- sr-users - kamailio.org

[Serusers] Outbound sip proxy
by harry gaillac 22 Nov '05

22 Nov '05

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[Serusers] Problems to install presence server in SER
by victoria beltran 22 Nov '05

22 Nov '05

Hi, I am trying to install the last unstable SER version with support for SIP-based presence. The server is a Fedora core 1 machine. I have installed the libcurl (curl-7.10.6-7) and libxml2 (libxml2 2.6.6-3) libraries. When I compile and install the libraries of SER (make f Makefile.ser install), it does several errors. The most important are: In file included from parse_pres_rules.c:35: /root/sip_router/lib/xcap/xml_utils.h:26:27: libxml/parser.h: No such file or di rectory /root/sip_router/lib/xcap/xml_utils.h:27:25: libxml/tree.h: No such file or dire ctory Could anybody help me? Thank you --------------------------------- Correo Yahoo! Comprueba qué es nuevo, aquí http://correo.yahoo.es

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[Users] OAM monitoring of OpenSER? (via SNMP?)
by dhsueh＠somanetworks.com 22 Nov '05

22 Nov '05

Hello, Does anyone know of a module or other code for OpenSER that collects and provides OAM monitoring information on the state of the server? The end goal would be to monitor the process via SNMP. Any information-gathering support will be of assistance. Ideally, I'd like to see information like that specified in the IETF draft SIP MIBs available for collection/retrieval. Thank you! -- Daniel Hsueh mailto:dhsueh@somanetworks.com SOMA Networks tel:+1-416-348-1631

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[Users] exec_dset and NOT changing ruri
by Iqbal 22 Nov '05

22 Nov '05

Hi Is this possible ? just want to use the features of dset, to pull in call info into a script Iqbal

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[Serusers] new postgres module
by Klaus Darilion 22 Nov '05

22 Nov '05

Hi! I've tried to port the new postgres module to openser and had some problems: .. symbol PQserverVersion not found .. after removing PQserverVersion from source, the module loads but there are other problems: 0(0) postgres:timestamp_format: Binary format expected but server sent text 0(0) postgres: Error: 0(0) postgres: Unknown error occurred, giving up Are there some known limitations? Does the new module requires postgresql 8.0, or a certain postgresql client library? regards klaus

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[Serusers] Mediaproxy monitor using Nagios
by Ryan Pagquil 22 Nov '05

22 Nov '05

Hi, I'm thinking of monitoring mediaproxy using nagios. I've already done monitoring SER using nagios using sipsak. I tried telnet-ing on 25060 but not successful. Please help. Thanks, Ryan

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[Serusers] Change ser database structure
by chew pei nee 22 Nov '05

22 Nov '05

Hi all, i would like to change the ser database structure. For example, i wan the field "username" in Subscriber table. Is there any way i can achieve this? Thanks in advance. rgds Penny

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[Serusers] Caller ID How??
by Eric Haskins 21 Nov '05

21 Nov '05

Hello, I have a question regarding caller ID currently I am catching some flak because when ou outbound calls ring to the PSTN they see ,for instance SER username is 6013, 6013 on their caller ID. How do you handle this to show a valid inbound number on a user by user basis. We have a couple different people that use our server for outbound termination only they want it for instance to show 888-555-1212. Is there anything I can do to force ser to show a fulle telephone number instead of a user. I was thinking of changing their username to be the phone number they want to show up but is their another way? Eric Haskins 978-375-6112 Cell

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

21 Nov '05

Remarque : message transféré en pièce jointe. ___________________________________________________________________________ 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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[Users] reading passed env variables from within php script
by Iqbal 21 Nov '05

21 Nov '05

Hi I am trying to read the env variables which are passed in exec_msg, from within the script, and have tried $_ENV etc from with the script, but all blank, is there and easier/better way of getting the details in the message into a script iqbal

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[Users] Regular Expression based groups
by Bogdan-Andrei Iancu 21 Nov '05

21 Nov '05

Hi, I just committed a new group feature: so far we have group support with fixed matching: user X belongs to group G; the new regexp support allows you to fetch all the groups for which the user matched a RE (if the URI match the Re, means the belongs that group). See http://www.openser.org/docs/modules/1.1.x/group.html Application example: imagine you have routing point between several termination points (gateways, media servers, application servers, etc) and you have large blocks of numbers/users (K or 10K order). you know that certain range of users must be routed to specific destinations. So use RE based groups to define the range of users -> flexible and scalable routing and simple to provision. regards, bogdan

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

21 Nov '05

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[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

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1 0

[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@.*$" : "^sip:.*192\.168\.1\.10$" "^sip:foo@.*$" : "^sip:.*192\.168\.1\.15$" "^sip:boo@.*$" : "^sip:.*192\.168\.1\.20$" "^sip:boo@.*$" : "^sip:.*192\.168\.1\.30$" but it doesn't work. SER lets foo@ and boo@ register using whatever IP they want to. Where is the mistake? Am I missing something? Thank you Ciao FF

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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 fereign keys and there are other kind of limitation. Let me know if you can suggest a good way to do it. I need the transactional functions, foreign keys and to be sure that data are flushed to disk almost immediately after the query executed (this is something that the cluster do not assure to you because it keep in memory lot of data and flush it to disk in big chunks) for the application that run on top of SER/OpenSer. The goal of my application is to have 2 OpenSer (balanced using SRV records or something like this) servers working on two separate mysql but dealing with the same data. For now the best solution for reliability that i found is to have 2 OpenSer servers to operate on the 1st mysql and the 2nd in sync as slave with one way replication and in case of failure to move the IP address of mysql on the slave and restart the slave as master. All this with Heartbeat package. Bye, Marcello >Hi Daryl, > >the 2-way replication in Mysql works and it might be used along with >OpenSER - depends of your configuration. > >regards, >bogdan > >Daryl Sanders wrote:

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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 continuously retransmits INVITES to SER but nothing happens. I would appreciate your help ramin

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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 config. Thanks! - Daryl

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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! > > -- > Pol Novell > UPF - Universitat Pompeu Fabra. >

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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 is quite a daunting task given the number of possible call flow scenarios so if you wish that your calls always work regardless of UA type or call flow you will enable a media relay because Stun is not reliable 100%. Adrian >>>>>>>>>> Terrible question: Is mediaproxy better than rtp or STUN server for NAT problem ??? Radius Authentication work fine with Mediaproxy, STUN and RTP ? In Mediaproxy Server, radius auth supported only in non-free version (7.000,00 euro) ! My work is: Private Network for more than 1000 users Public SER SERVER Public Radius+MySQL Public Patton MediaGW Please, answer ! Thanks Antonello

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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..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 > > > >

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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 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. 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.

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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=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.

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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 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 > > >

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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 Effective is STUN? > To: users(a)openser.org, serusers(a)iptel.org > Message-ID: <437E6583.5050905(a)galaxytelecom.net> > Content-Type: text/plain; charset=ISO-8859-1 > > 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: [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 INVITE into INVITE 600001 @ gnomemeeting.net. Is that possible? -- _ Damien Sandras (o- GnomeMeeting: http://www.gnomemeeting.org/ //\ FOSDEM 2005 : http://www.fosdem.org v_/_ H.323 phone : callto:ils.seconix.com/dsandras@seconix.com

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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 asterisk ? I've been trying mangle and textops modules but i really need to be adviced. One box --------------------------- | ---------------- | | | asterisk pbx | | | ---------------- | | || | | ---------- ---------- | | SER |====|NAT box |==== private network | ---------- ---------- --------------------------- Regards Harry ___________________________________________________________________________ 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] 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 lost? Has anybody here done this successfully? Thanks for your patience and time. -- Regards, Jan Henkins

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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. 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

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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 (use ser_mysql.sh reinstall) 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration I've used the template database files Daniel provided as well, they result in the same fault. openser.cfg notes: modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("usrloc|auth_db", "db_url", "dbtext:///ext/pbx/lib/openser/dbtext") I'm probably using wrong database files, but I couldn't find any proper samples so far. Suggestions anyone ? Thanks, Florian

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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 server? Please help, Colin Jordan DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

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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

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[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, because, for the phones, the call just goes to the private SER IP, and they dont know nothing besides that. Does someone have a setup like this with Cisco phones? Thanks Joao

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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: 192.168.0.47 ##################################################################### 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

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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/username", "i:19/usr_preferences"); }; Or some such (as I say, I'm playing loose and fast with syntax here)... N. On Wed, 16 Nov 2005 17:30:16 +0200, Kyriakos Mavromichalis wrote > Hi, > actually I was thinking that users would press *21*1*number to > forward to , number would be stored to database. For disabling it > could be something like *21*2 , which in that case , the record in > db with the call forwarding option for the user who sent the message > would be deleted. The problem is how do i write and delete in the > usr_preferences table from the ser.cfg file? Of course the users > will also have the option of enabling call forwarding from a web > interface but adding this its much more practical. > > K. > > On Wednesday 16 November 2005 16:47, you wrote: > > If you do a search on the list, I THINK I posted a way to do call > > forwarding (the way I was doing it anyway)... there are likely other ways. > > How are you going to take input for the forwarding, though? > > > > Will they dial *21 and it will forward to something they've predefined on a > > website somewhere? Will they dial *21 and then the number to which they > > want to forward? How do you turn it off? Etc, etc. > > > > N. > > > > > > On Wed, 16 Nov 2005 16:44:15 +0200, Kyriakos Mavromichalis wrote > > > > > 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 > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > -- > Kyriakos Mavromichalis > Otenet Telecom

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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: "System Settings"-> "Network" -> "Listen RTP port" ) Thanks Joao Pereira

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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 source packages on http://developer.berlios.de/projects/ser/ <http://developer.berlios.de/projects/ser/> . Are pre-built executable SER packages no longer distributed? Thanks. Doug Davey

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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 functionality and it's too small to be a separate module. any pro or against comments / suggestions before committing? regards, Bogdan

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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 > > 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 the architecture of this > nathelper module. > > Thanks in advance! > > Best regards > > Serge > >

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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 the architecture of this > nathelper module. > > Thanks in advance! > > Best regards > > Serge > >

1 0

[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 asterisk ? I've been trying mangle and textops modules but i really need to be advice. One box --------------------------- | ---------------- | | | asterisk pbx | | | ---------------- | | || | | ---------- ---------- | | SER |====|NAT box |==== private network | ---------- ---------- --------------------------- Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[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 asterisk ? I've been trying mangle and textops modules but i really need to be advice. One box --------------------------- | ---------------- | | | asterisk pbx | | | ---------------- | | || | | ---------- ---------- | | SER |====|NAT box |==== private network | ---------- ---------- --------------------------- Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[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!

2 1

[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 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

2 1

[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.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

3 3

[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 found this case? I have tried to use command "isdn overlap-receiving" on serial interface. Thank you Ake

1 0

[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? Has anyone encountered this? If it is python problem, then why it can distribute the load evenly when stream locally? Kernel Problem?? Code??? Thanks for any help available. Regards, TC Chan

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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 supports the UPDATE method. Thanks in advance, Regards / Met vriendelijke groet, --------------------------------------------- R.L.L.M. Voermans Manager Network Connectivity Intern Global-e Raadhuisstraat 32 5126 CJ Gilze (NL) T: +31-(0)161-888888 F: +31-(0)161-888899 E: r.voermans(a)global-e.nl <mailto:r.voermans@global-e.nl> W: http://www.global-e.nl <http://www.global-e.nl/> ---------------------------------------------

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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 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 > >

1 0

[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. 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> > > >

3 3

[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 requested by server; consider upgrading MySQL client Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: db_init(): Could not create a connection Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: ERROR: uridb_db_ver: unable to open database connection Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: ERROR: uri_db:mod_init(): Error while querying table version Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: init_mod(): Error while initializing module uri_db When those situation occur SER always unable to start, please see this message: Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed I think it is to do with mysql connection. I will appreciate if someone, please help me how to fix this. I am sure a number of other people will be facing this issue as well. Please help !! Please Help! Thanks Dhiman

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 requested by server; consider upgrading MySQL client Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: db_init(): Could not create a connection Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: ERROR: uridb_db_ver: unable to open database connection Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: ERROR: uri_db:mod_init(): Error while querying table version Nov 17 20:26:12 voice /usr/local/sbin/ser[2753]: init_mod(): Error while initializing module uri_db When those situation occur SER always unable to start, please see this message: Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed I think it is to do with mysql connection. I will appreciate if someone, please help me how to fix this. I am sure a number of other people will be facing this issue as well. Please help !! Please Help!

1 0

[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 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

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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 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 >

5 6

[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>

3 3

[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

3 5

[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

1 1

[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

1 0

[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) "A" INVITEs "B" through SER 2) "B" says that its trying and ringing to "A" 3) User at "B" picks-up 4) I check the 200-ok at SER. If it´s ok, let it flow to "A". If it´s not, I want to drop it and terminate the call establishment. DO ANYBODY DID THAT OR CAN HELP ME WITH ANY CLUE/STARTING POINT? best regards to all. Ricardo Poppi.

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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

2 2

[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 scenario? thanks klaus BYE from callee (5060) to proxy U 2005/11/09 16:47:48.254756 83.136.33.19:5060 -> 83.136.32.83:5060 BYE sip:83.136.32.83:5060;r2=on;ftag=a613b273;nat=yes;lr=on SIP/2.0. Via: SIP/2.0/UDP 83.136.33.19:5060;branch=z9hG4bK3dc73e84. From: <sip:01505641636@enum.at>;tag=000dedfb04cc011e597bda33-277ea073. To: klaus enum.at<sip:klaus@enum.at>;tag=a613b273. Call-ID: 7f49f21ed451bb7a. Date: Wed, 09 Nov 2005 15:47:47 GMT. CSeq: 101 BYE. User-Agent: CSCO/6. Content-Length: 0. RTP-RxStat: Dur=??,Pkt=??,Oct=??,LatePkt=??,LostPkt=??,AvgJit=??. RTP-TxStat: Dur=??,Pkt=??,Oct=??. Route: <sip:83.136.32.83:6060;r2=on;ftag=a613b273;nat=yes;lr=on>, <sip:klaus@84.20.167.143:7404>. . BYE from proxy to caller: is sent from 5060 althouth it should be sent from 6060. # U 2005/11/09 16:47:48.264940 83.136.32.83:5060 -> 84.20.167.143:7404 BYE sip:klaus@84.20.167.143:7404 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:83.136.32.83;ftag=000dedfb04cc011e597bda33-277ea073;nat=yes;lr=on>. Via: SIP/2.0/UDP 83.136.32.83;branch=z9hG4bKaf5b.fbc8b274.0. Via: SIP/2.0/UDP 83.136.33.19:5060;branch=z9hG4bK3dc73e84. From: <sip:01505641636@enum.at>;tag=000dedfb04cc011e597bda33-277ea073. To: klaus enum.at<sip:klaus@enum.at>;tag=a613b273. Call-ID: 7f49f21ed451bb7a. Date: Wed, 09 Nov 2005 15:47:47 GMT. CSeq: 101 BYE. User-Agent: CSCO/6. Content-Length: 0. RTP-RxStat: Dur=??,Pkt=??,Oct=??,LatePkt=??,LostPkt=??,AvgJit=??. RTP-TxStat: Dur=??,Pkt=??,Oct=??. .

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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 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

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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 | ---------------- || ---------- ---------- | SER |=======|NAT box |======== private network ---------- ---------- ___________________________________________________________________________ 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] 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 is looking for... I already set in radiusd.conf: password_attribute = userPassword But still the same error.... Any clues? 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 possible!" Jesus Christ in Matthews 19:26

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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: 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

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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 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 > >

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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 , actually im a newbe on openser but I can see a lot of potential if we had a basic manual and soma basic setup examples. Thanks a lot for your help. Fernando Rodriguez V.

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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 have mysql started on my system. -- Mahesh. __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

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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.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 >

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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 >>> 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

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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 have a known location, dial them. Don't know if this would give errors when the same user is connected to multiple ser's? thx Arne

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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 calls because I get the invite, I send it to the UA, and then I do another dial if there is a forward. That of coarse costs me twice as much. I have tried... if(t_check_status("302")){ # what to do if device tells us to forward to another number xlog("A redirect"); get_redirects("*","redirect"); rewritehost("our.pstn.gw.ip"); t_relay(); return; I simply need it to get the number to redirect to from the contact portion and forward the invite to that number @ our pstn gateway. Any help greatly appreciated.

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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; } else { log(1," sipldap call"); }; }; I'm getting "Not Found", but when I'm doing (transferring to PSTN route without sl_send_reply) (!lookup("location")) { if (!exec_dset("/etc/ser/sipldap")) { #sl_send_reply("404", "Not Found"); route(5); break; } else { log(1," sipldap call"); }; }; route[5] { rewritehostport(xxxxxxxxxxx); I do extract LDAP telephone number (so "sipldap" script works OK) and do get call transferred to PSTN GW but still getting "ERROR: exec_str:" thanx davor

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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 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

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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/ 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.

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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 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 >

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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".

1 0

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 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 > > >> > > >> > > >> >

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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 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 >> >> >>

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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"); break; }; The SIP_DOMAIN is set to voip Anyone facing this issue too? Thanks Regards, Chia -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/172 - Release Date: 11/15/2005

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[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"); break; }; The SIP_DOMAIN is set to voip Anyone facing this issue too? Thanks Regards, Chia -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/172 - Release Date: 11/15/2005

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[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 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

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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

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[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

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[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/mysql.so>: libmysqlclient.so.14: cannot open shared object file: No such file or directory' 'libmysqlclient.so.14' is in '/usr/local/lib/mysql/' Any idea ? Thank. -- Murdej Ukrutny

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[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, 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

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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 asterisk, and then sending the bye to the other client (the one who didn't hang up). on ser: ser receives the invite, looking up the address of the callee, and sending the invite through to the callee, the contact header allways stays the caller's uri, but ser put's a record-route in the sipmessage, so that the bye first is send to the ser machine, and then to the other party. Am I seeing things correct here? What are the pro/contra's of both procedures? Imo gives the asterisk way a lot more overhead, but is more secure (2 clients keep communicating with asterisk, and are unable to fake things (or less easely), there were an not compliant client can bypass the record-route of ser, and so avoiding accounting. Again, am I correct here? thx Arne

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; } else { log(1," sipldap call"); }; }; I'm getting "Not Found", but when I'm doing (transferring to PSTN route without sl_send_reply) (!lookup("location")) { if (!exec_dset("/etc/ser/sipldap")) { #sl_send_reply("404", "Not Found"); route(5); break; } else { log(1," sipldap call"); }; }; route[5] { rewritehostport(xxxxxxxxxxx); I can extract LDAP telephone number and do get call transferred to PSTN GW but still getting "ERROR: exec_str:" thanx davor

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: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

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