sr-users November 2005

sr-users@lists.kamailio.org

245 participants
571 discussions

[Serusers] Concurrent call limitation
by Marcello Lupo 15 Nov '05

15 Nov '05

Hi to all, any idea as to limit the concurrent calls outgoing to a particular PSTN gateway supposing to let ser becoming dialogs aware so to have the status of active open calls?? Thanks, Bye, Marcello

1 0

[Users] Usrloc question
by Darren Nay 15 Nov '05

15 Nov '05

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

1 0

[Serusers] Illegal redirection
by Joao Pereira 15 Nov '05

15 Nov '05

Hello I have two Cisco 7940 phones with private addresses (10.0.11.239 and 10.0.11.140) connected to SER also with private address (10.0.0.135), but in another network. My SER is with the default configuration. Between the networks I have a Checkpoint Firewall-1NG The Cisco IP phones can register because the REGISTER packets arent blocked. But the INVITEs never reach SER (I checked with ngrep), because the Firewall drops them, saying there was an illegal redirection. The most strange part, is that, when I try to make a phone call from PhoneA(10.0.11.239) to PhoneB(10.0.11.240), the INVITE is dropped before reaching SER, and it says "Illegal redirection 10.0.0.135->10.0.11.240". How can the firewall know that the INVITE was going to be redirected by SER to PhoneB(10.0.11.240) ???? my ser.cfg (the default one): # $Id: ser.cfg,v 1.25 2004/11/30 16:28:24 andrei Exp $ # simple quick-start config script # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) listen = 10.0.0.135 /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) # ------------------ module loading ---------------------------------- 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/textops.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic 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 >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # 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 the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; }

3 4

[Users] OpenSER Clients
by Gary Major 15 Nov '05

15 Nov '05

2 1

[Serusers] dynamic sip port for clients behind same firewall
by Imran Akbar 15 Nov '05

15 Nov '05

Hi, The problem I'm facing is that if I have multiple clients behind the same firewall, calls between them don't work as each is using the same sip port. The clients can be made to use a random port, but the registrar server has to listen on all ports and respond to the client using the port it was sent on... does SER support this? Is there any other way to solve this problem? Thanks, Imran

2 1

Re: [Users] Forwarding..
by Bogdan-Andrei Iancu 15 Nov '05

15 Nov '05

Hi Sebastian, not sure if you can do this - a kind of middle registrar; but you can try to experiment by using the latest UAC module which provides the FROM mangling and authentication features. http://www.openser.org/docs/modules/1.1.x/uac.html regards, bogdan BASTARD Sebastien wrote: > Ok, it seems to work...but I have another question: > I would like to have a static configuration on the PC on a LAN with a > local account test1(a)test.net <mailto:test1@test.net>. There is openser > running on a computer of this LAN which is managing domain test.net > <http://test.net>. > I would like to have an association of accounts on the openser server > (I know asterisk is able to do it): > SIP client connect to my local openser with account test1(a)test.net > <mailto:test1@test.net> > OPENSER associates test1(a)test.net <mailto:test1@test.net> with > test1(a)iptel.org <mailto:test1@iptel.org> and inform iptel.org > <http://iptel.org> that this account is managed locally: it's like > openser act as a client / iptel ser server. > Then I can be called from outside without problem. > > Do you think it's possible to do it or to implement a module for that. > > Thanks a lot for your help > Sebastien > > On 11/9/05, *BASTARD Sebastien* <sebastien.bastard(a)gmail.com > <mailto:sebastien.bastard@gmail.com>> wrote: > > Ok, i'm going to try. > > Thanks for your help. > > sebastien > > > On 11/8/05, *Bogdan-Andrei Iancu* < bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi, > > if I'm not wrong, you are talking about routing outbound requests > (requests which are not targeting your domain). If so, the > routing will > be done via DNS based on the domain name; you do not have to > specify > each of them. Just do like: > > if (!uri==myself) { > # this is an outbound call > t_relay(); > exit; > } > # process inbound calls > ........ > > regards, > bogdan > > BASTARD Sebastien wrote: > >> Hi, >> >> I would like to make run openser as a sip client when a client is >> connecting to it but with an unmanaged domain. I tried with the >> following configuration but with no result: >> >> if (!uri==myself) { >> # mark routing logic in request >> append_hf("P-hint: outbound\r\n"); >> # if you have some interdomain connections via TLS >> if(uri=~" iptel.org <http://iptel.org>") { >> forward("iptel.org <http://iptel.org>",5060); >> exit; >> } #else if(uri=~"@tls_domain2.net") { >> # t_relay_to_tls("IP_domain2","port_domain2"); >> # exit; >> #} >> route(1); >> }; >> >> Thanks for any help. >> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org <mailto:Users@openser.org> >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

1 0

[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

5 9

[Users] Does openser support fax ?
by Nguyen Duc Phi 15 Nov '05

15 Nov '05

Hi all, Does openser support fax? if it support How can I config openser ? Best Regards.

4 4

[Users] t_relay to multiple path
by Jayesh Nambiar 15 Nov '05

15 Nov '05

hi everyone, In my configuration, I trust some source ips and if call comes from these src ips, without any authentication they are t_relayed to a gateway ip address for pstn termination. Is there any potential problems by trusting some clients by src_ips, I do so as I dont want them to login from anywhere else. I also wanted to know that if I do not get reply from a particular gateway, can I forward them to another gateway. I dont want to use LCR for this. I thought of using failure route, but the call does not fail it keeps on sending invite. What I wanted to do is, if I dont get a reply within 5 seconds, it will try the next t_relay_to_udp statement. the logic goes something like this: if(allow_trusted()) { if(uri=="sip:011...") { t_relay_to_udp("to a pstn gateway"); # if no reply for 5 seconds t_relay_to_udp("to another gateway) break; }; }; can somebody pls help on this. I am currently using 0.9.5 and will soon be upgrading to 1.0.0. Thanks a lot in advance Jayesh --------------------------------- Enjoy this Diwali with Y! India Click here

3 2

[Users] what about mediaproxy module?
by messa＠innsof.com 15 Nov '05

15 Nov '05

Hi all I want to know how i can configure the NAT helper module to make call between 2 LANs separated by internet,or simply to do Nat Transversal? Can the mediaproxy module help me to do NAT transversal? Regards!

2 1

← Newer
1
...
32
33
34
35
36
37
38
...
58
Older →

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users November 2005