sr-users October 2005

sr-users@lists.kamailio.org

214 participants
458 discussions

[Serusers] ser as outbound proxy only
by jeff kwong 26 Nov '05

26 Nov '05

Hi to everybody! Im currently trying to setup an outbound proxy using ser. Im still a newbie with this ser and im not yet that good with the scripting. Could anyone tell me how to configure ser for outbound proxy only using rtpproxy? There are samples files in the Getting started manual but im having troubles eliminating the authentication/mysql part in script since im not yet familiar which modules depends on which. A striped down version of the nat-rtpproxy.cfg file would be really appreciated. Thanks in advance. _jeff

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

15 Nov '05

Hi, A newbie question: Is there a way for OpenSER to use SIP headers (other than the URI set) that have been modified/processed by an external command (exec_dst, exec_msg)? Reading the documentation and mailing list archives revealed some questions similar to mine but no answer. The problem is: Delegate the change of one or more SIP message headers (or the whole SIP message) to an external application via exec module. Two straight-forward solutions: 1. exec_rmsg() - for "replace message". A merge of the functionality of exec_dst() and exec_msg(), i.e. OpenSER uses the output of exec_rmsg(command) to replace the current SIP message. This is imho an extremly flexible solution but slightly difficult to implement as it requires at least parsing the command's output - similar to a message received by OpenSER. 2. The command executed by exec_msg() can modify the environment variables SIP_HF_<hf_name> and OpenSER reuses these values after the exec() returned control to SER. Setvars could be used to switch between the current, read-only meaning of env-variables (e.g., modparam("exec", "setvars", 1)) and a read-write behavior (e.g. modparam("exec", "setvars", 2). This solution does not require any special handling except that OpenSER must read back the content of all environment variables which were passed to the command. Probably quite easy to implement but has the drawback that no new header fields can be added. Imho no major limitation as users can add their (dummy) header fields prior to invoking the command via exec() Or is there any other existing/planned procedure for SER to use the output of external commands for rewriting an existing SIP message. tia best regards --Joachim

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

11 Nov '05

Hi, We'd like to force the following SIP INVITE routing scenario: UA1->P1->P2->R->P2->P1->UA2. Unfortunately the Route is not known apriori and must be built dynamically. P1 knows the Route from P1 to R, R knows the Route from R to P1/UA2. The INVITE sent by UA1 is loosely routed to R based on the Route (Route: <sip:P2;lr>,<sip:R;lr>) that is added by P1 to UA1's INVITE. This works. The message arrives correctly to the registrar, Route header contains Route: <sip:R;lr> as it should. What we'd like to do: 1) Lookup the contact (lookup("location");) 2) Rewrite the Route field to the value Route: <sip:P2;lr>,<sip:P1;lr> that is stored by R in an AVP. 3) Forward the Invite on this route to UA2 Can someone please give some hints on how this is typically done in OpenSER? Although we rewrite the Invite's Route header field (new value stored in an AVP) at the Registrar, loose_route() does not work. We tried several alternatives including re-writing the Route header, pushing hard-coded strings into the Route, etc. but none of these solutions does what we want. What does work is if R self-forwards the Invite and only then adds a new Route header. Apparently loose_route() in R acts on the route header of the incoming INVITE - if present - and ignores route header changes done within the script. Thanks in advance --Joachim

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

10 Nov '05

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

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[Serusers] Prepaid approach
by Ashutosh kumar 09 Nov '05

09 Nov '05

Hi, While trying to implement prepaid solution using SER, I decided to go be a intuitive approach which is as follows. -When the user registers, he is placed in a "voip" groups table of ser (or radius server database), i.e he can make only pc-to-pc calls. - Later , or otherwise, when the user registers, he is shifted to a "pstn" groups table of ser , i.e noew he can make only pc-to-pstn calls. -Henceforth, whenever the user logins and tries to make pstn call, the call is approved only if group_radius_is_user_in(username) succeds, else the call rejected by SER. -To restrict the user from making pstn calls when his credits are zero (or beyond a threshold) , a dedicated cron job is scripted to move users from "pstn" group to "voip" when their account_cerdit=0. Am I right in using this approach, or are there any foreseeable problems which I might be overlooking. Thanks. Regards, Ashutosh Kumar

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[Users] acc error
by Jayesh Nambiar 08 Nov '05

08 Nov '05

hi all, I compiled the latest version od openser from cvs head. I've set the acc. I've loaded module after tm and mysql module. I added the db_url parameter, db_flag, db_table_acc, db_extra, radius_config, service_type and radius_flag. I've compiled acc with both mysql and radius support. But while starting openserctl I get the following error: Oct 27 10:41:06 localhost /usr/local/sbin/openser[6575]: ACC - initializing Oct 27 10:41:06 localhost /usr/local/sbin/openser[6575]: ERROR: acc: can't get code for the Sip-Method attribute Oct 27 10:41:06 localhost /usr/local/sbin/openser[6575]: init_mod(): Error while initializing module acc can somebody pls help me with error as I need accounting in both radius and acc. I am eagerly waiting for the 1.0 release. Its tommorow right!!!! jayesh --------------------------------- Enjoy this Diwali with Y! India Click here

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[Serusers] ser+nat+siproxd+asterisk
by jeff kwong 07 Nov '05

07 Nov '05

Hi Guys! I just would like to share that I was able to get a working setup using SER as Softswitch, Asterisk as PSTN gateway and SIPROXD on my NAT Router. SIPROXD is an open source ALG and it effectively handles sip nat traversals. With it I dont have to run a seperate mediaproxy. When making calls from SIP UA to PSTN, RTP is as below: UA---NAT/SIPROXD---ASTERISK for 2 UA behind the same NAT: UA1--NAT---UA2 and for 2 UA behind different NATs: UA1--NAT1----NAT2---UA2 Thus there is less latency on signals and less traffic on SER. My question is, from the experience of other guys here, what do you think is the drawback or advantages of using SIPROXD together with SER to solve SIP NAT issues compared to other methods like using mediaproxy and rtpproxy?Will I still be able to do other SER features like accounting? Thanks! _jeff

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[Serusers] ser+asterisk example
by julien bos 05 Nov '05

05 Nov '05

Hi all, I am working with asterisk to active voicemail for ser. I am newbie for it. So can you give me an example of ser.cfg for do that? Thank you very much for your instruction.

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[Serusers] FW: EmbeddedObj.o Sipstack Make error
by administrator 03 Nov '05

03 Nov '05

_____ From: administrator [mailto:administrator@hellasfon.com] Sent: Wednesday, October 26, 2005 11:49 AM To: 'b2bua(a)vovida.org' Subject: EmbeddedObj.o Make error Hello everybody, I am trying to install b2bus from cvs and I get the following error: make[1]: Leaving directory `/root/vocal/sip/sipstack' make[1]: Entering directory `/root/vocal/sip/sipstack' g++ -Wall -fPIC -Wno-deprecated -D_REENTRANT -DUSE_PTHREADS -g -I../../build -I../../build/../sdp/sdp2 -I../../build/../util -I../../build/../util/threads -I../../build/../util/logging -I../../build/../util/crypto -I../../build/../util/statistics -I../../build/../util/snmp -I../../build/../util/signals -I../../build/../util/behavior -I../../build/../util/io -I../../build/../util/services -I../../build/../util/transport -I../../build/../util/config -I../../build/../util/dnssrv -I../../build/../util/deprecated -I../../build/../util/adt -I../../build/../contrib/libxml2.Linux.i686 -I../../build/../contrib/libxml2.Linux.i686/include/libxml -I../../build/../contrib/libxml2.Linux.i686/include -DVOCAL_USE_DEPRECATED -DVOCAL_USING_PENTIUM -DOLD_PROV -c -o obj.debug.Linux.i686/EmbeddedObj.o EmbeddedObj.cxx BaseUrl.hxx:146: error: specialization of βtemplate<class _Key> struct __gnu_cxx::hashβ in different namespace /usr/lib/gcc/i386-redhat-linux/4.0.1/../../../../include/c++/4.0.1/ext/hash_ fun.h:71: error: from definition of βtemplate<class _Key> struct __gnu_cxx::hashβ EmbeddedObj.cxx: In member function βData Vocal::EmbeddedObj::doReverseEscape(const std::string&)β: EmbeddedObj.cxx:213: warning: comparison between signed and unsigned integer expressions make[1]: *** [obj.debug.Linux.i686/EmbeddedObj.o] Error 1 make[1]: Leaving directory `/root/vocal/sip/sipstack' make: *** [sip] Error 2 I user SER(0.9.4) and freeRadius (latest) on Fedora Core 4. Please Help ASAP. Thank you in advance Yiannis Marios

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sr-users October 2005