rmain

2 1

[Serusers] Cisco IP phone NAT config
by Joao Pereira 18 Nov '05

18 Nov '05

Hello, I have SER in bridging mode with two IPs (private and public). To dial the world, my Cisco IP phones must contact the SER private IP, and the call is then proxyed by SER. All other SIP clients can do it, but the Cisco phones dont.... What should I put in the configuration file? For now I have this: # NAT/Firewall Traversal nat_enable: 1 nat_address: "" voip_control_port: 5060 start_media_port: 8000 end_media_port: 9000 nat_received_processing: 0 ...but I m not really using NAT, … [View More]

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[Users] Problem with the registration of X-PRO client with Openser
by Lokesh Kumar 18 Nov '05

18 Nov '05

Hello Friends, I am having trouble with registration of X-PRO softphone with openser, where my softphone is on 192.168.0.47 and openser is on 192.168.0.60. My softphone says "Login Timed out , Contact Network admin". Here I am posting ngrep output also.Openser is having default configs, but I had added this softphone as a user also. Can anyone help me please. Best Regards Lokesh [root@virtualmachine configs]# ngrep 192.168.0.47 interface: eth0 (192.168.0.0/255.255.255.0) match: 192.168.0.… [View More]47 ##################################################################### U 192.168.0.47:5060 -> 192.168.0.60:5060 REGISTER sip:192.168.0.60 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.47:5060;rport;branch=z9hG4bK96 DBF3C437034F28A50968410DEB564C..From: Alexandre Alves <sip:3501@192.168.0.60>..To: Alexandre Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" <sip:3501@192.168.0.47:5060>..Cal l-ID: CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq: 26821 REGISTER..Expires: 1800..Ma x-Forwards: 70..User-Agent: X-PRO build 1101..Content-Length: 0.... # I 192.168.0.60 -> 192.168.0.47 3:3 ....E...o\....H..../...<......<PREGISTER sip:192.168.0.60 SIP/2.0..Via: SIP/2.0/UDP 192.168. 0.47:5060;rport;branch=z9hG4bK96DBF3C437034F28A50968410DEB564C..From: Alexandre Alves <sip:3 501(a)192.168.0.60>..To: Alexandre Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" < sip:3501@192.168.0.47:5060>..Call-ID: CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq: 2 6821 REGISTER..Expires: 1800..Max-Forwards: 70..User-Agent: X-PRO build 1101..Content-Length : 0.... ##### U 192.168.0.47:5060 -> 192.168.0.60:5060 REGISTER sip:192.168.0.60 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.47:5060;rport;branch=z9hG4bK96 DBF3C437034F28A50968410DEB564C..From: Alexandre Alves <sip:3501@192.168.0.60>..To: Alexandre Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" <sip:3501@192.168.0.47:5060>..Cal l-ID: CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq: 26821 REGISTER..Expires: 1800..Ma x-Forwards: 70..User-Agent: X-PRO build 1101..Content-Length: 0.... # I 192.168.0.60 -> 192.168.0.47 3:3 ....E...o]....H..../...<......<PREGISTER sip:192.168.0.60 SIP/2.0..Via: SIP/2.0/UDP 192.168. 0.47:5060;rport;branch=z9hG4bK96DBF3C437034F28A50968410DEB564C..From: Alexandre Alves <sip:3 501(a)192.168.0.60>..To: Alexandre Alves <sip:3501@192.168.0.60>..Contact: "Alexandre Alves" < sip:3501@192.168.0.47:5060>..Call-ID: CAF0E758953441DD8ECB6C118A4A8331@192.168.0.60..CSeq: 2 6821 REGISTER..Expires: 1800..Max-Forwards: 70..User-Agent: X-PRO build 1101..Content-Length : 0.... ##exit 78 received, 0 dropped [View Less]

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[Users] exec_dset and added HF
by Medve Istvan 18 Nov '05

18 Nov '05

Hi, I would like to add some extra header fields with avp_pushto into the received SIP message before I call exec_dset. My tests show that added header fields cannot read from environment variables. Shortly I would like to use an external script to manipulate ruri based on AVPs received from radius (eg. rpid). Can I do this without sql or radius request from external script ? regards, imedve

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Re: [Serusers] call forwarding and *21
by sip 18 Nov '05

18 Nov '05

THAT'S not too bad. Basically, what they dial will be the RURI. So if they dial *21*1*5551212 your SIP RURI becomes sip:*21*1*5551212@your.server All you'd need is a check for the URI and then strip out the unnecessary bits and use something like an AVPop to store the db info... something like: (note, I'm just tossing stuff out for an idea... this syntax may be completely wrong) if(uri =~ "^sip:\*21\*1") { strip(6); avp_write("$ruri","i:19"); .... avp_db_store("$from/… [View More]

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[Serusers] Cisco phones port range
by Joao Pereira 18 Nov '05

18 Nov '05

Hi Im using Cisco IP 7940 (with SIP firmware) and I want to force him to put the media stream in some specific port. To do it I put this in the Cisco configuration file: start_media_port: 8000 end_media_port: 9000 but the Cisco IP phone boots and doesnt accept these ports, and assumes the defaults (16384-32766). Even when I put these ports directly in the phone configuration, he doesnt accept them. How can I change the RTP ports in the Cisco IP phone? ( Like in Xlite we do: … [View More]

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[Serusers] SER download site continuously "busy"
by Davey, Douglas A (Doug) 18 Nov '05

18 Nov '05

I have been unable to download SER from ftp://ftp.berlios.de/pub/ser/ <ftp://ftp.berlios.de/pub/ser/> for several days. Several colleagues have reported the same problem. The ftp server appears to be "busy" at all times even though it is never near its maximum of 95 users. I have downloaded SER in the past for several projects without problems, so this seems to be a relatively recent change. Are there mirror sites from which to download a SER installation package? I found … [View More]

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Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting
by 'Jan Janak' 18 Nov '05

18 Nov '05

On 17-11-2005 10:21, Lenir wrote: > In this case the radius proxy wont work, because you never can anticipate > who hangs up the call, thus radius wont know who hung up the call...Besides > all other voice applications/hardware (SIP and H323) that use radius do not > behave like that, the Called-Station-ID ALWAYS remains the same, as with the > Calling-Station-ID. Could you name those SIP applications that behave the way you describe ? Jan.

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[Users] new group feature
by Bogdan-Andrei Iancu 18 Nov '05

18 Nov '05

Hi, I have implemented a new group feature. So far we have group support with fixed matching: user X belongs to group G, etc.. This proves to be inefficient when you have huge block of users belonging to small numbers of groups. So I did some regular expression based groups: if user@domain matched a RE -> belongs to group G very usefull if try to provisions routing logic based on groups. I developed this new functionality inside the group module since there is about same … [View More]

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[Users] Re: How does openser use the nathelper
by Klaus Darilion 18 Nov '05

18 Nov '05

Hi messa! I suggest you to take a look on the getting_started document from onsip.org. It uses mediaproxy module and the mediaproxy (just another flavour of nathelper+rtpproxy). You can't integrate only nathelper to vocal. Use openser+nathelper as replacement for vocal or together with vocal. btw: you could resolve the problem by using clients which support ICE. regards klaus PS: please no direct email. Send your emails always to the list. messa(a)innsof.com wrote: > Hi Klaus > … [View More]

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[Users] Re: How does openser use the nathelper
by Klaus Darilion 18 Nov '05

18 Nov '05

messa(a)innsof.com wrote: > Hi Klaus > > I want you to help me. I want to integrate nathelper in my vocal > system to > fix the problem of NAT traversal because the STUN solution is not good for > me. With stun, i cannot make in the same moment a call to another client > in a remote network and to a client in the same network. ANd NAthelper do > it good for openser. So i want to know, the architecture of openser. How > call the nathelper module. I want to have … [View More]

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[Serusers] Contact field in SIP HF between asterisk + ser
by harry gaillac 18 Nov '05

18 Nov '05

Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and … [View More]

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[Serusers] Contact field in SIP HF between asterisk + ser
by harry gaillac 18 Nov '05

18 Nov '05

Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and … [View More]

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[Serusers] RTPproxy how to
by crazy 18 Nov '05

18 Nov '05

hi all i donot know howto set the port of RTPproxy, my server public ip such as 202.97.123.123, i want to set RTPproxy use the 5080-5082 ports,what can i do for it. client can use 202.97.123.123:5080-5082 as a outbound proxy!

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[Users] Re: [Serdev] How to gaurd PSTN gateway with openser + NAT
by Lokesh Kumar 18 Nov '05

18 Nov '05

Hello Nils, Attached is the mentioned files you asked me to send. As you said, after disabling gnutls in configure, I can compile it. It works fine . Thanks for your kind help. One important question I would like to ask you. Suppose I am having 100 SIP clients, all of them are behind their respective 100 NAT's, The NAT devices are assigned public IP dynamically. What besides openser do I need on my SIP server to traverse those NAT's. Assume NAT's does not support STUN. Do I need to put STUN … [View More]or Mediaproxy/RTPProxy server on my SIP server. If not , then how could i can manage such network. Thanks a lot for your kind help Best Regards Lokesh ----- Original Message ----- From: "Nils Ohlmeier" <lists(a)ohlmeier.org> To: "Lokesh Kumar" <lokesh(a)interacesso.pt> Sent: Thursday, November 17, 2005 10:14 PM Subject: Re: [Serdev] How to gaurd PSTN gateway with openser + NAT > Hello Lokesh, > > could you do me a favor and send me config.h and config.log from the directory > where you tried to compile 0.9.2? > > After you send me the files you could try the following command > './configure --disable-gnutls' and afterwards 'make'. I think that should help > to compile sipsak-0.9.2 on your system. > > Thanks > Nils > > On Friday 18 November 2005 19:54, you wrote: > > Hello Nils, > > > > When I do make in the source directory of sipsak-0.9.2, it gives me error > > like this. > > > > /usr/lib/libgnutls-openssl.so: undefined reference to > > 'asn1_delete_structue' /usr/lib/libgnutls-openssl.so: undefined reference > > to 'asn1_create_element' /usr/lib/libgnutls-openssl.so: undefined reference > > to 'asn1_der_decoding' collect2: ld returned 1 exit status > > make[1]: ***[spsak] Error 1 > > make[1]: Leaving directory '/usr/src/sipsak-0.9.2' > > make: *** [all] Error 2 > > > > But there is no such problem with 0.9.0 version. > > > > Regards > > > > Lokesh > > > > > > ----- Original Message ----- > > From: "Nils Ohlmeier" <lists(a)ohlmeier.org> > > To: <serdev(a)iptel.org> > > Cc: "Lokesh Kumar" <lokesh(a)interacesso.pt>; <deepak.dhiman(a)uf4.net>; > > <serusers(a)iptel.org> > > Sent: Thursday, November 17, 2005 6:05 PM > > Subject: Re: [Serdev] How to gaurd PSTN gateway with openser + NAT > > > > > Hello Lokesh, > > > > > > On Friday 18 November 2005 01:27, Lokesh Kumar wrote: > > > > I installed openser, it is working fine. sipsak-0.9.0 is also working > > > > fine, > > > > > > previously i was using 0.9.2 version of sipsak that is having problem. > > > > > > I would really appreciate it if you could describe we what problems you > > > > had > > > > > with sipsak 0.9.2. Then I could try to fix it, if it is a bug in sipsak. > > > > > > Thanks > > > Nils [View Less]

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[Users] Scripting tutorial
by Bogdan-Andrei Iancu 18 Nov '05

18 Nov '05

Hi, thanks to Norman Brandinger we have a basic tutorial about OpenSER scripting. The document is linked in doc section on the project page, along with the other tutorials: http://www.openser.org/docs also you can accessing directly at: http://www.openser.org/docs/scripting.html Again, many thanks to Norman for his contribution. Best regards, Bogdan

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[Users] Compile problem on SLES 9 EM64T
by Marcello Lupo 18 Nov '05

18 Nov '05

Hi i'm getting this errors of files not found compiling OPenser stable 1.0.0 on Suse Linux Enterprise Server 9 x86_64 . Can you address me to solve the issue? After the compilation the Openser compile but i don't know if it can be an operational problem. After make proper i get: venus:/data/src/openser-1.0.0 # make Makefile.rules:91: action.d: No such file or directory Makefile.rules:91: crc.d: No such file or directory Makefile.rules:91: daemonize.d: No such file or directory Makefile.… [View More]rules:91: data_lump.d: No such file or directory Makefile.rules:91: data_lump_rpl.d: No such file or directory Makefile.rules:91: dprint.d: No such file or directory Makefile.rules:91: dset.d: No such file or directory Makefile.rules:91: error.d: No such file or directory Makefile.rules:91: fifo_server.d: No such file or directory Makefile.rules:91: flags.d: No such file or directory Makefile.rules:91: forward.d: No such file or directory Makefile.rules:91: hash_func.d: No such file or directory Makefile.rules:91: ip_addr.d: No such file or directory Makefile.rules:91: items.d: No such file or directory Makefile.rules:91: main.d: No such file or directory Makefile.rules:91: md5.d: No such file or directory Makefile.rules:91: md5utils.d: No such file or directory Makefile.rules:91: modparam.d: No such file or directory Makefile.rules:91: msg_translator.d: No such file or directory Makefile.rules:91: pass_fd.d: No such file or directory Makefile.rules:91: proxy.d: No such file or directory Makefile.rules:91: qvalue.d: No such file or directory Makefile.rules:91: re.d: No such file or directory Makefile.rules:91: receive.d: No such file or directory Makefile.rules:91: resolve.d: No such file or directory Makefile.rules:91: route.d: No such file or directory Makefile.rules:91: route_struct.d: No such file or directory Makefile.rules:91: script_cb.d: No such file or directory Makefile.rules:91: socket_info.d: No such file or directory Makefile.rules:91: sr_module.d: No such file or directory Makefile.rules:91: stats.d: No such file or directory Makefile.rules:91: tcp_main.d: No such file or directory Makefile.rules:91: tcp_read.d: No such file or directory Makefile.rules:91: timer.d: No such file or directory Makefile.rules:91: tsend.d: No such file or directory Makefile.rules:91: udp_server.d: No such file or directory Makefile.rules:91: unixsock_server.d: No such file or directory Makefile.rules:91: usr_avp.d: No such file or directory Makefile.rules:91: mem/f_malloc.d: No such file or directory Makefile.rules:91: mem/mem.d: No such file or directory Makefile.rules:91: mem/memtest.d: No such file or directory Makefile.rules:91: mem/q_malloc.d: No such file or directory Makefile.rules:91: mem/shm_mem.d: No such file or directory Makefile.rules:91: mem/vq_malloc.d: No such file or directory Makefile.rules:91: parser/hf.d: No such file or directory Makefile.rules:91: parser/msg_parser.d: No such file or directory Makefile.rules:91: parser/parse_allow.d: No such file or directory Makefile.rules:91: parser/parse_content.d: No such file or directory Makefile.rules:91: parser/parse_cseq.d: No such file or directory Makefile.rules:91: parser/parse_disposition.d: No such file or directory Makefile.rules:91: parser/parse_diversion.d: No such file or directory Makefile.rules:91: parser/parse_event.d: No such file or directory Makefile.rules:91: parser/parse_expires.d: No such file or directory Makefile.rules:91: parser/parse_fline.d: No such file or directory Makefile.rules:91: parser/parse_from.d: No such file or directory Makefile.rules:91: parser/parse_hname2.d: No such file or directory Makefile.rules:91: parser/parse_hostport.d: No such file or directory Makefile.rules:91: parser/parse_methods.d: No such file or directory Makefile.rules:91: parser/parse_nameaddr.d: No such file or directory Makefile.rules:91: parser/parse_param.d: No such file or directory Makefile.rules:91: parser/parse_refer_to.d: No such file or directory Makefile.rules:91: parser/parser_f.d: No such file or directory Makefile.rules:91: parser/parse_rpid.d: No such file or directory Makefile.rules:91: parser/parse_rr.d: No such file or directory Makefile.rules:91: parser/parse_sipifmatch.d: No such file or directory Makefile.rules:91: parser/parse_to.d: No such file or directory Makefile.rules:91: parser/parse_uri.d: No such file or directory Makefile.rules:91: parser/parse_via.d: No such file or directory Makefile.rules:91: parser/digest/digest.d: No such file or directory Makefile.rules:91: parser/digest/digest_parser.d: No such file or directory Makefile.rules:91: parser/digest/param_parser.d: No such file or directory Makefile.rules:91: parser/contact/contact.d: No such file or directory Makefile.rules:91: parser/contact/parse_contact.d: No such file or directory Makefile.rules:91: db/db.d: No such file or directory Makefile.rules:91: db/db_fifo.d: No such file or directory Makefile.rules:91: db/db_id.d: No such file or directory Makefile.rules:91: db/db_pool.d: No such file or directory Makefile.rules:91: lex.yy.d: No such file or directory Makefile.rules:91: cfg.tab.d: No such file or directory bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex I checked in the main dir of openser src and the files are present. May be the Makefile is searching somewhere else ? Thanks, BYe, Marcello [View Less]

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[Serusers] Cisco 3640 connect to PBX
by Akarach Rattanawaree 18 Nov '05

18 Nov '05

Hello, I have Cisco 3640 with Voice and E1 card connected to Nortel PBX for call out from PBX to PSTN. I have problem when 1. user hang-up the phone and got dial tone from PBX 2. dial prefix 9 and telephone number (9,024455566) 3. user will got dial tone again from Cisco 4. If user dial telephone number again, it can go to PSTN. (024455566) User ------ > PBX ----- > Cisco 3640 ----- > SIP Server ----- > PSTN Have you ever … [View More]

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[Serusers] Mediaproxy on Multiprocessor System
by tcchan 18 Nov '05

18 Nov '05

Hi All, I run mediaproxy on a 2 CPUs machine (Fedora Code 3). When I stress the mediaproxy with rtpgenerator locally, i.e., just rtpgenerator.py --count=50, I noticed that the load is distributed evenly between the 2 CPUs. But when I stress remotely, using a seperate PC, i.e., rtpgenerator.py --ip=x.x.x.x --proxy=y.y.y.y --count=50, where x.x.x.x is rptgenerator's ip and y.y.y.y is mediaproxy's ip, I noticed that the load is mainly taken by CPU0. Can someone tell me what happenned? … [View More]

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[Serusers] UPDATE instead of re-INVITE and RTPProxy
by Ronald Voermans 18 Nov '05

18 Nov '05

Hello all, I have a problem with some user-agents (namely Grandstream and Cisco ATA 18X). My SIP provider provides a UPDATE instead of a re-INVITE, because the mentioned user-agents support the UPDATE method. However, SER/RTPProxy doesn't know how to handle this UPDATE request, and forwards the request to the user-agent. Result: one-way-voice. Does someone know how to solve this. I solved it for now by replacing the UPDATE in the request, so the SIP provider doesn't see that my user-agents … [View More]

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Re: [Serusers] mediaproxy for incoming
by Greger V. Teigre 18 Nov '05

18 Nov '05

You need in ser.cf both: alias="mydomain.com" alias="server.mydomain.com" g-) ----- Original Message ----- From: "George Lambson" <LambsonGE(a)mtc.byu.edu> To: <greger(a)teigre.com> Cc: <serusers(a)lists.iptel.org> Sent: Sunday, November 13, 2005 1:50 AM Subject: Re: [Serusers] mediaproxy for incoming > Any idea what I might be doing wrong? > > I have had some issues with setting up the served domain. I use x-lite for > my SIP UAs and have to put the host name … [View More]

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[Serusers] Usrloc question
by Darren Nay 18 Nov '05

18 Nov '05

Does anyone have any advice for me on issue? Thanks for your help! Darren Nay "Darren Nay" wrote: > > Hey all, > > > > I have a question regarding usrloc. I have run into a problem.. > > > > We have static routes sent to an asterisk server for all of our SIP > usernames. In addition our IAD’s will register with the same > username, so that calls coming into our switch for that username will > be routed to both the asterisk box and the SIP IAD. … [View More]

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[Serusers] Need help: SER and MYSQL issue: PID file /var/run/ser.pid does not exist
by Dhiman D Chowdhury 18 Nov '05

18 Nov '05

Hi all, I need help to fix the following issue which only occurs when I am loading mysql.so. The problem is that serctl will fail to start SER and sometimes if SER start with service SER command. SER process is not found. I have observed that for all those issues, the cause is as system log as follows: Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: Maxfwd module- initializing Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: new_connection(): Client does not support authentication protocol … [View More]

1 0

[Serusers] Need help: SER and MYSQL issue: PID file /var/run/ser.pid does not exist
by Dhiman D Chowdhury 18 Nov '05

18 Nov '05

Hi all, I need help to fix the following issue which only occurs when I am loading mysql.so. The problem is that serctl will fail to start SER and sometimes if SER start with service SER command. SER process is not found. I have observed that for all those issues, the cause is as system log as follows: Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: Maxfwd module- initializing Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: new_connection(): Client does not support authentication protocol … [View More]

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[Serusers] mediaproxy module: RTP packet mapping after reINVITE with new SDP
by Joel Rosenfield 18 Nov '05

18 Nov '05

As I am new to this list, please tell me if there is a more appropriate list to which I should post this question. Does the mediaproxy handle changes in the media description within the same call dialog? If not, are there plans to do so? Here is what I have done: I am using the SER mediaproxy module v1.4.2 and a co-located proxydispatcher and mediaproxy to accomplish NAT traversal between a 3PCC and an remote ATA. All basic call scenarios work fine. However, when the module receives a … [View More]

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[Users] How to gaurd PSTN gateway with openser + NAT
by Lokesh Kumar 17 Nov '05

17 Nov '05

Hello Friends, I installed openser, it is working fine. sipsak-0.9.0 is also working fine, previously i was using 0.9.2 version of sipsak that is having problem. Now I want to ask you 2 important things. 1>> Forget Asterisk for sometime, I am having a openser with one public IP and one private IP, and I am having one voip gateway which is connected to E1 PRI, and voip gateway is also having private IP. So, my question is can I forward all the calls coming on public IP of openser to … [View More]voip gateway via private IP's of openser and voip gateway.After that voip gateway will forward the call to E1 PRI. SER Admin guide mentioned about how to gaurd your PSTN gateway with SER, but the documentation is not suffciente there. So, can you please suggest in this scenario, how do i can configure request routing in openser, like pstn.cfg file. 2>> How do I have to register SIP phone on openser. Where the sip phones are behind the NAT, but NAT device supports STUN and UPnP. Do I need to setup seperate STUN server or I can run this service on openser server. Please help. Regards Lokesh ----- Original Message ----- From: <deepak.dhiman(a)uf4.net> To: "Lokesh Kumar" <lokesh(a)interacesso.pt> Sent: Tuesday, November 15, 2005 8:42 PM Subject: Re: Need Help in configuring Asterisk with SER and PSTN gateway > Hi Lokesh ! > > here i am writing my own experience baed views and facts: - > > 1) openser i certainly better than ser with better nat handling tricks, it > also has got TLS support first time ina sip server and certain other > enhanced feature are alo there, that can be find out on www.voip-info.org > > 2)sipsak is working fine on my pc and i really had no problem in installing > it, i have installed vesion sipsak-0.9.0 > > 3) i have not istalled any ser administrative program, so i can't tell you > which one is better. > > 4) in case of Radius server try with radius 0.9.0. > > 5) Finally, ser doesn't have any pstn support so you can't connect it to the > pstn gateway, > you can forward all sip packets to asterisk server and it would forward it > to pstn gateway and i think that's is the possible solution of your problem. > > regards, > > Deepak > > Lokesh Kumar writes: > > > Hello Deepak, > > > > Here are some of the questions which if you can respond then i will be quite > > happy. > > > > 1>> What is the difference between SER and openSER and which one is best to > > adopt in terms of security and ease of use and customization. > > 2>> I am unable to install sipsak on SER, it gives me error, when i execute > > make. > > 3>> Which is the best web tool to manage SER, I tried SERweb and SERadmin, > > but unfortunately unable to configure them with SER > > 4>> How do I have to setup a Voip trunk between asterisk and SER, where SER > > forwards any requests from asterisk to PSTN gateway. > > 5>> How do I have to inter conect Radius with SER. > > > > I had searched everything on the internet but sufficiente documents are not > > available. > > > > Looking for your reply > > Thanks a lot > > Lokesh > > > > > > ----- Original Message ----- > > From: <deepak.dhiman(a)uf4.net> > > To: "Lokesh Kumar" <lokesh(a)interacesso.pt> > > Sent: Monday, November 14, 2005 8:49 PM > > Subject: Re: Need Help in configuring Asterisk with SER and PSTN gateway > > > > > >> Hi Lokesh ! > >> > >> I think u can easily configure it, let me know what have u tried so far so > >> that i can tell u exactly where is the problem.. > >> > >> regard, > >> > >> Deepak > >> > >> Lokesh Kumar writes: > >> > >> > Hello, > >> > > >> > Deepak, > >> > > >> > I got your mail from the asterisk lists. I wanted to ask you about how > > you had configured asterisk with SER. I am facing lots of difficulties with > > configuration of asterisk with SER. Can you please guide me . I will be very > > thankfull to you. > >> > > >> > Regards > >> > > >> > >> > >> > >> Software Engg. Trainee > >> Trail Ridge Software India Pvt. Ltd. > >> Noida > >> > > > > > > Software Engg. Trainee > Trail Ridge Software India Pvt. Ltd. > Noida > [View Less]

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[Serusers] Ser Business Mailing List ?
by Rehan Ahmed 17 Nov '05

17 Nov '05

Hello All, Is there a SER Business Mailing list for commerical only emails on ? Rehan www.supertec.com <http://www.supertec.com> www.didxchange.com <http://www.didxchange.com>

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[Users] exec_dset syntax error
by Iqbal 17 Nov '05

17 Nov '05

Hi can anyone see what wrong with exec_dset("/path/to/program.php") unless of course exec_dset only likes shell scripts Iqbal

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[Users] exec-msg
by Iqbal 17 Nov '05

17 Nov '05

Hi Anyoe got shell script to parse all the parameters sent, from my understanding the entire sip message is sent, into the script, I just want to pull the username out from From: iqbal

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[Users] A tutorial on the OSP peering module is now available
by Dmitry Isakbayev 17 Nov '05

17 Nov '05

Dear Users and Developers, A tutorial on the OSP peering module is now available at http://openser.org/docs/osp/Multi-Lateral_Peering_with_OpenSER_1_0_0.pdf. The module enables secure, multi-lateral peering with SIP devices that support OSP (i.e. OpenSER 1.1.x, OpenSER 1.0.x, (Open)SER 0.9.x, Asterisk 1.2). Open source OSP servers are available at www.sipfoundry.org/OSP. Regards, Dmitry

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[Serusers] Dropping 200-ok?
by Ricardo Poppi 17 Nov '05

17 Nov '05

Hi list. I want to make an analisis at a 200-ok response from an INVITE. I implemented at ser 0.9.4 an "on_reply_route" to check if the 200-ok contains the SDP address of the gateway I want to connect. It its true, I let the 200-ok flows. If it´s not, I want to drop this and terminate the call (SIP ERROR or whatever). Here is the example ============= ------------- "A" =======> | "SER" | =========> "B" ------------- 1) "… [View More]

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[Users] obscufucation...spelt that wrong ;-)
by Iqbal 17 Nov '05

17 Nov '05

Hi Any chance in next versions of openser, of having a module if activated that it would be able to read/create a unreadable config file, but would be readable by ser itself. Or is there already a way of doing this iqbal

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[Users] in dialog request when openser listens on multiple ports
by Klaus Darilion 17 Nov '05

17 Nov '05

Hi! I've configure openser to listen also on port udp:6060. The INVITE comes in to 6060, forwarded to other client on port 5060. Responses and ACK will be handled correctly - on call leg uses port 6060, the other call leg 5050. If now the callee sends a BYE (Cisco, strict router) the BYE will be forwarded from port 5060 instead of 6060. I didn't find a problem in the BYE request sent to the openser. Is openser behaving wrong here? Or do I have to configure special routing for this … [View More]

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Re: [Serusers] Re: [Serdev] ACK before INVITE on re-INVITE causes CiscoAS's to return a 500 error
by Evan Borgström 17 Nov '05

17 Nov '05

Hey Ettore, Yes, both INVITEs contain valid and correct SDP info. We have a maintenance scheduled to upgrade IOS on the Cisco devices so hopefully that will help it. -Evan Ettore Benedetti wrote: > Hallo Evan, > > I reckon Nils is right on the matter. The ACK of the dialog-forming INVITE is actually > the first in-dialog request and, as a matter of fact, it is a separate transaction. As far > as the UAS is concerned, the dialog is established as soon as the 200 response is … [View More]sent. > The Cisco GW just misbehaves provided the first INVITE from Asterisk did contain > an SDP offer (didn't it?). > > > Ettore Benedetti > > THALES COMMUNICATIONS B.V. > Bestevaer 46, 1271 ZA Huizen > The Netherlands > > Unclassified > > >>>> Evan Borgstrom <evan.borgstrom(a)ca.mci.com> 11/16/05 5:27:52 PM >>> > Hey Nils, > > You are correct that each message is handled by a different process, as > it to be expected, and I know that SER is not call statefull so my > concern lies around the fact that SER is delivering an INVITE which > requires FAR more processing and at least 2 database hits before a > loosely routed ACK which requires little to none for processing. > > I'm going to differ with you, and my self from last night, on the views > about the RFC as it does indeed cover the problem (I was just looking at > it the wrong way before). A re-INVITE cannot modify a dialog that does > not exist, and without the ACK being delivered in response to the 200 OK > the dialog is not yet created which is why the Cisco is returning a > CheckRequest failure since there's no dialog to map the re-INVITE to. > So, the device is correct in returning an error. > > When the Cisco returns the 500 error it's only for the re-INVITE so the > initial call stays established and a BYE is still required to tear it > down. The problem is that Asterisk is expecting that to work and has > already re-INVITED to the client so there's no audio and to the caller > the call failed. > > -Evan > > Nils Ohlmeier wrote: >> Evan, >> >> I would not invest the time to look into the SER side of your problem. Because >> this is a general problem. Instead of SER any other network element (router, >> switch, etc.) could change the order of the packets. >> BTW I guess this problem occurs on your SER because the ACK and the INVITE are >> handled by different SER processes. And as SER is not call-statefull there is >> no parameter to solve this situation. >> >> I fear you are right that this problem is not covered by any RFC, but the >> solutions seems logical to me: >> if this a generic problem, then either the UAC is not allowed to send a >> re-INVITE until the first INVITE is completed, or the UAS has to accept a >> re-INVITE after sending the 200 OK. But how does the the UAC know that the >> ACK reached the UAS so that it can send the re-INVITE now? It cant know that, >> so it would have to wait for a timer before it could send the re-INVITE. >> On the other side the INVITE transaction is completed for the UAS by sending >> the 200 OK (the ACK is a new transaction, according to the branch). >> Thus I think the UAS (the Cisco gateway) should accept the re-INVITE without >> complaining. >> >> Just for curiosity: is the call established or does the Cisco gateway drop the >> call after sending the 500? >> >> Regards >> Nils >> >> On Wednesday 16 November 2005 07:51, Greger V. Teigre wrote: >>> Evan, >>> I'm not able to match your text with the messages you show. The message >>> timestamped 16:51:51.559657, does it go to Cisco? It is shown to go to >>> Asterisk and it doesn't make sense. >>> However, until the Cisco gw receives the ACK from Asterisk, no call has >>> been established and the reINVITE cannot be matched. IMO, this is not a >>> Cisco bug. It seems very strange indeed that ser should send a reINVITE >>> before the ACK in the same call. Even without FIFO, an INVITE is likely to >>> take more time to process than a loose routed ACK. I would try to log more >>> thoroughly in SER to see what happens in processing the ACK and the >>> reINVITE. It could be a problem in your ser.cfg, but it seems strange >>> indeed. >>> g-) >>> >>> ----- Original Message ----- >>> From: "Evan Borgstrom" <evan.borgstrom(a)ca.mci.com> >>> To: <serusers(a)lists.iptel.org>; <serdev(a)lists.iptel.org> >>> Sent: Wednesday, November 16, 2005 12:05 AM >>> Subject: [Serdev] ACK before INVITE on re-INVITE causes Cisco AS's to >>> return a 500 error >>> >>>> Hey all, >>>> >>>> This has more to do with Asterisk & Cisco AS's IMO but I thought it >>>> would be interesting to get some responses from this group before I >>>> begin the battle with Cisco to get a bug report underway. >>>> >>>> We have an Asterisk box acting as a PBX sending calls to our SER >>>> instances. The asterisk box has "reinvite=yes" in the sip.conf file >>>> which causes Asterisk to send a re-INVITE for the call once it receives >>>> the 200 OK. This re-INVITE has it's SDP changed to that of the IP Phone >>>> behind the Asterisk box so that all media doesn't need to traverse it. >>>> This works fine except in certain instances where messages get out of >>>> order and it causes our Cisco GW's to error the call with a generic 500 >>>> message. When the messages are in the correct order the 500 error does >>>> not appear. >>>> >>>> So here's the situation, user behind the asterisk box picks up the >>>> phone and dials a PSTN destination. Asterisk fires off the initial >>>> invite and everything proceeds as normal and we eventually receive the >>>> 200 OK message to which the Asterisk box replies with ACK and a new >>>> INVITE that come in this order (X = Asterisk, Y = SER, Z = Cisco): >>>> >>>> U 2005/11/15 16:51:51.557322 X.X.X.X:5060 -> Y.Y.Y.Y:5060 >>>> ACK >>>> >>>> # >>>> U 2005/11/15 16:51:51.557948 X.X.X.X:5060 -> Y.Y.Y.Y:5060 >>>> INVITE >>>> >>>> SER accordingly processes the two messages and replies 100 Trying to >>>> the new INVITE message. Here's where the problem occurs, SER then sends >>>> these messages out to the PSTN gateway but the order has been reversed: >>>> >>>> # >>>> U 2005/11/15 16:51:51.558937 Y.Y.Y.Y:5060 -> X.X.X.X:5060 >>>> SIP/2.0 100 trying >>>> >>>> # >>>> U 2005/11/15 16:51:51.559000 Y.Y.Y.Y:5060 -> Z.Z.Z.Z:5060 >>>> INVITE >>>> >>>> # >>>> U 2005/11/15 16:51:51.559657 Y.Y.Y.Y:5060 -> X.X.X.X:5060 >>>> ACK >>>> >>>> When this happens the Cisco GW replies with a 500 error message and the >>>> following is sent to the Cisco's logs: >>>> >>>> # >>>> U 2005/11/15 16:51:51.711355 Z.Z.Z.Z:5060 -> Y.Y.Y.Y:5060 >>>> SIP/2.0 500 Server Internal Error. >>>> >>>> >>>> Nov 15 21:51:51.613: HandleUdpSocketReads :Msg enqueued for SPI with >>>> IPaddr: Y.Y.Y.Y:5060 >>>> Nov 15 21:51:51.613: *****CCB found in UAS Request table. ccb=0x653BCD0C >>>> Nov 15 21:51:51.613: CCSIP-SPI-CONTROL: act_sentsucc_new_message >>>> Nov 15 21:51:51.617: CCSIP-SPI-CONTROL: act_sentsucc_new_message_request >>>> Nov 15 21:51:51.617: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE >>>> Nov 15 21:51:51.617: sip_stats_status_code >>>> Nov 15 21:51:51.617: sipSPICheckRequest: CheckRequest fail on method 102 >>>> error code: 2 and status: 500 >>>> >>>> >>>> Is there any parameter that I can set to ensure that message leave SER >>>> in the order they enter? Should the Cisco be able to handle this >>>> condition (I check a number of different sections of the RFC but >>>> couldn't find anything specific for either side)? >>>> >>>> Thanks, >>>> Evan >>>> >>>> _______________________________________________ >>>> Serdev mailing list >>>> serdev(a)lists.iptel.org >>>> http://lists.iptel.org/mailman/listinfo/serdev >>> _______________________________________________ >>> Serdev mailing list >>> serdev(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serdev > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers [View Less]

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[Serusers] mangler module
by harry gaillac 17 Nov '05

17 Nov '05

Hello, I try to change the contact field in SIP HF with mangler module if (src_ip == 10.0.0.0/8) encode_contact("enc_prefix","193.175.135.38"); However asterisk receive a bad AOR because of "enc_prefix" . How can I rewrite contact field with public ip without adding this prefix in th user part ? Regards Harry Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 ---------------- | asterisk pbx | ---… [View More]

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[Serusers] RADIUS/LDAP
by Felipe Louback 17 Nov '05

17 Nov '05

I'm putting my SER to authenticate on a LDAP base using RADIUS. But when I try to authenticate any client, I get the following error: auth: type "LDAP" Processing the authenticate section of radiusd.conf modcall: entering group Auth-Type for request 0 rlm_ldap: - authenticate rlm_ldap: Attribute "User-Password" is required for authentication. modcall[authenticate]: module "ldap" returns invalid for request 0 Into my LDAP, the correct field is userPassword and not User-Password, as RADIUS … [View More]

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[Serusers] Compile problem on SLES 9 EM64T
by Marcello Lupo 17 Nov '05

17 Nov '05

Hi i'm getting this errors of files not found compiling SER 0.9.4 (both from onsip.org or not) and Openser stable 1.0.0 on Suse Linux Enterprise Server 9 x86_64 . Can you address me to solve the issue? After the compilation the SER compile but i don't know if it can be an operational problem. After make proper i get: venus:/data/src/ser-0.9.4 # make prefix=/data/ser Makefile.rules:80: action.d: No such file or directory Makefile.rules:80: crc.d: No such file or directory Makefile.rules:80:… [View More] daemonize.d: No such file or directory Makefile.rules:80: data_lump.d: No such file or directory Makefile.rules:80: data_lump_rpl.d: No such file or directory Makefile.rules:80: dprint.d: No such file or directory Makefile.rules:80: dset.d: No such file or directory Makefile.rules:80: error.d: No such file or directory Makefile.rules:80: fifo_server.d: No such file or directory Makefile.rules:80: flags.d: No such file or directory Makefile.rules:80: forward.d: No such file or directory Makefile.rules:80: hash_func.d: No such file or directory Makefile.rules:80: ip_addr.d: No such file or directory Makefile.rules:80: main.d: No such file or directory Makefile.rules:80: md5.d: No such file or directory Makefile.rules:80: md5utils.d: No such file or directory Makefile.rules:80: modparam.d: No such file or directory Makefile.rules:80: msg_translator.d: No such file or directory Makefile.rules:80: pass_fd.d: No such file or directory Makefile.rules:80: proxy.d: No such file or directory Makefile.rules:80: qvalue.d: No such file or directory Makefile.rules:80: re.d: No such file or directory Makefile.rules:80: receive.d: No such file or directory Makefile.rules:80: resolve.d: No such file or directory Makefile.rules:80: route.d: No such file or directory Makefile.rules:80: route_struct.d: No such file or directory Makefile.rules:80: script_cb.d: No such file or directory Makefile.rules:80: socket_info.d: No such file or directory Makefile.rules:80: sr_module.d: No such file or directory Makefile.rules:80: stats.d: No such file or directory Makefile.rules:80: tcp_main.d: No such file or directory Makefile.rules:80: tcp_read.d: No such file or directory Makefile.rules:80: timer.d: No such file or directory Makefile.rules:80: tsend.d: No such file or directory Makefile.rules:80: udp_server.d: No such file or directory Makefile.rules:80: unixsock_server.d: No such file or directory Makefile.rules:80: usr_avp.d: No such file or directory Makefile.rules:80: mem/f_malloc.d: No such file or directory Makefile.rules:80: mem/mem.d: No such file or directory Makefile.rules:80: mem/memtest.d: No such file or directory Makefile.rules:80: mem/q_malloc.d: No such file or directory Makefile.rules:80: mem/shm_mem.d: No such file or directory Makefile.rules:80: mem/vq_malloc.d: No such file or directory Makefile.rules:80: parser/hf.d: No such file or directory Makefile.rules:80: parser/msg_parser.d: No such file or directory Makefile.rules:80: parser/parse_allow.d: No such file or directory Makefile.rules:80: parser/parse_content.d: No such file or directory Makefile.rules:80: parser/parse_cseq.d: No such file or directory Makefile.rules:80: parser/parse_disposition.d: No such file or directory Makefile.rules:80: parser/parse_diversion.d: No such file or directory Makefile.rules:80: parser/parse_event.d: No such file or directory Makefile.rules:80: parser/parse_expires.d: No such file or directory Makefile.rules:80: parser/parse_fline.d: No such file or directory Makefile.rules:80: parser/parse_from.d: No such file or directory Makefile.rules:80: parser/parse_hname2.d: No such file or directory Makefile.rules:80: parser/parse_hostport.d: No such file or directory Makefile.rules:80: parser/parse_methods.d: No such file or directory Makefile.rules:80: parser/parse_nameaddr.d: No such file or directory Makefile.rules:80: parser/parse_param.d: No such file or directory Makefile.rules:80: parser/parse_rpid.d: No such file or directory Makefile.rules:80: parser/parse_rr.d: No such file or directory Makefile.rules:80: parser/parse_sipifmatch.d: No such file or directory Makefile.rules:80: parser/parse_to.d: No such file or directory Makefile.rules:80: parser/parse_uri.d: No such file or directory Makefile.rules:80: parser/parse_via.d: No such file or directory Makefile.rules:80: parser/parser_f.d: No such file or directory Makefile.rules:80: parser/digest/digest.d: No such file or directory Makefile.rules:80: parser/digest/digest_parser.d: No such file or directory Makefile.rules:80: parser/digest/param_parser.d: No such file or directory Makefile.rules:80: parser/contact/contact.d: No such file or directory Makefile.rules:80: parser/contact/parse_contact.d: No such file or directory Makefile.rules:80: db/db.d: No such file or directory Makefile.rules:80: db/db_fifo.d: No such file or directory Makefile.rules:80: lex.yy.d: No such file or directory Makefile.rules:80: cfg.tab.d: No such file or directory bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -DNAME='"ser"' -DVERSION='"0.9.4"' -DARCH='"x86_64"' -DOS='"linux"' -DCOMPILER='"gcc 3.3.3"' -D__CPU_x86_64 -D__OS_linux -DCFG_DIR='"/data/ser/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c action.c -o action.o ... ... I checked in the main dir of SER/openser src and the files are present. May be the Makefile is searching somewhere else ? Thanks, Bye, Marcello [View Less]

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Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting
by Klaus Darilion 17 Nov '05

17 Nov '05

Hi! I think we should leave (open)ser radius as it is. If you need adopt the accounting packets to fit your billing application, I would add a radius proxy, which translates the acounting requests according to your needs and forward it to the billing application. regards klaus Lenir wrote: >>We could determine who sent the BYE from the value of ftag parameter in >>Route header field and swap the values, the question is whether this >>makes sense. The billing system could … [View More]take all info from start requests > > > I think it would make sense just from a uniformity stand point. > > >>and use the stop request to calculate the session duration only. > > > I don't know if that's the case, but this would probably vary from billing > system to billing system. I think if we keep it uniform, then it would never > be a problem. Specially, for billing systems that only support stop-only > records and billing systems that authenticate by using PIN prefixes, > specially when one of the parties is not a user but a PSTN number. On my > billing platform it wont work, it complains about the discrepancy, and the > fact that when the callee hangs up it doesn't recognize that the CDR is for > a PSTN user. > > Lenir > > -----Original Message----- > From: 'Jan Janak' [mailto:jan@iptel.org] > Sent: Wednesday, November 16, 2005 3:45 PM > To: Lenir > Cc: serdev(a)iptel.org; serusers(a)iptel.org; users(a)openser.org; > devel(a)openser.org > Subject: Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius > Accounting > > Yes, this can happen if the callee sends BYE. In this case the URIs in >>From and To header fields are swapped. > > The only SIP RADIUS related document (which has expired a long time ago) > says that Calling-Station-Id is the URI from From header field of any > SIP request. Thus -- according this I-D it is correct. But use of RADIUS > with SIP is underspecified so there is no real standard for this. > > I don't think it matters that much, because other attributes, namely > Acct-Session-Id are used to match start and stop radius requests. > > We could determine who sent the BYE from the value of ftag parameter in > Route header field and swap the values, the question is whether this > makes sense. The billing system could take all info from start requests > and use the stop request to calculate the session duration only. > > Jan. > > On 16-11-2005 15:04, Lenir wrote: > >>Ok, that put the right things in place. However, here's a twist: >>Either way, wit switching Sip-Translated-Request-URI and Called-Station-ID >>around, or not, the CDRs are sent differently depending on who "hands up" >>the call. For example, notice the User-Name, Calling-Station-ID and >>Called-Station-ID: >> >>User 1000 calls 15615551212, and user 1000 hangs up: >>===================================================== >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39228, id=206, >>length=306 >> Acct-Status-Type = Start >> Service-Type = Sip-Session >> Acct-Terminate-Cause = 200 >> Error-Cause = Invite >> User-Name = "1000(a)xx.xx.xx.xx" >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" >> Sip-Translated-Request-URI = >>"sip:15615551212@xx.xx.xx.xx;user=phone" >> Called-Station-Id = >>"sip:949449852150#15615551212@gateway;user=phone" >> Acct-Session-Id = > > "000d2890-d47f01f5-6410e8bd-4baf0435(a)66.232.8.116" > >> Sip-To-Tag = "33C5A3C-76A" >> Sip-From-Tag = "000d2890d47f24f85e6c0b91-0092ab53" >> Sip-CSeq = "102" >> NAS-IP-Address = xx.xx.xx.xx >> NAS-Port = 5060 >> Acct-Delay-Time = 0 >>rlm_sql (sql): Reserving sql socket id: 15 >>rlm_sql (sql): Released sql socket id: 15 >>Sending Accounting-Response of id 206 to xx.xx.xx.xx:39228 >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39228, id=207, >>length=300 >> Acct-Status-Type = Stop >> Service-Type = Sip-Session >> Acct-Terminate-Cause = 200 >> Error-Cause = 8 >> User-Name = "1000(a)xx.xx.xx.xx" >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" >> Sip-Translated-Request-URI = >>"sip:15615551212@xx.xx.xx.xx;user=phone" >> Called-Station-Id = >>"sip:949449852150#15615551212@gateway:5060" >> Acct-Session-Id = > > "000d2890-d47f01f5-6410e8bd-4baf0435(a)66.232.8.116" > >> Sip-To-Tag = "33C5A3C-76A" >> Sip-From-Tag = "000d2890d47f24f85e6c0b91-0092ab53" >> Sip-CSeq = "103" >> NAS-IP-Address = xx.xx.xx.xx >> NAS-Port = 5060 >> Acct-Delay-Time = 0 >> >> >>User 1000 calls 15615551212, and user 15615551212 hangs up: >>=========================================================== >> >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39227, id=200, >>length=306 >> Acct-Status-Type = Start >> Service-Type = Sip-Session >> Acct-Terminate-Cause = 200 >> Error-Cause = Invite >> User-Name = "1000(a)xx.xx.xx.xx" >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" >> Sip-Translated-Request-URI = >>"sip:15615551212@xx.xx.xx.xx;user=phone" >> Called-Station-Id = >>"sip:949449852150#15615551212@gateway;user=phone" >> Acct-Session-Id = > > "000d2890-d47f01f4-372dff4b-793d4acc(a)66.232.8.116" > >> Sip-To-Tag = "33B37E4-77B" >> Sip-From-Tag = "000d2890d47f24f639ccef77-4ef08b02" >> Sip-CSeq = "102" >> NAS-IP-Address = xx.xx.xx.xx >> NAS-Port = 5060 >> Acct-Delay-Time = 0 >>rlm_sql (sql): Reserving sql socket id: 1 >>rlm_sql (sql): Released sql socket id: 1 >>Sending Accounting-Response of id 200 to xx.xx.xx.xx:39227 >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39227, id=201, >>length=286 >> Acct-Status-Type = Stop >> Service-Type = Sip-Session >> Acct-Terminate-Cause = 200 >> Error-Cause = 8 >> User-Name = "15615551212(a)xx.xx.xx.xx" >> Calling-Station-Id = "sip:15615551212@xx.xx.xx.xx;user=phone" >> Sip-Translated-Request-URI = "sip:1000@xx.xx.xx.xx" >> Called-Station-Id = "sip:1000@66.232.8.116:5061" >> Acct-Session-Id = > > "000d2890-d47f01f4-372dff4b-793d4acc(a)66.232.8.116" > >> Sip-To-Tag = "000d2890d47f24f639ccef77-4ef08b02" >> Sip-From-Tag = "33B37E4-77B" >> Sip-CSeq = "101" >> NAS-IP-Address = xx.xx.xx.xx >> NAS-Port = 5060 >> Acct-Delay-Time = 0 >> >> >>Shouldn't it always be that the User-Name attribute is the user who >>initiated the call? >> >>Lenir >> >>-----Original Message----- >>From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On > > Behalf > >>Of 'Jan Janak' >>Sent: Wednesday, November 16, 2005 1:53 PM >>To: lsantiago(a)globalgatewaycom.com >>Cc: serdev(a)iptel.org; serusers(a)iptel.org; users(a)openser.org; >>devel(a)openser.org >>Subject: Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius >>Accounting >> >>Either reconfigure the billing application to use >>Sip-Translated-Request-URI (recommended) or change the ID of the >>attribute in the radiusclient dictionary as suggested by Bogdan. >> >>In this particular example you should set Called-Station-Id to 107. >> >> Jan. >> >>On 16-11-2005 13:09, 'Lenir' wrote: >> >>>Ok, so how do I tell acc.so to send Sip-Translated-Request-URI as the >>>Called-Station-ID instead? >>> >>>Lenir Santiago, Partner >>>Global Gateway Communications, Inc. >>>2500 Quantum Lakes Drive, Ste. 203 >>>Boynton Beach, FL 33426 >>>Office: 561-853-2213 >>>Cell: 561-722-0966 >>>Fax: 877-875-0060 >>>http://www.globalgatewaycom.com >>> >>>-----Original Message----- >>>From: 'Jan Janak' [mailto:jan@iptel.org] >>>Sent: Wednesday, November 16, 2005 11:29 AM >>>To: Bogdan-Andrei Iancu >>>Cc: Lenir; 'Lenir Santiago'; serdev(a)iptel.org; serusers(a)iptel.org; >>>devel(a)openser.org; users(a)openser.org >>>Subject: Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius >>>Accounting >>> >>>Why would you do something like this ? All the info that is needed is in >>>Sip-Translated-Ruquest-ID below. You can see it in the RADIUS dump: >>> >>>Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway:5060" >>> >>> Jan. >>> >>>On 16-11-2005 16:50, Bogdan-Andrei Iancu wrote: >>> >>>>Hi Lenir, >>>> >>>>you can keep the modified To value into an AVP and use the extra >>>>accounting feature to log it. See >>>> http://www.openser.org/docs/modules/1.1.x/acc.html#AEN115 >>>>and >>>> http://www.openser.org/docs/modules/1.1.x/acc.html#AEN350 >>>> >>>>the problem is how to set it as Called-Station-ID attribute in order > > to > >>>>avoid overlapping. You can do a trick by playing with the radius >>>>dictionary (on proxy side) : >>>> changed the Called-Station-ID to a different ID attribute >>>> set extra accounting with the original attribute ID of >>> >>>Called-Station-ID >>> >>>>in this way the acc will set de default TO as something else and you > > AVP > >>>>as Called-Station-ID ;) >>>> >>>>regards, >>>>bogdan >>>> >>>> >>>>Lenir wrote: >>>> >>>> >>>>>Well, if userA dials to:30, 30 gets translated to:15615551212 and SER >>> >>>dials >>> >>>>>15615551212, how can I make the acc.so module send the modified >>>>>to:15615551212 as the Called-Station-ID attribute? >>>>> >>>>>This is important for billing, specially if someone forwards to a > > long > >>>>>distance or international long distance number. >>>>> >>>>>-----Original Message----- >>>>>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@iptel.org] > > On > >>>>>Behalf Of Jan Janak >>>>>Sent: Wednesday, November 16, 2005 3:15 AM >>>>>To: Lenir Santiago >>>>>Cc: serdev(a)iptel.org; serusers(a)iptel.org; users(a)openser.org; >>>>>devel(a)openser.org >>>>>Subject: [Serusers] Re: [Serdev] Inaccurate Radius Accounting >>>>> >>>>>Hello, >>>>> >>>>>Called-Station-Id attribute always contains the To URI (not >>>>>Request-URI). You can find the outbound Request-URI in >>>>>Sip-Translated-Request-URI attribute. >>>>> >>>>>Even if you rewrite To header field, it will never be accounted in >>>>>Called-Station-Id. >>>>> >>>>>Jan. >>>>> >>>>>On 15-11-2005 21:59, Lenir Santiago wrote: >>>>> >>>>> >>>>> >>>>>>Hey guys, >>>>>> >>>>>> >>>>>> >>>>>>I'm using Radius to do pretty much everything regarding my users, > > i.e. > >>>>>>registrations, authorization, features and preferences. I > > implemented > >>my >> >>>>>> >>>>>> >>>>> >>>>>own >>>>> >>>>> >>>>> >>>>>>way of doing speeddial by using AVPs provided by radius and AVPOPs >>> >>>module, >>> >>>>>>Here is the snippet: >>>>>> >>>>>> >>>>>> >>>>>>AVP-SIP = speedial:30:15615551212 >>>>>> >>>>>> >>>>>> >>>>>> avp_subst("s:caller_speedial/i:99", > > "/(.*):(.*)/\2/"); > >>>>>>#15615551212 >>>>>> >>>>>> avp_printf("i:98","$rU:*"); #30 >>>>>> >>>>>> avp_printf("i:97","sip:$avp($temp2)@$td"); # >>>>>>sip:15615551212@xx.xx.xx.xx >>>>>> >>>>>> avp_printf("i:96","<sip:$avp($temp2)@$td>\r\n"); # >>>>>><sip:15615551212@xx.xx.xx.xx>. >>>>>> >>>>>> xlog("L_N"," SPEEDIALING: $fU is speed-dialing >>>>>>$hdr(To)->$avp($temp2)\n"); >>>>>> >>>>>> if (avp_check("s:caller_speedial", "fm/$temp1/g")) { >>>>>> >>>>>> >>>>>> >>>>>> #avp_pushto("$To","i:96"); >>>>>> >>>>>> #avp_pushto("$ruri","i:97"); >>>>>> >>>>>> #avp_pushto("$duri","i:97"); >>>>>> >>>>>> #subst('/^To: (.*)(a)(.*)/To: "$avp($temp2)" >>>>>><$avp($temp2)@\2/'); >>>>>> >>>>>> #append_hf("To: \"5615551212\" >>>>>><15615551212(a)xx.xx.xx.xx>;user=phone"); >>>>>> >>>>>> >>>>>>#subst_uri('/^sip:(.*)@(.*)$/sip:$avp($temp2)@\2/i'); >>>>>> >>>>>> #subst_user('/(.*)/$avp($temp2)/'); >>>>>> >>>>>> #remove_hf("To"); >>>>>> >>>>>> rewriteuser("15615551212"); >>>>>> >>>>>> xlog("L_N"," SPEEDIALING: $fU is dialing >>>>>> >>>>>> >>>>> >>>>>$hdr(To) >>>>> >>>>> >>>>> >>>>>>$avp($temp2)\n"); >>>>>> >>>>>> avp_delete("$temp1/g"); >>>>>> >>>>>> avp_delete("$temp2/g"); >>>>>> >>>>>> >>>>>> >>>>>> }; >>>>>> >>>>>> >>>>>> >>>>>>In this scenario, im forcing it instead of getting the value from > > AVPs > >>>>>> >>>>>> >>>>> >>>>>just >>>>> >>>>> >>>>> >>>>>>for ease of showing my point. The problem that I'm having is that no >>>>>> >>>>>> >>>>> >>>>>matter >>>>> >>>>> >>>>> >>>>>>what I do to change the To header or change the R-URI, radius ALWAYS >>> >>>sends >>> >>>>>>the accounting records with the original R-URI, in my case >>> >>>30(a)xx.xx.xx.xx, >>> >>>>>>how can I get acc.so to send the final resulting URI >>>>>>(15615551212(a)xx.xx.xx.xx) in the Called-Station-Id AVP? >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, >> >>id=97, >> >>>>>>length=298 >>>>>> >>>>>> Acct-Status-Type = Start >>>>>> >>>>>> Service-Type = Sip-Session >>>>>> >>>>>> Acct-Terminate-Cause = 200 >>>>>> >>>>>> Error-Cause = Invite >>>>>> >>>>>> User-Name = "1000(a)xx.xx.xx.xx" >>>>>> >>>>>> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" >>>>>> >>>>>> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" >>>>>> >>>>>> Sip-Translated-Request-URI = >>>>>>"sip:949449852150#15615551212@gateway;user=phone" >>>>>> >>>>>> Acct-Session-Id = >>>>>> >>>>>> >>>>> >>>>>"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" >>>>> >>>>> >>>>> >>>>>> Sip-To-Tag = "230383F4-2AF" >>>>>> >>>>>> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" >>>>>> >>>>>> Sip-CSeq = "102" >>>>>> >>>>>> NAS-IP-Address = xx.xx.xx.xx >>>>>> >>>>>> NAS-Port = 5060 >>>>>> >>>>>> Acct-Delay-Time = 0 >>>>>> >>>>>>rlm_sql (sql): Reserving sql socket id: 12 >>>>>> >>>>>>rlm_sql (sql): Released sql socket id: 12 >>>>>> >>>>>>Sending Accounting-Response of id 97 to xx.xx.xx.xx:39216 >>>>>> >>>>>>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, >> >>id=98, >> >>>>>>length=292 >>>>>> >>>>>> Acct-Status-Type = Stop >>>>>> >>>>>> Service-Type = Sip-Session >>>>>> >>>>>> Acct-Terminate-Cause = 200 >>>>>> >>>>>> Error-Cause = 8 >>>>>> >>>>>> User-Name = "1000(a)xx.xx.xx.xx" >>>>>> >>>>>> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" >>>>>> >>>>>> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" >>>>>> >>>>>> Sip-Translated-Request-URI = >>>>>>"sip:949449852150#15615551212@gateway:5060" >>>>>> >>>>>> Acct-Session-Id = >>>>>> >>>>>> >>>>> >>>>>"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" >>>>> >>>>> >>>>> >>>>>> Sip-To-Tag = "230383F4-2AF" >>>>>> >>>>>> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" >>>>>> >>>>>> Sip-CSeq = "103" >>>>>> >>>>>> NAS-IP-Address = xx.xx.xx.xx >>>>>> >>>>>> NAS-Port = 5060 >>>>>> >>>>>> Acct-Delay-Time = 0 >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>Thanks in advance! >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>Lenir >>>>>>_______________________________________________ >>>>>>Users mailing list >>>>>>Users(a)openser.org >>>>>>http://openser.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>>> >>>>>> >>> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > [View Less]

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[Users] Multiple mediaproxy instances
by Fernando Rodriguez 17 Nov '05

17 Nov '05

Hello, I have an openser box that has multiple interfaces each one with its own ip address and i was wondering how to run various mediaproxy instances in the same machine per ip address so I can have communication between interfaces and proxy all traffic coming and going between the UA, I have been trying to get information on this and I found that is valid to do this but I cant find any info on how to doit, actually if people can give me hint or a basic setup I could write a howto , … [View More]

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[Users] Can I run this SIP server without running mysql.
by Mahesh Krishnan 17 Nov '05

17 Nov '05

I have the SIP server running. But I am not able to add any users. Do I have to install mysql to add users. Unfortunately the documentation is also not good. It mentions only how to start the server and then thats it. I stop it basically by killing it. [Ctrl-C]. I installed it by compiling and linking it from source. Basically this is what I am looking at. 1. Installed and started SIP server. -- Works 2. try to add a user with openserctl -- fails. no mysql connection. I do not … [View More]

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Re: [Users] Usrloc question
by Daniel-Constantin Mierla 17 Nov '05

17 Nov '05

Hello, there is a solution using the branch_route[] in openser. For each branch created after the usrloc lookup you can make different checks. So, in the branch route you have to check the IP in the R-URI and if matches the asterisk IP, then drop it (http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook#drop). There is an example how to check source and destination addresses, although in your case you can do it even without avps: http://www.voice-system.ro/docs/avpops/ar01s08.… [View More]html#ex_org_dst Cheers, Daniel On 11/15/05 17:05, Darren Nay wrote: > > Hey all, > > I have a question regarding usrloc. I have run into a problem.. > > We have static routes sent to an asterisk server for all of our SIP > usernames. In addition our IAD’s will register with the same username, > so that calls coming into our switch for that username will be routed > to both the asterisk box and the SIP IAD. This way, whichever endpoint > (IAD or asterisk) answers the call first will take the call. > > For example. > > root:/ # serctl ul show +18646404810 > > <sip:+18646404810@192.168.1.60>;q=1;expires=-1012151 > > <sip:+18646404810@192.168.1.157:5060>;q=;expires=403 > > 192.168.1.60 is the asterisk server. This is a static route added by > serctl. > > 192.168.1.157 is my IAD which registers with the switch every 10 minutes. > > So when calls are made to (864) 640-4810 then SER will send an INVITE > to both location. > > I explained all of this just to explain now what my problem is, and > ask if anyone may know a possible solution. > > Now, we also use asterisk to perform call fwd’ing functions. Asterisk > will answer the call and then originate another call out back to SER > to a new location. Now the problem! (finally!) This call fwd’ing > method works very well in most cases, except that if the call fwd’ing > is being sent to another location registered with SER then it will be > redirected back to asterisk again, albeit to a different URI, and > asterisk will kill the call because it thinks that it has looped > (which I guess it has… sort of). > > So, I’m wondering if there is possibly a way to retrieve only the > usrloc locations that don’t contain the IP address 192.168.1.60 in the > contact URI? This way I can just check if the src_ip is 192.168.1.60 > and if so then retrieve all the usrloc locations – without asterisk – > and the call will not be redirected back to asterisk. > > Is this possible? Or if anyone has any other ideas that may help then > I am definitely open to suggestions. > > Thanks for your help!! > > Darren Nay > > Ionosphere, Inc > > VoIP Network Development > > dnay(a)ionosphere.net <mailto:dnay@ionosphere.net> > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > [View Less]

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[Serusers] Re: [Serdev] ACK before INVITE on re-INVITE causes CiscoAS's to return a 500 error
by Ettore Benedetti 17 Nov '05

17 Nov '05

Hallo Evan, I reckon Nils is right on the matter. The ACK of the dialog-forming INVITE is actually the first in-dialog request and, as a matter of fact, it is a separate transaction. As far as the UAS is concerned, the dialog is established as soon as the 200 response is sent. The Cisco GW just misbehaves provided the first INVITE from Asterisk did contain an SDP offer (didn't it?). Ettore Benedetti THALES COMMUNICATIONS B.V. Bestevaer 46, 1271 ZA Huizen The Netherlands Unclassified … [View More]>>> Evan Borgstrom <evan.borgstrom(a)ca.mci.com> 11/16/05 5:27:52 PM >>> Hey Nils, You are correct that each message is handled by a different process, as it to be expected, and I know that SER is not call statefull so my concern lies around the fact that SER is delivering an INVITE which requires FAR more processing and at least 2 database hits before a loosely routed ACK which requires little to none for processing. I'm going to differ with you, and my self from last night, on the views about the RFC as it does indeed cover the problem (I was just looking at it the wrong way before). A re-INVITE cannot modify a dialog that does not exist, and without the ACK being delivered in response to the 200 OK the dialog is not yet created which is why the Cisco is returning a CheckRequest failure since there's no dialog to map the re-INVITE to. So, the device is correct in returning an error. When the Cisco returns the 500 error it's only for the re-INVITE so the initial call stays established and a BYE is still required to tear it down. The problem is that Asterisk is expecting that to work and has already re-INVITED to the client so there's no audio and to the caller the call failed. -Evan Nils Ohlmeier wrote: > Evan, > > I would not invest the time to look into the SER side of your problem. Because > this is a general problem. Instead of SER any other network element (router, > switch, etc.) could change the order of the packets. > BTW I guess this problem occurs on your SER because the ACK and the INVITE are > handled by different SER processes. And as SER is not call-statefull there is > no parameter to solve this situation. > > I fear you are right that this problem is not covered by any RFC, but the > solutions seems logical to me: > if this a generic problem, then either the UAC is not allowed to send a > re-INVITE until the first INVITE is completed, or the UAS has to accept a > re-INVITE after sending the 200 OK. But how does the the UAC know that the > ACK reached the UAS so that it can send the re-INVITE now? It cant know that, > so it would have to wait for a timer before it could send the re-INVITE. > On the other side the INVITE transaction is completed for the UAS by sending > the 200 OK (the ACK is a new transaction, according to the branch). > Thus I think the UAS (the Cisco gateway) should accept the re-INVITE without > complaining. > > Just for curiosity: is the call established or does the Cisco gateway drop the > call after sending the 500? > > Regards > Nils > > On Wednesday 16 November 2005 07:51, Greger V. Teigre wrote: >> Evan, >> I'm not able to match your text with the messages you show. The message >> timestamped 16:51:51.559657, does it go to Cisco? It is shown to go to >> Asterisk and it doesn't make sense. >> However, until the Cisco gw receives the ACK from Asterisk, no call has >> been established and the reINVITE cannot be matched. IMO, this is not a >> Cisco bug. It seems very strange indeed that ser should send a reINVITE >> before the ACK in the same call. Even without FIFO, an INVITE is likely to >> take more time to process than a loose routed ACK. I would try to log more >> thoroughly in SER to see what happens in processing the ACK and the >> reINVITE. It could be a problem in your ser.cfg, but it seems strange >> indeed. >> g-) >> >> ----- Original Message ----- >> From: "Evan Borgstrom" <evan.borgstrom(a)ca.mci.com> >> To: <serusers(a)lists.iptel.org>; <serdev(a)lists.iptel.org> >> Sent: Wednesday, November 16, 2005 12:05 AM >> Subject: [Serdev] ACK before INVITE on re-INVITE causes Cisco AS's to >> return a 500 error >> >>> Hey all, >>> >>> This has more to do with Asterisk & Cisco AS's IMO but I thought it >>> would be interesting to get some responses from this group before I >>> begin the battle with Cisco to get a bug report underway. >>> >>> We have an Asterisk box acting as a PBX sending calls to our SER >>> instances. The asterisk box has "reinvite=yes" in the sip.conf file >>> which causes Asterisk to send a re-INVITE for the call once it receives >>> the 200 OK. This re-INVITE has it's SDP changed to that of the IP Phone >>> behind the Asterisk box so that all media doesn't need to traverse it. >>> This works fine except in certain instances where messages get out of >>> order and it causes our Cisco GW's to error the call with a generic 500 >>> message. When the messages are in the correct order the 500 error does >>> not appear. >>> >>> So here's the situation, user behind the asterisk box picks up the >>> phone and dials a PSTN destination. Asterisk fires off the initial >>> invite and everything proceeds as normal and we eventually receive the >>> 200 OK message to which the Asterisk box replies with ACK and a new >>> INVITE that come in this order (X = Asterisk, Y = SER, Z = Cisco): >>> >>> U 2005/11/15 16:51:51.557322 X.X.X.X:5060 -> Y.Y.Y.Y:5060 >>> ACK >>> >>> # >>> U 2005/11/15 16:51:51.557948 X.X.X.X:5060 -> Y.Y.Y.Y:5060 >>> INVITE >>> >>> SER accordingly processes the two messages and replies 100 Trying to >>> the new INVITE message. Here's where the problem occurs, SER then sends >>> these messages out to the PSTN gateway but the order has been reversed: >>> >>> # >>> U 2005/11/15 16:51:51.558937 Y.Y.Y.Y:5060 -> X.X.X.X:5060 >>> SIP/2.0 100 trying >>> >>> # >>> U 2005/11/15 16:51:51.559000 Y.Y.Y.Y:5060 -> Z.Z.Z.Z:5060 >>> INVITE >>> >>> # >>> U 2005/11/15 16:51:51.559657 Y.Y.Y.Y:5060 -> X.X.X.X:5060 >>> ACK >>> >>> When this happens the Cisco GW replies with a 500 error message and the >>> following is sent to the Cisco's logs: >>> >>> # >>> U 2005/11/15 16:51:51.711355 Z.Z.Z.Z:5060 -> Y.Y.Y.Y:5060 >>> SIP/2.0 500 Server Internal Error. >>> >>> >>> Nov 15 21:51:51.613: HandleUdpSocketReads :Msg enqueued for SPI with >>> IPaddr: Y.Y.Y.Y:5060 >>> Nov 15 21:51:51.613: *****CCB found in UAS Request table. ccb=0x653BCD0C >>> Nov 15 21:51:51.613: CCSIP-SPI-CONTROL: act_sentsucc_new_message >>> Nov 15 21:51:51.617: CCSIP-SPI-CONTROL: act_sentsucc_new_message_request >>> Nov 15 21:51:51.617: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE >>> Nov 15 21:51:51.617: sip_stats_status_code >>> Nov 15 21:51:51.617: sipSPICheckRequest: CheckRequest fail on method 102 >>> error code: 2 and status: 500 >>> >>> >>> Is there any parameter that I can set to ensure that message leave SER >>> in the order they enter? Should the Cisco be able to handle this >>> condition (I check a number of different sections of the RFC but >>> couldn't find anything specific for either side)? >>> >>> Thanks, >>> Evan >>> >>> _______________________________________________ >>> Serdev mailing list >>> serdev(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serdev >> _______________________________________________ >> Serdev mailing list >> serdev(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serdev _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers [View Less]

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[Serusers] ser High Availability
by Arne Van Theemsche 17 Nov '05

17 Nov '05

Hi I've been thinking on making a HA setup with multiple ser's. Wat would be the best solution? 1) Sharing a usrloc database? Then I think the only way is to use db_mode 1 (every read out of a database), but would cause a lot of overhead because of every request needs a sqlquery. Method 2 doesn't seem to work, because if I delete a record manually from database, the internal DB doesn't get synced to that 2) evey SER automaticly dials also to all ser's in the clusters, and the one's who … [View More]

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[Users] 302 Redirect behavior
by Brandon Price 17 Nov '05

17 Nov '05

I am trying to use the uac_redirect module to duplicate the behavior that asterisk performs when it receives a 302-Moved Temporarily from a registered user agent. It basically extracts the number to forward to from the SIP contact address in the 302 reply. For example if "sip: 13143212222(a)ser1.voipnoc.net" is the sip contact address then 13143212222 is the number (of coarse). It then executes whatever code matches that pattern. This works perfectly for me but ends up making 2 … [View More]

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[Serusers] ERROR: exec_str: cmd /etc/ser/sipldap failed. exit_status=-1, errno=10: No child processes
by davor jovanovic 17 Nov '05

17 Nov '05

Hi all, I'm using SER 0.9.3 and trying to do LDAP query but I keep getting this message: 0(17313) INFO: dont_fork turned on, living on 0(17313) exec_str: rtrim 0(17313) ERROR: exec_str: cmd /etc/ser/sipldap failed. exit_status=-1, errno=10: No child processes can someone explain to me why is that and how can I fix it? when I'm doing lookup with: (!lookup("location")) { if (!exec_dset("/etc/ser/sipldap")) { sl_send_reply("404", "Not Found"); break; … [View More]

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[Serusers] Can rtpproxy scale?
by tcchan 17 Nov '05

17 Nov '05

Hi All, Can anybody tell me whether it is possible to have multiple rtpproxy running with ser, i.e., can rtpproxy scales as mediaproxy? I remembered sometime ago (ser ver 0.84) that it wasn't possible and I switched to mediaproxy because of that. If it is possible to scale now, since which ver? Thanks for any help available. Regards, TC Chan

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Re: [Serusers] problem with rtpproxy
by Greger V. Teigre 17 Nov '05

17 Nov '05

Rajesh, Again, copy the list. The config you sent is NOT the onsip.org config and as I have said: You call force_rtp_proxy several times. g-) ----- Original Message ----- From: rajesh volluru To: Greger V. Teigre Sent: Thursday, November 10, 2005 10:52 PM Subject: Re: [Serusers] problem with rtpproxy hi I see that ser is appending to new IP address multiple times in the Invite message instead of rewriting the address. I had copied entire ser.cfg iam using. Please look at … [View More]it and let me know if anything is wrong. sdpData arg - v=0 o=Rajesh 0 0 IN IP4 192.168.1.223 s=- c=IN IP4 68.225.20.25069.229.23.5769.229.23.5769.229.23.57 t=0 0 m=audio 351843518435236 RTP/AVP 0 3 4 5 6 8 15 18 m=video 35238 RTP/AVP 26 34 31 a=recvonly a=direction:active a=oldmediaip:192.168.1.223 a=nortpproxy:yes a=nortpproxy:yes java.lang.NumberFormatException: For input string: "351843518435236" Thanks Rajesh # $Id: nat-rtpproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=9 fork=yes log_stderror=yes listen=69.229.23.57 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("69.229.23.57:5060;nat=yes"); if(method == "INVITE") { fix_nated_sdp("3"); force_rtp_proxy(); t_on_reply("1"); }; } else if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); end_media_session(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); #fix_nated_sdp("3"); }; force_rtp_proxy("l"); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("adsl-69-229-23-57.dsl.irvnca.pacbell.net","subscriber")) { www_challenge("adsl-69-229-23-57.dsl.irvnca.pacbell.net","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("adsl-69-229-23-57.dsl.irvnca.pacbell.net","subscriber")) { www_challenge("adsl-69-229-23-57.dsl.irvnca.pacbell.net","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("adsl-69-229-23-57.dsl.irvnca.pacbell.net","subscriber")) { proxy_challenge("adsl-69-229-23-57.dsl.irvnca.pacbell.net","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); use_media_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } "Greger V. Teigre" <greger(a)teigre.com> wrote: Please copy the list. If you use a vanilla onsip.org config, it's pretty thoroughly tested by now, so I would be tempted to say that there is something wrong with the client or the setup of the client. g-) ----- Original Message ----- From: rajesh volluru To: Greger V. Teigre Sent: Wednesday, November 09, 2005 2:04 AM Subject: Re: [Serusers] problem with rtpproxy hi Thanks for your reply. I did configure ser.cfg as said at onsip.org for rtp proxy. But, i still get below exception when I am behind NAT. net.java.sip.communicator.media.MediaException: Incorrect SDP data! at net.java.sip.communicator.media.MediaManager.openMediaStreams(MediaManager.java:307) at net.java.sip.communicator.SipCommunicator.callStateChanged(SipCommunicator.java:758) at net.java.sip.communicator.sip.Call.fireCallStatusChangedEvent(Call.java:250) at net.java.sip.communicator.sip.Call.setState(Call.java:156) at net.java.sip.communicator.sip.CallProcessing.processAck(CallProcessing.java:566) at net.java.sip.communicator.sip.SipManager.processRequest(SipManager.java:1532) at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:204) at gov.nist.javax.sip.EventScanner.run(EventScanner.java:352) at java.lang.Thread.run(Unknown Source) Caused by: javax.sdp.SdpParseException: Could not parse message at javax.sdp.SdpFactory.createSessionDescription(SdpFactory.java:114) at net.java.sip.communicator.media.MediaManager.openMediaStreams(MediaManager.java:302) ... 8 more Please let me know the possible reason for this error. When I start ser, it shows the message ' RTP proxy found. support for RTP enabled'. Thanks Rajesh "Greger V. Teigre" <greger(a)teigre.com> wrote: Are you using the configs from onsip.org? If not, you may find some help there ;-) g-) ----- Original Message ----- From: rajesh volluru To: Sip Users Sent: Tuesday, November 08, 2005 12:37 AM Subject: [Serusers] problem with rtpproxy hi I had installed ser with media and rtp proxy. Below is the output that i get when I start ser. However, iam not able receive any audio transmission when I connect through sip communicator when iam behind NAT. I am able to register with the server. 0(0) fixing /usr/local/lib/ser/modules/maxfwd.so mf_process_maxfwd_header 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so client_nat_test 0(0) fixing /usr/local/lib/ser/modules/textops.so search 0(0) module - fixing ^Record-Route: 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) fixing /usr/local/lib/ser/modules/nathelper.so fix_nated_sdp 0(0) fixing /usr/local/lib/ser/modules/rr.so record_route 0(0) fixing /usr/local/lib/ser/modules/rr.so loose_route 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup 0(0) get_connection(): Connection not found in the pool 0(0) preload_udomain():! ! Table is empty 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/auth_db.so www_authorize 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/auth.so www_challenge 0(0) fixing /usr/local/lib/ser/modules/registrar.so save 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/textops.so search 0(0) module - fixing ^Route: 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so use_media_proxy 0(0) fixing /usr/local/lib/ser/modules/tm.so t_on_reply 0(0) fixing /usr/local/lib/ser/modules/tm.so t_relay 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_reply_error 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so use_media_proxy 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so client_nat_test 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) DEBUG: udp_init: trying SO_RCVBUF: 221184 0(0) DEBUG: setting SO_RCVBUF; set=221184,verify=221184 0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232 0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=221184 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF i! ! s finally 221184 0(0) DBG: open_uac_fifo: opening fifo... 0(0) DEBUG: FIFO created @ /tmp/ser_fifo 0(0) DEBUG: fifo /tmp/ser_fifo opened, mode=432 0(0) init_unixsock_socket: No unix domain socket will be opened 2(0) INFO: fifo process starting: 19946 2(0) DEBUG: init_mod_child (-2): tm 2(0) DEBUG: callid: '57b7f7f7-0(a)69.229.23.57' 2(0) DEBUG: init_mod_child (-2): usrloc 2(0) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): tm 0(19941) DEBUG: callid: '57b7f7f7-19941(a)69.229.23.57' 0(19941) DEBUG: init_mod_child (1): usrloc 0(19941) get_connection(): Connection not found in the pool 1(0) DEBUG: init_mod_child (-1): tm 1(19945) DEBUG: callid: '57b7f7f7-19945(a)69.229.23.57' 1(19945) DEBUG: init_mod_child (-1): usrloc 1(19945) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): auth_db 0(19941) get_connection(): Connection not found in the pool 1(19945) DEBUG: init_mod_child (-1): auth_db 1(19945) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): domain 0(19941) get_connection(): Connection found in the pool 0(19941) DEBUG: init_mod_child (1): nathelper 2(19946) DEBUG: init_mod_child (-2): auth_db 2(19946) get_connection(): Connection not found in the pool 1(19945) DEBUG: init_mod_child (-1): domain 1(19945) DEBUG: init_mod_child (-1): nathelper 0(19941) rtpp_test: RTP proxy found, support for it enabled 1(19945) rtpp_test: RTP proxy found, support for it enabled 2(19946) DEBUG: init_mod_child (-2): domain 2(19946) DEBUG: init_mod_child (-2): nathelper 2(19946) rtpp_test: RTP proxy found, support for it enabled 2(19946) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(19946) DEBUG: register_fifo_cmd: new command (print) registered 2(19946) DEBUG: register_fifo_cmd: new command (uptime) registered 2(19946) DEBUG: register_fifo_cmd: new command (version) registered 2(19946) DEBUG: register_fifo_cmd: new command (pwd) registered 2(19946) DEBUG: register_fifo_cmd: new command (arg) registered 2(19946) DEBUG: register_fifo_cmd: new command (which) registered 2(19946) DEBUG: register_fifo_cmd: new command (ps) registered 2(19946) DEBUG: register_fifo_cmd: new command (kill) registered 2(19946) WARNING: no fifo_db_url given - fifo DB commands disabled! Let me know if you find any configuration error in SER. Please let me if you need any more information. I am trying to configure SER from long time. Please help me. Thanks Rajesh ---------------------------------------------------------------------- Enjoy this Diwali with Y! India Click here ---------------------------------------------------------------------- Enjoy this Diwali with Y! India Click here ---------------------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------------- Enjoy this Diwali with Y! India Click here ------------------------------------------------------------------------------ Enjoy this Diwali with Y! India Click here [View Less]

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RE: [Serusers] SEMS:announcement plug in
by rajesh.rajguru＠wipro.com 17 Nov '05

17 Nov '05

Hi, I get following from SEMS's log, but still announcement is not being heared What could be the problem ---------------------------------------------- (23556) DEBUG: write_to_fifo (SerClient.cpp:252): Write to fifo: completed (23556) DEBUG: getReturnCode (AmRequest.cpp:391): response from Ser: 200 fifo_t reply succeeded (23556) DEBUG: setRAddr (AmRtpStream.cpp:402): RTP remote address set to 10.111 17.16:32900 (23556) DEBUG: negotiate (AmSession.cpp:184): Sending Rtp data to 10.111.17.16/ … [View More]2900 (23556) DEBUG: onSessionStart (Announcement.cpp:116): starting sending announce ..(/usr/local/lib/sems/audio/6789.wav) (23556) DEBUG: trying to read WAV file (23556) DEBUG: tag = <RIFF> (23556) DEBUG: file size = <129710> (23556) DEBUG: tag = <WAVE> (23556) DEBUG: tag = <fmt > (23556) DEBUG: chunk_size = <18> (23556) DEBUG: fmt = <01> (23556) DEBUG: channels = <1> (23556) DEBUG: rate = <8000> (23556) DEBUG: bits/sample = <16> (23556) DEBUG: tag = <fact> (23556) DEBUG: chunk size = <4> (23556) DEBUG: tag = <data> (23556) DEBUG: chunk size = <129660> (23556) DEBUG: ilbc with format parameters : '', mode=30. (23556) DEBUG: streamGet (AmAudio.cpp:548): 0-stuffing packet: adding 420 bytes (packet size=480) (23556) DEBUG: convert (AmAudio.cpp:246): end of stream (23556) DEBUG: onSessionStart (Announcement.cpp:118): ... announce reaches its nd. (23556) DEBUG: sendRequest (AmRequest.cpp:138): dummy sendrequest ----------------------------------------------------- Regards, Rajesh -----Original Message----- From: Xavier TRENTIN [mailto:xtrentin@softiphone.net] Sent: Tuesday, November 15, 2005 11:09 PM To: Rajesh Rajguru (WT01 - Voice & Next Generation Networks) Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SEMS:announcement plug in Try to record your .wav in PCM 8Kbit/s format and configure your phone to use PCMU codec. B.R. xavier rajesh.rajguru(a)wipro.com a écrit : >*********************** >Pas de virus detecté dans l'attachement no filename > >Votre Mail a été vérifié par InterScan. >***********-*********** > > > >Hi, > >We are trying to forward INVITE to SEMS and play announcement. >It doesn't play anything. > >The UA which is sending INVITE is getting time out > >But the same .wav file for announcement if I open from sound recorder >manually ,then it plays. > >How to check that there is problem with codec ? > > >Regards, >Rajesh > >-----Original Message----- >From: Xavier TRENTIN [mailto:xtrentin@softiphone.net] > >Sent: Monday, November 14, 2005 10:00 PM >To: Rajesh Rajguru (WT01 - Voice & Next Generation Networks) >Cc: serusers(a)lists.iptel.org >Subject: Re: [Serusers] SEMS:announcement plug in > >Does it play nothing or does it hang off? >Perhaps the problem could be the Codec you used to record the file. > >B.R. >Xavier. > >rajesh.rajguru(a)wipro.com a écrit : > > > >>*********************** >>Pas de virus detecté dans l'attachement no filename >> >>Votre Mail a été vérifié par InterScan. >>***********-*********** >> >> >> >>Hi, >> >>In our ser.cfg following is mentioned for announcement >> >>---- >>if(!t_write_req("/tmp/am_fifo","announcement")) { >> t_reply("500","error contacting sems"); >>----- >> >>we are using fifo for communicating between ser and sems. >>ser and sems are running on the same server (Fedora) When we start >> >> > > > >>sems, the default announcement file mentioned in sems.conf >> >> > > > >>(default_announce=default_en.wav) is not played. >>Also there is no error in /var/log/messages wrt announcement plug in >> >>What could be the problem ? >> >>How does sems decides which sound recorder to play ? >> >> >>The ll /tmp/am_fifo shows following >> >> >>prw-r--r-- 1 root root 0 Nov 14 09:48 /tmp/am_fifo >> >>Regards, >>Rajesh >> >> >> >> >>Confidentiality Notice >> >> >>The information contained in this electronic message and any >> >> > > > >>attachments to this message are intended for the exclusive use of the >> >> > > > >>addressee(s) and may contain confidential or privileged information. >>If >> >> > > > >>you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments. >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> > > > > > > > > >Confidentiality Notice > > >The information contained in this electronic message and any >attachments to this message are intended for the exclusive use of the >addressee(s) and may contain confidential or privileged information. If >you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments. > > > Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments. [View Less]

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[Serusers] MYSQL Keeps needing to get restarted
by Goran Donev 16 Nov '05

16 Nov '05

I have a SER server works fine when I start everything manually. Yet when the computer reboots I have to go in a restart the Mysql service again because when I try to create a user it says can not connect to Mysql database. Is there a anywhere I can have Linux automatically start Mysql.

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Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting
by 'Jan Janak' 16 Nov '05

16 Nov '05

Yes, this can happen if the callee sends BYE. In this case the URIs in >From and To header fields are swapped. The only SIP RADIUS related document (which has expired a long time ago) says that Calling-Station-Id is the URI from From header field of any SIP request. Thus -- according this I-D it is correct. But use of RADIUS with SIP is underspecified so there is no real standard for this. I don't think it matters that much, because other attributes, namely Acct-Session-Id are used to … [View More]match start and stop radius requests. We could determine who sent the BYE from the value of ftag parameter in Route header field and swap the values, the question is whether this makes sense. The billing system could take all info from start requests and use the stop request to calculate the session duration only. Jan. On 16-11-2005 15:04, Lenir wrote: > Ok, that put the right things in place. However, here's a twist: > Either way, wit switching Sip-Translated-Request-URI and Called-Station-ID > around, or not, the CDRs are sent differently depending on who "hands up" > the call. For example, notice the User-Name, Calling-Station-ID and > Called-Station-ID: > > User 1000 calls 15615551212, and user 1000 hangs up: > ===================================================== > rad_recv: Accounting-Request packet from host 216.82.124.29:39228, id=206, > length=306 > Acct-Status-Type = Start > Service-Type = Sip-Session > Acct-Terminate-Cause = 200 > Error-Cause = Invite > User-Name = "1000(a)216.82.124.29" > Calling-Station-Id = "sip:1000@216.82.124.29" > Sip-Translated-Request-URI = > "sip:15617220966@216.82.124.29;user=phone" > Called-Station-Id = > "sip:949449852150#15617220966@216.82.124.30;user=phone" > Acct-Session-Id = "000d2890-d47f01f5-6410e8bd-4baf0435(a)66.232.8.116" > Sip-To-Tag = "33C5A3C-76A" > Sip-From-Tag = "000d2890d47f24f85e6c0b91-0092ab53" > Sip-CSeq = "102" > NAS-IP-Address = 216.82.124.29 > NAS-Port = 5060 > Acct-Delay-Time = 0 > rlm_sql (sql): Reserving sql socket id: 15 > rlm_sql (sql): Released sql socket id: 15 > Sending Accounting-Response of id 206 to 216.82.124.29:39228 > rad_recv: Accounting-Request packet from host 216.82.124.29:39228, id=207, > length=300 > Acct-Status-Type = Stop > Service-Type = Sip-Session > Acct-Terminate-Cause = 200 > Error-Cause = 8 > User-Name = "1000(a)216.82.124.29" > Calling-Station-Id = "sip:1000@216.82.124.29" > Sip-Translated-Request-URI = > "sip:15617220966@216.82.124.29;user=phone" > Called-Station-Id = > "sip:949449852150#15617220966@216.82.124.30:5060" > Acct-Session-Id = "000d2890-d47f01f5-6410e8bd-4baf0435(a)66.232.8.116" > Sip-To-Tag = "33C5A3C-76A" > Sip-From-Tag = "000d2890d47f24f85e6c0b91-0092ab53" > Sip-CSeq = "103" > NAS-IP-Address = 216.82.124.29 > NAS-Port = 5060 > Acct-Delay-Time = 0 > > > User 1000 calls 15615551212, and user 15615551212 hangs up: > =========================================================== > > rad_recv: Accounting-Request packet from host 216.82.124.29:39227, id=200, > length=306 > Acct-Status-Type = Start > Service-Type = Sip-Session > Acct-Terminate-Cause = 200 > Error-Cause = Invite > User-Name = "1000(a)216.82.124.29" > Calling-Station-Id = "sip:1000@216.82.124.29" > Sip-Translated-Request-URI = > "sip:15617220966@216.82.124.29;user=phone" > Called-Station-Id = > "sip:949449852150#15617220966@216.82.124.30;user=phone" > Acct-Session-Id = "000d2890-d47f01f4-372dff4b-793d4acc(a)66.232.8.116" > Sip-To-Tag = "33B37E4-77B" > Sip-From-Tag = "000d2890d47f24f639ccef77-4ef08b02" > Sip-CSeq = "102" > NAS-IP-Address = 216.82.124.29 > NAS-Port = 5060 > Acct-Delay-Time = 0 > rlm_sql (sql): Reserving sql socket id: 1 > rlm_sql (sql): Released sql socket id: 1 > Sending Accounting-Response of id 200 to 216.82.124.29:39227 > rad_recv: Accounting-Request packet from host 216.82.124.29:39227, id=201, > length=286 > Acct-Status-Type = Stop > Service-Type = Sip-Session > Acct-Terminate-Cause = 200 > Error-Cause = 8 > User-Name = "15617220966(a)216.82.124.29" > Calling-Station-Id = "sip:15617220966@216.82.124.29;user=phone" > Sip-Translated-Request-URI = "sip:1000@216.82.124.29" > Called-Station-Id = "sip:1000@66.232.8.116:5061" > Acct-Session-Id = "000d2890-d47f01f4-372dff4b-793d4acc(a)66.232.8.116" > Sip-To-Tag = "000d2890d47f24f639ccef77-4ef08b02" > Sip-From-Tag = "33B37E4-77B" > Sip-CSeq = "101" > NAS-IP-Address = 216.82.124.29 > NAS-Port = 5060 > Acct-Delay-Time = 0 > > > Shouldn't it always be that the User-Name attribute is the user who > initiated the call? > > Lenir > > -----Original Message----- > From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf > Of 'Jan Janak' > Sent: Wednesday, November 16, 2005 1:53 PM > To: lsantiago(a)globalgatewaycom.com > Cc: serdev(a)iptel.org; serusers(a)iptel.org; users(a)openser.org; > devel(a)openser.org > Subject: Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius > Accounting > > Either reconfigure the billing application to use > Sip-Translated-Request-URI (recommended) or change the ID of the > attribute in the radiusclient dictionary as suggested by Bogdan. > > In this particular example you should set Called-Station-Id to 107. > > Jan. > > On 16-11-2005 13:09, 'Lenir' wrote: > > Ok, so how do I tell acc.so to send Sip-Translated-Request-URI as the > > Called-Station-ID instead? > > > > Lenir Santiago, Partner > > Global Gateway Communications, Inc. > > 2500 Quantum Lakes Drive, Ste. 203 > > Boynton Beach, FL 33426 > > Office: 561-853-2213 > > Cell: 561-722-0966 > > Fax: 877-875-0060 > > http://www.globalgatewaycom.com > > > > -----Original Message----- > > From: 'Jan Janak' [mailto:jan@iptel.org] > > Sent: Wednesday, November 16, 2005 11:29 AM > > To: Bogdan-Andrei Iancu > > Cc: Lenir; 'Lenir Santiago'; serdev(a)iptel.org; serusers(a)iptel.org; > > devel(a)openser.org; users(a)openser.org > > Subject: Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius > > Accounting > > > > Why would you do something like this ? All the info that is needed is in > > Sip-Translated-Ruquest-ID below. You can see it in the RADIUS dump: > > > > Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway:5060" > > > > Jan. > > > > On 16-11-2005 16:50, Bogdan-Andrei Iancu wrote: > > > Hi Lenir, > > > > > > you can keep the modified To value into an AVP and use the extra > > > accounting feature to log it. See > > > http://www.openser.org/docs/modules/1.1.x/acc.html#AEN115 > > > and > > > http://www.openser.org/docs/modules/1.1.x/acc.html#AEN350 > > > > > > the problem is how to set it as Called-Station-ID attribute in order to > > > avoid overlapping. You can do a trick by playing with the radius > > > dictionary (on proxy side) : > > > changed the Called-Station-ID to a different ID attribute > > > set extra accounting with the original attribute ID of > > Called-Station-ID > > > in this way the acc will set de default TO as something else and you AVP > > > > as Called-Station-ID ;) > > > > > > regards, > > > bogdan > > > > > > > > > Lenir wrote: > > > > > > >Well, if userA dials to:30, 30 gets translated to:15615551212 and SER > > dials > > > >15615551212, how can I make the acc.so module send the modified > > > >to:15615551212 as the Called-Station-ID attribute? > > > > > > > >This is important for billing, specially if someone forwards to a long > > > >distance or international long distance number. > > > > > > > >-----Original Message----- > > > >From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@iptel.org] On > > > >Behalf Of Jan Janak > > > >Sent: Wednesday, November 16, 2005 3:15 AM > > > >To: Lenir Santiago > > > >Cc: serdev(a)iptel.org; serusers(a)iptel.org; users(a)openser.org; > > > >devel(a)openser.org > > > >Subject: [Serusers] Re: [Serdev] Inaccurate Radius Accounting > > > > > > > >Hello, > > > > > > > >Called-Station-Id attribute always contains the To URI (not > > > >Request-URI). You can find the outbound Request-URI in > > > >Sip-Translated-Request-URI attribute. > > > > > > > >Even if you rewrite To header field, it will never be accounted in > > > >Called-Station-Id. > > > > > > > > Jan. > > > > > > > >On 15-11-2005 21:59, Lenir Santiago wrote: > > > > > > > > > > > >>Hey guys, > > > >> > > > >> > > > >> > > > >>I'm using Radius to do pretty much everything regarding my users, i.e. > > > >>registrations, authorization, features and preferences. I implemented > my > > > >> > > > >> > > > >own > > > > > > > > > > > >>way of doing speeddial by using AVPs provided by radius and AVPOPs > > module, > > > >>Here is the snippet: > > > >> > > > >> > > > >> > > > >>AVP-SIP = speedial:30:15615551212 > > > >> > > > >> > > > >> > > > >> avp_subst("s:caller_speedial/i:99", "/(.*):(.*)/\2/"); > > > >>#15615551212 > > > >> > > > >> avp_printf("i:98","$rU:*"); #30 > > > >> > > > >> avp_printf("i:97","sip:$avp($temp2)@$td"); # > > > >>sip:15615551212@xx.xx.xx.xx > > > >> > > > >> avp_printf("i:96","<sip:$avp($temp2)@$td>\r\n"); # > > > >><sip:15615551212@xx.xx.xx.xx>. > > > >> > > > >> xlog("L_N"," SPEEDIALING: $fU is speed-dialing > > > >>$hdr(To)->$avp($temp2)\n"); > > > >> > > > >> if (avp_check("s:caller_speedial", "fm/$temp1/g")) { > > > >> > > > >> > > > >> > > > >> #avp_pushto("$To","i:96"); > > > >> > > > >> #avp_pushto("$ruri","i:97"); > > > >> > > > >> #avp_pushto("$duri","i:97"); > > > >> > > > >> #subst('/^To: (.*)(a)(.*)/To: "$avp($temp2)" > > > >><$avp($temp2)@\2/'); > > > >> > > > >> #append_hf("To: \"5615551212\" > > > >><15615551212(a)xx.xx.xx.xx>;user=phone"); > > > >> > > > >> > > > >>#subst_uri('/^sip:(.*)@(.*)$/sip:$avp($temp2)@\2/i'); > > > >> > > > >> #subst_user('/(.*)/$avp($temp2)/'); > > > >> > > > >> #remove_hf("To"); > > > >> > > > >> rewriteuser("15615551212"); > > > >> > > > >> xlog("L_N"," SPEEDIALING: $fU is dialing > > > >> > > > >> > > > >$hdr(To) > > > > > > > > > > > >>$avp($temp2)\n"); > > > >> > > > >> avp_delete("$temp1/g"); > > > >> > > > >> avp_delete("$temp2/g"); > > > >> > > > >> > > > >> > > > >> }; > > > >> > > > >> > > > >> > > > >>In this scenario, im forcing it instead of getting the value from AVPs > > > >> > > > >> > > > >just > > > > > > > > > > > >>for ease of showing my point. The problem that I'm having is that no > > > >> > > > >> > > > >matter > > > > > > > > > > > >>what I do to change the To header or change the R-URI, radius ALWAYS > > sends > > > >>the accounting records with the original R-URI, in my case > > 30(a)xx.xx.xx.xx, > > > >>how can I get acc.so to send the final resulting URI > > > >>(15615551212(a)xx.xx.xx.xx) in the Called-Station-Id AVP? > > > >> > > > >> > > > >> > > > >> > > > >> > > > >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, > id=97, > > > >>length=298 > > > >> > > > >> Acct-Status-Type = Start > > > >> > > > >> Service-Type = Sip-Session > > > >> > > > >> Acct-Terminate-Cause = 200 > > > >> > > > >> Error-Cause = Invite > > > >> > > > >> User-Name = "1000(a)xx.xx.xx.xx" > > > >> > > > >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" > > > >> > > > >> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" > > > >> > > > >> Sip-Translated-Request-URI = > > > >>"sip:949449852150#15615551212@gateway;user=phone" > > > >> > > > >> Acct-Session-Id = > > > >> > > > >> > > > >"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" > > > > > > > > > > > >> Sip-To-Tag = "230383F4-2AF" > > > >> > > > >> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" > > > >> > > > >> Sip-CSeq = "102" > > > >> > > > >> NAS-IP-Address = xx.xx.xx.xx > > > >> > > > >> NAS-Port = 5060 > > > >> > > > >> Acct-Delay-Time = 0 > > > >> > > > >>rlm_sql (sql): Reserving sql socket id: 12 > > > >> > > > >>rlm_sql (sql): Released sql socket id: 12 > > > >> > > > >>Sending Accounting-Response of id 97 to xx.xx.xx.xx:39216 > > > >> > > > >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, > id=98, > > > >>length=292 > > > >> > > > >> Acct-Status-Type = Stop > > > >> > > > >> Service-Type = Sip-Session > > > >> > > > >> Acct-Terminate-Cause = 200 > > > >> > > > >> Error-Cause = 8 > > > >> > > > >> User-Name = "1000(a)xx.xx.xx.xx" > > > >> > > > >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" > > > >> > > > >> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" > > > >> > > > >> Sip-Translated-Request-URI = > > > >>"sip:949449852150#15615551212@gateway:5060" > > > >> > > > >> Acct-Session-Id = > > > >> > > > >> > > > >"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" > > > > > > > > > > > >> Sip-To-Tag = "230383F4-2AF" > > > >> > > > >> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" > > > >> > > > >> Sip-CSeq = "103" > > > >> > > > >> NAS-IP-Address = xx.xx.xx.xx > > > >> > > > >> NAS-Port = 5060 > > > >> > > > >> Acct-Delay-Time = 0 > > > >> > > > >> > > > >> > > > >> > > > >> > > > >>Thanks in advance! > > > >> > > > >> > > > >> > > > >> > > > >> > > > >>Lenir > > > >>_______________________________________________ > > > >>Users mailing list > > > >>Users(a)openser.org > > > >>http://openser.org/cgi-bin/mailman/listinfo/users > > > >> > > > >> > > > >> > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > [View Less]

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Recall: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting
by Lenir 16 Nov '05

16 Nov '05

Lenir would like to recall the message, "[Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting".

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Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting
by 'Jan Janak' 16 Nov '05

16 Nov '05

Either reconfigure the billing application to use Sip-Translated-Request-URI (recommended) or change the ID of the attribute in the radiusclient dictionary as suggested by Bogdan. In this particular example you should set Called-Station-Id to 107. Jan. On 16-11-2005 13:09, 'Lenir' wrote: > Ok, so how do I tell acc.so to send Sip-Translated-Request-URI as the > Called-Station-ID instead? > > Lenir Santiago, Partner > Global Gateway Communications, Inc. > 2500 Quantum … [View More]Lakes Drive, Ste. 203 > Boynton Beach, FL 33426 > Office: 561-853-2213 > Cell: 561-722-0966 > Fax: 877-875-0060 > http://www.globalgatewaycom.com > > -----Original Message----- > From: 'Jan Janak' [mailto:jan@iptel.org] > Sent: Wednesday, November 16, 2005 11:29 AM > To: Bogdan-Andrei Iancu > Cc: Lenir; 'Lenir Santiago'; serdev(a)iptel.org; serusers(a)iptel.org; > devel(a)openser.org; users(a)openser.org > Subject: Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius > Accounting > > Why would you do something like this ? All the info that is needed is in > Sip-Translated-Ruquest-ID below. You can see it in the RADIUS dump: > > Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway:5060" > > Jan. > > On 16-11-2005 16:50, Bogdan-Andrei Iancu wrote: > > Hi Lenir, > > > > you can keep the modified To value into an AVP and use the extra > > accounting feature to log it. See > > http://www.openser.org/docs/modules/1.1.x/acc.html#AEN115 > > and > > http://www.openser.org/docs/modules/1.1.x/acc.html#AEN350 > > > > the problem is how to set it as Called-Station-ID attribute in order to > > avoid overlapping. You can do a trick by playing with the radius > > dictionary (on proxy side) : > > changed the Called-Station-ID to a different ID attribute > > set extra accounting with the original attribute ID of > Called-Station-ID > > in this way the acc will set de default TO as something else and you AVP > > as Called-Station-ID ;) > > > > regards, > > bogdan > > > > > > Lenir wrote: > > > > >Well, if userA dials to:30, 30 gets translated to:15615551212 and SER > dials > > >15615551212, how can I make the acc.so module send the modified > > >to:15615551212 as the Called-Station-ID attribute? > > > > > >This is important for billing, specially if someone forwards to a long > > >distance or international long distance number. > > > > > >-----Original Message----- > > >From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@iptel.org] On > > >Behalf Of Jan Janak > > >Sent: Wednesday, November 16, 2005 3:15 AM > > >To: Lenir Santiago > > >Cc: serdev(a)iptel.org; serusers(a)iptel.org; users(a)openser.org; > > >devel(a)openser.org > > >Subject: [Serusers] Re: [Serdev] Inaccurate Radius Accounting > > > > > >Hello, > > > > > >Called-Station-Id attribute always contains the To URI (not > > >Request-URI). You can find the outbound Request-URI in > > >Sip-Translated-Request-URI attribute. > > > > > >Even if you rewrite To header field, it will never be accounted in > > >Called-Station-Id. > > > > > > Jan. > > > > > >On 15-11-2005 21:59, Lenir Santiago wrote: > > > > > > > > >>Hey guys, > > >> > > >> > > >> > > >>I'm using Radius to do pretty much everything regarding my users, i.e. > > >>registrations, authorization, features and preferences. I implemented my > > >> > > >> > > >own > > > > > > > > >>way of doing speeddial by using AVPs provided by radius and AVPOPs > module, > > >>Here is the snippet: > > >> > > >> > > >> > > >>AVP-SIP = speedial:30:15615551212 > > >> > > >> > > >> > > >> avp_subst("s:caller_speedial/i:99", "/(.*):(.*)/\2/"); > > >>#15615551212 > > >> > > >> avp_printf("i:98","$rU:*"); #30 > > >> > > >> avp_printf("i:97","sip:$avp($temp2)@$td"); # > > >>sip:15615551212@xx.xx.xx.xx > > >> > > >> avp_printf("i:96","<sip:$avp($temp2)@$td>\r\n"); # > > >><sip:15615551212@xx.xx.xx.xx>. > > >> > > >> xlog("L_N"," SPEEDIALING: $fU is speed-dialing > > >>$hdr(To)->$avp($temp2)\n"); > > >> > > >> if (avp_check("s:caller_speedial", "fm/$temp1/g")) { > > >> > > >> > > >> > > >> #avp_pushto("$To","i:96"); > > >> > > >> #avp_pushto("$ruri","i:97"); > > >> > > >> #avp_pushto("$duri","i:97"); > > >> > > >> #subst('/^To: (.*)(a)(.*)/To: "$avp($temp2)" > > >><$avp($temp2)@\2/'); > > >> > > >> #append_hf("To: \"5615551212\" > > >><15615551212(a)xx.xx.xx.xx>;user=phone"); > > >> > > >> > > >>#subst_uri('/^sip:(.*)@(.*)$/sip:$avp($temp2)@\2/i'); > > >> > > >> #subst_user('/(.*)/$avp($temp2)/'); > > >> > > >> #remove_hf("To"); > > >> > > >> rewriteuser("15615551212"); > > >> > > >> xlog("L_N"," SPEEDIALING: $fU is dialing > > >> > > >> > > >$hdr(To) > > > > > > > > >>$avp($temp2)\n"); > > >> > > >> avp_delete("$temp1/g"); > > >> > > >> avp_delete("$temp2/g"); > > >> > > >> > > >> > > >> }; > > >> > > >> > > >> > > >>In this scenario, im forcing it instead of getting the value from AVPs > > >> > > >> > > >just > > > > > > > > >>for ease of showing my point. The problem that I'm having is that no > > >> > > >> > > >matter > > > > > > > > >>what I do to change the To header or change the R-URI, radius ALWAYS > sends > > >>the accounting records with the original R-URI, in my case > 30(a)xx.xx.xx.xx, > > >>how can I get acc.so to send the final resulting URI > > >>(15615551212(a)xx.xx.xx.xx) in the Called-Station-Id AVP? > > >> > > >> > > >> > > >> > > >> > > >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=97, > > >>length=298 > > >> > > >> Acct-Status-Type = Start > > >> > > >> Service-Type = Sip-Session > > >> > > >> Acct-Terminate-Cause = 200 > > >> > > >> Error-Cause = Invite > > >> > > >> User-Name = "1000(a)xx.xx.xx.xx" > > >> > > >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" > > >> > > >> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" > > >> > > >> Sip-Translated-Request-URI = > > >>"sip:949449852150#15615551212@gateway;user=phone" > > >> > > >> Acct-Session-Id = > > >> > > >> > > >"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" > > > > > > > > >> Sip-To-Tag = "230383F4-2AF" > > >> > > >> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" > > >> > > >> Sip-CSeq = "102" > > >> > > >> NAS-IP-Address = xx.xx.xx.xx > > >> > > >> NAS-Port = 5060 > > >> > > >> Acct-Delay-Time = 0 > > >> > > >>rlm_sql (sql): Reserving sql socket id: 12 > > >> > > >>rlm_sql (sql): Released sql socket id: 12 > > >> > > >>Sending Accounting-Response of id 97 to xx.xx.xx.xx:39216 > > >> > > >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=98, > > >>length=292 > > >> > > >> Acct-Status-Type = Stop > > >> > > >> Service-Type = Sip-Session > > >> > > >> Acct-Terminate-Cause = 200 > > >> > > >> Error-Cause = 8 > > >> > > >> User-Name = "1000(a)xx.xx.xx.xx" > > >> > > >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" > > >> > > >> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" > > >> > > >> Sip-Translated-Request-URI = > > >>"sip:949449852150#15615551212@gateway:5060" > > >> > > >> Acct-Session-Id = > > >> > > >> > > >"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" > > > > > > > > >> Sip-To-Tag = "230383F4-2AF" > > >> > > >> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" > > >> > > >> Sip-CSeq = "103" > > >> > > >> NAS-IP-Address = xx.xx.xx.xx > > >> > > >> NAS-Port = 5060 > > >> > > >> Acct-Delay-Time = 0 > > >> > > >> > > >> > > >> > > >> > > >>Thanks in advance! > > >> > > >> > > >> > > >> > > >> > > >>Lenir > > >>_______________________________________________ > > >>Users mailing list > > >>Users(a)openser.org > > >>http://openser.org/cgi-bin/mailman/listinfo/users > > >> > > >> > > >> > [View Less]

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Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting
by Bogdan-Andrei Iancu 16 Nov '05

16 Nov '05

Hi Lenir, you can keep the modified To value into an AVP and use the extra accounting feature to log it. See http://www.openser.org/docs/modules/1.1.x/acc.html#AEN115 and http://www.openser.org/docs/modules/1.1.x/acc.html#AEN350 the problem is how to set it as Called-Station-ID attribute in order to avoid overlapping. You can do a trick by playing with the radius dictionary (on proxy side) : changed the Called-Station-ID to a different ID attribute set extra … [View More]accounting with the original attribute ID of Called-Station-ID in this way the acc will set de default TO as something else and you AVP as Called-Station-ID ;) regards, bogdan Lenir wrote: >Well, if userA dials to:30, 30 gets translated to:15615551212 and SER dials >15615551212, how can I make the acc.so module send the modified >to:15615551212 as the Called-Station-ID attribute? > >This is important for billing, specially if someone forwards to a long >distance or international long distance number. > >-----Original Message----- >From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@iptel.org] On >Behalf Of Jan Janak >Sent: Wednesday, November 16, 2005 3:15 AM >To: Lenir Santiago >Cc: serdev(a)iptel.org; serusers(a)iptel.org; users(a)openser.org; >devel(a)openser.org >Subject: [Serusers] Re: [Serdev] Inaccurate Radius Accounting > >Hello, > >Called-Station-Id attribute always contains the To URI (not >Request-URI). You can find the outbound Request-URI in >Sip-Translated-Request-URI attribute. > >Even if you rewrite To header field, it will never be accounted in >Called-Station-Id. > > Jan. > >On 15-11-2005 21:59, Lenir Santiago wrote: > > >>Hey guys, >> >> >> >>I'm using Radius to do pretty much everything regarding my users, i.e. >>registrations, authorization, features and preferences. I implemented my >> >> >own > > >>way of doing speeddial by using AVPs provided by radius and AVPOPs module, >>Here is the snippet: >> >> >> >>AVP-SIP = speedial:30:15615551212 >> >> >> >> avp_subst("s:caller_speedial/i:99", "/(.*):(.*)/\2/"); >>#15615551212 >> >> avp_printf("i:98","$rU:*"); #30 >> >> avp_printf("i:97","sip:$avp($temp2)@$td"); # >>sip:15615551212@xx.xx.xx.xx >> >> avp_printf("i:96","<sip:$avp($temp2)@$td>\r\n"); # >><sip:15615551212@xx.xx.xx.xx>. >> >> xlog("L_N"," SPEEDIALING: $fU is speed-dialing >>$hdr(To)->$avp($temp2)\n"); >> >> if (avp_check("s:caller_speedial", "fm/$temp1/g")) { >> >> >> >> #avp_pushto("$To","i:96"); >> >> #avp_pushto("$ruri","i:97"); >> >> #avp_pushto("$duri","i:97"); >> >> #subst('/^To: (.*)(a)(.*)/To: "$avp($temp2)" >><$avp($temp2)@\2/'); >> >> #append_hf("To: \"5615551212\" >><15615551212(a)xx.xx.xx.xx>;user=phone"); >> >> >>#subst_uri('/^sip:(.*)@(.*)$/sip:$avp($temp2)@\2/i'); >> >> #subst_user('/(.*)/$avp($temp2)/'); >> >> #remove_hf("To"); >> >> rewriteuser("15615551212"); >> >> xlog("L_N"," SPEEDIALING: $fU is dialing >> >> >$hdr(To) > > >>$avp($temp2)\n"); >> >> avp_delete("$temp1/g"); >> >> avp_delete("$temp2/g"); >> >> >> >> }; >> >> >> >>In this scenario, im forcing it instead of getting the value from AVPs >> >> >just > > >>for ease of showing my point. The problem that I'm having is that no >> >> >matter > > >>what I do to change the To header or change the R-URI, radius ALWAYS sends >>the accounting records with the original R-URI, in my case 30(a)xx.xx.xx.xx, >>how can I get acc.so to send the final resulting URI >>(15615551212(a)xx.xx.xx.xx) in the Called-Station-Id AVP? >> >> >> >> >> >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=97, >>length=298 >> >> Acct-Status-Type = Start >> >> Service-Type = Sip-Session >> >> Acct-Terminate-Cause = 200 >> >> Error-Cause = Invite >> >> User-Name = "1000(a)xx.xx.xx.xx" >> >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" >> >> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" >> >> Sip-Translated-Request-URI = >>"sip:949449852150#15615551212@gateway;user=phone" >> >> Acct-Session-Id = >> >> >"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" > > >> Sip-To-Tag = "230383F4-2AF" >> >> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" >> >> Sip-CSeq = "102" >> >> NAS-IP-Address = xx.xx.xx.xx >> >> NAS-Port = 5060 >> >> Acct-Delay-Time = 0 >> >>rlm_sql (sql): Reserving sql socket id: 12 >> >>rlm_sql (sql): Released sql socket id: 12 >> >>Sending Accounting-Response of id 97 to xx.xx.xx.xx:39216 >> >>rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=98, >>length=292 >> >> Acct-Status-Type = Stop >> >> Service-Type = Sip-Session >> >> Acct-Terminate-Cause = 200 >> >> Error-Cause = 8 >> >> User-Name = "1000(a)xx.xx.xx.xx" >> >> Calling-Station-Id = "sip:1000@xx.xx.xx.xx" >> >> Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" >> >> Sip-Translated-Request-URI = >>"sip:949449852150#15615551212@gateway:5060" >> >> Acct-Session-Id = >> >> >"000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" > > >> Sip-To-Tag = "230383F4-2AF" >> >> Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" >> >> Sip-CSeq = "103" >> >> NAS-IP-Address = xx.xx.xx.xx >> >> NAS-Port = 5060 >> >> Acct-Delay-Time = 0 >> >> >> >> >> >>Thanks in advance! >> >> >> >> >> >>Lenir >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> [View Less]

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[Serusers] ser-0.9.3 vs ser-0.9.4
by Chia Huey Lim 16 Nov '05

16 Nov '05

Hi list, Is there any major difference for www_authorize and www_challenge function in ser 0.9.3 and ser 0.9.4 ? When I am using ser 0.9.3, have no problem with the registration, but when I upgrade to ser 0.9.4 version, Most of the UAs failed to register, and the 401 rate is extremely high. Once I downgrade back to ser 0.9.3, then no more issue. I m using if (!www_authorize("voip","subscriber")) { www_challenge("voip","0"); … [View More]

1 0

[Serusers] ser-0.9.3 vs ser-0.9.4
by Chia Huey 16 Nov '05

16 Nov '05

Hi list, Is there any major difference for www_authorize and www_challenge function in ser 0.9.3 and ser 0.9.4 ? When I am using ser 0.9.3, have no problem with the registration, but when I upgrade to ser 0.9.4 version, Most of the UAs failed to register, and the 401 rate is extremely high. Once I downgrade back to ser 0.9.3, then no more issue. I m using if (!www_authorize("voip","subscriber")) { www_challenge("voip","0"); … [View More]

1 0

[Serusers] ACK before INVITE on re-INVITE causes Cisco AS's to return a 500 error
by Evan Borgstrom 16 Nov '05

16 Nov '05

Hey all, This has more to do with Asterisk & Cisco AS's IMO but I thought it would be interesting to get some responses from this group before I begin the battle with Cisco to get a bug report underway. We have an Asterisk box acting as a PBX sending calls to our SER instances. The asterisk box has "reinvite=yes" in the sip.conf file which causes Asterisk to send a re-INVITE for the call once it receives the 200 OK. This re-INVITE has it's SDP changed to that of the IP Phone behind the … [View More]Asterisk box so that all media doesn't need to traverse it. This works fine except in certain instances where messages get out of order and it causes our Cisco GW's to error the call with a generic 500 message. When the messages are in the correct order the 500 error does not appear. So here's the situation, user behind the asterisk box picks up the phone and dials a PSTN destination. Asterisk fires off the initial invite and everything proceeds as normal and we eventually receive the 200 OK message to which the Asterisk box replies with ACK and a new INVITE that come in this order (X = Asterisk, Y = SER, Z = Cisco): U 2005/11/15 16:51:51.557322 X.X.X.X:5060 -> Y.Y.Y.Y:5060 ACK # U 2005/11/15 16:51:51.557948 X.X.X.X:5060 -> Y.Y.Y.Y:5060 INVITE SER accordingly processes the two messages and replies 100 Trying to the new INVITE message. Here's where the problem occurs, SER then sends these messages out to the PSTN gateway but the order has been reversed: # U 2005/11/15 16:51:51.558937 Y.Y.Y.Y:5060 -> X.X.X.X:5060 SIP/2.0 100 trying # U 2005/11/15 16:51:51.559000 Y.Y.Y.Y:5060 -> Z.Z.Z.Z:5060 INVITE # U 2005/11/15 16:51:51.559657 Y.Y.Y.Y:5060 -> X.X.X.X:5060 ACK When this happens the Cisco GW replies with a 500 error message and the following is sent to the Cisco's logs: # U 2005/11/15 16:51:51.711355 Z.Z.Z.Z:5060 -> Y.Y.Y.Y:5060 SIP/2.0 500 Server Internal Error. Nov 15 21:51:51.613: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: Y.Y.Y.Y:5060 Nov 15 21:51:51.613: *****CCB found in UAS Request table. ccb=0x653BCD0C Nov 15 21:51:51.613: CCSIP-SPI-CONTROL: act_sentsucc_new_message Nov 15 21:51:51.617: CCSIP-SPI-CONTROL: act_sentsucc_new_message_request Nov 15 21:51:51.617: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE Nov 15 21:51:51.617: sip_stats_status_code Nov 15 21:51:51.617: sipSPICheckRequest: CheckRequest fail on method 102 error code: 2 and status: 500 Is there any parameter that I can set to ensure that message leave SER in the order they enter? Should the Cisco be able to handle this condition (I check a number of different sections of the RFC but couldn't find anything specific for either side)? Thanks, Evan [View Less]

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[Serusers] avp_write/avp_pushto in database/memory?
by Ramin Nikaeen 16 Nov '05

16 Nov '05

Valued Colleagues, Do the avp_write and avp_pushto involve Database access operations?! For example: avp_write("sip:22@10.1.10.60:5060", "s:callfwd"); avp_pushto("$ruri", "s:callfwd"); Does avp_write store the string "sip:22@10.1.10.43:5060" for "s:callfwd" in database and avp_pushto looks up the string related to "s:callfwd" from the database or is the information stored in memory?! I want to avoid database access to ensure improved performance. Thanks ramin

2 1

[Serusers] call forwarding and *21
by Kyriakos Mavromichalis 16 Nov '05

16 Nov '05

Can someone point me to any resources on how to implement services like callforwarding by pressing *21 through the users' UA? Basically I need my users to be able to activate different services via the phone's keys. thank you -- Kyriakos Mavromichalis Otenet Telecom

3 3

[Serusers] Problem with running SER with mysql module
by Murděj Ukrutný 16 Nov '05

16 Nov '05

Hi, I am a beginner here. Could anyone please answer my question? I have downloaded the SER 0.9.4 and compile with mysql module it on SUSE Linux Enterprise Server 9. Mysql 4.1.15 compiled from source. I have problem whith running SER. Command 'serctl start' finished with message 'Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed' Message in /var/logs/messages is 'Nov 16 05:34:34 suse1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/… [View More]

3 2

[Serusers] Mediaproxy High Load
by Adrian Georgescu 16 Nov '05

16 Nov '05

The CPU will show in Linux high usage as soon as you get over 20 calls but you will be able to load the box up to 200 streams, the CPU will show the same high value while the proxy works fine. Use the rtpgenerator to test after what amount of calls your box will start loosing packets we reckon 70 streams per 1 GHz. Regards, Adrian <<<<<< Hi All, I tested the load of the mediaproxy (ver 1.4.2) and found that it is quite high. To give an idea, on a P4 2.4GHz machine, … [View More]

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[Serusers] difference between callsetup asterisk & ser and related security risks
by Arne Van Theemsche 16 Nov '05

16 Nov '05

Hi, I've been spending days on tcpdumping callsetups with ser OR asterisk as registar. This is my conclusion, can somebody confirm this? On asterisk: -asterisk maintains 2 independendly sipsessions between 2 clients, where asterisks stays the contact in the sipheaders. After picking up the call, asterisk sends an re-invite in the sip-body (address), and remains the contact for sipmessages. If someone hangs up, asterisk sends an re-invite to the other party, for redirecting stream back to … [View More]

3 2

[Serusers] ERROR: exec_str:
by davor jovanovic 16 Nov '05

16 Nov '05

Hi all, I'm using SER 0.9.3 and trying to do LDAP query but I keep getting this message: 0(17313) INFO: dont_fork turned on, living on 0(17313) exec_str: rtrim 0(17313) ERROR: exec_str: cmd /etc/ser/sipldap failed. exit_status=-1, errno=10: No child processes can someone explain to me why is that and how can I fix it? when I'm doing lookup with: (!lookup("location")) { if (!exec_dset("/etc/ser/sipldap")) { sl_send_reply("404", "Not Found"); break; … [View More]

1 0

[Serusers] Inaccurate Radius Accounting
by Lenir Santiago 16 Nov '05

16 Nov '05

Hey guys, I'm using Radius to do pretty much everything regarding my users, i.e. registrations, authorization, features and preferences. I implemented my own way of doing speeddial by using AVPs provided by radius and AVPOPs module, Here is the snippet: AVP-SIP = speedial:30:15615551212 avp_subst("s:caller_speedial/i:99", "/(.*):(.*)/\2/"); #15615551212 avp_printf("i:98","$rU:*"); #30 avp_printf("i:97","sip:$avp($temp2)@$td"); # sip:… [View More]15615551212@xx.xx.xx.xx avp_printf("i:96","<sip:$avp($temp2)@$td>\r\n"); # <sip:15615551212@xx.xx.xx.xx>. xlog("L_N"," SPEEDIALING: $fU is speed-dialing $hdr(To)->$avp($temp2)\n"); if (avp_check("s:caller_speedial", "fm/$temp1/g")) { #avp_pushto("$To","i:96"); #avp_pushto("$ruri","i:97"); #avp_pushto("$duri","i:97"); #subst('/^To: (.*)(a)(.*)/To: "$avp($temp2)" <$avp($temp2)@\2/'); #append_hf("To: \"5615551212\" <15615551212(a)xx.xx.xx.xx>;user=phone"); #subst_uri('/^sip:(.*)@(.*)$/sip:$avp($temp2)@\2/i'); #subst_user('/(.*)/$avp($temp2)/'); #remove_hf("To"); rewriteuser("15615551212"); xlog("L_N"," SPEEDIALING: $fU is dialing $hdr(To) $avp($temp2)\n"); avp_delete("$temp1/g"); avp_delete("$temp2/g"); }; In this scenario, im forcing it instead of getting the value from AVPs just for ease of showing my point. The problem that I'm having is that no matter what I do to change the To header or change the R-URI, radius ALWAYS sends the accounting records with the original R-URI, in my case 30(a)xx.xx.xx.xx, how can I get acc.so to send the final resulting URI (15615551212(a)xx.xx.xx.xx) in the Called-Station-Id AVP? rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=97, length=298 Acct-Status-Type = Start Service-Type = Sip-Session Acct-Terminate-Cause = 200 Error-Cause = Invite User-Name = "1000(a)xx.xx.xx.xx" Calling-Station-Id = "sip:1000@xx.xx.xx.xx" Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway;user=phone" Acct-Session-Id = "000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" Sip-To-Tag = "230383F4-2AF" Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" Sip-CSeq = "102" NAS-IP-Address = xx.xx.xx.xx NAS-Port = 5060 Acct-Delay-Time = 0 rlm_sql (sql): Reserving sql socket id: 12 rlm_sql (sql): Released sql socket id: 12 Sending Accounting-Response of id 97 to xx.xx.xx.xx:39216 rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=98, length=292 Acct-Status-Type = Stop Service-Type = Sip-Session Acct-Terminate-Cause = 200 Error-Cause = 8 User-Name = "1000(a)xx.xx.xx.xx" Calling-Station-Id = "sip:1000@xx.xx.xx.xx" Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway:5060" Acct-Session-Id = "000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" Sip-To-Tag = "230383F4-2AF" Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" Sip-CSeq = "103" NAS-IP-Address = xx.xx.xx.xx NAS-Port = 5060 Acct-Delay-Time = 0 Thanks in advance! Lenir [View Less]

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[Users] no voice when using nathelper with rtpproxy
by messa＠innsof.com 16 Nov '05

16 Nov '05

Hi all I install rtpproxy in the same host (with the public IP) than openser. the installation of rtpproxy put rtpproxy.sock in /var/run directory.But when i make a call between 2 clients (each client in a different LAN), i have no voice after signalisation. And openser give me these errors: [root@Is-dev01 openser-1.0.0-tls]# /usr/local/sbin/openser Listening on udp: 62.128.173.101 [62.128.173.101]:5060 tcp: 62.128.173.101 [62.128.173.101]:5060 Aliases: … [View More]

3 2

[Serusers] Mediaproxy High Load
by tcchan 16 Nov '05

16 Nov '05

Hi All, I tested the load of the mediaproxy (ver 1.4.2) and found that it is quite high. To give an idea, on a P4 2.4GHz machine, 20 sessions give a reading of about 35%, and it will go to almost 90% for 50 sessions. If we have a system that say can accommodate 10000 users (behind nat) or concurrent calls, that probably means we need about 10000/50=200 machines. That is way too many boxes. Does anybody has a solution for this? Thanks for any help available. Regards, TC Chan

1 0

[Serusers] Incoming message
by Lazy.charles 16 Nov '05

16 Nov '05

1 0

[Serusers] Inaccurate Radius Accounting
by lenirsantiago＠yahoo.com 16 Nov '05

16 Nov '05

Hey guys, I'm using Radius to do pretty much everything regarding my users, i.e. registrations, authorization, features and preferences. I implemented my own way of doing speeddial by using AVPs provided by radius and AVPOPs module, Here is the snippet: AVP-SIP = speedial:30:15615551212 avp_subst("s:caller_speedial/i:99", "/(.*):(.*)/\2/"); #15615551212 avp_printf("i:98","$rU:*"); #30 avp_printf("i:97","sip:$avp($temp2)@$td"); # sip:… [View More]15615551212@xx.xx.xx.xx avp_printf("i:96","<sip:$avp($temp2)@$td>\r\n"); # <sip:15615551212@xx.xx.xx.xx>. xlog("L_N"," SPEEDIALING: $fU is speed-dialing $hdr(To)->$avp($temp2)\n"); if (avp_check("s:caller_speedial", "fm/$temp1/g")) { #avp_pushto("$To","i:96"); #avp_pushto("$ruri","i:97"); #avp_pushto("$duri","i:97"); #subst('/^To: (.*)(a)(.*)/To: "$avp($temp2)" <$avp($temp2)@\2/'); #append_hf("To: \"5615551212\" <15615551212(a)xx.xx.xx.xx>;user=phone"); #subst_uri('/^sip:(.*)@(.*)$/sip:$avp($temp2)@\2/i'); #subst_user('/(.*)/$avp($temp2)/'); #remove_hf("To"); rewriteuser("15615551212"); xlog("L_N"," SPEEDIALING: $fU is dialing $hdr(To) $avp($temp2)\n"); avp_delete("$temp1/g"); avp_delete("$temp2/g"); }; In this scenario, im forcing it instead of getting the value from AVPs just for ease of showing my point. The problem that I'm having is that no matter what I do to change the To header or change the R-URI, radius ALWAYS sends the accounting records with the original R-URI, in my case 30(a)xx.xx.xx.xx, how can I get acc.so to send the final resulting URI (15615551212(a)xx.xx.xx.xx) in the Called-Station-Id AVP? rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=97, length=298 Acct-Status-Type = Start Service-Type = Sip-Session Acct-Terminate-Cause = 200 Error-Cause = Invite User-Name = "1000(a)xx.xx.xx.xx" Calling-Station-Id = "sip:1000@xx.xx.xx.xx" Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway;user=phone" Acct-Session-Id = "000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" Sip-To-Tag = "230383F4-2AF" Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" Sip-CSeq = "102" NAS-IP-Address = xx.xx.xx.xx NAS-Port = 5060 Acct-Delay-Time = 0 rlm_sql (sql): Reserving sql socket id: 12 rlm_sql (sql): Released sql socket id: 12 Sending Accounting-Response of id 97 to xx.xx.xx.xx:39216 rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=98, length=292 Acct-Status-Type = Stop Service-Type = Sip-Session Acct-Terminate-Cause = 200 Error-Cause = 8 User-Name = "1000(a)xx.xx.xx.xx" Calling-Station-Id = "sip:1000@xx.xx.xx.xx" Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway:5060" Acct-Session-Id = "000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" Sip-To-Tag = "230383F4-2AF" Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" Sip-CSeq = "103" NAS-IP-Address = xx.xx.xx.xx NAS-Port = 5060 Acct-Delay-Time = 0 Thanks in advance! Lenir [View Less]

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[Users] Inaccurate Radius Accounting
by lenirsantiago＠yahoo.com 16 Nov '05

16 Nov '05

Hey guys, I'm using Radius to do pretty much everything regarding my users, i.e. registrations, authorization, features and preferences. I implemented my own way of doing speeddial by using AVPs provided by radius and AVPOPs module, Here is the snippet: AVP-SIP = speedial:30:15615551212 avp_subst("s:caller_speedial/i:99", "/(.*):(.*)/\2/"); #15615551212 avp_printf("i:98","$rU:*"); #30 avp_printf("i:97","sip:$avp($temp2)@$td"); # sip:… [View More]15615551212@xx.xx.xx.xx avp_printf("i:96","<sip:$avp($temp2)@$td>\r\n"); # <sip:15615551212@xx.xx.xx.xx>. xlog("L_N"," SPEEDIALING: $fU is speed-dialing $hdr(To)->$avp($temp2)\n"); if (avp_check("s:caller_speedial", "fm/$temp1/g")) { #avp_pushto("$To","i:96"); #avp_pushto("$ruri","i:97"); #avp_pushto("$duri","i:97"); #subst('/^To: (.*)(a)(.*)/To: "$avp($temp2)" <$avp($temp2)@\2/'); #append_hf("To: \"5615551212\" <15615551212(a)xx.xx.xx.xx>;user=phone"); #subst_uri('/^sip:(.*)@(.*)$/sip:$avp($temp2)@\2/i'); #subst_user('/(.*)/$avp($temp2)/'); #remove_hf("To"); rewriteuser("15615551212"); xlog("L_N"," SPEEDIALING: $fU is dialing $hdr(To) $avp($temp2)\n"); avp_delete("$temp1/g"); avp_delete("$temp2/g"); }; In this scenario, im forcing it instead of getting the value from AVPs just for ease of showing my point. The problem that I'm having is that no matter what I do to change the To header or change the R-URI, radius ALWAYS sends the accounting records with the original R-URI, in my case 30(a)xx.xx.xx.xx, how can I get acc.so to send the final resulting URI (15615551212(a)xx.xx.xx.xx) in the Called-Station-Id AVP? rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=97, length=298 Acct-Status-Type = Start Service-Type = Sip-Session Acct-Terminate-Cause = 200 Error-Cause = Invite User-Name = "1000(a)xx.xx.xx.xx" Calling-Station-Id = "sip:1000@xx.xx.xx.xx" Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway;user=phone" Acct-Session-Id = "000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" Sip-To-Tag = "230383F4-2AF" Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" Sip-CSeq = "102" NAS-IP-Address = xx.xx.xx.xx NAS-Port = 5060 Acct-Delay-Time = 0 rlm_sql (sql): Reserving sql socket id: 12 rlm_sql (sql): Released sql socket id: 12 Sending Accounting-Response of id 97 to xx.xx.xx.xx:39216 rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=98, length=292 Acct-Status-Type = Stop Service-Type = Sip-Session Acct-Terminate-Cause = 200 Error-Cause = 8 User-Name = "1000(a)xx.xx.xx.xx" Calling-Station-Id = "sip:1000@xx.xx.xx.xx" Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway:5060" Acct-Session-Id = "000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" Sip-To-Tag = "230383F4-2AF" Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" Sip-CSeq = "103" NAS-IP-Address = xx.xx.xx.xx NAS-Port = 5060 Acct-Delay-Time = 0 Thanks in advance! Lenir [View Less]

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RE: [Serusers] LCR question
by Bjorn Asmul 16 Nov '05

16 Nov '05

What indexes would you recommend on the LCR table? -- Bjorn -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Juha Heinanen Sent: Saturday, November 12, 2005 8:48 AM To: Stefan Prelle Cc: serusers(a)lists.iptel.org Subject: [Serusers] LCR question Stefan Prelle writes: > I am having huge performance problems. The OpenSER drops to effectivly > 1 CPS. My lcr table contains only 20 entries and I have just 2 gateways &… [View More]

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[Serusers] Am I looking in the wrong place for SEMS?
by sip 15 Nov '05

15 Nov '05

I thought I'd regrab code for SEMS lately and make sure I had the most up to date source, but I noticed the sems.berlios.de page is conspicuously blank. Has the page for SEMS been moved somewhere? While I'm at it, does anyone know of a conference-capable application (hacked SEMS, Asterisk, other) that does multicast RTP? Just playing around with a few ideas, so I was curious if anyone's run across one or two. N.

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[Serusers] Usrloc question
by Darren Nay 15 Nov '05

15 Nov '05

Hey all, I have a question regarding usrloc. I have run into a problem.. We have static routes sent to an asterisk server for all of our SIP usernames. In addition our IAD's will register with the same username, so that calls coming into our switch for that username will be routed to both the asterisk box and the SIP IAD. This way, whichever endpoint (IAD or asterisk) answers the call first will take the call. For example. root:/ # serctl ul show +18646404810 <sip:+… [View More]

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[Serusers] Info on t_fork();
by Leo Papadopoulos 15 Nov '05

15 Nov '05

Dear Serusers: I cannot seem to find information on t_fork. There is only a small reference to it the the TM.so documentation, but no description of t_fork itself. Does anyone know where I can find information about this? Leo P.

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RE: [Serusers] SEMS:announcement plug in
by rajesh.rajguru＠wipro.com 15 Nov '05

15 Nov '05

Hi, We are trying to forward INVITE to SEMS and play announcement. It doesn't play anything. The UA which is sending INVITE is getting time out But the same .wav file for announcement if I open from sound recorder manually ,then it plays. How to check that there is problem with codec ? Regards, Rajesh -----Original Message----- From: Xavier TRENTIN [mailto:xtrentin@softiphone.net] Sent: Monday, November 14, 2005 10:00 PM To: Rajesh Rajguru (WT01 - Voice & Next Generation Networks) Cc: … [View More]serusers(a)lists.iptel.org Subject: Re: [Serusers] SEMS:announcement plug in Does it play nothing or does it hang off? Perhaps the problem could be the Codec you used to record the file. B.R. Xavier. rajesh.rajguru(a)wipro.com a écrit : >*********************** >Pas de virus detecté dans l'attachement no filename > >Votre Mail a été vérifié par InterScan. >***********-*********** > > > >Hi, > >In our ser.cfg following is mentioned for announcement > >---- > if(!t_write_req("/tmp/am_fifo","announcement")) { > t_reply("500","error contacting sems"); >----- > >we are using fifo for communicating between ser and sems. >ser and sems are running on the same server (Fedora) When we start >sems, the default announcement file mentioned in sems.conf >(default_announce=default_en.wav) is not played. >Also there is no error in /var/log/messages wrt announcement plug in > >What could be the problem ? > >How does sems decides which sound recorder to play ? > > >The ll /tmp/am_fifo shows following > > >prw-r--r-- 1 root root 0 Nov 14 09:48 /tmp/am_fifo > >Regards, >Rajesh > > > > >Confidentiality Notice > > >The information contained in this electronic message and any >attachments to this message are intended for the exclusive use of the >addressee(s) and may contain confidential or privileged information. If >you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments. > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > > > Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments. [View Less]

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[Users] a rtpproxy!
by messa＠innsof.com 15 Nov '05

15 Nov '05

Hi effectively, the first error is " ca't connect to rtpproxy". Where can i download a rtpproxy? Can i installed rtpproxy on the same host? regards! Serge!

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[Serusers] [Help] Setting up SER in Redirect mode
by anonymous anonymous 15 Nov '05

15 Nov '05

Hello all, I am a student currently studying SIP with SER on a Slackware 10.2 system. I could have my test network working fine in proxy/registrar mode but I am having trouble in configuring SER for redirect mode. Actually I am totally new at SER administration and don't feel really comfortable in implementing code blocks into ser.cfg When I use the following ser.cfg file, SER simply ignore the redirect block and call alice at 192.168.1.2 whereas the call should be redirected to 192.168.1.… [View More]4 I've been driftng all around the Internet gathering hints and tips for a few days but I'm still at that point. I guess it's a programming problem but can't find what's wrong exactly. Thank you in advance for your help. Matthieu Here is my uname -a result: Linux PC1 2.4.31 #6 Sun Jun 5 19:04:47 PDT 2005 i686 unknown unknown GNU/Linux My SER build: version: ser 0.9.4 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197.2.1 2005/07/25 16:56:24 andrei Exp $ main.c compiled on 15:54:20 Sep 6 2005 with gcc 4.0.1 My ser.cfg file based upon default ser.cfg: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=8 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ listen=192.168.1.1 check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # 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" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; # redirect block, maybe at the wrong place if (uri=="sip:alice@192.168.1.2") { rewritehost("192.168.1.4"); sl_send_reply("302", "Redirect"); } }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } _________________________________________________________________ Recherchez sur votre PC comme sur le Web avec Windows Desktop Search ! http://desktop.msn.fr/ [View Less]

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[Serusers] Concurrent call limitation
by Marcello Lupo 15 Nov '05

15 Nov '05

Hi to all, any idea as to limit the concurrent calls outgoing to a particular PSTN gateway supposing to let ser becoming dialogs aware so to have the status of active open calls?? Thanks, Bye, Marcello

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[Users] Usrloc question
by Darren Nay 15 Nov '05

15 Nov '05

Hey all, I have a question regarding usrloc. I have run into a problem.. We have static routes sent to an asterisk server for all of our SIP usernames. In addition our IAD's will register with the same username, so that calls coming into our switch for that username will be routed to both the asterisk box and the SIP IAD. This way, whichever endpoint (IAD or asterisk) answers the call first will take the call. For example. root:/ # serctl ul show +18646404810 <sip:+… [View More]

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[Serusers] Illegal redirection
by Joao Pereira 15 Nov '05

15 Nov '05

Hello I have two Cisco 7940 phones with private addresses (10.0.11.239 and 10.0.11.140) connected to SER also with private address (10.0.0.135), but in another network. My SER is with the default configuration. Between the networks I have a Checkpoint Firewall-1NG The Cisco IP phones can register because the REGISTER packets arent blocked. But the INVITEs never reach SER (I checked with ngrep), because the Firewall drops them, saying there was an illegal redirection. The most strange part,… [View More] is that, when I try to make a phone call from PhoneA(10.0.11.239) to PhoneB(10.0.11.240), the INVITE is dropped before reaching SER, and it says "Illegal redirection 10.0.0.135->10.0.11.240". How can the firewall know that the INVITE was going to be redirected by SER to PhoneB(10.0.11.240) ???? my ser.cfg (the default one): # $Id: ser.cfg,v 1.25 2004/11/30 16:28:24 andrei Exp $ # simple quick-start config script # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) listen = 10.0.0.135 /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } [View Less]

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[Users] OpenSER Clients
by Gary Major 15 Nov '05

15 Nov '05

2 1

[Serusers] dynamic sip port for clients behind same firewall
by Imran Akbar 15 Nov '05

15 Nov '05

Hi, The problem I'm facing is that if I have multiple clients behind the same firewall, calls between them don't work as each is using the same sip port. The clients can be made to use a random port, but the registrar server has to listen on all ports and respond to the client using the port it was sent on... does SER support this? Is there any other way to solve this problem? Thanks, Imran

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Re: [Users] Forwarding..
by Bogdan-Andrei Iancu 15 Nov '05

15 Nov '05

Hi Sebastian, not sure if you can do this - a kind of middle registrar; but you can try to experiment by using the latest UAC module which provides the FROM mangling and authentication features. http://www.openser.org/docs/modules/1.1.x/uac.html regards, bogdan BASTARD Sebastien wrote: > Ok, it seems to work...but I have another question: > I would like to have a static configuration on the PC on a LAN with a > local account test1(a)test.net <mailto:test1@test.net>. … [View More]There is openser > running on a computer of this LAN which is managing domain test.net > <http://test.net>. > I would like to have an association of accounts on the openser server > (I know asterisk is able to do it): > SIP client connect to my local openser with account test1(a)test.net > <mailto:test1@test.net> > OPENSER associates test1(a)test.net <mailto:test1@test.net> with > test1(a)iptel.org <mailto:test1@iptel.org> and inform iptel.org > <http://iptel.org> that this account is managed locally: it's like > openser act as a client / iptel ser server. > Then I can be called from outside without problem. > > Do you think it's possible to do it or to implement a module for that. > > Thanks a lot for your help > Sebastien > > On 11/9/05, *BASTARD Sebastien* <sebastien.bastard(a)gmail.com > <mailto:sebastien.bastard@gmail.com>> wrote: > > Ok, i'm going to try. > > Thanks for your help. > > sebastien > > > On 11/8/05, *Bogdan-Andrei Iancu* < bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi, > > if I'm not wrong, you are talking about routing outbound requests > (requests which are not targeting your domain). If so, the > routing will > be done via DNS based on the domain name; you do not have to > specify > each of them. Just do like: > > if (!uri==myself) { > # this is an outbound call > t_relay(); > exit; > } > # process inbound calls > ........ > > regards, > bogdan > > BASTARD Sebastien wrote: > >> Hi, >> >> I would like to make run openser as a sip client when a client is >> connecting to it but with an unmanaged domain. I tried with the >> following configuration but with no result: >> >> if (!uri==myself) { >> # mark routing logic in request >> append_hf("P-hint: outbound\r\n"); >> # if you have some interdomain connections via TLS >> if(uri=~" iptel.org <http://iptel.org>") { >> forward("iptel.org <http://iptel.org>",5060); >> exit; >> } #else if(uri=~"@tls_domain2.net") { >> # t_relay_to_tls("IP_domain2","port_domain2"); >> # exit; >> #} >> route(1); >> }; >> >> Thanks for any help. >> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org <mailto:Users@openser.org> >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > > [View Less]

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[Users] Exec module: Use output of an external command within SER?
by Joachim Fabini 15 Nov '05

15 Nov '05

Hi, A newbie question: Is there a way for OpenSER to use SIP headers (other than the URI set) that have been modified/processed by an external command (exec_dst, exec_msg)? Reading the documentation and mailing list archives revealed some questions similar to mine but no answer. The problem is: Delegate the change of one or more SIP message headers (or the whole SIP message) to an external application via exec module. Two straight-forward solutions: 1. exec_rmsg() - for "replace message"… [View More]

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[Users] Does openser support fax ?
by Nguyen Duc Phi 15 Nov '05

15 Nov '05

Hi all, Does openser support fax? if it support How can I config openser ? Best Regards.

4 4

[Users] t_relay to multiple path
by Jayesh Nambiar 15 Nov '05

15 Nov '05

hi everyone, In my configuration, I trust some source ips and if call comes from these src ips, without any authentication they are t_relayed to a gateway ip address for pstn termination. Is there any potential problems by trusting some clients by src_ips, I do so as I dont want them to login from anywhere else. I also wanted to know that if I do not get reply from a particular gateway, can I forward them to another gateway. I dont want to use LCR for this. I thought of using failure route, … [View More]

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[Users] what about mediaproxy module?
by messa＠innsof.com 15 Nov '05

15 Nov '05

Hi all I want to know how i can configure the NAT helper module to make call between 2 LANs separated by internet,or simply to do Nat Transversal? Can the mediaproxy module help me to do NAT transversal? Regards!

2 1

[Users] how to configure openser to use nathelp
by messa＠innsof.com 15 Nov '05

15 Nov '05

Hi all I install openser in a host with private ip. i can make a call between clients in the same LAN. But now, i install openser in a host with public IP. I want to make a call between a client in LAN A and a client in a LAN B. the 2 LANs are separated by Internet.What can i use the nathelp module and configure openser to make this call possible? help me please!

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[Users] Multi-domain setup problem
by Daryl Sanders 15 Nov '05

15 Nov '05

Hi Everyone, I just added a second domain to my openser box and screwed things up somehow. Maybe someone can help figure out what's going on. I'm using the new onsip config (unmodified except IPs & replacing breaks w/returns). I'm using mediaproxy for NAT. Everything worked properly prior to adding the new domain. I did add the new domain to the domain table, and have loaded the domain module with modparams as follows. modparam("domain", "db_url", "mysql://openser:mypass@myhost/openser") … [View More]

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[Serusers] DNS problem
by Grabias Dariusz 15 Nov '05

15 Nov '05

Hello , I would like to use domain name of SER instead of IP adress during indetification I set dns=yes,rev_dns=yes and alias="sip.siptest" in ser.cfg but something strange has happened. Sip Client nr1(it's number 204) sends " INVITE sip:203@sip.siptest " to SER and SER sends "INVITE sip:203@192.168.1.204" to sip client nr2 (it's number 203). It looks very suspicious because SER should send " INVITE sip:204@sip.siptest " to client nr 2. It results in trouble when you try to phone from one … [View More]

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[Serusers] Multiplexing RTP streams
by Noel Sharpe 15 Nov '05

15 Nov '05

Hi All I was wondering if any thought (effort) has been given to trying to combine multiple rtp streams that are going to the same destination to save on IP overhead from the many small packets. I guess this would be a use case for RTPProxy (or mediaproxy) or it could be implemented as an application infront of rtpproxy. I know a number of vendors of hardware gateways (quintum/mediatrix) do this, and report massive savings (50% if memory serves) of bandwidth. I doubt very much that … [View More]

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[Serusers] Unable to stop SER: Need help
by Dhiman D Chowdhury 15 Nov '05

15 Nov '05

Hi all, I will appreciate your kind help. I am unable to stop ser and tried the followings: 1. service ser stop. [It failed but ser restart wroks] 2. killall ser. [ser: no process killed] Can anybody please help? Thanks Dhiman

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[Serusers] how to make use of the function, is_from_local, in ser?
by zhangshuai 15 Nov '05

15 Nov '05

Dear all, We want to use multiple mediaproxy by using dispatcher. We set SIP_DOMIAN = mydomain.com, set up two user: A(a)mydomain.com & B(a)mydomain.com. They both register successfully. When user A & B dials for each other, the functions of is_from_local and is_destination_local both return False, so the dispatcher only return "warning: no proxy available for call: mydomain.com -> *.*.*.*" In mysql, the domain table is empty, too. How come? Thanks in advance. My ser config … [View More]

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[Users] use of nathelper!
by messa＠innsof.com 15 Nov '05

15 Nov '05

Hi all I want to use nathelper for nat transversal. But when i make a call, i have this error: 0(11117) ERROR: force_rtp_proxy2: no available proxies 0(11117) ERROR: force_rtp_proxy2: no available proxies 0(11117) ERROR: force_rtp_proxy2: no available proxies 0(11117) ERROR: unforce_rtp_proxy: no available proxies 0(11117) ERROR: force_rtp_proxy2: no available proxies 0(11117) ERROR: unforce_rtp_proxy: no available proxies 0(11117) ERROR: force_rtp_proxy2: no available proxies 0(… [View More]

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[Users] nat transversal
by messa＠innsof.com 15 Nov '05

15 Nov '05

Hi all Thanks for your help. I make a test in local (one xlite client calls another one, both in the same LAN than the openser) and everything goes well. But when i install openser in a machine with a public IP to make call between 2 clients in 2 differents LANs, i can't make a call. I don't know why? it is a client or an openser configuration problem? Can you help me? best regards!

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[Users] modules_api
by Dariusz Dwornikowski 14 Nov '05

14 Nov '05

where can i find modules api for openser? -- Dariusz 'tdi' Dwornikowski mailto : tdi(a)vercom.pl gg: 2266034 JID : tdi(a)gentoo.pl

3 2

[Serusers] SEMS:announcement plug in
by rajesh.rajguru＠wipro.com 14 Nov '05

14 Nov '05

Hi, In our ser.cfg following is mentioned for announcement ---- if(!t_write_req("/tmp/am_fifo","announcement")) { t_reply("500","error contacting sems"); ----- we are using fifo for communicating between ser and sems. ser and sems are running on the same server (Fedora) When we start sems, the default announcement file mentioned in sems.conf (default_announce=default_en.wav) is not played. Also there is no error in /var/log/messages wrt announcement plug in What could be the problem ? … [View More]

2 1

[Serusers] prefix and the numeration plan
by Ldd 14 Nov '05

14 Nov '05

Hi, My carrier sent the information below, as installs? following prefix 45264 # and the numeration plan below: 45264# CC + AC + NNNNNNNN I try : if (uri=~"^sip:00[0-9].*@.*") { if (is_user_in("From", "ld")) { sl_send_reply("403", "Payment required"); break; }; setflag(1); prefix("45264#") rewritehostport("ip-server:5060"); if (!t_relay()) { … [View More]

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[Users] substr in uri
by Iqbal 14 Nov '05

14 Nov '05

Hi I am doing the following and for some reason it doesn't seem to work (please ignore syntax) if (uri=0[1-9] { substr/^0/00/} and then if {uri=00) Now should the second if statement now match, or not, it my config it doesn't seem to. iqbal

2 4

[Users] Geting status and reason in failure_route
by Taras Bendik 14 Nov '05

14 Nov '05

Hi ----------------------------- failure_route[1] { if (t_check_status("408|500|503")) { t_reply("503", "Sorry"); } } ----------------------------- Is it posible to add loging in failure route, so User get 503, but I get acctual info on what happen, because $rs $rr work only in onreply_route, but I need them or any equivalent in failure_route? So I can log exactly what happen. Regards, Taras

2 2

[Users] $ruri
by Taras Bendik 14 Nov '05

14 Nov '05

Hi Im using avp_write("sip:123456@1.2.3.4:4321", "i:100"); avp_pushto("$ruri","i:100"); t_relay(); And this is sended to 1.2.3.4:4321 like : INVITE sip:123456@1.2.3.4:4321 SIP/2.0 And this works fine. However can I make it some how to generate message like this : INVITE sip:123456 SIP/2.0 ??? Or in this case I need to use : avp_write("sip:123456", "i:100"); avp_pushto("$ruri","i:100"); t_relay_to_udp("1.2.3.4","4321"); Regards, Taras

2 1

[Users] LCR documentation, examples, pointers?
by rushowr 14 Nov '05

14 Nov '05

Hello all, Hate to be a bother to the list, but the documentation for the LCR module (and googling afterwards) left me slightly puzzled. . . The way I read it, it seems like EVERY user on the system would have to have an entry in the gw table, in order to match the FROM URI and get a gateway prioritized.... Anyone have some light to shed on LCR? Maybe a solid doc somewhere? Other than that, please keep up the excellent work! O_SER is definitely a wonderful piece of work! SKM

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[Serusers] ACC early_media issue ...
by Mamen xu 14 Nov '05

14 Nov '05

Hello, all: I am using SER interworking with Cisco26 as PSTN gateway. So far, everything goes OK but the Radius billing. There is only STOP Radius request with no START Radius request at first. I think it's due to 183 with SDP issue. But it seems early_media in ACC module didn't give it a remedy. BTW, START/STOP runs OK in intra-domain call..which including 180 without SDP in signaling loop. My ser.cfg is as below: modparam("acc", "log_level",1) modparam("acc","radius_flag",1) modparam(… [View More]

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[Serusers] Audio player for SEMS
by hemal.sheth＠wipro.com 14 Nov '05

14 Nov '05

We are trying to use SEMS along with SER in our lab setup We have configured a dedicated SEMS for one IP phone to start with And enabled the announcement plug-in But the announcement message is not played Where does SEMS take the default player from? Thanks Hemal Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you … [View More]

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[Users] Re: nat transversal with mediaproxy
by Daryl Sanders 13 Nov '05

13 Nov '05

On 11/13/05, serge messa <sergepmessa(a)yahoo.fr> wrote: > Hi > > Thanks for your reply. must i make changes in the > mediaproxy.ini , the mediaproxy.cfg or openser.cfg > file? I had to configure mediaproxy.ini, but I am running mediaproxy on a few different, boxes. You might not if you are only running it local. Can anyone confirm this? > I want to know if my openser.cfg config file have any > problem? > > Have you use openser with the mediaproxy module? … [View More]

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[Serusers] Question Compiling acc.so
by Leo Papadopoulos 13 Nov '05

13 Nov '05

Dear serusers, I am running ser 0.9.3 I want to enable the accounting package to write to MySQL. I realize I need to recompile to do this. Can I just compile ONLY the acc and somehow install this acc module? If so, how? Is it just easier or better to recompile all of SER? ________________________________ Telecom CTO Leo Papadopoulos 10 Old Stone Court Ridgefield, CT 06877 Tel: 203-438-8117 Cell: 203-788-6364 E-mail: leo(a)telecomCTO.com Web site: www.telecomCTO.com

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[Serusers] Re: NAT Problem and Stun Server
by Elangainesan Shanmugam 13 Nov '05

13 Nov '05

Dear serusers, I am running SER 9.4 . it works great. I am running SER outside on public address I am also having small problem behind NAT. To resolv this I am running Stun server - public address with Stun Client - private address ( behind NAT address ) The clients were not able to find its location. But If i do enter Public ip addresss for that system by finding via www.ip2location.com , Both incomming and out going works. The Problem is every time , i have to manually enter the … [View More]ipaddress. Not sure, if I have to put both Stun server and Stun Client on the public network. And not sure, Stun Client process gives the ip address of the location Please help me Regards Nesan ----- Original Message ----- From: <serusers-request(a)lists.iptel.org> To: <serusers(a)lists.iptel.org> Sent: Sunday, November 13, 2005 5:00 AM Subject: Serusers Digest, Vol 31, Issue 15 > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. LCR question (Juha Heinanen) > 2. Question Compiling acc.so (Leo Papadopoulos) > 3. Re: NAT Traversal (Thomas Britis) > 4. Multi-domain setup problem (Daryl Sanders) > 5. Re: Question Compiling acc.so (sip) > 6. Re: mediaproxy for incoming (George Lambson) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sat, 12 Nov 2005 15:48:01 +0200 > From: Juha Heinanen <jh(a)tutpro.com> > Subject: [Serusers] LCR question > To: Stefan Prelle <s.prelle(a)broadnet.de> > Cc: serusers(a)lists.iptel.org > Message-ID: <17269.62097.141780.968729(a)rautu.tutpro.com> > Content-Type: text/plain; charset=us-ascii > > Stefan Prelle writes: > > > I am having huge performance problems. The OpenSER drops to effectivly > > 1 CPS. My lcr table contains only 20 entries and I have just 2 gateways > > configured, so this shouldn't be a problem. > > currently load_gws() makes a complex mysql query, but only one mysql > query per second seems very slow to me. load_gws could be rewritten > totally in c, but so far nobody has had time to do so. > > -- juha > > > > ------------------------------ > > Message: 2 > Date: Sat, 12 Nov 2005 10:26:34 -0500 > From: "Leo Papadopoulos" <leo(a)telecomcto.com> > Subject: [Serusers] Question Compiling acc.so > To: <serusers(a)lists.iptel.org> > Message-ID: <006201c5e79d$77ea4250$0a01010a@leoathlon> > Content-Type: text/plain; charset="us-ascii" > > Dear serusers, > > I am running ser 0.9.3 > I want to enable the accounting package to write to MySQL. I realize I need > to recompile to do this. > > Can I just compile ONLY the acc and somehow install this acc module? If so, > how? > Is it just easier or better to recompile all of SER? > > ________________________________ > Telecom CTO > Leo Papadopoulos > 10 Old Stone Court > Ridgefield, CT 06877 > Tel: 203-438-8117 > Cell: 203-788-6364 > E-mail: leo(a)telecomCTO.com > Web site: www.telecomCTO.com > > > > > ------------------------------ > > Message: 3 > Date: Sat, 12 Nov 2005 09:36:54 -0200 > From: Thomas Britis <thomas(a)tcnet.com.br> > Subject: Re: [Serusers] NAT Traversal > To: "Greger V. Teigre" <greger(a)teigre.com> > Cc: serusers(a)lists.iptel.org > Message-ID: <4375D3D6.1070805(a)tcnet.com.br> > Content-Type: text/plain; charset=us-ascii; format=flowed > > I tried. Everything goes fine until I change the default configuration > from onsip.org. All the configuration files are basical and needs a lot > of changes to make me happy, heh. > > If anyone has an idea, it would be great. > > Thank you. > > Greger V. Teigre wrote: > > Seems like you could use the Getting Started document from ONsip.org ;-) > > g-) > > ----- Original Message ----- From: "Thomas Britis" <thomas(a)tcnet.com.br> > > To: <serusers(a)lists.iptel.org> > > Sent: Friday, November 11, 2005 5:58 PM > > Subject: [Serusers] NAT Traversal > > > > > >> Hi, > >> > >> I'm working on a popular cenario (I think) of SER's implementation but > >> I'm having some difficult on making clients behind NAT to work well. > >> I tried to use rtpproxy and mediaproxy (not together, of course) but > >> without success. > >> Here is my scenario: SER with public IP address with rtpproxy and/or > >> mediaproxy up and running. Clients are behind NAT with NO firewall > >> blocking any ports. Or even one client behind NAT and other with valid > >> IP (by the way, all clients that are not behind nat works fine). The one > >> behind nat do not receive any voice but can send it. > >> > >> I'm attaching my ser.cfg (both, the rtpproxy and mediaproxy one) and > >> appreciate any help. > >> > >> Thank's in advance. > >> -- > >> Thomas Storino Britis > >> TCNet Informatica e Telecomunicacoes LTDA > >> > > > > > > -------------------------------------------------------------------------- ------ > > > > > > > >> # > >> # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ > >> # > >> # simple quick-start config script > >> # > >> > >> # ----------- global configuration parameters ------------------------ > >> > >> debug=3 # debug level (cmd line: -dddddddddd) > >> fork=yes > >> log_stderror=no # (cmd line: -E) > >> > >> # TCNet Configuration > >> # hostname matching an alias will satisfy the condition uri==myself. > >> listen="200.167.20.34" > >> alias="e-voip.com.br" > >> alias="200.167.20.34" > >> uid="ser" > >> gid="ser" > >> > >> check_via=no # (cmd. line: -v) > >> dns=no # (cmd. line: -r) > >> rev_dns=no # (cmd. line: -R) > >> port=5060 > >> children=4 > >> fifo="/tmp/ser_fifo" > >> fifo_db_url="mysql://ser:pass@localhost/ser" > >> > >> # Modules > >> loadmodule "/tcnet/ser/lib/ser/modules/domain.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/mysql.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/sl.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/tm.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/rr.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/maxfwd.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/usrloc.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/registrar.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/auth.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/auth_db.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/uri.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/uri_db.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/nathelper.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/mediaproxy.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/textops.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/avpops.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/acc.so" > >> > >> # Modules parameter > >> # auth_db > >> modparam("auth_db|uri_db|usrloc", "db_url", > >> "mysql://ser:pass@localhost/ser") > >> modparam("auth_db", "calculate_ha1", 1) > >> modparam("auth_db", "password_column", "password") > >> > >> # mediaproxy > >> modparam("mediaproxy", "mediaproxy_socket", > >> "/var/run/proxydispatcher.sock") > >> modparam("mediaproxy", "natping_interval", 30) > >> modparam("mediaproxy", "sip_asymmetrics", > >> "/tcnet/mediaproxy/sip-asymmetrics-clients") > >> modparam("mediaproxy", "rtp_asymmetrics", > >> "/tcnet/mediaproxy/rtp-asymmetrics-clients") > >> > >> # usrloc > >> modparam("usrloc", "db_mode", 2) > >> > >> # rr > >> modparam("rr", "enable_full_lr", 1) > >> > >> # registrar > >> modparam("registrar", "nat_flag", 6) > >> > >> # acc > >> modparam("acc", "radius_config", > >> "/usr/local/etc/radiusclient-ng/radiusclient.conf") > >> modparam("acc", "log_level", 1) > >> modparam("acc", "log_flag", 1) > >> modparam("acc", "radius_flag", 1) > >> modparam("acc", "radius_missed_flag", 2) > >> modparam("acc", "log_fmt", "miocfst") > >> modparam("acc", "failed_transactions", 1) > >> > >> # tm > >> modparam("tm", "fr_inv_timer", 27) > >> modparam("tm", "fr_inv_timer_avp", "inv_timeout") > >> > >> # main routing logic > >> route { > >> > >> if (method == "BYE" || method == "CANCEL") { > >> end_media_session(); > >> setflag(1); # accounting > >> setflag(2); # missed call > >> }; > >> > >> # initial sanity checks -- messages with > >> # max_forwards==0, or excessively long requests > >> if (!mf_process_maxfwd_header("10")) { > >> if (method!="ACK") { > >> sl_send_reply("483", "Too Many Hops"); > >> break; > >> }; > >> }; > >> if (msg:len > max_len) { > >> if (method!="ACK") { > >> sl_send_reply("513", "Message too big"); > >> break; > >> }; > >> }; > >> > >> if (method!="REGISTER") { > >> record_route (); > >> }; > >> > >> if (loose_route()) { > >> if (method=="BYE" || method=="CANCEL") { > >> end_media_session(); > >> setflag(1); > >> setflag(2); > >> }; > >> if (method=="INVITE" || method=="ACK") { > >> use_media_proxy(); > >> }; > >> route(1); > >> break; > >> }; > >> > >> if (uri == myself) { > >> if (method == "REGISTER") { > >> if (search("^(Contact|m): > >> .*(a)(200\.167\.20\.34|evoip03\.tcnet\.com\.br|sip\.e-voip\.com\.br)")) { > >> log("LOG: alert: someone trying to set aor==contact\n"); > >> sl_send_reply("476", "No Server Address in Contacts Allowed"); > >> break; > >> }; > >> > >> route(2); # WWW authentication > >> > >> if (!save("location")) { > >> sl_reply_error(); > >> }; > >> break; > >> }; > >> > >> if (method == "INVITE") { > >> route(3); # PROXY authentication > >> if (uri=~"^sip:0") { > >> route(4); # Asterisk > >> }; > >> > >> consume_credentials(); > >> > >> #lookup("location"); > >> if (!lookup("location")) { > >> if (!lookup("aliases")) { > >> if (uri=~"^sip:[3789][0-9]") { > >> prefix("035"); > >> route(4); # Asterisk > >> } else { > >> sl_send_reply ("404", "Not Found"); > >> break; > >> }; > >> }; > >> }; > >> setflag(1); > >> setflag(2); > >> route(1); > >> break; > >> }; > >> }; > >> route(1); > >> } > >> > >> # Fix NATED client on reply route. > >> onreply_route[1] { > >> if (client_nat_test("1") && status=~"(183)|(2[0-9][0-9])") { > >> fix_contact(); > >> }; > >> use_media_proxy (); > >> } > >> > >> # Normal relay > >> route[1] { > >> > >> if (uri =~ "[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")) { > >> sl_send_reply("479", "We don't forward to private IP addresses"); > >> break; > >> }; > >> > >> t_on_reply("1"); > >> t_on_failure("1"); > >> > >> if (!t_relay()) { > >> sl_reply_error(); > >> }; > >> } > >> > >> # WWW authentication > >> route[2] { > >> if (client_nat_test("3")) { > >> setflag(6); > >> force_rport(); > >> fix_contact(); > >> }; > >> > >> sl_send_reply("100", "Trying"); > >> > >> if (!www_authorize("", "subscriber")) { > >> www_challenge("", "0"); > >> break; > >> }; > >> > >> if (!check_to()) { > >> sl_send_reply ("401", "Unauthorized"); > >> break; > >> }; > >> > >> } > >> > >> # PROXY authentication > >> route[3] { > >> if (!proxy_authorize("", "subscriber")) { > >> proxy_challenge("", "0"); > >> break; > >> }; > >> } > >> > >> # Asterisk - E-Voip > >> route[4] { > >> t_on_reply("1"); > >> rewritehost("200.167.20.26"); > >> avp_write("i:45", "inv_timeout"); > >> } > >> > >> failure_route[1] { > >> end_media_session(); > >> } > >> > > > > > > -------------------------------------------------------------------------- ------ > > > > > > > >> # > >> # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ > >> # > >> # simple quick-start config script > >> # > >> > >> # ----------- global configuration parameters ------------------------ > >> > >> debug=3 # debug level (cmd line: -dddddddddd) > >> fork=yes > >> log_stderror=no # (cmd line: -E) > >> > >> # TCNet Configuration > >> # hostname matching an alias will satisfy the condition uri==myself. > >> listen="200.167.20.34" > >> alias="e-voip.com.br" > >> alias="200.167.20.34" > >> uid="ser" > >> gid="ser" > >> > >> check_via=no # (cmd. line: -v) > >> dns=no # (cmd. line: -r) > >> rev_dns=no # (cmd. line: -R) > >> port=5060 > >> children=4 > >> fifo="/tmp/ser_fifo" > >> fifo_db_url="mysql://ser:pass@localhost/ser" > >> > >> # Modules > >> loadmodule "/tcnet/ser/lib/ser/modules/mysql.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/sl.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/tm.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/rr.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/maxfwd.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/usrloc.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/registrar.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/auth.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/auth_db.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/uri.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/uri_db.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/nathelper.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/textops.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/avpops.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/acc.so" > >> > >> # Modules parameter > >> # auth_db > >> modparam("auth_db|uri_db|usrloc", "db_url", > >> "mysql://ser:pass@localhost/ser") > >> modparam("auth_db", "calculate_ha1", 1) > >> modparam("auth_db", "password_column", "password") > >> > >> # nathelper > >> modparam("nathelper", "natping_interval", 30) > >> modparam("nathelper", "ping_nated_only", 1) > >> modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") > >> > >> # usrloc > >> modparam("usrloc", "db_mode", 2) > >> > >> # rr > >> modparam("rr", "enable_full_lr", 1) > >> > >> # registrar > >> modparam("registrar", "nat_flag", 6) > >> > >> # acc > >> modparam("acc", "radius_config", > >> "/usr/local/etc/radiusclient-ng/radiusclient.conf") > >> modparam("acc", "log_level", 1) > >> modparam("acc", "log_flag", 1) > >> modparam("acc", "radius_flag", 1) > >> modparam("acc", "radius_missed_flag", 2) > >> modparam("acc", "log_fmt", "miocfst") > >> modparam("acc", "failed_transactions", 1) > >> > >> # tm > >> modparam("tm", "fr_inv_timer", 27) > >> modparam("tm", "fr_inv_timer_avp", "inv_timeout") > >> > >> # main routing logic > >> route { > >> > >> if (method == "BYE" || method == "CANCEL") { > >> unforce_rtp_proxy(); > >> setflag(1); # accounting > >> setflag(2); # missed call > >> }; > >> > >> # initial sanity checks -- messages with > >> # max_forwards==0, or excessively long requests > >> if (!mf_process_maxfwd_header("10")) { > >> sl_send_reply("483", "Too Many Hops"); > >> break; > >> }; > >> if (msg:len > max_len) { > >> sl_send_reply("513", "Message too big"); > >> break; > >> }; > >> > >> if (method!="REGISTER") { > >> record_route (); > >> }; > >> > >> if (loose_route()) { > >> if (method == "BYE") { > >> setflag(1); > >> setflag(2); > >> unforce_rtp_proxy(); > >> }; > >> route(1); > >> break; > >> }; > >> > >> if (uri == myself) { > >> if (method=="INVITE") { > >> fix_nated_sdp ("1"); > >> }; > >> > >> if (method == "REGISTER") { > >> if (search("^(Contact|m): > >> .*(a)(200\.167\.20\.34|evoip03\.tcnet\.com\.br|sip\.e-voip\.com\.br)")) { > >> log("LOG: alert: someone trying to set aor==contact\n"); > >> sl_send_reply("476", "No Server Address in Contacts Allowed"); > >> break; > >> }; > >> > >> route(2); # WWW authentication > >> > >> if (!save("location")) { > >> sl_reply_error(); > >> }; > >> break; > >> }; > >> > >> if (method == "INVITE") { > >> route(3); # PROXY authentication > >> if (uri=~"^sip:0") { > >> route(4); # Asterisk > >> }; > >> > >> consume_credentials(); > >> > >> #lookup("location"); > >> if (!lookup("location")) { > >> if (!lookup("aliases")) { > >> if (uri=~"^sip:[3789][0-9]") { > >> prefix("035"); > >> route(4); # Asterisk > >> } else { > >> sl_send_reply ("404", "Not Found"); > >> break; > >> }; > >> }; > >> }; > >> setflag(1); > >> setflag(2); > >> route(1); > >> break; > >> }; > >> }; > >> route(1); > >> } > >> > >> # Fix NATED client on reply route. > >> onreply_route[1] { > >> if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { > >> if (!search("Content-Length:[ ]*0")) { > >> force_rtp_proxy(); > >> }; > >> }; > >> } > >> > >> # Normal relay > >> route[1] { > >> > >> if (uri =~ "[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")) { > >> sl_send_reply("479", "We don't forward to private IP addresses"); > >> break; > >> }; > >> > >> t_on_reply("1"); > >> > >> if (isflagset(6) || isflagset(5)) { > >> log (1, "oi"); > >> force_rtp_proxy (); > >> }; > >> > >> if (!t_relay()) { > >> sl_reply_error(); > >> }; > >> } > >> > >> # WWW authentication > >> route[2] { > >> if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { > >> setflag(6); > >> fix_nated_register(); > >> force_rport(); > >> }; > >> > >> sl_send_reply("100", "Trying"); > >> > >> if (!www_authorize("", "subscriber")) { > >> www_challenge("", "0"); > >> break; > >> }; > >> > >> if (!check_to()) { > >> sl_send_reply ("401", "Unauthorized"); > >> break; > >> }; > >> > >> if (isflagset(5)) { > >> setflag(6); > >> }; > >> } > >> > >> # PROXY authentication > >> route[3] { > >> if (nat_uac_test("19")) { > >> setflag(6); > >> }; > >> > >> if (!proxy_authorize("", "subscriber")) { > >> proxy_challenge("", "0"); > >> break; > >> }; > >> } > >> > >> # Asterisk - E-Voip > >> route[4] { > >> if (isflagset(6)) { > >> force_rport(); > >> fix_nated_contact(); > >> force_rtp_proxy(); > >> }; > >> > >> t_on_reply("1"); > >> rewritehost("200.167.20.26"); > >> avp_write("i:45", "inv_timeout"); > >> } > >> > > > > > > -------------------------------------------------------------------------- ------ > > > > > > > >> _______________________________________________ > >> Serusers mailing list > >> serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > >> > > > > > > > > -- > Thomas Storino Britis > TCNet Informatica e Telecomunicacoes LTDA > > > > ------------------------------ > > Message: 4 > Date: Sat, 12 Nov 2005 14:59:01 -0700 > From: Daryl Sanders <daryl.sanders(a)gmail.com> > Subject: [Serusers] Multi-domain setup problem > To: users(a)openser.org, SER Users <serusers(a)lists.iptel.org> > Message-ID: > <cffd7ecd0511121359l3d7a135ev84f81087534e3a84(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi Everyone, > > I just added a second domain to my openser box and screwed things up somehow. > Maybe someone can help figure out what's going on. > I'm using the new onsip config (unmodified except IPs & replacing > breaks w/returns). > I'm using mediaproxy for NAT. Everything worked properly prior to > adding the new domain. > I did add the new domain to the domain table, and have loaded the > domain module with modparams as follows. > > modparam("domain", "db_url", "mysql://openser:mypass@myhost/openser") > modparam("domain", "db_mode", 1) > modparam("domain", "domain_table", "domain") > modparam("domain", "domain_col", "domain") > > Existing domain: sip1.mydomain.com > New domain: sip1.mydomain.com > > user1: X-Lite (behind NAT) > private-IP (192.168.x.x) > Registers using domain sip1.mydomain.com > > user2: Linksys PAP2-NA > public-IP (same subnet as Utsarcom) > Registers using domain sip1.mydomain.com > > user3: Utstarcom iAN-02EX > public-IP (63.225.x.x) > Registers using domain sip2.mydomain.com > > Here is what happens now... > > - All users can register fine. > - user1 & user2 can only receives calls from user3 (can't call others > or pstn , 407 Proxy Auth Required) > - user3 is the only one who can call others (calls to user1,user2, & > pstn are fine) > > It's probably just something simple I missed, but I can't seem to > figure it out. As always, thanks for your assistance. > > - Daryl > > > > ------------------------------ > > Message: 5 > Date: Sat, 12 Nov 2005 17:02:34 -0500 > From: "sip" <sip(a)arcdiv.com> > Subject: Re: [Serusers] Question Compiling acc.so > To: "Leo Papadopoulos" <leo(a)telecomcto.com>, <serusers(a)lists.iptel.org> > Message-ID: <20051112215625.M80825(a)infinideas.com> > Content-Type: text/plain; charset=iso-8859-1 > > It's actually pretty easy to just recompile/install a module at a time. The > recompilation part is in the docs (INSTALL file). > > Basically, from the root directory of your source code... > > make modules=modules/acc modules > > That would recompile the acc module. My recommendation, however, is to go > INTO the src/modules/acc directoy and do a make clean (it will whine about how > you should run this from the root directory, but ignore that -- that's just a > stock message and doesn't pertain to make cleans). THEN recompile the module > from the root of the src tree to make sure any stale object files are no > longer lying about mucking up your recompile. > > Once it's been compiled, stop ser (important step), and just copy the > resulting .so (which will be in the modules/acc directory) file into the > /usr/local/lib/ser/modules directory and you're good to go. > > Restart ser and watch for weird error messages in the log. :) > > STOPPING ser is very important if you're replacing one of its modules, or it > will panic, die, and leave a bunch of core files sitting in your / tree. > Always messy. > > N. > > > On Sat, 12 Nov 2005 10:26:34 -0500, Leo Papadopoulos wrote > > Dear serusers, > > > > I am running ser 0.9.3 > > I want to enable the accounting package to write to MySQL. I realize > > I need to recompile to do this. > > > > Can I just compile ONLY the acc and somehow install this acc module? > > If so, how? Is it just easier or better to recompile all of SER? > > > > ________________________________ > > Telecom CTO > > Leo Papadopoulos > > 10 Old Stone Court > > Ridgefield, CT 06877 > > Tel: 203-438-8117 > > Cell: 203-788-6364 > > E-mail: leo(a)telecomCTO.com > > Web site: www.telecomCTO.com > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > ------------------------------ > > Message: 6 > Date: Sat, 12 Nov 2005 17:50:37 -0700 > From: George Lambson <LambsonGE(a)mtc.byu.edu> > Subject: Re: [Serusers] mediaproxy for incoming > To: greger(a)teigre.com > Cc: serusers(a)lists.iptel.org > Message-ID: <s3762bd8.008(a)mtcemail.mtc.byu.edu> > Content-Type: text/plain; charset=US-ASCII > > Any idea what I might be doing wrong? > > I have had some issues with setting up the served domain. I use x-lite for my SIP UAs and have to put the host name of my SER server to get them to log in, even though I set the environment variable for the SIP domain of the server to my domain. Do you think that it might be related? > > Thanks, > George > > >>> "Greger V. Teigre" <greger(a)teigre.com> 11/11/2005 12:19:59 AM >>> > I believe the standard NAT traversal configs from onsip.org should handle > that. > g-) > ----- Original Message ----- > From: "George Lambson" <LambsonGE(a)mtc.byu.edu> > To: <serusers(a)lists.iptel.org> > Sent: Friday, November 11, 2005 12:51 AM > Subject: [Serusers] mediaproxy for incoming > > > > Is anyone using mediaproxy for INCOMING calls? > > > > What I mean is: is SER capable of accepting INVITE messages from an > > unregistered UA that is behind a NAT Firewall and connecting them with a > > registered UA? > > > > I would like to be able to receive incoming calls for my domain from > > anyone on the internet and connect them to the proper user. Please tell me > > anyone if you are successfully doing this. > > > > Thanks, > > George > > > > _______________________________________________ > > 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 > > > > ------------------------------ > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > End of Serusers Digest, Vol 31, Issue 15 > **************************************** > [View Less]

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Re: [Serusers] mediaproxy for incoming
by George Lambson 13 Nov '05

13 Nov '05

Any idea what I might be doing wrong? I have had some issues with setting up the served domain. I use x-lite for my SIP UAs and have to put the host name of my SER server to get them to log in, even though I set the environment variable for the SIP domain of the server to my domain. Do you think that it might be related? Thanks, George >>> "Greger V. Teigre" <greger(a)teigre.com> 11/11/2005 12:19:59 AM >>> I believe the standard NAT traversal configs from onsip.org … [View More]

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[Users] openser config file to use mediaproxy
by messa＠innsof.com 12 Nov '05

12 Nov '05

Hi all I want to use madiaproxy module to do nat transversal. But it not work for me. I run the media proxy server (proxy server) in the same host than openser on a host with public IP and i set the openser configuration file to above: # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes … [View More]log_stderror=no # (cmd line: -E) #Uncomment these lines to enter debugging mode fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" #loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" #loadmodule "/usr/local/lib/openser/modules/auth.so" #loadmodule "/usr/local/lib/openser/modules/auth_radius.so" #loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" #loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" #loadmodule "/usr/local/lib/openser/modules/uri_db.so" #loadmodule "/usr/local/lib/openser/modules/uri_radius.so" #modparam("auth_db|usrloc|uri_db", "db_url","mysql://ser:heslo@localhost/ser") #modparam("auth_db", "calculate_ha1", 1) #modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "rtpproxy_sock","unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 0) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { #---------------------------------------------------------------- # Sanity Check Section #----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; #----------------------------------------------------------------- # Record Route Section #----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); }; #----------------------------------------------------------------- # Loose Route Section #----------------------------------------------------------------- if (loose_route()) { if (has_totag() && method=="INVITE") { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); exit; }; #----------------------------------------------------------------- # Call Type Processing Section #----------------------------------------------------------------- if (uri==myself) { if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); }; } route[1] { #----------------------------------------------------------------- # Default Message Handler #----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { #----------------------------------------------------------------- # REGISTER Message Handler #---------------------------------------------------------------- if (!search("^Contact: \*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); # fix_nated_contact(); force_rport(); }; sl_send_reply("100", "Trying"); # if (!www_authorize("","subscriber")) { # www_challenge("","0"); # exit; # }; # if (!check_to()) { # sl_send_reply("401", "Unauthorized"); # exit; # }; # consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { #----------------------------------------------------------------- # INVITE Message Handler #----------------------------------------------------------------- if (nat_uac_test("19")) { setflag(6); }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; # if (!proxy_authorize("","subscriber")) { # proxy_challenge("","0"); # break; # } else if (!check_from()) { # if(!check_from()) { # sl_send_reply("403", "Use From=ID"); # exit; # }; # consume_credentials(); if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } onreply_route[1] { if (isflagset(6) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ 0")) { force_rtp_proxy(); }; } else if (nat_uac_test("1")) { fix_nated_contact(); }; } What's wrong? your comments are welcome. Thanks Serge MESSA [View Less]

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[Serusers] NAT Traversal
by Thomas Britis 12 Nov '05

12 Nov '05

Hi, I'm working on a popular cenario (I think) of SER's implementation but I'm having some difficult on making clients behind NAT to work well. I tried to use rtpproxy and mediaproxy (not together, of course) but without success. Here is my scenario: SER with public IP address with rtpproxy and/or mediaproxy up and running. Clients are behind NAT with NO firewall blocking any ports. Or even one client behind NAT and other with valid IP (by the way, all clients that are not behind nat … [View More]works fine). The one behind nat do not receive any voice but can send it. I'm attaching my ser.cfg (both, the rtpproxy and mediaproxy one) and appreciate any help. Thank's in advance. -- Thomas Storino Britis TCNet Informatica e Telecomunicacoes LTDA # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) # TCNet Configuration # hostname matching an alias will satisfy the condition uri==myself. listen="200.167.20.34" alias="e-voip.com.br" alias="200.167.20.34" uid="ser" gid="ser" check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:pass@localhost/ser" # Modules loadmodule "/tcnet/ser/lib/ser/modules/domain.so" loadmodule "/tcnet/ser/lib/ser/modules/mysql.so" loadmodule "/tcnet/ser/lib/ser/modules/sl.so" loadmodule "/tcnet/ser/lib/ser/modules/tm.so" loadmodule "/tcnet/ser/lib/ser/modules/rr.so" loadmodule "/tcnet/ser/lib/ser/modules/maxfwd.so" loadmodule "/tcnet/ser/lib/ser/modules/usrloc.so" loadmodule "/tcnet/ser/lib/ser/modules/registrar.so" loadmodule "/tcnet/ser/lib/ser/modules/auth.so" loadmodule "/tcnet/ser/lib/ser/modules/auth_db.so" loadmodule "/tcnet/ser/lib/ser/modules/uri.so" loadmodule "/tcnet/ser/lib/ser/modules/uri_db.so" loadmodule "/tcnet/ser/lib/ser/modules/nathelper.so" loadmodule "/tcnet/ser/lib/ser/modules/mediaproxy.so" loadmodule "/tcnet/ser/lib/ser/modules/textops.so" loadmodule "/tcnet/ser/lib/ser/modules/avpops.so" loadmodule "/tcnet/ser/lib/ser/modules/acc.so" # Modules parameter # auth_db modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:pass@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") # mediaproxy modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "natping_interval", 30) modparam("mediaproxy", "sip_asymmetrics", "/tcnet/mediaproxy/sip-asymmetrics-clients") modparam("mediaproxy", "rtp_asymmetrics", "/tcnet/mediaproxy/rtp-asymmetrics-clients") # usrloc modparam("usrloc", "db_mode", 2) # rr modparam("rr", "enable_full_lr", 1) # registrar modparam("registrar", "nat_flag", 6) # acc modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_fmt", "miocfst") modparam("acc", "failed_transactions", 1) # tm modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # main routing logic route { if (method == "BYE" || method == "CANCEL") { end_media_session(); setflag(1); # accounting setflag(2); # missed call }; # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too Many Hops"); break; }; }; if (msg:len > max_len) { if (method!="ACK") { sl_send_reply("513", "Message too big"); break; }; }; if (method!="REGISTER") { record_route (); }; if (loose_route()) { if (method=="BYE" || method=="CANCEL") { end_media_session(); setflag(1); setflag(2); }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; route(1); break; }; if (uri == myself) { if (method == "REGISTER") { if (search("^(Contact|m): .*(a)(200\.167\.20\.34|evoip03\.tcnet\.com\.br|sip\.e-voip\.com\.br)")) { log("LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed"); break; }; route(2); # WWW authentication if (!save("location")) { sl_reply_error(); }; break; }; if (method == "INVITE") { route(3); # PROXY authentication if (uri=~"^sip:0") { route(4); # Asterisk }; consume_credentials(); #lookup("location"); if (!lookup("location")) { if (!lookup("aliases")) { if (uri=~"^sip:[3789][0-9]") { prefix("035"); route(4); # Asterisk } else { sl_send_reply ("404", "Not Found"); break; }; }; }; setflag(1); setflag(2); route(1); break; }; }; route(1); } # Fix NATED client on reply route. onreply_route[1] { if (client_nat_test("1") && status=~"(183)|(2[0-9][0-9])") { fix_contact(); }; use_media_proxy (); } # Normal relay route[1] { if (uri =~ "[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")) { sl_send_reply("479", "We don't forward to private IP addresses"); break; }; t_on_reply("1"); t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; } # WWW authentication route[2] { if (client_nat_test("3")) { setflag(6); force_rport(); fix_contact(); }; sl_send_reply("100", "Trying"); if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; if (!check_to()) { sl_send_reply ("401", "Unauthorized"); break; }; } # PROXY authentication route[3] { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; } # Asterisk - E-Voip route[4] { t_on_reply("1"); rewritehost("200.167.20.26"); avp_write("i:45", "inv_timeout"); } failure_route[1] { end_media_session(); } # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) # TCNet Configuration # hostname matching an alias will satisfy the condition uri==myself. listen="200.167.20.34" alias="e-voip.com.br" alias="200.167.20.34" uid="ser" gid="ser" check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:pass@localhost/ser" # Modules loadmodule "/tcnet/ser/lib/ser/modules/mysql.so" loadmodule "/tcnet/ser/lib/ser/modules/sl.so" loadmodule "/tcnet/ser/lib/ser/modules/tm.so" loadmodule "/tcnet/ser/lib/ser/modules/rr.so" loadmodule "/tcnet/ser/lib/ser/modules/maxfwd.so" loadmodule "/tcnet/ser/lib/ser/modules/usrloc.so" loadmodule "/tcnet/ser/lib/ser/modules/registrar.so" loadmodule "/tcnet/ser/lib/ser/modules/auth.so" loadmodule "/tcnet/ser/lib/ser/modules/auth_db.so" loadmodule "/tcnet/ser/lib/ser/modules/uri.so" loadmodule "/tcnet/ser/lib/ser/modules/uri_db.so" loadmodule "/tcnet/ser/lib/ser/modules/nathelper.so" loadmodule "/tcnet/ser/lib/ser/modules/textops.so" loadmodule "/tcnet/ser/lib/ser/modules/avpops.so" loadmodule "/tcnet/ser/lib/ser/modules/acc.so" # Modules parameter # auth_db modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:pass@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") # nathelper modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") # usrloc modparam("usrloc", "db_mode", 2) # rr modparam("rr", "enable_full_lr", 1) # registrar modparam("registrar", "nat_flag", 6) # acc modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_fmt", "miocfst") modparam("acc", "failed_transactions", 1) # tm modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # main routing logic route { if (method == "BYE" || method == "CANCEL") { unforce_rtp_proxy(); setflag(1); # accounting setflag(2); # missed call }; # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message too big"); break; }; if (method!="REGISTER") { record_route (); }; if (loose_route()) { if (method == "BYE") { setflag(1); setflag(2); unforce_rtp_proxy(); }; route(1); break; }; if (uri == myself) { if (method=="INVITE") { fix_nated_sdp ("1"); }; if (method == "REGISTER") { if (search("^(Contact|m): .*(a)(200\.167\.20\.34|evoip03\.tcnet\.com\.br|sip\.e-voip\.com\.br)")) { log("LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed"); break; }; route(2); # WWW authentication if (!save("location")) { sl_reply_error(); }; break; }; if (method == "INVITE") { route(3); # PROXY authentication if (uri=~"^sip:0") { route(4); # Asterisk }; consume_credentials(); #lookup("location"); if (!lookup("location")) { if (!lookup("aliases")) { if (uri=~"^sip:[3789][0-9]") { prefix("035"); route(4); # Asterisk } else { sl_send_reply ("404", "Not Found"); break; }; }; }; setflag(1); setflag(2); route(1); break; }; }; route(1); } # Fix NATED client on reply route. onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; } # Normal relay route[1] { if (uri =~ "[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")) { sl_send_reply("479", "We don't forward to private IP addresses"); break; }; t_on_reply("1"); if (isflagset(6) || isflagset(5)) { log (1, "oi"); force_rtp_proxy (); }; if (!t_relay()) { sl_reply_error(); }; } # WWW authentication route[2] { if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; if (!check_to()) { sl_send_reply ("401", "Unauthorized"); break; }; if (isflagset(5)) { setflag(6); }; } # PROXY authentication route[3] { if (nat_uac_test("19")) { setflag(6); }; if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; } # Asterisk - E-Voip route[4] { if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); rewritehost("200.167.20.26"); avp_write("i:45", "inv_timeout"); } [View Less]

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[Serusers] avpops question
by Andy Pyles 12 Nov '05

12 Nov '05

hi I have an avpops question I have an avp object of boolean type called forwardcalls. My question is how do I check if a specific user has this enabled or not? I have the following logic: if (avp_db_load("$ruri/username", "s:callfwd")) { xlog("L_ERR", "call forward [%Tf] method <%rm> r-uri <%ru>\n"); if ( avp_check("forwarded_calls", "eq/0/gi" )) { log(1, "forward calls is disabled!\n\n"); break; } else { setflag(22); avp_pushto("$ruri", "s:callfwd"); route(6); … [View More]

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[Serusers] Foward Call to iConnectHere
by b in 12 Nov '05

12 Nov '05

I want to modify my ser.cfg so that if a user dials *5(then their iConnectHere username, password and the pstn phone number they want to call), the sip server will forward to information to iConnectHere, where the call is sent to the pstn phone number. Ex: If a user's iConnectHere username is 12345678 and their password is 1122 and they want to call the pstn number 9-999-999-9999, they will dial *512345678112299999999999 to call the desired pstn telephone number. How can I modify my ser.cfg to … [View More]

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[Serusers] SER as an ALG. (now w/version info)
by Kyle Bresin 11 Nov '05

11 Nov '05

One quick addition, I'm using the latest versions of SER and mediaproxy that I could find: ser 0.9.4 and mediaproxy 1.4.2. Original problem follows: I have been trying to configure SER w/MediaProxy module/server to be a drop in NAT solution in front of asterisk. With the twist that I want to have asterisk continue acting as a registrar rather than SER. Outbound calls from natted clients works perfectly currently. The problem I'm having is with SER passing the REGISTERS through to … [View More]asterisk. SER/MP isn't rewriting the address in the packets to go back through SER, but instead sets it to be the NAT'd IP of the client. So, when asterisk receives a call, and tries to send it to the external nat'd IP of the client, there is no open session from asterisk, so the firewall denies the request. What I need to happen, is for SER/MP to rewrite the packet with it's own IP, so that asterisk sends it back to SER, and SER forwards it back down to the client, from a source IP which has an open firewall session. Any help would be greatly appreciated, hopefully it's just a matter of me having an extraneous function call somewhere, or being ignorant of the right function to use. Or, in ascii diagram form: NAT_IP client --- EXTERNAL_FW_IP ----- SER_IP ----- ASTERISK SER is giving asterisk the EXTERNAL_FW_IP for Registers, and I want it to give Asterisk the SER IP. Something that would make SER rewrite *all* register packets to it's own IP would even work, since clients that don't need NAT won't be sent to SER in the first place. My ser.cfg and mediaproxy.ini follow. Lemme know if you need any more info from me. And extra thanks to anyone who has taken the time reading this, I truly appreciate it. kyle. # ### ser.cfg ### # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5070 #children=4 fifo="/tmp/ser_fifo" #fifo_db_url="mysql://ser:ser/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/dbtext.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" #loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" # !!! Media Proxy loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" #modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") #modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/ser/sip-asymmetric-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 20) # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (client_nat_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER fix_contact(); # Rewrite contact with source IP of signalling force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed # use_media_proxy(); }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # if (!method=="REGISTER") record_route(); record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; route(1); } route[1] { if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { use_media_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { use_media_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (client_nat_test("1")) { fix_contact(); use_media_proxy(); }; } ; ; Configuration file for MediaProxy ; [Dispatcher] start = yes socket = /var/run/proxydispatcher.sock ;group = ser defaultProxy = /var/run/mediaproxy.sock [MediaProxy] start = yes socket = /var/run/mediaproxy.sock ;group = ser ;listen = None ;allow = None ;proxyIP = 127.0.0.1 portRange = 35000:65000 ;TOS = 0xb8 ;idleTimeout = 60 ;holdTimeout = 3600 ;forceClose = 0 accounting = off [Accounting] user = dbuser password = dbpass host = dbhost database = radius table = radacct [View Less]

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[Serusers] SER as an ALG.
by Kyle Bresin 11 Nov '05

11 Nov '05

I have been trying to configure SER w/MediaProxy module/server to be a drop in NAT solution in front of asterisk. With the twist that I want to have asterisk continue acting as a registrar rather than SER. Outbound calls from natted clients works perfectly currently. The problem I'm having is with SER passing the REGISTERS through to asterisk. SER/MP isn't rewriting the address in the packets to go back through SER, but instead sets it to be the NAT'd IP of the client. So, when … [View More]asterisk receives a call, and tries to send it to the external nat'd IP of the client, there is no open session from asterisk, so the firewall denies the request. What I need to happen, is for SER/MP to rewrite the packet with it's own IP, so that asterisk sends it back to SER, and SER forwards it back down to the client, from a source IP which has an open firewall session. Any help would be greatly appreciated, hopefully it's just a matter of me having an extraneous function call somewhere, or being ignorant of the right function to use. Or, in ascii diagram form: NAT_IP client --- EXTERNAL_FW_IP ----- SER_IP ----- ASTERISK SER is giving asterisk the EXTERNAL_FW_IP for Registers, and I want it to give Asterisk the SER IP. Something that would make SER rewrite *all* register packets to it's own IP would even work, since clients that don't need NAT won't be sent to SER in the first place. My ser.cfg and mediaproxy.ini follow. Lemme know if you need any more info from me. And extra thanks to anyone who has taken the time reading this, I truly appreciate it. kyle. # ### ser.cfg ### # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5070 #children=4 fifo="/tmp/ser_fifo" #fifo_db_url="mysql://ser:ser/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/dbtext.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" #loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" # !!! Media Proxy loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" #modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") #modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/ser/sip-asymmetric-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 20) # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (client_nat_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER fix_contact(); # Rewrite contact with source IP of signalling force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed # use_media_proxy(); }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # if (!method=="REGISTER") record_route(); record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; route(1); } route[1] { if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { use_media_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { use_media_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (client_nat_test("1")) { fix_contact(); use_media_proxy(); }; } ; ; Configuration file for MediaProxy ; [Dispatcher] start = yes socket = /var/run/proxydispatcher.sock ;group = ser defaultProxy = /var/run/mediaproxy.sock [MediaProxy] start = yes socket = /var/run/mediaproxy.sock ;group = ser ;listen = None ;allow = None ;proxyIP = 127.0.0.1 portRange = 35000:65000 ;TOS = 0xb8 ;idleTimeout = 60 ;holdTimeout = 3600 ;forceClose = 0 accounting = off [Accounting] user = dbuser password = dbpass host = dbhost database = radius table = radacct [View Less]

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RE: [Serusers] Multiple backup routes
by Bjorn Asmul 11 Nov '05

11 Nov '05

The LCR module is your best bet. -- Bjorn -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Marcello Lupo Sent: Friday, November 11, 2005 6:45 AM To: serusers(a)lists.iptel.org Subject: [Serusers] Multiple backup routes Hi, if in a system i need to have more that one PSTN gateway to try for a destination in sequential order (note that gateways are dinamyc and may change call by call) and the order is important because them … [View More]

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Re: [Serusers] DID exchange or buy
by Rehan Ahmed 11 Nov '05

11 Nov '05

Available on Special Order, but not for exchange at this time. Rehan On 11/11/05, George Lambson <LambsonGE(a)mtc.byu.edu> wrote: > > Do you offer toll free #'s in Australia? > > George Lambson > Missionary Training Center > Provo, Utah > > >>> Rehan Ahmed <rehan33(a)gmail.com> 11/10/2005 3:11 PM >>> > We would be interested in numbers from ANYWHERE in the world, :) > How many can you give? > Minimum i would like to exchange is … [View More]

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[Serusers] Compiling ser and mysql module on RH ES em64t
by Thomas Nilsen 11 Nov '05

11 Nov '05

Hello everyone! Im changing platforms and I have problem compiling ser for my new arch. I now have em64t servers with rh es. Actually the problem is compiling ser with mysql module. - Mysql em64t rpm installed from mysql site. - ser 0.9.4 sourcetree. When i compile everything works ok. But making mysql module fails. It complains about some relocation problem between 32 and 64 bit libs. I cant understand this since mysql is em64 version and so is ser. Off course bin … [View More]

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[Users] OpenSer as a UA?
by Chris Hastie 11 Nov '05

11 Nov '05

Hello I'm struggling to understand how all the bits of a VOIP system fit together and whether OpenSer can act as a UA and authenticate to a remote proxy. What I want to achieve is this. Incoming SIP calls are first routed to a SIP phone on the network. If this is unavailable, the call is then redirected to another proxy of a voip service provider with whom I have an accout, whence it should be routed to my mobile phone. I guess I can rewrite the call URI to mymobilenumber(a)proxy.voipprovider.… [View More]

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[Users] which clients can i use for openser?
by messa＠innsof.com 11 Nov '05

11 Nov '05

Hi all I want to know which client i can use with openser and how i can configure it. Help me please! Regards! Serge MESSA OVONO

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[Users] client configuration
by messa＠innsof.com 11 Nov '05

11 Nov '05

Hi Bogdan Thanks for your reply. I know Xlite! But tell me how i can add users in openser and how i must configure my xlite client. I want to know if in the outbound proxy, i must put the ip adress of the external part of the media proxy ( proxy server in the openser documentation). best regards Serge

2 2

[Serusers] Multiple backup routes
by Marcello Lupo 11 Nov '05

11 Nov '05

Hi, if in a system i need to have more that one PSTN gateway to try for a destination in sequential order (note that gateways are dinamyc and may change call by call) and the order is important because them can be ordered for costs of the call from the less cost to the higher. There is a way for example to append all of them as branch and try them in sequence?? Otherwise i have to process them in a separate route block and have a temporary table in a database where take next gateways to … [View More]

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Re: [Users] How to add Route headers?
by Bogdan-Andrei Iancu 11 Nov '05

11 Nov '05

right...you need the new header on the top... I'm afraid there is no way to control the insertion point from the script (default is at the end)....... maybe having an extra parameter on the function to say top or bottom?.....not sure.... regards, bogdan Gustavo García Bernardo wrote: >Hi Bogdan, > >I had seen that function, but i think append_hf appends headers at the >bottom of the message, and I need to put those Route headers above any other >existing Route header. > … [View More]>Any other solution? > >Thanks once more. > >regards >G. > >-----Mensaje original----- >De: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Enviado el: jueves, 10 de noviembre de 2005 13:00 >Para: Gustavo García Bernardo >CC: users(a)openser.org >Asunto: Re: [Users] How to add Route headers? > >Hi Gustavo, > >if you have fixed content, it's even simpler. Use the textops module to >add the headers. See the append_hf() function: >http://www.openser.org/docs/modules/1.1.x/textops.html#AEN189 > >regards, >bogdan > >Gustavo García Bernardo wrote: > > > >>Hi Bogdan, thx for your response >> >>I need to put two fixed route headers, like these: >> >>Route: user(a)as.domain.com >>Route: user(a)cscf.domain.com >> >>Can I do that with AVPops module? I think is not suitable for this task. >> >>BR, >> >>G. >> >>-----Mensaje original----- >>De: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>Enviado el: jueves, 10 de noviembre de 2005 0:28 >>Para: Gustavo García Bernardo >>CC: users(a)openser.org >>Asunto: Re: [Users] How to add Route headers? >> >>Hi Gustavo, >> >>you can generate the body of the Route header(s) into AVP(s) be loading >>them from database of by building them via avpops functions (like >>avp_printf()). >>then just set the AVP(s) into Route header(s) via the avp_pushto() >> >> >function: > > >>http://www.voice-system.ro/docs/avpops/ar01s06.html#avp_pushto >> >>mostly depends of the content you want to push into the headers. >> >>regards, >>bogdan >> >>Gustavo García Bernardo wrote: >> >> >> >> >> >>>Hi, >>> >>>Is posible to add more route headers to a request before relaying it? >>> >>>I’m trying to implement the IMS mechanism before relaying a request to >>>an application server, which consists in adding two route headers. >>> >>>Thx in advance. >>> >>>G. >>> >>>------------------------------------------------------------------------ >>> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> >>> >> >> >> >> > > > > [View Less]

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RE: [Serusers] Test tool to generate 30 simultaneous SIP calls
by Rodríguez García, Marcos 11 Nov '05

11 Nov '05

Does this tool generate also RTP traffic in thoese simultaneous calls? Thanks, Marcos -----Mensaje original----- De: Atle Samuelsen [mailto:clona@cyberhouse.no] Enviado el: viernes, 11 de noviembre de 2005 11:11 Para: Rodríguez García, Marcos CC: serusers(a)lists.iptel.org Asunto: Re: [Serusers] Test tool to generate 30 simultaneous SIP calls signalling, you could use sipp.. /A * Rodr?guez Garc?a, Marcos <mrogarcia(a)indra.es> [051111 11:03]: > Do you know any test tool that I … [View More]could use to generate 30 simultaneous SIP calls from one PC? > > > > Thanks in advance, > > Marcos Rodr?guez > > ------------------------------------------------------------------------------------------------------------------- > Este correo electr?nico y, en su caso, cualquier fichero anexo al mismo, contiene informaci?n de car?cter confidencial exclusivamente dirigida a su destinatario o destinatarios. Queda prohibida su divulgaci?n, copia o distribuci?n a terceros sin la previa autorizaci?n escrita de Indra. En el caso de haber recibido este correo electr?nico por error, se ruega notificar inmediatamente esta circunstancia mediante reenv?o a la direcci?n electr?nica del remitente. POR FAVOR, ANTES DE IMPRIMIR ESTE CORREO ELECTR?NICO CONSIDERE SU APORTACI?N A LA CONSERVACI?N DEL MEDIO AMBIENTE POR LA REDUCCI?N DE CONSUMO DE PAPEL. > > The information in this e-mail and in any attachments is confidential and solely for the attention and use of the named addressee(s). You are hereby notified that any dissemination, distribution or copy of this communication is prohibited without the prior written consent of Indra. If you have received this communication in error, please, notify the sender by reply e-mail. PLEASE CONSIDER YOUR ENVIRONMENTAL RESPONSIBILITY BEFORE PRINTING THIS E-MAIL. > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers ------------------------------------------------------------------------------------------------------------------- Este correo electrónico y, en su caso, cualquier fichero anexo al mismo, contiene información de carácter confidencial exclusivamente dirigida a su destinatario o destinatarios. Queda prohibida su divulgación, copia o distribución a terceros sin la previa autorización escrita de Indra. En el caso de haber recibido este correo electrónico por error, se ruega notificar inmediatamente esta circunstancia mediante reenvío a la dirección electrónica del remitente. POR FAVOR, ANTES DE IMPRIMIR ESTE CORREO ELECTRÓNICO CONSIDERE SU APORTACIÓN A LA CONSERVACIÓN DEL MEDIO AMBIENTE POR LA REDUCCIÓN DE CONSUMO DE PAPEL. The information in this e-mail and in any attachments is confidential and solely for the attention and use of the named addressee(s). You are hereby notified that any dissemination, distribution or copy of this communication is prohibited without the prior written consent of Indra. If you have received this communication in error, please, notify the sender by reply e-mail. PLEASE CONSIDER YOUR ENVIRONMENTAL RESPONSIBILITY BEFORE PRINTING THIS E-MAIL. [View Less]

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[Users] Registrar-initiated Route changes on Invite?
by Joachim Fabini 11 Nov '05

11 Nov '05

Hi, We'd like to force the following SIP INVITE routing scenario: UA1->P1->P2->R->P2->P1->UA2. Unfortunately the Route is not known apriori and must be built dynamically. P1 knows the Route from P1 to R, R knows the Route from R to P1/UA2. The INVITE sent by UA1 is loosely routed to R based on the Route (Route: <sip:P2;lr>,<sip:R;lr>) that is added by P1 to UA1's INVITE. This works. The message arrives correctly to the registrar, Route header contains Route: … [View More]

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[Users] xlog and request mothod in responses
by Klaus Darilion 11 Nov '05

11 Nov '05

Hi! I use xlog to log certain responses. I want to log the request method of a response. Using $rm in the reply prints <null>. Using $cs prints only the CSeq number, but no the method. Is there a way to achieve this? regards klaus

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[Users] Help with NATHelper module and fix_nated_contact() function
by Anthony Chapellier 11 Nov '05

11 Nov '05

Hi, NatHelper documentation says fix_nated_contact is rewriting contact header with request source address:port but it seems the function doesn't work properly with my SER config... Indeed, when you analyze a SIP Request Packet you have the following : Message Header : Contact : <sip:352621@192.170.10.32:5060> #192.170.10.32:5060 is the private IP/Port and if this function was working properly, I understand you should get the following : Message Header : Contact: <sip:… [View More]

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[Users] How to change contact
by Taras Bendik 11 Nov '05

11 Nov '05

Hi I need something similar so fix_nated_contact but with parametars so I can set up my own values for host:port. Can I do this in OpenSer? Can avp_pushto do this? Regards, Taras

2 1

[Serusers] scope & database operations for avp_write
by Ramin Nikaeen 11 Nov '05

11 Nov '05

Valued Colleagues, Can anyone tell me what is the scope of avp_write?! Is it global / per-transaction / per-message?! I would like to have something similar to it with global scope so that I can set the string "callfwd" once by executing: avp_write("sip:211@10.1.10.22:5060", "s:callfwd"); and then keep using it for each SIP message processing using: avp_pushto("$ruri", "s:callfwd"); Also, do the avp_write and avp_pushto involve Database access operations?! i.e. Does … [View More]

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[Serusers] jabber and pa modules
by Matteo Piazza 11 Nov '05

11 Nov '05

Hi, I have this problem with the jabber and PA mudules, I use Jive messenger as Jabber server on the same machine Someone can me help? ========================= log debug ========================= 0(15774) parse_headers: flags=-1 0(15774) check_via_address(192.168.9.181, 192.168.9.181, 0) 0(15774) REGISTER received -> reply okay 0(15774) XJAB: Going OFFline in Jabber network!!! 0(15774) XJAB: go offline in Jabber network 0(15774) parse_headers: flags=12 0(15774) DEBUG: … [View More]add_param: tag=67b87d 0(15774) end of header reached, state=29 0(15774) XJAB:xj_wlist_check: entry does not exist for <sip:308@192.168.9.97> 0(15774) XJAB:xjab_manage_sipmsg: no open Jabber session for <sip:308@192.168.9.97>! 0(15774) parse_headers: flags=-1 =========================== ser.cfg ========================== modparam("jabber","db_url","mysql://root:****@localhost/sip_jab") modparam("jabber","jaddress","localhost") modparam("jabber","jport",5222) modparam("jabber","workers",2) modparam("jabber","max_jobs",10) modparam("jabber","cache_time",200) modparam("jabber","delay_time",60) modparam("jabber","jdomain","192.168.9.97") route{ #if ( !mf_process_maxfwd_header("10") ) #{ # sl_send_reply("483","To Many Hops"); # drop(); #}; # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { if (t_newtran()) { save("location"); log("REGISTER received -> reply okay\n"); }; if(search("registration")) { log("XJAB: Going ONline in Jabber network!!!\n"); if(jab_go_online()) { sl_send_reply("200", "Accepted"); } else { sl_send_reply("404","Not found"); }; } else { log("XJAB: Going OFFline in Jabber network!!!\n"); if(jab_go_offline()) { sl_send_reply("200", "Accepted"); } else { sl_send_reply("404","Not found"); }; }; break; }; if (method=="SUBSCRIBE") { if (t_newtran()) { handle_subscription("registrar"); }; break; }; if(!lookup("location")) { sl_send_reply("404","Not found"); break; }; ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it ======================================= [View Less]

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[Serusers] prob jabber xml version
by Matteo Piazza 11 Nov '05

11 Nov '05

I try to use jabber module, but the jabber server send me this string: <?xml version='1.0' encoding='UTF-8'?><stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:client" from="panoramix.create-net.it" id="2abf6ae" xml:lang="en"> The jabber module want a string as the follow: <?xml version='1.0'?>.......... Any idea? -- ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - … [View More]

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[Serusers] Accounting Problem
by Rommel Catabian 11 Nov '05

11 Nov '05

Im very new to SER. My colleague who just left the company already set up the system. We can now call sip-to-sip and sip-to-pstn. However, our problem is that we dont have accounting record for the pstn calls. For sip-to-sip, accounting is OK. >From the radius log of the pstn calls, I can see only STOP request but no START request. My question: What triggers the SER to send START accounting request to the RADIUS? All traffic to pstn goes to a gateway operated by a third party service … [View More]

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Re: [Users] enable NAT OPTION ping
by Bogdan-Andrei Iancu 11 Nov '05

11 Nov '05

Hi Raymond, to you experience the same behaviour if you disable SIP ping (by not setting flag 5) and falling back to UDP ping? regards, bogdan Raymond Chen wrote: >Hi Bogdan, > >No, registration happens every minute. > >Thanks > >Ray > > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: 2005年11月10日 23:01 >To: Raymond Chen >Cc: users(a)openser.org >Subject: Re: [Users] enable NAT OPTION ping > … [View More]>Hi Raymond, > >maybe the registration expires - check if when the PING stops, the >registered contact is still valid (not expired). > >regards, >bogdan > >Raymond Chen wrote: > > > >>Dear all, >> >>I want to force openser to ping all registered users with OPTIONS >>message, my configuration as follow, flag in location table is 512 for >>sip_nat_flag, and 513 for NAT. I can see the OPTIONS message send to >>the users and got replied for a few minutes, after that, it stop >>sending. Is there something I’ve done wrong? >> >>Thanks >> >>Ray >> >>modparam("registrar", "nat_flag", 6) >> >>modparam("registrar", "sip_natping_flag", 5) >> >>modparam("nathelper", "ping_nated_only", 0) >> >>modparam("nathelper", "natping_interval", 30) >> >>modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") >> >>modparam("nathelper", "sipping_method", "OPTIONS") >> >>modparam("nathelper", "sipping_from", "sip:keepitopen@xxxxxx.net") >> >>if (nat_uac_test("31")) { >> >>if (method == "REGISTER" || !search("^Record-Route:")) { >> >>fix_nated_contact(); # Rewrite contact with source IP of signalling >> >>if (method=="REGISTER") { >> >>fix_nated_register(); >> >>} else if (method == "INVITE") { >> >>fix_nated_sdp("1"); # Add direction=active to SDP >> >>}; >> >>force_rport(); # Add rport parameter to topmost Via >> >>append_hf("P-hint: Proxied\r\n"); >> >>setflag(6); # Mark as NATed >> >>}; >> >>}; >> >>if (uri==myself) { >> >>if (method=="REGISTER") { >> >>if (!www_authorize("", "subscriber")) { >> >>www_challenge("", "1"); >> >>exit(); >> >>}; >> >>Setflag(5); >> >>save("location"); >> >>return; >> >>}; >> >>alias_db_lookup("aliases"); >> >>if (lookup("location")) { >> >>append_hf("P-hint: usrloc applied\r\n"); >> >>route(1); >> >>return; >> >>}; >> >>}; >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> > > > > > > > [View Less]

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Re: [Serusers] Call forwarding problem
by rpagquil＠philonline.com 11 Nov '05

11 Nov '05

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[Serusers] Test tool to generate 30 simultaneous SIP calls
by Rodríguez García, Marcos 11 Nov '05

11 Nov '05

Do you know any test tool that I could use to generate 30 simultaneous SIP calls from one PC? Thanks in advance, Marcos Rodríguez ------------------------------------------------------------------------------------------------------------------- Este correo electrónico y, en su caso, cualquier fichero anexo al mismo, contiene información de carácter confidencial exclusivamente dirigida a su destinatario o destinatarios. Queda prohibida su divulgación, copia o distribución a terceros sin … [View More]

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[Serusers] Call forwarding problem
by rpagquil＠philonline.com 11 Nov '05

11 Nov '05

1 0

[Users] UAC problem. UAC don't send final ACK if UAS send OK
by jltorre＠yahoo.com 11 Nov '05

11 Nov '05

uac module configuration is by default without modparam, and the problem is when the callee respond the call, the caller hear ring sound and callee voice, both of then, and UAC ( caller) don't do final ACK if reveive OK from UAS, because OK trace from UAS may be not correct. We only use uac_replace_from("","usuario(a)domain.com") for INVITE,ACK,CANCEL or BYE method in route section.We have use openser release 1.0.0 Have you a solution or is it a bug ? Thank in advance Jose … [View More]

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Re: [Serusers] Calls from UAs behind NAT getting a 503 error...
by Klaus Darilion 11 Nov '05

11 Nov '05

Kyriakos wrote: > So Klaus, basically what i need is an extra handler for ACK? > > route[6] { > > # ------------------------------------------------------------------------ > # ACK Handler > # ------------------------------------------------------------------------ > if (method=="ACK") { > t_relay(); > return; > }; That should work > Does this affect the call flow? I mean Ive read that responses to ACK > requests are not that important. per standard: … [View More]

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[Serusers] Expire for User agent question
by rpagquil＠philonline.com 11 Nov '05

11 Nov '05

2 1

[Serusers] SER and LDAP
by Felipe Louback 11 Nov '05

11 Nov '05

Is there any way to authenticate SER users in LDAP without the use of RADIUS? If there is, could anyone point out where I could find information about it? I've searched some documents but found nothing about authentication with LDAP without using RADIUS.... Regards, Felipe -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "Humanly speaking it is impossible; but with God anything is … [View More]

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[Serusers] question about relay
by yserge 11 Nov '05

11 Nov '05

Hi, I am using SER as a an entry point for the requests from the network and every request from the network is being relayed to another ( clusterized ) SIP server ( not a SER ) with multiple IP addresses in DNS. The server is running on 4 machines, any number of which can be down at any moment of time. Suppose, the DNS name is "sip.myserver.com". The incoming requests from the network have SER as host:port of RURI.. I was trying to relay the request using t_relay_to_udp("sip.myserver.… [View More]

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[Serusers] mediaproxy for incoming
by George Lambson 11 Nov '05

11 Nov '05

Is anyone using mediaproxy for INCOMING calls? What I mean is: is SER capable of accepting INVITE messages from an unregistered UA that is behind a NAT Firewall and connecting them with a registered UA? I would like to be able to receive incoming calls for my domain from anyone on the internet and connect them to the proper user. Please tell me anyone if you are successfully doing this. Thanks, George

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[Serusers] can't connect to RTP proxy
by lo ™ 11 Nov '05

11 Nov '05

hi all, i'm quite new to sip and i'm trying to setup ser with nat support. i read about stun but to rewrite contact with received and rport tags i guess i need nathelper module that, in turn, needs rtpproxy, doesn't it? i got ser-0.9.0, i downloaded rtpproxy from cvs.ser.berlios.de<http://cvs.ser.berlios.de>and i use the same nathelper module configuation and nat routing you can find here: http://www.informatik.uni-bremen.de/~prelle/terena/cookbook/main/ch04s07.ht… unlucky ser.log … [View More]

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RE: [Serusers] Accounting Problem
by Rommel Catabian 11 Nov '05

11 Nov '05

Thanks for the immediate reply. We have this on the file ser.cfg: modparam("acc", "radius_flag", 2) setflag(2); ---->>> could this be on the wrong location? To route traffic to PSTN, we have this on the ser config: ----- if (uri=~"^sip:90087(.+)@.*") { rewritehostport("XXX.XXX.XXX.XXX:5060"); forward(uri:host, uri:port); exec_msg('(/usr/local/etc/ser/logs.csh)'); break; ------ Here is our ser.cfg file: --------… [View More]------------------ debug=3 fork=yes log_stderror=yes check_via=no dns=no rev_dns=no port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://XXXX:XXXX@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/exec.so" modparam("usrloc", "db_mode", 2) # 0 Disable Authentication ; 2 Enable Authentication modparam("usrloc", "db_url", "mysql://XXXX:XXXX@localhost/ser") modparam("auth_db", "db_url", "mysql://XXXX:XXXX@localhost/ser") modparam("rr", "enable_full_lr", 1) modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("acc", "radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 2) modparam("acc", "radius_missed_flag", 3) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 0) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (nat_uac_test("3")) { if (method == "REGISTER" || ! search("^Record-Route:")) { fix_nated_contact(); if (method == "INVITE") { fix_nated_sdp("1"); }; force_rport(); setflag(6); }; }; lookup("aliases"); if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; setflag(2); setflag(5); if (method=="INVITE") { exec_msg('(/usr/local/etc/ser/record.pl)'); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); break; }; if (uri=~"^sip:90087(.+)@.*") { rewritehostport("202.64.248.209:5060"); forward(uri:host, uri:port); exec_msg('(/usr/local/etc/ser/logs.csh)'); break; }; #--------- IVR Start ----------- if (uri=~"^sip:222@.*") { if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; if (method=="ACK"){ t_relay(); break; }; t_reply("100","Trying -- just wait a minute !"); if(!t_write_unix("/tmp/am_sock","ivr")) { t_reply("500","error contacting sems"); }; break; }; #--------- IVR End ----------- if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("sip.easy.net.ph","subscriber")) { www_challenge("sip.easy.net.ph", "0"); break; }; save("location"); break; }; if (method=="INVITE") { exec_msg('(/usr/local/etc/ser/record.pl)'); }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { if (method=="INVITE") { exec_msg('/usr/local/etc/ser/record.pl'); }; if (method=="CANCEL") { exec_msg('/usr/local/etc/ser/record.pl'); }; if (method=="BYE") { exec_msg('/usr/local/etc/ser/record.pl'); }; if (method=="ACK") { exec_msg('/usr/local/etc/ser/record.pl'); }; setflag(2); if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(6)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); } else if (nat_uac_test("1")) { fix_nated_contact(); }; } --- Matt Simpson <msimpson(a)commpartners.us> wrote: > > >My question: > >What triggers the SER to send START accounting > request > >to the RADIUS? > > > >Thanks, > > > >Rommel > > Rommel, there should be a statement at the top like: > modparam("acc","radius_flag",1) > > The "1" is the flag to account for a call in radius > > Then you would use: > setflag(1); > > to account for the call. It is hard to tell you > where to put the > statement without seeing your conf but that should > help. > > __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com [View Less]

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[Serusers] SER between softswitches
by Matt Simpson 11 Nov '05

11 Nov '05

I'm using SER to shuttle calls between softswitches. I have "softswitch A" sending calls to the SER proxy, which then does an enum lookup to make a routing decision and then passes the call on to the appropriate softswitch. I have been having some strange problems with the setup and some questions about my config. I sometimes get: Nov 10 21:14:01 cpser ser[22087]: 483 Too many Hops Nov 10 21:14:01 cpser ser[22087]: Warning: sl_send_reply: I won't send a reply for ACK!! This mostly … [View More]happens after I do a serctl restart to reload my config. I assume that is because ACKs are coming in that the stateful transaction processor doesn't know where to forward them to since I restarted? There's no other way to reload a config other than a full restart, correct? Also I see a lot of "408 Request Timeout" from far end gateways. Is there something wrong with my config? Here is my config: modparam("acc","radius_flag",1) modparam("acc","failed_transactions",0) modparam("tm", "fr_inv_timer", 5 ) route{ if ( !mf_process_maxfwd_header("10") ) { xlog("L_ERR","483 Too many Hops"); sl_send_reply("483","Too Many Hops"); drop(); }; if (msg:len >= 2048 ) { xlog("L_ERR","513 Message too Big"); sl_send_reply("513", "Message too big"); break; }; setflag(1); if (loose_route()) { t_relay(); break; }; if (method =="BYE") { if ( !t_relay()) { sl_reply_error(); } break; }; if (method =="CANCEL") { if ( !t_relay()) { sl_reply_error(); } break; }; if (method =="ACK") { if ( !t_relay()) { sl_reply_error(); } break; }; if (method =="INVITE") { record_route(); if (uri=~"^sip:[2-9]+@*") { prefix("+"); if ( !enum_query("e164.internal.net")) { strip(1); prefix("1"); rewritehost("192.168.10.5"); xlog("L_ERR","NANPA number to %tu from %fu carrier %ru\n"); t_on_failure("1"); t_relay(); break; } xlog("L_ERR","NANPA-ENUM number to %tu from %fu carrier %ru\n"); t_on_failure("1"); t_relay(); }; # end if block method = invite } } failure_route[1] { if(t_check_status("501|502|503|504|505|513")) { setflag(1); revert_uri(); rewritehost("192.168.50.3"); append_branch(); xlog("L_ERR","FAIL-5XX number to %tu from %fu carrier %ru\n"); t_relay(); } } [View Less]

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[Serusers] ACK Message problem
by Eddie Chan 11 Nov '05

11 Nov '05

Hi, I was making test call from a UA1(10.0.2.12) via SER (192.168.168.87) to another UA2 (192.168.168.33) where UA1 is a program written by us and UA2 is a softphone. When I cancel the call, I am having problem sending an Ack to SER. In the trace attached, when UA1 send a Cancel, it replies with 200 canceling in response to this Cancel, and 487 in response to the original INVITE message. Then UA1 send back an ACK in response to 487 but SER keep resending 487 message. It seems like … [View More]

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[Serusers] rewrite contact header of INVITE message
by George Lambson 10 Nov '05

10 Nov '05

Hi all, How do I have SER rewrite the contact header of an INVITE message? If I do that, will mediaproxy work for the session? Thanks in advance, George

1 0

[Serusers] ACK Message problem
by Eddie Chan 10 Nov '05

10 Nov '05

Hi, I was making test call from a UA1(10.0.2.12) via SER (192.168.168.87) to another UA2 (192.168.168.33) where UA1 is a program written by us and UA2 is a softphone. When I cancel the call, I am having problem sending an Ack to SER. In the trace attached, when UA1 send a Cancel, it replies with 200 canceling in response to this Cancel, and 487 in response to the original INVITE message. Then UA1 send back an ACK in response to 487 but SER keep resending 487 message. It seems like … [View More]

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Re: [Serusers] DID exchange or buy
by Rehan Ahmed 10 Nov '05

10 Nov '05

We would be interested in numbers from ANYWHERE in the world, :) How many can you give? Minimum i would like to exchange is 50 number at a time. Rehan On 11/11/05, Victor Stanescu <victor.stanescu(a)gtstelecom.ro> wrote: > > Dear Rehan, > > Are you interested in Romanian numbers also? > > Rehan Ahmed wrote: > > Dear Leon, > > > > We would like to exchange numbers from you, which numbers do you have > > avaialble ? > > > > We can … [View More]

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[Serusers] Mediaproxy module and SIP over TCP
by HERNAN HEVIA 10 Nov '05

10 Nov '05

Hello, I've found the following issue: When my gateway (it has a public IP address) sends an Invite to SER over a TCP connection, the function client_nat_test(2) always returns true, even though the VIA HF ahs the gateway public IP address. Any idea of why it is doing so? Does it have to do with port comparisson? I tried to fix it using sip_asymmetrics file, but it didn't work. Any help on this file syntax? Thank you very much in advance Hernán

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[Users] Thanks for the help
by Krishna Chava 10 Nov '05

10 Nov '05

Hi everyone, Thanks for your comments.. i had a working openser now thanks and am having registrations working.. Will post my problems if i had any while going through... thanks krishna

1 0

[Serusers] SER + SipXchange
by Kathleen McCallum 10 Nov '05

10 Nov '05

Hello everyone, I must use SER as a proxy server to send every request to SipXchange (except the MESSAGE request that will be handled by SER) Can I get some help on this?, Is there any example I could use as a reference?. The user must be registered in SipXchange (in SER it would be optional) If the request is "MESSAGE" then SER will handle the request. This is what I have done if (!method=="MESSAGE") { xlog("L_INFO","Foward process\n"); forward(SipXchangeServer,… [View More]

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[Serusers] replace function
by Xavier TRENTIN 10 Nov '05

10 Nov '05

Hi, I use ser 0.10.99+cvs20050428. When I use replace("foo","bar") in ser.cfg, is it supposed to replace all occurences of "foo" in the headers/body of the SIP message? It seems that it only replace the first occurence. Moreover, when I write : replace("foo","bar"); replace("foo","bar"); in my ser.cfg, it replace "foo" with "barbar". B.R. Xavier.

1 0

[Users] roadmap to v1.1.0
by Daniel-Constantin Mierla 10 Nov '05

10 Nov '05

Hi everybody, we are trying to collect the items to be included in the roadmap to the next release. I have written a draft with the ideas discussed on the mailing list, the items which were not accomplished in v1.0.0 and a few more. There is no weight attached, and the presence in the roadmap does not imply that the feature will be for sure implemented until the next release. The roadmap should guide the development of the project. Please send your suggestions, if possible with some … [View More]description. As a general rule, the features to be implemented must follow the standard, if they are related directly to SIP, or they must have a clear and common usage in real cases. Cheers, Daniel Draft of the roadmap to OpenSERv1.1.0 ------------------------------------- OpenSER core: - generic communication interface which must offer an abstract layer between core/modules and transports (e.g., fifo, unix sockets) - move the code of the implemented trasports as module - NAPTR lookup - TLS multi-domain support - TLS configuration values to be set via a module, to keep core less exposed to often changes - better error reporting and handling - pseudo-variables to be introduced in different modules - security checks of destination addess (white/black lists) - memory defragmentation - tcp enhancements - OpenSER command line interface (terminal) OpenSER modules: [acc] - possibility to disable values from db_fmt [avpops] - more coherent format of the parameters (=>to be used the one from pseudo-variables) - ability to take the name of the AVP to be loaded from the value of another AVP - global avps - avps to be stored during run time, shared between processes - local avps - avps to be stored locally, specific per script, not per transaction [cpl-c] - possibility to configure table and columns' names - import registrar paramters insted of redefine [dbtext] - replace support - reload support [dispatcher] - serial forking when selected destination fails - database support [enum] [-] - parallel/serial forking based on order and preference fields [postgres] - connection pool - shift to memory manager used by openser [tm] - unification of t_relay_to_*() in a form of t_relay_to("proto:host:port") [uac] - qop authentication support - proper CSeq value after authentication challenge [usrloc] - better handling of the natted contacts, to uniquely identify a contact - cacheless usrloc - path support for registrations - sip instance support - possibility to attach to a contact a set of values (similar to log_extra in acc) [View Less]

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[Users] FQDN in record route
by ranveer kunal 10 Nov '05

10 Nov '05

Hi folks, I want my record route to be FQDN(domain name) instead of ip address, so i tried using record_route_preset("mysip.test"). now my problem is when i do this it also appends my name rkunal(a)mysip.test to the record route field. Is there any way of configuring openser to send FQDN in record route instead of ip address and I don't want username there. Regards, Ranveer.

2 2

[Serusers] Rewriting SDP Infos for NAT Traversal
by Anthony Chapellier 10 Nov '05

10 Nov '05

Hi, When an INVITE request is incoming, I need to rewrite VIA Header Field, Contact Header Field, (c=) field in SDP message core and (o=) field in SDP message core too with IP public adress of UAC. Those are needed by my SIP Server before SER Proxy forward the request to him. I got some success with rewriting Contact Header File by using fix_nated_contact() function and (c=) field with fix_nated_sdp("1") function. But I don't succeed at all in rewriting (o=) field and partially for VIA … [View More]

2 3

Re: [Users] enable NAT OPTION ping
by Bogdan-Andrei Iancu 10 Nov '05

10 Nov '05

Hi Raymond, maybe the registration expires - check if when the PING stops, the registered contact is still valid (not expired). regards, bogdan Raymond Chen wrote: > Dear all, > > I want to force openser to ping all registered users with OPTIONS > message, my configuration as follow, flag in location table is 512 for > sip_nat_flag, and 513 for NAT. I can see the OPTIONS message send to > the users and got replied for a few minutes, after that, it stop > sending. … [View More]

2 1

Re: [Users] Re: [Devel] roadmap to v1.1.0
by Bogdan-Andrei Iancu 10 Nov '05

10 Nov '05

HI Raymond, I think was already mentioned in usrloc section as "cacheless usrloc" - integrating this into usrloc will be a more elegant way to do it than having another module. regards, bogdan Raymond Chen wrote: >How about usrloc-cl? > >Ray > >-----Original Message----- >From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf >Of Klaus Darilion >Sent: 2005年11月7日 18:22 >To: daniel(a)voice-system.ro >Cc: devel(a)openser.org; users(a)… [View More]

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RE: [Users] Routing to NATed gateways
by Leonid Fainshtein 10 Nov '05

10 Nov '05

Great! Bogdan, you are genius! I'll try it. Thank you very much! Leonid -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Thursday, November 10, 2005 17:05 To: Leonid Fainshtein Cc: users(a)openser.org Subject: Re: [Users] Routing to NATed gateways if you want to try the next gateway if lookup() fails, you may try to simulate loops by recursive call of the same route block. regards, bogdan Leonid Fainshtein wrote: >Hi Bogdan, >Yes, it does. … [View More]Please pay attention that the called number must be >restored in branch_route[]. >Also, I still don't know what to do if lookup() fails (the gateway is >temporary down and therefore is not registered...) Any idea? Is it >planned to add loops possibility to script language? > >Best regards, >Leonid > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Thursday, November 10, 2005 15:36 >To: Leonid Fainshtein >Cc: users(a)openser.org >Subject: Re: [Users] Routing to NATed gateways > >Hi Leonid, > >does the logic solved your problem? on a first view, the script look ok. >regarding the failure of lookup(); you can stick to sending an error >reply or you can try the next gateway provided by LCR. > >regards, >bogdan > >Leonid Fainshtein wrote: > > > >>In other words, the simplified configuration may be as the following: >> >> # Save original user name (the called phone number). >> avp_printf("$orig_called_num", "$rU"); >> if (load_gws()) >> { >> if (next_gw()) >> { >> if (lookup("location")) >> { >> avp_pushto("$ruri/username", "$orig_called_num"); >> t_on_failure("2"); >> route(x); # NAT and other usual staff there >> exit; >> } >> else >> { >> xlog("L_INFO", "I have to think what to do in this >>situation.\n"); >> sl_send_reply("404", "User Not Found"); >> } >> } >> } >> >>failure_route[2] >>{ >> if (method=="INVITE" && t_check_status("408|500|503")) >> { >> if (!next_gw()) >> { >> t_reply("503", "Service not available, no more gateways"); >> } >> else >> { >> if (!lookup("location")) >> { >> xlog("L_INFO", "I have to think what to do in this >>situation.\n"); >> t_reply("503", "Service not available, no more >>gateways"); >> exit; >> }; >> t_on_failure("2"); >> t_on_branch("2"); >> t_relay(); >> } >> } >>} >> >>branch_route[2] >>{ >> avp_pushto("$ruri/username", "$orig_called_num"); >>} >> >>Am I correct? Actually I don't know what to do if one of the gateways >>is off-line (lookup() failed). >> >>Best regards, >>Leonid Fainshtein >>-----Original Message----- >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>Sent: Tuesday, November 08, 2005 5:30 PM >>To: Leonid Fainshtein >>Cc: users(a)openser.org >>Subject: Re: [Users] Routing to NATed gateways >> >>Hi Leonid, >> >>the scenario you are probing might be already doable. Set in the LCR >>table some IP addresses which will be used as aliases: private >>addresses in 10.10.x.x class for example. Then register this IPs in >>usrloc pointing to the real GWs address. >>So you can do lcr (next_gw) and then lookup. >> >>for this to work, lcr must operate on RURI and not on DST_URI (not >>sure >> >> > > > >>how exactly is working). >> >>regards, >>bogdan >> >>Leonid Fainshtein wrote: >> >> >> >> >> >>>Hi, >>>I'd like to use OpenSER for routing calls to NATed gateways. I also >>>want to support the LCR feature. Unfortunately, the current LCR >>>module >>> >>> > > > >>>doesn't support NAT. I mean that the gateway IP addresses must be >>>defined explicitly in the "gtw" table. Now I'm looking for a way to >>>force OpenSer to use information from the "location" table. >>>I want to write a module that will have the similar functionality >>>like >>> >>> > > > >>>the current LCR module has but resolving procedure will be more >>>complicated (not based on prefix and From only). The module should >>>build avp list where each gateway will have symbolic user name. For >>>example, g1, g2 etc. The gateways will also be registered on the >>>proxy >>> >>> > > > >>>with those names. Then I think it will be possible to call >>>lookup("location") after successful call the next_gtw() from the >>> >>> >>> >>> >>script. >> >> >> >> >>>Is it feasible? Is there another way to solve my problem? >>> >>>Thank you in advance, >>>Leonid Fainshtein >>> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> >>> >>> >> >> >> >> > > > > [View Less]

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RE: [Users] Routing to NATed gateways
by Leonid Fainshtein 10 Nov '05

10 Nov '05

Hi Bogdan, Yes, it does. Please pay attention that the called number must be restored in branch_route[]. Also, I still don't know what to do if lookup() fails (the gateway is temporary down and therefore is not registered...) Any idea? Is it planned to add loops possibility to script language? Best regards, Leonid -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Thursday, November 10, 2005 15:36 To: Leonid Fainshtein Cc: users(a)openser.org Subject:… [View More] Re: [Users] Routing to NATed gateways Hi Leonid, does the logic solved your problem? on a first view, the script look ok. regarding the failure of lookup(); you can stick to sending an error reply or you can try the next gateway provided by LCR. regards, bogdan Leonid Fainshtein wrote: >In other words, the simplified configuration may be as the following: > > # Save original user name (the called phone number). > avp_printf("$orig_called_num", "$rU"); > if (load_gws()) > { > if (next_gw()) > { > if (lookup("location")) > { > avp_pushto("$ruri/username", "$orig_called_num"); > t_on_failure("2"); > route(x); # NAT and other usual staff there > exit; > } > else > { > xlog("L_INFO", "I have to think what to do in this >situation.\n"); > sl_send_reply("404", "User Not Found"); > } > } > } > >failure_route[2] >{ > if (method=="INVITE" && t_check_status("408|500|503")) > { > if (!next_gw()) > { > t_reply("503", "Service not available, no more gateways"); > } > else > { > if (!lookup("location")) > { > xlog("L_INFO", "I have to think what to do in this >situation.\n"); > t_reply("503", "Service not available, no more >gateways"); > exit; > }; > t_on_failure("2"); > t_on_branch("2"); > t_relay(); > } > } >} > >branch_route[2] >{ > avp_pushto("$ruri/username", "$orig_called_num"); >} > >Am I correct? Actually I don't know what to do if one of the gateways >is off-line (lookup() failed). > >Best regards, >Leonid Fainshtein >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Tuesday, November 08, 2005 5:30 PM >To: Leonid Fainshtein >Cc: users(a)openser.org >Subject: Re: [Users] Routing to NATed gateways > >Hi Leonid, > >the scenario you are probing might be already doable. Set in the LCR >table some IP addresses which will be used as aliases: private >addresses in 10.10.x.x class for example. Then register this IPs in >usrloc pointing to the real GWs address. >So you can do lcr (next_gw) and then lookup. > >for this to work, lcr must operate on RURI and not on DST_URI (not sure >how exactly is working). > >regards, >bogdan > >Leonid Fainshtein wrote: > > > >>Hi, >>I'd like to use OpenSER for routing calls to NATed gateways. I also >>want to support the LCR feature. Unfortunately, the current LCR module >>doesn't support NAT. I mean that the gateway IP addresses must be >>defined explicitly in the "gtw" table. Now I'm looking for a way to >>force OpenSer to use information from the "location" table. >>I want to write a module that will have the similar functionality like >>the current LCR module has but resolving procedure will be more >>complicated (not based on prefix and From only). The module should >>build avp list where each gateway will have symbolic user name. For >>example, g1, g2 etc. The gateways will also be registered on the proxy >>with those names. Then I think it will be possible to call >>lookup("location") after successful call the next_gtw() from the >> >> >script. > > >>Is it feasible? Is there another way to solve my problem? >> >>Thank you in advance, >>Leonid Fainshtein >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> > > > > [View Less]

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[Serusers] ser authenticating with asterisk before relaying through
by Arne Van Theemsche 10 Nov '05

10 Nov '05

Hi When my SER passes through a call to asterisk, this is done without authenticating to asterisk. I use the hostname= and insecure=very method. Can't I force ser and asterisk to use authentication? thx Arne

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[Serusers] Re: [Serweb-users] whats this "DB Error:no such table" error???
by Karel Kozlik 10 Nov '05

10 Nov '05

hi, > according to the "usr_preferences.sql" > (serweb/scripts/sql/usr_preferences)the attribute name and types are all > dumped into > a table called "preference_types" which is not there in the mysql DB. > this isnt a default column to be included in the tables. > ny ideas to create this table? yes ser_mysql.sh reinstall Karel > > thank you > > > ------------------------------------------------------------------------ > *From:* Karel Kozlik [… [View More]

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[Users] Please Help Me
by Ferianto monster 10 Nov '05

10 Nov '05

I am a college student in Polithecnic Caltex of Riau in Indonesia. I am in the third years now. In my last project, I want to build the VOIP with TLS system at my college because i think it is interesting. I have studied about it since one month , but i get many problems to undestand about the system. Here is some obstacles questions about the system that i have met: 1. How can the attacker eavesdrop the communication in VoIP (The way)? 2. How can TLS prevent the attacker to eavesdrop the … [View More]

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RE: [Users] Routing to NATed gateways
by Leonid Fainshtein 10 Nov '05

10 Nov '05

In other words, the simplified configuration may be as the following: # Save original user name (the called phone number). avp_printf("$orig_called_num", "$rU"); if (load_gws()) { if (next_gw()) { if (lookup("location")) { avp_pushto("$ruri/username", "$orig_called_num"); t_on_failure("2"); route(x); # NAT and other usual staff there exit; } else … [View More] { xlog("L_INFO", "I have to think what to do in this situation.\n"); sl_send_reply("404", "User Not Found"); } } } failure_route[2] { if (method=="INVITE" && t_check_status("408|500|503")) { if (!next_gw()) { t_reply("503", "Service not available, no more gateways"); } else { if (!lookup("location")) { xlog("L_INFO", "I have to think what to do in this situation.\n"); t_reply("503", "Service not available, no more gateways"); exit; }; t_on_failure("2"); t_on_branch("2"); t_relay(); } } } branch_route[2] { avp_pushto("$ruri/username", "$orig_called_num"); } Am I correct? Actually I don't know what to do if one of the gateways is off-line (lookup() failed). Best regards, Leonid Fainshtein -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Tuesday, November 08, 2005 5:30 PM To: Leonid Fainshtein Cc: users(a)openser.org Subject: Re: [Users] Routing to NATed gateways Hi Leonid, the scenario you are probing might be already doable. Set in the LCR table some IP addresses which will be used as aliases: private addresses in 10.10.x.x class for example. Then register this IPs in usrloc pointing to the real GWs address. So you can do lcr (next_gw) and then lookup. for this to work, lcr must operate on RURI and not on DST_URI (not sure how exactly is working). regards, bogdan Leonid Fainshtein wrote: >Hi, >I'd like to use OpenSER for routing calls to NATed gateways. I also >want to support the LCR feature. Unfortunately, the current LCR module >doesn't support NAT. I mean that the gateway IP addresses must be >defined explicitly in the "gtw" table. Now I'm looking for a way to >force OpenSer to use information from the "location" table. >I want to write a module that will have the similar functionality like >the current LCR module has but resolving procedure will be more >complicated (not based on prefix and From only). The module should >build avp list where each gateway will have symbolic user name. For >example, g1, g2 etc. The gateways will also be registered on the proxy >with those names. Then I think it will be possible to call >lookup("location") after successful call the next_gtw() from the script. >Is it feasible? Is there another way to solve my problem? > >Thank you in advance, >Leonid Fainshtein > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > > [View Less]

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[Serusers] Important please note: unable to load mysql.so and SER Version 9.3.0: MySQL 4.1.15.
by Dhiman D Chowdhury 10 Nov '05

10 Nov '05

Dear friends, If you are new to SER like I am, please note the followings: Setup: Red Hat Linux 9.0 or Red Hat Enterprise ES 3 and SER version 0.9.3. 1. When you try to load mysql.so module for your ser, you get an error. bad config. if you do ser -D -E . The result shows that mysql.so was unable to load due to libmysqlclient.so.12 was not found. Please note that the above mentioned Red Hat Linux includes a MySQL server version 3.23 which does not include the above file. If you do prefer to … [View More]

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Re: [Serusers] FW: Call logs
by Steve Blair 10 Nov '05

10 Nov '05

Colin: Ahh, I've only done work with accounting on 0.9.x systems. This might explain why my instructions didn't help. I've copied the list on this reply in the hope that someone may know the answer since I will be out of the office all day today. I'll check back with you tomorrow. -Steve Colin Jordan wrote: >Hi Steve > >Thanks for this. The SER version is 8.1. Nothing appear in the database. > >Thanks for helping, > >Colin >

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[Users] How to change Contact
by Taras Bendik 10 Nov '05

10 Nov '05

Hi I need something similar so fix_nated_contact but with parametars so I can set up my own values for host:port. Can I do this in OpenSer? Can avp_pushto do this? Regards, Taras

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[Users] Compile openser with tls error ?
by Nguyen Duc Phi 10 Nov '05

10 Nov '05

Hi all, I am compile openser-1.0.0-tls on Red Hat Linux 3.2.3-52 openssl-0.9.7a-33.17 openssl-devel-0.9.7a-33.17 It's show error : gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"openser"' -DVERSION='"1.0.0-tls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -… [View More]DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DUSE_TLS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/ssl/include -c main.c -o main.o In file included from /usr/include/openssl/ssl.h:179, from tls/tls_init.h:27, from main.c:112: /usr/include/openssl/kssl.h:72:18: krb5.h: No such file or directory In file included from /usr/include/openssl/ssl.h:179, from tls/tls_init.h:27, from main.c:112: /usr/include/openssl/kssl.h:134: syntax error before "krb5_enctype" /usr/include/openssl/kssl.h:136: syntax error before '*' token /usr/include/openssl/kssl.h:137: syntax error before '}' token /usr/include/openssl/kssl.h:149: syntax error before "kssl_ctx_setstring" /usr/include/openssl/kssl.h:149: syntax error before '*' token /usr/include/openssl/kssl.h:150: syntax error before '*' token /usr/include/openssl/kssl.h:151: syntax error before '*' token /usr/include/openssl/kssl.h:151: syntax error before '*' token /usr/include/openssl/kssl.h:152: syntax error before '*' token /usr/include/openssl/kssl.h:153: syntax error before "kssl_ctx_setprinc" /usr/include/openssl/kssl.h:153: syntax error before '*' token /usr/include/openssl/kssl.h:155: syntax error before "kssl_cget_tkt" /usr/include/openssl/kssl.h:155: syntax error before '*' token /usr/include/openssl/kssl.h:157: syntax error before "kssl_sget_tkt" /usr/include/openssl/kssl.h:157: syntax error before '*' token /usr/include/openssl/kssl.h:159: syntax error before "kssl_ctx_setkey" /usr/include/openssl/kssl.h:159: syntax error before '*' token /usr/include/openssl/kssl.h:161: syntax error before "context" /usr/include/openssl/kssl.h:162: syntax error before "kssl_build_principal_2" /usr/include/openssl/kssl.h:162: syntax error before "context" /usr/include/openssl/kssl.h:165: syntax error before "kssl_validate_times" /usr/include/openssl/kssl.h:165: syntax error before "atime" /usr/include/openssl/kssl.h:167: syntax error before "kssl_check_authent" /usr/include/openssl/kssl.h:167: syntax error before '*' token /usr/include/openssl/kssl.h:169: syntax error before "enctype" In file included from tls/tls_init.h:27, from main.c:112: /usr/include/openssl/ssl.h:909: syntax error before "KSSL_CTX" /usr/include/openssl/ssl.h:931: syntax error before '}' token make: *** [main.o] Error 1 Could you help me fix this problem ? Thank in advance! Best Regard. [View Less]

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[Users] Problem with OpenSER
by Krishna Chava 10 Nov '05

10 Nov '05

Hi, I am new to OpenSER or SER and am having problems registering users to OpenSER. i had the Default user Admin registered to my softphone but am unable to register my other users.. i am using mysql with openSER.. i saw the difference between the admin user that i had default in Subsriber table and the user 'abc' that serctl added when i sed the command serctl add 'abc' 'abc' 'a(a)a.com' <mailto:'a(a)a.com'> and i see the user added to the subscriber table with 1 difference....… [View More]

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[Serusers] "Click to call" with SER ?
by Christophe Irles 10 Nov '05

10 Nov '05

Hi, Is it possible to create a "click to call" button (from a contact web list for example) with SER since I know that the current user as as SIP device already registered ? Which module must be used ? Is there any examples / URL on this subject ? Thanks, Christophe

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[Users] accuracy of retransmission timer
by Klaus Darilion 10 Nov '05

10 Nov '05

Hi! I often see retransmission of of requests sent by openser after ~100ms. Is this normal? regards klaus

2 1

Re: [Users] How to add Route headers?
by Bogdan-Andrei Iancu 10 Nov '05

10 Nov '05

Hi Gustavo, if you have fixed content, it's even simpler. Use the textops module to add the headers. See the append_hf() function: http://www.openser.org/docs/modules/1.1.x/textops.html#AEN189 regards, bogdan Gustavo García Bernardo wrote: >Hi Bogdan, thx for your response > >I need to put two fixed route headers, like these: > >Route: user(a)as.domain.com >Route: user(a)cscf.domain.com > >Can I do that with AVPops module? I think is not suitable for this task. … [View More]

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[Users] How can i run openser ?
by messa＠innsof.com 10 Nov '05

10 Nov '05

Hi all I compile and install Openser-1.0.1-tls.tar.gz successfully. But in the documentation, i can't find how i can run and use it? Can you tell me? regards! Serge MESSA

2 1

[Serusers] RV: [Serdev] Parse Referred-By
by Ing. Francisco Talavera 10 Nov '05

10 Nov '05

Hi all, I need to parse the header field Referred-By, can any one tell me. How can I do this? For example, which files I must to edit because the SER-0.9.0 can NOT parse this header field. Thanks in advance, Ing. Francisco Talavera Dpto. Investigación y Desarrollo Conexión S.A.

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[Users] compile error
by Nguyen Duc Phi 10 Nov '05

10 Nov '05

Hi all, I am compile openser-1.0.0-tls on Red Hat Linux 3.2.3-52 openssl-0.9.7a-33.17 openssl-devel-0.9.7a-33.17 It's show error : gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"openser"' -DVERSION='"1.0.0-tls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -… [View More]DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DUSE_TLS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/ssl/include -c main.c -o main.o In file included from /usr/include/openssl/ssl.h:179, from tls/tls_init.h:27, from main.c:112: /usr/include/openssl/kssl.h:72:18: krb5.h: No such file or directory In file included from /usr/include/openssl/ssl.h:179, from tls/tls_init.h:27, from main.c:112: /usr/include/openssl/kssl.h:134: syntax error before "krb5_enctype" /usr/include/openssl/kssl.h:136: syntax error before '*' token /usr/include/openssl/kssl.h:137: syntax error before '}' token /usr/include/openssl/kssl.h:149: syntax error before "kssl_ctx_setstring" /usr/include/openssl/kssl.h:149: syntax error before '*' token /usr/include/openssl/kssl.h:150: syntax error before '*' token /usr/include/openssl/kssl.h:151: syntax error before '*' token /usr/include/openssl/kssl.h:151: syntax error before '*' token /usr/include/openssl/kssl.h:152: syntax error before '*' token /usr/include/openssl/kssl.h:153: syntax error before "kssl_ctx_setprinc" /usr/include/openssl/kssl.h:153: syntax error before '*' token /usr/include/openssl/kssl.h:155: syntax error before "kssl_cget_tkt" /usr/include/openssl/kssl.h:155: syntax error before '*' token /usr/include/openssl/kssl.h:157: syntax error before "kssl_sget_tkt" /usr/include/openssl/kssl.h:157: syntax error before '*' token /usr/include/openssl/kssl.h:159: syntax error before "kssl_ctx_setkey" /usr/include/openssl/kssl.h:159: syntax error before '*' token /usr/include/openssl/kssl.h:161: syntax error before "context" /usr/include/openssl/kssl.h:162: syntax error before "kssl_build_principal_2" /usr/include/openssl/kssl.h:162: syntax error before "context" /usr/include/openssl/kssl.h:165: syntax error before "kssl_validate_times" /usr/include/openssl/kssl.h:165: syntax error before "atime" /usr/include/openssl/kssl.h:167: syntax error before "kssl_check_authent" /usr/include/openssl/kssl.h:167: syntax error before '*' token /usr/include/openssl/kssl.h:169: syntax error before "enctype" In file included from tls/tls_init.h:27, from main.c:112: /usr/include/openssl/ssl.h:909: syntax error before "KSSL_CTX" /usr/include/openssl/ssl.h:931: syntax error before '}' token make: *** [main.o] Error 1 Could you help me fix this problem ? Thank in advance! Best Regard. [View Less]

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Re: [Serusers] problem with rtpproxy
by Greger V. Teigre 10 Nov '05

10 Nov '05

Please copy the list. If you use a vanilla onsip.org config, it's pretty thoroughly tested by now, so I would be tempted to say that there is something wrong with the client or the setup of the client. g-) ----- Original Message ----- From: rajesh volluru To: Greger V. Teigre Sent: Wednesday, November 09, 2005 2:04 AM Subject: Re: [Serusers] problem with rtpproxy hi Thanks for your reply. I did configure ser.cfg as said at onsip.org for rtp proxy. But, i still get below … [View More]exception when I am behind NAT. net.java.sip.communicator.media.MediaException: Incorrect SDP data! at net.java.sip.communicator.media.MediaManager.openMediaStreams(MediaManager.java:307) at net.java.sip.communicator.SipCommunicator.callStateChanged(SipCommunicator.java:758) at net.java.sip.communicator.sip.Call.fireCallStatusChangedEvent(Call.java:250) at net.java.sip.communicator.sip.Call.setState(Call.java:156) at net.java.sip.communicator.sip.CallProcessing.processAck(CallProcessing.java:566) at net.java.sip.communicator.sip.SipManager.processRequest(SipManager.java:1532) at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:204) at gov.nist.javax.sip.EventScanner.run(EventScanner.java:352) at java.lang.Thread.run(Unknown Source) Caused by: javax.sdp.SdpParseException: Could not parse message at javax.sdp.SdpFactory.createSessionDescription(SdpFactory.java:114) at net.java.sip.communicator.media.MediaManager.openMediaStreams(MediaManager.java:302) ... 8 more Please let me know the possible reason for this error. When I start ser, it shows the message ' RTP proxy found. support for RTP enabled'. Thanks Rajesh "Greger V. Teigre" <greger(a)teigre.com> wrote: Are you using the configs from onsip.org? If not, you may find some help there ;-) g-) ----- Original Message ----- From: rajesh volluru To: Sip Users Sent: Tuesday, November 08, 2005 12:37 AM Subject: [Serusers] problem with rtpproxy hi I had installed ser with media and rtp proxy. Below is the output that i get when I start ser. However, iam not able receive any audio transmission when I connect through sip communicator when iam behind NAT. I am able to register with the server. 0(0) fixing /usr/local/lib/ser/modules/maxfwd.so mf_process_maxfwd_header 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so client_nat_test 0(0) fixing /usr/local/lib/ser/modules/textops.so search 0(0) module - fixing ^Record-Route: 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) fixing /usr/local/lib/ser/modules/nathelper.so fix_nated_sdp 0(0) fixing /usr/local/lib/ser/modules/rr.so record_route 0(0) fixing /usr/local/lib/ser/modules/rr.so loose_route 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup 0(0) get_connection(): Connection not found in the pool 0(0) preload_udomain():! Table is empty 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/auth_db.so www_authorize 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/auth.so www_challenge 0(0) fixing /usr/local/lib/ser/modules/registrar.so save 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf 0(0) fixing /usr/local/lib/ser/modules/textops.so search 0(0) module - fixing ^Route: 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so use_media_proxy 0(0) fixing /usr/local/lib/ser/modules/tm.so t_on_reply 0(0) fixing /usr/local/lib/ser/modules/tm.so t_relay 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_reply_error 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so use_media_proxy 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so client_nat_test 0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) DEBUG: udp_init: trying SO_RCVBUF: 221184 0(0) DEBUG: setting SO_RCVBUF; set=221184,verify=221184 0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232 0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=221184 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF i! s finally 221184 0(0) DBG: open_uac_fifo: opening fifo... 0(0) DEBUG: FIFO created @ /tmp/ser_fifo 0(0) DEBUG: fifo /tmp/ser_fifo opened, mode=432 0(0) init_unixsock_socket: No unix domain socket will be opened 2(0) INFO: fifo process starting: 19946 2(0) DEBUG: init_mod_child (-2): tm 2(0) DEBUG: callid: '57b7f7f7-0(a)69.229.23.57' 2(0) DEBUG: init_mod_child (-2): usrloc 2(0) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): tm 0(19941) DEBUG: callid: '57b7f7f7-19941(a)69.229.23.57' 0(19941) DEBUG: init_mod_child (1): usrloc 0(19941) get_connection(): Connection not found in the pool 1(0) DEBUG: init_mod_child (-1): tm 1(19945) DEBUG: callid: '57b7f7f7-19945(a)69.229.23.57' 1(19945) DEBUG: init_mod_child (-1): usrloc 1(19945) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): auth_db 0(19941) get_connection(): Connection not found in the pool 1(19945) DEBUG: init_mod_child (-1): auth_db 1(19945) get_connection(): Connection not found in the pool 0(19941) DEBUG: init_mod_child (1): domain 0(19941) get_connection(): Connection found in the pool 0(19941) DEBUG: init_mod_child (1): nathelper 2(19946) DEBUG: init_mod_child (-2): auth_db 2(19946) get_connection(): Connection not found in the pool 1(19945) DEBUG: init_mod_child (-1): domain 1(19945) DEBUG: init_mod_child (-1): nathelper 0(19941) rtpp_test: RTP proxy found, support for it enabled 1(19945) rtpp_test: RTP proxy found, support for it enabled 2(19946) DEBUG: init_mod_child (-2): domain 2(19946) DEBUG: init_mod_child (-2): nathelper 2(19946) rtpp_test: RTP proxy found, support for it enabled 2(19946) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(19946) DEBUG: register_fifo_cmd: new command (print) registered 2(19946) DEBUG: register_fifo_cmd: new command (uptime) registered 2(19946) DEBUG: register_fifo_cmd: new command (version) registered 2(19946) DEBUG: register_fifo_cmd: new command (pwd) registered 2(19946) DEBUG: register_fifo_cmd: new command (arg) registered 2(19946) DEBUG: register_fifo_cmd: new command (which) registered 2(19946) DEBUG: register_fifo_cmd: new command (ps) registered 2(19946) DEBUG: register_fifo_cmd: new command (kill) registered 2(19946) WARNING: no fifo_db_url given - fifo DB commands disabled! Let me know if you find any configuration error in SER. Please let me if you need any more information. I am trying to configure SER from long time. Please help me. Thanks Rajesh -------------------------------------------------------------------------- Enjoy this Diwali with Y! India Click here -------------------------------------------------------------------------- Enjoy this Diwali with Y! India Click here -------------------------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ------------------------------------------------------------------------------ Enjoy this Diwali with Y! India Click here [View Less]

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[Users] rport error
by Marco 10 Nov '05

10 Nov '05

Hi all. I've a problem with a call using mediaproxy. The call got established, but after a few seconds got disconnected and I obtain the following errors. The problem appears with several clients. Please can someone enlighten me on where I err? TIA Nov 8 23:43:20 fw /usr/sbin/openser[11625]: ERROR: parse_via: invalid port number <12797rport> Nov 8 23:43:20 fw /usr/sbin/openser[11625]: ERROR: parse_via on: <SIP/2.0/UDP 213.144.x.x:12797rport;branch=z9hG4bK510915090^M Route: <… [View More]

2 3

[Users] How to add Route headers?
by Gustavo García Bernardo 09 Nov '05

09 Nov '05

Hi, Is posible to add more route headers to a request before relaying it? Im trying to implement the IMS mechanism before relaying a request to an application server, which consists in adding two route headers. Thx in advance. G.

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