- sr-users - kamailio.org

[Serusers] Ser2.1.0.
by Aleš Podskalský 23 Dec '09

23 Dec '09

Hi all, I tried to install the newest snapshot of the Ser (version 2.1.0+cvs20091126). Server has been instaled correctly, but I can't add new user to the database. I noticed that serctl was changed with new tool called sercmd. With this tool I should be able to create new user using appropriate command, but I can't find some help for it. So my question is, if this version is unable for normal use by then, or if I can fill user data in some another way and how. Thank a lot. Ales P.

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[Kamailio-Users] adding inbound and outbound routes seperately
by Rodney 23 Dec '09

23 Dec '09

Ok, I have kamailio and siremis, and I need to know how to set up a separate in bound and outbound route. Peering with inbound and registering with outbound Any ideas guys? R

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Re: [Kamailio-Users] PCRE Regex / Transformations Bug? (Daniel-Constantin Mierla)
by Brandon Armstead 23 Dec '09

23 Dec '09

Inaki: That is what I was looking for and it does appear you are correct! Daniel, I could see that being as a useful addition, do you imagine there being much involved to implement such a method? Thank you guys for all of your help! Sincerely, Brandon Armstead

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[Kamailio-Users] Question about Contact in NAT
by Ricardo Martinez 22 Dec '09

22 Dec '09

Hello list. I know that probably this questions is more related to the SIP protocol, but is causing me problems and this is a very supportive list. I'm using a HT-286 from Grandstream behind NAT. This is the REGISTER sequence : U 200.120.46.99:29786 -> 200.100.154.35:5060 REGISTER sip:myproxy.mydomain.net SIP/2.0. Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKafcb21cb60db91bb. From: <sip:5501234567@myproxy.mydomain.net;user=phone>;tag=478555f337ca8f10. To: <sip:5501234567@myproxy.mydomain.net;user=phone>. Contact: *. Supported: replaces, timer. Call-ID: a51e541e3b0f66fd(a)192.168.1.100. CSeq: 100 REGISTER. Expires: 0. User-Agent: Grandstream HT287 1.1.0.42 DevId 000b822197fe. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE. Content-Length: 0. . U 200.100.154.35:5060 -> 200.120.46.99:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKafcb21cb60db91bb;received=200.120.46.99. From: <sip:5501234567@myproxy.mydomain.net;user=phone>;tag=478555f337ca8f10. To: <sip:5501234567@myproxy.mydomain.net;user=phone>. Call-ID: a51e541e3b0f66fd(a)192.168.1.100. CSeq: 100 REGISTER. Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 200.100.154.35:5060 "Noisy feedback tells: pid=29777 req_src_ip=200.120.46.99 req_src_port=29786 in_uri=sip:myproxy.mydomain.net out_uri=sip:myproxy.mydomain.net via_cnt==1". . U 200.100.154.35:5060 -> 200.120.46.99:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKafcb21cb60db91bb;received=200.120.46.99. From: <sip:5501234567@myproxy.mydomain.net;user=phone>;tag=478555f337ca8f10. To: <sip:5501234567@myproxy.mydomain.net;user=phone>;tag=6eb14299de2cc731d17 3c135064daec5.1fe9. Call-ID: a51e541e3b0f66fd(a)192.168.1.100. CSeq: 100 REGISTER. WWW-Authenticate: Digest realm="myproxy.mydomain.net", nonce="4b30e9281c52f640a2713c3011bb835d06005946", qop="auth". Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 200.100.154.35:5060 "Noisy feedback tells: pid=29777 req_src_ip=200.120.46.99 req_src_port=29786 in_uri=sip:myproxy.mydomain.net out_uri=sip:myproxy.mydomain.net via_cnt==1". As you can see the initial REGISTER has a Contact header like this : Contact: *. Is this ok?. Is this ok by RFC? What does this means? In my cfg file I have : route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; So I think is not matching the regexp "(!search("^Contact:\ +\*")" therefore is not fixing the contact. Any ideas? Ricardo.-

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[Kamailio-Users] append_hf
by Corey Edwards 22 Dec '09

22 Dec '09

I'm upgrading from OpenSER 1.2.2 to Kamailio 1.5.3 and I've run into a problem with append_hf. It seems that my header only gets appended once in a while. Here's the shortened version of what I'm doing: failure_route[1]{ if (t_check_status("(486)|(600)")){ # busy handler route(15); } } route[15]{ # look up forwarding number .... if (does_uri_exist()){ append_hf("Diversion: <sip:$var(orig_user)@$Ri>;user-id=\"$var(orig_user)@$Ri\";reason=busy\r\n"); t_relay(); } } Unfortunately, the Diversion header doesn't get set every time. I can't see any difference in the logs from when it does get set and when it doesn't. I added an xlog right above append_hf and confirmed that append_hf runs each time. On a whim I added a branch_route and called it using t_on_branch. I then added duplicate code to call append_hf. I now seem to be getting more instances of Diversion being set (twice!), but still not every time. Is there something obvious I'm missing here? Any way I can ensure those headers get set every time? Corey

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[Kamailio-Users] listing users that were added
by Rafi Gordon 22 Dec '09

22 Dec '09

Hello, I had added successfully several users by : kamctl add xxx yyy How can I list all users that exist in the DB? Rgs, Rafi

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[Kamailio-Users] Question about AVP in route()
by Ricardo Martinez 21 Dec '09

21 Dec '09

Hello. I have the next situation. In my config file I have something like this.. route { ... If INVITE route(3); } ... route[1] { Route(39); t_relay() } route[3] { ... different check for the INVITE radius_load_caller_avps("$fU@$fd") (I load the next AVPS from the radius server : Attributes: SIP-AVP = "tra:sip:0217001234567@10.0.0.211:5060" SIP-AVP = "channels:1") is_avp_set("$avp(s:channels)/n") <-- this AVP seem to be load and is set OK. route(10); } route[10] { rewritehost("myproxy.mydomain.com"); route(1); } route[39] { is_avp_set("$avp(s:channels)/n") } The last check in the route(39) is not working!. Why? Ricardo Martinez Ingeniero de Desarrollo VOISSNET S.A.

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[Kamailio-Users] Presence_Dialoginfo versioning
by kamailio.org＠spam.lublink.net 21 Dec '09

21 Dec '09

Hey, I have noticed that the XML body of the dialog messages contains a version attribute. The server is counting the versions using the latest subscription as a reference point, and the phone is counting the versions from the first subscription ( at reboot ). Which is the correct way to count these versions? Consider : 10:00 Subscription 10:05 Notification version 1 10:35 Notification version 2 11:01 Subscription 11:05 Notification version X Is X = 3 because it is the third notification or 1 because it is the first after the last subscription? If it's version 1, it could confuse the phone cause a notification that is sent at the same time as the notification would have a confusing version. On the other hand, each subscription, has it's own versioning, so would it not logically follow that the different subscriptions for the same device have seperate versioning? From my phone : BLF Notify received for line: 3 has older version: 3 last version:135 To confirm my theory, I redialed the number 133 times, and once the version number had run up to 136, the lights started flashing again. Is this a bug in Grandstream, Kamailio, or perhaps an RFC which was unclear? Thanks, David

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[Kamailio-Users] Clients with late offer-answer
by Klaus Darilion 21 Dec '09

21 Dec '09

Hi! I want to test rtpproxy with late offer-answer (SDP in 200-ok and ACK). Thus, I'm looking for a SIP client which support that. Do you know any? thanks klaus

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[Kamailio-Users] add Allow header to sl_send_reply
by Andreas Heise 19 Dec '09

19 Dec '09

Hello, if I create a response with sl_send_reply then the message contains not an Allow header. How could I add an Allow header to this responses? thanks in advance, Andreas

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[Kamailio-Users] log not showing when using uac_replace_from( ) function
by Robert R 19 Dec '09

19 Dec '09

Hi, I simply trying to modify URI and FROM headers as follow. tcpdump packet capture shows the values are changed for both URI and FROM headers. However, the syslog only shows the new value for URI and not for FROM header. How can I verify new value in FROM header in the log file? # main request routing logic route { $rU = avp(i:1); $rd = avp(i:2); .... uac_replace_from("$avp(i:3)"); xlog ("fu after perl: $fu"); xlog ("ru after perl: $ru"); ... } Thanks, Rob

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[Kamailio-Users] Transparent bridge mode
by Sungjoo Kim 17 Dec '09

17 Dec '09

Hello All, I'm trying to setup a test scenario, where i have Kamailio and rtpproxy running on one CentOS box(server1) and asterisk running on another CentOS box(server2). Server1 has 2 NIC's installed, eth0 connected to the WAN and eth1 connected to Server2 in a bridge mode. I will be modifying rtpproxy to intercept data flowing to Server2 and process them before i let them continue to flow along their original path. Server2 has to have a Public IP. For this i am having to use 4 Public IP's- eth0,eth1 and bridge br0 on Server1 and eth0 on Server2. Is there an alternate way of doing this ? Do you suspect there will be a problem with setting things up this way while trying to use rtpproxy to intercept data. Thanks and regards, Vikram.

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[Kamailio-Users] Kamailio + CSQL
by Ramu 17 Dec '09

17 Dec '09

Hi All, Did anybody tried to setup Kamailio with CSQL database. Please post your experiences using CSQL and what are parameters we need to change in modules to connect CSQL DB Server. Thanks, Ramu

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[Kamailio-Users] Kamailio and RTPProxy
by Bruce McAlister 16 Dec '09

16 Dec '09

Hi All, I am trying to setup kamailio 1.5.3-notls for use with rtpproxy 1.2.1. I have a partially working setup, however, when I have a softphone configured behind nat it appears that rtpproxy is using the internal ip (prior to nat) to setup the sdp ports. I have rtpproxy configured in bridge mode as follows: rtpproxy -f -F -l public_ip/internal_ip -s udp:*:11211 I have kamailio's nathelper module setup as follows: modparam("nathelper", "rtpproxy_sock", "udp:internal_ip:11211") I then use a softphone (x-lite) behind a nat firewall (ip = 192.168.11.2) to establish a call, and the log of rtpproxy is as follows: INFO:handle_command: new session ZmRkNjRlNjM0NzExZjQ5MWUxYjllOTFhZWFmZGZkMjY., tag 8330585e;1 requested, type strong INFO:handle_command: new session on a port 52572 created, tag 8330585e;1 INFO:handle_command: pre-filling caller's address with 192.168.11.2:35498 INFO:handle_command: lookup on ports 52572/40380, session timer restarted INFO:handle_command: pre-filling callee's address with internal_ip:11250 INFO:handle_delete: forcefully deleting session 1 on ports 52572/40380 INFO:remove_session: RTP stats: 906 in from callee, 0 in from caller, 906 relayed, 0 dropped INFO:remove_session: RTCP stats: 3 in from callee, 0 in from caller, 3 relayed, 0 dropped INFO:remove_session: session on ports 52572/40380 is cleaned up In the log above it looks like rtpproxy is using 192.168.11.2:35498 as the remote ip/port combination to contact for the sdp session of the x-lite softphone. I would have thought that it would use the IP address that initial request was received from to establish this connection (via the fix_nated_contact() function). My route logic in kamailio is a slightly modified default configuration that is installed with kamailio : route{ . . . # NAT detection route(4); . . . route(1); } route[1]{ if (check_route_param("nat=yes")) { setbflag(6); } # if (isflagset(5) || isbflagset(6)) { route(5); # } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } Would the "fix_nated_contact" function in the "nat detection" route not fix the contact before passing over the request to rtpproxy to establish the sdp session? route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ # force_rtp_proxy(); if(src_ip==internal_ip_of_media_server) force_rtp_proxy("ocfaei"); else force_rtp_proxy("ocfaie"); } if (!has_totag()) add_rr_param(";nat=yes"); return; } onreply_route[1] { xdbg("incoming reply\n"); # if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { if (status=~"(183)|(2[0-9][0-9])") { # force_rtp_proxy(); if(src_ip == internal_ip_of_media_server) force_rtp_proxy("ocfaei"); else force_rtp_proxy("ocfaie"); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } } Any pointers would be greatly appreciated. Thanks Bruce

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[Kamailio-Users] Problem
by Ricardo Martinez 15 Dec '09

15 Dec '09

Hello. What this error means? Dec 14 15:20:19 kamailio: ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/nat_traversal.so>: /usr/local/lib/kamailio/modules_k/nat_traversal.so: undefined symbol: register_module_stats Dec 14 15:20:19 kamailio: : <core> [cfg.y:3210]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 48, column 12-29: failed to load module I have : # ------------------ module loading ---------------------------------- #set module path mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/module s/" # Uncomment this if you want to use SQL database loadmodule "db_mysql.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "siputils.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "domain.so" loadmodule "permissions.so" loadmodule "group.so" loadmodule "mi_fifo.so" loadmodule "avpops.so" loadmodule "xlog.so" loadmodule "dialog.so" loadmodule "nathelper.so" loadmodule "nat_traversal.so" loadmodule "mediaproxy.so" I'm using Kamailio version: kamailio 3.0.0-rc2 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. @(#) $Id$ main.c compiled on 18:09:39 Dec 9 2009 with gcc 4.3.2 In the directory /usr/local/lib/kamailio/modules_K/ the file nat_traversal.so exists : [root@ nat_traversal]# ls /usr/local/lib/kamailio/modules_k/ acc.so cfgutils.so drouting.so mi_datagram.so permissions.so registrar.so speeddial.so uac.so alias_db.so db_text.so exec.so mi_fifo.so pike.so rr.so sqlops.so uri_db.so auth_db.so dialog.so group.so msilo.so presence_mwi.so rtimer.so sst.so userblacklist.so auth_diameter.so dispatcher.so htable.so nathelper.so pua_mi.so siptrace.so statistics.so usrloc.so auth.so diversion.so imc.so nat_traversal.so pv.so siputils.so textops.so xlog.so benchmark.so domainpolicy.so kex.so path.so qos.so sl.so tmx.so call_control.so domain.so maxfwd.so pdt.so ratelimit.so sms.so uac_redirect.so Can someone help me? Regards, Ricardo

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Re: [Kamailio-Users] kamailio load problem
by Daniel-Constantin Mierla 15 Dec '09

15 Dec '09

Hello, cc-ed back the list as others may look for same issues in the future ... On 12/15/09 11:30 AM, alex pappas wrote: > Daniel hi, > > In my config I don't have any authentication. I also configured in > mysql the log-slow-query with long_query_time = 2 > In my kamailio.cfg is a lot of ACC loging: > > Log acc to a file (extra fields) > Log to mysql with axtra fields(I disabled this for the moment because > was causing high load) > Log to a Radius (the radius is in the same machine and as far as I > know the radius after i swriting his data to a mysql also) I diden't > know that we have radius installed localy > > From the log-slow-query now I see a message about sip_trace query wich > obiesly took time more than 2 sec. Now I know that I'm doing also > sip_trace inserts in mysql sip_trace is writing big amount of data, this can make mysql do a lot of work. Try without it and see the difference. Also, doing acc to syslog brings some penalty if syslog is not configured in asyncronous mode (see my previous email). Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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Re: [Kamailio-Users] kamailio load problem
by Daniel-Constantin Mierla 15 Dec '09

15 Dec '09

Hello, cc-ed back the list as others may look for same issues in the future ... On 12/15/09 11:30 AM, alex pappas wrote: > Daniel hi, > > In my config I don't have any authentication. I also configured in > mysql the log-slow-query with long_query_time = 2 > In my kamailio.cfg is a lot of ACC loging: > > Log acc to a file (extra fields) > Log to mysql with axtra fields(I disabled this for the moment because > was causing high load) > Log to a Radius (the radius is in the same machine and as far as I > know the radius after i swriting his data to a mysql also) I diden't > know that we have radius installed localy > > From the log-slow-query now I see a message about sip_trace query wich > obiesly took time more than 2 sec. Now I know that I'm doing also > sip_trace inserts in mysql sip_trace is writing big amount of data, this can make mysql do a lot of work. Try without it and see the difference. Also, doing acc to syslog brings some penalty if syslog is not configured in asyncronous mode (see my previous email). Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] debian packages for kamailio 3.0.0-rc3
by Daniel-Constantin Mierla 15 Dec '09

15 Dec '09

Hello, I have uploaded debian packages for stable version: http://www.kamailio.org/pub/kamailio/3.0.0-rc3/packages/debian-lenny/ If you want to create debs by yourself, get Kamailio 3.0 branch from GIT, see here: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git And then run: make deb Testing and feedback is very much appreciated. Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] kamailio load problem
by alex pappas 15 Dec '09

15 Dec '09

Dear users, I'm using kamailio in the following scenario: 1. A call comes in and is checked against PDT module. From this module depend from the prefix I'm getting back the domain which is used as the dialplan group. 2. After that the call is checked against Dialplan module using the previous value from PDT as Dialplan dpid. From the Dialplan module i get back the the attributes values. 3. these attributes values are user in the dispathcher module where I continue with failover scenarios. Also the script is working with a radius for accounting purposes. This scenario works perfecttly and it is still in a testing mode. An the problem: I'm running SIPP against this scenario and until the CPS is 5 everything looks good. The scenario for SIPP is the simplest. uac and uas default. Uac is sending a call and uas is answering. When the CPS goes above 9 then I have call failures. All the statistics are going down. I'm cheking for CPU load or other system problems but I don't see anything strange. Does anyone understand why this is happening? I'm attachng the 2 routes that i'm using: test values in order to understand the script *PDT value = 1 (used for the dialplan group) Dialplan value = 100.101.102.# Dispatcher values: 100 = 10.0.0.5 101 = 10.0.0.6 102 = 10.0.0.7* Thank you in advance Alex route[7] { if(prefix2domain("2", "0")) { $var(dial_grp) = $(rd{s.int}); if(dp_translate("$var(dial_ grp)", "$rU/$rU")) { $var(i) = 0; while($(avp(s:dest){s.select,$var(i),.})!="#") { $avp(s:dstgrp) = $(avp(s:dest){s.select,$var(i),.}{s.int}); $var(i) = $var(i) + 1; } $avp(s:user) = $rU; if(ds_select_domain("$avp(s:dstgrp)", "4")) { if($(ru{uri.param,prefix})!=null) { $ru = "sip:" + $(ru{uri.param,prefix}) + $rU + "@" + $rd; } else { $ru = "sip:" + $rU + "@" + $rd; } $avp(s:dstgrp) = null; t_on_failure("2"); t_relay(); exit; } } sl_send_reply("404", "No route found"); exit; } } failure_route[2] { if(t_was_cancelled()) { exit; } if(t_check_status("4[0-9][0-9]|5[0-9][0-9]")) { if(ds_next_domain()) { if($(ru{uri.param,prefix})!=null) { $ru = "sip:" + $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; } else { $ru = "sip:" + $avp(s:user) + "@" + $rd; } t_on_failure("2"); append_branch(); t_relay(); exit; } else { if($avp(s:dstgrp)!=null) { if(ds_select_domain("$avp(s:dstgrp)", "4")) { if($(ru{uri.param,prefix})!=null) { $ru = "sip:" + $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; } else { $ru = "sip:" + $avp(s:user) + "@" + $rd; } $avp(s:dstgrp) = null; t_on_failure("2"); append_branch(); t_relay(); exit; } } else { t_reply("444", "No more tries for you!"); } } }

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[Kamailio-Users] Reboot phones
by David 14 Dec '09

14 Dec '09

Hey, I have noticed that when I try to use kamctl to reboot phones, it comes back with simply "400". I tried using the fifo interface and the phone asked the server to identify itself using the phone's username and password. How do people generally use the kamctl to reboot phones? Thanks, David

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[Kamailio-Users] Kamailio 3.0.0-RC3 Released
by Daniel-Constantin Mierla 14 Dec '09

14 Dec '09

Kamailio v3.0.0-RC3 is out - the tarball with sources is available at: http://www.kamailio.org/pub/kamailio/3.0.0-rc3/src/kamailio-3.0.0-rc3_src.t… Lot of fixes to packaging, code and documentation have been committed since RC1, Kamailio 3.0.0-RC3 becoming ready for pre-production phase. If you like to work with sources, then use the tutorial guiding the installation of Kamailio 3.0 branch from GIT repository: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git Two wiki pages were created to collect what is new in Kamailio 3.0 and how to migrate to this version: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 Feel free to contribute to wiki pages, helping to build migration tutorials. This is the last RC before full 3.0.0 relase. Please report any issue you find to sr-dev(a)lists.sip-router.org Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] preparing RC3 for kamailio 3.0
by Daniel-Constantin Mierla 14 Dec '09

14 Dec '09

Hello, I plan to release Kamailio 3.0, hopefully today, if no outstanding issue out there. Ending of last week, I managed to get debian packaging working (tested on Ubuntu) using Andrei's new Makefile system, with module groups. If you are aware of issues you want fixed before 3.0RC3, drop an email. This is supposed to be latest RC before full 3.0.0 release. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] Bflags
by David 14 Dec '09

14 Dec '09

Hey, I am having some issues with NAT and I think the issue is with Bflags. So I have some clarifications : 1. What does the flags column in location contain? 2. What does the cflags column in location contain ? I am guessing that cflags represents the bflags? So if I have a value of 192, does that mean flags 7 and 8 are set ? This confuses me cause I set flags 6 and 7. Do the bflags appear magically in the routing scripts or is there a command that I have to execute for the script to have the bflags? Thanks! David

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What to download?
by Antonio Goméz Soto 13 Dec '09

13 Dec '09

Hi, I'm a longtime asterisk user, needing to scale up. I figured it would be smart to use and develop for sip-router, because it'll come out soon. But I've never used one of ser, openser, or siblings, and now I'm confused by the download links on you site. I see kamailio_3.0, master, sr_3.0, stable and unstable top level directories. What should I download? What's the diference between them? Thanks, Antonio

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[Serusers] Warning 392 : Noisy Feedback Tells
by Abdul Khadar 11 Dec '09

11 Dec '09

Hi, I need help for registering with iptel.org I have created one user called hitechabdul with iptel.org When I am trying to register with this domain, It is returing error *401 Unauthorized *with *Warning 392 : Noisy Feedback Tells* please find attached ethereal packet and help me to resolve this problem. The details filling with register message : username : hitechabdul password : xxxxxx Domain Name : iptel.org Other than this Is I have to given any more data ? -- thanks & regards, Abdul Khadar Shaik,

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[Kamailio-Users] Rewrite $tU
by David 11 Dec '09

11 Dec '09

Hey, I need to rewrite the $tU variable on a packet before it is relayed. It is ok if the change is only on the relayed packet and the $tU variable change. The important thing is that I modify the To header, it seems that rewritting $tU would be the easiest. I know that user agents do not like this changing, but I am changing it on an intermediary server and when the reply comes back, I change it back to the original value. Thanks! David

5 6

Case sensitivity in usrloc "location" table
by Bruce McAlister 10 Dec '09

10 Dec '09

Hi All, I am trying to integrate Kamailio (1.5.2-notls) into our existing database structure using a view for the subscriber table and using the location table to store the registration information. I am successfully able to register a phone using this method. One of our requirements is to set an "outboundproxy" and "outboundproxyport" parameter in the location table for a particular sip account (username). I am trying to do this with the sqlops module. What I have come across though is that when the registration information is saved to the location table the username is stored in lower case. So when my sqlops query is run against the username it does not match. Is there a way to tell the usrloc module to store the username in the same case as it is received? For example, in the subscriber table I have view defined that for my user account returns the following information: ------------- select * from subscriber where username = 'BruceNW9440'; username | domain | password | ha1 | ha1b -------------+---------------+------------+----------------------------------+---------------------------------- BruceNW9440 | 10.10.10.10 | 00minger00 | 1f2d688f65be04a61cab36a1fb3f426a | 6529d45f5c418daf35be7b3dd4a858cb (1 row) ------------- Now, if I check the above against the location table: -------------- select * from location where username = 'BruceNW9440'; 0 rows returned -------------- If I run the same query using "brucenw9440" as the username (lowercase) then the correct information is returned. I have the following log line in my kamailio configuration script: xlog("L_INFO","Username: $au, OutboundProxy: $Ri, OutboundProxyPort: $Rp"); The details of this log appear correct when it is logged: Dec 9 14:21:03 ibis /opt/kamailio/sbin/kamailio[16972]: [ID 197553 local0.info] Username: BruceNW9440, OutboundProxy: 10.10.10.10, OutboundProxyPort: 5060 However, when my sqlopt query is run it does not find a match because of the incorrect case: sql_query("bfl", "UPDATE location SET outboundproxy = '$Ri', outboundproxyport = '$Rp' WHERE username = '$au'", "res"); I've had a read through the usrloc module documentation and I cannot seem to find a reference to any options that will preserve the case. I also cannot find if there is a parameter to save('location') that will preserve the case. I cant seem to find a reference to the save() function on the wiki. I must be missing something small. We have some sip usernames that are case sensitive so any help in being able to resolve this would be greatly appreciated. Thanks Bruce

4 9

[Kamailio-Users] Question regarding TM module DNS failover with SRV records
by Nathan Angelacos 10 Dec '09

10 Dec '09

According to the 1.5.x TM module documentation, DNS failover is triggered by a 503 reply or a timeout. When using SRV records, it appears a 503 triggers the failure_route instead. Given: host -t srv _sip._udp.voip.example.net _sip._udp.voip.example.net has SRV record 1 0 5060 fast-server.example.net. _sip._udp.voip.example.net has SRV record 2 0 5060 slow-server.example.net. On an invite to voip.example.net, if fast-server does not respond within "fr_timer" seconds, t_relay fires off a serial branch (t_on_branch is triggered) and the invite goes to slow-server. If fast-server responds with a 503, the t_on_failure route is triggered instead, and the invite is not sent on to slow-server. If its not a bug in the module, does anyone have clues of how to coax the next DNS SRV record out of the t_on_failure route? Or what can be done to get t_relay to consume the 503? Thanks!

3 7

[Kamailio-Users] ldap_search get result URL
by Andrey Utkin 10 Dec '09

10 Dec '09

How can i access LDAP URI of ldap_search() results in routing script? I need to do recursive search in company directory within persons that belong to groups, i need to get group id without storing groupid as person attribute, but as an element of resulting URL. Query base name: dc=example,dc=com. Query result is cn=user1,ou=group1,dc=example,dc=com. I can easily get user1 attributes, but how to get ou?

3 3

[Kamailio-Users] kam 1.5 dispatcher priority usage
by Robert McGilvray 10 Dec '09

10 Dec '09

What is the function of the priority field in dispatcher module? The docs are pretty light on the subject and it's not clear to me what it's meant to do. I'm looking for a way to assign a priority to preferred gateways within the same group. Thanks Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

4 4

[Kamailio-Users] Siremis login page once and again.
by Vicente 10 Dec '09

10 Dec '09

Hello, I've installed siremis 0.9.4 with kamailio 3.0.0rc2 on a test computer and it was working without a problem. But now I'm trying to install it on a server and I can't go beyond the login screen. If I try a wrong password I get the error message. But if I use the right one I'm redirected to the login screen again and again. Both systems are ubuntu Linux with the following differences: Test computer Server Ubuntu 9.10 Ubuntu 8.10 apache 2.2.12 apache2.2.9 PHP 5.2.10 PHP 5.2.6 mysqld 5.1.37 mysqld 5.0.67 The bigger difference that I can see is mysqld version.But I've installed PHPOpenBiz framwork and the demo app creates and populates the demo data base. So it doesn't look like related to this. I've used the apache configuration on the siremis install page to set up apache for both systems. Although on the server there are some previous settings, I've created a new virtual server for siremis. apache logs only show: File does not exist: /var/www/siremis-0.9.4/siremis-web/js/sound.js But this error also appears on the working computer. And doesn't seem to be related with the problem. The log_DEBUG.html on siremis only shows the query to the database. ¿ Is there any log file on siremis that can point to the problem ? ¿ Has anyone else had this problem ? Sorry if I ask something silly but I've been trying to find what I'm doing wrong for two days. Best regards. Vicente.

2 1

Perl Module Logging Query
by Bruce McAlister 09 Dec '09

09 Dec '09

Hi All, I have been trying to use the perl module with Kamailio 1.5.3-notls. I have managed to load the module successfully. I have a simple script with a single function that strips the "<" or ">" from a contact header and logs the info, then sets an AVP. Kamailio seems to load the perl module and the script properly, but when it runs I get an error on the logging side of it. I cannot for the life of me see where the error lies. Could one of you guru's out there please cast an eye over my config and see if you can spot something I am missing. My kamailio configuration for perl is as such: ---- loadmodule "perl.so" modparam("perl", "modpath", "/opt/kamailio/lib64/perl") modparam("perl", "filename", "/opt/kamailio/perl-scripts/kamailio-perl.pl") ---- My perl script looks as follows: ---- use OpenSER ; use OpenSER::Message ; sub clean_contact { my $contact_header = shift; log("contact header passed: $contact_header\n"); $contact_header =~ s/^<|>$//g; log("contact header fixed: $contact_header\n"); OpenSER::AVP::add("clnct", "$contact_header"); return 1; } ---- I have the following section in the route logic that calls this function as follows: ---- perl_exec_simple("clean_contact","$ct"); ---- When the perl script executes, it gets to the first log line and dies with the following message: ---- ERROR:core:XS_OpenSER__Message_log: perl error: Can't take log of 0 at /opt/kamailio/perl-scripts/kamailio-perl.pl line 7. ---- I have had a look at the logging.pl script in the sample directory for the perl module, and have tried all the methods shown in the example, but I just cannot seem to get past this particular logging issue. Any tips would be greatly appreciated. Thanks Bruce

1 1

[Kamailio-Users] Problem with REGISTER
by Uriel Rozenbaum 09 Dec '09

09 Dec '09

Hello guys, I'm having a strange problem with my Kamailio 1.5.3 when trying to register an Audiocodes Gateway. I'm using authentication for REGISTER messages. The GW sends a REGISTER without credentials, so I do a proxy_challenge. Then the GW sends a REGISTER with credentials but the proxy_authorize is returning -4. I did an NGREP to check the messages and they don't seem malformed, can you tell me if there is something on the functions that might be loosing the Proxy-Authorization header? This happens only with this GW, not if i register a softphone (XLITE). Within route block: ... if (is_method("REGISTER")) { route(11); save("location"); exit(); } ... Route(11): route[11] { # ----- AUTENTICACION DE UN MENSAJE ----- if (!proxy_authorize("", "subscriber")) { switch($retcode) { case -1: xlog("L_INFO", "AUTH: Invalid username\n"); sl_send_reply("403", "Forbidden"); exit; case -2: xlog("L_INFO", "AUTH: Invalid password\n"); sl_send_reply("403", "Forbidden"); exit; case -3: xlog("L_INFO", "AUTH: Nonce ambiguo\n"); break; case -4: xlog("L_INFO", "AUTH: No credentials\n"); break; #default: #xlog("L_INFO", "AUTH: Unknown error\n"); #break; } #xlog("L_DBG", "AUTH: -Data- Auth Username = $au\n"); #xlog("L_DBG", "AUTH: -Data- Domain = $fd\n"); #xlog("L_DBG", "AUTH: -Data- From Username = $fU\n"); proxy_challenge("192.168.200.130","0"); exit; }; xlog("L_INFO", "AUTH: Valid username and credentials\n"); consume_credentials(); } NGREP: # U 10.0.1.200:5060 -> 192.168.200.130:5060 REGISTER sip:192.168.200.130 SIP/2.0. Via: SIP/2.0/UDP 10.0.1.200;branch=z9hG4bKac26395445. Max-Forwards: 70. From: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200> >;tag=1c26389068. To: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200>>. Call-ID: 154992670511200004613(a)10.0.1.200. CSeq: 68 REGISTER. Contact: <sip:testdevice@10.0.1.200:5060>;expires=180. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Expires: 180. User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006. Content-Length: 0. . # U 192.168.200.130:5060 -> 10.0.1.200:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 10.0.1.200;branch=z9hG4bKac26395445. From: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200> >;tag=1c26389068. To: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200> >;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.ee1a. Call-ID: 154992670511200004613(a)10.0.1.200. CSeq: 68 REGISTER. Proxy-Authenticate: Digest realm="192.168.200.130", nonce="4b142e8900000000188436d32ee472791b706f36952f4162". Server: Kamailio (1.5.3-notls (i386/linux)). Content-Length: 0. . # U 10.0.1.200:5060 -> 192.168.200.130:5060 REGISTER sip:192.168.200.130 SIP/2.0. Via: SIP/2.0/UDP 10.0.1.200;branch=z9hG4bKac26766380. Max-Forwards: 70. From: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200> >;tag=1c26389068. To: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200>>. Call-ID: 154992670511200004613(a)10.0.1.200. CSeq: 69 REGISTER. Proxy-Authorization: Digest username="testdevice",realm="192.168.200.130",nonce="4b142e8900000000188436d32ee472791b706f36952f4162",uri="sip:192.168.200.130",algorithm=MD5,response="9fa0904b94f94152619a16c46ffa6485". Contact: <sip:testdevice@10.0.1.200:5060>;expires=180. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Expires: 180. User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006. Content-Length: 0. . # U 192.168.200.130:5060 -> 10.0.1.200:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 10.0.1.200;branch=z9hG4bKac26766380. From: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200> >;tag=1c26389068. To: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200> >;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.b7c8. Call-ID: 154992670511200004613(a)10.0.1.200. CSeq: 69 REGISTER. Proxy-Authenticate: Digest realm="192.168.200.130", nonce="4b142e8900000001515ab7891a3d534cc3f15e5627322da3". Server: Kamailio (1.5.3-notls (i386/linux)). Content-Length: 0. . Thanks in advance! Uriel

3 5

Upgrade Serweb from version 0.9.4 to version 2.0.1-dev1
by inge 09 Dec '09

09 Dec '09

Hi, I have to upgrade Serweb to the version 2.0.1-dev1 (that work with SER 2.0). Is it possible to use Serweb 0.9.4 (using PHP v.5.1.6) with SER 2.0 however the database "ser" has a new structure? Regards, David.

1 0

Build error: "no type or storage class" when building "presence" modules on solaris (pa module)
by Bruce McAlister 09 Dec '09

09 Dec '09

Hi All, I have come across another build issue when building the presence module bundle on Solaris 10. The specific error is as follows: config.mak included make[1]: Entering directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/modules_s/pa' make[1]: Leaving directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/modules_s/pa' config.mak included make[1]: Entering directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/modules_s/pa' /usr/sfw/bin/gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m32 -minline-all-stringops -falign-loops -mtune=athlon -Wall -I/usr/include/libxml2 -I/usr/local/include/libxml2 -I/usr/local/include -I../../lib -I../.. -DNAME='"ser"' -DVERSION='"2.99.99-pre3"' -DARCH='"i386"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/usr/sfw/bin/gcc 3.4.3"' -D__CPU_i386 -D__OS_solaris -DSER_VER=2099099 -DCFG_DIR='"/opt/kamailio/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -DSER -DSER_MOD_INTERFACE -DMOD_NAME='"pa"' -c async_auth.c -o async_auth.o In file included from async_auth.c:1: pa_mod.h:54: error: syntax error before "pa_dbf" pa_mod.h:54: warning: type defaults to `int' in declaration of `pa_dbf' pa_mod.h:54: warning: data definition has no type or storage class make[1]: *** [async_auth.o] Error 1 make[1]: Leaving directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/modules_s/pa' make: *** [modules_s] Error 1 I have read the INSTALL text file and it seems that the pa module requires libxml2. I have checked that this library is on the box. The version of libxml2 on Solaris is as follows: user@soldev:~ $ /usr/bin/xml2-config --version 2.6.23 user@soldev:~ $ /usr/bin/xml2-config --libs -L/usr/lib -R/usr/lib -lxml2 -lz -lpthread -lm -lsocket -lnsl user@soldev:~ $ /usr/bin/xml2-config --cflags -I/usr/include/libxml2 I am building the following version of SIPRouter: ser-2.99.99-pre3_src_2009-12-05_b8e67f The way I am building SIPRouter is as follows: make prefix=/opt/kamailio group_include="standard postgres presence" all If I manually change directory into the modules_s/pa directory and issue a "make" then the build still fails with the same error. If I remove the "presence" group then the build completes successfully. Any pointers for a resolution would be greatly appreciated. If you require any additional information please dont hesitate to ask. Thank you Bruce

3 4

PerlVDB Usage
by Bruce McAlister 09 Dec '09

09 Dec '09

Hi All, We are looking to integrate SIPRouter/Kamailio into our existing infrastructure. Currently we have a sip proxy and registrar that handles registration and routing. The main issue we have with integrating SIPRouter/Kamailio into our environment is that the table structure for kamailio's tables are vastly different to our existing database tables. We have several application built around our current database structure so I think it is fair to assume that we will be forced to use the existing structures. I had a look at perlvdb which seems to do what I had in plan, where the table structure is transparent to kamailio. so for example, when a registration request comes in and we then authenticate the user and want to save the location, could we use the perlvdb module to intercept the save and manipulate it so that it inserts/updates/reads data from a table structure that is different to what it is expecting? I've been trying to find an example of how to use perlvdb but have been unsucessfull in my google searches and mailing list searches. Does anyone have an example of how to use the perlvdb module? Thanks Bruce

4 6

[Kamailio-Users] check SIP agent health
by Robert R 08 Dec '09

08 Dec '09

Hi, I have two SIP agents for redundancy, node1 and node2. Kamailo is supposed to route SIP traffic to node1 unless if it is down. Then it will route the traffic to node2. Is there a way to script Kamailo to check if a SIP agent is alive or not. Thanks, R

2 1

[Kamailio-Users] Add a module's file descriptor to the main loop
by Olivier Détour 08 Dec '09

08 Dec '09

Hi, I would like to create a new module. This module has to read an external file descriptor and call a callback. I don't want to use extra process and shared memory or something like that. Is it possible to add FD in the main Select and call my callback ? Regards, -- Olivier Détour EPITA 2009

2 3

[Kamailio-Users] Problem with ACK
by Ricardo Martinez 07 Dec '09

07 Dec '09

Hello List. I'm testing the last kamailio versión 1.5.3. I'm using the default configuration file and a gateway sending calls to a registered device. Gw : 10.0.0.7 Proxy : 10.0.0.11 Registered device : 10.0.0.20 This is what I see : U 10.0.0.7:5060 -> 10.0.0.11:5060 INVITE sip:5501234567@10.0.0.11 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 INVITE. Supported: timer, replaces, early-session. Min-SE: 1800. Date: Mon, 07 Dec 2009 16:17:13 GMT. User-Agent: AddPac SIP Gateway. Contact: <sip:3341876@10.0.0.7>. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO. Content-Type: application/sdp. Content-Length: 238. Max-Forwards: 70. . v=0. o=3341876 1260202633 1260202633 IN IP4 10.0.0.7. s=AddPac Gateway SDP. c=IN IP4 10.0.0.7. t=1260202633 0. m=audio 23068 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. U 10.0.0.11:5060 -> 10.0.0.7:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 INVITE. Server: Kamailio (1.5.3-notls (i386/linux)). Content-Length: 0. . U 10.0.0.11:5060 -> 10.0.0.20:5060 INVITE sip:5501234567@10.0.0.20:5060;transport=udp SIP/2.0. Record-Route: <sip:10.0.0.11;lr=on>. Via: SIP/2.0/UDP 10.0.0.11;branch=z9hG4bKd506.b3a3dd83.0. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 INVITE. Supported: timer, replaces, early-session. Min-SE: 1800. Date: Mon, 07 Dec 2009 16:17:13 GMT. User-Agent: AddPac SIP Gateway. Contact: <sip:3341876@10.0.0.7>. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO. Content-Type: application/sdp. Content-Length: 238. Max-Forwards: 69. . v=0. o=3341876 1260202633 1260202633 IN IP4 10.0.0.7. s=AddPac Gateway SDP. c=IN IP4 10.0.0.7. t=1260202633 0. m=audio 23068 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. U 10.0.0.20:5060 -> 10.0.0.11:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.0.0.11;branch=z9hG4bKd506.b3a3dd83.0. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. Record-Route: <sip:10.0.0.11;lr=on>. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>;tag=3079903260. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.20:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1B6C7A>". Server: Aastra 57i/2.5.0.82. Supported: gruu, path. Content-Length: 0. . U 10.0.0.11:5060 -> 10.0.0.7:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. Record-Route: <sip:10.0.0.11;lr=on>. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>;tag=3079903260. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.20:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1B6C7A>". Server: Aastra 57i/2.5.0.82. Supported: gruu, path. Content-Length: 0. . U 10.0.0.20:5060 -> 10.0.0.11:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.11;branch=z9hG4bKd506.b3a3dd83.0. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. Record-Route: <sip:10.0.0.11;lr=on>. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>;tag=3079903260. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.20:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1B6C7A>". Server: Aastra 57i/2.5.0.82. Supported: gruu, path, timer, replaces. Content-Type: application/sdp. Content-Length: 210. . v=0. o=MxSIP 0 0 IN IP4 10.0.0.20. s=SIP Call. c=IN IP4 10.0.0.20. t=0 0. m=audio 3000 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U 10.0.0.11:5060 -> 10.0.0.7:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. Record-Route: <sip:10.0.0.11;lr=on>. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>;tag=3079903260. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.20:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1B6C7A>". Server: Aastra 57i/2.5.0.82. Supported: gruu, path, timer, replaces. Content-Type: application/sdp. Content-Length: 210. . v=0. o=MxSIP 0 0 IN IP4 10.0.0.20. s=SIP Call. c=IN IP4 10.0.0.20. t=0 0. m=audio 3000 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U 10.0.0.7:5060 -> 10.0.0.11:5060 ACK SIP/2.0. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>;tag=3079903260. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 ACK. Route: <sip:10.0.0.11;lr=on>. Content-Length: 0. Max-Forwards: 70. . The call is answered and then the ACK is sent without R-URI.. making the Proxy fails with : Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:core:parse_uri: uri too short: <> (0) Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:core:parse_sip_msg_uri: bad uri <> Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:rr:loose_route: failed to parse Request URI Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:core:parse_uri: uri too short: <> (0) Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:core:parse_sip_msg_uri: bad uri <> Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:pv:pv_get_ruri_attr: failed to parse the R-URI Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:core:parse_uri: uri too short: <> (0) Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:core:parse_sip_msg_uri: bad uri <> Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:pv:pv_get_ruri_attr: failed to parse the R-URI What could be the problem? I'm using the default configuration file, and this start happening to me since the upgrade from 1.5.2 to 1.5.3.. Hope that someone could help me. Ricardo Martinez.

2 1

[Serusers] Via: branch tags from SER for CANCELs/BYEs frequently don't match value in the INVITE
by Frank Durda IV 07 Dec '09

07 Dec '09

As you may recall, this is a problem where frequently (well, virtually all of the time) when SER passed a BYE or CANCEL message through, the device receiving such messages from SER would reject them, because the Via: header line that SER added to the BYE or CANCEL message didn't match the one that SER added to the INVITE message sent earlier. I brought this up a couple of times in the list previously. As there was not any response on this point at all, I have done some digging and have a simplified example and analysis of what SER appears to be doing wrong. First, here is an example of the problem occurring. Here are key header lines from the INVITE and a BYE message of a sample call: INVITE INPUT TO SER From: <sip:9715551212@10.11.12.13;isup-oli=62;user=phone>;tag=000a0285+1+9c6d0174+ff665515 To: <sip:3719110000@10.20.30.40;user=phone> Call-ID: 16305237000a0285 CSeq: 252558264 INVITE SER EMITTED THIS INVITE (to called party) Via: SIP/2.0/UDP 66.77.88.99;branch=z9hG4bKe47c.7be033a8ca37f0e8a1c6bfe63b043efe.0 From: <sip:9715551212@10.11.12.13;isup-oli=62;user=phone>;tag=000a0285+1+9c6d0174+ff665515 To: <sip:3719110000@10.20.30.40;user=phone> Call-ID: 16305237000a0285 CSeq: 252558264 INVITE BYE INPUT TO SER (from calling party) Call-ID: 16305237000a0285 From: <sip:9715551212@10.11.12.13;isup-oli=62;user=phone>;tag=000a0285+1+9c6d0174+ff665515 To: <sip:3719110000@10.20.30.40;user=phone>;tag=15358744 CSeq: 252558265 BYE SER EMITTED THIS BYE (to called party) Via: SIP/2.0/UDP 66.77.88.99;branch=z9hG4bKf47c.c79c6922afd022e9b4cdf00110f17fd5.0 Call-ID: 16305237000a0285 From: <sip:9715551212@10.11.12.13;isup-oli=62;user=phone>;tag=000a0285+1+9c6d0174+ff665515 To: <sip:3719110000@10.20.30.40;user=phone>;tag=15358744 CSeq: 252558265 BYE The called switch rejects the BYE with a 481 because top-most Via: in the BYE does not match the one from the INVITE. Here are the two VIAs added by SER that do not match: INVITE Via: SIP/2.0/UDP 66.77.88.99;branch=z9hG4bKe47c.7be033a8ca37f0e8a1c6bfe63b043efe.0 BYE Via: SIP/2.0/UDP 66.77.88.99;branch=z9hG4bKf47c.c79c6922afd022e9b4cdf00110f17fd5.0 A review of how SER constructs the branch=nnnn value reveals where the differences are coming from: #1. e47c vs f47c difference caused by the Cseq: number changing between INVITE and BYE messages. The Cseq: value is supposed to be different in the INVITE and BYE per RFC 3261. The Cseq: should not be used in generating the branch=nnnn value. In this example, the BYE just happened to get the very next sequence number, but it will always be a value other than the sequence number seen in the INVITE. #2. 7be033a8ca37f0e8a1c6bfe63b043efe vs c79c6922afd022e9b4cdf00110f17fd5 is caused by ;tag=15358744 added to BYE To: header line. At minimum, any tags present on the To: and From: header should not be used in generating the branch=nnnn value because of the handling of tags in called-party-BYE situations. Also, there is no requirement in RFC 3261 that the To: nor From: header lines be used in generating the branch=nnnn value at all. Based on what I am seeing, it appears that neither the Cseq: nor the complete To: or From: header lines should have ever been used in constructing a branch=nnnn value, because of the adverse affects on CANCEL and BYE messages. Four things appear to be key to causing the branch=nnnn computation to generate non-compliant results in this example, as well as the case of a CANCEL message sent prior to the completion of the INVITE. (There could be other scenarios.) #1. The Cseq: value of the BYE will be different than the INVITE (per RFC 3261). The Cseq: header line should not ever be used in generation the hash for a branch=nnnn value as it will prevent the branch in BYE from matching the one in the INVITE. #2. If the called party generated the BYE message, the tags on the To: and From: header lines will be exchanged (also per RFC 3261), so the entire To: and From: header line's content cannot be used for generating a hash for a branch=nnnn value that is expected to match earlier messages that had a branch=nnnn value computed prior to the tag swap. #3. If a tag was previously absent on the To:/From:, a BYE from either party will add one. Again, the entire To: and From: header lines cannot be used for generating a hash for a branch=nnnn value that matches earlier messages prior to the addition of any tag. #4. Any deviation in the From: and To: header lines between INVITE and BYE/CANCEL will yield a different branch=nnnn value, even though the parameters of the From: and To: header lines may still be perfectly legal and technically identical. Any deviation may be the result of a sloppy SIP implementation in the calling equipment, but isn't valid grounds for not handling the message correctly. Examples include things like alternate number formats (as in To: sip:8885551212@66.77.88.99:5060 (seen in INVITE) versus To: <sip:8885551212@66.77.88.99:5060> (seen in CANCEL from same calling device), or differences in the amount of whitespace (including trailing whitespace), which is supposed to be ignored in SIP messages but the MD5 generator doesn't appear to know that it should ignore whitespace. Any one or a combination of these items (and possibly others) will cause the current branch=nnnn generation code in SER to generate a different branch=nnnn value and a receiving system who is following RFC 3261 will reject the BYE or CANCEL message that has passed through SER. This is non-compliant behavior. For BYE, you only get into this situation if you are using Record-Route to force the BYE message to pass through SER so that it can tear down the call (unforce_rtp_proxy() and similar actions), but for a CANCEL message, anybody can run into this problem, if syn_branch=0. So, while this defect in the branch=nnnn generation doesn't affect everybody, it still appears to be wrong. Is there by chance a compliant version of char_msg_val() exists that doesn't have these issues? That is, it doesn't use the Cseq: header line, and if it uses the From: or To: header line at all, it computes using only the called/calling digits, and not anything else that may appear on those header lines. I also don't understand why this is being MD5'ed. The RFC only says that the branch must be unique, and there are lots of cheaper has algorithms out there. Thanks for reading this far and thanks in advance for anyone who has fixed this or can come up with a fix.

4 7

[Kamailio-Users] Add Reason Header in CANCEL request (forked call)
by heinz huber 07 Dec '09

07 Dec '09

Hi everyone, I've the following question: Is it possible to add a reason header in a CANCEL request? This is helpful in the following scenario: A call gets forked by the proxy (since there are more contacts for one AOR). On the of the SIP-destination-UAs accepts the call and sends a 200 OK to the proxy. The proxy afterwards generates the CANCEL request for all open UAC transactions. In this case it would be helpful to add additional infos, like "Call completed" elsewhere, so that the call is not listed in the "missing calls" list in the destinations UAs. I've found a linke in the tracker on http://sourceforge.net/tracker/?func=detail&aid=2581449&group_id=139143&ati… Has this feature been included in the last official kamailio version, or is it still open. thx && br Andy -- Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3.5 - sicherer, schneller und einfacher! http://portal.gmx.net/de/go/atbrowser

2 1

[Kamailio-Users] log not showing when using uac_replace_from( ) function
by Robert R 07 Dec '09

07 Dec '09

Hi, I simply trying to modify URI and FROM headers as follow. tcpdump packet capture shows the values are changed for both URI and FROM headers. However, the syslog only shows the new value for URI and not for FROM header. How can I verify new value in FROM header in the log file? # main request routing logic route { $rU = avp(i:1); $rd = avp(i:2); .... uac_replace_from("$avp(i:3)"); xlog ("fu after perl: $fu"); xlog ("ru after perl: $ru"); ... } Thanks, Rob

3 2

[Kamailio-Users] finding out which gw id was used
by Anders 07 Dec '09

07 Dec '09

Hi, I would like to know which gw was used for a call. Right now, it is doing this: sql_query("cb", "select id from gw where ip_addr='$rd'", "rb"); ...in route[1], but that's not good enough, because I have more than one gw with the same IP, so it just takes the first one available. If I could do something like "select id from gw where grp_id='grp_id_from_lcr'" , where grp_id_from_lcr is the the grp_id in the lcr table, which lead to the gw used... then I'd have the right id of the gw, since I use unique numbers for the grp_id in the gw table. How do I do this? Thanks!!

2 1

[Kamailio-Users] Can't connect with rtpproxy-0.2_2007-07-28
by RAJNIKANT VANZA 07 Dec '09

07 Dec '09

Hi Friends, i have problem about rtpproxy connection. /var/log/messages/ --------------------------- ERROR:nathelper:send_rtpp_command: can't connect to RTP proxy ERROR:nathelper:send_rtpp_command: proxy <unix:/var/run/rtpproxy.sock> does not respond, disable it WARNING:nathelper:rtpp_test: can't get version of the RTP proxy WARNING:nathelper:rtpp_test: support for RTP proxy <unix:/var/run/rtpproxy.sock> has been disabled temporarily i have installed successful rtpproxy-0.2_2007-07-28. run from /usr/local/sbin/rtpproxy in /var/run/ to files exists as -rw-r--r-- 1 root root 6 Dec 4 19:48 rtpproxy.pid srwxr-xr-x 1 root root 0 Dec 4 19:48 rtpproxy.sock kamailio.cfg ------------------ loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") -- Thanks and Regards Rajnikant Vanza Call : +91-97374 56583

2 1

Symbol referencing errors (building ser-2.99.99-pre3_src_2009-12-03_ca351a on Solaris)
by Bruce McAlister 07 Dec '09

07 Dec '09

Hi All, When attempting to build SIPRouter version ser-2.99.99-pre3_src_2009-12-03_ca351a on Solaris 10 I am presented with the following error: ---- config.mak included make[1]: Entering directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/modules/ctl' Makefile.defs defs skipped make[2]: Entering directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/utils/sercmd' make[2]: Leaving directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/utils/sercmd' Makefile.defs defs skipped make[2]: Entering directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/utils/sercmd' /usr/sfw/bin/gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m32 -minline-all-stringops -falign-loops -mtune=athlon -Wall -DNAME='"sercmd"' -DVERSION='"0.2"' -DMOD_NAME="utils/sercmd" -c parse_listen_id.c -o parse_listen_id.o /usr/sfw/bin/gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m32 -minline-all-stringops -falign-loops -mtune=athlon -Wall -DNAME='"sercmd"' -DVERSION='"0.2"' -DMOD_NAME="utils/sercmd" -c sercmd.c -o sercmd.o /usr/sfw/bin/gcc -m32 -O2 parse_listen_id.o sercmd.o -o sercmd Undefined first referenced symbol in file bind sercmd.o getservbyname parse_listen_id.o gethostbyname sercmd.o socket sercmd.o connect sercmd.o ld: fatal: Symbol referencing errors. No output written to sercmd collect2: ld returned 1 exit status make[2]: *** [sercmd] Error 1 make[2]: Leaving directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/utils/sercmd' ERROR: make -C ../../utils/sercmd install-if-newer failed make[1]: *** [install-utils] Error 1 make[1]: Leaving directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/modules/ctl' make: *** [install-modules] Error 1 ---- Here are the steps that I have followed to get to this point: [1] make prefix=/opt/kamailio [2] make prefix=/opt/kamailio 'include_modules=db_unixodbc db_postgres cpl-c snmpstats' modules [3] make basedir=/var/tmp/pkgbuild-user/SIPRouter-2.99.99-pre3-build prefix=/opt/kamailio 'include_modules=db_unixodbc db_postgres cpl-c snmpstats' install The above error I receive is during the last step ([3]), the installation phase of SIPRouter. I'm not sure where to go from here, although it looks like it may be a resolver library that is not included along the line. Any help would be appreciated. If you need any further information, please dont hesitate to ask. Thanks Bruce

2 4

[Kamailio-Users] [kamailio-users] Error : SIP/2.0 401 Unauthorized
by RAJNIKANT VANZA 05 Dec '09

05 Dec '09

Hi Friends, need to help. *I have problem about sip : SIP/2.0 401 Unauthorized* Is it require to nathelper module in kamailio ? *what can i write kamailio.cfg file when kamailio and Asterisk on same network?* Scenario is like as : ----------------------------- 1) kamailio server on 172.18.100.74 kamailio.cfg ( nathelpler module ) ----------------- loadmodule "nathelper.so" modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") 2) Asterisk server on 172.18.100.65 sip.conf ----------- [rajnikant] nat=yes disallow=all allow=alaw allow=ulaw allow=gsm type=peer context=default host=172.18.100.74 fromdomain=rajnikant.net mailbox=user@context -- Thanks and Regards Rajnikant Vanza

2 1

[Kamailio-Users] Kamailio RTP Proxy issues
by General Lee 04 Dec '09

04 Dec '09

Hi, I am currently integrating an H323 to SIP gateway with Kamailio and trying to route all calls through RTP Proxy. I have a problem where RTP Proxy treats both the incoming H323 gateway call (Caller) and outgoing SIP call (Callee) as the 'caller' in the RTP Proxy syslog. RTP Proxy doesn't assign a 'callee' therefore not able to setup a call (See syslog below). I have configured Kamailio to accept the 'ACK' with SDP and this is working correctly. When I enable H323 Fast connect and the SDP is included in the INVITE, the call connects correctly and is routed through RTP Proxy. I feel the problem is related to RTP Proxy receiving an INVITE from the H323-SIP gateway without SDP. Can anyone explain why RTP Proxy treats both the incoming H323 Gateway call and outgoing SIP call as the 'caller' in the RTP Proxy syslog. How can I make RTP Proxy treat the incoming H323 call as the 'callee'? Thanks, More information below ******************************************************************************* My H323 endpoints use H323 Slow Connect, so when the H323-SIP Gateway delivers the 'INVITE' to Kamailio there is no SDP included in the INVITE. I added a 'onreply_route' to the Kamailio configuration file which handles the 'ACK' with SDP which is working correctly. All of my SIP calls (Signalling + Media) are forced though RTP Proxy and I would like to force all H323-SIP Gateway calls through RTP Proxy. When placing a call from my H323 endpoint to my SIP UA, the RTP Proxy syslog records the incoming and outgoing call, however RTP Proxy states that the ‘callee’ is actually the 'caller'. The RTP Proxy syslog also states that the caller is the caller so there is no 'callee' (see below). In the syslogs both the ‘callee’ and ‘caller’ are recognised as the ‘caller’ so RTP Proxy has no callee to send the traffic back to. When the INVITE is received the ‘callee’ is populated in the syslogs as the ‘caller’. The H323 Gateway call is not recorded until after the ‘ACK’ with SDP is received from the gateway. Oct 27 17:33:05 rtpproxy[24086]: INFO:handle_command: pre-filling caller's address with 72.19.211.106:49620 (Should be callee) Oct 27 17:33:05 rtpproxy[24086]: INFO:handle_command: pre-filling caller's address with 72.19.211.106:49622 (Should be callee) Then after the ‘ACK’ is received from the Gateway, the H323 call is mentioned in the syslog as well as the other caller who is supposed to be the callee. Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling caller's address with 69.72.11.51:10204 Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling caller's address with 69.72.11.51:10214 Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling caller's address with 72.19.211.106:49620 Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling caller's address with 72.19.211.106:49622 -- View this message in context: http://www.nabble.com/Kamailio-RTP-Proxy-issues-tp26112049p26112049.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 12

Build Error for v3.0.0-rc2 (2009-12-03_3ed668) on Solaris
by Bruce McAlister 03 Dec '09

03 Dec '09

Hi All, I am trying to build Kamailio v3.0.0 RC2 on Solaris 10 (note: *not* OpenSolaris), but I have been faced with a couple issues I hope you can help me with. The first issue has to do with GNU as (/usr/sfw/bin/gas), there appears to be a check somewhere for what I assume to be the version number of the GNU assembler. When the make is running "/usr/sfw/bin/gas -V" it just hangs at this point with the following output from make: ---- make prefix=/opt/kamailio Makefile:248: config.mak: No such file or directory normal Makefile.defs exec target architecture <i386>, host architecture <x86_64> ---- I did a process listing and noticed the following process waiting: 1719 make prefix=/opt/kamailio 1763 /bin/sh -c if [ -z "/usr/sfw/bin/gas" ] ; then echo "as" ; else if /usr/sfw/bin 1764 /usr/sfw/bin/gas -V I ended up making a wrapper script that just output the version information for "/usr/sfw/bin/gas -V" which let me get a little further. The wrapper script looks as follows: ---- #/bin/ksh if [ "$1" = "-V" ]; then echo "GNU assembler version 2.15 (i386-pc-solaris2.10) using BFD version 2.15" exit else /usr/sfw/bin/gas.bin $@ fi ---- However, when I now try a "make prefix=/opt/kamailio" I am faced with the following error: ---- gcc: installation problem, cannot exec `/usr/sfw/bin/gas': Exec format error ---- The complete output of the make is as follows: ---- make prefix=/opt/kamailio Makefile:248: config.mak: No such file or directory normal Makefile.defs exec target architecture <i386>, host architecture <x86_64> making config... yacc -d -b cfg cfg.y conflicts: 5 shift/reduce flex cfg.lex Makefile.defs defs skipped /usr/sfw/bin/gcc -g -O9 -funroll-loops -Wcast-align -m32 -minline-all-stringops -falign-loops -mtune=athlon -Wall -DNAME='"kamailio"' -DVERSION='"3.0.0-rc2"' -DARCH='"i386"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/usr/sfw/bin/gcc 3.4.3"' -D__CPU_i386 -D__OS_solaris -DSER_VER=3000000 -DCFG_DIR='"/opt/kamailio/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DSTATISTICS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -c action.c -o action.o gcc: installation problem, cannot exec `/usr/sfw/bin/gas': Exec format error make: *** [action.o] Error 1 ---- If I remove the wrapper script then the make just hangs on the command "/usr/sfw/bin/gas -V". Would anyone know how I could work around the issue? Just an FYI, I am able to build Kamailio 1.5.2-notls on the same machine without any errors/hangs, although we are considering moving to Kamailio 3/Sip-Router when it is officially released, this is why I am trying to build v3.0.0 RC2 at the moment. If you need any further information please dont hesitate to ask, any help in resolving the issue would be greatly appreciated. Thanks Bruce

3 4

[Kamailio-Users] log not showing when using uac_replace_from( ) function
by Robert R 03 Dec '09

03 Dec '09

Hi, I simply trying to modify URI and FROM headers as follow. tcpdump packet capture shows the values are changed for both URI and FROM headers. However, the syslog only shows the new value for URI and not for FROM header. How can I verify new value in FROM header in the log file? # main request routing logic route { $rU = avp(i:1); $rd = avp(i:2); .... uac_replace_from("$avp(i:3)"); xlog ("fu after perl: $fu"); xlog ("ru after perl: $ru"); ... } Thanks, Rob

1 0

[Kamailio-Users] log not showing when using uac_replace_from( ) function
by Robert R 03 Dec '09

03 Dec '09

Hi, I simply trying to modify URI and FROM headers as follow. tcpdump packet capture shows the values are changed for both URI and FROM headers. However, the syslog only shows the new value for URI and not for FROM header. How can I verify new value in FROM header in the log file? # main request routing logic route { $rU = avp(i:1); $rd = avp(i:2); .... uac_replace_from("$avp(i:3)"); xlog ("fu after perl: $fu"); xlog ("ru after perl: $ru"); ... } Thanks, Rob

1 0

[Kamailio-Users] Kamailio Perl OpenSER::AVP
by Örn Arnarson 03 Dec '09

03 Dec '09

Hello everyone, I've been successfully using the kamailio Perl module, but now I'm trying to set AVPs. Unsuccessfully. Whenever I do "use OpenSER::AVP;" and restart kamailio, the following error messages appear and kamailio doesn't start: Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:core:XS_OpenSER__Message_log: perl error: Can't locate OpenSER/AVP.pm in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.10.0 /usr/local/share/perl/5.10.0 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.10 /usr/share/perl/5.10 /usr/local/lib/site_perl .) at /etc/kamailio/checkip.pl line 4. Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:core:XS_OpenSER__Message_log: perl error: Can't locate OpenSER/AVP.pm in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.10.0 /usr/local/share/perl/5.10.0 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.10 /usr/share/perl/5.10 /usr/local/lib/site_perl .) at /etc/kamailio/checkip.pl line 4. BEGIN failed--compilation aborted at /etc/kamailio/checkip.pl line 4. Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:perl:parser_init: failed to load perl file "/etc/kamailio/checkip.pl". Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:core:init_mod: failed to initialize module perl Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:core:main: error while initializing modules Now, I cannot find this AVP module within the perl package anywhere. I downloaded the source of 1.5.0 and it's nowhere to be found. It is, however, in the .1.5 documentation. Does anyone have any info on this for me? Best regards, Örn

3 3

[Kamailio-Users] Using Sip Message in a third application.
by Olivier Détour 03 Dec '09

03 Dec '09

Hello, I'm using Kamailio 1.5.3. I configured it in a proxy mode. I would like to know what is the best way to forward the SIP message to a C application via a socket, block kamailio.cfg script, and reforward it to kamailio to continue his treatment. I found a ugly bay using PV as lock with a while loop, but this solution is not acceptable for me. I saw SEAS module, but it is only for Java Application server. Do you know a simple way to do it with an existing module ? Do I have to write my own module ? Thanks for your response, Regards, -- Olivier Détour EPITA 2009

3 5

topoh: remove Via headers
by Klaus Darilion 03 Dec '09

03 Dec '09

Hi! IIRC correctly. topoh modules rewrites existing Via headers. Would it be possible to remove existing headers completely so that the forwarded request has only one Via header? regards klaus

1 0

[Kamailio-Users] Integrate Skype into Kamailio (Must support video)
by General Lee 03 Dec '09

03 Dec '09

Hi, I would like to integrate Skype into my Kamailio video system? Has anyone implemented this successfully before? Or suggest how this could be done? Key features I would like are: 1. Must support Video 2. Ideally support H264 video codec 3. Share presence information 4. Ability to make calls in both directions I would appreciate any information you could provide. Thanks, -- View this message in context: http://old.nabble.com/Integrate-Skype-into-Kamailio-%28Must-support-video%2… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

4 6

[Kamailio-Users] ERROR:tm:add_uac: maximum number of branches exceeded
by alex pappas 02 Dec '09

02 Dec '09

Hello there, I'm using dispatcher to serial fork a call through many gateways. For a call to number 2XXXXXXXXX , first a try to connect through gateways A,B,C after that D,F , after that E,R , after that M,N and finally after those K,L. When the call is trying gateway N then I'm getting the ERROR:tm:add_uac: maximum number of branches exceeded. I found on internet that I'm exceeding the MAX_BRANCHES: branch=t->nr_of_outgoings; if (branch==MAX_BRANCHES) { LOG(L_ERR, "ERROR:tm:add_uac: maximum number of branches exceeded\n"); ret=E_CFG; goto error; } >From a trace I got , I see that kamailio is sending to each destination IP 8 INVITES and if there is no answer to that is continuing with next destination gateway sending also there 8 INVITES. When the number of INVITES is 52 then I get this error. Does anyone can suggest a way to change the MAX_BRANCHES MAX number ? Is is important to keep open the INVITEs from the first gateways because is an LCR scenario and if I get an answer from these gateways is important to complete through them. Any help is apreciated Thanks Alexandros

2 6

[Kamailio-Users] Random looped back audio issues with RTPPROXY
by tony＠everestbn.com 02 Dec '09

02 Dec '09

Hello, I'm having random audio issues (<10% of the time) with blind call transfers on a Polycom phone. According to RTPPROXY's logs, it seems it is substituting the same IP and port for the caller's and callee's address causing one end to hear their own voice while the other end hears nothing. INFO:rxmit_packets: caller's address filled in: a.b.c.199:17856 (RTP) INFO:rxmit_packets: guessing RTCP port for caller to be 17857 INFO:rxmit_packets: callee's address filled in: a.b.c.199:17856 (RTP) INFO:rxmit_packets: guessing RTCP port for callee to be 17857 The SDP in the SIP messages has all the correct IPs and ports and it appears from a SIP overview everything should work fine. Here is some background information: I am using Kamailio 1.5.3 with RTPPROXY 1.2.1. SIP PATH is as follows: PSTN Gateway---Kamailio--SIP Proxy Server--Polycom Phone RTP PATH: PSTN Gateway---Kamailio----Polycom Phone *There are nor firewalls or filters. Call Flow: I place a call from the gateway to the ip phone. Call is answered. When I do a blind transfer from the Polycom, the SIP Proxy sends an invite with no SDP, the gateway responds with a 200 with SDP and the SIP Proxy then sends an ACK with SDP. As I mentioned, this works most of the time and the SIP messages look identical for when the transfer works and when it doesn't. When it doesn't work the gateway end hears their own voice while the recipient Polycom transfer (usually another phone on the same proxy) hears nothing. Here is a portion of my configuration file: ..... if (is_method("INVITE")) { if((search("^Content-Type:[ ]*application/sdp")) || (search("^Content-Type:application/sdp"))){ rtpproxy_offer("fcr"); setflag(12); } } if (is_method("ACK")) { if((search("^Content-Type:[ ]*application/sdp")) || (search("^Content-Type:application/sdp"))) { rtpproxy_answer("fcr"); } } t_on_reply("1"); .... } onreply_route[1] { if (status=~"(180)|(183)|(2[0-9][0-9])"){ if((search("^Content-Type:[ ]*application/sdp")) || (search("^Content-Type:application/sdp"))) { if (isflagset(12)) { rtpproxy_answer("fcr"); } else { rtpproxy_offer("fcrl"); } } } } Any help would be greatly appreciated! Thanks, Tony

2 1

Re: [Kamailio-Users] [sr-dev] sr and kamailio autogenerated tarballs
by Daniel-Constantin Mierla 02 Dec '09

02 Dec '09

On 12/2/09 1:50 PM, Andrei Pelinescu-Onciul wrote: > Daily snapshots of sr_3.0, kamailio_3.0 and sr master branch can be > downloaded from: http://sip-router.org/tarballs/sr/ . > Note that new tarballs are generated only if there are changes for the > corresponding branch (so if you don't see a tarball with today's date it > means there was no change from the previous one). > > The tarballs are generated via make tar, which makes sure the correct > repository version will be included in the compiled binary (make tar > generates first autover.h before creating the archive). > Thanks Andrei! I cc-ed Kamailio users mailing list, since there might be people interested as well. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Serusers] Memory leak issue
by Piyush.Bansal＠relianceada.com 02 Dec '09

02 Dec '09

Hi All, I'm using SER version 2.0.0-rc1 as a SIP server and in that facing lot of memory leaks issue. I'm not able to enable the memory logs in SER. Can anybody help me identifying memory leaks or tell me how to enable memlogs so that I myself will try removing all the leaks... Plz help me ASAP, its urgent... Thanx n regards, --Piyush The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

2 1

[Kamailio-Users] Kamailio -> Asterisk Redirect/Reinvite and Remote-Party-ID
by Florian Meister 02 Dec '09

02 Dec '09

Hi, basically I'm using this structure at the moment: SIP Users <----> Kamailio <-----> Asterisk <-----> PSTN I have to add a diversion-functionality at kamailio-level, so to simply rewrite $ru with something else defined in the database. That's working without problems. For billing issues, I also have to add a Remote-Party-ID header, set to the SIP user, which initiated the redirect in the database. Now to the problem: When a call is coming from PSTN, it's passing the asterisk server, then at kamailio level $ru is rewritten and sent back to asterisk (I'm talking about a redirect to a number in PSTN here) What I've seen from the logs is, that asterisk is seeing that it gets an invite back with the same call-id, and therefore it cancels the original invite and handles the whole call internally via the Local Channel. The Problem is, that in the invite sent from kamailio back to asterisk, I've set a Remote-Party-ID header to tell asterisk to set the Callerid correctly for billing purposes. Now it seems that asterisk is _ignoring_ this header from the second invite. So is this an expected behavior ? If yes, how to do it correctly ? Below you can see the verbose output of asterisk. Since the call is handled at "Local" Channels the function to read sip headers does not work. The only message I get is "thanks to SIP/tpsiptestproxyu01-00d0a0b8". -- Called tpsiptestproxyu01/+435572949012 -- Now forwarding DAHDI/2-1 to 'Local/066480588134@from-internal' (thanks to SIP/tpsiptestproxyu01-00d0a0b8) -- Executing [066480588134@from-internal:1] NoOp("Local/066480588134@from-internal-d69e;2", "435572501134") in new stack [Dec 1 15:14:50] WARNING[20506]: chan_sip.c:15797 func_header_read: This function can only be used on SIP channels. -- Executing [066480588134@from-internal:2] NoOp("Local/066480588134@from-internal-d69e;2", "") in new stack -- Executing [066480588134@from-internal:3] Dial("Local/066480588134@from-internal-d69e;2", "DAHDI/G0/066480588134") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G0/066480588134 -- DAHDI/124-1 is proceeding passing it to Local/066480588134@from-internal-d69e;2 -- Local/066480588134@from-internal-d69e;1 is proceeding passing it to DAHDI/2-1 In the SIP debug you can see that asterisk is cancelling the dialog with kamailio and doing it itself: 13:49:30.589054 IP [--ASTERISK--].5060 > [--KAMAILIO--].5060: SIP, length: 938 E.......@..L..,L..,N........INVITE sip:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]> Contact: <sip:435572501134@[--ASTERISK--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Remote-Party-ID: "435572501134" <sip:435572501134@[--ASTERISK--]>;privacy=off;screen=yes Date: Tue, 01 Dec 2009 12:49:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 930830518 930830518 IN IP4 [--ASTERISK--] s=Asterisk PBX 1.6.1.5 c=IN IP4 [--ASTERISK--] t=0 0 m=audio 16924 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 13:49:30.591037 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length: 342 E..r..@.@.[0..,N..,L.....^.aSIP/2.0 100 Trying Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 INVITE Server: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 13:49:30.594345 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length: 1093 E..a..@.@.XA..,N..,L.....M+LINVITE sip:066480588134@[--ASTERISK--]:5060;transport=udp SIP/2.0 Record-Route: <sip:[--KAMAILIO--];lr;ftag=as27658014> Via: SIP/2.0/UDP [--KAMAILIO--];branch=z9hG4bK06.05390227.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 Max-Forwards: 69 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]> Contact: <sip:435572501134@[--ASTERISK--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 01 Dec 2009 12:49:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 Remote-Party-ID: "435572949012" <sip:435572949012@tpseru01.tele.net>;party=caller;privacy=none;screen=yes v=0 o=root 930830518 930830518 IN IP4 [--ASTERISK--] s=Asterisk PBX 1.6.1.5 c=IN IP4 [--ASTERISK--] t=0 0 m=audio 16924 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 13:49:30.594605 IP [--ASTERISK--].5060 > [--KAMAILIO--].5060: SIP, length: 460 E.......@..)..,L..,N....... CANCEL sip:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 CANCEL User-Agent: Asterisk PBX 1.6.1.5 Remote-Party-ID: "435572501134" <sip:435572501134@[--ASTERISK--]>;privacy=off;screen=yes Content-Length: 0 13:49:30.596307 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length: 387 E.....@.@.[...,N..,L.......KSIP/2.0 200 canceling Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]>;tag=45db18648893e7acabf725621374d382-4ddb Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 CANCEL Server: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 13:49:30.596801 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length: 396 E.....@.@.Z...,N..,L.......kSIP/2.0 487 Request Terminated Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]>;tag=45db18648893e7acabf725621374d382-4ddb Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 INVITE Server: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 13:49:30.596842 IP [--ASTERISK--].5060 > [--KAMAILIO--].5060: SIP, length: 539 E..7....@.....,L..,N.....#.oACK sip:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]>;tag=45db18648893e7acabf725621374d382-4ddb Contact: <sip:435572501134@[--ASTERISK--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.5 Remote-Party-ID: "435572501134" <sip:435572501134@[--ASTERISK--]>;privacy=off;screen=yes Content-Length: 0 Thanks, Florian

6 9

[Kamailio-Users] client registers with private IP
by Joao Gomes Pereira 01 Dec '09

01 Dec '09

Hello I'm trying to register in Kamailio, an Asterisk working behind Nat. Here is the trunk: The problem is: Asterisk is registering with its internal IP (192.168.1.25), as you can see here: sip:s@192.168.1.25 Q= Expires:: 81 Callid:: 480b40aa13ddd8707787b21a69656535(a)127.0.0.1 Cseq:: 103 User-agent:: Asterisk PBX How can I force Asterisk to register with its public IP? Is there some configuration I can do in Kamailio? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira(a)startel.pt

3 2

[Kamailio-Users] CDRTool Mediaproxy etc.. AG stuff
by Brian Chamberlain 01 Dec '09

01 Dec '09

Hi List, Apologies if you get this twice, for some reason the last post didn't seem to appear on the list.. I am running Debian lenny. Can I just change the repos in sources.list from lenny to testing and unstable to get the correct python dependencies etc for the ag-project tools? Thanks, Brian

4 4

[Kamailio-Users] presence send system info
by alex pappas 01 Dec '09

01 Dec '09

Dear friends, I'm trying to send some system info through presence in my X-Lite client , following the example http://www.kamailio.org/dokuwiki/doku.php/presence:pua-modules#using_with_x… Does anyone knows what configuration should i have in the X-Lite side? Thanks Alex

2 7

[Kamailio-Users] t_on_failure() t_on_reply()
by BERGANZ François 01 Dec '09

01 Dec '09

Hello, Is it possible to do that : ? t_on_reply("1"); t_on_failure("1"); t_relay(); François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 3

[Kamailio-Users] Fwd: Binding to multiple IP addresses on the same interface.
by Geoffrey Mina 01 Dec '09

01 Dec '09

yes it does: [root@atl-sipgateway1 ~]# ip addr show dev eth0 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast qlen 1000 link/ether 00:1e:c9:b6:4c:fa brd ff:ff:ff:ff:ff:ff inet 67.220.XX.XX/27 brd 67.220.XX.XX scope global eth0 inet 67.220.XX.XX/27 brd 67.220.XX.XX scope global secondary eth0:1 inet6 fe80::21e:c9ff:feb6:4cfa/64 scope link valid_lft forever preferred_lft forever On Mon, Nov 30, 2009 at 12:15 PM, Alex Hermann <alex(a)speakup.nl> wrote: > On Sunday 29 November 2009 20:57:59 Geoffrey Mina wrote: >> I have a Kamailio machine that has multiple IPs assigned to a single >> interface. >> >> eth0 = 1.2.3.4 >> eth0:1 = 1.2.3.5 > > No idea if it has any influence on Kamailio behaviour, but those seem to be > two interfaces as far as applications are concerned. Assigning multiple IP's > to one interface means that for all IP's, the interface should be the same. > > Does 'ip addr show dev eth0' show 2 IP's? > > -- > Alex Hermann > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >

1 0

[Kamailio-Users] 487
by Anders 30 Nov '09

30 Nov '09

Hi, This question may belong in a general SIP forum - but I'll drop it here anyway, - hope I stir no big anger ;-) I'm running Kamailio 1.5, - use it for wholesale, and my customers typically send me calls from their Asterisk platform. I'm experiencing a very share of calls ending up in 487. This should be the client terminating the call - but is it my client (the provider with the Asterisk platform) or is it the final client - my client's client? Could Asterisk send a 487 for some reason? I've researched this (maybe not good enough!), but cannot find the answer: Does a 487 always mean that the client gave up and hung up? If this is the case, could high PDD be my problem - it's taking too long and the client gives up? Thanks!

2 1

[Kamailio-Users] Binding to multiple IP addresses on the same interface.
by Geoffrey Mina 30 Nov '09

30 Nov '09

Hello, I have a Kamailio machine that has multiple IPs assigned to a single interface. eth0 = 1.2.3.4 eth0:1 = 1.2.3.5 In my Kamailio config i have listen=eth0:5060 And it doesn't appear to be binding to my second IP address. What is the proper way to tell the server to bind to all IPs on a single interface? I don't want to specify the IP directly for binding as I use the same config across 4 Kamailio servers. listen=eth0:1:5060 (i think that's not correct) Thanks, Geoff p.s. I would do some testing, but this is a live machine and I don't want to bring it up and down.

4 7

Re: [Kamailio-Users] Compiling nonce.o (Auth Module) with TLS=1 fails
by Martin Koenig 30 Nov '09

30 Nov '09

Hello Laszlo, Good hint. We are building against openssl-0.9.7f-7.10. For testing purposes I have commented out the INCLUDE of md5.h in nonce.c, now the module compiles with the following warning: gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"auth"' -DNAME='"kamailio"' -DVERSION='"1.5.3-tls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DUSE_LOCAL_ROUTE -DF_MALLOC -DUSE_TLS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/ssl/include -I/usr/local/include -I/usr/include/openssl -c nonce.c -o nonce.o nonce.c: In function âcalc_nonceâ: nonce.c:116: warning: implicit declaration of function âMD5Initâ nonce.c:126: warning: implicit declaration of function âMD5Updateâ nonce.c:129: warning: implicit declaration of function âU_MD5Finalâ I'll try to see if this is functional now. However, it feels like a "dirty hack". Best regards, Martin ________________________________ Von: voipfreak.net(a)gmail.com [mailto:voipfreak.net@gmail.com] Im Auftrag von Laszlo Gesendet: Montag, 30. November 2009 17:57 An: Martin Koenig Betreff: Re: [Kamailio-Users] Compiling nonce.o (Auth Module) with TLS=1 fails Hi Martin, [...] Which version of openssl you have? You might find this interesting: http://bugs.gentoo.org/146780 (botton of the page) -Laszlo

1 0

[Kamailio-Users] Compiling nonce.o (Auth Module) with TLS=1 fails
by Martin Koenig 30 Nov '09

30 Nov '09

Hello all, I am trying to compile kamailio 1.5.3 with TLS=1. The whole kamailio package compiles just fine, with the exception of nonce.o: gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"auth"' -DNAME='"kamailio"' -DVERSION='"1.5.3-tls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DUSE_LOCAL_ROUTE -DF_MALLOC -DUSE_TLS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/ssl/include -I/usr/local/include -I/usr/include/openssl -c nonce.c -o nonce.o In file included from /usr/include/openssl/evp.h:84, from /usr/include/openssl/x509.h:67, from /usr/include/openssl/ssl.h:177, from ../sl/../../tls/tls_config.h:34, from ../sl/../../version.h:29, from ../sl/../../sr_module.h:54, from ../sl/sl_api.h:33, from auth_mod.h:39, from nonce.c:40: /usr/include/openssl/md5.h:105: error: conflicting types for âMD5_CTXâ ../../md5.h:42: error: previous declaration of âMD5_CTXâ was here nonce.c: In function âcalc_nonceâ: nonce.c:116: warning: passing argument 1 of âMD5Initâ from incompatible pointer type nonce.c:126: warning: passing argument 1 of âMD5Updateâ from incompatible pointer type nonce.c:128: warning: passing argument 1 of âMD5Updateâ from incompatible pointer type nonce.c:129: warning: passing argument 2 of âU_MD5Finalâ from incompatible pointer type make: *** [nonce.o] Error 1 Do you guys have any hints on what is going wrong? Best regards, Martin Koenig

1 0

[Kamailio-Users] Can kamailio bypass rtp media to backend sip server?
by Lei Tang 30 Nov '09

30 Nov '09

Hi everyone, I'm new to kamailio. I plan to setup a sip server cluster, Does someone can give me some suggestions if I can use kamailio as front server, which handle sip message and bypass rtp media message to backendserver, this mean the kamailio handle only sip message, the rtp packets are sent directed to backend server. The sip sequences is 1.softswitch send invite to kamailio 2.kamailio send invite to backend server. 3.backend server return 200 ok to kamailio 4.kamailio return 200 ok to softswitch, the ip and port in sip is from the backend server's msg, this make the softswitch send rtp direct to backend server. 5.rtp packets are transfered between softswitch and backend server. .... 6.backend server send bye to kamailio 7.kamailio send bye to softswitch. -- Lei.Tang lei.tlfly(a)gmail.com

1 0

[Kamailio-Users] Adding Columns to Database
by JR Richardson 29 Nov '09

29 Nov '09

Hi All, Using openser 1.1.0-notls (i386/linux) from debian etch package. Working with mysql database integration, working fine, using permissions module with 'trusted' table and alias_db module with the 'dbalias' table. I created the database and tables with openser_mysql cmd and things seemed to work out of the box. I ran into trouble when I dropped the database and manually created just the 'trusted' and 'dbalias' tables and added a few more columns in the tables [accountcode,notes,id(auto increment)]. When I restarted openser, no go, was getting table version and structure errors. So my question: is the table structure strict as per the openser_mysql script and are all 19 tables required for a properly running instance? I have not tried yet, but would it break if I added the new columns to the openser_mysql script created tables? Thanks. JR -- JR Richardson Engineering for the Masses

2 2

[Kamailio-Users] compile error for perl module on Fedora 11
by Zen Kato 28 Nov '09

28 Nov '09

I have installed the following packages on Fedora 11: perl-5.10.0-82.fc11.i586 perl-devel-5.10.0-82.fc11.i586 perl-libs-5.10.0-82.fc11.i586 perl-LDAP-0.34-5.fc11.noarch perl-IPC-Shareable-0.60-9.fc11.noarch But, when I compile perl modules: # make include_modules="jabber db_mysql perl perlvdb" all ....(snip)....... openserxs.c:1968: error: ‘PL_scopestack_ix’ undeclared (first use in this function) openserxs.c:1969: error: ‘XSRETURN_YES’ undeclared (first use in this function) make[1]: ディレクトリ `/usr/local/src/kamailio/1.5.3/kamailio-1.5.3-notls/modules/perl' から出ます make: *** [modules] エラー 2 What modules should I install on Fedora 11? Regards, Zen

1 1

Problems with PRESENCE/DIALOG functionality in sip-router
by klaus.feichtinger＠gmx.net 27 Nov '09

27 Nov '09

Hi all, I've tested the current release candidate for Kamailio these days and found that the presence/dialog state functionality is not given in the current version. The first (and worst) difference to the latest stable release is that the handling of "setflag(X)" for tracing the dialogs resulted in a crash of kamailio with following error message: ********************** EXCERPT ****************** Nov 24 13:35:47 lennysrv /usr/local/sbin/kamailio[13814]: : tm [t_hooks.c:190]: BUG:tm:register_tmcb: callback type TMCB_REQUEST_IN can't be register along with types Nov 24 13:35:47 lennysrv /usr/local/sbin/kamailio[13814]: ERROR: dialog [dlg_handlers.c:615]: failed cache in T the shortcut to dlg Nov 24 13:35:53 lennysrv /usr/local/sbin/kamailio[13815]: NOTICE: acc [acc.c:275]: ACC: call missed: timestamp=1259134133;method=INVITE;from_tag=4051501197-3958320-1259134127622;to_tag=3762694574-3958280-1259134127622;call_id=140363298-3958320-1259134127622(a)10.16.48.60;code=603;reason=Decline;src_user=5003;src_domain=10.16.48.205;dst_ouser=5002;dst_user=5002;dst_domain=10.16.48.60 Nov 24 13:35:56 lennysrv /usr/local/sbin/kamailio[13815]: INFO: pua_usrloc [ul_publish.c:203]: should not send ul publish Nov 24 13:35:56 lennysrv /usr/local/sbin/kamailio[13815]: INFO: <script>: --- pua-set-publish() was executed --- Nov 24 13:35:58 lennysrv kernel: [176699.348059] kamailio[13821]: segfault at c0c0c0c0 ip b7de43b3 sp bfb03eb8 error 5 in libc-2.7.so[b7d6e000+155000] Nov 24 13:35:58 lennysrv /usr/local/sbin/kamailio[13837]: : <core> [pass_fd.c:283]: ERROR: receive_fd: EOF on 12 Nov 24 13:35:58 lennysrv /usr/local/sbin/kamailio[13812]: ALERT: <core> [main.c:721]: child process 13821 exited by a signal 11 Nov 24 13:35:58 lennysrv /usr/local/sbin/kamailio[13812]: ALERT: <core> [main.c:724]: core was generated Nov 24 13:35:58 lennysrv /usr/local/sbin/kamailio[13812]: INFO: <core> [main.c:736]: INFO: terminating due to SIGCHLD Nov 24 13:35:58 lennysrv /usr/local/sbin/kamailio[13814]: INFO: <core> [main.c:787]: INFO: signal 15 received *************************************************** However, the module documentation of the source code (READMEs) does not include major news. The configuration is "the same" (regarding handle_publish, handle_subscribe, pua_set_publish) as I've used in the kamailio-1.5.3 version. But - no NOTIFY messages with XML bodies are sent to the subscribers / watchers. Instead of using the setflag(X) method I´ve tried using the method "dlg_setflag(X)" for prohibiting crashes - that was accepted by kamailio. Were these modules tested (in combination with the dialog module) in the new version 3.0 RC1? - presence - presence_xml - presence_dialoginfo - pua - pua_dialoginfo - pua_usrloc It does not work in the SER variant of sip-router, too. Yesterday I've downloaded the current version with GIT, compiled the code, adapted the configuration. But the result is the same as with kamailio-3 RC1 (crash and no XML). For being sure that I have not made any mistake in the configuration, I've tested the functionality in the version 1.5.3 of Kamailio today. It works fine - the subscribers receive NOTIFY messages with XML body after every PRESENCE and DIALOG state change. Can anybody give me a hint? Thanks in advance! regards, Klaus P.S. the tested versions: PROBLEMS: ======== version: kamailio 3.0.0-rc1 (i386/linux) 78f069 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. @(#) $Id$ main.c compiled on 13:34:37 Nov 20 2009 with gcc 4.3.2 version: ser 2.99.99-pre3 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. @(#) $Id$ main.c compiled on 11:12:49 Nov 23 2009 with gcc 4.3.2 SUCCESS: ======= version: kamailio 1.5.3-notls (i386/linux) flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 13:31:21 Nov 24 2009 with gcc 4.3.2 -- Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3.5 - sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser

3 4

Looking for documentation with SER 2.0
by inge 27 Nov '09

27 Nov '09

Hello here, I'm testing SER 2.0 on a dev plateform. I see that there's a lot of modifications beside SER 0.9.9. But I have a lot of pain to find documentation like SER Getting Started for 0.9.x to be able to understand the new routing method (ie. ser.cfg) and the databases scheme and all other parameters at the top of the config file. I install SERWeb too, the version 2.1, and it looks like very different too (don't know how to create an admin user, separate greeting/sip password, etc.). What's your recommandation about SERWeb version with SER2.0 ? Regards, Adrien .L

3 4

[Kamailio-Users] Inbound NAT detection and fix_nated_register() very poorly documented
by Alex Balashov 27 Nov '09

27 Nov '09

Greetings, I was faced with the following problem in Kamailio 1.5.2: I was using nathelper + rtpproxy + registrar and needed to relay media for calls going inbound to a NAT'd registrant through rtpproxy - only if the registrant is NAT'd. Here is the scenario: Media gateway Kamailio (on public IP) ----> registrar -----> NAT'd registrant (w/nathelper) I was handling registrations from NAT'd endpoints like this: if(nat_uac_test("1")) { force_rport(); fix_nated_contact(); } I was aware of fix_nated_register() but was not clear on its relationship to the 'received_param' parameter stored in 'location' and set as a modparam to both the 'registrar' and 'nathelper' modules, so I was not using it. Very quickly I ran into an obvious problem: if the contact stored in the 'contact' column in 'location' is already fixed up with the received IP:port, there is no way to know that the endpoint to which the call is going is behind NAT -- when lookup() is called, the RURI is set to the public IP:port. There are no flags of any kind that one can set during save() that persist while the contact binding is present for that AOR and can be resurrected on lookup(). Anyway, I eventually figured out how to fix this problem by empirical means, with no clear help from the nathelper documentation. It turns out that if I set the 'received_param' as a modparam to 'nathelper' and 'registrar' and handle registrations from NAT'd endpoints with fix_nated_register() instead, it will magically work. I made the following discoveries to arrive at this conclusion, both of which are not documented. This is why it will work: 1) When the original (RFC1918) 'contact' is stored in the 'location' table (if using DB, which I am), the 'received' parameter is stored alongside it and contains the public IP:port. When lookup() is called, the RURI domain is set to private address, e.g. if(!lookup("location")) { # Error handling here exit; } xlog("L_INFO", "[R-2:$ci] -> Registration resolved to RURI: $ru\n"): $ru here will contain something like this: sip:s@10.1.0.2:5060, the original and unmodified contact supplied by the UAC. But somehow, magically, when t_relay() is called the request will be relayed to a different RURI - one with the 'received' IP:port substituted in the domain portion. So, the request goes end up going to the correct place. This is, of course, because of the way the 'received' parameter is supposed to work when appended to a Contact URI. But the point is that it is not documented; nowhere in the documentation for lookup() or t_relay() does it say that this will transpire, and I have no way of knowing it except by observation. 2) For some reason, nat_uac_test("1") returns a positive result after the lookup() and confirms that the destination is NAT'd. This allows me to set a flag and then mangle the SDP in the reply to use rtpproxy for NAT traversal of media as well, e.g. route[2] { ... if(!lookup("location")) { # Error handling here exit; } if(nat_uac_test("1")) setflag(9); ... t_on_reply("1"); if(!t_relay()) sl_reply_error(); ... } ... onreply_route[1] { if(t_check_status("(180|183|200)")) { if(nat_uac_test("5")) fix_nated_contact(); if(search("Content-Type: application/sdp")) { if(isflagset(9)) { set_rtp_proxy_set("1"); force_rtp_proxy(); } } } } But if you examine carefully the documentation for nat_uac_test(), it says the following for bit flag 1: * 1- Contact header field is searched for occurrence of RFC1918 addresses. To me this means that nat_uac_test() should not work after the lookup() above. The only Contact header value that is present in the inbound INVITE handler is the Contact URI of the media gateway, which is a public IP address! Yet for some reason it works, as if by magic; apparently it is somehow implicit that a hidden "received" attribute of the RURI is also part of this check. This is also not documented, and is completely counterintuitive. Is there any possibility of clearing up documentation as to this point? Perhaps I have done something wrong here unknowingly; I have no idea. I know that my solution works but I cannot justify why in terms of the documentation. Is it very much to ask that the documentation be explicit about hidden but critically important mysteries like this? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

2 6

[Kamailio-Users] tmx.so: undefined symbol: _tm_table
by Andrey Utkin 27 Nov '09

27 Nov '09

Hello all. I try to use config from http://sip-router.org/wiki/migration/kamailio-3.0-config I use a kamailio compiled from git://git.sip-router.org/sip-router, branch kamailio_3.0, commit 9b482424cb8aff4050426033908e7e243f03ae83 Kamailio fails to launch with such error: root@oba:~# kamailio 0(24231) DEBUG: <core> [cfg.y:1551]: loading modules under /usr/lib/kamailio loading modules under /usr/lib/kamailio 0(24231) DEBUG: <core> [cfg.y:1544]: loading module modules/db_mysql.so 0(24231) DEBUG: <core> [sr_module.c:349]: load_module: module file not found <modules/db_mysql.so> 0(24231) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </usr/lib/kamailio/modules/db_mysql.so> 0(24231) DEBUG: <core> [cfg.y:1544]: loading module modules_k/mi_fifo.so 0(24231) DEBUG: <core> [sr_module.c:349]: load_module: module file not found <modules_k/mi_fifo.so> 0(24231) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </usr/lib/kamailio/modules_k/mi_fifo.so> 0(24231) DEBUG: <core> [cfg.y:1544]: loading module modules_k/kex.so 0(24231) DEBUG: <core> [sr_module.c:349]: load_module: module file not found <modules_k/kex.so> 0(24231) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </usr/lib/kamailio/modules_k/kex.so> 0(24231) DEBUG: <core> [cfg.y:1544]: loading module modules_k/sl.so 0(24231) DEBUG: <core> [sr_module.c:349]: load_module: module file not found <modules_k/sl.so> 0(24231) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </usr/lib/kamailio/modules_k/sl.so> 0(24231) DEBUG: <core> [cfg.y:1544]: loading module modules/tm.so 0(24231) DEBUG: <core> [sr_module.c:349]: load_module: module file not found <modules/tm.so> 0(24231) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </usr/lib/kamailio/modules/tm.so> 0(24231) DEBUG: <core> [cfg.y:1544]: loading module modules_k/tmx.so 0(24231) DEBUG: <core> [sr_module.c:349]: load_module: module file not found <modules_k/tmx.so> 0(24231) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </usr/lib/kamailio/modules_k/tmx.so> 0(24231) ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/lib/kamailio/modules_k/tmx.so>: /usr/lib/kamailio/modules_k/tmx.so: undefined symbol: _tm_table 0(24231) : <core> [cfg.y:3210]: parse error in config file //etc/kamailio/kamailio.cfg, line 117, column 12-29: failed to load module .... Has anybody succeed to load this module from last git sources?

2 1

[Kamailio-Users] Presence
by Andrey Utkin 27 Nov '09

27 Nov '09

Hello, all. My objective is to make a kamailio installation where phones call to each other by number and are able to monitor others using SUBSCRIBE. I use Kamailio 3.0.0 rc1 pulled from git. My config file is http://pastebin.ca/1675069 I had bad results with establishing calls, possibly because i frequently restarted kamailio, only once i had INVITE passing from one phone to other. RTPProxy mustn't work on a box with grey IP, i know. But my greatest convern now is notifications. I just want to say that i never got NOTIFYs with xml containing bodies, what was usual in Asterisk. First of all NOTIFYs that go immediately after SUBSCRIBEs also have no xml body. Please direct me what is wrong in my configuration regarding presence setup. Tell me what system info to provide.

2 2

[Kamailio-Users] we don't follow the RFCs!
by BERGANZ François 27 Nov '09

27 Nov '09

Hello all, As I could see in the source, we dont follow the RFCs. If I send an INVITE, kamailio response a Giving a try It isnt in the RFCs! It is normaly Trying. I could change it in -src/modules/tm/t_funcs.c -line: static str relay_reason_100 = str_init("Giving a try"); I hope you understand what I mean François BERGANZ Ingénieur R&D VoIP Acropolis Telecom P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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Re: [Kamailio-Users] i want to get user defined header value
by Alex Balashov 27 Nov '09

27 Nov '09

Please don't forget to copy your responses back to the list so that others can benefit from this discussion. I must have misunderstood your intended meaning of "user-defined." $hdr(...) takes only string literals as arguments, not pseudovariables. I do not believe that what you are wanting is possible. RAJNIKANT VANZA wrote: > Hi Alex, > > i have tested like : > > 1. ${hdr(TEST)) > 2. $hdr($avp(s:TEST)) > > but it's not work. > if u know any other suggestion, send me. > > On Mon, Nov 23, 2009 at 3:56 PM, Alex Balashov > <abalashov(a)evaristesys.com <mailto:abalashov@evaristesys.com>> wrote: > > RAJNIKANT VANZA wrote: > > i want to get user-defined header value. > e.g. i have create header name : TEST and its value : vtec in > asterisk. > then how can i get this value from this header in kamailio. > > > Try: $hdr(TEST) > > -- > Alex Balashov - Principal > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > > > > > -- > Thanks and Regards > Rajnikant Vanza > Call : +91-97374 56583 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

5 11

[Kamailio-Users] prb while loading presence.so
by ioanna lampropoyloy 27 Nov '09

27 Nov '09

Openser cant start while i am trying to load presence.so I get the following error Nov 24 12:57:11 localhost openser: ERROR: load_module: could not open module <//lib/openser/modules/presence.so>: //lib/openser/modules/presence.so: undefined symbol: ctime_buf Nov 24 12:57:11 localhost openser: parse error (58,13-14): failed to load module and i also get errors which refer to the parameters of presence since the module cant be loaded. Any ideas?

2 1

[Kamailio-Users] delete the SDP?
by BERGANZ François 27 Nov '09

27 Nov '09

Hello, How can I delete all the SDP ? Thank you François BERGANZ P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

3 2

how i should process replies on nathelper pings?
by Andres Moya 27 Nov '09

27 Nov '09

Hi. nathelper sending empty OPTIONS messages to clients. It have only one VIA, server's IP address(195.167.152.55). Then it got reply i have: U 195.167.152.55:5060 -> 81.195.165.250:5060 OPTIONS sip:81.195.165.250:5060 SIP/2.0..Via: SIP/2.0/UDP 195.167.152.55:5060;branch=0..From: sip:pinger@riki.ru;tag=e64c73d5..To: sip:81.195.165.250:5060..Call-ID: 680bed34-d2e5c741-0eff517(a)195.167.152 .55..CSeq: 1 OPTIONS..Content-Length: 0.... # U 81.195.165.250:5060 -> 195.167.152.55:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 195.167.152.55:5060;branch=0..Contact: <sip:10.0.52.103:5060>..To: <sip:81.195.165.250:5060>;tag=021c6509..From: <sip:pinger@riki.ru>;tag=e64c73d5..Call-ID: 680bed34-d2e 5c741-0eff517@195.167.152.55..CSeq: 1 OPTIONS..Accept: application/sdp, application/sdp..Accept-Language: en..Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE..User-Agen t: Zoiper rev.5324..Allow-Events: presence..Content-Length: 0.... Nov 25 23:59:26 v /usr/local/sbin/kamailio[9697]: ERROR: <core> [forward.c:695]: ERROR: forward_reply: no 2nd via found in reply # U 195.167.152.55:5070 -> 86.154.150.209:5060 OPTIONS sip:86.154.150.209:5060 SIP/2.0..Via: SIP/2.0/UDP 195.167.152.55:5070;branch=0..From: sip:pinger@riki.ru;tag=f64c73d5..To: sip:86.154.150.209:5060..Call-ID: 680bed34-e2e5c741-7eff517(a)195.167.152 .55..CSeq: 1 OPTIONS..Content-Length: 0.... # U 86.154.150.209:5060 -> 195.167.152.55:5070 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 195.167.152.55:5070;branch=0..To: <sip:86.154.150.209>;tag=931uruqh0thc6crbneub..From: <sip:pinger@riki.ru>;tag=f64c73d5..Call-ID: 680bed34-e2e5c741-7eff517(a)195.1 67.152.55..CSeq: 1 OPTIONS..Content-Length: 0.... Nov 25 23:59:33 v /usr/local/sbin/kamailio[9698]: ERROR: <core> [forward.c:695]: ERROR: forward_reply: no 2nd via found in reply Reply should be consumed by nathelper? or i should eat it somewhere? i can't do so in request route. and i can't use reply route because i can't set t_on_reply, it is not my request. There should be easy way... tired get ERROR: forward_reply: no 2nd via found in reply thanks

3 7

[Kamailio-Users] Dialplan regular expressions
by alex pappas 26 Nov '09

26 Nov '09

Dear all, I'm having difficulties in the dialplan module. My problem is in the regular expressions added to the database. I tried some examples from previous users but they don't work for me. example: dpid 10 pr 100 match_op 0 match_exp ^123(.*)$ match_len 0 subst_exp ^123 repl_exp 0030 Previous example from Daniel that I found in user lists: s/subst_exp/repl_exp/ subst_exp - is regular expression that can be used to group parts of the input repl_exp - is used to build the output Example: you want to remove the first 0 and prefix +44 to a number: subst_exp=^0(.+)$ repl_exp=+44\1 Can someone provide an example that works with the latest kamailio please? Also where I can find more docs about dialplan? Thank you all Alex

3 7

[Serusers] newbie
by Carlos Felipe Solano Fernandez 26 Nov '09

26 Nov '09

Greetings. I had to read some thigs about SER. I like configure a sip server but I don't know that I need. I like to take calls (pc-phone)(ip phone-phone) from my customers, and resend them to a voip provider. and I need a billing system (preaid), to control my customers, I need to configure several rates for calls. Can somebody tell me that I need to make this? I'm sorry but my english isn't very well. (I speak spanish) Atte, Felipe Solano Explore the seven wonders of the world Learn more! _________________________________________________________________ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx

1 0

[Kamailio-Users] RTCP
by kamailio.org＠spam.lublink.net 26 Nov '09

26 Nov '09

Hey, I was wondering how I can log all the RTCP messages and automatically associate them with the CDRs? Thanks, David

5 6

[Kamailio-Users] tm retransmits and times out even though reponse is received
by Alex Hermann 26 Nov '09

26 Nov '09

Hello, I'm seeing an increasing amount of spurious timeouts on my proxies. Even though the proxy receives a "100 Trying" response immediately after relaying the request, tm will send retransmits and times out eventually. Below is a commented log snippet. The timeout occurs on the proxy named 'balancer1'. Both are on the same host. I verified that the 100 Trying response is received in a t_on_reply block, so it is recognized by tm as belonging to a transaction. When needed, I'll provide a debug log when it occurs again. Meanwhile, any idea what is causing this? This is on kamailio 1.5+svn5768 fr_timer = 10 fr_timer_avp = 2 # Initial request received on balancer May 15 12:32:01 balancer1[24170]: [2 INVITE] - New request. <sip:uac@domain> => <sip:uas@domain> May 15 12:32:01 balancer1[24170]: [2 INVITE] Relaying # Initial request received on proxy May 15 12:32:01 proxy1[23803]: [2 INVITE] - New request. <sip:uac@domain> => <sip:uas@domain> # Balancer receives 100 response May 15 12:32:01 balancer1[24166]: [2 INVITE] - Reply. 100 Trying # Proxy relays request May 15 12:32:01 proxy1[23803]: [2 INVITE] Relaying # Proxy receives retransmission from balancer ??? May 15 12:32:02 proxy1[23804]: [2 INVITE] - New request. <sip:uac@domain> => <sip:uas@domain> # Balancer times out ??? May 15 12:32:02 balancer1[24171]: [2 INVITE] Failure (Status: 408) May 15 12:32:02 balancer1[24171]: [2 INVITE] No more gateways May 15 12:32:02 proxy1[23804]: [2 CANCEL] - New request. <sip:uac@domain> => <sip:uas@domain> -- Greetings, Alex Hermann

2 4

[Kamailio-Users] DSYSLOG_ASYNC option in Makefile.defs
by Torben Friese 26 Nov '09

26 Nov '09

Hi all, I have a problem with the Kamailio 1.5, if the DSYSLOG_ASYNC option is set I receive no log messages on a lenny-64bit system(fresh installed). Rsyslog is the standard log daemon in lenny. If I start directly from the command line with -dd I receive log messages. Any hints or ideas what is going wrong here ? The same option on a etch system works fine. best regards and thanking you in anticipation Torben

2 1

[Kamailio-Users] Commercial 24x7x365 Support
by Martin Werne 26 Nov '09

26 Nov '09

Hi! I have a customer (service-provider) that want to implement redundant openSER type of solution. On two locations (Europe and USA). They need access to high level support 24 hours a day, 365 days a year (by phone and remote-assistance via VPN). Do you know of any companies that delivers this kind of service? Anyone that has a list of companies that delivers solutions based on openSER? As I'm only one man, I'm not able to do this by myself...... ;-) Best regards, Martin

2 1

[Kamailio-Users] Core dump on xl_get_pai in Openser 1.2.2
by Marcello Lupo 26 Nov '09

26 Nov '09

Hi to all, i implemented the P-Asserted-Identity and P-Preferred-Identity Header management. On the development server all is working properly but when i activate the new config on our production system i start to get crashes of the openser process: openser[31483]: segfault at 0000000000000000 rip 0000000000487853 rsp 00007fff94c9f1a0 error 4 openser[6511]: segfault at 0000000000000025 rip 00000000004214ce rsp 00007fff80f05480 error 4 openser[8967] general protection rip:4214ce rsp:7fff05fbb530 error:0 openser[9872] general protection rip:4214ce rsp:7fffd2b630d0 error:0 I checked the core dump with gdb and i have this info from the core: Core was generated by `/usr/sbin/openser'. Program terminated with signal 11, Segmentation fault. #0 xl_get_pai (msg=0x2b4ad9030d00, res=0x7fffd2b63190, param=<value optimized out>, flags=<value optimized out>) at items.c:1050 1050 res->rs.len = get_pai(msg)->uri.len; We are using the 1.2.2 version and i know it is quite old but i like to understand if it happen for some known bug or if i made something wrong. For the moment is not possible for us to manage the upgrade to a newer version of the proxy till we will not have it redundant on another location. Thank you for help. Bye, MArcello

2 3

[Kamailio-Users] Grails Runtime Exception while adding user with SerMyAdmin
by Fred 25 Nov '09

25 Nov '09

Hello, when i push create button, I get the following: Grails Runtime Exception Error Details Message: org.apache.commons.codec.digest.DigestUtils Caused by: java.lang.NoClassDefFoundError: org.apache.commons.codec.digest.DigestUtils Class: UserController At Line: [226] Code Snippet: Stack Trace org.codehaus.groovy.runtime.InvokerInvocationException: java.lang.NoClassDefFoundError: org.apache.commons.codec.digest.DigestUtils at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:290) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationDispatcher.invoke(ApplicationDispatcher.java:630) at org.apache.catalina.core.ApplicationDispatcher.processRequest(ApplicationDispatcher.java:436) at org.apache.catalina.core.ApplicationDispatcher.doForward(ApplicationDispatcher.java:374) at org.apache.catalina.core.ApplicationDispatcher.forward(ApplicationDispatcher.java:302) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.jsecurity.web.servlet.JSecurityFilter.doFilterInternal(JSecurityFilter.java:384) at org.jsecurity.web.servlet.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:183) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:233) at org.apache.catalina.core.StandardContextValve.invoke(StandardContextValve.java:175) at org.apache.catalina.core.StandardHostValve.invoke(StandardHostValve.java:128) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:102) at org.apache.catalina.core.StandardEngineValve.invoke(StandardEngineValve.java:109) at org.apache.catalina.connector.CoyoteAdapter.service(CoyoteAdapter.java:286) at org.apache.coyote.http11.Http11Processor.process(Http11Processor.java:844) at org.apache.coyote.http11.Http11Protocol$Http11ConnectionHandler.process(Http11Protocol.java:583) at org.apache.tomcat.util.net.JIoEndpoint$Worker.run(JIoEndpoint.java:447) at java.lang.Thread.run(Thread.java:619) Caused by: java.lang.NoClassDefFoundError: org.apache.commons.codec.digest.DigestUtils at UserController$_closure11.class$(UserController.groovy) at UserController$_closure11.doCall(UserController.groovy:226) at UserController$_closure11.doCall(UserController.groovy) ... 28 more Unfortunately, i am not familiar with such java stuff. It would be great if one of you can give me a hint or something. I dont have any idea how to solve this problem. Thanks for any comment! Best regards, Fred

3 2

[Kamailio-Users] /usr/local/sbin/kamdbctl create error
by vivi 25 Nov '09

25 Nov '09

Hi all I have specify the wanted db type (DBENGINE=MYSQL) in the /usr/local/etc/kamailio/kamctlrc then I using "/usr/local/sbin/kamdbctl create" to create MySQL database, but I got this error: ERROR: database engine not specified, please setup one in the config script root@acer:/usr/local/src/kamailio-1.5.0/sip-server# vim /usr/local/etc/kamailio/kamctlrc root@acer:/usr/local/src/kamailio-1.5.0/sip-server# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR: Creating core database and grant privileges failed! Cheers, vivi

3 2

[Kamailio-Users] Call duration with RADIUS support
by mbsip 25 Nov '09

25 Nov '09

Hi, I am struggling with kamailio and radius configuration to achieve proper Call duration notification. Here is what i have: - kamailio 1.6 - radiusclient-ng - freeradius on different server Kamailio configuration has following radius-extra: modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$ruri; \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") which produce following radius data START and STOP: Mon Nov 23 16:26:38 2009 Acct-Status-Type = Start Service-Type = IAPP-Register Sip-Response-Code = 200 Sip-Method = Invite Event-Timestamp = 1258990835 Sip-From-Tag = "c3716c01" Sip-To-Tag = "11544D08-C80" Acct-Session-Id = "Nzg5MmZlMDNkMGUwODgwN2ZkZWZkMmQ3NzYwYzNkNWY." User-Name = "Mac(a)xx.yy.zz.vv" Calling-Station-Id = "sip:Mac@xx.yy.zz.vv" Called-Station-Id = "sip:123456789@xx.yy.zz.vv" Sip-Translated-Request-URI = "sip:123456789@aa.bb.cc.dd" X-Ascend-Send-Secret = 0x3135372e32352e3139322e32 X-Ascend-Receive-Secret = 0x3a7c X-Ascend-Netware-timeout = 0x3c7369703a4d616369656a403139322e3136382e332e3132353a36323034323e NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = xx.yy.zz.vv Client-IP-Address = xx.yy.zz.vv Acct-Unique-Session-Id = "a62f94528d34acfe" Timestamp = 1258989998 Mon Nov 23 16:26:38 2009 Acct-Status-Type = Stop Service-Type = IAPP-Register Sip-Response-Code = 200 Sip-Method = 8 Event-Timestamp = 1258990836 Sip-From-Tag = "c3716c01" Sip-To-Tag = "11544D08-C80" Acct-Session-Id = "Nzg5MmZlMDNkMGUwODgwN2ZkZWZkMmQ3NzYwYzNkNWY." User-Name = "Mac(a)xx.yy.zz.vv" Calling-Station-Id = "sip:Mac@xx.yy.zz.vv" Called-Station-Id = "sip:123456789@xx.yy.zz.vv" Sip-Translated-Request-URI = "sip:123456789@aa.bb.cc.dd:5060" X-Ascend-Send-Secret = 0x3135372e32352e3139322e32 X-Ascend-Receive-Secret = 0x3a7c X-Ascend-Netware-timeout = 0x3c7369703a4d616369656a403139322e3136382e332e3132353a36323034323e NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = xx.yy.zz.vv Client-IP-Address = xx.yy.zz.vv Acct-Unique-Session-Id = "a62f94528d34acfe" Timestamp = 1258989998 The task is to upload all stored radius messages into MYSQL with proper call duration info. I've ran through many posts and found a few way-outs: 1) to use AVPOPS module: modparam("acc", "radius_extra", "Acct-Session-Time=$avp(call_length)") modparam( "avpops", "avp_aliases", "start_timestamp=i:100") modparam( "avpops", "avp_aliases", "end_timestamp=i:101") modparam( "avpops", "avp_aliases", "call_length=i:102") avp_op("$avp(end_timestamp)","sub/$avp(start_timestamp)"); avp_op("$avp(end_timestamp)/$avp(call_length)","sub/$avp(start_timestamp)"); 2) to use DIALOG module modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_extra", "Acct-Session-Time=$avp(i:921)") if (is_method("BYE")) { xlog("L_INFO", "dialog_status:$DLG_status duration=$DLG_lifetime\n"); $avp(i:921) = $DLG_lifetime{s.int}; } else { $avp(i:921) = 0; } setflag(1); 3) to use a script or db triggers(after STOP). Could you advice which one is better/work properly. Maybe there are any other configuration i could use. Thanks in advance, Maciej

2 2

[Kamailio-Users] Same "RADIUS Problem"
by Masoom Alam 25 Nov '09

25 Nov '09

I am having the same problem: I am configuring RADIUS with openser and CDRtool. when i dial a number it loges into in mysql of RADIUS server but it gives the following error. and when i check CDRs of CDRtool it shows all calls "*inprogress". * Tue Nov 24 19:47:37 2009 : Error: rlm_sql (sql) in sql_accounting: stop packet with zero session length. [user '1000@ip address'<%271000@ip%20address%27>, nas 'ipaddress'] Can any body help me, that whether the RADIUS server is not sending the INVITE and BYE traffic correctly or not singallng the CDRTool as well. Thanking you in advance.

2 1

[Kamailio-Users] RADIUS problem
by toqeer ali 24 Nov '09

24 Nov '09

I am configuring RADIUS with openser and CDRtool. when i dial a number it loges into in mysql of RADIUS server but it gives the following error. and when i check CDRs of CDRtool it shows all calls "*inprogress". * Tue Nov 24 19:47:37 2009 : Error: rlm_sql (sql) in sql_accounting: stop packet with zero session length. [user '1000@ip address'<'1000@ip%20address'>, nas 'ipaddress'] -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

1 0

[Kamailio-Users] please help
by Masoom Alam 24 Nov '09

24 Nov '09

Hi i am configuring RADIUS accounting with openser and i facing the problem with radius_extra module in acc all other parameters are ok but when i add $hdr something i got the following error. parameters with avp are fine but problem is with $hdr. modparam("acc", "radius_extra", ......; X-RTP-Stat=$hdr (X-RTP-Stat; ......) Nov 24 13:30:50 kamailio /sbin/openser[3760]: xl_fill_extra_spec: extra item [hdr] not found Nov 24 13:30:50 kamailio /sbin/openser[3760]: xl_lookup_spec_name: not found PV [hdr] Nov 24 13:30:50 kamailio /sbin/openser[3760]: ERROR:xl_parse_spec: error searching pvar "hdr" Regards, MM Alam SERG Pakistan http://serg.imsciences.edu.pk http://www.imsciences.edu.pk/profiles/masoom_alam.html

2 1

[Kamailio-Users] Strip core function with variable
by Antonio Reale 24 Nov '09

24 Nov '09

Hi all, strip core function accepts only an integer as parameter. What do you thing about implementing the strip function with a variable as parameter in a future release? strip($var(x)); or something like that... Is this doable? Best regards. -- Antonio

2 2

Re: [SR-Users] [sr-dev] recursive calls to failure_route strange behavior
by Miklos Tirpak 23 Nov '09

23 Nov '09

[please keep the list CC-d because others may also be interested in the solution or may know the answer better.] On 11/20/2009 07:06 PM, Andres Moya wrote: > Can i ask one more question here. It is complicated, i am using > uac_replace_from and uac_auth. > I am using failure route to process authentication if necessary and > redirect on next carrier. > > If i authenticated with one provider, then fot let say 415 ( i set only > speex in UAC to get it ;) ). Ok SER send request to second provider, i use > uac_replace_from once again in my LOAD_AUTH route > then uac_auth again. > > Ok. now i see from ngrep that uac_replace_from did nothing in from field > and use user@domain for first provider, authentication failed :( Which authentication fails? The first or the second one? The first should work, at least the from header should be rewritten by the function. The second authentication will not work this way (if it requires a different from header) because the proxy "remembers" for the header changes done before the first t_relay() function call and applies the same header modifications also for any other branch added from failure route. Hence, the outgoing SIP request to the second provider will contain the same from HF as the request to the first provider. The easiest way is to apply the header modifications in branch route if you do not need to reuse them later from failure route. Modifications done in branch routes are valid only within that branch. > > I moved uac_replace_from to my failure route to call once again before > uac_auth, but got config error as i can't use uac_replace_from in > failure_route. I think you already use this function from failure route because it is in a route block that is included from failure route. The only difference is that the syntax checker does not recognize the issue. I am not familiar with uac_replace_from() but after having a quick look at the function I think it is safe to use it from here. > Ok i will use textops to rewrite, but it is ugly? The main difference is that uac_replace_from() restores the original From HF when the response if forwarded. If you use textops module then you need to restore the header manually. > no. Maybe i should > call uac_replace_from in branch route? I would suggest this way. Both for the first and for the second provider. Miklos > > Thanks

3 6

Re: [Kamailio-Users] i want to get user defined header value
by RAJNIKANT VANZA 23 Nov '09

23 Nov '09

On Mon, Nov 23, 2009 at 5:54 PM, RAJNIKANT VANZA <rajnivanza(a)gmail.com>wrote: > On Mon, Nov 23, 2009 at 4:20 PM, Thiago Rondon <thiago(a)aware.com.br>wrote: > >> >> Ranikant, >> >> You can you use Perl module for this, please look at : >> >> http://kamailio.org/docs/modules/1.6.x/perl.html >> >> See the script example, headers.pl >> >> -Thiago Rondon >> >> >> Alex Balashov escreveu: >> >> Please don't forget to copy your responses back to the list so that >>> others can benefit from this discussion. >>> >>> I must have misunderstood your intended meaning of "user-defined." >>> $hdr(...) takes only string literals as arguments, not pseudovariables. >>> >>> I do not believe that what you are wanting is possible. >>> >>> RAJNIKANT VANZA wrote: >>> >>> Hi Alex, >>>> >>>> i have tested like : >>>> >>>> 1. ${hdr(TEST)) >>>> 2. $hdr($avp(s:TEST)) >>>> >>>> but it's not work. >>>> if u know any other suggestion, send me. >>>> >>>> On Mon, Nov 23, 2009 at 3:56 PM, Alex Balashov < >>>> abalashov(a)evaristesys.com <mailto:abalashov@evaristesys.com>> wrote: >>>> >>>> RAJNIKANT VANZA wrote: >>>> >>>> i want to get user-defined header value. >>>> e.g. i have create header name : TEST and its value : vtec in >>>> asterisk. >>>> then how can i get this value from this header in >>>> kamailio. >>>> >>>> >>>> Try: $hdr(TEST) >>>> >>>> -- Alex Balashov - Principal >>>> Evariste Systems >>>> Web : http://www.evaristesys.com/ >>>> Tel : (+1) (678) 954-0670 >>>> Direct : (+1) (678) 954-0671 >>>> >>>> >>>> >>>> >>>> -- >>>> Thanks and Regards >>>> Rajnikant Vanza >>>> Call : +91-97374 56583 >>>> >>> > > Hi friends, > > thank for reply. > > i have add header through asterisk api by below statement and its display > correct on asterisk CLI > but not able to get header value through kamailio: > > *asteisk statement is :* SipAddheader(TEST=vtec); > above statement in header name is "TEST" and its value is "vtec". > > i want get "header TEST" from sip message when kamailio receive sip message > (INVITE) from asterisk server. > > > -- > Thanks and Regards > Rajnikant Vanza > Call : +91-97374 56583 > -- Thanks and Regards Rajnikant Vanza Call : +91-97374 56583

1 0

[Kamailio-Users] i want to get user defined header value
by RAJNIKANT VANZA 23 Nov '09

23 Nov '09

Hi friends, i want to get user-defined header value. e.g. i have create header name : TEST and its value : vtec in asterisk. then how can i get this value from this header in kamailio. -- Thanks and Regards Rajnikant Vanza

2 1

[Kamailio-Users] how can i get sip header value
by RAJNIKANT VANZA 23 Nov '09

23 Nov '09

Hi Friends, i am a new user for kamailio. let me know, how can i get sip header value ? -- Thanks and Regards Rajnikant Vanza

2 1

[Kamailio-Users] [kamailio-users] Read SIP Header from call request
by Chandrakant Solanki 23 Nov '09

23 Nov '09

Hi I am using kamailio 1.5.0.2 How can i read SIP Header in kamailio for every sip request. Is there any method / function which will help to read SIP Header...?? -- Regards, Chandrakant Solanki

2 1

[Kamailio-Users] TCP problem/question
by Torben Friese 21 Nov '09

21 Nov '09

Hi all, I need a hint regarding TCP handling. I want to receive messages as TCP and relay or forward them as UDP. This is realized with dispatcher module. A explicit entry with sip:IP;transport=udp in the dispatcher list. Receiving via TCP works fine, but the proxy send the request also via TCP out. Any ideas where I can search or how to debug the problem ? best regards Torben

3 2

[Kamailio-Users] ERROR:mi_fifo:mi_fifo_server: fifo command must begin with :: early_dialogs (1.5 rev 5834)
by Iñaki Baz Castillo 21 Nov '09

21 Nov '09

Hi, I use the Munin plugin I did in various servers running Kamailio 1.5 (Debian 64 bits). The munin plugins performs the Mi command: kamctl fifo get_statistics inuse_transactions In one of them the installed revision is 5834 and the plugin works perfectly, no errors in 3 months with high traffic. However in the other server (revision 5925) there are no traffic yet but the plugin fails sometimes (maybe once each 2-3 days). Sometimes it gets an error: ERROR:mi_fifo:mi_fifo_server: fifo command must begin with :: early_dialogs and sometimes it gets wrong value: kamctl fifo get_statistics inuse_transactions => 2819624 (with no traffic !!!) In the second case the Munin grapsh gets corrupted due to so high value. Could it be a bug in dialog module? Should I upgrade to revision 5939 in which I see a change for modules/dialog/dlg_handlers.c: ------------------------------------------------------------------------------------------------------------------------- ~# svn diff -r HEAD modules/dialog/dlg_handlers.c Index: modules/dialog/dlg_handlers.c =================================================================== --- modules/dialog/dlg_handlers.c (revisión: 5939) +++ modules/dialog/dlg_handlers.c (copia de trabajo) @@ -824,14 +824,13 @@ if(dlg->toroute>0 && dlg->toroute<RT_NO) { + dlg_set_ctx_dialog(dlg); fmsg = faked_msg_next(); if (exec_pre_req_cb(fmsg)>0) { - dlg_set_ctx_dialog(dlg); LM_DBG("executing route %d on timeout\n", dlg->toroute); set_route_type(REQUEST_ROUTE); run_top_route(rlist[dlg->toroute], fmsg); - dlg_set_ctx_dialog(0); exec_post_req_cb(fmsg); } } ------------------------------------------------------------------------------------------------------------------------ I see some differences between the dialog module versions running in both servers, but I think the main difference is a bug fix for the case in which there is empty Record-Route in the response (Kamailio crashed before the fix): ----------------------------------------------------------------------------------------------------------------------- # svn diff -r 5834:5925 modules/dialog/dlg_handlers.c Index: modules/dialog/dlg_handlers.c =================================================================== --- modules/dialog/dlg_handlers.c (revisión: 5834) +++ modules/dialog/dlg_handlers.c (revisión: 5925) @@ -163,10 +163,10 @@ int populate_leg_info( struct dlg_cell *dlg, struct sip_msg *msg, struct cell* t, unsigned int leg, str *tag) { - unsigned int skip_recs; - str cseq; - str contact; - str rr_set; + unsigned int skip_recs = 0; + str cseq = {0, 0}; + str contact = {0, 0}; + str rr_set = {0, 0}; /* extract the cseq number as string */ if (leg==DLG_CALLER_LEG) { @@ -184,48 +184,41 @@ /* extract the contact address */ if (!msg->contact&&(parse_headers(msg,HDR_CONTACT_F,0)<0||!msg->contact)){ LM_ERR("bad sip message or missing Contact hdr\n"); - goto error0; + } else { + if ( parse_contact(msg->contact)<0 || + ((contact_body_t *)msg->contact->parsed)->contacts==NULL || + ((contact_body_t *)msg->contact->parsed)->contacts->next!=NULL ) { + LM_ERR("bad Contact HDR\n"); + } else { + contact = ((contact_body_t *)msg->contact->parsed)->contacts->uri; + } } - if ( parse_contact(msg->contact)<0 || - ((contact_body_t *)msg->contact->parsed)->contacts==NULL || - ((contact_body_t *)msg->contact->parsed)->contacts->next!=NULL ) { - LM_ERR("bad Contact HDR\n"); - goto error0; - } - contact = ((contact_body_t *)msg->contact->parsed)->contacts->uri; - /* extract the RR parts */ if(!msg->record_route && (parse_headers(msg,HDR_RECORDROUTE_F,0)<0) ){ LM_ERR("failed to parse record route header\n"); - goto error0; - } - - if (leg==DLG_CALLER_LEG) { - skip_recs = 0; } else { - /* was the 200 OK received or local generated */ - skip_recs = dlg->from_rr_nb + - ((t->relaied_reply_branch>=0)? - (t->uac[t->relaied_reply_branch].added_rr):0); - } + if (leg==DLG_CALLEE_LEG) { + /* was the 200 OK received or local generated */ + skip_recs = dlg->from_rr_nb + + ((t->relaied_reply_branch>=0)? + (t->uac[t->relaied_reply_branch].added_rr):0); + } - if(msg->record_route){ - if( print_rr_body(msg->record_route, &rr_set, leg, - &skip_recs) != 0 ){ - LM_ERR("failed to print route records \n"); - goto error0; + if(msg->record_route){ + if( print_rr_body(msg->record_route, &rr_set, leg, + &skip_recs) != 0 ){ + LM_ERR("failed to print route records \n"); + } } - } else { - rr_set.s = 0; - rr_set.len = 0; } if(leg==DLG_CALLER_LEG) dlg->from_rr_nb = skip_recs; LM_DBG("route_set %.*s, contact %.*s, cseq %.*s and bind_addr %.*s\n", - rr_set.len, rr_set.s, contact.len, contact.s, - cseq.len, cseq.s, + rr_set.len, (rr_set.len)?rr_set.s:"", + contact.len, (contact.len)?contact.s:"", + cseq.len, (cseq.len)?cseq.s:"", msg->rcv.bind_address->sock_str.len, msg->rcv.bind_address->sock_str.s); ----------------------------------------------------------------------------------------------------------------------- Perhaps a regression in rev 5925 so I should upgrade? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 13

change from header in branches works strange
by Andres Moya 21 Nov '09

21 Nov '09

Hi, it is me again. I gave up on uac_replace_from, and trying with subst('/^From:(.*)sip:[^@]*@[a-zA-Z0-9\.]+(.*)$/From:\1sip:$avp(i:21)@$avp(i:24)\2/i'); first time it substitute ok, on next branch i got result: From: <sip:andres.moya4@sipdiscount.com;transport=UDP>;tag=ee12f132.sip:andres.moya4@sipdiscount.com Something really strange, looks like there is no way i can authenticate to different SIP providers in one transaction :( Anyone know how i can hack From field actually?

1 0

crazy uac_replace_from
by Andres Moya 21 Nov '09

21 Nov '09

Hi, not sleeping second night. I am trying to authenticate to sequence of SIP provider, if one fail - another. I am calling uac_replace_from for each. uac_replace_from("sip:$avp(i:21)@$avp(i:24)"); first time it set ok: From: <sip:andres.moya4@sipdiscount.com>;tag=572ac509. second time i am getting From: <sip:andres.moya4@sipdiscount.comsip:21100036838@terrasip.net>;tag=572ac509. grrrr.... is it a joke :) Someone ... please help

1 0

[Kamailio-Users] No free TCP receiver
by luis.guaman＠interlancompu.com 19 Nov '09

19 Nov '09

3 2

[Kamailio-Users] Dialog timed calls
by Jason Penton 19 Nov '09

19 Nov '09

Hi All, Has anyone else noticed the significant inaccuracy of the timeout in the dialog module? We are using: modparam("dialog", "timeout_avp", "$avp(i:10)") However, the actual call is torn down a NUMBER of seconds past the timeout supplied. In some instances, for long calls, it is out by 30seconds or more....? For shorter calls it is accurate with a second or so. Anyone have any input, ideas? Cheers Jason

2 1

[Kamailio-Users] How to configure alias_db_lookup in config script
by JR Richardson 18 Nov '09

18 Nov '09

Hi All, I have a real simple load balancer proxy setup using the dispatcher module, works great. I want to do an alias lookup before I send the calls through to the dispatcher, but I can't seem to find a good example of how that should be setup in the config script. I've tried different things, the aliase_db_lookup is working but if a user is not in the database, the call loops and does not continue on to the dispatcher. Here is my current route script (this does not work): 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; alias_db_lookup("dbaliases"); t_on_failure("1"); t_relay(); ds_select_dst("1", "4"); t_on_failure("1"); t_relay(); } failure_route[1] { ds_next_dst(); forward(); } I think there should be an 'if' 'then' or 'if' 'else' separating the alias_db_lookup and the ds_select functions, but I'm not sure of the syntax and where the curly brackets need to be. Thanks. JR -- JR Richardson Engineering for the Masses

2 5

[Kamailio-Users] openser freeradius connectivity error
by toqeer ali 18 Nov '09

18 Nov '09

Hi all, I configured freeradius for accounting of openser. but when i make a call the call establishwa and gives this error for Freeradius. Nov 18 06:50:58 kamailio /sbin/openser[11590]: rc_ip_hostname: couldn't look up host by addr: 0A000005 Nov 18 06:50:58 kamailio /sbin/openser[11590]: rc_send_server: no reply from RADIUS server unknown:1813 Nov 18 06:50:58 kamailio /sbin/openser[11590]: ERROR:acc:acc_rad_request: radius-ing failed Nov 18 06:51:20 kamailio /sbin/openser[11593]: rc_ip_hostname: couldn't look up host by addr: 0A000005 Nov 18 06:51:20 kamailio /sbin/openser[11593]: rc_send_server: no reply from RADIUS server unknown:1813 Nov 18 06:51:20 kamailio /sbin/openser[11593]: ERROR:acc:acc_rad_request: radius-ing failed please help -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

2 3

[Kamailio-Users] Play Early Media
by alex pappas 18 Nov '09

18 Nov '09

Dear all, I'm trying to play Early Media instead of Ringing WHEN an 180 Ringing OR 182 Call Queue message is received from the destination UAC. In the onreply_route[1] I added the following lines in bold: I use kamailio 1.5 and I have NAT enabled. onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"*(180)|*(183)|(2[0-9][0-9])") { force_rtp_proxy(); } * if (status=~"(180)") { rtpproxy_stream2uac("/usr/local/etc/rtpproxy/ast-ivr.alaw",""); } else { rtpproxy_stop_stream2uac(); }* if (isbflagset(6)) { fix_nated_contact(); } } When I get from a UAC an 180 Ringing does not play the media and also in the syslog I see the following ERRORs: Nov 17 14:11:29 YGOR /usr/local/sbin/kamailio[4681]: ERROR:nathelper:extract_body: message body has length zero Nov 17 14:11:29 YGOR /usr/local/sbin/kamailio[4681]: ERROR:nathelper:force_rtp_proxy: can't extract body from the message Nov 17 14:11:29 YGOR /usr/local/sbin/kamailio[4681]: ERROR:nathelper:select_rtpp_node: script error -no valid set selected Nov 17 14:11:29 YGOR /usr/local/sbin/kamailio[4681]: ERROR:nathelper:rtpproxy_stream: no available proxies Thanks in advance for any help Alex

4 8

[Kamailio-Users] Dialplan Module Confusion
by JR Richardson 18 Nov '09

18 Nov '09

Hi All, I'm new to the list and fairly new to the vastness which is kamailio and the tons of modules and options. I'm trying to figure out the simplest method to accomplish the following scenario: SIP call comes in, match DID in database (MySQL), rewrite ruri with associated database entry matching DID, relay call to specified gateway, if no matching entry for DID is in database, forward call to dispatcher module and continue processing call (I already use the dispatcher module for load balancing, just need to implement DID call routing before I send calls to dispatcher) I've been reading about the following modules, dialplan, LCR, drouting, alias_db. I think dialplan is what I really need, but I'm confused about what entries to put in the database to match on 10 digit DID: match-op = 0 (equal) match_exp= 1112223333 match_len= 10 subst_exp= ?? is this where I would put the ruri sip:1112223333@10.10.10.10<sip%3A1112223333(a)10.10.10.10>(I'm not sure of the syntax here) repl_exp= ?? not sure what this is for can more than one database entry have the same id? and also I could use a practical example of how this would work int he config file: dp_translate("240", "$ruri.user/$ruri.user"); t_relay?? Any guidance on this would be much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

2 1

[Kamailio-Users] RADIUS for billing
by Rajesh 17 Nov '09

17 Nov '09

Dear All, I am trying to configure openSER for billing.After a lot og googling, i came to know that RADIUS is needed for anabling billing module. I would like to know if RADIUS is a MUST for billing or can i manage using mysql alone. I have configured accounting module in openser.can i use mysql and a CDR tool for billing?.. -- Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

3 4

[Kamailio-Users] dbtext, domain and caching
by Andreas Granig 17 Nov '09

17 Nov '09

Hi, I was trying to use caching in domain module with the dbtext backend, but on startup I get the error ERROR:domain:reload_domain_table: Database problem Looks like something is wrong with the column type when it's checked in line if ((ROW_N(row) == 1) && (VAL_TYPE(val) == DB_STRING)) { ... } The db file for domain module looks like: id(int) domain(str) last_modified(int) 1:test.local:0 Any hints on that? Probably a problem with dbtext module, since it works with mysql? Best regards, Andreas

2 5

[Kamailio-Users] Test a string concatenated with an AVP
by Antonio Reale 17 Nov '09

17 Nov '09

Hi all, actually I test if the R-URI contains the value *44* with: if (uri=~"^sip:\*44\*[0-9]+@") { ... } Now I want to read the value 44 or *44* from an AVP. Tried using directly the avp or assigning it to a variable, without success. ex.: $var(x) = "sip:" + $avp(s:code) + "[0-9]+@"; How can I solve this problem? Thank you very much. Regards. -- Antonio

2 6

[Serusers] (no subject)
by Chahn Kim 16 Nov '09

16 Nov '09

Hi all, Need some help from experts. I have beeb using $hdr(via[0]) to fetch the VIA header to get top via field, which will show entire top via filed as; <SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK-9aa0fded>. Now I am in need of getting IP:port# portion only to get assigend to a variable. What would be the most effecient way of doing it? I didn't find any other pseudo variable to provide it based on docu. Thanks in advance. John Chahn Kim email: johnk(a)koreanet.us

1 1

[Kamailio-Users] Question about force_send_socket() as it relates to record_route()
by Geoffrey Mina 16 Nov '09

16 Nov '09

Hello, I have a scenario where I need to send traffic to my carrier from two different source IP addresses. The carrier charges me one rate for dialer based traffic and another rate for standard outbound origination. I am planning on using the force_send_socket() method to manually set the source IP address. My question is how does this correlate to my usage of the record_route() function? Do I need to force_send_socket() BEFORE inserting the Record-Route header? Are there any other things I need to be concerned with while using the TM module? Thanks!

2 3

[Kamailio-Users] Freeradius Configuration
by toqeer ali 16 Nov '09

16 Nov '09

I am configuring Freeradius server with openser... By default there is openser dictionary file that the path of that file should be included freeradius master dictionary file... now i am confuse that command $INCLUDE {path of the openser dictioanry file} how i will include that over the network in dictionary file of Freeradius... if i copy the Freeradius server then it gives error... Basically my radius and openser are both on diffrent servers. -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

2 1

[Kamailio-Users] Possible setups question
by Varga Levente 16 Nov '09

16 Nov '09

Hello! I'm new to Kamailio so I want to ask a few questions before to be sure that I'm not trying to do something that cannot be done using Kamailio. The question would be if it is possible to do the following things using Kamailio: I want to setup the following configuration, two or more Asterisk servers with one redundant server in standby using Kamailio to route and partition the number. of users in a fixed manner. What I mean by that for instance if I have 200 users I want that the first 100 user to register and use server number one and the other 100 user to use the server number two. A user cannot migrate from one server to another they have to be always on the designated server from start. Kamailio would act as a unified front-end/gateway for the client (they will use the same entry point not being aware that there a number of Asterisk server beside it). Also there is a need for a N+1 redudancy scheme, that is if one of the servers fail or get very busy the new calls and registration should be redirected to the standby server on the fly (user account data replication would be taken care of on the spot). Can all this be done using Kamailio as a gateway and SIP router? Thank you! levi -- Lege, lege, relege, ora, labora et invenies.

3 2

[Kamailio-Users] please help me
by GEMINORIUM crux 15 Nov '09

15 Nov '09

hi i am transferring tls protocol to udp protocol. client <---tls----> kamilio <----udp---> asterisk how can i add in contact "transport=tls" when i am sending 200 OK message. please help me. any suggestion is appreciated.

2 1

[Kamailio-Users] CDRTool Mediaproxy etc.. AG stuff
by Brian Chamberlain 14 Nov '09

14 Nov '09

Hi List, I am running Debian lenny. Can I just change the repos in sources.list from lenny to testing and unstable to get the correct python dependencies etc for the ag-project tools? Thanks, Brian

1 0

[Serusers] CNAM in SER
by Brian Artigas 13 Nov '09

13 Nov '09

Hello, Our VoIP provider doesn't provide CNAM to us so we have to access an outside provider for this. Can anyone point me in the right direction for running an http request and passing the result back out so the pbx can pick grab the callerid. I know that OpenSER they has the Utils module but SER doesn't have that. Any ideas? Thanks in advance, Brian Artigas Allstate Computers, LLC brian(a)allstatecomputers.com www.allstatecomputers.com P: 561.743.1521 F: 561.575.9234

1 0

[Kamailio-Users] Remote management and security
by Brian 13 Nov '09

13 Nov '09

Hi, Does anyone know how I can do remote management of Kamailio in a secure way? Looking at the documentation for MI_DATAGRAM and MI_XMLRPC, there does not seem to be a way to authenticate the request, or a way to restrict requests to specific IP ranges. I want to be able to send management commands remotely, but I don't to make this wide open. Thanks, Brian.

3 3

Re: [Kamailio-Users] Register Request Forward
by toqeer ali 13 Nov '09

13 Nov '09

My rtpproxy works now basically i am using kamailio 1.2 with LCR, but when i dial one number via LCR it dial this number and it works but when i dial second number it says "*filtered destination*", i have to restart the service and then it dial second number... i think LCR not de-alocating the resources any idea ? 2009/11/12 Denis Putyato <denis7979(a)mail.ru> > 1) I would done not ” modparam("nathelper", "rtpproxy_sock", "udp: > 127.0.0.1:8899")” but “modparam("nathelper", "rtpproxy_sock", > "unix:/var/run/rtpproxy.sock")” > > 2) http://www.kamailio.org/docs/modules/1.5.x/nathelper.html#id2468157 > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Thursday, November 12, 2009 2:10 PM > > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Thanks Denis, > > > > It is indeed the problem of NAT. We have tried with live ip and its work > fine. However, behind the NAT it is not working. I have installed RTPproxy , > but still it is not working. Here are the configuration that we use in our > cfg file. > > > > modparam("nathelper","received_avp", "$avp(i:42)") > modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:8899") > modparam("nathelper", "natping_interval", 30) > modparam("nathelper", "ping_nated_only", 0) > modparam("nathelper", "sipping_bflag", 7) > modparam("nathelper", "sipping_from", "sip:pinger@127.0.0.1<sip%3Apinger(a)127.0.0.1> > ") > > route { > > ## NAT Detection > # > force_rport(); > if (nat_uac_test("19")) { > if (method=="REGISTER") { > fix_nated_register(); > } else { > fix_nated_contact(); > }; > setflag(5); > }; > > > ................. > > > > and rtpproxy is also configured on the same server. > > > > > > > 2009/11/12 Denis Putyato <denis7979(a)mail.ru> > > No two way communication is a problem of NAT in general. If session has > been established then it’s not codec problem. > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Thursday, November 12, 2009 9:25 AM > > > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > i have a problem with codec. when i call to canada from kamailio to PSTN, i > could not listen his/her voice as they listen( no two way communication) > > but when call to other countries it works... Basically call to canada > support g729 can i define any codec information in kamailio to G729. > > > > Basically the problem is only with local number when you dial with > extension 1 and when you dial with 08 it works... 08 is international > gateway of canada... > > > > X-lite i am using as a softphone. > > > > please help > > > > 2009/11/11 Denis Putyato <denis7979(a)mail.ru> > > “can i use regular expression in $fu=XXX” > > > > You can use regexp. $fu=~”expression” > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 3:08 PM > > > *To:* Denis Putyato > *Cc:* users@lists.kamailio..org <users(a)lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Hmmm > > > > it works now i just remove Record Route function and it works,,, now i can > dial from 602 to 601... thanks :). > > > > i will start work on how i will register multiple extension via kamailio to > asterisk ... can i use regular expression in $fu=XXX instead of actual > extension extension... > > 2009/11/11 Denis Putyato <denis7979(a)mail.ru> > > Registration is a process for telling SoftSwitch (in your variant - > Asterisk) where (on which username, IP address and port) the SoftSwitch must > send INVITE when received incoming call for 601. Try to understand where > your asterisk try to send INVITE when you try to call from 602 to 601. > > ngrep utility will help you (http://ngrep.sourceforge.net/download.html) > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 2:18 PM > > > *To:* Denis Putyato > *Cc:* users@lists.kamailio..org <users(a)lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Here is the update: > > > > what i have done so for is that I have "Register extension 601" to asterisk > via kamailio and it is registered!. > > > > When i call to 602 (configured on 192.168.0.1 -- asterisk box) it calls and > make connection and voice call also works (RTP Traffic is ok). > > > > However, when i call to 601 from 602 (which is directly connected to > asterisk without kamailio ), 601 can't receive call via kamailio. This is > the main problem. Please guide me that how i will define this in kamailio > (ip = 192.168.0.2) that xlite with extension 601 can receive call through > 602 which is another extension on Asterisk. > > > > In a nutshell: 601 (registered through Kamailio to Asterisk) ---can call > ---> 602 (registered via Asterisk) > > However: 602----cannot call ------> 601 > > > > please help. > > > > > > 2009/11/11 Denis Putyato <denis7979(a)mail.ru> > > Try to use dlg_bridge(); function in dialog module > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 12:13 PM > > > *To:* Denis Putyato > *Cc:* users@lists.kamailio..org <users(a)lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > > "that client send INVITE to asterisk via kamailio but no replay is in on > this INVITE" > > > > basically Asterisk receive the request and send back to kamailio but > kamailio can't send back to xlite.. > > > > 2009/11/11 Denis Putyato <denis7979(a)mail.ru> > > «but call does not route back to me» > > > > You mean, that incoming call from asterisk to client via kamailio doesn’t > work or that client send INVITE to asterisk via kamailio but no replay is in > on this INVITE? > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 10:16 AM > > > *To:* Denis Putyato > *Cc:* users@l > > ists.kamailio.org > > > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Thanks Again for the prompt reply... Basicaly i want to route Register > Requests for multiple asterisk via kamailio ... by using pattern matching > of extensions... > > > > is i am on the right path ?? or i can achieve this from dispatcher or load > balancer module .... Basically i want to register specific extensions > to specific asterisk boxes.... > > > > > > from your previous reply i can send call to asterisk and register to as > well but call does not route back to me... any clue ... > > 2009/11/11 Denis Putyato <denis7979(a)mail.r > > “The secret is that the other sip server was asterisk and i want route > call to asterisk via kamailio :).” > > I don’t understand why do you want to do any session with asterisk via > kamailio… not directly to asterisk… but any way the decision is same – > send() or rewritehost() > > > > route { > > if ($fU==”…”&&is_method(“INVITE”)) > > record_route(); > > rewritehost(“192.168.0..1”); > > t_relay(); > > } > > > > Do this before authentication checking is made in kamailio. > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 6:49 AM > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Thanks it Registered but i can't call on this .. "authentication Required > from proxy". what i should do now. > > > > The secret is that the other sip server was asterisk and i want route call > to asterisk via kamailio :). > > 2009/11/10 Denis Putyato <denis7979(a)mail.ru> > > Hi > > If not secret, why do you want to use such scheme? > > > > Your decision > > send(); - core function. > > > > { > > If (is_method(“REGISTER”)&&$fU==”….”) send(“udp:192.168.0.1:5060”); > > } > > > > or > > > > { > > If (is_method(“REGISTER”)&&$fU==”….”) { > > rewritehost(“192.168..0.1”); > > t_relay(); > > } > > } > > > > *From:* users-bounces(a)lists.kamailio.org [mailto: > users-bounces(a)lists.kamailio.org] *On Behalf Of *toqeer ali > *Sent:* Tuesday, November 10, 2009 1:56 PM > *To:* users(a)lists.kamailio.org > *Subject:* [Kamailio-Users] Register Request Forward > > > > Hi all, > > > > My question is how to forward Register Request to another sip server. > > > > For example one user (X-lite) want to register with 192.168.0.1(sip > server1) via 192.168.0.2(sip server 2). > > it means that in the domain part of xlite is 192.168...0.2 and > it basically register with 192.168.0.1. > > 192.168.0.2 will forward Register request to 192.168...0.1. > > it will be very helpful if any one can answer. > > > > Thanks > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

2 1

[Kamailio-Users] SKYPE or voipstunt
by Masoom Alam 13 Nov '09

13 Nov '09

whether skype or voipstunt are using SIP protocols. I have experienced that in many countries, where voice ports are blocked by the Telecom providers but still voipstunt and skype works for calling to PSTN. Please give me some links, for further reading on this.

2 1

[Kamailio-Users] Register Request Forward
by toqeer ali 12 Nov '09

12 Nov '09

Hi all, My question is how to forward Register Request to another sip server. For example one user (X-lite) want to register with 192.168.0.1(sip server1) via 192.168.0.2(sip server 2). it means that in the domain part of xlite is 192.168.0.2 and it basically register with 192.168.0.1. 192.168.0.2 will forward Register request to 192.168.0.1. it will be very helpful if any one can answer. Thanks -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

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[Kamailio-Users] record-route
by toqeer ali 12 Nov '09

12 Nov '09

Let me describe my scenario: SipUser1 -------register request------------> kamailio (SIP Proxy) ------- forwarded register request-----------> Asterisk 1. In the above scenario, when I register to Asterisk through Kamailio, it works fine when I give the "record-route()" function in the cfg file. Further, using this, I can also call to another extension 602 which is also registered to Asterisk but 602 cannot call to 601 which is registered via kamailio. 2. However, If I remove the "record-route()" function without unregistering 601 and restarting the server, 602 can call to 601 . route { if ($fU=="601"&&is_method("INVITE")) record_route(); rewritehost("10.0.10.111"); //asterisk host route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } should i add anything in script further... -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

2 1

[Kamailio-Users] Listening on more than one port
by Sven Schulz 12 Nov '09

12 Nov '09

Is it possible to have kamailio (1.4.4) to listen on more than one port? Is it as simple as adding another ³port=xxxx² in the config? Sven Schulz Penn State University Telecommunications and Network Services 814.865.6116 sip:sven@psu.edu

2 1

[Kamailio-Users] sending calls to voip provider
by Andres Moya 11 Nov '09

11 Nov '09

Hi, I am with openser about my first 40 hours. So far i managed to install 3.0. Was able to make calls between registered phones. Enabled accounting, etc. Looks i have control ;) Now i have a very simple question. I want to send calls for land lines via voip provider, sipdiscount.com in my case. I want to enable so registered client agent can send invite to that server, but it should re authenticate somehow. I was looking for proxy_authorize www_authorize but it looks like it authenticate my phones only. I was trying rewrite rewritehostport, but of course was getting authentications errors. Is there any manual, previous discussion on this topic? Thank you.

1 0

[Kamailio-Users] change contact address
by GEMINORIUM crux 11 Nov '09

11 Nov '09

hi 1. i want to change contact address .i want to add transport protocol how can i do this ? 2. i am getting Error on $var(query) = "INSERT INTO sip_user(sipuser) VALUES(' "+$fU+"' )"; if(!avp_db_query("$var(query)","$avp(s:test)")){ xlog("ERROR IN Query = $var(query)"); sl_send_reply("400","Bad Request"); exit; } but if i run that query in mysql it give me expected output. why i am getting error ? pls help me ? any suggestion is appreciated. thanks

1 0

[Kamailio-Users] Problem in configuring LCR module
by Masoom Alam 11 Nov '09

11 Nov '09

Hi there, I am a first time user of Kamilio 1.5. We are trying to configure the LCR, we get an error in the syslog. The error in the syslog is: Nov 6 15:12:09 kam /sbin/kamailio[5396]: ERROR:lcr:mod_init: AVP fr_inv_timer_avp has not been defined Nov 6 15:12:09 kam /sbin/kamailio[5396]: ERROR:core:init_mod: failed to initialize module lcr Nov 6 15:12:09 kam /sbin/kamailio[5396]: ERROR:core:main: error while initializing modules We have already configured FR_inv_timer_avp parameter like this in the Kamilio.cfg: modparam("tm", "fr_inv_timer_avp", "$avp(i:70") The "tm" module is already loaded in the Kamilio.cfg. Please guide, where we are making problems?

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[Kamailio-Users] Translate "486 Busy Here" to "600 Busy Everwhere"
by geoffreymina＠gmail.com 11 Nov '09

11 Nov '09

Hello, I have a need to translate a 486 from my asterisk server to a 600 when sending upstream to my carrier. The problem I am seeing is that when I return 486, the carrier re-presents the call to my next server in the SRV rotation... I really want the 486 to be a final disposition, so I am assuming sending a 600 would solve my issue. I am not sure how to go about forwarding the message, but modifying the SIP code... is that even allowed? This would be something like what I'm trying to do: failure_route[1]{ if(t_check_status("

3 3

Re: [Kamailio-Users] Translate "486 Busy Here" to "600 Busy Everwhere"
by Geoffrey Mina 10 Nov '09

10 Nov '09

Apologies... sent directly to Alex and not the list. On Tue, Nov 10, 2009 at 12:02 PM, Geoffrey Mina <geoffreymina(a)gmail.com> wrote: > Alex, > Thanks for the info. We are on the same page about what a proxy > should, and should not do... however, I am strugging with my ITSP > (Level3) a bit on this. Unfortunately, Asterisk is my media end-point > here and there is no mechanism in Asterisk to send back a 600. Level3 > is exhibiting very odd behaivor, and I have a ticket open with them, > but who knows if it will ever go anywhere. > > CallID: 1627795841_10332(a)4.55.4.227 > Level3 --> INVITE --> Kamailio --> Asterisk > Asterisk --> 183 Progress --> Kamailio --> Level3 > Asterisk --> 486 Busy --> Kamailio --> Level3 > Level3 --> ACK --> Kamailio --> Asterisk > > [at this point Level3 IMMEDIATELY sends me another presentation of the > call with a different CallID and the process repeats] > > CallID: 604627041_120490736(a)4.55.20.99 > Level3 --> INVITE --> Kamailio --> Asterisk > Asterisk --> 183 Progress --> Kamailio --> Level3 > Asterisk --> 486 Busy --> Kamailio --> Level3 > Level3 --> ACK --> Kamailio --> Asterisk > > > I am pretty sure I am in the right here, and Level3 shouldn't be > failing over in a 486 scenario... unfortunately my customer doesn't > care why our systems aren't talking properly :) > > Thanks! > > > > On Tue, Nov 10, 2009 at 10:38 AM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> geoffreymina(a)gmail.com wrote: >>> >>> Hello, >>> I have a need to translate a 486 from my asterisk server to a 600 when >>> sending upstream to my carrier. The problem I am seeing is that when I >>> return 486, the carrier re-presents the call to my next server in the SRV >>> rotation... I really want the 486 to be a final disposition, so I am >>> assuming sending a 600 would solve my issue. >>> >>> I am not sure how to go about forwarding the message, but modifying the >>> SIP code... is that even allowed? >>> >>> This would be something like what I'm trying to do: >>> >>> failure_route[1]{ >>> if(t_check_status(" >> >> Sure, you can do that. It is not a good idea to do so in principle because >> it violates how proxies should behave; proxies are very thin interoperation >> layers that are designed to statefully pass requests and replies back and >> forth more or less strictly as intended by the endpoints involved, and a >> B2BUA would more ideally fit your goal. Nevertheless, it is possible. >> >> failure_route[1] { >> >> if(t_was_cancelled()) >> exit; >> >> if(t_check_status("486")) { >> t_reply("603", "Declined"); >> exit; >> } >> } >> >> >> -- >> Alex Balashov - Principal >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> >

2 2

[Kamailio-Users] Double acc records in DB
by Alex Hermann 10 Nov '09

10 Nov '09

Hello, I noticed that sometimes I get 2 identical acc records in the database, although I get only 1 in the syslog (I use both db_flag and log_flag). Of course, the second record in the DB shouldn't be there. There are no retransmissions of neither request nor response and only a single branch exists. How is this possible, and more importantly, how can I prevent the double records? This is on Kamailio 1.4+svn. -- Greetings, Alex Hermann

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[Kamailio-Users] Question about translating 486 to 600
by Geoffrey Mina 10 Nov '09

10 Nov '09

Hello, I have a need to translate a 486 from my asterisk server into a 600 to my ITSP. I haven't figured out a way to force asterisk to send the 600, so I guess I need to rely on Kamailio to do that for me. The problem is that when the INVITE is sent to me, if I return a 486, my ITSP immediately sends a new INVITE to the next gateway in my SRV records. I want the 486 to be a final disposition in this scenario, so I am guessing I need to send 600 for the carrier to not try another gateway. How would I go about doing this? failure_route[1]{ if(t_check_status("486")){ t_relay("600","Busy Here"); } } I know the above example isn't valid... but that is basically what I need to accomplish. Thanks!

1 0

[Kamailio-Users] avp
by Anders 10 Nov '09

10 Nov '09

Hi all, Here I am - back with more "Kamailio 101"-questions! ;-) I need to strip the first character from the uri and write this to the 'acc' table. What I'm doing so far in kamailio.cfg is: modparam("acc", "db_extra", "............., $avp(s:user_ip_tag)"); So I've added the avp-thing in that line. And independently I've created a new column in the 'acc' table called "user_ip_tag". If the above is correct, my next step is to write something to this variable. In the routing, I have this: if (uri=~"^sip:[1-9]00*") { do_something_here_to_take_first_character_of_the_uri_and_put_to_avp...but_how? strip(1); } What I want to do, is to take the 1-9 digit, if it starts with that and has to zeroes after, and put that to the avp - and the strip it off before I move on in the routing. Yes, I have browsed the documentation, but I'm not sharp enough to understand it, it seems. (or alternatively, the documentation is not made for beginners). Thanks!

2 2

[Kamailio-Users] accounting
by Rajesh 09 Nov '09

09 Nov '09

Dear all, I would like to know if i can get the duration between two calls in Accounting module..I am in an attempt to do "Billing" module for OpenSER and hence would like to know the same. -- Regards, Rajesh

4 8

SR or Kamailio.
by Vicente 09 Nov '09

09 Nov '09

Hi, This is my first message to the list. So, sorry if I say something stupid. I've installed sip-router from GIT repository. It is now compiled and installed. I can see that bins ser, sercmd and ser_mysql.sh has been copied to /usr/local/sbin. ser runs properly but sercmd tries to use /tmp/ser_ctl (ERROR: connect_unix_sock: connect(/tmp/ser_ctl): No such file or directory [2]) while /temp/sip-router_ctl is the socket generated by ser. ser_mysql.sh also fails with error: ./ser_mysql.sh: 326: Bad substitution The INSTALL file on the repository instructs to use serctl utility to watch server health but no binary has been generated with that name. Is it replaced by sercmd? What is the status of sip-router.org? Is it ready for production? Or is it a better option to use Kamailio at the moment? Best regards.

3 3

[Kamailio-Users] How to recognize the caller ID
by Masoom Alam 09 Nov '09

09 Nov '09

Hi Every one, Is the following scenario possible and if yes, how? I want to identify the caller ID inside the Kamailio so that a specific caller can be routed to an Asterisk box. Actually, the caller -- a SIP client or an IP phone client is not registered at Kamailio, but Kamailio has to act as a Proxy to a group of Asterisk servers. The sketch in my mind is to use some if-then-else statements within the [route] where user can call a particular number say NXXXXXXX201 and then the Kamailio can strip off the first number and re-route it to an Asterisk server. Regards, MM Alam SERG Pakistan http://serg.imsciences.edu.pk http://www.imsciences.edu.pk/profiles/masoom_alam.html

3 2

[Serusers] 500 internal error
by Sabatella, Michael 08 Nov '09

08 Nov '09

Hi, I am getting a 500 internal server error. I am not sure why as my config did not change. I do not want to restart the ser proxy. Could I get a list of causes for a 500 server error? Thanks, Michael Sabatella Network Services Engineer Office: 1-201-253-2280 Cell: 1-908-380-5321 ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

2 3

[Kamailio-Users] A architectural question
by Masoom Alam 08 Nov '09

08 Nov '09

Hi there, Is it possible that Kamailio is used in front of a group of Asterisk servers. That is, Kamailio as a REGISTRAR Translation can proxy the SIP registration and identify which PBX to send on LAN by the ID before the EXT underlined from IP PHONE. Please advise with some links through which it is possible to handle such an architecture. Regards, MM Alam SERG Pakistan http://serg.imsciences.edu.pk http://www.imsciences.edu.pk/profiles/masoom_alam.html

2 2

[Kamailio-Users] OCS?
by Jay R. Worthington 06 Nov '09

06 Nov '09

Hi, has someone managed to "federate" kamailio with an Office Communications Server? I'm mainly looking for IM&Presence (to gate into ejabberd), voice would be nice to have... Regards, Jay

2 1

[Kamailio-Users] routing question
by Andrew Gallo 06 Nov '09

06 Nov '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Greetings: Hope you don't mind a question from someone just getting his feet wet :) First off, if I want Kamailio to act as a SIP router among, say, a group of Asterisk PBXs (meaning, Kamailio will have no user registrations, but just pass SIP messages to and from the Asterisks), would I do this in the least cost routing table or in carrier routes table, or via another method? (assume for the moment that all the PBXs are under my control, these aren't connections outside my organization, no connections to the PSTN). I figured it would be with lcr, so I tried compiling 3.0.0RC1 with that module removed from the exclude list. I received the following error on 'make install' make[1]: Entering directory `/usr/local/src/kamailio-3.0.0/kamailio/modules/lcr' make[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/kamailio/modules/lcr' config.mak included make[1]: Entering directory `/usr/local/src/kamailio-3.0.0/kamailio/modules/lcr' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m32 - -minline-all-stringops -falign-loops -ftree-vectorize - -fno-strict-overflow -mtune=athlon64 -Wall -DNAME='"kamailio"' - -DVERSION='"3.0.0-rc1"' -DARCH='"i386"' -DOS='linux_' - -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux - -DSER_VER=3000000 -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC - -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP - -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER - -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS - -DSTATISTICS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 - -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN - -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL - -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL - -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT - -I/usr/local/include -I/usr/local/include -I/opt/include - -I/usr/sfw/include -DOPENSER_MOD_INTERFACE -DMOD_NAME='"lcr"' -c hash.c - -o hash.o In file included from hash.c:26: lcr_mod.h:34:18: error: pcre.h: No such file or directory In file included from hash.c:26: lcr_mod.h:51: error: expected specifier-qualifier-list before âpcreâ hash.c:34: error: expected declaration specifiers or â...â before âpcreâ hash.c: In function âlcr_hash_table_insertâ: hash.c:56: error: âstruct lcr_infoâ has no member named âfrom_uri_reâ hash.c:56: error: âfrom_uri_reâ undeclared (first use in this function) hash.c:56: error: (Each undeclared identifier is reported only once hash.c:56: error: for each function it appears in.) hash.c:58: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:59: error: âstruct lcr_infoâ has no member named âfirst_gwâ hash.c:60: error: âstruct lcr_infoâ has no member named âpriorityâ hash.c:66: error: âstruct lcr_infoâ has no member named ânextâ hash.c: In function âlcr_hash_table_contents_freeâ: hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:107: error: âstruct lcr_infoâ has no member named âfrom_uri_reâ hash.c:108: error: âstruct lcr_infoâ has no member named âfrom_uri_reâ hash.c:110: error: âstruct lcr_infoâ has no member named ânextâ make[1]: *** [hash.o] Error 1 make[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/kamailio/modules/lcr' make: *** [install-modules] Error 1 If I leave lcr in the exclude list, I can make/make install, but if I try to add an entry into the lcr table, I get this error: # kamctl lcr addgw labCM 172.16.59.6 5060 1 2 1 ERROR 1054 (42S22) at line 1: Unknown column 'ping' in 'field list' ERROR: lcr - SQL Error System info: Kamailio 3.0.0 RC1 from GIT debian 5.0.3 (2.6.26-2-686) gcc -v Using built-in specs. Target: i486-linux-gnu Configured with: ../src/configure -v --with-pkgversion='Debian 4.3.2-1.1' --with-bugurl=file:///usr/share/doc/gcc-4.3/README.Bugs - --enable-languages=c,c++,fortran,objc,obj-c++ --prefix=/usr - --enable-shared --with-system-zlib --libexecdir=/usr/lib - --without-included-gettext --enable-threads=posix --enable-nls - --with-gxx-include-dir=/usr/include/c++/4.3 --program-suffix=-4.3 - --enable-clocale=gnu --enable-libstdcxx-debug --enable-objc-gc - --enable-mpfr --enable-targets=all --enable-cld - --enable-checking=release --build=i486-linux-gnu --host=i486-linux-gnu - --target=i486-linux-gnu Thread model: posix gcc version 4.3.2 (Debian 4.3.2-1.1) Thank you! -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAkrx/CMACgkQQr/gMVyFYySWQgCeJoW2VLWRI85bHdC9ThwnEwLc pWQAoJZVtBEzdRI8+3xBoLnlWJxNXbj/ =Wwjk -----END PGP SIGNATURE-----

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[Kamailio-Users] Dispatcher module question
by DJB 05 Nov '09

05 Nov '09

Is there a way that I can pass variable into ds_select_dst(set, alg) on the set parameter so I can query from database table for setid; for instance, ds_select_dst("avp(customer_setid)", "4"). I've tried, but it did not work. Please advise. Thank you.

3 5

difference between memlog and memdbg
by Klaus Darilion 05 Nov '09

05 Nov '09

Hi! What's the difference betwenn memlog and memdbg? thanks klaus

2 2

[Kamailio-Users] uac_replace_from check
by jh＠tutpro.com 05 Nov '09

05 Nov '09

if i have understood correctly, if uac_replace_from() is called in "auto" mode, encoded from uri is added into an extra param of r-r header when the request is forwarded towards uas. does the module check that reply to the request from uas really includes the added parameter in its r-r header or is correct operation of in-dialog requests at the mercy of the uas? -- juha

7 37

Re: [Kamailio-Users] LCR problem with "id" column and match priorities
by jh＠tutpro.com 05 Nov '09

05 Nov '09

it turned out that the problems ricardo reported on gw selection in lcr module were due to the fact that he has the same gw (same in terms of ip address) in more than one gw group and in each group the gw has different trip/tag attributes. load_gws() function tries the same gateway only once, but it is not currently well defined from which group gw's attributes are taken when request is sent to the gateway. currently the attributes are taken from the gw group, which happens to be the first in in-memory gw table. one could think that this is a feature, i.e., if the same gw is in many gw groups, it must have the same attributes in every group. the other alternative is that this is a bug, i.e., same gw's attributes can differ from group to group and the ones that are used must be the ones of the chosen group. i have a fix written for sr_3.0 lcr module, but i'm hesitating to introduce the fix also to version 1.5, because the fix has not been tested well yet. any opinions on what should be done? -- juha

3 9

Re: [Kamailio-Users] syntax error
by Daniel-Constantin Mierla 05 Nov '09

05 Nov '09

don't forget -- always cc the mailing list. On 05.11.2009 5:39 Uhr, GEMINORIUM crux wrote: > Daniel, > i am transferring tls protocol to udp protocol, > sorry, i don't mention it. > > client <---tls----> kamilio <----udp---> asterisk > > thanks for your reply. > > > On Wed, Nov 4, 2009 at 7:38 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > > On 04.11.2009 14:34 Uhr, GEMINORIUM crux wrote: > > hi, > i am new to kamailio, > > getting syntax error: > $var(test) = "udp:"+$rd+":"+$rp; > t_relay_to_udp($var(test)); > > t_relay_to_udp($rd,$rp); > > working with string. > > is there any reason ? how to resolve it ? > i have to use database so i need to use variable. > > any suggestion is appreciated. > > you have to do: > > $du = "sip:"+$rd+":"+$rp; > t_relay(); > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > * http://www.asipto.com/ > > -- Daniel-Constantin Mierla * http://www.asipto.com/

2 3

[Kamailio-Users] vim syntax file for 3.0
by Daniel-Constantin Mierla 04 Nov '09

04 Nov '09

Hello, I updated the vim syntax file for Kamailio 3.0 and SR3.0/devel. I attached as well a script to auto-detect the type of .cfg file based on matching compatibility string (i.e., #!XYZ), loadmodule or main route line. To install, copy the files in your home dir - see the hints inside each file to know which one to place where: .vim/syntax/ser.vim .vim/ftdetect/ser.vim If you don't install the file type detect script, then you can use ":setf ser" once you opened the config file. Enjoy, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ " Copy this file to $HOME/.vim/ftdetect/ser.vim func! s:cfgType() let max = line("$") > 200 ? 200 : line("$") for n in range(1, max) if getline(n) =~ '^\s*#!$KAMAILIO\|OPENSER\|SER\|ALL\|MAXCOMPAT$' set filetype=ser return elseif getline(n) =~ '^\s*modparam\s*(\s*"[^"]\+"' set filetype=ser return elseif getline(n) =~ '^\s*route\s*{\s*' set filetype=ser return endif endfor setf cfg endfunc au BufNewFile,BufRead *.cfg call s:cfgType() " -*- vim -*- " FILE: ser.vim " LAST MODIFICATION: 2009-05-28 18:30 " (C) Copyright 2008 Stanisław Pitucha <viraptor(a)gmail.com> " (C) Copyright 2009 Daniel-Constantin Mierla <miconda(a)gmail.com> " Version: 1.01 " USAGE: " " Save this file to $VIMFILES/syntax/ser.vim. Either add a detection " script to filetypes.vim, or set filetype manually to "ser" when " editing SIP Router configuration file with 'setf ser'. " " List of keyword and core functions taken from latest dev version of " SIP Router. Module functions not included. " " Tested only on vim 7.1 " " Example: "setf ser" " " REQUIREMENTS: " vim (>= 7) if exists("b:current_syntax") finish endif syn match serConfigParamLine '^[^=]\+=.*$' contains=serCoreParameter,serString,serConfigConstant,serSpecial,serNumber,serCppComment,serHashComment syn region serConfigModparam start='^\s*modparam\s*(' end=')' contains=serString,serNumber syn match serConfigModule '^\s*loadmodule\s*"[^"]\+"' contains=serString syn keyword serTodo TODO FIXME XXX contained syn match serOperator '!\|&&\|||\|=[~=]\?\|>\|<\|+\|-\|/\|\*\||\|&\|^\|\~\|defined\|eq\|ieq\|ne\|ine\|mod' display contained syn region serCppComment start='/\*' end='\*/' contains=serTodo syn match serHashDefine '#!define\s\|#!ifdef\s\|#!ifndef\s\|#!endif\|#!else\|#!KAMAILIO\|#!OPENSER\|#!SER\|#!MAXCOMPAT\|#!ALL' " syn match serHashDefine '^\s*#!.+$' syn match serHashComment '#[^!].*$\|#$' contains=serTodo syn match serStringEscape '\\.' contained syn match serNumber '[0-9]\+' contained syn region serString matchgroup=Normal start='"' skip='\\"' end='"' contained contains=serVariable,serStringEscape syn match serVariable "$[a-zA-Z_][a-zA-Z0-9_]*$([^)]\+)$\?" contained syn match serIdentifier '[a-zA-Z_][a-zA-Z0-9_]*' contained syn keyword serStatement route if else switch case default break exit return drop while include_file contained syn keyword serSpecial yes no on off true false enabled disabled contained syn keyword serCoreKeyword af dst_ip dst_port from_uri method msg:len proto status snd_af snd_ip snd_port snd_proto src_ip src_port to_af to_ip to_port to_proto to_uri uri uri:host uri:port contained syn keyword serCoreValue udp UDP tcp TCP tls TLS sctp SCTP inet INET inet6 INET6 sslv23 SSLv23 SSLV23 sslv2 SSLv2 SSLV2 sslv3 SSLv3 SSLV3 tlsv1 TLSv1 TLSV1 max_len myself contained syn keyword serCoreFunction forward forward_tcp forward_udp forward_tls forward_sctp send send_tcp log error exec force_rport add_rport force_tcp_alias add_tcp_alias udp_mtu udp_mtu_try_proto setflag resetflag isflagset flags bool setavpflag resetavpflag isavpflagset avpflags rewritehost sethost seth rewritehostport sethostport sethp rewritehostporttrans sethostporttrans sethpt rewriteuser setuser setu rewriteuserpass setuserpass setup rewriteport setport setp rewriteuri seturi revert_uri prefix strip strip_tail userphone append_branch set_advertised_address set_advertised_port force_send_socket contained syn keyword serCoreParameter debug fork log_stderror log_facility listen alias auto_aliases dns rev_dns dns_try_ipv6 dns_try_naptr dns_srv_lb dns_srv_loadbalancing dns_udp_pref dns_udp_preference dns_tcp_pref dns_tcp_preference dns_tls_pref dns_tls_preference dns_sctp_pref dns_sctp_preference dns_retr_time dns_retr_no dns_servers_no dns_use_search_list dns_search_full_match dns_cache_init use_dns_cache use_dns_failover dns_cache_flags dns_cache_negative_ttl dns_cache_min_ttl dns_cache_max_ttl dns_cache_mem dns_cache_gc_interval dns_cache_del_nonexp dns_cache_delete_nonexpired dst_blacklist_init use_dst_blacklist dst_blacklist_mem dst_blacklist_expire dst_blacklist_ttl dst_blacklist_gc_interval port statistics maxbuffer children check_via phone2tel syn_branch memlog mem_log memdbg mem_dbg sip_warning server_signature reply_to_via user uid group gid chroot workdir wdir mhomed disable_tcp tcp_children tcp_accept_aliases tcp_send_timeout tcp_connect_timeout tcp_connection_lifetime tcp_poll_method tcp_max_connections tcp_no_connect tcp_source_ipv4 tcp_source_ipv6 tcp_fd_cache tcp_buf_write tcp_async tcp_conn_wq_max tcp_wq_max tcp_rd_buf_size tcp_wq_blk_size tcp_defer_accept tcp_delayed_ack tcp_syncnt tcp_linger2 tcp_keepalive tcp_keepidle tcp_keepintvl tcp_keepcnt tcp_crlf_ping disable_tls tls_disable enable_tls tls_enable tlslog tls_log tls_port_no tls_method tls_verify tls_require_certificate tls_certificate tls_private_key tls_ca_list tls_handshake_timeout tls_send_timeout disable_sctp enable_sctp sctp_children sctp_socket_rcvbuf sctp_socket_receive_buffer sctp_socket_sndbuf sctp_socket_send_buffer sctp_autoclose sctp_send_ttl sctp_send_retries advertised_address advertised_port disable_core_dump open_files_limit shm_force_alloc mlock_pages real_time rt_prio rt_policy rt_timer1_prio rt_fast_timer_prio rt_ftimer_prio rt_timer1_policy rt_ftimer_policy rt_timer2_prio rt_stimer_prio rt_timer2_policy rt_stimer_policy mcast_loopback mcast_ttl tos pmtu_discovery exit_timeout ser_kill_timeout max_while_loops stun_refresh_interval stun_allow_stun stun_allow_fp server_id description descr desc loadpath mpath contained syn region serBlock start='{' end='}' contained contains=serBlock,@serCodeElements syn match serRouteBlock '$failure_\|onreply_\|branch_\|event_\|onsend_$\?route$\s*\[[^\]]\+\]$\?' contained contains=serNumber,serString,serIdentifier syn region serRrouteBlockFold matchgroup=serRouteBlock start="$failure_\|onreply_\|branch_\|event_\|onsend_$\?route$\s*\[[^\]]\+\]$\?\s*\n\?{" matchgroup=NONE end="}" contains=serBlock,@serCodeElements syn cluster serCodeElements contains=serHashDefine,serCppComment,serHashComment,serNumber,serString,serVariable,serOperator,serStatement,serKeyword,serCoreKeyword,serCoreValue,serCoreFunction,serIdentifier hi def link serCppComment Comment hi def link serHashComment Comment hi def link serHashDefine Special hi def link serTodo Todo hi def link serConfigModparam Function hi def link serConfigModule Keyword hi def link serKeyword Keyword hi def link serCoreKeyword Special hi def link serCoreValue Special hi def link serCoreFunction Function hi def link serRouteBlock Type hi def link serRrouteBlockFold Type hi def link serIdentifier Identifier hi def link serSpecial Special hi def link serCoreParameter Keyword hi def link serOperator Operator hi def link serStatement Conditional hi def link serNumber Number hi def link serVariable Identifier hi def link serString String hi def link serStringEscape Special let b:current_syntax = "ser"

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[Kamailio-Users] Social Networking Event
by Daniel-Constantin Mierla 04 Nov '09

04 Nov '09

Hello, several folks working with Kamailio, SIP Router, SER, OpenIMSCore, SEMS and Asterisk are in Berlin next week, so we think of having a dinner (or beer) meeting Thursday, 19:00, Nov 12, 2009. If happens that you are around and want to join, please send me an email to make sure you get a seat. As usual for this kind of event, free participation with everyone paying for himself/herself. Just to have an idea about how it could be, see some photos from previous similar events: http://www.asipto.com/gallery/v/social_meeting_barcelona2008/ http://www.asipto.com/gallery/v/FOSDEM2009/ From private discussions so far, we should be already about 10 people, confirmed including myself, Olle E. Johansson (Asterisk) and Elena-Ramona Modroiu. There is a lot to celebrate, from integration work within SIP Router Project to upcoming Kamailio 3.0 (now RC1). Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] [Fwd: Re: uac_replace_from again]
by marius zbihlei 04 Nov '09

04 Nov '09

Forgot to forward to list

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[Kamailio-Users] syntax error
by GEMINORIUM crux 04 Nov '09

04 Nov '09

hi, i am new to kamailio, getting syntax error: $var(test) = "udp:"+$rd+":"+$rp; t_relay_to_udp($var(test)); t_relay_to_udp($rd,$rp); working with string. is there any reason ? how to resolve it ? i have to use database so i need to use variable. any suggestion is appreciated. thanks

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[Kamailio-Users] Kamailio 1.5.0 and NAT
by Путято Денис 04 Nov '09

04 Nov '09

Hello everybody I have such problem with kamailio and client which behind NAT This is REGISTER message which came from client. This message came from public IP (for example, 1.1.1.1) and source port 44197. (This information I got from sip_trace) REGISTER sip:sip.ptl.ru:5060;transport=UDP SIP/2.0 From: <sip:xxxxxxx@sip.ptl.ru:5060>;tag=a82eda3-13c4-47798948-1a1296-5259a46 To: <sip:xxxxxxx@sip.ptl.ru:5060> Call-ID: a82eda3-13c4-47798948-1a1296-9641c88(a)sip.ptl.ru CSeq: 446229198 REGISTER Via: SIP/2.0/UDP 10.130.237.163:5060;branch=z9hG4bK-47799e27-6b9841-84448db Max-Forwards: 70 Supported: timer Contact: <sip:xxxxxxx@10.130.237.163:5060;transport=UDP> Expires: 60 Content-Length: 0 This is Unauthorized message was sent by kamailio to 1.1.1.1 and: destination port not 44197 (as I expected) but 5060!!! SIP/2.0 401 Unauthorized From: <sip:xxxxxxx@sip.ptl.ru:5060>;tag=a82eda3-13c4-47798948-1a1296-5259a46 To: <sip:xxxxxxx@sip.ptl.ru:5060>;tag=fdc454a3cdd8f9e5718382b74f33aeed.3b81 Call-ID: a82eda3-13c4-47798948-1a1296-9641c88(a)sip.ptl.ru CSeq: 446229198 REGISTER Via: SIP/2.0/UDP 10.130.237.163:5060;branch=z9hG4bK-47799e27-6b9841-84448db;received=1.1.1.1 WWW-Authenticate: Digest realm="domain name", nonce="4af0129b000000047c1020258f9f0efc4de1b76663dd2ab4" Server: Kamailio (1.5.0-notls (i386/linux)) Content-Length: 0 And as you understand 401 reply, which was sent to 1.1.1.1:5060 can't reach client's equipment. A piece of kamailio.cfg relating to REGISTER processing: route(0) { :. if (!is_method("INVITE")) route(8); :. if (is_method("REGISTER")) { route(11); exit; } :.. } route(8) { if (nat_uac_test("5")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } route(11) { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if (isflagset(5)) setbflag(6); if (!save("location")) { sl_reply_error(); exit; } } Hearty thanks for any help.

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[Kamailio-Users] 404 sparks next gw and 503
by Anders 03 Nov '09

03 Nov '09

Hi, This list becomes a mix of high-level tech and low-level beginner questions... - and I'm in the latter group. Making two lists might be over-kill, though. Here's my issue: My call termination provider is rightfully returning a 404 for a call to a number that doesn't exist. This currently seems to spark Kamailio to look for the next gw, but in this case there is no next gw set up in the lcr/gw - so it returns a 503 to the caller. If I'm confident that my provider is right that there is no such number, and 404 is the correct answer, wouldn't it be better to return a 404 to the caller rather than a 503? With 503 stats also get screwed up, because my client can just see a bunch of 503s, which is not a very detailed picture of the real situation. If I should return a 404, how would I go about this? What would I do in cfg to ensure that a 404 return does not spark it to look for a next gw, but sends 404 back to the caller? Alternatively, I could try to figure out how to get the return from my provider saved to the acc table on the INVITE record, and I'd have something more detailed to run stats on. Does that make sense? Thanks!

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[Kamailio-Users] OpenSER/Kamailio behind NAT
by systeo 02 Nov '09

02 Nov '09

Hi, I successfully installed Kamailio on the Amazon computing cloud (EWS), and it runs fine. However, I can't get to it (register) from any UACs from the public internet. In amazon, all the instances (i.e. "virtual machines) are in a "private" IP, kindda like the private IP you get in a LAN. However, Amazoin gives you a "public IP" that can be mapped to any machine instance in your account, i.e. the machine that kamailio is running. However, when i use this public IP in the configuration of the UACs, it cannot get to it. How can I get kamailio to listen on this public IP? Secondly, and related to the first, when I run kamailio in my LAN (behind a router), say on 192.168.1.113, and all my UACs are in the same LAN, say 192.168.1.112, 192.168.1.114, etc, I can register ok and make calls just fine. However, when I try to come in from the public internet, no UACs can register to my Kamailio server (running on 192.168.1.113), even though I am forwarding port 5060 from my router to 192.168.1.113 (where kamailio is running). I think this problem is related to the first one, in that I cannot get Openser (kamailio) to listen on a public IP, when it is behing a "LAN". Can anyone help? Lastly, the same issue with rtpproxy also behind the LAN, but it needs a pulic IP, right? Please help...anyonw. Hassen BEN SLIMA Analyst developer TEL : 00216 74 863 507 FAX : 00216 74 863 506 E-MAIL : <mailto:tarak@systeo.biz> hassen(a)systeo.biz Web : <http://www.systeo.biz> www.systeo.biz Adresse : Route de Tunis km 10, Technopôle de Sfax - 3021 - Sakiet EZZIT - Sfax - Tunisie <http://www.systeo.biz/> Sans titre-4

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[Kamailio-Users] force_rtp_proxy() usage with IP that is not natively homed
by Alex Balashov 02 Nov '09

02 Nov '09

Greetings, Please excuse the apparent silliness of this request, but I am curious if there is a combination of flags passed to force_rtp_proxy() and/or rtpproxy invocation options that can allow force_rtp_proxy() to substitute an SDP endpoint IP that is not a physical device/interface on the host on which rtpproxy is running? Let me explain further: 1. I have a host with a single network interface that has private IP 192.168.1.254. 2. I am receiving an inbound call from outside the network, coming in on a public IP address 4.4.4.4 that is a 1-to-1 DNAT/SNAT to 192.168.1.254. 3. The call is forwarded to a PBX on the same LAN as the native interface, e.g. 192.168.1.250. 4. I would like to set up SDP in such a way that 192.168.1.250 sendsmedia to the 192.168.1.254 interface while the outside host initiating the call sends its media to 4.4.4.4. Under normal circumstances in which both the public and private interface were physically present on the machine, this would call for the use of "bridging" mode (rtpproxy invoked with -l 4.4.4.4/192.168.1.254 and use of "ie" and "ei" flags with force_rtp_proxy() - we have previously established that rtpproxy_offer/answer do not actually seem to work as advertised). But is it possible to "bridge" to an IP that is not routed to the box? I expected that force_rtp_proxy()'s second parameter - ip_address - would do this. However, this does not appear to let me override the SDP endpoint with any IP address I want; it seems that I am still limited to choosing from those IPs physically available on the rtpproxy host. I would guess that this is because ip_address is passed as a parameter through rtpproxy's control API socket and is validated by rtpproxy itself, and if it is not a local existing address, is overridden with one that is. Is there a clever combination of flags that can be used to accomplish this, or is it a lost cause? Thanks! -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

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[Serusers] error: SER/FIFO not accessible: 2
by Brian Artigas 02 Nov '09

02 Nov '09

Hello, I'm trying to add a user to the database and getting the following error back: error: SER/FIFO not accessible: 2 Here is an excerpt from my ser.cfg file: debug=3 fork=no log_stderror=yes listen=192.168.1.30 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo_mode=0777 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) Anyone have any ideas? Thanks, Brian

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[Kamailio-Users] Broken Debian packaging in Kamailio 3.0.0-RC
by Kim Philby 31 Oct '09

31 Oct '09

Hi! I've tried to make Debian packages for latest Kamailio, but some issues arised. First, packages are named ser-stuff, shouldn't they be called kamailio-stuff even though Kamailio now uses SIP-Router core? Also, Debian package making seems to be broken: # install the *.sql files for ser_mysql.sh script cp -f scripts/mysql/*.sql /home/kmphlb/src/kamailio-3.0.0/kamailio/debian/ser/usr/lib/ser/ cp: target «/home/kmphlb/src/kamailio-3.0.0/kamailio/debian/ser/usr/lib/ser/» is not a directory make[1]: *** [install] Error 1 Directories are named 'ser'. Shouldn't it be just 'kamailio'? Regards, -- Kim Philby

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[Kamailio-Users] Kamailio 3.0.0 RC1
by Daniel-Constantin Mierla 30 Oct '09

30 Oct '09

Hello, I created a tarball with sources of Kamailio 3.0.0 RC1 in order to easy the usage and testing: http://www.kamailio.org/pub/kamailio/3.0.0-rc1/src/kamailio-3.0.0-rc1_src.t… There is also now a tutorial to guide the installation of kamailio 3.0 branch from GIT repository: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git Two wiki pages were created to collect what is new in Kamailio 3.0 and how to migrate to this version: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 Both documents need a lot of care since many new features are missing or are not detailed, and definitely update guidelines do not cover all changes. Help is very much appreciated there. As we keep hunting and fixing issues, improve documentation, I hope to get to full release Kamailio 3.0.0 very soon. Enjoy RC1 meanwhile and report any issue you find to sr-dev(a)lists.sip-router.org. Thanks, Daniel

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[Kamailio-Users] Route header problem
by Alex Rendour 30 Oct '09

30 Oct '09

Hello, We have a problem with route on header. I receive an ack from X.X.X.X (asterisk) on Y.Y.Y.Y (kamailio 1.5.3). Y.Y.Y.Y relay ACK to Z.Z.Z.Z Y.Y.Y.Y this route from X.X.X.X: <sip:Y.Y.Y.Y;lr=on;ftag=as7e58db45>,<sip:11dc0d7918b56374@Z.Z.Z.Z;lr> And forward as the same to Z.Z.Z.Z. The route header must be : <sip:11dc0d7918b56374@Z.Z.Z.Z;lr> I think my kamailio must delete the first part of the route header, like this explaination: http://www.tech-invite.com/Ti-sip-dialog.html#inv-ack Could you help me? Regards, -- alex

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[Kamailio-Users] kam 1.5 syslog on solaris 10
by Robert McGilvray 29 Oct '09

29 Oct '09

I'm having some issues getting kam to log correctly to syslog after it daemonizes. I get all the initialization entries in the proper logfile but nothing after that - I have xlog statements throughout the config which only show up in syslog if fork=no. I've tested my syslog config with the system logger which works correctly executed as the kamailio user. If I run syslog in debug I don't see it receiving anything from kam after initialization. Any ideas? log_stderror=no log_facility=LOG_LOCAL1 debug=7 fork=yes children=4 /etc/syslog.conf (with tabs) local1.debug /var/log/globeop/kamailio.log This is the correct format of syslog in solaris. It only uses wildcard (*) in facility not levels. Specifying debug will include everything at higher levels. From the man page: For a given facility and level, syslogd matches all messages for that level and all higher levels. For example, an entry that specifies a level of crit also logs messages at the alert and emerg levels. This config is working correctly on sol 10 with openser 1.3. ykt1sipprd2# tail -f /var/log/globeop/kamailio.log Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 951507 local1.debug] DBG:core:grep_aliases: comparing host [2:loghost:5060] with us [1:loghost:5060] Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 951507 local1.debug] DBG:core:grep_aliases: comparing host [2:loghost:5060] with us [1:ykt1sipprd2:5060] Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 793175 local1.debug] DBG:core:grep_aliases: no match for: [2:loghost:5060] Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 634971 local1.debug] DBG:core:socket2str: <tcp:172.20.151.148:5060> Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 797031 local1.debug] DBG:core:fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048 Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 596959 local1.debug] DBG:core:fm_malloc_init: F_HASH_SIZE=2067, fm_block size=16560 Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 362601 local1.debug] DBG:core:fm_malloc_init: params (fc400000, 33554432), start=fc400000 Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 551434 local1.debug] DBG:core:shm_mem_init_mallocs: success Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 123545 local1.info] INFO:core:init_tcp: using /dev/poll as the TCP io watch method (auto detected) Oct 27 09:50:46 ykt1sipprd2 kamailio[3815]: [ID 721503 local1.warning] WARNING:core:daemonize: pid file contains old pid, replacing pid -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

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[Kamailio-Users] How to take MAC address
by Chandrakant Solanki 29 Oct '09

29 Oct '09

Hi I want to know the MAC address of registered client on kamailio... suppose i have 3 Client registered with 192.168.1.2, 192.168.1.3 and 192.168.1.4 from these 3 Client, i am registered successfully with 101, 102, 103(a)test.com But for additional authentication ... i also need MAC address for this Client. Is there any module, method, function or variable to take MAC address or is there any alternative for the above case. Waiting for favorable reply... -- Regards, Chandrakant Solanki

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[Kamailio-Users] Pass calls from/to another realm to gateway
by Alexander 29 Oct '09

29 Oct '09

I've encountered one problem I can not solve :( The situation is following: we've got Kamailio working together with our own SIP platform. Our platform is about various call processing business logic and billing. All calls from SIP users should pass through our platform. For now we successfully can make and receive calls inside our domain - NAT is handled fine in most cases, instant messages are handled and so on. It looks like that: - Kamailio receives a call from caller - Call is redirected to our platform - Platform redirects call back to Kamailio and it looks for callee. I've tried to implement that logic. It seems to work in some cases, but I encounter one problem. Sometimes Kamailio can not forward a call to the platform via rewritehostport(). It tries to forward SIP request, but nothing happens - only retransmission handler is called. I mean, after call to t_relay(), I see this: Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb618de58, INVITE sip:200213@62.117.120.101:5061 SIP/2.0... Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb618de58, INVITE sip:200213@62.117.120.101:5061 SIP/2.0... Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb618de58, INVITE sip:200213@62.117.120.101:5061 SIP/2.0... SIP requests seem to be correct, and configuration file works ok from time to time. The problem I've described happens when Kamailio receives a call from another domain (for example, from sipbroker). Inside our domain everything works fine and INVITE passes to our platform correctly. I've attached configuration file and debug output. Files were written for and by Opensips, but with Kamailio I encounter the same problem. Is this a problem with configuration file or something else?

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[Kamailio-Users] roadmap to kamailio 3.0.0
by Daniel-Constantin Mierla 29 Oct '09

29 Oct '09

Hello, I am thinking of planning to release Kamailio 3.0.0 next Wednesday or day after (October 28 or 29). With some delays, testing is going fine in my side, still some bits to be done, here is what I collected: - taking in consideration $du updates in branch route - import fixes from sr_3.0 branch - complete core statistics and handling of drop reply I believe is feasible, in the worse case we can package it at respective date and allow a bit more time for testing, but we unfreeze code and can go for new development. I encourage people to start writing migration documents. Let's see other opinions as well. Cheers, Daniel

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Re: [Kamailio-Users] How to take MAC address
by Zahid Mehmood 28 Oct '09

28 Oct '09

I know that the Polycom phones can be configured to add the MAC address in one of the headers (probably the user agent hdr). -- Zahid ----- Original Message ----- From: users-bounces(a)lists.kamailio.org <users-bounces(a)lists.kamailio.org> To: users(a)lists.kamailio.org <users(a)lists.kamailio.org> Sent: Wed Oct 28 02:58:05 2009 Subject: Re: [Kamailio-Users] How to take MAC address On Wednesday 28 October 2009 06:35:28 Chandrakant Solanki wrote: > Hi > > I want to know the MAC address of registered client on kamailio... > > suppose i have 3 Client registered with 192.168.1.2, 192.168.1.3 and > 192.168.1.4 > > from these 3 Client, i am registered successfully with 101, 102, > 103(a)test.com > > But for additional authentication ... i also need MAC address for this > Client. > > Is there any module, method, function or variable to take MAC address or is > there any alternative for the above case. > > > Waiting for favorable reply... You will not have luck outsite "your proxy network" ... I mean ... that if your proxy is on 192.168.0.10/24 and your clients on any other network, you will always get's the same MAC, the mac of the 192.168.0.10/24 gateway. Your only hope is that SOME SIP devices add their MAC as part of some subscribe or register petition. I don't see any security increase on this, because MAC's could be easily spoofed. -- Raúl Alexis Betancor Santana Dimensión Virtual _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

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Re: [Kamailio-Users] Asterisk to Kamailio Call forwarding error
by Daniel-Constantin Mierla 28 Oct '09

28 Oct '09

On 28.10.2009 11:23 Uhr, Chandrakant Solanki wrote: > > > On Wed, Oct 28, 2009 at 3:50 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > > > On 28.10.2009 11:14 Uhr, Chandrakant Solanki wrote: > > > > On Wed, Oct 28, 2009 at 3:18 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com> > <mailto:miconda@gmail.com <mailto:miconda@gmail.com>>> wrote: > > > > On 28.10.2009 10:37 Uhr, Chandrakant Solanki wrote: > > Hi > > Call Sequence is ... > > Client A/B register on Kamailio and both are registered > successfully.. > Firewall is stopped. > > Now... Client A call to Client B.. > > Client A => Kamailio => Asterisk => Kamailio => Client B > > Above is call sequence, when call goes from Asterisk to > Kamailio ... then found below error.... > > ERROR:tm:t_forward_nonack: no branch for forwarding > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: > ERROR:tm:w_t_relay: t_forward_nonack failed > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ***** > OUT route[8] > > what processing are you doing in kamailio for this case? Do > you do > same failure routing to another destination if B does not > answer? > Such error occurs when r-uri is changed in failure route but no > append_branch() was executed. > > Cheers, > Daniel > > > > hi > > where could i put this thing.... > > Here is my config file for kamailio... > > http://pastebin.com/m34e07510 > > Please help me out ... > > > your config is big to troubleshoot it quick and easy. Definitely > you have call forwarding, what version of kamailio are you using? > the header of file says openser 1.2. > > I suggest you use ngrep/wireshark to watch the sip traffic and put > xlog() messages in the config to correlate and get hints about > what happesn. > > Cheers, > Daniel > > PS. Keep cc-ing to mailing list, there are people that can help > better or faster inc ase they use the sip wizard config. > > -- > Daniel-Constantin Mierla > * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin > * http://www.asipto.com/index.php/sip-router-masterclass/ > > > hi > > i m using kamailio-1.5.2-tls... > if you don't keep mailing list in the discussion thread, your messages will be ignored... Daniel

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Re: [Kamailio-Users] Asterisk to Kamailio Call forwarding error
by Daniel-Constantin Mierla 28 Oct '09

28 Oct '09

On 28.10.2009 11:14 Uhr, Chandrakant Solanki wrote: > > > On Wed, Oct 28, 2009 at 3:18 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > > > On 28.10.2009 10:37 Uhr, Chandrakant Solanki wrote: > > Hi > > Call Sequence is ... > > Client A/B register on Kamailio and both are registered > successfully.. > Firewall is stopped. > > Now... Client A call to Client B.. > > Client A => Kamailio => Asterisk => Kamailio => Client B > > Above is call sequence, when call goes from Asterisk to > Kamailio ... then found below error.... > > ERROR:tm:t_forward_nonack: no branch for forwarding > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: > ERROR:tm:w_t_relay: t_forward_nonack failed > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ***** OUT route[8] > > what processing are you doing in kamailio for this case? Do you do > same failure routing to another destination if B does not answer? > Such error occurs when r-uri is changed in failure route but no > append_branch() was executed. > > Cheers, > Daniel > > > > hi > > where could i put this thing.... > > Here is my config file for kamailio... > > http://pastebin.com/m34e07510 > > Please help me out ... > your config is big to troubleshoot it quick and easy. Definitely you have call forwarding, what version of kamailio are you using? the header of file says openser 1.2. I suggest you use ngrep/wireshark to watch the sip traffic and put xlog() messages in the config to correlate and get hints about what happesn. Cheers, Daniel PS. Keep cc-ing to mailing list, there are people that can help better or faster inc ase they use the sip wizard config. -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] kamctl online
by Chandrakant Solanki 28 Oct '09

28 Oct '09

hello I would like to print user@domainname while i execute kamctl online command... Any Idea -- Regards, Chandrakant Solanki

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[Kamailio-Users] Asterisk to Kamailio Call forwarding error
by Chandrakant Solanki 28 Oct '09

28 Oct '09

Hi Call Sequence is ... Client A/B register on Kamailio and both are registered successfully.. Firewall is stopped. Now... Client A call to Client B.. Client A => Kamailio => Asterisk => Kamailio => Client B Above is call sequence, when call goes from Asterisk to Kamailio ... then found below error.... ERROR:tm:t_forward_nonack: no branch for forwarding Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ERROR:tm:w_t_relay: t_forward_nonack failed Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ***** OUT route[8] -- Regards, Chandrakant Solanki

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Re: [Kamailio-Users] hung dialogs + no BYE
by Daniel-Constantin Mierla 27 Oct '09

27 Oct '09

On 27.10.2009 21:22 Uhr, Anders wrote: > That was exactly the problem Daniel - no BYE was ever sent from the > UA, so that's what we need to fix! > please keep cc-ing the mailing list. You can fix the problem by identifying the sip devices that do not send BYE. You are doing record routing, right? In 1.5, dialog can send BYE at call timeout -- does not help much you, but can close eventual open channels in gateways. The missing BYE happens when call is between two SIP phones? Or between sip phone and pstn gateway/media server? Cheers, Daniel > Thanks!! > > On Tue, Oct 27, 2009 at 4:16 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: > >> On 26.10.2009 16:41 Uhr, Anders wrote: >> >>> Hi, >>> >>> I have two issues, and I think they are connected. The number of >>> Active Dialogs keeps growing - as if some of them are hung. Not all of >>> them, but some of them. At the same time, I have seen that from a >>> specific customer, there is no BYE in the 'acc' table in the >>> accounting. So, my conclusion - no BYE means it's not finished means >>> it's hung... - right? >>> >>> Any ideas where to look? >>> >>> >>> >> if you don't get a BYE in acc table then maybe was not sent and that keeps >> the dialog active. You can set a max time per call -- timeout -- for each >> dialog: >> >> http://kamailio.org/docs/modules/1.5.x/dialog.html >> >> However, is good to identify why BYE is not coming, maybe is a fraud attempt >> or a broken sip device. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin >> * http://www.asipto.com/index.php/sip-router-masterclass/ >> >> >> -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

2 2

[Kamailio-Users] newbie question
by Luis Jimenez 27 Oct '09

27 Oct '09

Hello, list. Exactly what happens with a dialog (at accounting level) when i reload kamailio with dialogs in process? Thanks in advance. Luis Jimenez

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[Kamailio-Users] hung dialogs + no BYE
by Anders 27 Oct '09

27 Oct '09

Hi, I have two issues, and I think they are connected. The number of Active Dialogs keeps growing - as if some of them are hung. Not all of them, but some of them. At the same time, I have seen that from a specific customer, there is no BYE in the 'acc' table in the accounting. So, my conclusion - no BYE means it's not finished means it's hung... - right? Any ideas where to look? Thanks!

2 1

[Kamailio-Users] LCR problem with "id" column and match priorities
by jh＠tutpro.com 24 Oct '09

24 Oct '09

ricardo, in situation where you see higher priority valued gateway group selected first, do you know if the lower priority valued group would be used if none of the gateways in the higher priority valued gateway group can serve the request? if it is used, then both exist in the hash table. the list in the hash table is sorted by prefix length and priority. -- juha

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[Kamailio-Users] acc: accounting flag for CANCEL
by Alex Balashov 24 Oct '09

24 Oct '09

If I have acc set to log CANCEL'd calls, do I need to set the accounting flag on handling CANCEL request, since it is a new transaction relative to the initial INVITE, or will it get logged automatically? Thanks, -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

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[Kamailio-Users] basic SIP forwarding with Asterisk
by Jeff Brower 23 Oct '09

23 Oct '09

All- Can we use Asterisk combined with Kamailio as follows: __________ ___________ | | | | SIP ___| |___ SIP ___| Kamailio |___ SIP | | | rtpproxy | | Asterisk | | | | | | | | | RTP ___| |___ RTP ___| DSP card |___ RTP (G711) |__________| (G711) |___________| (G729, G723, GSM-AMR, EVRC) We've already implemented an rtpproxy interface to the DSP card, which has its own GbE port. Our question is whether we can we perform some type of basic SIP forwarding or "SIP pass-thru", but still invoke rtpproxy for call setup and tear-down and/or when media attributes for the call change? We're getting a lot of requests from customers who -- for whatever reasons -- need to use or continue to use Asterisk, but need to also add transcoding, ec, encryption, and other compute-intensive requirements that Asterisk doesn't support (or at least doesn't support at higher capacity or without going unstable). Thanks. -Jeff

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[Kamailio-Users] Anybody using domainpolicy module?
by Klaus Darilion 23 Oct '09

23 Oct '09

Hi! The domainpolicy draft is obsolete and the module needs adaptions to work with sip-router. Thus, we plan to remove it, as I guess nobody uses it at all. So, if you use this module, please speek up NOW! regards klaus

1 0

[Kamailio-Users] Search for text with a space in it
by CSB 22 Oct '09

22 Oct '09

OpenSER 1.3 When I try to match a phrase with a space in it I don't get a match. I'm looking for "User-Agent: My UA" None of the following seem to work: if (search("User-Agent:\sMy\sUA")) { if (search("User-Agent:[ ]My[ ]UA")) { if (search("User-Agent: My UA")) { Could anyone point out my obvious mistake? Thanks

2 1

[Kamailio-Users] Kamailio ignores Via in replies?
by Alex Hermann 22 Oct '09

22 Oct '09

Hello, while debugging a problem today I encountered a rather strange issue. The client's UA/router was messing up the Via header, with as result that all the IP's in the Via's were replaced by the public IP of the DSL modem. Now the weird part: because of the mangled Via's, the expected result is that the reply would be routed back to the IP of the client. Instead, the replies were still following the "correct" return path through a series of OpenSER/Kamailio proxies. How is this possible? According to rfc3261, a proxy must forward the reply to the next Via in the received _response_. It seems Kamailio is incorrectly keeping the Via from the request and sends the reply to the stored address of the Via from the _request_. Btw, no 'received' parameter is involved. Can anyone with inside knowledge of Kamailio share his/her view on this issue? And possibly, how to fix it. I suspect I can use check_via=1 to drop incoming replies with incorrect via, although the behaviour of check_via is undocumented (what does it do when check fails?). And why is the default value 0 (as has every example config on the web)? There's also reply_to_via, but according to docs that's only for locally generated replies. Also, its use is beyond me, as the correct value is dependent on the src_ip of the request and thus can't possibly be a global option. Alex.

2 1

[Kamailio-Users] SIP message
by Rajesh 22 Oct '09

22 Oct '09

Dear All, I am using a separate Location server.I would like to know which SIP message is used to call the location server.To be more clear, as INVITE messages are used to make calls, I would like to know the similar SIP message which is used to store the user details in the location table..Is that REGISTER message ?...Correct me if iam wrong.. -- Regards, Rajesh

2 1

[Kamailio-Users] Kamailio v1.5.3 released
by Daniel-Constantin Mierla 21 Oct '09

21 Oct '09

A new release in 1.5 series is out. Kamailio 1.5.3 is based on the latest version of branch 1.5, including many fixes in code and documentation, therefore those running 1.5.0, 1.5.1 or 1.5.2 are advised to upgrade. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.5.3/src/ Detailed changelog: http://www.kamailio.org/pub/kamailio/1.5.3/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.5.3/ Project site at SourceForge.net (still using old name): http://sourceforge.net/projects/openser/ Binaries and packages will be uploaded at: http://www.kamailio.org/pub/kamailio/1.5.3/ Modules' documentation: http://www.kamailio.org/docs/modules/1.5.x/ What is new in 1.5.x release series is summarized in the announcement of v1.5.0: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v1.5.0-Release-Notes Enjoy and stay tunned -- new major release Kamailio 3.0.0 is going to be out very soon, Daniel

1 0

[Kamailio-Users] LCR problem with "id" column and match priorities.
by Ricardo Martinez 21 Oct '09

21 Oct '09

Hello List. I'm having a weird behavior with the LCR module. I have a particular prefix routed as follows (according to the LCR tables) : mysql> select * from lcr where prefix=0251; +-----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +-----+--------+----------+--------+----------+ | 89 | 1351 | | 51 | 1 | | 127 | 1351 | | 43 | 2 | +-----+--------+----------+--------+----------+ The gw's group are as follows : mysql> select * from gw where grp_id=51; +-----+----------------+--------+---------------+----------+------+----- -------+-----------+-------+------+--------+------+-------+ | id | gw_name | grp_id | ip_addr | hostname | port | uri_scheme | transport | strip | tag | weight | ping | flags | +-----+----------------+--------+---------------+----------+------+----- -------+-----------+-------+------+--------+------+-------+ | 100 | GW1.TERMN1 | 51 | XX.XXX.XXX.16 | | 5060 | 1 | 1 | 4 | 1351 | 25 | 1 | 0 | | 101 | GW2.TERMN1 | 51 | XX.XXX.XXX.13 | | 5060 | 1 | 1 | 4 | 1351 | 25 | 1 | 0 | | 102 | GW3.TERMN1 | 51 | XX.XXX.XXX.32 | | 5060 | 1 | 1 | 4 | 1351 | 25 | 1 | 0 | | 103 | GW4.TERMN1 | 51 | XX.XXX.XXX.35 | | 5060 | 1 | 1 | 4 | 1351 | 25 | 1 | 0 | +-----+----------------+--------+---------------+----------+------+----- -------+-----------+-------+------+--------+------+-------+ 4 rows in set (0.00 sec) mysql> mysql> select * from gw where grp_id=43; +----+------------------+--------+---------------+----------+------+---- --------+-----------+-------+------+--------+------+-------+ | id | gw_name | grp_id | ip_addr | hostname | port | uri_scheme | transport | strip | tag | weight | ping | flags | +----+------------------+--------+---------------+----------+------+---- --------+-----------+-------+------+--------+------+-------+ | 76 | GW1.TERMN222 | 43 | XX.XXX.XXX.16 | | 5060 | 1 | 1 | 4 | 1343 | 25 | 1 | 0 | | 77 | GW2.TERMN222 | 43 | XX.XXX.XXX.13 | | 5060 | 1 | 1 | 4 | 1343 | 25 | 1 | 0 | | 78 | GW3.TERMN222 | 43 | XX.XXX.XXX.32 | | 5060 | 1 | 1 | 4 | 1343 | 25 | 1 | 0 | | 79 | GW4.TERMN222 | 43 | XX.XXX.XXX.35 | | 5060 | 1 | 1 | 4 | 1343 | 25 | 1 | 0 | +----+------------------+--------+---------------+----------+------+---- --------+-----------+-------+------+--------+------+-------+ 4 rows in set (0.00 sec) So, when a call with prefix 1351 arrived at the proxy, the LCR module is changing the RURI for 1343 (matching the 2nd priority for that prefix). This keep happening even if change the priority value 2 for priority value 5. I even changed the priority 1 for priority 0 as follows : +-----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +-----+--------+----------+--------+----------+ | 89 | 1351 | | 51 | 0 | | 127 | 1351 | | 43 | 1 | +-----+--------+----------+--------+----------+ But the LCR module keep using the "second" priority. After doing some tests I've found two ways to use the "first" priority for the prefix. 1.- Removing the second priority 2.- Re-ordering the "id" column and using correlative values as indicated in the next table : mysql> select * from lcr where prefix=0251 order by id; +-----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +-----+--------+----------+--------+----------+ | 127 | 1351 | | 51 | 1 | | 128 | 1351 | | 43 | 2 | +-----+--------+----------+--------+----------+ So, when the prefix are with "no correlatives" values in the "id" column seems to be a problem to match the first priority. But when the values are correlative there is no problem. Is this the correct behavior? Maybe I'm missing something? What happens for example if I need to add a third priority to a certain prefix after doing several changes in the LCR table? I need to re-order every time the "id" values to make them correlatives? I really hope that someone could give me a hand here. Ricardo Martinez.-

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[Serusers] question regarding sip protocol
by Sabatella, Michael 21 Oct '09

21 Oct '09

Hi, I am new to the sip protocol. I have been doing packet traces to understand how the protocol works. I am using the ser sip proxy version 0.9.6. I notice that once my phone registers with a proxy I get a 200 ok. Every 20 seconds after that the client sends a packet to the proxy. It is an Request: OPTIONS packet. My proxy responds with a Status:404 Not found. I tried to do some research on this but could not find anything useful. Does this response of a 404 matter? How can you fix this? I have read that the client does this as an application level ping to make sure the proxy is up. If I have 100s of people registering that is a lot of extra traffic. Is there a way to stop this options query or have the proxy respond properly? Thanks Michael Network Services Engineer ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

2 1

[Kamailio-Users] preparing Kamailio 1.5.3
by Daniel-Constantin Mierla 21 Oct '09

21 Oct '09

Hello, Kamailio 1.5.3 will be released today. Please commit anything you have for branch 1.5 on Sourceforge.net svn in the next two hours. Afterwards we will start preparing the release. Cheers, Daniel

1 0

[Kamailio-Users] max channel capacity
by Jeff Brower 21 Oct '09

21 Oct '09

All- On a reasonably fast server, say quad-core, what is the approx maximum number of G711 IP calls when using Kamailio and rtpproxy? What if rtpproxy runs on a second server? No echo can, no packet concealment, etc... just G711. -Jeff

3 5

Re: [Kamailio-Users] use case for settlement free peering
by Daniel-Constantin Mierla 20 Oct '09

20 Oct '09

welcome -- cc-ing mailing list is recommended all the time -- help people to get the conclusion of discussions. On 20.10.2009 20:04 Uhr, A G wrote: > Great. Thanks for the feedback! > > On Mon, Oct 19, 2009 at 3:47 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: > >> Hello, >> >> On 19.10.2009 20:18 Uhr, A G wrote: >> >>> Greetings: >>> >>> I'm looking for advice on a project/proof of concept I'm working on. >>> I would like to create a settlement-free peering fabric for voice >>> traffic between and among some peer institutions in my area. Because >>> this is more of a side-project for cost cutting measure, I'm primarily >>> looking at open source software, though commercial product >>> recommendations would be helpful as well. >>> >>> The organizations I would like to connect have their own PBXs with >>> large blocks of numbers (whole NPA-NXXs), with no number portability >>> in or out. I imagine both at the individual PBXs and peering fabric, >>> the number routing would be static. To put another way, we would >>> manually configure which connections the block of telephone numbers is >>> reachable at. >>> >>> Here is the required ASCII art diagram :) >>> >>> >>> +-------+ >>> | PBX | >>> +-------+ >>> | >>> +-------+ >>> | SBC | >>> +-------+ >>> | >>> | >>> +---+ +---+ .--------. +---+ +---+ >>> | P | | S | / \ | S | | P | >>> | B |--| B |------ ( ???? )----------| B |--| B | >>> | X | | C | \ / | C | | X | >>> +---+ +---+ `---------' +---+ +---+ >>> | >>> | >>> +-------+ >>> | SBC | >>> +-------+ >>> | >>> +-------+ >>> | PBX | >>> +-------+ >>> >>> >>> >>> >>> For scalability reasons, a full mesh of connections between and among >>> the SBCs is not an attractive option. >>> >>> Here's what I think I need: >>> Basic SIP routing >>> TCP, TLS, and UDP support >>> >>> What would be nice to have: >>> IPv6 >>> CDR >>> >>> What is probably not needed: >>> User agent client registration, presence, IM, voice mail >>> >>> I see there are several different open source voice projects. >>> Do you think this is an appropriate use for Kamailio? >>> >>> >> Kamailio can be used in such scenario. It has a lot of features to help you >> routing calls -- see modules such as lcr, carrierroute, dialplan. >> >> If you do heavy tls/tcp, then upcoming Kamailio 3.0 will have lot of >> improvements in this areas. IPv6 is supported as well for core routing. You >> can generate CDRs using acc module and some stored procedures -- see for >> example: >> http://siremis.asipto.com/install-accounting/ >> >> Cheers, >> Daniel >> >> >>> I'm seeking comments on what you would use for this situation. >>> Are there any existing projects along these lines? >>> Is there one project that is better than another for this application? >>> >>> >>> >> -- >> Daniel-Constantin Mierla >> * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin >> * http://www.asipto.com/index.php/sip-router-masterclass/ >> >> >> -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

2 1

Re: [Kamailio-Users] Problem with LCR "ping"
by Alexandr Dubovikov 20 Oct '09

20 Oct '09

Hi Ricardo, you use ping_interval 10 seconds. For 126 gateways it's extrem short . Can you set this param up to 60 seconds and check again ? Wbr, Alexandr P.S. will be nice to see the result from "ngrep -W byline -t -l port 5060", if it possible :-) > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of Ricardo Martinez > Sent: Thursday, October 15, 2009 4:48 PM > To: users(a)lists.kamailio.org > Subject: [Kamailio-Users] Problem with LCR "ping" > > Hello list. > > I have a question/problem regarding the LCR module. I'm using Kamailio version 1.5.2. The LCR tables have : > > > > gw table : 126 records > > lrc tables : 120 records. > > > > I'm using the "ping" (ping = 1) feature in all my gateways, part of the cfg file is as follows : > > > > modparam("lcr","db_url","mysql://openserro:openserro@localhost/openser") > > modparam("tm", "fr_inv_timer_avp", "$avp(i:704)") > > modparam("lcr", "gw_uri_avp", "$avp(i:709)") > > modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)") > > modparam("lcr", "ruri_user_avp", "$avp(i:500)") > > modparam("lcr", "flags_avp", "$avp(i:712)") > > modparam("lcr", "fetch_rows", 3000) > > modparam("lcr", "ping_interval", 10) > > modparam("lcr", "ping_from", "sip:pinger@mydomain.net") > > modparam("lcr", "ping_method", "OPTIONS") > > > > From time to time I can see 3 o 4 gateways marked as down (with the command kamctl lcr dump). I'm not manually monitoring the gateways every minute so is possible that in a particular moment the gateways could be unreachable, but when I run the command "kamctl lcr dump" the gateways are on-line, that's for sure. > > So I'm not understanding why LCR keep marking the gateways as down, even if they are answering to the OPTIONS request. Could this be a bug in the LCR module? Or maybe I'm missing something?. The silly part of all this is the gateways marked as down are always the same. Could this be related to a memory issue?? > > > > Can someone guide me here?. > > > > Thanks in advance, > > Regards, > > Ricardo Martinez.- > > > > -- Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info AD1-UANIC * ICQ: 122351182 * http://www.start4.info

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[Serusers] SER Staring Problem
by sourav.das1＠wipro.com 20 Oct '09

20 Oct '09

Hi, I am trying to install the SER through the following steps. Download and extract the package : * Download the source package containing mysql support for SER from: http://ftp.iptel.org/pub/ser/latest/ <http://ftp.iptel.org/pub/ser/latest/> * Save the downloaded package at /usr/src folder * Extract the package * Change directory to the extracted package Make and install Make the program and the modules and then install the program in the folder /usr/local. # make && make modules # make prefix=/usr/local install After installing the SER on the Linux system I am trying to start SER through the following command. /usr/local/sbin/serctl start While executing the above command I am getting the following error. Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed Please let me know how to resolve this or something is missing. Regards, Sourav Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[Kamailio-Users] kamailio 1.5.3 scheduled for next wednesday
by Henning Westerholt 20 Oct '09

20 Oct '09

Hi all, as it has been three month since the last minor release of the kamailio 1.5 branch i'd suggest that we do a 1.5.3 release soon. I want to propose next wednesday the 14.10.2009 as date for the release. So if you have any pending patches in the trunk or in your internal repositories that you want to like to see in this release, please commit them until 12:00 UTC this day. Thank you, Henning -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

4 8

[Kamailio-Users] use case for settlement free peering
by A G 19 Oct '09

19 Oct '09

Greetings: I'm looking for advice on a project/proof of concept I'm working on. I would like to create a settlement-free peering fabric for voice traffic between and among some peer institutions in my area. Because this is more of a side-project for cost cutting measure, I'm primarily looking at open source software, though commercial product recommendations would be helpful as well. The organizations I would like to connect have their own PBXs with large blocks of numbers (whole NPA-NXXs), with no number portability in or out. I imagine both at the individual PBXs and peering fabric, the number routing would be static. To put another way, we would manually configure which connections the block of telephone numbers is reachable at. Here is the required ASCII art diagram :) +-------+ | PBX | +-------+ | +-------+ | SBC | +-------+ | | +---+ +---+ .--------. +---+ +---+ | P | | S | / \ | S | | P | | B |--| B |------ ( ???? )----------| B |--| B | | X | | C | \ / | C | | X | +---+ +---+ `---------' +---+ +---+ | | +-------+ | SBC | +-------+ | +-------+ | PBX | +-------+ For scalability reasons, a full mesh of connections between and among the SBCs is not an attractive option. Here's what I think I need: Basic SIP routing TCP, TLS, and UDP support What would be nice to have: IPv6 CDR What is probably not needed: User agent client registration, presence, IM, voice mail I see there are several different open source voice projects. Do you think this is an appropriate use for Kamailio? I'm seeking comments on what you would use for this situation. Are there any existing projects along these lines? Is there one project that is better than another for this application? Thank you

2 1

Presence gateway between SIP and XMPP
by afshin afzali 18 Oct '09

18 Oct '09

Hi Guys, You stated : "*This module is a gateway for presence between SIP and XMPP. * * It translates one format into another and uses xmpp, pua and presence modules to manage the transmition of presence state information*" My first question is that does it enable exchange of presence data between SIP and XMPP in two ways ? If yes how I could use the pua_xmpp_notify and pua_xmpp_req_winfo APIs to do so ? Regards, -- Afshin

1 0

[Kamailio-Users] Odd behavior svn rev 2:5901M
by brandon＠cryy.com 16 Oct '09

16 Oct '09

Hello, I am noticing very rarely however it is happening where kamailio will route to the completely wrong sip uri. In this scenario I see it routing to one of our voicemail applications, thus placing voicemails into the wrong mailbox identifier. What is odd in this particular scenario I was able to capture is that it attempts to deliver for the same call-id to the right location and wrong location within a very very very small time frame causing our voicemail application to pick the wrong branch. What is even odder is this is a recovery from a 487 request cancelled so it is coming from failure route, appending a branch, so there should only be a single branch to voicemail. I clearly see in the logs after setting the ruri that it is correct, I have no idea where it is getting this ghost mailbox uri identifier, it is almost as some how the call got mixed up with a completely different call. Just throwing this out here to see if anyone else has experienced this. I will attach supporting evidence to this email for this scenario shortly. The last thing to mention for now is that it is definitely random and not sure-proof reproducible. Thank you for your time and input ahead of time. Look forward to hearing back from you all! Have a great weekend. Sent from my Verizon Wireless BlackBerry

1 0

[Kamailio-Users] Problem with LCR "ping"
by Ricardo Martinez 16 Oct '09

16 Oct '09

Hello list. I have a question/problem regarding the LCR module. I'm using Kamailio version 1.5.2. The LCR tables have : gw table : 126 records lrc tables : 120 records. I'm using the "ping" (ping = 1) feature in all my gateways, part of the cfg file is as follows : modparam("lcr","db_url","mysql://openserro:openserro@localhost/openser") modparam("tm", "fr_inv_timer_avp", "$avp(i:704)") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "fetch_rows", 3000) modparam("lcr", "ping_interval", 10) modparam("lcr", "ping_from", "sip:pinger@mydomain.net") modparam("lcr", "ping_method", "OPTIONS") >From time to time I can see 3 o 4 gateways marked as down (with the command kamctl lcr dump). I'm not manually monitoring the gateways every minute so is possible that in a particular moment the gateways could be unreachable, but when I run the command "kamctl lcr dump" the gateways are on-line, that's for sure. So I'm not understanding why LCR keep marking the gateways as down, even if they are answering to the OPTIONS request. Could this be a bug in the LCR module? Or maybe I'm missing something?. The silly part of all this is the gateways marked as down are always the same. Could this be related to a memory issue?? Can someone guide me here?. Thanks in advance, Regards, Ricardo Martinez.-

4 5

[Kamailio-Users] dispatcher failover between two groups
by Robert McGilvray 16 Oct '09

16 Oct '09

Kam 1.5 I'm using dialplan module to return a setid for the matching regex which is passed onto dispatcher to select the proper gateways. This works fine including failover to the next gateway in the group in failure_route. What I can't quite wrap my head around is how I can setup failover between different dispatcher groups. The scenario is as follows: 2 external PSTN gateways (sip trunks) in group 1 6 internal gateways with PSTN access via PRIs. Two gateways per group for a total of three separate setids in dispatcher. I'm doing some LCR across the 6 internal gateways which are all in different cities via regex matching in dialplan. What I'd like to do is try one of the internal groups first if it matches the local NPA/NXX, and if ALL gateways in that group fail move on to the external gateway group, and try all those gateways (and vice versa in my external gateways fail for non matching numbers). I thought of two possible solutions but I'm not sure if either will work how I expect. 1) Use the priority field in dispatcher to setup a large gateway group that has all the gateways I want to use. - I tested this out in kam 1.5 and the priority field seems to be ignored when hasing on call-id. - What is the intended usage of priority? Will setting two gateways to 0 and the rest to 10 cause the load balancing functions to only select the ones with 0? It seems to be ignoring priority and using all 8 gateways 2) Once in a failure_route can I move the processing into a request_route and call ds_select_dst again with a different setid and append it as a branch? Is there a cleaner solution? Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

1 0

[Kamailio-Users] Codecs
by Anders 16 Oct '09

16 Oct '09

Hi, A very basis 101-question, which I couldn't find the answer to elsewhere: how/where do I see/set codecs allowed and disallowed in Kamailio? Thanks!

2 1

Re: [Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks
by Örn Arnarson 15 Oct '09

15 Oct '09

Hi, I have exactly the same scenario; no NAT involved. Just trying to bridge between these two interfaces. Now, to clarify better what I have done; As I understand it, force_rtp_proxy / rtpproxy_offer needs to be used to inform RTP proxy that it will have to relay the packets. So, that pretty much makes it necessary to use that call. As I understand it, you can't call both fix_nated_sdp and force_rtp_proxy, as the SDP will be mangled (two IP addresses, one after the other). However, I can't get the SDP rewritten properly without using fix_nated_sdp. No matter how I call force_rtp_proxy, I can't specify my own IP address. It works as expected when using fix_nated_sdp though. Here's how I call it: force_rtp_proxy("coe","10.252.1.8"); I've tried multiple variants of flags, c, co, o, coe, coi, i, no flags, etc etc, but no matter what I do, the public interface IP address does not change to the one supplied in the function, and remains the public IP address. So, to summarize; I can change all the headers to be correct with the fix_nated_sdp function. When using that function, RTPProxy does not relay the RTP streams. When I use force_rtp_proxy / rtpproxy_offer, RTPProxy does indeed relay the streams, but does not alter the SDP IP address information to the required specifications. Is there something that I'm missing in the usage of the force_rtp_proxy function, or is it not working as documented? Best regards, Örn 2009/10/14 Alex Balashov <abalashov(a)evaristesys.com>: > Örn Arnarson wrote: > >> Oh, I forgot one thing; >> >> I'm using fix_nated_sdp, as previously mentioned. I have a hunch that >> because I'm not using force_rtp_proxy/rtpproxy_offer it's not being >> relayed (a hunch, because I don't fully understand how rtpproxy >> works). Problem is though, I can't get the IP addresses correct by >> using rtpproxy_offer. > > On behalf of Joe (same company) I will add that there is no NAT scenario > here. Both the private and public network are directly routable from the > rtpproxy box; there is no IP address oversubscription, and no reason to > ignore ports or addresses in SDP and replace them with received parameters. > This is not a far-end NAT traversal solution. > > What is basically being asked here is how to bridge between two interfaces > (internal, external) without NAT using rtpproxy, and if anyone can provide a > canonical implementation or example. > > We've got it running in bridge mode, e.g. > > rtpproxy -F -s udp:10.x.x.x:5050 -l 10.x.x.x/89.y.y.y > > ... and basically, are just wondering how to get this to work on the > Kamailio side using rtpproxy_offer()/rtpproxy_answer(). > > The RTP proxy machine is a dedicated host; the SIP proxies are virtual. > > > -- > Alex Balashov - Principal > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 >

1 0

[Kamailio-Users] SIP-T messages
by Luis Jimenez 15 Oct '09

15 Oct '09

Hi, can some one please guide me on how can i parse the received SIP-T messages from the SDP in Kamailio. Thanks in advance Luis Jimenez

2 1

[Kamailio-Users] How to remove route header field
by Mythili D 14 Oct '09

14 Oct '09

hii..Good morning. When we are making a call from one UA to other UA, it passes through proxies. ACK message also passes through the proxies. But i need the ACK message not to go through proxies. It must directly go from UA to UA. How can i make in cfg. There must not be route header filed in SIP meessage flow. Please help me. Thanks & Regards, Mythili.K

2 1

[Kamailio-Users] AVP problem
by Ricardo Martinez 13 Oct '09

13 Oct '09

Hello list. I'm getting this error trying to start the kamailio process /usr/local/sbin/kamailio[29243]: INFO:core:init_mod: initializing module mediaproxy /usr/local/sbin/kamailio[29243]: ERROR:core:pv_parse_spec: error searching pvar "avp" /usr/local/sbin/kamailio[29243]: ERROR:core:pv_parse_spec: wrong char [s/115] in [$avp(s:signaling_ip)] at [5 (5)] /usr/local/sbin/kamailio[29243]: CRITICAL:mediaproxy:mod_init: invalid AVP specification for signaling_ip_avp: `$avp(s:signaling_ip)' /usr/local/sbin/kamailio[29243]: ERROR:core:init_mod: failed to initialize module mediaproxy This is the line in the kamailio.cfg file : modparam("mediaproxy", "signaling_ip_avp", "$avp(s:signaling_ip)") I'm using kamailio 1.5.2 What I'm missing here?. Thanks, Ricardo Martinez

2 2

[Kamailio-Users] SIMPLE vs XMPP: The Resolution
by Iñaki Baz Castillo 13 Oct '09

13 Oct '09

Architectures of XMPP and SIMPLE (OMA specifications) for a full rich presence environment: - XMPP: http://oversip.net/public/XMPP.jpeg - SIMPLE (OMA): http://oversip.net/public/SIP_SIMPLE_OMA.jpeg -- Iñaki Baz Castillo <ibc(a)aliax.net>

8 19

[Kamailio-Users] SIMPLE vs XMPP: The Resolution
by Iñaki Baz Castillo 13 Oct '09

13 Oct '09

Attached the architecture of XMPP and SIMPLE (OMA specifications) for a presence environment. Self explanatory. -- Iñaki Baz Castillo <ibc(a)aliax.net>

4 7

[Kamailio-Users] Connecting two or more proxies
by Mythili D 13 Oct '09

13 Oct '09

Hi Good morning. I'm going to connect two proxies. I'm having two servers with openser installed. One user (UAC) will register with one server in one system which is not having openser and the other user (UAS) will register with second server in other syatem which is not having openser. I want to connect the two proxies through cfg file. Hoiw to give the connection. My IP's are UAC registered with 1000(a)192.168.64.77 at other system (192.168.64.85) 192.168.64.77 is one proxy. 192.168.64.94 is secind proxy. UAS registerd with 10(a)192.168.64.94 at other system (192.168.64.128) Now the person at 85 registered with 77 is going to call the person at 128 registered with 94. I have configuered my cfg's in this way: In 77 cfg file, i configuered *if(is_method("INVITE") && !src_ip=="192.168.64.77") * *{ * * rewritehost("192.168.64.77"); * *} * ** In 99 cfg file, i configuered *if(is_method("INVITE") && !src_ip=="192.168.64.94") * *{ * *rewritehost("192.168.64.94"); * *}* Is there any other way to connect. Please help me as soon as possible. Thanks & Regards, Mythili.K

2 1

[Kamailio-Users] keep tos value from incoming SIP message
by kévin l'heveder 13 Oct '09

13 Oct '09

Hi all, I found out how to set the tos value in a outgoing SIP message with the tos parameter from core-cookbook. But my issue is to forward an incoming SIP message keeping the tos value unchange. Is there already a way to do that, something already implemented? Or is there a way to get the tos value from incoming message and dynamically change the tos parameter? Regards,

2 1

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