sr-users November 2005

sr-users@lists.kamailio.org

245 participants
571 discussions

[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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sr-users November 2005