sr-users August 2006

sr-users@lists.kamailio.org

184 participants
361 discussions

[Users] Limiting Call Volume and CPS
by Brian Jones 24 Aug '06

24 Aug '06

Hi, I was wondering if anyone had any suggestions on how I could use OpenSER to limit call volume and call per second on a peer basis. Thanks! Brian.

3 4

[Users] Weird forwarding with Route header
by Jeremie Le Hen 24 Aug '06

24 Aug '06

Hi list, my proxy (yy.yy.yy.yy) receives a message like this: BYE sip:12345678@xx.xx.xx.xx SIP/2.0 Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on> Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bK07cb.466d8587.0 Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf Route: <sip:xx.xx.xx.xx;ftag=1056999653;lr=on> My ruleset is something like this (with the REGISTER logic pulled out) : %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; force_rport(); if (!method == "REGISTER") { record_route(); }; if (loose_route()) { route(1); }; loopkup("location"); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% I would expect the packet to leave my router, back to xx.xx.xx.xx : BYE sip:12345678@xx.xx.xx.xx SIP/2.0 Record-Route: <sip:yy.yy.yy.yy;lr;ftag=2873864> Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on> Via: SIP/2.0/UDP yy.yy.yy.yy;branch=zu83fdhDB8d.67235676.0 Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bK07cb.466d8587.0 Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf Route: <sip:xx.xx.xx.xx;ftag=1056999653;lr=on> But the outgoing packet is : BYE sip:xx.xx.xx.xx;ftag=1056999653;lr=on SIP/2.0 Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on> Via: SIP/2.0/UDP yy.yy.yy.yy;branch=z9hG4bK07cb.02005244.0 Via: SIP/2.0/UDP xx.xx.xx.xx;rport=5060;branch=z9hG4bK07cb.466d8587.0 Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf Do you have any idea about what is going on ? Thank you. Best regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

2 2

[Users] dropping a 200
by Simon Morvan 24 Aug '06

24 Aug '06

Just to be sure : there's no way to call drop() in an onreply_route block when it's not a provisionnal reply ? -- Simon.

2 1

[Users] how to correctly handle NAT on inbound call from another SIP-proxy?
by Gerd Feiner 24 Aug '06

24 Aug '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hi list, we have some trouble handling NAT correctly when receiving a call from another SIP-proxy. the mediaproxy-module does not seem to rewrite SDP when it finds another proxy's record-route header, though it invokes mediaproxy and sends uncorrected SDP to the other SIP-server resulting in one-way-audio as the provided c=-field includes the RFC1918-address of the client. when receiving calls through our Cisco AS5350 everything is fine ('cause it does not add the record-route). the addition of the header cannot be circumvented bacause the other provider wants to bill calls (obviously) and therefor needs to be in the SIP-dialog. is there a neat solution to this problem or do we have to get a session-border-controller or some B2BUA-config for handling the received calls? and by the way, how do i distinguish external INVITEs destined for one of our clients (uri==myself) from the INVITEs being sent out to the callee upon an external INVITE? if there was a way we could invoke mediaproxy from there? how is the recommended setup? brgds, g.feiner -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE7aVOqvDCGdeEb8oRArDjAJ9ssHEHBAIGmSB34oHQxuuT0ke15wCgmKvB chyomyJgqU9NVMep7QV1nFI= =r5Xb -----END PGP SIGNATURE-----

1 0

[Users] Q:ABOUT:MODULE:auth_radius
by Alejandro Sanchez 24 Aug '06

24 Aug '06

Hello. openser 1.1 radiusclient-ng 5 freeradius 1.1.1 I need to enable the modules *_radius for my openser instalation a follow the guide for enable radius then when i try to start openser i get the following error. 0(0) ERROR: auth_radius: can't get code for the Digest-Response attribute Anyone knows what is the cause for this error? Thank's Alex. ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y <b >2GB</b> extra a tu correo por $100 al mes. http://net.yahoo.com.mx

2 1

[Serusers] multiple domains in Ser
by Ricardo Carvalho 24 Aug '06

24 Aug '06

Hi, Is there a way to coexist multiple domains in Ser, distinguishing users even if they have the same username in subscriber table but different domains? I'll explain better, I have subscriber table with: | username | domain | e-mail | ------------------------------------------------ | user_A | domain_A | user_A@domain_A | ------------------------------------------------ | user_A | domain_B | user_A@domain_B | and in domain table there are both domains domain_A and domain_B. But when someone calls user_A, (with or without writing their domain), both phones ring (if both users have their phones registered), when I want only one to ring because they are different users!! Is there a way in Ser to do this right? Regards, Ricardo.

2 2

Re: [Serusers] multiple domains in Ser
by Ricardo Carvalho 24 Aug '06

24 Aug '06

It's true Asterisk can use different contexts, but that doesn't solve my problem because I use Ser + Asterisk and Ser is used for routing SIP messages between SIP UAs, not Asterisk. In my architecture, Asterisk is only used for advanced services like voicemail or call conferencing and for gateway, not to route messages between UAs. Ricardo. openser(a)comcast.net wrote: > The way to do that is intergrate Ser with Asterisk and in Asterisk you > can build context and keep the same numbers for different context like > a pbx. > > > -------------- Original message -------------- > From: Ricardo Carvalho <rcarvalho(a)iric.up.pt> > > > Hi, > > > > Is there a way to coexist multiple domains in Ser, > distinguishing users > > even if they have the same username in subscriber table but > different > > domains? > > > > I'll explain better, I have subscriber table with: > > > > | username | domain | e-mail | > > ------------------------------------------------ > > | user_A | domain_A | user_A@domain_A | > > ------------------------------------------------ > > | user_A | domain_B | user_A@domain_B | > > > > and in domain table there are both domains domain_A and domain_B. > > > > But when someone calls user_A, (with or without writing their > domain), both > > phones ring (if both users have their phones > > registered), when I w! ant onl y one to ring because they are > different users!! > > Is there a way in Ser to do this right? > > > > Regards, > > > > Ricardo. > > > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Serusers] Voice Mail problem
by ram 24 Aug '06

24 Aug '06

Hi I have existing SER Running i have local users ( extensions) and DID ( all other countries) iam able to make calls in and out , there is no problem but when the user not available, it has to send to Voice mail, but its only sending the users who are locally start with 9*, how can i send the user start with 9 or any DID not availble send to voicemail here is my config if (!lookup("location") && ( uri=~"^sip:9.*@domain.com" ) ) { # Voicemail specific configuration - begin if(method=="ACK" || method=="INVITE" || method=="BYE"){ setflag(2); if(t_newtran()){ t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log(1,"**************** vm start - begin ******************\n"); if(!vm("/tmp/am_fifo","voicemail")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log(1,"**************** vm start - end ******************\n"); break; }; if(method=="BYE"){ log(1,"**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log(1,"**************** vm end - end ******************\n"); break; }; } else { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); }; break; }; break; }; any help will be great Ram

2 1

[Serusers] Broken ser.cfg
by Jean-Michel Hiver 23 Aug '06

23 Aug '06

Can somebody please take a look at the routing logic below... I am trying to use exec_dset along with another program to do some LCR with SER but it doesn't work properly (here is a capture of what it produces: http://pastebin.ca/146771 Any ideas of what's doing wrong? Code: # -------------------------------------- route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (method=="REGISTER") { sl_send_reply("501", "Not Implemented"); break; }; if (method=="OPTIONS") { sl_send_reply("501", "Not Implemented"); break; }; record_route(); setflag(1); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method=="INVITE") { route(2); break(); }; route(1); break(); } route[1] { if (!t_relay()) { sl_reply_error(); }; } route[2] { # replaces the INVITE URI with the right host exec_dset ("/usr/local/Asterisk-LCR/bin/ser-lcr"); if (uri=~"^sip:STOP@127") { sl_send_reply("503", "Service Unavailable"); break; }; t_on_failure("2"); if(isflagset(1)) { append_branch(); }; setflag(1); t_relay(); } # /usr/local/Asterisk-LCR/bin/ser-lcr will give us the next uri to try # in case of failure, or send sip:STOP@127.0.0.1 if no more routes failure_route[2] { route(2); } # --------------------------------------

1 0

[Serusers] Need config help for PSTN call routing blocks to service provider.
by Chahn John Kim 23 Aug '06

23 Aug '06

Hi, I am new to SER and need experts' help in PSTN call routing block to call termination service provider in a scenario below: We have our own sip realm/server that will handle all VoIP calls among our own subscribers; sip.ours.com. This will forward all PSTN calls, authenticated and authorized by it, to sip.theirs.com, which will authorize usning our IP address and forward to pstngateway.theirs.com. What I have been given by SP is: sip.theirs.com, IP, myuserID and password. What they are given by me is our IP and hostaname. sip.ours.com and IP. My first try was simply; rewritehostport(sip.theirs.com:5060); t_relay(); ,which fails. Can someone tell me proper approach in ser commands based on the scenario above? Thank you so much in advance. John K

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sr-users August 2006