sr-users June 2008

sr-users@lists.kamailio.org

124 participants
230 discussions

[Serusers] registration to ser server fails

by Shrirang Yardi

Hello, I am a complete novice to VoIP and am trying to setup SER for the first time...so some of my questions may sound naive. I have installed ser-0.96 on x86 machine running fedora core 9. I followed the getting started guide to start ser using the hello-world.cfg config file. The service starts succesfully: Listening on udp: <ser_ip> [ser_ip]:5060 tcp: <ser_ip> [ser_ip]:5060 where ser_ip is the IP address of the server. I verified that the ser processes are actually running. I then used a UA from ekiga.net to register with the server, but the registration fails - reason: timeout. I dont have any problems pinging the ser_ip. What am I missing? Any help will be appreciated. Thanks, Shri

16 years, 5 months

[OpenSER-Users] Lost BYEs and Carriers

by David Villasmil

Hello all, Now I got a question: I understand that sometimes, when a UAC droppes out of the internet, a BYE message will be lost. But, if I were to use Openser as a Carrier switch, this is NOT supposed to happen, right? as all VoIP Carriers MUST be on the internet permanently. Any thoughts? David

16 years, 5 months

[OpenSER-Users] concurrent call limit per user / customer

by Patrick Miccio

Hi @ all, I was wondering if anyone of you guys already implemented a system where one could limit the maximum number of concurrent calls? With ISDN-terminaladapters or analog-telefon-adapters it is no problem because the hardware itself is the limiting device, but with the growing amount of IP PBXs the need becomes obvious. I am playing with the thought of writing something myself, but I am more a copy&paste programmer, so it will take quite some time :D Maybe someone here likes the idea? Here are my thoughts of how a module could provide a simple functionality: INVITE: check 200 OK: create Re-INVITE: update BYE: delete must provide: create_call() update_call() delete call() check_limit() create must save uuid,from,to,callid,timestamp in DB, from/to just for human readability ;) update must update timestamp with uuid/callid match ( we enforce 300sec re-invite call refresh ) delete must delete the DB entry with uuid/callid match check must count DB entries with uuid match and return amount of active calls. cheers, Patrick.

16 years, 5 months

[OpenSER-Users] Problem with OPENXCAP

by ADALA Asma

Hello everybody; I've installed *openser v1.3.2* and it works perfectly. Then, i''ve installed *openxcap v0.9.9. i'm using Uctimsclient as ims client. * When i try to upload or download an xml document using uctimsclient, i have this error in log messages : *error: XCAP root not found for request URI: http://127.0.0.1/xcap-root/ *My xcap root is: *127.0.0.1:8000/xcap-root* and i set the same in my client. *Can anyone help me please. thx *

16 years, 5 months

[OpenSER-Users] multi-leg accounting

by Ruchir Lists

Hi All, I'm trying to configure OpenSER with multi-leg accounting. I'm using OpenSER 1.2 & radius. I'm using acc_radius table for writing cdr records. I've searched through several articles and mailing list posts about configuring multi-leg accounting and everywhere they talk about setting up multi-leg-info parameter of acc module to configure leg source & destination and setting up accounting flag. But this is not working for me. I get only BYE record in table if I use this way to write multiple records for one call in call forwarding scenario. I managed to get multiple records by calling acc_rad_request on INVITE, ACK, BYE & CANCEL. For instance, I have 4 users; 90001, 90002, 90003, 90004. The call forwarding is setup as below: 90004 -> 90001 -> 90003. I dial 90004 from 90002 user and it forwards the call to 90001 & then 90003 and they're connected properlry. However I don't get leg source/destination properly and also I get 7-8 bye records for this call. Can anyone guide me in right direction if I'm doing anything wrong. Regards, Ruchir

16 years, 5 months

[OpenSER-Users] call functions with Perl (moduleFunction())

by Yazid Hadj Said

Hi, Please, i want to know why when i try to use remove_hf() and append_hf() functions with Perl (moduleFunction(func,string1,string2)), I have this message: Jun 12 11:03:53 [12838] ERROR:core:moduleFunc: Module function 'remove_hf' is unsafe. Call is refused. Jun 12 11:03:53 [12838] ERROR:core:XS_OpenSER__Message_moduleFunction: calling module function 'remove_hf' failed. Missing loadmodule? Jun 12 11:03:53 [12838] ERROR:core:moduleFunc: Module function 'append_hf' is unsafe. Call is refused. Jun 12 11:03:53 [12838] ERROR:core:XS_OpenSER__Message_moduleFunction: calling module function 'append_hf' failed. Missing loadmodule? Thanks, Yazid

16 years, 5 months

[OpenSER-Users] Dialog module, timeout_avp and record_route()

by kionez

Hi all, i'm trying to debug a strange behaviour of dialog mode. If I use "record_route" in my script, to force any message to toward the server, dialog module stops to check "timeout_avp", if I comment it out, when timeout_avp is reached, the dialog module notify me in sylog (as shown in log.txt) It seems that dialog module has some problems to work if i use record_route()... I try to use version 1.3.1 and 1.3.2, but i see the same behaviour. So, where i'm wrong? ;) Here's relevant part of config, and in openser.txt there's my routing script ---- loadmodule "rr.so" modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "dialog.so" modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "timeout_avp", "$avp(s:dlgtimeout)") modparam("dialog", "dlg_extra_hdrs", "Hint: credit expired\r\n") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_url", "mysql://dbuser:dbpass@localhost/openser") modparam("dialog", "db_mode", 1) ---- thanx in advance k. # With record_route() Jun 11 09:36:07 New request - M=INVITE RURI=sip:alice@192.168.27.74 F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.82 ID=2112176925 Jun 11 09:36:07 BRANCH - S=<null> R=sip:alice@192.168.1.201:5060;line=2c542c23af8e7c7 D=<null> F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.82 ID=2112176925 Jun 11 09:36:07 Reply - S=101 R=<null> D=Dialog Establishement F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=2112176925 Jun 11 09:36:07 Reply - S=100 R=<null> D=Trying F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=2112176925 Jun 11 09:36:07 Reply - S=180 R=<null> D=Ringing F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=2112176925 Jun 11 09:36:09 Reply - S=200 R=<null> D=OK F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=2112176925 Jun 11 09:36:09 New request - M=ACK RURI=sip:alice@192.168.1.201:5060 F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.82 ID=2112176925 Jun 11 09:36:26 New request - M=BYE RURI=sip:alice@192.168.1.201:5060 F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.82 ID=2112176925 Jun 11 09:36:26 Reply - S=200 R=<null> D=OK F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=2112176925 # Without record_route() Jun 11 09:36:49 New request - M=INVITE RURI=sip:alice@192.168.27.74 F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.82 ID=1675712119 Jun 11 09:36:49 BRANCH - S=<null> R=sip:alice@192.168.1.201:5060;line=2c542c23af8e7c7 D=<null> F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.82 ID=1675712119 Jun 11 09:36:49 Reply - S=100 R=<null> D=Trying F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=1675712119 Jun 11 09:36:49 Reply - S=101 R=<null> D=Dialog Establishement F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=1675712119 Jun 11 09:36:49 Reply - S=180 R=<null> D=Ringing F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=1675712119 Jun 11 09:36:51 Reply - S=200 R=<null> D=OK F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=1675712119 Jun 11 09:36:56 WARNING:dialog:dlg_ontimeout: timeout for dlg with CallID '1675712119' and tags '53233056' '453709088' ####### Routing Logic ######## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } t_on_reply("1"); t_on_failure("1"); } if(!is_method("REGISTER")) { #record_route(); } if(is_method("CANCEL") || is_method("ACK")) { t_on_reply("1"); if(t_check_trans()) { if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } t_check_trans(); if(is_method("REGISTER")) { sl_send_reply("100", "Trying"); if(!save("location")) { xlog("L_ERR", "Register FAIL - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Register OK - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } if(is_method("INVITE")) { $avp(s:dlgtimeout)=5; $avp(s:ringtimeout)=10; setflag(4); sl_send_reply("100", "Trying"); } if(is_method("BYE")) { route(1); } if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } route(1); } route[1] { if (is_method("INVITE")) { t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } branch_route[1] { xlog("L_INFO", "BRANCH - S=$rs R=$ru D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); } onreply_route[1] { xlog("L_INFO", "Reply - S=$rs R=$ru D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); } failure_route[1] { if (t_was_cancelled()) { exit; } }

16 years, 5 months

[OpenSER-Users] stun and symmetric

by Pezhman Lali

Dear, I find that stun and symmetric nat can not work together , but some big sip providers like sipgate or fwd, added stun settings to their ata as default without, any question about customer's nat type. where is the key ?

16 years, 5 months

Re: [Serusers] stun and s

by Pezhman Lali

> well, I would hope, if it's their ATA, they probably > know that it > doesn't do symmetric NAT.... How ??? can ATAs behind router or dsl modem, select nat type ? --- On Thu, 6/12/08, Harry Behrens <hb(a)4snewcom.de> wrote: > From: Harry Behrens <hb(a)4snewcom.de> > Subject: Re: [Serusers] stun and s > To: pezhman_lali(a)yahoo.com > Cc: serusers(a)lists.iptel.org > Date: Thursday, June 12, 2008, 11:49 AM > Pezhman Lali schrieb: > > Dear, > > I find that stun and symmetric nat can not work > together , > > but some big sip providers like sipgate or fwd, added > stun settings to their ata as default without, any question > about customer's nat type. > > > > > > well, I would hope, if it's their ATA, they probably > know that it > doesn't do symmetric NAT.... How ??? > > > where is the key ? > > > > > > > open? > > -h > > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > >

16 years, 5 months

[Serusers] stun and s

by Pezhman Lali

16 years, 5 months

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sr-users June 2008