sr-users October 2012

sr-users@lists.kamailio.org

87 participants
165 discussions

How can I directly access the domain of additional branches??
by Stephen Dodge (Bistech) 11 Oct '12

11 Oct '12

Hello, Could someone advise if it is possible to directly access the domain of additional branches as $rd references the domain of the main URI only. If it's not possible does anyone have any tips on how I can achieve this. Many Thanks. Steve. ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

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Kamailio direct interconnectivity with PRI
by SamyGo 11 Oct '12

11 Oct '12

Hello, I've a scenario in which I've to deploy a couple Sangoma PRI cards with kamailio. What I wish is that I've some drivers for this purpose and so I don't ned to install FreeSWITCH or Asterisk in between the PRIs and Kamailio. Kindly give any feedback on what are the possibilities and options. Thanks, Sammy

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Get sip clients to work with sip:user@domain and ldap
by Christophe ROY 11 Oct '12

11 Oct '12

Hello I'm quite new to SIP and TOIP so I may not understand the concepts correctly. I'm supposed to make work SIP clients (for now, linphone on android and windows) with a Kamailio 3.2 server and ldap authentication. Ldap authentication seems to be working fine, as the login/pass is validated against the Active Directory before registering the SIP client. However, Linphone seems to register the SIP client as sip:user@true_ip_address instead of sip:user@domain. If I try to make a call to sip:anotheruser@domain it doesn't work, and I can find in /var/log/syslog the following lines: Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) However, if I try to make a direct call to sip:anotheruser@his_true_IP, it works How can I make the registered users to be reachable with user@domain with Kamailio? And by the way, is there a way to get a list of registered users? Thank you for the help Christophe

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kamailio 3.3.1 - uac - remote register - init delay possible?
by René Bormann 11 Oct '12

11 Oct '12

Hello @all, want to use the remote register function of the UAC module. We need to register 42 accounts on one openSER server. At kamailio startup those 42 remote register requests will be startet at the same time. This leads to a kind of FlodAttact on the openSER server of our provider, which will block us :-( We would like to delay each initial remote register by 1 second. We searched the UAC module and taught that at some place the uac.so module has to load the uacreg table and do some for loop to start the remote register process. We found function child_init() in uac.c where the function uac_reg_load_db() is called followed uac_reg_timer(). Our change to uac_reg_timer() is the sleep(1). After compiling it seems to us - that this function is not called. Is there anybody how has a tip for us? Thanks, René /** * */ void uac_reg_timer(unsigned int ticks, unsigned int initial_reg_call) { LM_ERR("----------------------- uac_reg_timer() -- enter \n"); <------>int i; <------>reg_item_t *it = NULL; <------>time_t tn; <------>tn = time(NULL); <------>for(i=0; i<_reg_htable->htsize; i++) <------>{ <------><------>/* free entries */ <------><------>it = _reg_htable->entries[i].byuuid; <------><------>while(it) <------><------>{ LM_ERR("----------------------- uac_reg_timer() -- 1 \n"); <------><------><------>if (initial_reg_call == 1) <------><------><------>{ LM_ERR("----------------------- uac_reg_timer() -- 2 \n"); <------><------><------><------>sleep(1); <------><------><------>} <------><------><------>uac_reg_update(it->r, tn); <------><------><------>it = it->next; <------><------>} <------>} }

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MSRP Config
by Ed Brady 10 Oct '12

10 Oct '12

Hi, I am working on an MSRP project with Kamailio. I have Kamailio installed and configured for SIP with several test accounts created, but am unable to send MSRP messages so far. I am using Blink as a test client, though the final intentions of the project will be to use iDoubs and IMSDroid for client connections. We are having these developed to our needs right now though, so I must use another client to begin testing. I have been unable to find good documentation on the initial configuration of MSRP in Kamailio. I have seen the module documentation, but am unclear how to utilize the information there to configure Kamailio properly. Is there any documentation on getting started with a basic MSRP relay setup so that two clients can chat and send files? I am not new to SIP, but this is my first project with MSRP. Thanks, Ed

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OpenSER/Kamailio Support? - Forwarding SIP INVITE Packets
by Eddie Elric 10 Oct '12

10 Oct '12

Hello fellow OpenSER/Kamailio users, I hope this is the right method of contact - I was unable to find a forum on which to post any queries. In essence, I've set up a Kamailio server of the latest version - logging on to clients works fine, etc. The idea of my setting up was for forwarding SIP messages to our test server, that's where I got stuck. The following link explains everything on Kamailio it seems - http://www.kamailio.org/wiki/cookbooks/3.3.x/core#forward So, I've added to the kamailio.cfg the following - forward("10.0.0.10","5060"); But you see, I'm not sure where I should have added it. I've tried adding to a variety of places, such as - route[SIPOUT] and onreply_route[MANAGE_REPLY] - but none are working. Cheers guys/girls. Best, Eddie

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Kamailio/RTP Proxy issue
by Kamal Palei 10 Oct '12

10 Oct '12

Hi All Facing a strange issue in Kamailio / RTP proxy setup. Using 1. Kamailio - 3.1.5 2. RTP proxy 1.1 I have pne sip client, lets say it is 'A'. 'A' sent INVITE to Kamailio proxy. Kamailio proxy modified the IP address present in SDP to RTP proxy IP address, and sent to main proxy. Main proxy did respond back with 200 OK to INVITE. In 200 OK SDP, Main proxy put the IP address of Media server (media server ip address is different than main proxy server ip address). After the call establishment, it is expected that RTP proxy should send media packets to media server, instead it is sending to proxy server. Pleae do let know, how to put a work around to avoid this problem. Thanks Kamal NECS, Bangalore

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TCP replace alias can cause core
by Jijo 10 Oct '12

10 Oct '12

We found a problem regarding TCP connection alias in the following code at tcp_main.c: int tcpconn_finish_connect( struct tcp_connection* c, union sockaddr_union* from) { : : /* remove all the aliases except the first one and re-add them * (there shouldn't be more then the 3 default aliases at this * stage) */ for (r=1; r<c->aliases; r++){ a=&c->con_aliases[r]; tcpconn_listrm(tcpconn_aliases_hash[a->hash], a, next, prev); } c->aliases=1; As TCP_ALIAS_REPLACE flag is set for the default TCP options value, in the function _tcpconn_add_alias_unsafe() a TCP connection alias can be moved from connection A to connection B based on the TCP alias hash. In this case, the number of aliases is incremented in the connection A, and decremented from connection B. However, in the connection B the number of aliases can reach zero (no alias). And the code above can be executed for connection B setting the number of aliases to 1 unconditionally. When this case happens, the connection B keeps an invalid alias (already excluded from connection B by tcpconn_add_alias_unsafe() function called from connection A). When the connection A is released, the aliases are also released, and this memory area can be filled with different data. As connection B has references to an invalid alias it can try to access invalid areas, and can crash Kamailio. This access happens, for example, when another alias is added to connection B. To fix it we include a check before the code: if (c->aliases>0) { for (r=1; r<c->aliases; r++){ a=&c->con_aliases[r]; tcpconn_listrm(tcpconn_aliases_hash[a->hash], a, next, prev); memset(a,0xbb,sizeof(struct tcp_conn_alias)); } c->aliases=1; } Please let us know if any comments. Thanks Jijo

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[snmpstats] openserSIPRegRejectedRegistrations not being updated
by Anatoly Ivanov 10 Oct '12

10 Oct '12

Hi, I am observing weird behavior of snmpstats.so - I see openserSIPRegAcceptedRegistrations being updated, but openserSIPRegRejectedRegistrations stays at zero. Could somebody please explain under what conditions this counter is being incremented? I am using auth_radius, so I suspect I simply do not reject failed registartions the 'proper' way, thus causing the counter to stay at 0. Thank you, Anatoly.

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how to use fix_nated_sdp("2") with multiple media types
by MARINA SERRANO MONTES 10 Oct '12

10 Oct '12

Hi, I have a doubt about how to use "fix_nated_sdp("2")" function when we have an scenario with multiple media types (audio and video) and the param "c" in sdp for each media type has a different IP. We are using a SCOPIA 400 MCU to multi-party in a call, and it will set up different IP to send RTP depend on media type. I have tested this function, but it rewrites always all the "c" parameter with the same value. Is there any option to use the function or to rewrite "c" value for each media type independently? Many thanks. Br, Marina ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at: http://www.tid.es/ES/PAGINAS/disclaimer.aspx

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sr-users October 2012