- sr-users - kamailio.org

Best tool for crafting sip messages
by Anton Kvashenkin 13 Aug '12

13 Aug '12

Hello, guys. What is the best tool for crafting sip messages for testing purpose? For example, simple REGISTER or INVITE message. I'm new to kamailio (not sip) so it would be great for me to use debbuger module for step-by-step digging into cfg.

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$avp reset to <null> with no reason?
by Uri Shacked 13 Aug '12

13 Aug '12

Hi,**** ** ** I encountered a problem. While doing some avp's settings I notice that one avp that I use is changing to <null> without me doing anything.**** ** ** Here is the part of the cfg file:**** ………….**** ………….**** Route[SET]{**** …………..**** $avp(Sfeaturetype)=$avp(next_feature_type);**** …………**** Route(DIDSRV);**** }**** ** ** route[DIDSRV] {**** xlog("L_CRIT","$C(rg) SCRIPT: in DIDSRV type = $avp(Sfeaturetype), index = $avp(SfeatureIndx) $C(xx)\n");**** if($var(srvcount)>3){**** xlog("L_NOTICE","$C(rg)Too many services loops $C(xx)\n");**** update_stat("nts_subs_srv_loop", "+1");**** $avp(TRMCS)="10";**** t_reply(403,"too many loops");**** exit;**** }**** $var(srvcount)=($var(srvcount)+1);**** xlog("L_CRIT","$C(rg) SCRIPT: in DIDSRV before switch to $avp(SfeatureType) or $avp(next_feature_type) $C(xx)\n");**** switch ($avp(SfeatureType)){**** case "0" :**** ……….**** ………**** ** ** On the route "SET" I put the value from the $avp(next_feature_type) in $avp(Sfeaturetype).**** After that I call the route "DIDSRV" and use switch($avp(Sfeaturetype)). Before the switch I xlog the value of the avp twice.**** As you can see in the log below, on the first xlog print, the value is 0 (as I expect). On the second print it resets to <null>**** WHY?**** ** ** Log:**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:413]: qm_malloc(0x7fe6f9aff000, 72) returns address 0x7fe6fcc4d330 frag. 0x7fe6fcc4d300 (size=72) on 1 -th hit**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>: #033[0;31;42m SCRIPT: before DIDSRV type = 0, index = 8888888889 #033[0;39;49m**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>: #033[0;31;42m SCRIPT*: in DIDSRV type = 0*, index = 8888888889 #033<8888888889%C2%A0%20%23033> [0;39;49m**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:369]: qm_malloc(0x7fe7ff765010, 848) called from <core>: rvalue.c: rval_new_empty(236)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:413]: qm_malloc(0x7fe7ff765010, 848) returns address 0x7fe7ff8f25b0 frag. 0x7fe7ff8f2580 (size=1232) on 1 -th hit**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:437]: qm_free(0x7fe7ff765010, 0x7fe7ff8f25b0), called from <core>: rvalue.c: rval_destroy(141)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:461]: qm_free: freeing frag. 0x7fe7ff8f2580 alloc'ed from <core>: rvalue.c: rval_new_empty(236)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>: #033[0;31;42m SCRIPT*: in DIDSRV before switch to <null>* or 0 #033[0;39;49m **** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:369]: qm_malloc(0x7fe7ff765010, 928) called from <core>: rvalue.c: rval_new_empty(236)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:413]: qm_malloc(0x7fe7ff765010, 928) returns address 0x7fe7ff8f25b0 frag. 0x7fe7ff8f2580 (size=1232) on 1 -th hit**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:437]: qm_free(0x7fe7ff765010, 0x7fe7ff8f25b0), called from <core>: rvalue.c: rval_destroy(141)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:461]: qm_free: freeing frag. 0x7fe7ff8f2580 alloc'ed from <core>: rvalue.c: rval_new_empty(236)

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Kamailio LCR Module test scenario
by Fatima Chahrour~Vanrise Support 12 Aug '12

12 Aug '12

Dears, Am trying to apply Kamailio LCR feature lab test, after am finally able to run Kamailio with no errors, am not able to reach my target clarified in this following scenario: Call 961312345 - route the call to 192.x.x.15 in first place and if failed route the call to 192.x.x.10. I applied needed configuration in lcr tables (attached image) and the routing script is: route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (is_method("INVITE")) { if (!load_gws(1)) { sl_send_reply("503", "Unable to load gateways"); exit; } } if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using ourUSRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } BUT when attempting a call am getting the message 'Call failed: Not found .' Any help to achieve the successful call using above scenario is highly appreciated. Thanks in advance. F.Chahrour

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postgres in kamailio 3.3.0-9.1
by Andrew O. Zhukov 08 Aug '12

08 Aug '12

The same trouble with: completely updated Centos 5 the last Kamailio RPM from http://download.opensuse.org/repositories/home:/kamailio:/telephony/RedHat_… and postgresql-libs.x86_64.8.1.23-5.el5_8 several month ago I try postgresql84-libs.x86_64 with the same result. It's not possible to use Kamailio RPM with postgres backend. Need to assemble it manually. Bruno Bresciani wrote: > Hi, > > I configure the Kamailio 3.1.2 with postgres but I cann't start. In the > log file is generated the following error: > > // > > ERROR: load_module: could not open module > </home2/local/kamailio/lib/kamailio/modules/db_ > postgres.so>: /home2/local/kamailio/lib/kamailio/modules/db_postgres.so: > undefined symbol: PQdescribePrepared Have you built Kamailio yourself? Looks like db_postgres.so can not find the postgresql libraries, or it was built with a different library version. Maybe ldd can give you some details: ldd /home2/local/kamailio/lib/kamailio/modules/db_postgres.so regards Klaus

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Re: [SR-Users] Problem Starting kamailio on Debian Squeeze [solved]
by JR Richardson 08 Aug '12

08 Aug '12

On Wed, Aug 8, 2012 at 12:13 PM, JR Richardson <jmr.richardson(a)gmail.com> wrote: > Hi All, > > I'm running redundant kamailio 3.0.4 servers in production, have been > for a long time with great success. They were installed on debian > Lenny. One of my servers crashed. I can't seem to do a debian lennyy > install because that version is archived now. I'm trying to install > on debian squeeze but still using kamailio 3.0.4 for compatibility > reasons. The install went pretty normal except during boot up, > kamailio starts before mysql and networking so it starts then exists. > > Once the server is fully booted, kamailio starts with the init script > without error. > > I have adjusted the init script to check for networking and mysql to > start first and performed an 'update-rc.d kamailio defaults' but still > no luck. > > Any help will be appreciated. > > Thanks. > > JR > -- > JR Richardson > Engineering for the Masses Yep, this is an http://wiki.debian.org/LSBInitScripts/DependencyBasedBoot issue. Thanks Alex. The old init scrip that ships with Kamailio 3.0.4 that I am using is not 100% compatible so I pulled a hack out of the hat and inserted 'sleep 20' before of 'check_fork ()' in the kamailio init script. Kamailio now waits long enough for all services to start before launching, simple but effective. Thanks. JR -- JR Richardson Engineering for the Masses

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Porblem Starting kamailio on Debian Squeeze
by JR Richardson 08 Aug '12

08 Aug '12

Hi All, I'm running redundant kamailio 3.0.4 servers in production, have been for a long time with great success. They were installed on debian Lenny. One of my servers crashed. I can't seem to do a debian lennyy install because that version is archived now. I'm trying to install on debian squeeze but still using kamailio 3.0.4 for compatibility reasons. The install went pretty normal except during boot up, kamailio starts before mysql and networking so it starts then exists. Once the server is fully booted, kamailio starts with the init script without error. I have adjusted the init script to check for networking and mysql to start first and performed an 'update-rc.d kamailio defaults' but still no luck. Any help will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

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Kernel Module Issue
by Vijay Thakur 08 Aug '12

08 Aug '12

Thanks for the hint. What is the file name in /etc/modprob.d/ where i am supposed to enter this module name to disable it. Thanks Vijay Tha Message: 1 Date: Mon, 06 Aug 2012 09:01:03 -0400 From: Richard Fuchs<rfuchs(a)sipwise.com> Subject: Re: [SR-Users] Kernel Droping SIP packet To:sr-users@lists.sip-router.org Message-ID:<501FC00F.6000604(a)sipwise.com> Content-Type: text/plain; charset="iso-8859-1" On 08/06/12 06:08, Vijay Thakur wrote: > Hi all, > > I have configure Kamailio 3.1.5 Server. All things are working fine. > When i make a call from Soft phone (X-Lite) to iphone, all is working > fine. But in other case call from iphone to Softphone is not working, > even not ringing. During checking the logs i am getting the error: > > Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= > MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 > DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF > PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 > RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) This is coming from nf_conntrack_sip, which is a netfilter connection tracking kernel module for SIP. I've never used it, but judging from what Google brings up, it seems to be very buggy. You should be able to just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work and/or if you want to keep it from auto-loading, you can blacklist it in /etc/modprobe.d/ and then reboot. HTH copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

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AVPOPS / TM behavior
by Brandon Armstead 08 Aug '12

08 Aug '12

Hello, I am curious if there is any documentation on how AVP's processing works in the following scenario below. UAC 1 -> KAMAILIO -> KAMAILIO -> DEST It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once I relay back to the same KAMAILIO proxy (self)? Is there any documentation on why or when this would occur? Is there a better way to handle such a scenario? i.e. more dynamic internal routing, vs relaying to self. Thanks as always in advance! Sincerely, Brandon Armstead

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local_timer: add_timeout: 0 expire timer added
by Barthel Marco (CI/AFU1) 08 Aug '12

08 Aug '12

Hi, Occasionally I can see the following log output from Kamailio in our systems: WARNING: <core> [local_timer.c:97]: WARNING: local_timer: add_timeout: 0 expire timer added Can someone explain what it means? Many thanks. Mit freundlichen Grüßen / Best regards Marco Barthel Robert Bosch GmbH (CI/AFU1) www.bosch.com<http://www.bosch.com> Tel. +49 711 811-3602341 marco.barthel(a)de.bosch.com<mailto:marco.barthel@de.bosch.com> Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart, HRB 14000; Aufsichtsratsvorsitzender: Franz Fehrenbach; Geschäftsführung: Volkmar Denner, Siegfried Dais; Stefan Asenkerschbaumer, Bernd Bohr, Rudolf Colm, Dirk Hoheisel, Christoph Kübel, Uwe Raschke, Wolf-Henning Scheider, Werner Struth, Peter Tyroller

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Re: [SR-Users] Kamailio with APNS
by Duong Manh Truong 08 Aug '12

08 Aug '12

As your first solution with UAC module: "As a plan B we instead opted for sending via uac module the invite to an oversip instance (it's open-source since a week or two), which triggers an HTTP request towards APNS, and on kamailio check with inv-timers every few seconds whether the client came online in the meanwhile, then complete the call. The drawback is that you can wait only for so many loop interations due to the max branch limitation in kamailio (you'd need to recompile it to set it higher)." I read about the UAC module but feeling very hard to implement it as your suggestion, http://kamailio.org/docs/modules/3.3.x/modules_k/uac.html Please give me and others more details about this? Which functions those we need ? Are they in: UAC module, Exec module, tm module ?? Thanks

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set_advertised_port using pseudovariable
by Neven Boric 07 Aug '12

07 Aug '12

Hi, I'm trying to use Kamailio as an outbound proxy behind a NAT (all clients and kamailio itself are behind the same NAT): UACs -> Kamailio -> NAT router -> PBX (hosted, public server) I figured I could detect the external source port used by the router by periodically sending an OPTIONS request to the public server, then capture the rport value in the reply, and use that port as an input to set_advertised_port. Because I'm doing this periodically, it should have the added value of keeping the NAT mapping alive. Otherwise, I should be able to detect it, as the rport value will change. I managed to periodically send the OPTIONS request, read the rport value on the reply message and store it on a pseudovariable using: $var(rport) = $sel(v.rport); But when I try to use that value to set the advertised port, I get an error, as if set_advertised_port only accepted literal values: if( $var(rport) != 0) { set_advertised_port($var(rport)); } Not starting Kamailio: invalid configuration file! -e 0(4692) : <core> [cfg.y:3591]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 504, column 24-34: syntax error 0(4692) : <core> [cfg.y:3594]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 504, column 35: bad argument, string expected ERROR: bad config file (2 errors) So, am I missing something or does set_advertised_port actually only accept literal values? Thanks Neven Boric

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Re: [SR-Users] sr-users Digest, Vol 87, Issue 18
by Vijay Thakur 07 Aug '12

07 Aug '12

Thanks all of your for reply. Here is the output of command: root@li496-23:~# lsmod | grep -i sip root@li496-23:~# This command is giving blank output. This is linnode server. So the zcat /proc/config.gz command reveals the kernel loaded modules. You can see complete output at pastebin: http://pastebin.com/a0iqpc4D. Thanks in advance Vijay Th. On Monday 06 August 2012 08:26 PM, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Kernel Droping SIP packet (Richard Fuchs) > 2. Re: Kernel Droping SIP packet (Alex Balashov) > 3. Re: Kernel Droping SIP packet (Konstantin M.) > 4. Re: Kernel Droping SIP packet (Alex Balashov) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 06 Aug 2012 09:01:03 -0400 > From: Richard Fuchs <rfuchs(a)sipwise.com> > Subject: Re: [SR-Users] Kernel Droping SIP packet > To: sr-users(a)lists.sip-router.org > Message-ID: <501FC00F.6000604(a)sipwise.com> > Content-Type: text/plain; charset="iso-8859-1" > > On 08/06/12 06:08, Vijay Thakur wrote: >> Hi all, >> >> I have configure Kamailio 3.1.5 Server. All things are working fine. >> When i make a call from Soft phone (X-Lite) to iphone, all is working >> fine. But in other case call from iphone to Softphone is not working, >> even not ringing. During checking the logs i am getting the error: >> >> Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= >> MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 >> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF >> PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 >> RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) > This is coming from nf_conntrack_sip, which is a netfilter connection > tracking kernel module for SIP. I've never used it, but judging from > what Google brings up, it seems to be very buggy. You should be able to > just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work > and/or if you want to keep it from auto-loading, you can blacklist it in > /etc/modprobe.d/ and then reboot. > > HTH > >

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LCR - inconsistent return results for gateway selection.
by Graham Wooden (personal) 06 Aug '12

06 Aug '12

Hello, I am noticing that some of the time (not always consistent) that load_gw() / next_gw() will not always return the gateway with the lowest priority number. For example, when I look up a route for 205595, I have three grp_ids (gateways) that services the destination; with them being separated with 1, 2, and 3 priorities: id prefix from_uri grp_id priority 7468 205595 NULL 1 1 7469 205595 NULL 2 2 7470 205595 NULL 7 3 About 1/2 of the time, it will chose the priority 1 gateway, as it should. But randomly, it will choose the other gateways on the first try. In this particular case, all three gateways are going to the same destination IP (just using different hostnames in the "gw" mySQL table) - so I know the gateways are up. Also, I am not noticing any failures and going to the next gateway, like it choosing #1, it failing and then going to #2. What kind of debugging can I do on this gateway selection process? I am running 1.5.5 notls, and have approximately 425K routes. Kamailio is started with "-m 512" for the additional memory. Thanks, -graham

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Re: [SR-Users] Kernel Droping SIP packet
by Alex Balashov 06 Aug '12

06 Aug '12

Can you paste this output too? lsmod | grep -i sip -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur(a)loopmethods.com> wrote:Thanks for prompt reply. Here is the out put if command iptables -L -n: Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server. Vijay TH -============================================================================== Date: Mon, 06 Aug 2012 06:35:04 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] Kernel Droping SIP packet To: sr-users(a)lists.sip-router.org Message-ID: <17t1fd0fbly84yy2vuy8qlcp.1344249304618(a)email.android.com> Content-Type: text/plain; charset="utf-8" You might consider pasting the actual output of: iptables -L -n This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur(a)loopmethods.com> wrote:Hi all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) I have not implemented any firewall. You can check the out put of my iptables : http://pastebin.com/i3zUfVeb The SIP server is hosted at linnode. With thanks in advance. Sorry dual posting. Vijay TH

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Re: [SR-Users] Kernel Droping SIP packet
by Alex Balashov 06 Aug '12

06 Aug '12

Ah, you short-circuited where I was going with this. Couldn't remember the name of the module. Yep, Vijay. What Richard said. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Richard Fuchs <rfuchs(a)sipwise.com> wrote:On 08/06/12 06:08, Vijay Thakur wrote: > Hi all, > > I have configure Kamailio 3.1.5 Server. All things are working fine. > When i make a call from Soft phone (X-Lite) to iphone, all is working > fine. But in other case call from iphone to Softphone is not working, > even not ringing. During checking the logs i am getting the error: > > Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= > MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 > DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF > PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 > RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) This is coming from nf_conntrack_sip, which is a netfilter connection tracking kernel module for SIP. I've never used it, but judging from what Google brings up, it seems to be very buggy. You should be able to just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work and/or if you want to keep it from auto-loading, you can blacklist it in /etc/modprobe.d/ and then reboot. HTH _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Kernel Droping SIP packet
by Vijay Thakur 06 Aug '12

06 Aug '12

Hi all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) I have not implemented any firewall. You can check the out put of my iptables : http://pastebin.com/i3zUfVeb The SIP server is hosted at linnode. With thanks in advance. Sorry dual posting. Vijay TH

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Re: [SR-Users] B2BUA issues
by Alex Balashov 06 Aug '12

06 Aug '12

For your relatively narrow, specific applications in your topology, no. You'd be better off installing SEMS per se. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/phillman25 <phillman25(a)gmail.com> wrote:Hello Alex Will try with SEMS first, found something called sip:provider CE v2.4 from http://www.sipwise.com/news/announcements/spce-v2_4-release/ if i'm not mistaken, this seems to combine Kamailio with SEMS? Do you think that this might be an easier installation rather than installing SEMS on its own as it seems to provide more documentation? Thanks again! On Mon, Aug 6, 2012 at 1:33 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: The short answer to your latter question is: yes. Cisco media and PSTN gateways have never hairpinned SIP-to-SIP calls well, even when officially supported. Asterisk has a lower learning curve due to the abundance of information and tutorials, but SEMS would make more sense, since all you need is a signaling B2BUA. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ phillman25 <phillman25(a)gmail.com> wrote: Hi Alex Thanks for your prompt reply. The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work. Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios? From your experience what do you think would be a better solution? Thanks again! Phillip ======================== Message: 2 Date: Mon, 06 Aug 2012 04:26:28 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] B2BUA issues To: sr-users(a)lists.sip-router.org Message-ID: <501F7FB4.8040700(a)evaristesys.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed What is the larger objective? Are you using the PGW purely as a B2BUA? If so, that's a colossally overblown waste of resources; just use something like SEMS or Asterisk. On 08/06/2012 04:24 AM, phillman25 wrote: > Dear List > > I am trying to accomplish the following: > > > Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco > PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> > Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) > > When trying the above scenario, the call is silent and drops after a few > seconds. In syslog i observe the following error: > > *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing > name-addr (sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060>>)* > > Looking at the sip trace i see that his might be caused by the ACK > message received from the ASTERISK PABX? : > > ACK sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport > Route: > <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@ > yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea> > Max-Forwards: 70 > From: "22498045" <sip:22498045@192.168.10.189 > <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea > To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 > Contact: <sip:22498045@192.168.10.189:5060 > <http://sip:22498045@192.168.10.189:5060>> > Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 > <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060> > CSeq: 102 ACK > User-Agent: FPBX-2.8.1(1.8.12.0) > Content-Length: 0 > > > After contacting Cisco they informed us that issue is cause by B2BUA > from our current version of Cisco PGW 2200 that doesn't support this > feature. Is there a module, solution that i can implement on Kamailio > that could temporarily resolve this issue? > > Thanking you in advance. > > Phillip

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Re: [SR-Users] Kernel Droping SIP packet
by Vijay Thakur 06 Aug '12

06 Aug '12

Thanks for prompt reply. Here is the out put if command iptables -L -n: *Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server. Vijay TH -============================================================================== * Date: Mon, 06 Aug 2012 06:35:04 -0400 From: Alex Balashov<abalashov(a)evaristesys.com> Subject: Re: [SR-Users] Kernel Droping SIP packet To:sr-users@lists.sip-router.org Message-ID:<17t1fd0fbly84yy2vuy8qlcp.1344249304618(a)email.android.com> Content-Type: text/plain; charset="utf-8" You might consider pasting the actual output of: iptables -L -n This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web:http://www.evaristesys.com/Vijay Thakur<vijay.thakur(a)loopmethods.com> wrote:Hi all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) I have not implemented any firewall. You can check the out put of my iptables :http://pastebin.com/i3zUfVeb The SIP server is hosted at linnode. With thanks in advance. Sorry dual posting. Vijay TH * *

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Re: [SR-Users] B2BUA issues
by Alex Balashov 06 Aug '12

06 Aug '12

The short answer to your latter question is: yes. Cisco media and PSTN gateways have never hairpinned SIP-to-SIP calls well, even when officially supported. Asterisk has a lower learning curve due to the abundance of information and tutorials, but SEMS would make more sense, since all you need is a signaling B2BUA. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/phillman25 <phillman25(a)gmail.com> wrote:Hi Alex Thanks for your prompt reply. The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work. Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios? From your experience what do you think would be a better solution? Thanks again! Phillip ======================== Message: 2 Date: Mon, 06 Aug 2012 04:26:28 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] B2BUA issues To: sr-users(a)lists.sip-router.org Message-ID: <501F7FB4.8040700(a)evaristesys.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed What is the larger objective? Are you using the PGW purely as a B2BUA? If so, that's a colossally overblown waste of resources; just use something like SEMS or Asterisk. On 08/06/2012 04:24 AM, phillman25 wrote: > Dear List > > I am trying to accomplish the following: > > > Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco > PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> > Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) > > When trying the above scenario, the call is silent and drops after a few > seconds. In syslog i observe the following error: > > *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing > name-addr (sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060>>)* > > Looking at the sip trace i see that his might be caused by the ACK > message received from the ASTERISK PABX? : > > ACK sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport > Route: > <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@ > yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea> > Max-Forwards: 70 > From: "22498045" <sip:22498045@192.168.10.189 > <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea > To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 > Contact: <sip:22498045@192.168.10.189:5060 > <http://sip:22498045@192.168.10.189:5060>> > Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 > <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060> > CSeq: 102 ACK > User-Agent: FPBX-2.8.1(1.8.12.0) > Content-Length: 0 > > > After contacting Cisco they informed us that issue is cause by B2BUA > from our current version of Cisco PGW 2200 that doesn't support this > feature. Is there a module, solution that i can implement on Kamailio > that could temporarily resolve this issue? > > Thanking you in advance. > > Phillip

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Re: [SR-Users] Kernel Droping SIP packet
by Alex Balashov 06 Aug '12

06 Aug '12

You might consider pasting the actual output of: iptables -L -n This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur(a)loopmethods.com> wrote:Hi all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) I have not implemented any firewall. You can check the out put of my iptables : http://pastebin.com/i3zUfVeb The SIP server is hosted at linnode. With thanks in advance. Sorry dual posting. Vijay TH _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] B2BUA issues
by phillman25 06 Aug '12

06 Aug '12

Hi Alex Thanks for your prompt reply. The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work. Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios? >From your experience what do you think would be a better solution? Thanks again! Phillip ======================== Message: 2 Date: Mon, 06 Aug 2012 04:26:28 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] B2BUA issues To: sr-users(a)lists.sip-router.org Message-ID: <501F7FB4.8040700(a)evaristesys.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed What is the larger objective? Are you using the PGW purely as a B2BUA? If so, that's a colossally overblown waste of resources; just use something like SEMS or Asterisk. On 08/06/2012 04:24 AM, phillman25 wrote: > Dear List > > I am trying to accomplish the following: > > > Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco > PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> > Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) > > When trying the above scenario, the call is silent and drops after a few > seconds. In syslog i observe the following error: > > *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing > name-addr (sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060>>)* > > Looking at the sip trace i see that his might be caused by the ACK > message received from the ASTERISK PABX? : > > ACK sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport > Route: > <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b. 66623da5>,<sip:22030305@ > yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx; lr=on;ftag=as166b1eea> > Max-Forwards: 70 > From: "22498045" <sip:22498045@192.168.10.189 > <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea > To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 > Contact: <sip:22498045@192.168.10.189:5060 > <http://sip:22498045@192.168.10.189:5060>> > Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 > <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060> > CSeq: 102 ACK > User-Agent: FPBX-2.8.1(1.8.12.0) > Content-Length: 0 > > > After contacting Cisco they informed us that issue is cause by B2BUA > from our current version of Cisco PGW 2200 that doesn't support this > feature. Is there a module, solution that i can implement on Kamailio > that could temporarily resolve this issue? > > Thanking you in advance. > > Phillip

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B2BUA issues
by phillman25 06 Aug '12

06 Aug '12

Dear List I am trying to accomplish the following: Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) When trying the above scenario, the call is silent and drops after a few seconds. In syslog i observe the following error: *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing name-addr (sip:22030305@192.168.10.189:5060>)* Looking at the sip trace i see that his might be caused by the ACK message received from the ASTERISK PABX? : ACK sip:22030305@192.168.10.189:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@ yyy.yyy.yyy.yyy ;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea> Max-Forwards: 70 From: "22498045" <sip:22498045@192.168.10.189>;tag=as166b1eea To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 CSeq: 102 ACK User-Agent: FPBX-2.8.1(1.8.12.0) Content-Length: 0 After contacting Cisco they informed us that issue is cause by B2BUA from our current version of Cisco PGW 2200 that doesn't support this feature. Is there a module, solution that i can implement on Kamailio that could temporarily resolve this issue? Thanking you in advance. Phillip

2 1

Question about Proxy-Authorization header in ACK
by Ivan Milivojevic 06 Aug '12

06 Aug '12

Hi all, I have a one question regarding ACK and Proxy-Authorization header. I am testing Kamailio 3.3 as SIP proxy,default config. I made one call with 2 phones where one has Public IP while other is behind NAT. SIP clients are Panasonic KX-UT133 and other is 1-NET (ex Sweden Mobile&CDMA provider ,over 50M users) Everything is working just fine, both RTP and SIP. On the other hand one of the clients,1-NET one, send Proxy-Authorization in ACK when he receives 200 OK from Kamailio. PUBLIC_IP_USER1 - user 1000 PUBLIC_IP_USER2 - user 1001 KAMAILIO ==> 1-NET SIP/2.0 200 OKr\n Record-Route: <sip: <sip:KAMAILIO_PUBLIC_IP;lr=on;nat=yes> KAMAILIO_PUBLIC_IP;lr=on;nat=yes> Via: SIP/2.0/UDP PUBLIC_IP_USER1:5064;rport=5064;branch=z9hG4bK934894606 Call-ID: 221374358 From: blabla <sip:1000@ <sip:1000@KAMAILIO_PUBLIC_IP> KAMAILIO_PUBLIC_IP>;tag=507511069 To: <sip:1001@ <sip:1001@KAMAILIO_PUBLIC_IP:5060> KAMAILIO_PUBLIC_IP:5060>;tag=2148378512 CSeq: 21 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,UPDATE,OPTIONS,NOTIFY,REFER Contact: <sip:1001@ <sip:1001@PUBLIC_IP_USER2:1027> PUBLIC_IP_USER2:1027> Require: replaces Content-Type: application/sdp Server: Panasonic_KX-UT133NE/01.081 (0080f0cedd83) Content-Length: 182 v=0 o=- 1343736535 1343736535 IN IP4 KAMAILIO_PUBLIC_IP s=- c=IN IP4 KAMAILIO_PUBLIC_IP t=0 0 m=audio 20412 RTP/AVP 18 a=rtpmap:18 G729/8000 a=sendrecv a=ptime:20 a=nortpproxy:yes 1-NET ==> KAMAILIO ACK sip:1001@ <sip:1001@PUBLIC_IP_USER2:1027> PUBLIC_IP_USER2:1027 SIP/2.0 Via: SIP/2.0/UDP PUBLIC_IP_USER1:5064;rport;branch=z9hG4bK642028490 Route: <sip: <sip:KAMAILIO_PUBLIC_IP;lr=on;nat=yes> KAMAILIO_PUBLIC_IP;lr=on;nat=yes> From: blabla <sip:1000@ <sip:1000@KAMAILIO_PUBLIC_IP> KAMAILIO_PUBLIC_IP>;tag=507511069 To: <sip:1001@ <sip:1001@KAMAILIO_PUBLIC_IP:5060> KAMAILIO_PUBLIC_IP:5060>;tag=2148378512 Call-ID: 221374358 CSeq: 21 ACK Contact: <sip:1000@ <sip:1000@PUBLIC_IP_USER1:5064> PUBLIC_IP_USER1:5064> [truncated] Proxy-Authorization: Digest username="1000", realm="KAMAILIO_PUBLIC_IP", nonce="UBfL8lAXysb5tJCs80ZnthyPl9IzmRZk", uri="sip:1001@KAMAILIO_PUBLIC_IP:5060", response="7824519cdad9f1c2c79027a2d7522344", algorithm=MD5, cnonce="0a4f113b", q Max-Forwards: 70 User-Agent: Serbia_2.00 Content-Length: 0 I attach txt file with call flow, I can send pcap also. I think that the issue is related with bad client but I need another opinion. Does anyone has an idea about this issue? Best Regards, Ivan

2 1

REGISTER replication to multiple hosts
by Henning Holtschneider 06 Aug '12

06 Aug '12

Hello, I'm trying to migrate an existing configuration from OpenSIPS 1.4.x to Kamailio 3.2.x. The old configuration replicates REGISTER requests to multiple servers using this code in the REGISTER routing block: add_sock_hdr("Local-Sock"); add_rcv_param(); append_branch("sip:1.1.1.1:5060"); t_replicate("sip:2.2.2.2:5060"); exit; From what I read in the documentation, the behaviour of append_branch() is different in Kamailio, so I modified my code like this: add_sock_hdr("Local-Sock"); add_rcv_param(); append_branch(); seturi(sip:1.1.1.1:5060"); t_replicate("sip:2.2.2.2:5060"); exit; Unfortunately, I do not see any REGISTER requests arriving on 1.1.1.1 so I assume that I made a mistake when forking the request. Can anyone point me in the right direction? Thanks, Henning Holtschneider -- LocaNet oHG - http://www.loca.net Lindemannstrasse 81, D-44137 Dortmund tel +49 231 91596-25, fax +49 231 91596-55 sip 25(a)voip.loca.net Registergericht Amtsgericht Dortmund HRA 14208 Geschäftsführer Sven Haufe, Henning Holtschneider

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Re: [SR-Users] sr-users Digest, Vol 87, Issue 9
by Vijay Thakur 06 Aug '12

06 Aug '12

Hi Konstantin, Thanks for your kind reply. You can check the iptables output at : http://pastebin.com/i3zUfVeb. I hope that this will give you enough clue in right direction. With Regards, Vijay Thakur ====================================================================================================================== Message: 3 Date: Fri, 3 Aug 2012 17:38:23 +0300 From: "Konstantin M."<evilzluk(a)gmail.com> Subject: Re: [SR-Users] kernel: nf_ct_sip: dropping packetIN= OUT=eth0 Error To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> Message-ID: <CAEYtTzdeY88X-GC+wDwLou+smsB60h+M36h7xwCu448V0YtD2g(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Looks like your firewall is passing a state RELATED,ESTABLISHED and don't have a permit rule for a state NEW. Check your iptables or simply do a command: iptables-save > /tmp/iptables.txt and paste this file (/tmp/iptables.txt) to pastebin 2012/8/3 Vijay Thakur<vijay.thakur(a)loopmethods.com> > Hello all, > > I have configure Kamailio 3.1.5 Server. All things are working fine. When > i make a call from Soft phone (X-Lite) to iphone, all is working fine. But > in other case call from iphone to Softphone is not working, even not > ringing. During checking the logs i am getting the error: > > Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= > MAC=f2:3c:91:ae:92:36:c8:4c:**75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 > DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP > SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK > URGP=0 OPT (0101080A000E20610932B25A) > > This is server is hosted on Linnode. > > Kindly guide me to solve the problem > > -- > Best Regards, > > Vijay Thakur > (Assistant Manager - Networks) > Mobile : +91 8744018065 > Mail :vijay.thakur@loopmethods.com

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Gorka left a message for you
by Badoo 05 Aug '12

05 Aug '12

Gorka left a message for you Only you can see the sender and content of your message, and you can delete it anytime. You can instantly reply using our message exchange system: http://eu1.badoo.com/0106730996/in/UJF4Z.6PYhQ/?lang_id=3&m=63&mid=501e9322… Some other people in the area who are on Badoo Ali (Jeddah, Saudi Arabia) Fnoo meni jnoo (Jeddah, Saudi Arabia) مجووووودي (Jeddah, Saudi Arabia) http://eu1.badoo.com/0106730996/in/UJF4Z.6PYhQ/?lang_id=3&m=63&mid=501e9322… If the link in this message does not work, try copying and pasting it into your browser. This email is part of our delivery procedure for the message sent by Gorka. If you have received this email by mistake, please ignore it. The message will be deleted soon. Have fun! The Badoo Team You have received this email from Badoo Trading Limited (postal address below). http://eu1.badoo.com/impersonation.phtml?lang_id=3&email=serusers%40iptel.o… Badoo Trading Limited is a limited company registered in England and Wales under CRN 7540255 with its registered office at 12 Red Lion Square, London, WC1R 4QD.

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Getting started with Kamailio
by Markus 05 Aug '12

05 Aug '12

Hello! I'm new to this list and Kamailio. First, thanks to all the programmers that developed this software. I'd like to ask several questions on how to get started with Kamailio and the basic understanding of it. I'm trying to figure out Kamailio only since yesterday. Right now I have a single Asterisk server that handles signaling and media. For accounting and user administration I'm using a2billing with MySQL on it. For redundancy and scalability I would like to create the following setup: France: BGP anycast 195.5.5.0/24 2 nodes with Kamailio (IP 195.5.5.5) active + standby 2 nodes with Asterisk + a2billing + DRBD for active-active MySQL cluster. Spain: BGP anycast 195.5.5.0/24 2 nodes with Kamailio (IP 195.5.5.5) active + standby 2 nodes with Asterisk + a2billing + DRBD for active-active MySQL cluster. (a2billing MySQL DB will be always synchronous over all nodes in Spain and France) This will give me: - Network redundancy (data center burns down in France, I stay online in Spain). This will be handled by BGP anycast. - Signaling redundancy (Kamailio active + standby). This will be handled by Linux-HA or something like that. - Media redundancy (2x Asterisk per country). This will be handled by Kamailio dispatcher. - Application/database redundancy (2x a2billing per country + synchronous MySQL DB everyhwere). This will be handled by DRBD and MySQL. = High redundancy, I can sleep and it can scale. What I would like to achieve besides the above: - Give only a single IP address to all customers and termination providers (the same IP address), 195.5.5.5. I'm a SIP noob, so I have to ask: - How do I do the Kamailio part? ;-) ... I have seen this how-to for Kamailio/Asterisk realtime: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb But it feels like "overkill". If I would not have any users that actually REGISTER (e.g. in a pure wholesale termination environment), I would not need Kamailio/Asterisk realtime integration. Correct? - Is there no way around changing the Asterisk side (activating realtime, new MySQL DB) when I have users that do need to REGISTER? If I would not be using a2billing I could probably handle all the registrations in Kamailio only? - Since Kamailio and Asterisk will not be on the same box, what is the recommended way for Kamailio securely communicating with the MySQL database on the Asterisk server? Does Kamailio support SSL with MySQL? - If I use RTPproxy on the Kamailio server, every customer and termination provider would connect to 1 single IP address, because both media and signaling comes from that IP. Correct? - If I don't use RTPproxy, and have canreinvite=yes on my Asterisk servers, customers would get the media, when placing PSTN calls, directly from my termination providers (I would like to avoid that). Correct? - If I don't use RTPproxy, and have canreinvite=no on my Asterisk servers, customers will get the media directly from my Asterisk servers, but termination providers that filter based on IP addresses they would have to allow all Asterisk IP addresses in their filters (same for customers, actually). Correct? Right now I didn't have to worry about such things because media and signaling were handled by a single Asterisk box with canreinvite=no everywhere. - Last question for now: why does it seem like important developers of SIP software such as Kamailio and yate are originating from Romania and are female? Just a coincidence? :-) Regards Markus

2 2

RTP over TCP
by Techie Sup 05 Aug '12

05 Aug '12

Hi All, Is there any possibility of sending RTP over TCP, because we are building a softphone based on PJSIP, but seems that some ISP are blocking RTP packets over UDP. Any solutions for this. Regards Gloria

2 3

kernel: nf_ct_sip: dropping packetIN= OUT=eth0 Error
by Vijay Thakur 03 Aug '12

03 Aug '12

Hello all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) This is server is hosted on Linnode. Kindly guide me to solve the problem -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail : vijay.thakur(a)loopmethods.com Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web: www.loopmethods.com LOOP Disclaimer ------------------------------------------------------------------------------------------------- This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

2 1

302 redirect with 2 numbers registered
by Mino Haluz 03 Aug '12

03 Aug '12

Hi, one number is registered on 2 phones. Phone1 has Always redirect set to another number. When incoming call is initiated, Phone2 is ringing and Phone1 sends 302 to the proxy. However the proxy does not send 302 to the caller (for ex. GW), but it waits for timeout of the Phone2. Then the proxy sends 302 to the caller. Can I do in kamailio, that it will ring on the Phone1 and also on the number where it is redirected? I know kamailio is a proxy and cannot initiate a call, but is there any solution? Thanks. Mino

2 1

Question on ACK behavior
by Varsha Venkatraramani 03 Aug '12

03 Aug '12

** ** * * ** ** Hi, ** ** I have a question regarding the below ACK response Kamilaio receives from a carrier. Can someone please help me understand why the “*Route: sip:callmanager@192.168.160.43:5060> *sent from Kamailio is missing a “<” as shown in the captures below? Our Internal proxy is treating that as a malformed header and dropping the packet. ** ** ** ** *ACK from CARRIER* * * U 2012/08/01 18:32:52.219852 4.55.18.227:5060 -> 192.168.160.47:5060**** *ACK sip:callmanager@192.168.160.43:5060 SIP/2.0.* Via: SIP/2.0/UDP 4.55.18.227:5060;branch=z9hG4bK04B0eef33040e9b8e70.**** From: sip:+14088442721@4.55.18.227:5060;tag=gK043001e3.**** To: sip:+19728931740@192.168.160.47:5060;tag=b307370c678f3b44.**** Call-ID: 295226_50030734(a)4.55.18.227.**** CSeq: 18079 ACK.**** Max-Forwards: 70.**** Route: <sip:192.168.160.47:5060;lr=on>.**** *Route: <sip:2c6c6d1ab58c623912f6b8a6ee526982@192.168.160.44:5060>.* Content-Length: 0.**** .**** *ACK FORWARDED TO SIP PROXY* ** ** U 2012/08/01 18:32:52.220612 192.168.160.47:5060 -> 192.168.160.44:5060**** *ACK sip:2c6c6d1ab58c623912f6b8a6ee526982@192.168.160.44:5060 SIP/2.0.* Via: SIP/2.0/UDP 192.168.160.47;branch=z9hG4bKcydzigwkX.**** Via: SIP/2.0/UDP 4.55.18.227:5060;branch=z9hG4bK04B0eef33040e9b8e70.**** From: sip:+14088442721@4.55.18.227:5060;tag=gK043001e3.**** To: sip:+19728931740@192.168.160.47:5060;tag=b307370c678f3b44.**** Call-ID: 295226_50030734(a)4.55.18.227.**** CSeq: 18079 ACK.**** Max-Forwards: 69.**** Content-Length: 0.**** *Route: sip:callmanager@192.168.160.43:5060>.* I have attached the config file for you reference. Kamailio version is 3.2.3 **** ** ** Thank You**** Varsha****

2 2

centos (fedora) service startup file
by andre 02 Aug '12

02 Aug '12

Somwhere losted since openser 1.0 file kamailio.init Please fix in function start: start() { echo -n $"Starting $prog: " ..... [ $RETVAL = 0 ] && touch something to [ $RETVAL = 0 ] && touch someting && success I mean call function success in the end of the expression. Thanks in advance.

1 0

Kamailio v3.3.1 Released
by Daniel-Constantin Mierla 02 Aug '12

02 Aug '12

Hello, Kamailio SIP Server v3.3.1 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.0. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.1. For more details about version 3.3.1 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/08/kamailio-v3-3-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

Outbound and Registrar. No binding update
by José Luis Millán 02 Aug '12

02 Aug '12

Hi, I'm using Kamilio 3.3.0 as registrar server. I`m using an outbound proxy so 'use_path' parameter or 'registrar' module is enabled. According to RFC 5626, a re-registration from a specific combination of AoR, instance_id and reg_id must update the binding. "" If the registrar receives a re-registration for a specific combination of AOR, and instance-id and reg-id values, the registrar MUST update any information that uniquely identifies the network flow over which the request arrived if that information has changed, and SHOULD update the time the binding was last updated. "" In my installation this is not fullfilled as shown: "" AOR:: jmillan Contact:: sip:jmillan@MY_IP;transport=ws;ov-ob=a570655c14 Q= Expires:: 181 Callid:: 1hgq3khalq2rzfr Cseq:: 108 User-agent:: JsSIP 0.1.0 Path:: <sip:OUTBOUND_IP:9090;transport=tcp;lr;ovid=de0c0b9f>,<sip:a570655c14@OUTBOUND_IP :10080;transport=ws;lr;ovid=de0c0b9f;ob> State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:KAM_IP:5060 Methods:: 783 Ruid:: uloc-50119d99-328e-1 Instance:: <urn:uuid:38dce009-ae1f-4fd1-91dc-99ed9affddc1> Reg-Id:: 1 Contact:: sip:jmillan@MY_IP;transport=ws;ov-ob=1dd97b4d51 Q= Expires:: 193 Callid:: 1hgq3khalq2rzfr Cseq:: 110 User-agent:: JsSIP 0.1.0 Path:: <sip:OUTBOUND_IP:9090;transport=tcp;lr;ovid=de0c0b9f>,<sip:1dd97b4d51@OUTBOUND_IP :10080;transport=ws;lr;ovid=de0c0b9f;ob> State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:KAM_IP:5060 Methods:: 783 Ruid:: uloc-50119d99-3290-2 Instance:: <urn:uuid:38dce009-ae1f-4fd1-91dc-99ed9affddc1> Reg-Id:: 1 "" The scenario is such that when the client looses the connection with the Outbound Server, it reconnects and re-registers to Kamailio in order to replace registration and be able to receive in-dialog messages. Any comment is wellcome. Regards,

4 19

Re: [SR-Users] Regarding - Add your own module
by Alex Balashov 02 Aug '12

02 Aug '12

Does TimesTen have a UnixODBC connector? If so, you can use existing Kamailio modules. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Ananth Kollipara <infoananthk(a)gmail.com> wrote:Hi Alex, Thanks for the response. Yes, i meant developing of a custom module and integrating with ser/kamailio. I am looking for the following. 1) On receiving 302 response, proxy shall use the contact header and re-send the msg to the correct destination. 2) Integration of kamailio with Times Ten database (query Times ten DB) Please advice. Regards, Ananth On Tue, Jul 31, 2012 at 3:30 PM, <sr-users-request(a)lists.sip-router.org> wrote: Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Add Own Plug-in (Ananth Kollipara) 2. Re: Add Own Plug-in (Alex Balashov) ---------------------------------------------------------------------- Message: 1 Date: Tue, 31 Jul 2012 14:33:52 +0530 From: Ananth Kollipara <infoananthk(a)gmail.com> Subject: [SR-Users] Add Own Plug-in To: sr-users(a)lists.sip-router.org Message-ID: <CACJ9FatODfYWEG+y6_U_=-xxsxt65Ca=5FV2NFBQrYw+grS=ZQ(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi, Can you please provide the steps to add a plug-in? Regards, Ananth

2 9

planning release of v3.3.1
by Daniel-Constantin Mierla 02 Aug '12

02 Aug '12

Hello, I am planning to package v3.3.1 out of the latest branch 3.3 this Thursday, August 2. If you are aware of issues that are not listed on the tracker (http://sip-router.org/tracker/), add them there to see what can be sorted out before. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 1

Regarding - Add your own module
by Ananth Kollipara 01 Aug '12

01 Aug '12

Hi Alex, Thanks for the response. Yes, i meant developing of a custom module and integrating with ser/kamailio. I am looking for the following. 1) On receiving 302 response, proxy shall use the contact header and re-send the msg to the correct destination. 2) Integration of kamailio with Times Ten database (query Times ten DB) Please advice. Regards, Ananth On Tue, Jul 31, 2012 at 3:30 PM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Add Own Plug-in (Ananth Kollipara) > 2. Re: Add Own Plug-in (Alex Balashov) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 31 Jul 2012 14:33:52 +0530 > From: Ananth Kollipara <infoananthk(a)gmail.com> > Subject: [SR-Users] Add Own Plug-in > To: sr-users(a)lists.sip-router.org > Message-ID: > <CACJ9FatODfYWEG+y6_U_=-xxsxt65Ca=5FV2NFBQrYw+grS= > ZQ(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > Can you please provide the steps to add a plug-in? > > Regards, > Ananth >

1 0

Re: [SR-Users] Segfault running 3.3.0 with registrar/p_usrloc/path
by DanB 01 Aug '12

01 Aug '12

Hey Guys, Just a quick follow up to see if anything happened with p_usrloc fixup. Any chance to expect it running on new 3.3.1 release? Ta, DanB

3 2

cannot allocate memory & increase shared memory
by Alex Solt 01 Aug '12

01 Aug '12

Hi, We are getting "cannot allocate memory" error in the log file. I was wondering which one is the right way to increase the shared memory: 1) adding the following to /etc/openser/openserctl STARTOPTIONS="-m 128" 2) modify the following in /etc/default/openser MEMORY=1024 Thanks,AS

2 1

New Wiki Site has problems?
by Andrew Pogrebennyk 31 Jul '12

31 Jul '12

At least some links at http://www.kamailio.org/wiki/ don't work at the moment: Install Kamailio v3.3.x From GIT Upgrade Kamailio v3.1.x to v3.2.0 Upgrade Kamailio v3.2.x to v3.3.0 you are redirected to some instruction from DokuWiki Installer when trying to visit them. Could somebody please check it? Thanks. Andrew

2 1

routing based on username
by Richard Zheng 31 Jul '12

31 Jul '12

Hi, We are trying to follow http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdbto setup Kamailio as front end and multiple PBXs as back end. We need to forward sip messages to the respective servers based on the contact. For example, users 101, 102 go to PBX at 1.1.1.1 and users 201, 202 go to PBX at 1.1.1.2. Where is a good place to save this info and how to route them in the config? Also we have some users with fixed IP addresses which don't use authentication. How do we route the calls based on source IP? Since we have plenty of these users, we'd like to save them in database too. Any advice is appreciated! Richard

3 2

Add Own Plug-in
by Ananth Kollipara 31 Jul '12

31 Jul '12

Hi, Can you please provide the steps to add a plug-in? Regards, Ananth

2 1

ACK not relayed
by David Notivol 31 Jul '12

31 Jul '12

In a UAC-Kamailio-UAS scenario, we've found a case where the ACK coming from uac is not relayed by our proxy to the uas. This is the log for the ACK message: Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:624]: SIP Request: Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:626]: method: <ACK> Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:628]: uri: <sip:59138553189@79.170.68.157> <sip:59138553189@79.170.68.157> Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:630]: version: <SIP/2.0> Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_via.c:1286]: Found param type 235, <rport> = <n/a>; state=6 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bKKc4gKXyemS9HD>; state=16 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=2 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: this is the first via Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [receive.c:149]: After parse_msg... Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [receive.c:190]: preparing to run routing scripts... Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: sl [sl_funcs.c:396]: DEBUG : sl_filter_ACK: to late to be a local ACK! Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: maxfwd [mf_funcs.c:85]: value = 65 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=6ne1x6d6-CC-23 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:188]: DEBUG: get_hdr_field: <To> [52]; uri=[sip:59138553189@79.170.68.157] Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: to body [ <sip:59138553189@79.170.68.157> <sip:59138553189@79.170.68.157>] Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:168]: get_hdr_field: cseq <CSeq>: <31362871> <ACK> Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:202]: DEBUG: get_hdr_body : content_length=0 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:104]: found end of header Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=0cS8trtaF196F Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: siputils [checks.c:106]: totag found Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: ERROR: <script>: ESTAMOS EN WITHIN Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: rr [loose.c:108]: No Route headers found Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: rr [loose.c:829]: There is no Route HF Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: ERROR: <script>: NO LOOSE ROUTE However, in a pcap trace I clearly see the Route header: This is the Record-Route in the 200 message Record-Route: <sip:79.170.68.157;lr;ftag=330cdf48-co1022-INS033> And this is the Route header in the ACk: Route: <sip:79.170.68.157;lr;ftag=330cdf48-co1022-INS033> I haven't been able to figure out what's going on. We have plenty of traffic working in that proxy. What am I missing? Thanks in advance David.

2 8

BYE Race Condition
by Brandon Armstead 31 Jul '12

31 Jul '12

Hello, I am running into an issue where there is a race condition happening. I am looking for opinions / ideas on how to handle the following below scenario. Scenario. UAC places an outbound call -> upstream carrier. The call is disconnected on both ends at the exact same time, UAC -> sends BYE upstream Upstream Carrier -> sends BYE downstream Upstream 200 OK's the BYE UAC sends 481 back to Upstream Carrier for their generated BYE. The upstream carrier is complaining about receiving the relayed 481 responses -- so my first thought was simply to drop() these from relaying upstream. I am curious how other people are handling this? Would you suggest simply dropping the relay from being sent back upstream on the 481? Would you simply always 200 OK a downstream BYE from trusted carriers regardless of UAC response, and create separate transaction to send BYE downstream? Thank you as always. Look forward to your thoughts / suggestions / ideas. Sincerely, Brandon Armstead

2 2

Error 403 not relaying
by Tayeb Meftah 30 Jul '12

30 Jul '12

hello, i'm using Kamailio with a simple registrare configuration i used the apt repository to install the developmant package of kamailio domain sip.tmvoip.net have 2 records, A record and AAA record (dual stacked) i configured it in my kamailio using siremis and in the kamctlrc config below: Kamailio.cfg: http://paste.debian.net/180972/ Kamctlrc: http://paste.debian.net/180973/ siptrace: http://paste.debian.net/180974/ note that 172.28.0.0/21 and 41.221.20.12/30 is routed, no nat. my current issue is if i register or i invite, i get error 403 not relaying trying to investigate this issu but i have no clue thank you everyone, -- Voice of the blind ™ broadcast freedom Phones: Algeria: +213-660-347746 USA, NewYork: +1(347)759-5883 iNum: +883510001288000 we're blind, but not deaf!

3 3

Re: [SR-Users] Can't initialize shared memory
by Uri Shacked 30 Jul '12

30 Jul '12

> > this is the log when debug is 4... > > Jul 24 10:19:37 RNDSRV kamailio: : <core> [cfg.y:3532]: parse error in > config file /usr/local/etc/kamailio/global_params.cfg, line 121, column 54: > Can't initialize shared memory > Jul 24 10:19:37 RNDSRV kamailio: ERROR: mi_fifo [mi_fifo.c:271]: FIFO stat > failed: Bad address > > > not much. > line 121 on the file is the modparam for mi_fifo indication the > /tmp/kamailio_fifo file. > > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > any ideas? > > >

2 1

Reply to one topic correctly
by Duong Manh Truong 30 Jul '12

30 Jul '12

Hi list, I am a newbie of the maillist, I'm wondering that whenever i reply to a specific topic with the subject of "RE: [SR-Users]....", my email come to maillist in the way dislike what i wish, my email appears like a NEW topic instead of SUB topic (reply message form) I try several ways but still not get the correct method of reply, please help me to fix this, thanks.

2 1

Re: [SR-Users] Kamailio with APNS
by Duong Manh Truong 29 Jul '12

29 Jul '12

Thank Klaus, I've try with t_suspend() and t_continue() as your guide, *#Failure route to react with negative messages failure_route[1] { if (t_check_status("5[0-9][2-9]")) { if(t_suspend()) { xlog("proceesing suspended in trasaction [$T(id_index):$T(id_label)]\n"); xlog("before sleep \n"); } if(async_sleep("20")) xlog("After sleep 20s \n"); if(t_continue('1','1','MYROUTE')) xlog("Continuation successfully \n"); exit; };* And i found that t_suspend() and async_sleep() runnung ok but t_continue() seems not running. I read about TMX module and TMX module Pseudo-Variables but still can not figure out it :( Secondly, in case of t_continue() runs successfully , please tell me how to let Kamailio resend the INVITE message to client B (before the t_continue(), i think so ) - while client B is waking up ? Which functions and modules that i need to do this? Thanks. ----------------------------------------------------------------- Hi! You have to use a failure route (http://kamailio.org/docs/modules/3.3.x/modules/tm.html#t_on_failure). In the failure route check if the reply code is 404 and if yes, trigger the APNS logic and then put the call e.g. on a stack and resume e.g. every second and ofrward to the "SIP server" again to see if the client is registered meanwhile. For the asynchronous processing take a look at these functions: t_suspend(),t_resume()[1]. For example you can call t_resume in a timer route every second. For the timer route, to know which transactions are currently suspended and are waiting for further processing, you could put the transaction identifiers in a message-queue [2] and read the queue from the timer route. regards Klaus [1] http://kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2535968 [2] http://kamailio.org/docs/modules/3.3.x/modules/mqueue.html On 24.07.2012 06:02, Duong Manh Truong wrote: >* Hi list,*>**>* My team are developing a SIP client for mobile running iOS ,*>**>* then having to deal with the "back ground mode" problem while this soft*>* is installed on mobiles*>**>* One of the solutions is using APNS. (Apple Push Notification Service)*>**>* I wanna to use Kamailio as SIP Redirect / Proxy for my current SIP server*>**>* The model is like that:*>**>* User A <---------> Kamailio <---------> SIP server <--------> User B*>**>* IF UserA call UserB, UserB is not available now (not registered yet),*>* normally, SIP server will reply Kamailio with "404 Not found"*>**>* In that case, i want to set up Kamailio do the communication with APNS*>* and waiting for UserB "Wake up" then RESEND the Invite message ......*>* (instead of forwarding this 404 message to UserA)*>**>* I am getting stuck at how Kamailio can control "404 Not found" message*>* which it receives from SIP server .*>**>* I do searching around several days but still can not get any clearer*>* points.*>**>* Please help me with some instruction/ configuration of Kamailio for this*>* case !*>**>* Thank you very much.*>**>**>* _______________________________________________*>* SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list*>* sr-users at lists.sip-router.org <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>* http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users*>**

1 0

How to change _only_ next hop and dpn't touch content of message?
by intel＠intrans.baku.az 29 Jul '12

29 Jul '12

I want retransmit some me ssages to other sip server as is. f.e server sip1.com receives invite message INVITE user(a)somewhere.org i want to send it to server sip2.com as is. Unfortunately, if i use rewritehostport() function, or modify $rp $rd pseudovariables fe $rp='sip2.com', kamailio changes invite uri, and send to sip2.com INVITE user(a)sip2.com The To: filed left untouched How can I resend message without modifications? I've tried also modify $dp and $dd pseudovariables, but then message wasn't sent anywhere,and in log I can see this message: Jul 27 15:44:54 dsdb01 /usr/sbin/kamailio[17786]: ERROR: tm [t_fwd.c:1523]: ERROR: t_forward_nonack: no branches for forwarding Jul 27 15:44:54 dsdb01 /usr/sbin/kamailio[17786]: ERROR: tm [tm.c:1375]: ERROR: w_t_relay_to: t_relay_to failed What else can I try? --

2 1

BYE Race Condition
by Brandon Armstead 28 Jul '12

28 Jul '12

*** Please forgive if this is a duplicate *** Hello, I am running into an issue where there is a race condition happening. I am looking for opinions / ideas on how to handle the following below scenario. Scenario. UAC places an outbound call -> upstream carrier. The call is disconnected on both ends at the exact same time, UAC -> sends BYE upstream Upstream Carrier -> sends BYE downstream Upstream 200 OK's the BYE UAC sends 481 back to Upstream Carrier for their generated BYE. The upstream carrier is complaining about receiving the relayed 481 responses -- so my first thought was simply to drop() these from relaying upstream. I am curious how other people are handling this? Would you suggest simply dropping the relay from being sent back upstream on the 481? Would you simply always 200 OK a downstream BYE from trusted carriers regardless of UAC response, and create separate transaction to send BYE downstream? Thank you as always. Look forward to your thoughts / suggestions / ideas. Sincerely, Brandon Armstead

1 0

lcr.dump_gws problem
by Gary Chen 27 Jul '12

27 Jul '12

Kamailio 3.2.0 When testing Kamailio 3.2.0, I notice that lcr.dump_gws command does not display ip_addr correctly. Here is my output: lcr_id: 3 gw_id: 15 gw_index: 2 gw_name: gateway_1 scheme: sip ip_addr: 1273060816.0.0.0 hostname: port: 5060 params: strip: 0 prefix: tag: flags: 0 defunct_until: 0 LCR is still working correctly. It just does not display IP in the right form. Does any body also has the same problem? Gary

3 4

Re: [SR-Users] Add CallerID to SIP header
by Dominic Raferd 27 Jul '12

27 Jul '12

1 0

Add CallerID to SIP header
by Dominic Raferd 27 Jul '12

27 Jul '12

I'm not yet using Kamailio. I am looking for something to do a job which sounds really simple but may not be. Our PBX connects to a SIP service (voicetrading.com) but does not supply our CallerID, this means that voicetrading.com provide some very weird callerids for the person at the other end (international numbers etc). Which is not good. There is no way to get our PBX to add CallerID itself, so I thought maybe I could set up a local sip proxy? or sip redirect? server (Kamailio?) to do nothing other than add in the callerid details to the sip header and then forward the sip message on to voicetrading i.e. be otherwise 'invisible'? If Kamailio is not the right thing to do this with, is there something else that could do it? Thanks for any suggestions. Dominic

2 1

Cant make from_uri, request_uri work for LCR module
by Antanas Masevicius 27 Jul '12

27 Jul '12

Hello, having trouble running LCR in kamailio 3.3 (head). Here is my DB excerpt which i use to reproduce the problem: INSERT INTO `lcr_rule` VALUES (2617960,90,'118','.*',1,1,'.*'); INSERT INTO `lcr_rule_target` VALUES (7771454,90,2617960,1424,0,100); INSERT INTO `lcr_gw` VALUES (1424,90,'OP1','10.10.10.10','',5060,NULL,1,0,NULL,'11#',1,NULL,NULL); I stripped all other LCR data from my DB and only this rule with from_uri and request_uri seems to bother me. Tried this version on debian lenny and debian squeeze. On Lenny i am not able to load lcr tables at all, get a segfault. On squeeze, lcr tables loads up, but this rule gets marked as disabled. Here is config part which i use to load a rule: if (load_gws(90, $rU, $fu)) { if (next_gw()) { t_on_failure("FAIL_GW"); } else { route(NO_ROUTE); } On squeeze, if i add following code to lcr_mod.c:1249 : LM_INFO("LOADING lcr rule <%u> \n", rule_id); right after: rule_id = (unsigned int)VAL_INT(ROW_VALUES(row)); rule_id is still printed, but i get segfault like this: Jul 24 10:17:43 sp1 kernel: [23482626.105865] kamailio[15136]: segfault at a4 ip 00007fde15cb8802 sp 00007fff39d68320 error 4 in lcr.so[7fde15ca8000+23000] Not sure what i do terribly wrong. Setting from_uri and request_uri to anythig but NULL makes the code to segfault with that print line added. Without print line, kamailio starts, but never loads that rule. Any help would be appreciated. best regards, Antanas Masevicius NTT

3 5

Re: [SR-Users] Cant make from_uri, request_uri work for LCR module
by Alex Balashov 26 Jul '12

26 Jul '12

When calling load_gws(), try putting $rU and $fu in double quotes, e.g. "$rU" rather than $rU bare. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Antanas Masevicius <antanas.masevicius(a)ntt.lt> wrote:Hello, having trouble running LCR in kamailio 3.3 (head). Here is my DB excerpt which i use to reproduce the problem: INSERT INTO `lcr_rule` VALUES (2617960,90,'118','.*',1,1,'.*'); INSERT INTO `lcr_rule_target` VALUES (7771454,90,2617960,1424,0,100); INSERT INTO `lcr_gw` VALUES (1424,90,'OP1','10.10.10.10','',5060,NULL,1,0,NULL,'11#',1,NULL,NULL); I stripped all other LCR data from my DB and only this rule with from_uri and request_uri seems to bother me. Tried this version on debian lenny and debian squeeze. On Lenny i am not able to load lcr tables at all, get a segfault. On squeeze, lcr tables loads up, but this rule gets marked as disabled. Here is config part which i use to load a rule: if (load_gws(90, $rU, $fu)) { if (next_gw()) { t_on_failure("FAIL_GW"); } else { route(NO_ROUTE); } On squeeze, if i add following code to lcr_mod.c:1249 : LM_INFO("LOADING lcr rule <%u> \n", rule_id); right after: rule_id = (unsigned int)VAL_INT(ROW_VALUES(row)); rule_id is still printed, but i get segfault like this: Jul 24 10:17:43 sp1 kernel: [23482626.105865] kamailio[15136]: segfault at a4 ip 00007fde15cb8802 sp 00007fff39d68320 error 4 in lcr.so[7fde15ca8000+23000] Not sure what i do terribly wrong. Setting from_uri and request_uri to anythig but NULL makes the code to segfault with that print line added. Without print line, kamailio starts, but never loads that rule. Any help would be appreciated. best regards, Antanas Masevicius NTT _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

5 8

siremis errors
by Tony Ferrario 26 Jul '12

26 Jul '12

I just installed a Siremis 3.2.1 on a server with an already working Kamailio 3.2.1. bellow is a typical error message I keep getting. Can you point me to a solution -Thanks [http://192.168.105.196/siremis/themes/default/images/icon_forbidden.gif] System Internal Error The detailed error message are below: [2012-07-26 19:14:11 (GMT)] An exception occurred while executing this script: Error message: #0, Error in query:: SELECT T0.id, T0.method, T0.from_tag, T0.to_tag, T0.callid, T0.sip_code, T0.sip_reason, T0.time, T0.src_ip, T0.dst_user, T0.dst_domain, T0.dst_ouser, T0.src_user, T0.src_domain, T0.cdr_id FROM `acc` T0 WHERE method='INVITE' AND T0.src_user='admin' LIMIT 10. SQLSTATE[42S22]: Column not found: 1054 Unknown column 'T0.src_ip' in 'field list' Script name and line number of error: /var/www/siremis-3.2.1/openbiz/bin/data/BizDataObj_Lite.php:447 function: _run_search ( Array(2) ) @ /var/www/siremis-3.2.1/openbiz/bin/data/BizDataObj_Lite.php 243 function: fetch ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyForm.php 690 function: fetchDataSet ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/FormRenderer.php 69 function: renderSmarty ( Object(EasyForm), "/var/www/siremis-3.2.1/siremis/modules/sipuser/template/grid.tpl" ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/FormRenderer.php 45 function: render ( Object(EasyForm) ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyForm.php 1648 function: renderHTML ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyForm.php 1538 function: render ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/ViewRenderer.php 83 function: renderSmarty ( Object(EasyView), "/var/www/siremis-3.2.1/siremis/modules/sipuser/template/view.tpl" ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/ViewRenderer.php 40 function: render ( Object(EasyView) ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyView.php 274 function: _render ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyView.php 237 function: render ( ) @ /var/www/siremis-3.2.1/openbiz/bin/BizController.php 221 function: renderView ( "sipuser.view.AccListView", "", "", Null, "" ) @ /var/www/siremis-3.2.1/openbiz/bin/BizController.php 107 function: dispatchRequest ( ) @ /var/www/siremis-3.2.1/openbiz/bin/BizController.php 32 function: include_once ( "/var/www/siremis-3.2.1/openbiz/bin/BizController.php" ) @ /var/www/siremis-3.2.1/siremis/bin/controller.php 6 function: include ( "/var/www/siremis-3.2.1/siremis/bin/controller.php" ) @ /var/www/siremis-3.2.1/siremis/bin/_forward.php 102 function: include ( "/var/www/siremis-3.2.1/siremis/bin/_forward.php" ) @ /var/www/siremis-3.2.1/siremis/index.php 3 ________________________________ Please ask system administrator for help... Tony Ferrario Manager SVT & Ops OnMobile Global Limited 3950 Civic Center Drive , Suite 200 San Rafael, Ca 94903 Phone: 415-223-8206 Cell: 707-508-7285 tony.ferrario(a)onmobile.com ________________________________ DISCLAIMER: The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. Further, this e-mail may contain viruses and all reasonable precaution to minimize the risk arising there from is taken by OnMobile. OnMobile is not liable for any damage sustained by you as a result of any virus in this e-mail. All applicable virus checks should be carried out by you before opening this e-mail or any attachment thereto. Thank you - OnMobile Global Limited.

1 0

Overload Control
by Kaplan, Michael A 26 Jul '12

26 Jul '12

Folks, I'm trying to figure out if either Kamailio or OpenSIPs provides the various overload control algorithms recommended in the IETF or elsewhere. I haven't found documentation to this effect. Has anyone looked into this or used such techniques? Thanks, Mike

3 6

rtimer execution period - round two
by Alex Balashov 25 Jul '12

25 Jul '12

Hi Daniel, I wanted to go back to this thread for a minute: http://lists.sip-router.org/pipermail/sr-users/2012-June/073663.html Sorry that I didn't reply to the thread; I no longer have the last message. In your reply, you said: > there is no timeout on the execution of that task. Of course, you > can start many rtimer processes to do same task, up to you, > via configuration parameters. A matter of how rtimer executed > route is built, a rtimer process may not rest (e.g., sleep) and > keep consuming mqueue items as long as the queue is not empty." However, I am seeing some behaviour that is rather different. Roughly, I have a single rtimer task that runs every 3 seconds and consumes an mqueue full of accounting events. The rtimer request route looks like something this: route[ACCT_WRITER] { while(mq_fetch("acct_write")) { # Some stuff to write $mqv(...) values to PostgreSQL. ... sql_query("ca", "SELECT * FROM acct_event_proc(...)"); ... mq_pv_free("account_write"); } } And the rtimer task is parameterised thusly: modparam("rtimer", "timer", "name=acct_write;interval=3;mode=1") modparam("rtimer", "exec", "timer=acct_write;route=ACCT_WRITER") And the mqueue: modparam("mqueue", "mqueue", "name=acct_write") I have a stored procedure - let's call it acct_event_proc() - that is invoked from within the rtimer task. When I looked at the live query activity from Kamailio to Postgres by doing a packet capture on loopback, like this: tcpdump -i any -A -s 0 -tttt -n 'tcp port 5432' | grep acct_event_proc What I saw was some events being written for about ~3 seconds and then the process would stop. Then, after about half a second, it would burst again for 3 seconds, then stop again. And so on and so forth. When I lowered the interval to 1 second, it got faster and more responsive, and would push more events through in one interval. But still, it appeared to be a limited "burst" of SELECT calls punctuated by 1 second of silence. When I checked out branch 3.3 and lowered the interval to 50000u, it got even faster, almost real-time enough to keep up. Still, I did not really know what was happening, so I added an MI command to HEAD to check mqueue size live: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commit;h=41f771… (I think it would be quite useful anyhow, as there is no runtime mqueue info available right now.) Using it, I confirmed my impression: the bigger the rtimer interval, the larger the queue grows, and the slower the events seem to be written overall. The shorter the rtimer interval, the faster they are dequeued, and the smaller the mqueue is overall, in the course of normal call processing. Thus, my impression is that the execution time of the rtimer task is bounded by the interval, and that the core kills the forked rtimer task currently running prior to starting another one. Can you shed some light? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

3 12

Cleanup location table on failed NAT ping
by David 25 Jul '12

25 Jul '12

Hello, I am using kamailio 3.2.3. I am using the nathelper module to "ping" users in the location table to keep the NAT bindings alive. In the event of 3 failed NAT pings, I would like to mark the entry as "unavailable" or completely delete it from the location table to avoid having stale data. Is this a feature that exists ? Is not, are their plans for it in the future ? When the REGISTER expires after 3600, if the phone doesn't register, the location entry gets cleaned up. So I have to wait up to one hour before the bad entry is purged. I was hoping to have it purged sooner. Thanks, David version: kamailio 3.2.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, 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, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 14:19:23 Apr 19 2012 with gcc 4.4.3

3 2

Error Loading db_postgres.so on Startup
by Akan 25 Jul '12

25 Jul '12

I am getting this error when trying to run kamailio on Centos 5.5. I have Postgresql 9.1.4 installed and I have tried with both OpenSSL v0.9.8e and v1.0.1c and I still get the same error. Any help would be appreciated. ERROR: <core> [sr_module.c:565]: ERROR: load_module: could not open module </usr/local/kamailio-3.2/lib64/kamailio/modules/db_postgres.so>: /usr/local/kamailio-3.2/lib64/kamailio/modules/db_postgres.so: undefined symbol: TLSv1_method Thanks Nathaniel

2 4

Kamailio with APNS
by Duong Manh Truong 24 Jul '12

24 Jul '12

Hi list, My team are developing a SIP client for mobile running iOS , then having to deal with the "back ground mode" problem while this soft is installed on mobiles One of the solutions is using APNS. (Apple Push Notification Service) I wanna to use Kamailio as SIP Redirect / Proxy for my current SIP server The model is like that: User A <---------> Kamailio <---------> SIP server <--------> User B IF UserA call UserB, UserB is not available now (not registered yet), normally, SIP server will reply Kamailio with "404 Not found" In that case, i want to set up Kamailio do the communication with APNS and waiting for UserB "Wake up" then RESEND the Invite message ...... (instead of forwarding this 404 message to UserA) I am getting stuck at how Kamailio can control "404 Not found" message which it receives from SIP server . I do searching around several days but still can not get any clearer points. Please help me with some instruction/ configuration of Kamailio for this case ! Thank you very much.

3 3

Kamailio with APNS
by Duong Manh Truong 24 Jul '12

24 Jul '12

Hi list, My team are developing a SIP client for mobile running iOS , then having to deal with the "back ground mode" problem while this soft is installed on mobiles One of the solutions is using APNS. (Apple Push Notification Service) I wanna to use Kamailio as SIP Redirect / Proxy for my current SIP server The model is like that: User A <---------> Kamailio <---------> SIP server <--------> User B IF UserA call UserB, UserB is not available now (not registered yet), normally, SIP server will reply Kamailio with "404 Not found" In that case, i want to set up Kamailio do the communication with APNS and waiting for UserB "Wake up" then RESEND the Invite message ...... (instead of forwarding this 404 message to UserA) I am getting stuck at how Kamailio can control "404 Not found" message which it receives from SIP server . I do searching around several days but still can not get any clearer points. Please help me with some instruction/ configuration of Kamailio for this case ! Thank you very much.

2 1

Re: [SR-Users] Kamailio with APNS
by Nguyen Thanh Tien 24 Jul '12

24 Jul '12

I think you can find the 404 message at the onreply_route[]. I tried and found the 200 OK here.

1 0

SIREMIS
by Adam Fyfe 24 Jul '12

24 Jul '12

Hi, I am testing Siremis, and have become stuck. How does one add an outbound SIP trunk for calls ? thanks

2 1

Kamailio setup problem
by Vijay Thakur 23 Jul '12

23 Jul '12

All Experts, I have installed Kamailio Server on Linode Cloud. I have followed the following URL for Setup: http://kb.asipto.com/kamailio:presence:k31-made-simple This is done in ubuntu 10.04. I am only able to make chat. Video chat, conference, Presence, Call, Screen Sharing is not working. But followed the same how to install kamailio in my local system and all features are working fine. Does there anyone who can point out the flaw in setup. Thanks and regards, Vijay Thakur

2 1

Can't initialize shared memory
by Uri Shacked 23 Jul '12

23 Jul '12

Hi, Anyone got this message " Can't initialize shared memory" ? When i run kamailio using "kamailio -m 512" it works OK When i run "/etc/init.d/kamailio start" i get the message above. my init script anf defaults are set with memory = 512. I played a little with 3.3.0 and now i am back to 3.2.4 can it be something to do with it? BR, Uri

2 1

Re: [SR-Users] How to rewritehostport() dynamically?
by Alex Balashov 20 Jul '12

20 Jul '12

Try modifying the request URI with $ru/$rU/$rd/$rp instead. See the pseudovariables page on the dokuwiki for more information. rewritehostport() is a legacy function that does not deal with variables. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/intel@intrans.baku.az wrote:Can I use rewritehostport() function with argument selected at time of call processing? I have a lua script, which gets address/port of next hop(from base) But how to call rewritehostport() with it? I can't find, how to call this function from lua script directly (I've tried sr.tm.rewritehostport() and rewritehostport() - no success) If i set pseudovariable (f.e $avp(relay)) and call rewritehostport("$avp(relay)") from kamailio.cfg later, i can see in logs such message: ERROR: tm [ut.h:327]: failed to resolve "$avp(relay)" Seems it use name of pv, not value? What should I do to call this function with correct parameter? -- _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

How to rewritehostport() dynamically?
by intel＠intrans.baku.az 20 Jul '12

20 Jul '12

Can I use rewritehostport() function with argument selected at time of call processing? I have a lua script, which gets address/port of next hop(from base) But how to call rewritehostport() with it? I can't find, how to call this function from lua script directly (I've tried sr.tm.rewritehostport() and rewritehostport() - no success) If i set pseudovariable (f.e $avp(relay)) and call rewritehostport("$avp(relay)") from kamailio.cfg later, i can see in logs such message: ERROR: tm [ut.h:327]: failed to resolve "$avp(relay)" Seems it use name of pv, not value? What should I do to call this function with correct parameter? --

1 0

Kamailio and IMS
by Apoorv Khare 20 Jul '12

20 Jul '12

Hi Everyone, I am trying to integrate IMS(*carstenbock/ims*) patch in Kamailio. But so far I could not find any proper documentation to load configuration files - pcscf, icscf, scscf, mgcf etc? Please let me know if there is anything available! Only relevant link I found - http://www.kamailio.org/w/2011/01/ims-extensions-available-for-testing/ AND this link does not seems to be update - http://www.ng-voice.com/our-solution/installation/ BR, Apoorv

3 5

uac module From header rewriting problems
by Timo Teras 20 Jul '12

20 Jul '12

Hi, I fixed some uac From rewriting issues in commit e1d1c774c9ac0b4d, however even with the latest versions I'm still seeing corruption in From headers when they are rewritten for the whole dialogue (from_restore_mode=auto, the default). I spent some time analyzing the problem and the problem seems to be connected with the fact that only the bitwise difference of the original and modified URI are stored (XOR). The only purpose for this appears to be the idea to save only the difference in the rr params instead of two full URIs. However, it appears that in many cases (especially call transfer) the From header will be altered during dialogue. This is explicitly allowed in RFC 3261 § 20.20: The From header field indicates the initiator of the request. This may be different from the initiator of the dialog. I've seen this in practise with few commercial SIP stacks. Sometimes the changes are minor, or even trivial - but with the difference encoding having even one character changed (e.g. case), being deleted or added will result in the whole header being corrupted. I'm now wondering if we should just store both URIs in the rr params. While taking little more space, it should fix us sending garbage to the wire (which usually results in dropped call as the remote will reject the garbage messages). Any better ideas? Cheers, Timo

3 7

Kamailio as SBC and far end nat
by Martin Schiøtz 19 Jul '12

19 Jul '12

Hi I'm trying to configure kamailio version 3.3.0 to be a Session Border Controller which only purpose is to handle far-end-nat. On the far end site I have a SIP client with IP '192.168.1.2' which is NAT'ed to IP 'zz.zz.85.230'. The SIP ALG is disabled on the NAT firewall. The SIP client is communicating with the kamailio server and the Kamailio server is forwarding every thing to an Avaya SIP server 'xx.xx.173.15' and vice versa. The kamilio is konfigured to do this stateless. The REGISTER and SUBSCRIBE part works fine. The kamailio is using the 'Via:' field and I also needed to do subst on the 'Contact:' field in order for our SIP server to communicate back to the kamailio. But when I get to the NOTIFY packet from the SIP server to the kamailio there is no 'Via:' field or any other information about the SIP clients IP/port and the Kamailio just sends back 'SIP/2.0 404 Not here.' to the SIP server. I guess that this can not be solved with a statless kamailio? I guess that I need some kind of state table in the kamailio - is that possible without doing REGISTER directly with the kamailio? Any other good ideas are welcome? Cheers, Martin Tell me if need more info the issue? ----- # T xx.xx.173.15:38059 -> yy.yy.159.211:5060 [AP] NOTIFY sip:2001012@yy.yy.159.211:5060;transport=TCP SIP/2.0. Call-ID: NjQ5ZDNlYzRjOWFjMzMzN2JlYzAwMTg4ODYwZTJlYWY.. From: "2001012" <sip:2001012@bsosip.domain.dk>;tag=-667195155*1*016asm-callprocessing.sar229435981~1342570131099~1170998434~1. To: "2001012" <sip:2001012@bsosip.domain.dk>;tag=7ca0af7d. CSeq: 3 NOTIFY. Contact: <sip:xx.xx.173.14:15060>. Via: SIP/2.0/TCP xx.xx.173.15;branch=z9hG4bK4DEAAD0E478B838008189457-AP;ft=xx.xx.173.15~13c4. Via: SIP/2.0/UDP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838008189457. Via: SIP/2.0/TCP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838018189455. Event: message-summary. Content-Type: application/simple-message-summary. Content-Length: 69. Subscription-State: active;expires=299. Max-Forwards: 68. User-Agent: AVAYA-SM-6.1.4.0.614005. . Messages-Waiting: no. Message-Account: sip:2001012@bsosip.domain.dk. ## T yy.yy.159.211:5060 -> xx.xx.173.15:38059 [AP] SIP/2.0 404 Not here. Call-ID: NjQ5ZDNlYzRjOWFjMzMzN2JlYzAwMTg4ODYwZTJlYWY.. From: "2001012" <sip:2001012@bsosip.domain.dk>;tag=-667195155*1*016asm-callprocessing.sar229435981~1342570131099~1170998434~1. To: "2001012" <sip:2001012@bsosip.domain.dk>;tag=7ca0af7d. CSeq: 3 NOTIFY. Via: SIP/2.0/TCP xx.xx.173.15;branch=z9hG4bK4DEAAD0E478B838008189457-AP;ft=xx.xx.173.15~13c4;rport=38059. Via: SIP/2.0/UDP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838008189457. Via: SIP/2.0/TCP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838018189455. Server: kamailio (3.3.0 (i386/linux)). Content-Length: 0. . # T xx.xx.173.15:38059 -> yy.yy.159.211:5060 [A] ...... # T yy.yy.159.211:5060 -> xx.xx.173.15:38059 [AP] SIP/2.0 404 Not here. Call-ID: NjQ5ZDNlYzRjOWFjMzMzN2JlYzAwMTg4ODYwZTJlYWY.. From: "2001012" <sip:2001012@bsosip.domain.dk>;tag=-667195155*1*016asm-callprocessing.sar229435981~1342570131099~1170998434~1. To: "2001012" <sip:2001012@bsosip.domain.dk>;tag=7ca0af7d. CSeq: 3 NOTIFY. Via: SIP/2.0/TCP xx.xx.173.15;rport=38059;branch=z9hG4bK4DEAAD0E478B838008189457-AP;ft=xx.xx.173.15~13c4. Via: SIP/2.0/UDP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838008189457. Via: SIP/2.0/TCP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838018189455. Server: kamailio (3.3.0 (i386/linux)). Content-Length: 0. -----

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Compile Kamailio from Source with STUN
by Vijay Thakur 19 Jul '12

19 Jul '12

Dear All Experts, I am following the steps given at http://kb.asipto.com/kamailio:presence:k31-made-simple. I want to compile kamailio with stun support. So guide me that what will be the command to do it. As per the given command : *make PREFIX=/usr/local/kamailio-3.1-xcap FLAVOUR=kamailio \ include_modules="db_mysql presence presence_xml xhttp xcap_server" cfg what text should i use to make stun module. * -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail : vijay.thakur(a)loopmethods.com Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web: www.loopmethods.com LOOP Disclaimer ------------------------------------------------------------------------------------------------- This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

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Kamailio v3.2.4 Released
by Daniel-Constantin Mierla 19 Jul '12

19 Jul '12

Hello, Kamailio v3.2.4 is out -- this is a minor release packaged from previous stable branch, GIT 3.2. Anyone running 3.2.x should update to it in order to benefit of latest fixes, there is no change required to be done in database structure or configuration file, just re-install over the old deployment. As usual, a backup is recommended to be in the safe side. More details about this release at: * http://www.kamailio.org/w/2012/07/kamailio-v3-2-4-released/ New installations should start with latest stable series 3.3.x, at this time latest Kamailio version being v3.3.0. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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Re: [SR-Users] Presence Subscription / Subscription-State: pending
by Daniel-Constantin Mierla 19 Jul '12

19 Jul '12

Hello, please keep the mailing list cc-ed, private emails from mailing list discussions will be ignored. On 7/19/12 8:28 AM, Durmuş Ali Öztürk wrote: > Hi Daniel, > > Thank you for your advise. I have found the problem, I thought it is > enough to subscribe with my own ID but fact is I have to subscribe to > the buddies of my contact list.This was not clear for me at the beginning. ok, great you found it. > > It is better to subscribe only to the presence.wmi? Or should I > susbcribe to presence and presence.wmi? Do you mean presence mwi? mwi is for message waiting indication, like voicemail, so you get notifications when someone leaves you a message on the media server. Cheers, Daniel > > Best Regards, > Ali > > 2012/7/18 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > when you updated the presence rules via xcap, did you executed the > function to update the notifiers as well? I mean the two function at: > > http://kamailio.org/docs/modules/stable/modules_k/presence.html#id2496049 > > Cheers, > Daniel > > > > On 7/18/12 9:29 AM, Durmuş Ali Öztürk wrote: >> Hi All, >> >> I am trying to subscribe to the presence event but unfortunaly I >> am getting following Notify with the "Subscription-State: >> pending" status. >> >> NOTIFY sip:5000@192.168.127.138:55583 >> <http://sip:5000@192.168.127.138:55583> SIP/2.0 >> Via: SIP/2.0/UDP 192.168.127.230;branch=z9hG4bK6a8e.19cd7bb7.0 >> To: sip:5000@192.168.127.230 >> <mailto:sip%3A5000@192.168.127.230>;tag=2786be25cb234561b8dea2e067fd769c-5000 >> From: sip:5000@192.168.127.230 >> <mailto:sip%3A5000@192.168.127.230>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6a8e >> CSeq: 2 NOTIFY >> Call-ID: e4ec6031-99e1-de11-8845-5000@vvt-laptop >> Content-Length: 0 >> User-Agent: kamailio (3.2.3 (i386/linux)) >> Max-Forwards: 70 >> Event: presence >> Contact: <sip:192.168.127.230:5060;transport=udp> >> Subscription-State: pending;expires=3670 >> >> >> For the implemantation I followed the guide from >> http://kb.asipto.com/kamailio:presence:k31-made-simple >> >> The xcap table has already the needed resource-lists >> and pres-rules record. >> >> Is there another guide exists for the presence module which can >> be helpfully for me? >> >> Thank you in advance >> ali >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu > Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw > > -- Daniel-Constantin Mierla -http://www.asipto.com http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw

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planning release of v3.2.4
by Daniel-Constantin Mierla 19 Jul '12

19 Jul '12

Hello, later this week (like Thursday) I am considering packaging v3.2.4 from the latest git branch 3.2. If anyone has patches to push it there, please do it asap in case you want them in this release. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 1

Presence Subscription / Subscription-State: pending
by Durmuş Ali Öztürk 18 Jul '12

18 Jul '12

Hi All, I am trying to subscribe to the presence event but unfortunaly I am getting following Notify with the "Subscription-State: pending" status. NOTIFY sip:5000@192.168.127.138:55583 SIP/2.0 Via: SIP/2.0/UDP 192.168.127.230;branch=z9hG4bK6a8e.19cd7bb7.0 To: sip:5000@192.168.127.230;tag=2786be25cb234561b8dea2e067fd769c-5000 From: sip:5000@192.168.127.230;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6a8e CSeq: 2 NOTIFY Call-ID: e4ec6031-99e1-de11-8845-5000@vvt-laptop Content-Length: 0 User-Agent: kamailio (3.2.3 (i386/linux)) Max-Forwards: 70 Event: presence Contact: <sip:192.168.127.230:5060;transport=udp> Subscription-State: pending;expires=3670 For the implemantation I followed the guide from http://kb.asipto.com/kamailio:presence:k31-made-simple The xcap table has already the needed resource-lists and pres-rules record. Is there another guide exists for the presence module which can be helpfully for me? Thank you in advance ali

2 1

Reply 503 was replaced to 500
by Uri Shacked 18 Jul '12

18 Jul '12

Hi, Kamailio server is behind our company's softswitch and acts as a sip application server. I notice that there are calls that the softswitch replied with 503 "service unavailable" and kamailio sent to the originator leg 500 "service unavaileable". When kamailio recieved 504 or 502 it sends them back as is. shouldn't it be the same with 503? It also does not have a "to tag" in the CDR. And the "to tag" in the 503 that was recieved is not equal to the 500 reply "to tag" kamailio sent back. any ideas? BR, Uri

2 6

Topoh address
by Alex Balashov 17 Jul '12

17 Jul '12

A whole ago, it was mentioned on the list that there was some bug or issue in Topoh that precluded the use of the actual host IP of the proxy as the Topoh spoof address. Is that still true in 3.3? Or has that been fixed some time ago? Thanks! -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/

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How to use regular expression for $var ?
by Gary Chen 17 Jul '12

17 Jul '12

Kamailio 3.3.0 I have a variable $var(s:dst). It can store either a number or IP. How do I check to determine whether it is a number of IP? I tried the following and it did not work: If ($var(s:dst) =~ "^\d+\.\d+\.\d+\.\d+$"){ It is a IP. }else{ It is a Number } Thanks Gary

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Does kamailio has some headers optimizing mechanism ?
by Konstantin M. 17 Jul '12

17 Jul '12

Hello, I have a following part of code: if (route(Test_TO_Dialer)) { append_hf("P-hint: Loose Routing. Re-INVITE Detected: Uplink --> Originator.\r\n"); xlog("********** loose_route(): Detected Re-INVITE from Uplink! **********\n"); setflag(1); # do accounting append_hf("P-hint: SDP was mangled\r\n"); python_exec("ParseSDP", $mb); fix_nated_contact(); setbflag(6); # Mark as NATed rtpproxy_manage(); dlg_manage(); } But in a logs I see: P-hint: Loose Routing. Re-INVITE Detected: Uplink --> Originator., SDP was mangled but it should be like this: P-hint: Loose Routing. Re-INVITE Detected: Uplink --> Originator. P-hint: SDP was mangled Does kamailio has some headers optimizing mechanism ? # ./kamailio -V version: kamailio 3.2.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, 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, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 15:37:11 Jun 9 2012 with gcc 4.4.3

2 1

How to use Kamailio
by piyush singhai 16 Jul '12

16 Jul '12

Hello I am very new in opensip. i am trying to install kamailio and freeswitch...kamailio is running fine...but i donot know how kamailio will communicate to my local sip server. kamctl start and stop ...starting and stopping the kamailio. i have installed kamalio from http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git --Piyush

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Interested Sip Server
by Tony Rosa 15 Jul '12

15 Jul '12

I am interested sip server software for residential application Tony Andrade IP Solution 617-386-9186

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Re: [SR-Users] : Adding "make menuconfig"
by Anton Kvashenkin 15 Jul '12

15 Jul '12

Or we can just "port" Opensips's "make menuconfig" :) 15.07.2012 2:05 пользователь "aft" <aftnix(a)gmail.com> написал: > Hi, > > Kamailio is a modular stack. So inclusion or exclusion of a module is > very important. At this moment editing modules.lst is very *primitive* > way of doing things. > > I like menuconfig style ncurses based interface. Which gives a very > clear view of "dependency" and "which and which" information about the > modules. > > As number of modules increases, most of us just compile most of them, or > take the default modules.lst as granted. > > If a menuconfig target is added, it should be easier to customize a > kamailio build. > > Adding a menuconfig target seems a trivial task for anyone who know > kamailio build system well plus some intro to ncurses programming. > > So i guess devs should look into it. > > cheers > aft > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

1 0

Adding "make menuconfig"
by aft 14 Jul '12

14 Jul '12

Hi, Kamailio is a modular stack. So inclusion or exclusion of a module is very important. At this moment editing modules.lst is very *primitive* way of doing things. I like menuconfig style ncurses based interface. Which gives a very clear view of "dependency" and "which and which" information about the modules. As number of modules increases, most of us just compile most of them, or take the default modules.lst as granted. If a menuconfig target is added, it should be easier to customize a kamailio build. Adding a menuconfig target seems a trivial task for anyone who know kamailio build system well plus some intro to ncurses programming. So i guess devs should look into it. cheers aft

2 1

Kamailio v3.1.6 released - the last packaging in 3.1.x series
by Daniel-Constantin Mierla 14 Jul '12

14 Jul '12

Hello, this morning I packaged the 3.1.6 to mark the last release in the 3.1.x series. At this moment, the latest stable series are 3.2.x and 3.3.x, so there is no intention of packing again a 3.1.x version -- although patches may still go in branch 3.1. Notes about v3.1.6 release were posted on the news section of the website: * http://www.kamailio.org/w/2012/07/kamailio-v3-1-6-released/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

Kamailio + dialog + dispatcher (limiting calls)
by George Breahna 13 Jul '12

13 Jul '12

I am trying to limit the total number of concurrent calls processed by Kamailio and I'm having a world of trouble. We are running Kamailio 3.2.3 on Debian Squeeze with the dispatcher module and recently enabled the dialog module. The call flow is extremely simple. We have two Asterisk servers to which we load balance the incoming calls to, nothing more. Our dialog configuration is the default kamailio config, with the following additions: modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_flag", FLD_DLG) modparam("dialog", "hash_size", 4096) modparam("dialog", "profiles_with_value", "callquota") modparam("dialog", "default_timeout", 300) modparam("dialog", "dlg_match_mode",2) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_mode", 0) in request_route we have: if ( method=="INVITE" ) { t_on_failure("FAIL_ONE"); ds_select_dst("1","4"); # sl_send_reply("100","Trying"); t_reply("100", "Trying"); dlg_manage(); setflag(FLT_ACC); # do accounting $var(SIZE) = 0; get_profile_size("callquota", "$rU", "$var(SIZE)"); if( $var(SIZE) >= MAX_NUMBER_OF_CALLS ){ xlog("DEBUG: Simultaneous calls limit reached for $rU: $var(SIZE)\n"); sl_send_reply("503", "Simultaneous calls limit reached"); exit; } set_dlg_profile("callquota","$rU"); if (get_profile_size("callquota","$var(SIZE)")) { xlog("DEBUG: there are $var(SIZE) total calls for $rU\n"); } forward();#uri:host, uri:port); exit(); } We also added a dlg_manage() in WITHINDLG if (is_method("BYE")) { dlg_manage(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } The problem is that some dialogs never get erased. Their total number always increases and their state is always 1 kamctl fifo profile_list_dlgs callquota dialog:: hash=2161:28334339 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 In the debug logs I see these entries for the dialog module: ### Starting Call Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=515 a=25 n=dlg_manage Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:523]: no dialog callid='1560428909_26358995(a)4.55.13.227' found Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:523]: no dialog callid='1560428909_26358995(a)4.55.13.227' found Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:553]: no dialog callid='1560428909_26358995(a)4.55.13.227' found Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:309]: new dialog on hash 2161 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_handlers.c:271]: route_set <sip:1.2.3.4;lr=on>, contact sip:+118884936245@4.55.13.227:5060, cseq 402 and bind_addr udp:10.10.10.1:5060 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:583]: ref dlg 0x7feb80ef0ae8 with 1 -> 1 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:603]: ref dlg 0x7feb80ef0ae8 with 1 -> 2 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:621]: unref dlg 0x7feb80ef0ae8 with 1 -> 1 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=525 a=27 n=set_dlg_profile Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:459]: ref dlg 0x7feb80ef0ae8 with 1 -> 2 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:461]: dialog id=28334339 found on entry 2161 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:621]: unref dlg 0x7feb80ef0ae8 with 1 -> 1 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:514]: ref dlg 0x7feb80ef0ae8 with 1 -> 2 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:517]: dialog callid='1560428909_26358995(a)4.55.13.227' found#012 on entry 2161, dir=1 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: CRITICAL: dialog [dlg_hash.c:650]: bogus event 6 in state 1 for dlg 0x7feb80ef0ae8 [2161:28334339] with clid '1560428909_26358995(a)4.55.13.227' and tags 'gK0251b91f' '' Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:808]: dialog 0x7feb80ef0ae8 changed from state 1 to state 1, due event 6 (ref 2) Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:621]: unref dlg 0x7feb80ef0ae8 with 1 -> 1 ### Hanging up now Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25456]: DEBUG: tm [uac.c:182]: DEBUG: dlg2hash: 29877 Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25456]: DEBUG: tm [uac.c:182]: DEBUG: dlg2hash: 29874 Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: DEBUG: dialog [dlg_hash.c:514]: ref dlg 0x7feb80ef0ae8 with 1 -> 2 Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: DEBUG: dialog [dlg_hash.c:517]: dialog callid='1560428909_26358995(a)4.55.13.227' found#012 on entry 2161, dir=1 Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: CRITICAL: dialog [dlg_hash.c:650]: bogus event 7 in state 1 for dlg 0x7feb80ef0ae8 [2161:28334339] with clid '1560428909_26358995(a)4.55.13.227' and tags 'gK0251b91f' '' Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: DEBUG: dialog [dlg_hash.c:808]: dialog 0x7feb80ef0ae8 changed from state 1 to state 1, due event 7 (ref 2) Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: DEBUG: dialog [dlg_hash.c:621]: unref dlg 0x7feb80ef0ae8 with 1 -> 1 Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=625 a=25 n=dlg_manage Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: DEBUG: dialog [dlg_hash.c:459]: ref dlg 0x7feb80ef0ae8 with 1 -> 2 Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: DEBUG: dialog [dlg_hash.c:461]: dialog id=28334339 found on entry 2161 Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: DEBUG: dialog [dlg_hash.c:621]: unref dlg 0x7feb80ef0ae8 with 1 -> 1 Is my config wrong ? Am I missing anything here ? Any advice would be much appreciated Thank you, George

2 2

what exactly is "shmeme:fragments" ?
by Uri Shacked 13 Jul '12

13 Jul '12

Hi, can someone please explain a little about the "shmem" params that i see when doing "kamctl fifo get_statistics shmeme:" BR, Uri

2 1

failure route correct..?
by Gertjan Wolzak 13 Jul '12

13 Jul '12

Hello All, I am trying to get a failure route to work, I have got it working partially. When a call comes in, first I check the db_alias, if that is positive I do a lookup location and relay if the location is valid. But sometimes the sip client is still registered in the location table, but not connected anymore, mainly with wifi connected clients. So, I have got the time out on 3 seconds (fr_timer). When that hits I have configured the following failure route: failure_route[NOTONLINE] { xlog("SCRIPT: Notonline failure route\n"); t_on_failure("STOP"); if (t_is_canceled()) { exit; } if (t_check_status("408")) { xlog("SCRIPT: Status is time out"); $rU = $avp(orig_called); /( called number and alias id not equal, so have to revert the rU back to the called number)/ prefix("9993"); /( needed to get the right manipulation done within asterisk)/ xlog("SCRIPT: uri is $ru"); $ru = "sip:" + $rU + "@w.x.y.x:5060"; xlog("SCRIPT: uri is $ru"); (w.x.y.z ip address of the asterisk box) append_branch(); t_relay_to_udp("w.x.y.x","5060"); break; } } I am not sure if the above is correct. I have based this on an old "voicemail" failure route I could find. It is working correct, the call is forwarded to an Asterisk box, where some manipulation is done and then send to an pstn gateway. The only problem I have is one way audio. RTP from the called number reaches the callee but not vice versa. Now I am wondering, can that be caused by the failure route, or should I be looking in another direction? Hope someone can give me a pointer. Thanks. Gertjan Wolzak

2 1

preparing release of v3.1.6
by Daniel-Constantin Mierla 13 Jul '12

13 Jul '12

Hello, later today I plan to package v3.1.6 as the last release in 3.1.x series. Based on our release policy to work on latest two stable branches, 3.1 got out of by release of 3.3.0. The branch 3.1 may still get patches, it is not a closed branch, but it is a matter of developers for willing to do that -- it is just not going to be packaged again in a numbered release, so people will have to use git branch. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

the 200 ok reply not include supported header
by mike 13 Jul '12

13 Jul '12

Hi all: i have the server loaded the path model , but the 200 ok didn't include the "Supported" header filed . REGISTER sip:example.com SIP/2.0 Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bKnashds7 Max-Forwards: 70 From: Bob <sip:bob@example.com>;tag=7F94778B653B To: Bob <sip:bob@example.com> Call-ID: 16CB75F21C70 CSeq: 1 REGISTER Supported: path, outbound Route: <sip:ep1.example.com;lr> Contact: <sip:bob@192.0.2.2;transport=tcp>;reg-id=1 ;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>" Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.0.2.15;branch=z9hG4bKnuiqisi Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bKnashds7 From: Bob <sip:bob@example.com>;tag=7F94778B653B To: Bob <sip:bob@example.com>;tag=6AF99445E44A Call-ID: 16CB75F21C70 CSeq: 1 REGISTER Supported: path, outbound (---------this line is not include-------------) Require: outbound Contact: <sip:bob@192.0.2.2;transport=tcp>;reg-id=1;expires=3600 ;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>" Path: <sip:VskztcQ/ how could i fix it ? thanks mike

2 1

Re: [SR-Users] sip over tls is not working
by Peter Dunkley 12 Jul '12

12 Jul '12

Hi, I have just built Kamailio from Git master, configured TCP and TLS, and an HTTP event route. I was able to connect to Kamailio over TLS using Google Chrome (and saw the "SSL Error: The site's security certificate is not trusted!" page). After I clicked on "Proceed anyway" I saw the log message I put in the HTTP event route come out. I also logged in with Jitsi using TLS and that worked fine too. I tried this with both the WebSocket module loaded and without it, and it worked in both cases. I also tried the above with TCP and that worked fine as well. As far as I can tell TLS is working in Kamailio Git master. Regards, Peter > On Wed, Jul 11, 2012 at 9:37 PM, Peter Dunkley > <peter.dunkley(a)crocodile-rcs.com> wrote: >> >> Hi, >> >> WebSockets over TLS works which requires establishing a TLS connection >> and >> exchanging an HTTP request and response. It doesn't sound like this >> connection is even getting passed the TLS handshake part? >> >> Peter >> > > Hi, > That was my first guess. I will run some tests with plain tcp socket > and post update. > > cheers. > >> >> On Wed, 2012-07-11 at 17:14 +0200, Klaus Darilion wrote: >> >> Maybe there were some changes fore websocket support which cause >> problems. Do plain TCP connections work? >> >> klaus >> >> On 11.07.2012 16:20, Aft nix wrote: >> > On Wed, Jul 11, 2012 at 6:56 PM, Klaus Darilion >> > <klaus.mailinglists(a)pernau.at> wrote: >> >> I just tested TLS with Kamailio 3.3.0 and Eyebeam and it works. Make >> >> sure to >> >> specify "ca_list" if intermediate certificates are used. >> >> >> > >> > I was working with master branch, not 3.3 branch. >> > >> >> >> >> regards >> >> Klaus >> >> >> >> On 09.07.2012 13:27, Aft nix wrote: >> >>> >> >>> Hi, >> >>> >> >>> I have enabled tls parameters as follows: >> >>> >> >>> in kamailio.cfg >> >>> >> >>> listen = tls:<IP>:<PORT> >> >>> >> >>> in tls.cfg >> >>> >> >>> [server:<IP>:<PORT>] >> >>> method = TLSv1 >> >>> verify_certificate = no >> >>> require_certificate = no >> >>> private_key = /usr/local/etc/kamailio/kamailio-selfsigned.key >> >>> certificate = /usr/local/etc/kamailio/ >> kamailio-selfsigned.pem >> >>> >> >>> Now if i try to connect to this interface using openssl s_client, it >> >>> does connects, >> >>> but now server certificate is sent from kamailio. >> >>> >> >>> kamailio log shows this : >> >>> >> >>> <core> [ip_addr.c:247]: tcpconn_new: new tcp connection: <CLIENT >> >>> IP> >> >>> <core> [tcp_main.c:10

2 1

Kamailio Unexpectedly Terminating
by Akan 12 Jul '12

12 Jul '12

I have 2 servers running Solaris and Kamailio 3.2.3 where on one Kamailio is terminating when it tries to save the location for a register request and the other is producing a core dump when processing an Option request. I have one server handling Register request while the other sip server forwards the register requests and handles the other requests. I have included the backtraces from the core dumps and the output from the log for the registrar server as well as the command that is causing kamailio to terminate: if (!save("location")) sl_reply_error(); 4(3364) ERROR: *** cfgtrace: c=[/opt/kamailio-3.2/etc/kamailio/kamailio.cfg] l=714 a=17 n=if 4(3364) ERROR: *** cfgtrace: c=[/opt/kamailio-3.2/etc/kamailio/kamailio.cfg] l=711 a=26 n=save 14(3374) : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 15 14(3374) DEBUG: <core> [tcp_main.c:3555]: DBG: handle_ser_child: dead child 4, pid 3364 (shutting down?) 14(3374) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (1003743d8, 15, 0, 0x0) fd_no=18 called 0(3360) ALERT: <core> [main.c:751]: child process 3364 exited by a signal 10 0(3360) ALERT: <core> [main.c:754]: core was not generated 0(3360) INFO: <core> [main.c:766]: INFO: terminating due to SIGCHLD 6(3366) INFO: <core> [main.c:817]: INFO: signal 15 received 1(3361) INFO: <core> [main.c:817]: INFO: signal 15 received 2(3362) INFO: <core> [main.c:817]: INFO: signal 15 received 3(3363) INFO: <core> [main.c:817]: INFO: signal 15 received 5(3365) INFO: <core> [main.c:817]: INFO: signal 15 received 7(3367) INFO: <core> [main.c:817]: INFO: signal 15 received 8(3368) INFO: <core> [main.c:817]: INFO: signal 15 received 9(3369) INFO: <core> [main.c:817]: INFO: signal 15 received 10(3370) INFO: <core> [main.c:817]: INFO: signal 15 received 11(3371) INFO: <core> [main.c:817]: INFO: signal 15 received 12(3372) INFO: <core> [main.c:817]: INFO: signal 15 received 13(3373) INFO: <core> [main.c:817]: INFO: signal 15 received 14(3374) INFO: <core> [main.c:817]: INFO: signal 15 received 0(3360) DEBUG: presence_xml [presence_xml.c:347]: start 0(3360) ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Permission denied (13) 0(3360) DEBUG: <core> [db_pool.c:102]: removing connection from the pool 0(3360) DEBUG: db_postgres [km_pg_con.c:122]: PQfinish(100842470) 0(3360) DEBUG: db_postgres [km_pg_con.c:126]: pkg_free(1004c1f30) 0(3360) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start 0(3360) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table 0(3360) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores 0(3360) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists 0(3360) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done Thanks Nathaniel

4 38

sip over tls is not working
by Aft nix 11 Jul '12

11 Jul '12

Hi, I have enabled tls parameters as follows: in kamailio.cfg listen = tls:<IP>:<PORT> in tls.cfg [server:<IP>:<PORT>] method = TLSv1 verify_certificate = no require_certificate = no private_key = /usr/local/etc/kamailio/kamailio-selfsigned.key certificate = /usr/local/etc/kamailio/kamailio-selfsigned.pem Now if i try to connect to this interface using openssl s_client, it does connects, but now server certificate is sent from kamailio. kamailio log shows this : <core> [ip_addr.c:247]: tcpconn_new: new tcp connection: <CLIENT IP> <core> [tcp_main.c:1089]: tcpconn_new: on port 40727, type 3 <core> [tcp_main.c:1400]: tcpconn_add: hashes: 2614:2652:2494, 2 <core> [io_wait.h:390]: DBG: io_watch_add(0x82535e0, 23, 2, 0xb5701580), fd_no=11 <core> [io_wait.h:617]: DBG: io_watch_del (0x82535e0, 23, -1, 0x0) fd_no=12 called <core> [tcp_main.c:4296]: tcp: DBG: sending to child, events 1 <core> [tcp_main.c:3963]: WARNING: send2child: no free tcp receiver, connection passed to the least busy one (3289651) <core> [tcp_main.c:3967]: selected tcp worker 0 0(8) for activity on [tls:<IP>:<PORT>], 0xb5701580 <core> [tcp_main.c:3576]: BUG: handle_ser_child: fd -1 for 0 (pid 2491) I'm using kamailio from git. its updated to the latest. Thanks in advance. -- -aft

4 11

Re: [SR-Users] GRUU support in Kamailio
by amar＠smarttel.com.np 11 Jul '12

11 Jul '12

A -original message- Subject: Re: [SR-Users] GRUU support in Kamailio From: Iñaki Baz Castillo <ibc(a)aliax.net> Date: 11/07/2012 10:08 pm 2012/7/11 Olle E. Johansson <oej(a)edvina.net>: > Thanks for your patience, Daniel! And even more thanks for the code! I will test it in depth very soon. -- Iñaki Baz Castillo <ibc(a)aliax.net> _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

GRUU support in Kamailio
by Olle E. Johansson 11 Jul '12

11 Jul '12

Hey, Been trying to understand the GRUU support in Kamailio. I see that the registrar saves sip.instance - but I don't see any functions to generate gruu URIs or store them. A public GRUU is valid forever for a sip.instance and should propably be saved into some sort of database table. A temporary GRUU is only valid for as long as a device registers with the same Call-ID. Please give me pointers to what I have missed! Thanks, /O --- * Olle E Johansson - oej(a)edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden

3 11

DB create statements
by Klaus Darilion 11 Jul '12

11 Jul '12

Hi! For some reason I need to the enter the SQL statements manually on the DB server (I do not want to copy the scripts to the DB server and there is no superuser access over the network) Is there a simple method to get all the SQL commands to create the kamailio database? I see there are all the files in "/usr/share/kamailio/postgres/" and lots of other SQL statements in /usr/lib/kamailio/kamctl/kamdbctl.pgsql. Does anyone know of a simple method to get all the raw SQL queries? Thanks Klaus

2 2

start kamailio
by Michal Yachimovitz 11 Jul '12

11 Jul '12

Hello I installed kamailio version 3.2.0 and followed the instructions in http://www.kamailio.org/wiki/install/3.2.x/git web site Whene I permormed the command /etc/init.d/kamailio start I got the following errors. The modules are in the correct directory so actually I do not understand the error. I will be happy if someone can help me solve this issue: [root@kamailio-dev-lab ~]# */etc/init.d/kamailio start * Not starting Kamailio: invalid configuration file! 0(8569) ERROR: <core> [sr_module.c:565]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules/db_mysql.so>: /usr/local/lib/kamailio/modules/db_mysql.so: undefined symbol: SSL_CTX_free 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 214, column 12-24: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <mi_fifo> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 217, column 12-23: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <kex> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 218, column 12-19: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <tmx> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 220, column 12-19: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <rr> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 222, column 12-18: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <pv> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 223, column 12-18: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <maxfwd> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 224, column 12-22: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <usrloc> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 225, column 12-22: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <registrar> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 226, column 12-25: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <textops> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 227, column 12-23: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <siputils> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 228, column 12-24: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <xlog> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 229, column 12-20: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <acc> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 234, column 12-19: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <auth_db> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 238, column 12-23: failed to load module 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <mi_fifo> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 287, column 54: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <rr> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 301, column 35: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <rr> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 303, column 35: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <registrar> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 307, column 44: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <registrar> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 313, column 42: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 318, column 33: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 319, column 32: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 320, column 36: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 324, column 38: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 326, column 30: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 327, column 37: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 330, column 45: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 331, column 45: Can't set module parameter 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 346, column 30-34: syntax error 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 346, column 30-34: Invalid arguments 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 346, column 35: ERROR: bad config file (30 errors) [root@kamailio-dev-lab ~]# Thanks in advance Michal Yachimovitz

2 3

Re: [SR-Users] is kamilio support RFC5626 now ?
by Daniel-Constantin Mierla 11 Jul '12

11 Jul '12

FYI: private emails sent from mailing lists discussions are simply ignored, keep the mailing list in the discussion. Thanks. On 7/11/12 9:13 AM, mike wrote: > yes , i know that the path could be added by the path module , but > ,the problem is the flow token,such as the path > > Path: <sip:VskztcQ/S8p4WPbOnHbuyh5iJvJIW3ib@ep1.example.com > <mailto:S8p4WPbOnHbuyh5iJvJIW3ib@ep1.example.com>;lr;ob> > > > the VskztcQ/S8p4WPbOnHbuyh5iJvJIW3ib is the token , so , where the > kamilio 3.3 handle it ? > > > On Wed, Jul 11, 2012 at 3:00 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > > On 7/11/12 4:23 AM, mike wrote: > > dear kamilio developers : > > i read the kamilio 3.3.0' s new features , it say it support > outbound , and i have place to handle reg-id and sip.instance > . so , i wonder if the kamilio3.3.0 support RFC5626 now，so > that the > kamilio can be used as a edge proxy . i ask why because i > can't find where kamilio add flow-token and path for the > register request. > > sip.instance and reg-id are handled by the registrar/usrloc > modules. Path has to be added by intermediary proxies and it is > also handled by registrar module. If the intermediary proxies are > kamailio, then adding path is possible via path module. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - > http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - > http://asipto.com/u/katu > Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - > http://asipto.com/u/kpw > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

usrloc concurrency
by Alex Balashov 11 Jul '12

11 Jul '12

(Reposting from sr-dev because, in retrospect, it makes more sense here.) Does anyone know of any issues running usrloc with DB backing ('location') in a multi-master DB-replicated environment? I am referring to plain usrloc, not p_usrloc. I have such an environment set up with db_mode = 1 (immediate write-through) and have run into a problem where Kamailio occasionally stops writing to the location table for no apparent reason. I am wondering if this is because another Kamailio instance bound to the secondary replica is performing a DB sync as well, and at some point they collide in some sort of race condition. Would db_mode 2 fix? Would anything fix? Is this even the problem? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com

2 3

How to proxy/authenticate on third party sip server?
by intel＠intrans.baku.az 10 Jul '12

10 Jul '12

Can anybody help me with configuration kamailio to authenticate call on 3rd party server? I want to authenticate(and proxy) call's on other server. for example, user A have account on my server(let it be A(a)myserver.com) and account on other server (let it be AA(a)hisserver.com) myserver.com keeps A's credentials for AA(a)hisserver.com when A is registered on myserver.com, he make call to B(a)othersip.com (using myserver.com as a proxy) Kamailio on myserver authenticate itself on hisserver as AA(a)hisserver.com, makes call to destination, and connect it to A. How can I setup kamailio for such behavior? Which module should I use? UAC seems can authenticate on another server, but I don't sure that it do what I want. (I've tried it, but without much success)

2 8

Does there exist a tutorial on the Kamailio SIP/XMPP Gateway configuration?
by Auro Tripathy 10 Jul '12

10 Jul '12

I'm finding that I may have to piece together (the configuration info) from version emails/presentations. Reluctantly, I'll take that approach...unless someone has a lucid HOWTO on the topic? Again, I'm new to the domain and I appreciate the subject-matter experts that have helped out so far. -Auro

2 1

[Fwd: [sr-dev] WebSocket Server support in Kamailio master]
by Peter Dunkley 09 Jul '12

09 Jul '12

-------- Forwarded Message -------- From: Peter Dunkley <peter.dunkley(a)crocodile-rcs.com> Reply-to: Development mailing list of the sip-router project <sr-dev(a)lists.sip-router.org> To: sr-dev(a)lists.sip-router.org Subject: [sr-dev] WebSocket Server support in Kamailio master Date: Sat, 7 Jul 2012 18:29:23 +0100 Hello, I have merged the pd/websocket branch into Kamailio master. This means that you can now connect SIP over WebSocket (draft-ibc-sipcore-sip-websocket-02) clients to Kamailio using the "ws://" and "wss://" protocols. Some customisation of the websocket module is possible through modparams, but for most users the defaults should be OK. The WebSocket module uses the xhttp and sl modules for the initial handshake, and (unless you have both a Kamailio installation and WebSocket SIP client supporting GRUU, Outbound[1], and Path[2]) nathelper for request routing and the core force_rport() function for response routing (a new nat_uac_test() has been added to detect whether a message has arrived on a WebSocket). There is an example kamailio.cfg in the websocket module directory. [1] Kamailio does not currently support Outbound [2] I have not updated the Path module for WebSockets I believe that, once Kamailio supports Outbound and WebSocket support is added to the Path module (and you have a SIP over WebSocket client that supports this), it will be possible to use the websocket module without the nathelper module and force_rport() and without needing to change the websocket module or Kamailio core code. If you want to use secure WebSockets (wss) as well as ordinary WebSockets just configure TLS and listen on an appropriate port. I have added WebSocket support to some modules, but there are definitely going to be others (modules/lcr, modules/sipcapture, modules_k/nat_traversal, modules_k/path, modules_k/seas, and modules_k/snmpstats, at least) that need updating too. WebSockets is an unusual transport, so I have put a few notes together for anyone who needs to use it in the code (including adding support to additional modules): - A WebSocket server cannot initiate a WebSocket connection. So a WebSocket connection (over TCP or TLS) is like a TCP/TLS connection coming from behind a NAT. This is why nathelper aliasing and force_rport() is used for the routing, and "set_..._no_connect()" is always used (it's set within the websocket module). - WebSocket (PROTO_WS) and secure WebSocket (PROTO_WSS) connections are just upgraded TCP and TLS connections, so there are no listening sockets for PROTO_WS and PROTO_WSS. This means that, when deciding on what transport is being used, you need to look at the proto set in the tcp_connection, receive_info, and/or dest_info structure for the message - looking at the socket_info structure (that the message has arrived on or will be sent on) will not give you the right answer. - Although WebSocket (PROTO_WS) and secure WebSocket (PROTO_WSS) are different internal protocols there is only one SIP transport type for both ";transport=ws" (WS and WSS are explicitly used in Via: headers though). This means that you can't tell whether the transport parameter in an R-URI, Route/Record-Route, or Contact-URI is for WebSockets or secure WebSockets. As long as the message makes it into the WebSocket module everything will be OK as that module sorts it all out, but it has led to slightly more complex checks being required in some of the code relating to record-routing to handle this - and it may have an effect on other modules too. Please give the new module a go and let me know about any issues you find, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd _______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Removing display-name with uac_replace_from
by Andreas Granig 09 Jul '12

09 Jul '12

Hi, in kamailio 3.3, doing uac_replace_from("", "$var(myuri)"); doesn't remove the old Display part anymore (although the documentation still says it should), the debug log is: DEBUG: uac [uac.c:499]: dsp=(nil) (len=0) , uri=0x7fff87f6b840 (len=28) DEBUG: uac [replace.c:324]: uri to replace [sip:a@example.org] DEBUG: uac [replace.c:325]: replacement uri is [sip:b@example.org] DEBUG: uac [replace.c:383]: encode is=<UEtlSyomCD8kIi8/SnlZMzNyS29QS2VLanBKeg--> len=40 So no hint that the old display is being removed. When doing uac_replace_from("foo", "$var(myuri)"); then there's an additional line like this: DEBUG: uac [replace.c:284]: removing display ["olddisplay"] which is missing when I put an empty string in the first parameter. Andreas

2 4

future features?
by 张 09 Jul '12

09 Jul '12

Hello Dear all, I'm new one to participate into this discussion. I have worked on VOIP and SIP communication solution for several years. I'm very excited about the new features in v3.3. They save me many work which I planed to dig hard. My question is whether there is a web page or discuss forum for future features for Kamailio? There is a fure feature page on sip-router.org but it's very out of date. I think it's important to catch up with potential wanted feature and the new draft which is closely watched by kamailio team. Thanks! Best Regards Victor

2 1

ACK and BYE messages uses wrong socket.
by Gary Chen 05 Jul '12

05 Jul '12

Kamailio 3.2.0 I am trying to setup kamailio to do the sip trunking. It receive the sip traffic from customer and then send it to carrier. I have two NIC interface's assigned with three IP's: Interface 1: ( Public IP's) x.x.130.34 x.x.130.36 floating IP interface 2: (private IP's) 10.10.1.31 .36 is a floating IP assigned by Linux-HA (heartbeat/pacemaker). I only want to use .36 to receive and send sip traffic. I uses force_send_socket() to send INVITE with .36 IP. But the ACK message always want to use .34 IP even the Route header has .36 in it unless I force it with force_send_socket() . How can I fix this problem? See below for the SIP messages: (x.x.128.205 is customer IP, x.x.129.200 is PSTN gateway IP) U x.x.128.205:51694 -> x.x.130.36:5060 INVITE sip:5033441174@x.x.130.36:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG 4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.3 6>..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A(a)x.x.128.20 5..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254113-3191264919-256 0877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, CO MET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 70..Remote-P arty-ID: <sip:5024427578@x.x.128.205>;party=calling;screen=no;privacy=off..Timestamp: 1341322661 ..Contact: <sip:5024427578@x.x.128.205:5060>..Expires: 180..Allow-Events: telephone-event..Conte nt-Type: application/sdp..Content-Length: 366....v=0..o=CiscoSystemsSIP-GW-UserAgent 9094 579 IN IP 4 x.x.128.205..s=SIP Call..c=IN IP4 x.x.128.205..t=0 0..m=audio 19312 RTP/AVP 125 0 18 100 10 1..c=IN IP4 x.x.128.205..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a =fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone-event/8000 ..a=fmtp:101 0-16.. # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9 hG4bK1D3CD1;rport=51694..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:502244117 4(a)x.x.130.36>..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..CSeq: 101 INVITE..Se rver: LVS Proxy 1.0..Content-Length: 0.... # U x.x.130.36:5060 -> x.x.129.200:5060 INVITE sip:15033441174@x.x.129.200:5060 SIP/2.0..Record-Route: <sip:x.x.130.36;lr=on>..Via: S IP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport =51694;branch=z9hG4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:50224 41174(a)x.x.130.36>..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3 E39A@x.x.128.205..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254 113-3191264919-2560877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANC EL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forw ards: 69..Remote-Party-ID: <sip:5024427578@x.x.128.205>;party=calling;screen=no;privacy=off..Tim estamp: 1341322661..Contact: <sip:5024427578@x.x.128.205:51694>..Expires: 180..Allow-Events: tel ephone-event..Content-Type: application/sdp..Content-Length: 375....v=0..o=CiscoSystemsSIP-GW-UserA gent 9094 579 IN IP4 10.200.1.51..s=SIP Call..c=IN IP4 10.200.1.51..t=0 0..m=audio 20464 RTP/AVP 12 5 0 18 100 101..c=IN IP4 10.200.1.51..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G7 29/8000..a=fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone- event/8000..a=fmtp:101 0-16..a=nortpproxy:yes.. U x.x.129.200:5060 -> x.x.130.36:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/UDP 216.4 9.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;tag=24513088 -D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Cal l-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPG ateway/IOS-12.x..CSeq: 101 INVITE..Allow-Events: telephone-event..Content-Length: 0.... # U x.x.129.200:5060 -> x.x.130.36:5060 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/ UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;ta g=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:4 1 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 101 INVITE..Require: 100rel..RSeq: 6708..Allow: INVITE, OPTIONS, B YE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: tele phone-event..Contact: <sip:15033441174@x.x.129.200:5060>..Record-Route: <sip:x.x.130.36;lr=on >..Content-Disposition: session;handling=required..Content-Type: application/sdp..Content-Length: 2 90....v=0..o=CiscoSystemsSIP-GW-UserAgent 2387 2116 IN IP4 x.x.129.200..s=SIP Call..c=IN IP4 216 .49.129.200..t=0 0..m=audio 18480 RTP/AVP 0 101 100..c=IN IP4 x.x.129.200..a=rtpmap:0 PCMU/8000. .a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194.. # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D3CD 1..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F 0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@216. 49.128.205..Timestamp: 1341322661..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 101 INVITE..Require: 10 0rel..RSeq: 6708..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: telephone-event..Contact: <sip:15033441174@x.x.129.200:50 60>..Record-Route: <sip:x.x.130.36;lr=on>..Content-Disposition: session;handling=required..Conte nt-Type: application/sdp..Content-Length: 305....v=0..o=CiscoSystemsSIP-GW-UserAgent 2387 2116 IN I P4 x.x.130.66..s=SIP Call..c=IN IP4 x.x.130.66..t=0 0..m=audio 19480 RTP/AVP 0 101 100..c=IN IP4 x.x.130.66..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpm ap:100 X-NSE/8000..a=fmtp:100 192-194..a=nortpproxy:yes.. # U x.x.128.205:51694 -> x.x.130.36:5060 PRACK sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9h G4bK1D4474..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130. 36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3 E39A@x.x.128.205..CSeq: 102 PRACK..RAck: 6708 101 INVITE..Route: <sip:x.x.130.36;lr=on>..Max- Forwards: 70..Content-Length: 0.... # U x.x.130.34:5060 -> x.x.129.200:5060 PRACK sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bK92c e.4def8436.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D4474..From: <sip:50 24427578(a)x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Dat e: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..CSeq : 102 PRACK..RAck: 6708 101 INVITE..Max-Forwards: 69..Content-Length: 0.... # U x.x.129.200:5060 -> x.x.130.34:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bK92ce.4def8436.0,SIP/2.0/UDP x.x.12 8.205:5060;rport=51694;branch=z9hG4bK1D4474..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59 ..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID : 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 102 PRACK..Content-Length: 0.... # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D4474..From: <sip: 5024427578(a)x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..D ate: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Se rver: Cisco-SIPGateway/IOS-12.x..CSeq: 102 PRACK..Content-Length: 0.... U x.x.129.200:5060 -> x.x.130.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/UDP x.x.12 8.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59 ..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID : 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPGatew ay/IOS-12.x..CSeq: 101 INVITE..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSC RIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: telephone-event..Contact: <sip:15033441174@216. 49.129.200:5060>..Record-Route: <sip:x.x.130.36;lr=on>..Content-Type: application/sdp..Content-L ength: 290....v=0..o=CiscoSystemsSIP-GW-UserAgent 2387 2116 IN IP4 x.x.129.200..s=SIP Call..c=IN IP4 x.x.129.200..t=0 0..m=audio 18480 RTP/AVP 0 101 100..c=IN IP4 x.x.129.200..a=rtpmap:0 PC MU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpmap:100 X-NSE/8000..a=fmtp:100 19 2-194.. # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: <sip: 5024427578(a)x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..D ate: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Ti mestamp: 1341322661..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 101 INVITE..Allow: INVITE, OPTIONS, B YE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: tele phone-event..Contact: <sip:15033441174@x.x.129.200:5060>..Record-Route: <sip:x.x.130.36;lr=on >..Content-Type: application/sdp..Content-Length: 305....v=0..o=CiscoSystemsSIP-GW-UserAgent 2387 2 116 IN IP4 x.x.130.66..s=SIP Call..c=IN IP4 x.x.130.66..t=0 0..m=audio 19480 RTP/AVP 0 101 10 0..c=IN IP4 x.x.130.66..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16 ..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=nortpproxy:yes.. # U x.x.128.205:51694 -> x.x.130.36:5060 ACK sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG4 bK1D5DA6..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36 >;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E3 9A@x.x.128.205..Route: <sip:x.x.130.36;lr=on>..Max-Forwards: 70..CSeq: 101 ACK..Content-Lengt h: 0.... # U x.x.130.34:5060 -> x.x.129.200:5060 ACK sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bKcydzi gwkX..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D5DA6..From: <sip:5024427578 @x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Max-Forwards : 69..CSeq: 101 ACK..Content-Length: 0.... # U x.x.128.205:51694 -> x.x.130.36:5060 BYE sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG4 bK1D61CE7..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.3 6>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E 39A@x.x.128.205..User-Agent: Cisco-SIPGateway/IOS-12.x..Max-Forwards: 70..Route: <sip:x.x.130 .36;lr=on>..Timestamp: 1341322666..CSeq: 103 BYE..Reason: Q.850;cause=16..Content-Length: 0.... # U x.x.130.34:5060 -> x.x.129.200:5060 BYE sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bKa2ce. 4ba2ee65.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D61CE7..From: <sip:502 4427578(a)x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date : Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..User- Agent: Cisco-SIPGateway/IOS-12.x..Max-Forwards: 69..Timestamp: 1341322666..CSeq: 103 BYE..Reason: Q .850;cause=16..Content-Length: 0.... # U x.x.129.200:5060 -> x.x.130.34:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bKa2ce.4ba2ee65.0,SIP/2.0/UDP x.x.12 8.205:5060;rport=51694;branch=z9hG4bK1D61CE7..From: <sip:5024427578@x.x.128.205>;tag=24513088-D5 9..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:46 GMT..Call-I D: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Server: Cisco-SIPGateway/IOS-12.x..Timestamp : 1341322666..Content-Length: 0..CSeq: 103 BYE.... # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D61CE7..From: <sip :5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F.. Date: Tue, 03 Jul 2012 13:37:46 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..S erver: Cisco-SIPGateway/IOS-12.x..Timestamp: 1341322666..Content-Length: 0..CSeq: 103 BYE....

3 6

Event when location expires?
by Niklas Larsson 03 Jul '12

03 Jul '12

Hi all, is there a way to get a event / log / something when an registry expires? /niklas

2 1

Releasing v3.1.6 - end of packaging maintenance for 3.1 branch
by Daniel-Constantin Mierla 03 Jul '12

03 Jul '12

Hello, in the near future I will package 3.1.6 to mark the end of releasing from 3.1 branch. Of course, patches can still go into that branch, but there is no intent to package again from it. If anyone is willing to push other commits before the packages, please do it in then next 1-2 days. I will do it when I get enough spare time for it, with a short notice before it in that day. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

Binding arbitrary TLS socket
by Andreas Granig 02 Jul '12

02 Jul '12

Hi, Got some problems when I tried this, not sure how this is intended to work: listen=tls:192.168.51.133:4343 When starting kamailio 3.3, I get the following error: ERROR: <core> [tcp_main.c:2915]: ERROR: tcp_init: bind(10, 0x7f5eea45db24, 16) on 192.168.51.133:4343 : Address already in use ERROR: tls [tls_init.c:314]: Error while initializing TCP part of TLS socket 192.168.51.133:4343 What works though is if I explicitely bind a TCP socket one port below the TLS socket, like this: listen=tcp:192.168.51.133:4342 listen=tls:192.168.51.133:4343 Is this how it's intended to work? Is there a specific reason for this behavior? Andreas

2 2

cfgutils lock / unlock + $sht() variables.
by Vitaliy Aleksandrov 02 Jul '12

02 Jul '12

Hi all, I use $sht() pseudo-variabled provided by htable module for caching various information about sip customers. I thought that htable has some locking mechanism inside it and today while i was reading core-cookbook i payed attention to one example with the lock/unlock functions and $sht(): lock("calls-to::10.10.10.10"); $sht(a=>calls-to::10.10.10.10) = $sht(a=>calls-to::10.10.10.10) + 1; unlock("calls-to::10.10.10.10"); According to that example an explicit locking mechanism should be used when i want to write something to a hash table. On the other hand there are no locks in htable documentation. So the question is when do the explicit locks should be used and for what purpose ?

2 1

TCP Forwarding error: cannot forward to af 2, proto 2 no correspondinglistening socket
by Omar Labs 188 Gmail 02 Jul '12

02 Jul '12

tried many versions in version 1.4 i have this error: Jun 29 21:25:04 [7309] ERROR:core:forward_request: cannot forward to af 2, proto 2 no correspondinglistening socket i do have the TCP port listen explicitly the line for disable-tcp is commented. listen=udp:192.168.169.169:5060 listen=udp:192.168.169.169:5060 if ( src_ip == 192.168.169.180 ) { forward("tcp:192.168.170.156:5060"); } I tried already many options related the forward command , using variables and using forward(); ``Listening on udp: 192.168.169.169 [192.168.169.169]:5060 tcp: 192.168.169.169 [192.168.169.169]:5060 Aliases: tcp: loghost:5060 tcp: kamtest01:5060 udp: loghost:5060 udp: kamtest01:5060 bash-3.00# telnet 192.168.169.169 5060 Trying 192.168.169.169... Connected to 192.168.169.169. Escape character is '^]'. Connection to 192.168.169.169 closed by foreign host. bash-3.00# kamctl ps Process:: ID=0 PID=4541 Type=attendant Process:: ID=1 PID=4542 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=2 PID=4543 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=3 PID=4544 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=4 PID=4545 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=5 PID=4546 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=6 PID=4547 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=7 PID=4548 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=8 PID=4549 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=9 PID=4550 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=10 PID=4551 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=11 PID=4552 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=12 PID=4553 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=13 PID=4554 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=14 PID=4555 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=15 PID=4556 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=16 PID=4557 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=17 PID=4558 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=18 PID=4559 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=19 PID=4560 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=20 PID=4561 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=21 PID=4562 Type=timer Process:: ID=22 PID=4563 Type=MI FIFO Process:: ID=23 PID=4564 Type=TCP receiver Process:: ID=24 PID=4565 Type=TCP receiver Process:: ID=25 PID=4566 Type=TCP receiver Process:: ID=26 PID=4567 Type=TCP receiver Process:: ID=27 PID=4568 Type=TCP receiver Process:: ID=28 PID=4569 Type=TCP receiver Process:: ID=29 PID=4570 Type=TCP receiver Process:: ID=30 PID=4571 Type=TCP receiver Process:: ID=31 PID=4572 Type=TCP receiver Process:: ID=32 PID=4573 Type=TCP receiver Process:: ID=33 PID=4574 Type=TCP main O

2 1

How to access reason phrase of winning reply in failure_route
by Barthel Marco (CI/AFU1) 02 Jul '12

02 Jul '12

All, how do I access the reason phrase of winning reply in failure_route? The reply code can be accessed via $T_reply_code. Is there something similar for the reason phrase? Mit freundlichen Grüßen / Best regards Marco Barthel Robert Bosch GmbH (CI/AFU1) www.bosch.com Tel. +49 711 811-3602341 marco.barthel(a)de.bosch.com Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart, HRB 14000; Aufsichtsratsvorsitzender: Franz Fehrenbach; Geschäftsführung: Volkmar Denner, Siegfried Dais; Stefan Asenkerschbaumer, Bernd Bohr, Rudolf Colm, Dirk Hoheisel, Christoph Kübel, Uwe Raschke, Wolf-Henning Scheider, Werner Struth, Peter Tyroller

2 1

Re: [SR-Users] how to talk across domains, my domain to say, iptel.org
by Alex Balashov 29 Jun '12

29 Jun '12

Note that from a security point of view, this is not necessarily a desirable trait, without appropriate programmatic restrictions and/or ACLs. -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Auro Tripathy <Auro.Tripathy(a)wdc.com> wrote:Thanks Klaus, non-local routing appears to work out-of-the-box. Thank you again for confirming what I observed. -Auro From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Friday, June 29, 2012 3:52 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Auro Tripathy Subject: Re: [SR-Users] how to talk across domains, my domain to say, iptel.org The default configuration supports that already. Usually inbound routing does not differ if the call comes from a local subscriber or from an external domain. Outbound routing is performed if the domain in the request URI (ruri) is non-local. regards Klaus Am 29.06.2012 19:46, schrieb Auro Tripathy: Hi All: I’m a newbie, excuse the simple (no pun intended) question. I installed Kamailio and I’m able to set it up so that all subscribers in the *same* domain can talk/discover-presence, etc. What do I do so that subscribers in other domains, for example the iptel.org domain, can talk to subscribers in my domain? -Auro _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

how to talk across domains, my domain to say, iptel.org
by Auro Tripathy 29 Jun '12

29 Jun '12

Hi All: I'm a newbie, excuse the simple (no pun intended) question. I installed Kamailio and I'm able to set it up so that all subscribers in the *same* domain can talk/discover-presence, etc. What do I do so that subscribers in other domains, for example the iptel.org domain, can talk to subscribers in my domain? -Auro

2 2

Serctl
by Angel Valdes 29 Jun '12

29 Jun '12

Please is any variable available to stop screen scrolling when you use serctl showdb and you have that many users list bigger than your screen and you need to see the top part of the list Thanks Angel Valdes IT Manager 201-920-2908 Talent is a gift that brings with it an obligation to serve the world, and not ourselves, for it is not of our making. Jose Marti

2 1

Re: [SR-Users] Nathelper module, FLT_NATS, FLT_NATB
by Richard Brady 28 Jun '12

28 Jun '12

Klaus / Daniel Thanks again for assistance with this. I've tried the solution based on add_contact_alias() and handle_ruri_alias() and it works perfectly. Richard On 22 June 2012 13:47, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > > > On 22.06.2012 13:50, Richard Brady wrote: >> >> Thanks guys, fantastic answers. >> >> You mention that NAT detection happens before save() and the flag is set >> by lookup() which makes much more sense. However, if Kamailio is not the >> registrar, as is the case with my current project, those functions are >> not called, so an alternative is needed. There are clearly several >> options. >> >> The solution I have gone for is to replace fix_nated_register() with >> fix_nated_contact() so that the REGISTER request is relayed with a >> modified Contact header containing the external ip:port of the client. > > > The cleanest solution would be to use add_contact_alias() and > handle_ruri_alias(). The do not change the contact but put the public > address into a uri parameter. Thus, the URI seen by the client is always the > one it sends: > http://www.kamailio.org/docs/modules/3.2.x/modules_k/nathelper.html#id25504… > > >> That is then stored by the registrar (FreeSWITCH in my case) and used >> later to originate calls for that user. The FreeSWITCH know to send >> those calls to Kamailio through either use of the Path header and module >> in Kamailio, or through static configuration of fs_path or proxy >> parameters in FreeSWITCH. > > > This is fine. > > >> The works for the first INVITE to the registered client behind NAT. But >> that client sends back a 200 OK with a Contact header containing its >> private IP address, and so fix_nated_contact() needs to be invoked on >> that response, and normally it would be due to FLB_NATB being set, but >> if Kamailio was not the registrar then that flag is not set. So I need >> to detect NAT on the client at the time of receiving the reply, or >> alternatively by having the registrar store a cookie and setting it >> based on that. > > > You are correct. For in-dialog messages received from SIP clients I would > always use add_contact_alias() and remove the NAT flags completely. > >> >> I suppose then my next question then is can I call nat_uac_test() on a >> UAS? > > > Today, almost any SIP clients are SIP symmetric. This means, that they > receive SIP from the some port/connection where they send. Thus, skip the > NAT tests completely and always use add_contact_alias() for messages receive > from SIP clients and handle_ruri_alias() for messages sent to SIP clients. > > Depending on if you use Freeswitch also as media relay you can also remove > the media proxy stuff from the Kamailio config. > > On important thing is NAT-keep-alive. This is usually done by nathelper > module by querying the location table for contact with NAT-flag set. Thus, > either you do NAT keep-alive by freeswitch (e.g sending OPTIONS requests) or > do it in your Kamailio proxy (e.g. set the nat_bflag > http://www.kamailio.org/docs/modules/3.2.x/modules_k/usrloc.html#id2541477 > and call save("location","0x02") before relaying REGISTER to freeswitch. > Then Kamailio will do NAT-pinging. Note, if you want to keep-alive only for > successfully registered clients, you man want to call save() in the > reply-route of the REGISTER request). > > regards > Klaus

4 5

Git Error on Kamailio-3.3 Install
by SamyGo 28 Jun '12

28 Jun '12

Hello, I'm setting up new Kamailio 3.3 from git ref-link: http://www.kamailio.org/wiki/install/3.3.x/git I'm using CentOS 5.8 and when I execute the line # git checkout -b 3.3 origin/3.3 it gives me error. fatal: git checkout: updating paths is incompatible with switching branches. Did you intend to checkout 'origin/3.3' which can not be resolved as commit? Can I skip it ! o need to do something about it !! Regards, Sammy G.

3 6

Segfault running 3.3.0 with registrar/p_usrloc/path
by Dan-Cristian Bogos 28 Jun '12

28 Jun '12

Hey Guys, After upgrading from 3.2.x to 3.3.0 (out of debian packages and fixing db side) Kamailio is crashing on processing REGISTER. This crash only happens for updates (or if a stale contact for the same AOR is already present), not initial REGISTERs. Attached you can find the log taken with debug=6 as well as the REGISTER package producing the segfault. Kamailio version: """ root@PrxyDev:/etc/slocator# kamailio -V version: kamailio 3.3.0 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_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, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 22:17:07 Jun 18 2012 with gcc 4.4.5 """ Syslog msg in debug=2: """ Jun 27 15:46:22 PrxyDev kernel: [11330.002854] kamailio[6959]: segfault at 0 ip (null) sp bf8d531c error 4 in kamailio[8048000+1ea000] Jun 27 15:46:22 PrxyDev /usr/sbin/kamailio[6970]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 10 Jun 27 15:46:22 PrxyDev /usr/sbin/kamailio[6957]: ALERT: <core> [main.c:785]: child process 6959 exited by a signal 11 Jun 27 15:46:22 PrxyDev /usr/sbin/kamailio[6957]: ALERT: <core> [main.c:788]: core was not generated """ Thanks in advance for any tip! DanB

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Re: [SR-Users] TM module timer question
by Gertjan Wolzak 28 Jun '12

28 Jun '12

Thanks guys for the info. Will try again and keep on it. Rgds, Gertjan On 6/28/12 08:55:27 AM Daniel-Constantin Mierla wrote; Hello, On 6/27/12 6:08 PM, Corey Edwards wrote: > On 06/26/2012 08:36 AM, Gertjan Wolzak wrote: >> Which timer do I need to use to get a failure route to reroute the >> call to a pstn gateway? >> >> >> >> I have tried the fr_timer which works, I set it to 5000, downside is >> that when I am in wifi range, I only have 5 seconds to answer the call. >> >> >> >> The fr_inv_timer is not the right one I think, as I will not get any >> message from the called number, as that device is not connected?. > I'm not an expert on this, but I'll post my experience and I hope that > helps. I'm using Kamailio 1.5, so perhaps YMMV. > > I have fr_timer set to 2 seconds and fr_inv_timer set to 60 seconds. > The latter can be overridden by fr_inv_timer_avp so users can alter > their own timers. I do not let users change fr_timer. > > The behavior I see is the device has 2 seconds to give a provisional > response to an INVITE. If the device has powered off, changed IP, etc. > this timer will quickly return an error. > > If the device is responsive and, for example, sends a 180 Ringing, the > fr_timer ceases to apply and the fr_inv_timer waits 60 seconds before > giving up and triggering a 408 Timeout. it is the same in 3.x, these two parameters are the right to be used in this way. Have in mind that in 3.x the accuracy is in milliseconds: <http://kamailio.org/docs/modules/stable/modules/tm.html#fr_timer> http://kamailio.org/docs/modules/stable/modules/tm.html#fr_timer Another new features in 3.x is that apart of using avps to set it per transaction (still in seconds), there is a new function that can be used to set it in milliseconds per transaction: <http://kamailio.org/docs/modules/stable/modules/tm.html#t_set_fr> http://kamailio.org/docs/modules/stable/modules/tm.html#t_set_fr Cheers, Daniel

1 0

system call
by Kelvin Chua 28 Jun '12

28 Jun '12

hi guys, i'm currently using cron to alert the admin via email of any irregularities (e.g. pike) but i'm wondering, is it possible to do a system call from the cfg script?

2 1

Question about git update
by Ricardo Martinez 28 Jun '12

28 Jun '12

Hello. Can someone help me on how to update a module patch using GIT? I need to update the module dialog….. Currently I have the directory /usr/local/src/kamailio-3.2 And under this I have kamailio/ all this was created using : git clone --depth 1 git://git.sip-router.org/sip-router kamailio cd kamailio git checkout -b 3.2 origin/3.2 I’ll appreciate any help. Thanks in advance, Ricardo Martinez.-

2 1

Questions about login to account
by Daniel Butenko 28 Jun '12

28 Jun '12

3 2

Re: [SR-Users] TM module timer question
by Corey Edwards 28 Jun '12

28 Jun '12

On 06/26/2012 08:36 AM, Gertjan Wolzak wrote: > Which timer do I need to use to get a failure route to reroute the call > to a pstn gateway? > > > > I have tried the fr_timer which works, I set it to 5000, downside is > that when I am in wifi range, I only have 5 seconds to answer the call. > > > > The fr_inv_timer is not the right one I think, as I will not get any > message from the called number, as that device is not connected…. I'm not an expert on this, but I'll post my experience and I hope that helps. I'm using Kamailio 1.5, so perhaps YMMV. I have fr_timer set to 2 seconds and fr_inv_timer set to 60 seconds. The latter can be overridden by fr_inv_timer_avp so users can alter their own timers. I do not let users change fr_timer. The behavior I see is the device has 2 seconds to give a provisional response to an INVITE. If the device has powered off, changed IP, etc. this timer will quickly return an error. If the device is responsive and, for example, sends a 180 Ringing, the fr_timer ceases to apply and the fr_inv_timer waits 60 seconds before giving up and triggering a 408 Timeout. Corey

2 1

A question about installing kamalio
by Aurel Sandu 28 Jun '12

28 Jun '12

Hello all of you, I tried to install kamallio 3.3 version released a couple days ago . I followed the steps mentioned here https://www.kamailio.org/wiki/install/3.3.x/git Everything works fine until I have to create mysql database . Everything goes fine until I have to install mysql database (step 7) : I run "sudo /usr/local/sbin/kamdbctl create" for database creation but I get this : INFO: test server charset /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 105: mysql: command not found /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 106: mysql: command not found 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 ... /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 71: mysql: command not found ERROR: Creating core database and grant privileges failed! It seems that mysql -serverside- is not installed (my opinion ...) .What do you think? Many Thanks, Aurel ps: please excuse me if this is not right forum for this question but I did not find the right one ....

2 1

Re: [SR-Users] dlg_refer
by Alex Balashov 28 Jun '12

28 Jun '12

It sends an actual REFER. -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Uri Shacked <ushacked(a)gmail.com> wrote:Hi, before trying dlg_refer, does anyone know, does it send REFER to the caller/callee? or doeas it do reinvite? i ask this because my class 4 softswitch does not support REFER.... thanks, Uri

1 0

dlg_refer
by Uri Shacked 28 Jun '12

28 Jun '12

Hi, before trying dlg_refer, does anyone know, does it send REFER to the caller/callee? or doeas it do reinvite? i ask this because my class 4 softswitch does not support REFER.... thanks, Uri

1 0

Subscription-State: terminated;reason=terminated?
by Min Wang 27 Jun '12

27 Jun '12

HI when I removed 102 from 101's contact list (using jitsi nightly 1.1 build), kamailio 3.3 send out NOTIFY to 102 like this: NOTIFY sip:102@192.168.122.147:5060;transport=udp;registering_acc=192_168_122_32 SIP/2.0. Via: SIP/2.0/UDP 192.168.122.32;branch=z9hG4bK1bfb.afbf0a85.0. To: sip:102@192.168.122.32;tag=f6a40771. From: sip:101@192.168.122.32;tag=a6a1c5f60faecf035a1ae5b6e96e979a-5724. CSeq: 4 NOTIFY. Call-ID: c7c52dd058268596ec84dd3c645a2f17(a)0.0.0.0. Content-Length: 0. User-Agent: kamailio (3.3.0-rc0 (x86_64/linux)). Max-Forwards: 70. Event: presence. Contact: <sip:192.168.122.32:5060;transport=udp>. Subscription-State: terminated;reason=terminated. <----------------- Note the reason code is:terminated. From rfc3265, The defined reason codes are: deactivated/ probation/rejected/ timeout/giveup/noresource What is the reason to send: reason=terminated instead one of the well-defined reason codes? There was a discussion regarding at: http://sip-router.org/tracker/index.php?do=details&task_id=133 <http://sip-router.org/tracker/index.php?do=details&task_id=133> but I did not see the explaination of reason=terminated. Thanks min

3 13

default kamailio configuration
by Vitaliy Aleksandrov 27 Jun '12

27 Jun '12

Hi All, After reading default kamailio configuration i can't understand why does kamailio remove preloaded route headers from the incoming initial INVITE before calling record_route(). remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); What will happen if an INVITE comes from another SIP proxy which wants to stay in the middle of the dialogue ? For example, during Re-INVITE UA from our local site will construct in-dialog INVITE with R-URI taken from the remote contact and send it to our local proxy according to RR. Local proxy will process it in route[WITHNDLG] where loose_route() function will remove top most RR header because it points to us and since there is no more RR records forward the request to a destination from the R-URI. As i understand in that case remote proxy will not receive any in-dialog requests from us.

3 3

kamailio.org server maintenance - estimated 1 hour downtime, 26/27 June
by Daniel-Constantin Mierla 27 Jun '12

27 Jun '12

Hello, during the night of 26/27th of June, Central European Standard Time (GMT+1, with daylight saving time on), the server hosting kamailio.org and lists.sip-router.org will be down for approx 1 hour. This is a planned maintenance work to upgrade the server to a new infrastructure. Hopefully everything goes smooth and gets back online quickly, the work should start about 2:00am on the 27th of June, but don't get nervous if takes longer or happens at different time. Among the most important affected services: - the mailing lists (all mailed in this message) - main website and kamailio's wiki systems - download folders - documentations (for modules and other html tutorials) The sip-router.org server will NOT be affected, so next services will be available: - sip-router.org web site and its wiki - GIT repository - issue tracker system If things are not going as expected, tomorrow we will post news about on this server, at http://sip-router.org This is a good opportunity to thank again to voztele.com for sponsoring the server and taking care of the maintenance work. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 1

Re: [SR-Users] Alias possible for presence...
by Daniel-Constantin Mierla 26 Jun '12

26 Jun '12

Hello, On 6/20/12 9:19 AM, Gertjan Wolzak wrote: > > Well, the account name is not public, but I would like to have the > possibility to have a "private" account for registering and a > "public"account to publish. > > The goal is to have others to be able to send messages to your > "public" id, while you log in with your "private" id. > most of user agent have two fields, one to be used as SIP id and another one for authentication id. > I know, think the next step is to get an communicator that has that > possibility..... > > So I want to be able to publish the "public" account and not the > "private" one. > > For example: > > I register to the presence server with uid gertjan and my password, > but I want to be known to others as newbie. > > So my friend can send an invitation/message to newbie as well as to > gertjan . > > Just like the db_alias which is used in the kamalio for linking sip > accounts to aliases. > For presence, people have to subscribe to a SIP id, in this case it your public ID known by everybody. The easiest will be that the user agent will publish in behalf of the public id. But ultimately, you can replace the values in headers and xml body (see textops/xmlops functions), then do presence handling, after applying the changes to the message (see textopsx module). Cheers, Daniel > Rgds, > > Gertjan > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* dinsdag 19 juni 2012 10:27 > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Cc:* Gertjan Wolzak > *Subject:* Re: [SR-Users] Alias possible for presence... > > Hello, > > On 6/14/12 10:28 AM, Gertjan Wolzak wrote: > > Hello All, > > I have the following challenge. > > We are using kamailio 3.2 and have enabled presence. Which works fine. > > But we would like to be able to work with aliases, so that the > account name and "presentation" name can be different. > > Any Ideas... Or did I miss the documentation on that? > > the account name is public or not? If it is some internal id, > eventually is used for authentication, but for presence the contacts > should know the public id. > > Or maybe you can give some example to understand how you use aliases > in the context of presence... > > Cheers, > Daniel > > > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu > Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw > > > -- > This message was scanned by Foize and is believed to be clean. > Click here to report this message as spam. > <http://filter.foize.com/cgi-bin/learn-msg.cgi?id=> > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

2 2

Kamailio : Listening on multiple ports behind NAT
by Reda Aouad 26 Jun '12

26 Jun '12

Hello, I am trying to implement the following configuration : - Kamailio as a SIP proxy/registrar behind a one-to-one NAT (port number is not modified) listening on ports 5060 and 53 (and more ports in the future) - aliases correctly configured : alias= udp: public_ip:53 alias= udp: public_ip:5060 alias= udp: hostname:53 alias= udp: hostname:5060 - listen directive correctly on private ip address and both ports : listen=udp:private_ip:53 listen=udp:private_ip:5060 - advertised_address=public_ip - record_route_preset("public_ip") is used to announce the public IP address in the RR header - user A : registered on port 5060 - user B : registered on port 53 Suppose user A tries to call user B. The Record-Route header in the INVITE forwarded from Kamailio to user B should contain the port number on which user B is connected (53), to force user B to send future requests to that port number. But I have no method to know which port user B is connected to, and that problem is aggravated when user B has multiple registrations on different port numbers and parralel forking is done. Declaring advertised_port doesn't solve the problem. I cannot force port number 53 in record_route_preset("public_ip:53") since it wouldn't work when user B calls user A. Using the record_route( ) function, Kamailio doesn't use the advertised_address to construct the RR header. Another problem is that the record_route_preset function clears the DID cookie set by the dialog module, which makes Kamailio fallback to SIP elements to match the request to an existing dialog, thus dialog matching becomes slower, and performance is an issue for me. Any suggestions? I know that one solution would be to run Kamailio with a public IP address and no NAT, but unfortunately it's not possible. I suggest that the function record_route( ) takes a public IP address as a parameter, still doing what it does (correct record routing and cookie addition did=xxx and loose route lr=on), but only replacing the private IP address on which Kamailio listens with a public IP address. Or that the record_route( ) function uses the advertised_address to construct the RR header. Thank you RA

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Kamailio v3.3.0 packages
by Daniel-Constantin Mierla 25 Jun '12

25 Jun '12

Hello, I'm summarizing the status of packages for latest release v3.3.0. Debian/Ubuntu packages are ready in the usual APT repository, including nightly builds of the GIT branch 3.3, details at: * http://www.kamailio.org/wiki/packages/debs RPMs for CentOS/RedHat, Fedora, openSuse (various versions for each OS) are now available on opensuse build factory repository, details at: * http://www.kamailio.org/wiki/packages/rpms There you find the note about Peter Dunkley's repository, which has also builds for Raspberry Pi. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

Over 100% of CPU usage on kamailio
by Konstantin M. 25 Jun '12

25 Jun '12

Hello, Is there any reason to consume over 135% of CPU usage on kamailio ? (sreenshot: http://i48.tinypic.com/2u72hr8.png). Any ideas ? Thanks!

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Fwd: Can Kamailio be used to redirect media between a client that switches from wifi to 3g/gsm
by Shaun Clark 25 Jun '12

25 Jun '12

Forgot to post the response to the list as well. Date: Fri, Jun 22, 2012 at 6:57 AM Subject: Re: [SR-Users] Can Kamailio be used to redirect media between a client that switches from wifi to 3g/gsm To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Thanks for the response! I see a series of what I believe are re-REGISTER statements: Message sent: (to dest=75.101.244.XXX:5060) REGISTER sip:75.101.244.XXX SIP/2.0 Via: SIP/2.0/UDP 10.165.27.161:2407;rport;branch=z9hG4bK1839704852 From: <sip:990XX@75.101.244.XXX>;tag=1689684502 To: <sip:990XX@75.101.244.XXX> Call-ID: 1867622191 CSeq: 1 REGISTER Contact: <sip:990XX@10.165.27.161:2407;line=daeb0d9351eff22> Max-Forwards: 70 User-Agent: Linphone/3.4.0 (eXosip2/unknown) Expires: 3600 Content-Length: 0 Received message: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.165.27.161:2407;received=32.158.143.61 ;rport=2407;branch=z9hG4bK1839704852 From: <sip:990XX@75.101.244.XXX>;tag=1689684502 To: <sip:990XX@75.101.244.XXX>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.e10d Call-ID: 1867622191 CSeq: 1 REGISTER WWW-Authenticate: Digest realm="75.101.244.XXX", nonce="4fe476e500000e00aa8700d49b8c668f4c6f1e6a367f2XXX" Server: Kamailio Content-Length: 0 REGISTER sip:75.101.244.XXX SIP/2.0 Via: SIP/2.0/UDP 10.165.27.161:2407;rport;branch=z9hG4bK123406454 From: <sip:990XX@75.101.244.XXX>;tag=1689684502 To: <sip:990XX@75.101.244.XXX> Call-ID: 1867622191 CSeq: 2 REGISTER Contact: <sip:990XX@10.165.27.161:2407;line=daeb0d9351eff22> Authorization: Digest username="990XX", realm="75.101.244.XXX", nonce="4fe476e500000e00aa8700d49b8c668f4c6f1e6a367f2XXX", uri="sip:75.101.244.XXX", response="1e1d558894f2c05c322c76efbb2f9XXX", algorithm=MD5 Max-Forwards: 70 User-Agent: Linphone/3.4.0 (eXosip2/unknown) Expires: 3600 Content-Length: 0 Received message: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.165.27.161:2407;received=32.158.143.61 ;rport=2407;branch=z9hG4bK123406454 From: <sip:990XX@75.101.244.XXX>;tag=1689684502 To: <sip:990XX@75.101.244.XXX>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.8b5a Call-ID: 1867622191 CSeq: 2 REGISTER Contact: <sip:990XX@10.165.27.161:2407;line=daeb0d9351eff22>;expires=120, <sip:990XX@50.43.101.83:51879;line=59ecc207f06f4e9>;expires=81 Server: Kamailio Content-Length: 0 But after this I would expect to see an INVITE but one is never sent, but if I switch back to the original IP on that device the call is reconnected, so it proves we're missing an INVITE I believe. What do I need to do on the server side to force a re-INVITE to be sent after this registration occurs? Thanks! On Fri, Jun 22, 2012 at 1:14 AM, Klaus Darilion < klaus.mailinglists(a)pernau.at> wrote: > Hi Shaun! > > Your problem description is too short to give you any good help. > > Use tcpdump (or other tools) to capture the scenario with Asterisk and > Kamailio. Then compare them to find out why it doesn't work. > > Is media sent directly to Asterisk then it ca not be the problem of > Kamailio. > > I hope the mobile client is smart enough to also send a reINVITE when > getting the new IP address (of the mobile connection) with proper Contact > header - otherwise it can not receive SIP requests from Asterisk. > > regards > Klaus > > > On 20.06.2012 18:07, Shaun Clark wrote: > >> The use case is that I have a SIP client registered to Kamailio talking >> to an Asterisk box connected to the PSTN. The client is a mobile phone >> and the user is connected to wifi. The user then steps out of wifi range >> and the phone drops the connection and picks up the 3g data connection. >> I want the media stream to reconnect to the client and the call to >> resume without having to redial. This works now if the client is >> directly connected to the Asterisk machine, but not when I am routing >> through my Kamailio server. How do I go about this, examples are always >> appreciated, thanks! >> >> >> ______________________________**_________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >> >> > > --

2 1

ERROR: uri2dst: bad_uri
by Barthel Marco (CI/AFU1) 25 Jun '12

25 Jun '12

Hi, we are using kamailio at Bosch with some several thousand users. From time to time we encounter the following in the logs of kamailio: /usr/sbin/kamailio[9247]: ERROR: tm [ut.h:255]: ERROR: uri2dst: bad_uri: sip: /usr/sbin/kamailio[9247]: ERROR: tm [t_fwd.c:1528]: ERROR: t_forward_nonack: failure to add branches Am I right that this means a relay/forward happened and either $ru or $du was just set to "sip:" ? Thanks in advance. Mit freundlichen Grüßen / Best regards Marco Barthel Robert Bosch GmbH (CI/AFU1) www.bosch.com Tel. +49 711 811-3602341 marco.barthel(a)de.bosch.com Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart, HRB 14000; Aufsichtsratsvorsitzender: Hermann Scholl; Geschäftsführung: Franz Fehrenbach, Siegfried Dais; Stefan Asenkerschbaumer, Bernd Bohr, Rudolf Colm, Volkmar Denner, Christoph Kübel, Uwe Raschke, Wolf-Henning Scheider, Werner Struth, Peter Tyroller

2 1

Nathelper module, FLT_NATS, FLT_NATB
by Richard Brady 22 Jun '12

22 Jun '12

Hi folks I've configured Kamailio as a simply multi-homed proxy offering NAT traversa, TCP connection management, etc to FreeSWICH behind it. FreeSWITCH acts as the registrar. I'm trying to understand the default config and the following things don't make sense to me (I can't find documentation on these): 1. The nat=yes flag. What does it mean exactly and why is it on the RR and not on the Contact/R-URI? Also, when double RR is enabled, why does it get added to both headers? 2. The FLT_NATS and FLB_NATB flags. I'd like to know what this stands for, I guess "flag for transaction - NAT" and "flag for branch - NAT" but what do the S and the B stand for? More importantly what do these flags mean? Is it correct to say that the first indicates the UAC / A-leg / requester is behind NAT and the second that the UAS / B-leg / responder is behind NAT? 3. In the the NATMANAGE route of the default config we have: if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } Which to me means "if the message is a request inside an existing dialog ..." and I'm thinking, isn't this a bit late in the dialog to be setting NAT flags? I would expect to set the flag for the A-leg on the incoming request, and the flag for the B-leg on the incoming reply. I can't point in waiting for the ACK to try and work out whether this leg is behind NAT. And also it seem this set of if statements will also match incoming requests from the A-leg, so why are we setting a brach flag? Does that leg also also get the properties of a branch? 4. When I get a 200 OK coming from behind NAT it is not subjected to fix_nated_contact(), and this seems to be because it doesn't have the FLB_NATB flag set: if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } Any help clarifying would be much appreciated. Thanks, Richard -- Richard Brady

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Can Kamailio be used to redirect media between a client that switches from wifi to 3g/gsm
by Shaun Clark 22 Jun '12

22 Jun '12

The use case is that I have a SIP client registered to Kamailio talking to an Asterisk box connected to the PSTN. The client is a mobile phone and the user is connected to wifi. The user then steps out of wifi range and the phone drops the connection and picks up the 3g data connection. I want the media stream to reconnect to the client and the call to resume without having to redial. This works now if the client is directly connected to the Asterisk machine, but not when I am routing through my Kamailio server. How do I go about this, examples are always appreciated, thanks!

2 1

Geographically Distributed Active Sites/Servers
by SamyGo 22 Jun '12

22 Jun '12

Hello, I've been pondering over an architecture of distributed VoIP services on different geographic regions. I'm sure its nothing new and there are lots of guys have ideas to share. There are multiple SIP enabled Regions/locations, each has its own media services and full services capability. Now Imagine I've unique SIP users registering on each Region. What I want to achieve is as follows: 1- User A from Region-A dials User-B registering on Region-C 2- Region-A inquires all Regional-Nodes and in response only one Regional Node (Region-C) raises hand that it has this particular user registered. 3- Upon this reply Region-A Kamailio relays the call to Region-C kamailio and we get a successful call. The SIP users are mobile and can register to different regions depending upon their location so thats why I need step-2. Offline SIP user's will be handled by the destination number area code and routed to the closest serving VoIP region. How can we use the new modules like DMQ or Presence to achieve this !1 Regards, Sammy Go.

1 0

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Charles Chance 22 Jun '12

22 Jun '12

Hi Daniel It is fixed :-) What was the problem? Thanks, Charles Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Hello, can you try with latest master branch -- it should be fixed by: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=1d89d7b… Cheers, Daniel On 5/29/12 5:52 PM, Daniel-Constantin Mierla wrote: Hello, finally had a chance to look over it. I guess it is some race, resulting in a memory space being overwritten. The 200ok for publish with ringing state is processed more or less at the time when 200ok for invite comes and dialog changes state. Can you re-compile with MEMDBG=1 (edit the Makefile.defs or give it in the command line parameter) and try again? Maybe the memory manager can catch the fault operation and provide more details. Cheers, Daniel On 5/17/12 6:58 PM, Charles Chance wrote: Hi Daniel, I have sent the messages over to you. I look forward to hear what you can find. Best, Charles From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: 17 May 2012 15:06 To: Charles Chance Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List' Subject: Re: FW: (Devel) Segmentation fault using pua_dialoginfo Hello, the content of ps->param is pretty much all invalid. So looks like a callback on a freed/invalid parameter, could be a double-callback execution. Set debug=3 and send all the messages for such case -- it might be quite a lot of them, so send them to my email address, the mailing list has a limit of message size. Cheers, Daniel On 5/17/12 2:13 PM, Charles Chance wrote: Hi, Requested output as follows: (gdb) frame 1 #1 publ_cback_func (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) at send_publish.c:246 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); (gdb) p *ps $2 = {req = 0x0, rpl = 0xb7c30340, param = 0xb3dc3ff4, code = 412, flags = 0, branch = 0, t_rbuf = 0x0, dst = 0x0, send_buf = {s = 0x0, len = 0}} (gdb) p *(ua_pres_t*)(*ps->param) $3 = {id = {s = 0x20455942 <Address 0x20455942 out of bounds>, len = 980445555}, pres_uri = 0x40323531, event = 875444279, expires = 875703856, desired_expires = 858861105, flag = 808794676, db_flag = 1394618421, cb_param = 0x322f5049, next = 0xa0d302e, ua_flag = 979462486, etag = { s = 0x50495320 <Address 0x50495320 out of bounds>, len = 808333871}, tuple_id = {s = 0x5044552f <Address 0x5044552f out of bounds>, len = 775436064}, body = 0x322e3034, content_type = { s = 0x312e3134 <Address 0x312e3134 out of bounds>, len = 1648047155}, watcher_uri = 0x636e6172, call_id = { s = 0x397a3d68 <Address 0x397a3d68 out of bounds>, len = 1647593320}, to_tag = {s = 0x3435634b <Address 0x3435634b out of bounds>, len = 842149473}, from_tag = { s = 0x63306461 <Address 0x63306461 out of bounds>, len = 808334648}, cseq = 1767246349, version = 1394621025, outbound_proxy = 0x322f5049, extra_headers = 0x552f302e, record_route = { s = 0x31205044 <Address 0x31205044 out of bounds>, len = 825111097}, remote_contact = {s = 0x312e3836 <Address 0x312e3836 out of bounds>, len = 976498224}, contact = { s = 0x36373135 <Address 0x36373135 out of bounds>, len = 1701985073}} Yes, it is a test server so very happy to arrange for remote access if required. In the meantime, I will do a little more digging to try to find out why entity is null. Cheers, Charles -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2176 / Virus Database: 2425/5004 - Release Date: 05/16/12 -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

2 1

New developer - Vicente Hernando
by Daniel-Constantin Mierla 21 Jun '12

21 Jun '12

Hello, I am introducing Vicente as a new registered developer - he submitted lately very useful patches to ndb_redis module (e.g., array support in replies, redis free function for config), new ones being on the pipe -- he can present better his plans or you just watch the commits. Mainly he will be in charge with ndb_redis module, being a heavy user of it, but there have been submissions from him to other parts of code, we welcome improvements and new good features anywhere! Welcome and looking forward to your contributions! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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$200 bounty
by copycall 21 Jun '12

21 Jun '12

hello, install and configure kamailio on a godaddy load-balanced cloud server and create a vpn with the customer site. the objective is to use the customer's sip provider, which requires a static ip address, with a dhcp connection. please contact me off-list if you are interested. dave(a)copycall.com thank you, dave

10 25

Bug with server originated call
by David 20 Jun '12

20 Jun '12

Hello, I am using the example that came with my kamailio deb file to launch a call server side. I send a dummy INVITE to the phone, when the phone responds 200 OK, I reply with a REFER to send the phone to the desired number. ( ctd.sh.gz ) My problem is that if I don't answer the ctd call, the phone rings for a long time ( or endlessly on certain devices ). I tried adding an Expires: 10, and it did not make any difference. Is there a way to cancel this call from the ctd script ? David

2 2

Kamailio v3.3.0 Released - new major version is out
by Daniel-Constantin Mierla 20 Jun '12

20 Jun '12

Kamailio v3.3.0 is out – a major release with a very large number of new features and improvements. This release is a result of about 6 months of development and 2 months of testing from the teams of Kamailio (OpenSER) and SIP Express Router (SER) projects. Version 3.3.0 comes with 7 brand new modules in addition to a lot of fresh features in core and old modules. You can read detailed release notes at: * http://www.kamailio.org/w/kamailio-v3-3-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy SIP routing in a secure, flexible and easier way with Kamailio v3.3.0! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Re: [SR-Users] simple test case: add user/delete user on kamailio 3.3 presence server
by Daniel-Constantin Mierla 20 Jun '12

20 Jun '12

Hello, On 6/19/12 4:49 PM, Min Wang wrote: > Hi > > Here is another similar case which does not work. maybe it is related > the above issue as well. > > Jitsi 1.1 client: 101/103 , with xcap as storage > proxy: kamailio 3.3 > ip network is: 192.168.122.0 > > the setup is like the following: > > 101(.224) --- proxy (.32) --- 103 (.1) > > After all presence related db are truncated, kamailio was restart. > then bring 101,103 online. > > On 101 add 103, 103 prompt up authorization window, click ok. > now on 103, I can see 101 online status > > but on 101, I can not see 103 status <----the first trace file: > the jitsi-101-add-103-confirm-2.pcap > > (it is grey, even manually change 103 status from online to away > etc, on 101, the 103 always showed grey. <--- second trace file: > jitsi-101-103-change.pcap) > > Please see the attached pcap files. > > > the watcher table shows: > > id presentity_uri watcher_username watcher_domain > event status reason inserted_time > 1 sip:103@192.168.122.32 101 192.168.122.32 presence > 2 1340114579 > 2 sip:101@192.168.122.32 103 192.168.122.32 presence > 1 1340114582 the first record above shows that the status is 2, which means 'pending'. The subscription of 101 was not accepted yet by presentity 103. For the second record, status is 1, which means active subscription. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

2 3

simple test case: add user/delete user on kamailio 3.3 presence server
by Min Wang 20 Jun '12

20 Jun '12

4 7

rtimer question
by Alex Balashov 20 Jun '12

20 Jun '12

Hi, I use 'rtimer' and 'mqueue' to asynchronously defer certain heavy DB tasks on which call processing does not depend. I was hoping to get a clarification on the exact behaviour of 'rtimer' when tasks don't complete as fast as they should. Let's say that I set the rtimer route interval to half a second (500 ms), as afforded by new Kamailio 3.3.0 capabilities. What happens if the tasks I am running inside the rtimer route don't complete in that interval? Is that rtimer thread forceably terminated, or is it allowed to run as long as necessary to complete the task? Is there an absolute timeout? If the latter, I assume that another, parallel rtimer route will be started in the background after 500 ms. I am given to understand that there is a thread pool. How big is it? Can I configure its size anywhere? Is there a practical limit? Finally, is there 100% assurance that 'mqueue' is safe to consume for multiple rtimer processes, possibly ones tripping over each other due to unpredictable delays? Or is it designed only to provide blocking mutex for data passed from a SIP worker thread to a different kind of thread? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 1

kamailio core request dropping - no reply
by Uri Shacked 20 Jun '12

20 Jun '12

Hi, I am testing kamailio replies when an INVITE or another request arrives with lets say, "VIA" header missing.... The core drops the request. But, there is no reply for the originator (so it keep on resending the request...) Why? Uri

4 7

Siremis routing support does not include drouting
by Yufei Tao 20 Jun '12

20 Jun '12

Hello Seems Siremis 3.2.1 doesn't support configuration of drouting. Any plans that it will in the future? I hope to use it to make my drouting configuration easier. What I want to do is to route prefixed numbers to external number at external domains, depending on the prefix (numbers+alphabets+dot and maybe+underscore+hyphen as well). Maybe I should use PDT module instead as it is supported by Siremis? Thanks! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

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Re: [SR-Users] Alias possible for presence...
by Daniel-Constantin Mierla 20 Jun '12

20 Jun '12

Hello, On 6/14/12 10:28 AM, Gertjan Wolzak wrote: > > Hello All, > > I have the following challenge. > > We are using kamailio 3.2 and have enabled presence. Which works fine. > > But we would like to be able to work with aliases, so that the account > name and "presentation" name can be different. > > Any Ideas... Or did I miss the documentation on that? > the account name is public or not? If it is some internal id, eventually is used for authentication, but for presence the contacts should know the public id. Or maybe you can give some example to understand how you use aliases in the context of presence... Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

2 1

Re: [SR-Users] simple test case: add user/delete user on kamailio 3.3 presence server
by Min Wang 19 Jun '12

19 Jun '12

HI Since my previous email has attachment , it is still waiting for the approval. So here is again the text part ( please see the email below) According to the trace, here is the call flow for this case: 101(.224) --- proxy (.32) --- 103 (.1) 101/103 are jitsi 1.1 nightly build client, the configure is to use SIMPLE/xcap on clients. the call flow for 101 adding 103 as contact 101 proxy 103 | | | user-add-103 | | | | | | xcap put pres/rls | | |--------------------------> | | | | | | SUBSCRIBE (presence/103) | | |---------------------------> | | | 202 ok | | |<----------------------------| | | NOTIFY( presenc/pending) | | |<----------------------------| | | | NOTIFY ( winfo/101 pending) | | |---------------------------------->| | 200 ok | | |--------------------------> | | | | 200 OK | | |<----------------------------------| | | auth window pop | | user accept | | SUBSRIBE(presence/101) | | |<----------------------------------| | | 202 OK | | |---------------------------------->| | NOTIFY (winfo,103 active) | | |<-------------------------- | | | 200 ok | | |--------------------------> | | | | NOTIFY (presence/101) | | |---------------------------------> | | | 200 OK | | |<----------------------------------| | NOTIFY (presence/103 open) | | |<-------------------------- | | ****** ( missing part) | 200 ok | | |--------------------------> | | As on the 103, auth windows pop up, then click ok to accept (thus it will added 101 as contact), in this case it does not do any xcap operation? Is it the correct call flow? Thanks min On 06/19/2012 04:49 PM, Min Wang wrote: > Hi > > Here is another similar case which does not work. maybe it is related > the above issue as well. > > Jitsi 1.1 client: 101/103 , with xcap as storage > proxy: kamailio 3.3 > ip network is: 192.168.122.0 > > the setup is like the following: > > 101(.224) --- proxy (.32) --- 103 (.1) > > After all presence related db are truncated, kamailio was restart. > then bring 101,103 online. > > On 101 add 103, 103 prompt up authorization window, click ok. > now on 103, I can see 101 online status > > but on 101, I can not see 103 status <----the first trace file: the > jitsi-101-add-103-confirm-2.pcap > > (it is grey, even manually change 103 status from online to away > etc, on 101, the 103 always showed grey. <--- second trace file: > jitsi-101-103-change.pcap) > > Please see the attached pcap files. > > > the watcher table shows: > > id presentity_uri watcher_username watcher_domain > event status reason inserted_time > 1 sip:103@192.168.122.32 101 192.168.122.32 > presence 2 1340114579 > 2 sip:101@192.168.122.32 103 192.168.122.32 > presence 1 1340114582 > > the active wacher table show: > > id presentity_uri watcher_username watcher_domain > to_user to_domain event event_id to_tag from_tag > callid local_cseq remote_cseq contact record_route > expires status reason version socket_info > local_contact from_user from_domain updated updated_winfo > 1 sip:101@192.168.122.32 101 192.168.122.32 > 101 192.168.122.32 message-summary > a6a1c5f60faecf035a1ae5b6e96e979a-7e2e 8b6ca8e7 > 9e8288d04d0dab18fda28f5fb24b40d8(a)0.0.0.0 1 2 > sip:101@192.168.122.224:5060;transport=udp;registe... > 1340117922 1 1 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 101 192.168.122.32 -1 -1 > 2 sip:101@192.168.122.32 101 192.168.122.32 > 101 192.168.122.32 presence.winfo > a6a1c5f60faecf035a1ae5b6e96e979a-8a99 9975004d > 9b8cb0735bc559232639aa28c16a6f4b(a)0.0.0.0 2 2 > sip:101@192.168.122.224:5060;transport=udp;registe... > 1340117921 1 2 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 101 192.168.122.32 -1 -1 > 3 sip:103@192.168.122.32 103 192.168.122.32 > 103 192.168.122.32 message-summary > a6a1c5f60faecf035a1ae5b6e96e979a-0117 9d873afb > 30fa1ce8cca7a7c50fea11e688ec3d5e@0:0:0:0:0:0:0:0 1 2 > sip:103@192.168.122.1:5060;transport=udp;registeri... > 1340117959 1 1 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 103 192.168.122.32 -1 -1 > 4 sip:103@192.168.122.32 103 192.168.122.32 > 103 192.168.122.32 presence.winfo > a6a1c5f60faecf035a1ae5b6e96e979a-9a71 101d721c > 148be0ec1a4acc087c0083324398cf14@0:0:0:0:0:0:0:0 2 2 > sip:103@192.168.122.1:5060;transport=udp;registeri... > 1340117959 1 2 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 103 192.168.122.32 -1 -1 > 5 sip:103@192.168.122.32 101 192.168.122.32 > 103 192.168.122.32 presence > a6a1c5f60faecf035a1ae5b6e96e979a-5ab7 e9099546 > 6ed81a0fa2042996295974b671a8074f(a)0.0.0.0 1 2 > sip:101@192.168.122.224:5060;transport=udp;registe... > 1340118179 2 1 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 101 192.168.122.32 -1 -1 > 6 sip:101@192.168.122.32 103 192.168.122.32 > 101 192.168.122.32 presence > a6a1c5f60faecf035a1ae5b6e96e979a-98a8 > > > > > thanks > > min > > On 06/19/2012 01:00 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> you have to share the content of the messages sent, providing just >> the url is not enough to see what happens. >> >> watchers table is for the list of contacts and their global status in >> regard to presentity, active_watchers is for keeping the presence >> dialogs. >> >> Cheers, >> Daniel >> >> On 6/19/12 12:47 PM, Min Wang wrote: >>> Hi >>> >>> (1) I tested more. It is due to the difference between bria 3 and >>> jitsi while handling the adding/removing contact: >>> >>> if contact is added/removed: >>> >>> for jitsi, it send out: >>> >>> PUT /xcap-root/resource-lists/users/sip:101@192.168.122.32/index >>> PUT /xcap-root/pres-rules/users/sip:101@192.168.122.32/presrules >>> >>> >>> for bria 3, it only send out >>> >>> PUT >>> /xcap-root/resource-lists/users/101(a)192.168.122.32/contacts-resource-list.xml >>> HTTP/1.0. >>> PUT >>> /xcap-root/resource-lists/users/101(a)192.168.122.32/contacts-resource-list.xml >>> >>> >>> it does NOT send out the pres-rules. >>> >>> Is it the good behavior ( from RFC point of view) ? >>> should I call the pres_update_watchers even though its is for >>> resource-lists? >>> >>> (2) BTW, what is the exact purpose of watcher table? >>> >>> e.g: on 101, if delete 103, watcher table become: >>> >>> id presentity_uri watcher_username >>> watcher_domain event status reason inserted_time >>> 1 sip:103@192.168.122.32 101 192.168.122.32 >>> presence 1 1340096051 >>> 2 sip:101@192.168.122.32 103 192.168.122.32 >>> presence 3 terminated 1340096181 >>> >>> it is item 2 become terminated instead of item 1. If watcher >>> table is for subscription, then seems item 1 should be terminated? >>> Look at the code, it seems somehow serve as mixed role of >>> subscription/auth/xcap purpose, not for subscription state? >>> >>> and is the active_watcher mainly for the state machine of >>> http://tools.ietf.org/html/rfc3857#section-4.7.1? >>> >>> thanks. >>> >>> >>> min >>> >>> >>> >>> >>> >>> >>> >>> >>> On 06/19/2012 10:41 AM, Daniel-Constantin Mierla wrote: >>>> Hello, >>>> >>>> On 6/18/12 2:11 PM, Andreas Granig wrote: >>>>> Hi, >>>>> >>>>> On 06/15/2012 05:25 PM, Min Wang wrote: >>>>>> | 1 | sip:103@192.168.122.32 | 101 | 192.168.122.32 | >>>>>> presence | 2 | | 1339772803 | >>>>>> >>>>>> then I deleted the 103 from the contact list, the watcher >>>>>> table still >>>>>> shows the same. >>>>> For local storage, I'd expect an unsubscribe (subscribe with >>>>> Expires=0) >>>>> from 101 to 103. The obvious work-flow would be to set it to >>>>> "terminated" in watchers table, and in case of a subsequent >>>>> re-subscribe >>>>> it should be changed back to "pending" state, although the >>>>> state-machine >>>>> doesn't indicate a transition from "terminated" back to "pending". >>>>> Isn't >>>>> this the case? >>>>> >>>>> For xcap storage, there are other ways to block/remove a contact >>>>> on/from >>>>> the list. As Iñaki pointed out in >>>>> http://lists.opensips.org/pipermail/devel/2009-August/003868.html, >>>>> the >>>>> xcap server needs to notify the sip server about the change, which in >>>>> turn will notify the other party (103) that it's no longer allowed to >>>>> see 101's state. If the xcap_server module of kamailio is used, >>>>> there is >>>>> the following code snippet in some examples floating around on the >>>>> internet: >>>>> >>>>> >>>>> switch($rm) { >>>>> case "PUT": >>>>> xcaps_put("$var(uri)", "$hu", "$rb"); >>>>> if($xcapuri(u=>auid)=~"pres-rules") { >>>>> pres_update_watchers("$var(uri)", "presence"); >>>>> pres_refresh_watchers("$var(uri)", "presence", 1); >>>>> } >>>>> >>>>> So, shouldn't this update the watchers accordingly? Anyways, also in >>>>> this case the watcher state should change to "terminated", and in >>>>> case >>>>> of a re-subscribe it should go back to pending if xcap rules are >>>>> allowing this. >>>>> >>>>> Maybe someone with good xcap_server/presence insights could >>>>> elaborate on >>>>> that? >>>> to summarize, you say that when the contact is removed from the >>>> buddy list, the watcher table is not updated to state terminated by >>>> pres_update_watchers(...)? >>>> >>>> Cheers, >>>> Daniel >>>> >>> >> >> -- >> Daniel-Constantin Mierla -http://www.asipto.com >> http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu >> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw >

1 0

About the subscribe/200-ok contact address on kamailio 3.3
by Min Wang 19 Jun '12

19 Jun '12

HI I have tried the kamailio 3.3, user 101(10.15.20.131) send subscribe to (10.15.20.137), kamailio return 200 OK, but with contact header as: Contact: <sip:10.15.20.137:5060;transport=udp>. is it correct? Since before subscription expired, the client will send the re-subscribe using that contact address, is it right? Should kamailio return the contact as sip:101@10.15.20.137 in this case? Thanks. min ///////////////////////////// pkt captured //// U 10.15.20.131:54740 -> 10.15.20.137:5060 SUBSCRIBE sip:101@10.15.20.137 SIP/2.0. Via: SIP/2.0/UDP 192.168.122.224:54740;branch=z9hG4bK-d8754z-87f9c58f1c70cbe2-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:101@10.15.20.131:54740>. To: <sip:101@10.15.20.137>. From: "101"<sip:101@10.15.20.137>;tag=4f7671df. Call-ID: MWRjNjU2NGNmMWE0Nzk3ODU2ZTZjZWViYmViYmQ1NmY. CSeq: 2 SUBSCRIBE. Expires: 60. Accept: application/watcherinfo+xml. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Proxy-Authorization: Digest username="101",realm="10.15.20.137",nonce="T9n8u0/Z+4/OOU2Qeo4acn7OE+7pb5A1",uri="sip:101@10.15.20.137",response="93caab1f09daa88a1619bb0613b9a5ff",algorithm=MD5. Supported: eventlist. User-Agent: Bria 3 release 3.4.1 stamp 66953. Event: presence.winfo. Content-Length: 0. . # U 10.15.20.137:5060 -> 10.15.20.131:54740 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.122.224:54740;branch=z9hG4bK-d8754z-87f9c58f1c70cbe2-1---d8754z-;rport=54740;received=10.15.20.131. To: <sip:101@10.15.20.137>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-21e3. From: "101"<sip:101@10.15.20.137>;tag=4f7671df. Call-ID: MWRjNjU2NGNmMWE0Nzk3ODU2ZTZjZWViYmViYmQ1NmY. CSeq: 2 SUBSCRIBE. Expires: 60. Contact: <sip:10.15.20.137:5060;transport=udp>. Server: kamailio (3.3.0-rc0 (x86_64/linux)). Content-Length: 0. . .... U 10.15.20.131:54740 -> 10.15.20.137:5060 SUBSCRIBE sip:10.15.20.137:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.122.224:54740;branch=z9hG4bK-d8754z-59fd0765744f37a3-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:101@10.15.20.131:54740>. To: <sip:101@10.15.20.137>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-21e3. From: "101"<sip:101@10.15.20.137>;tag=4f7671df. Call-ID: MWRjNjU2NGNmMWE0Nzk3ODU2ZTZjZWViYmViYmQ1NmY. CSeq: 8 SUBSCRIBE. Expires: 60. Accept: application/watcherinfo+xml. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Proxy-Authorization: Digest username="101",realm="10.15.20.137",nonce="T9n8u0/Z+4/OOU2Qeo4acn7OE+7pb5A1",uri="sip:10.15.20.137:5060;transport=udp",response="f9ef9d95891bbe5722bb47fbe93b20e9",algorithm=MD5. Supported: eventlist. User-Agent: Bria 3 release 3.4.1 stamp 66953. Event: presence.winfo. Content-Length: 0.

2 1

IP ToS
by Olle E. Johansson 19 Jun '12

19 Jun '12

I know that you can set the TOS for packets sent from Kamailio in the config section. I got the question if you can read the TOS in the IP packet of an incoming response or reply in the config scripts? And can you set it per message/transaction? /O

2 1

install problem
by copycall 19 Jun '12

19 Jun '12

hello, i ran into a problem installing kamailio when i got to #7 "create mysql database": < openser.cfg.5 > /usr/local/share/man//man5/ openser.cfg.5 chmod 644 /usr/local/share/man//man5/openser.cfg.5 sed -e "s#/etc/openser/openser\.cfg#/usr/local/etc/openser/openser.cfg#g" \ -e "s#/usr/sbin/#/usr/local/sbin/#g" \ -e "s#/usr/lib/openser/modules/#/ usr/local/lib64/openser/modules/#g" \ -e "s#/usr/share/doc/openser/#/ usr/local/share/doc/openser/#g" \ < scripts/openserctl.8 > /usr/local/share/man//man8/ openserctl.8 chmod 644 /usr/local/share/man//man8/openserctl.8 sed -e "s#/etc/openser/openser\.cfg#/usr/local/etc/openser/openser.cfg#g" \ -e "s#/usr/sbin/#/usr/local/sbin/#g" \ -e "s#/usr/lib/openser/modules/#/ usr/local/lib64/openser/modules/#g" \ -e "s#/usr/share/doc/openser/#/ usr/local/share/doc/openser/#g" \ < utils/openserunix/openserunix.8 > \ /usr/local/share/man//man8/openserunix.8 chmod 644 /usr/local/share/man//man8/openserunix.8 root@copycall:/usr/local/src/openser-1.3.0/sip-server# echo $PATH /usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin root@copycall:/usr/local/src/openser-1.3.0/sip-server# cd root@copycall:~# /usr/local/etc/kamailio/kamctlrc bash: /usr/local/etc/kamailio/kamctlrc: No such file or directory root@copycall:~# /usr/local/sbin/kamdbctl create bash: /usr/local/sbin/kamdbctl: No such file or directory any solution will be greatly appreciated, please include all the code as kamailio is new to me. thank you in advance, dave

2 1

Releasing v3.3.0 later today
by Daniel-Constantin Mierla 18 Jun '12

18 Jun '12

Hello, quick note about packaging the 3.3.0 today, therefore if anyone has commits to push to GIT branch 3.3 after 12:00GMT, please write first to sr-dev mailing list. Once the announcement about the release is sent out, the commits can be pushed again directly. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

2 1

Sipcapture and Siptrace Compile Error
by Akan 18 Jun '12

18 Jun '12

When compiling kamailio v3.3, I am getting this error from sipcapture and siptrace. I am compiling on solaris 10, 64bit Sun server with gnu tools. CC (gcc) [M sipcapture.so] sipcapture.o In file included from sipcapture.c:76:0: sipcapture.h:48:4: error: unknown type name 'u_int32_t' sipcapture.h:49:4: error: unknown type name 'u_int32_t' gmake[1]: *** [sipcapture.o] Error 1 In file included from siptrace.c:54:0: ../../modules/sipcapture/sipcapture.h:48:4: error: unknown type name 'u_int32_t' ../../modules/sipcapture/sipcapture.h:49:4: error: unknown type name 'u_int32_t' gmake[1]: *** [siptrace.o] Error 1 Thanks Nathaniel

2 4

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Charles Chance 18 Jun '12

18 Jun '12

Hello Daniel, Sorry for the delay! I will get around to testing with the latest master branch sometime over the next couple of days and will report back here. Many thanks, Charles Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Hello, can you try with latest master branch -- it should be fixed by: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=1d89d7b… Cheers, Daniel On 5/29/12 5:52 PM, Daniel-Constantin Mierla wrote: Hello, finally had a chance to look over it. I guess it is some race, resulting in a memory space being overwritten. The 200ok for publish with ringing state is processed more or less at the time when 200ok for invite comes and dialog changes state. Can you re-compile with MEMDBG=1 (edit the Makefile.defs or give it in the command line parameter) and try again? Maybe the memory manager can catch the fault operation and provide more details. Cheers, Daniel On 5/17/12 6:58 PM, Charles Chance wrote: Hi Daniel, I have sent the messages over to you. I look forward to hear what you can find. Best, Charles From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: 17 May 2012 15:06 To: Charles Chance Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List' Subject: Re: FW: (Devel) Segmentation fault using pua_dialoginfo Hello, the content of ps->param is pretty much all invalid. So looks like a callback on a freed/invalid parameter, could be a double-callback execution. Set debug=3 and send all the messages for such case -- it might be quite a lot of them, so send them to my email address, the mailing list has a limit of message size. Cheers, Daniel On 5/17/12 2:13 PM, Charles Chance wrote: Hi, Requested output as follows: (gdb) frame 1 #1 publ_cback_func (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) at send_publish.c:246 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); (gdb) p *ps $2 = {req = 0x0, rpl = 0xb7c30340, param = 0xb3dc3ff4, code = 412, flags = 0, branch = 0, t_rbuf = 0x0, dst = 0x0, send_buf = {s = 0x0, len = 0}} (gdb) p *(ua_pres_t*)(*ps->param) $3 = {id = {s = 0x20455942 <Address 0x20455942 out of bounds>, len = 980445555}, pres_uri = 0x40323531, event = 875444279, expires = 875703856, desired_expires = 858861105, flag = 808794676, db_flag = 1394618421, cb_param = 0x322f5049, next = 0xa0d302e, ua_flag = 979462486, etag = { s = 0x50495320 <Address 0x50495320 out of bounds>, len = 808333871}, tuple_id = {s = 0x5044552f <Address 0x5044552f out of bounds>, len = 775436064}, body = 0x322e3034, content_type = { s = 0x312e3134 <Address 0x312e3134 out of bounds>, len = 1648047155}, watcher_uri = 0x636e6172, call_id = { s = 0x397a3d68 <Address 0x397a3d68 out of bounds>, len = 1647593320}, to_tag = {s = 0x3435634b <Address 0x3435634b out of bounds>, len = 842149473}, from_tag = { s = 0x63306461 <Address 0x63306461 out of bounds>, len = 808334648}, cseq = 1767246349, version = 1394621025, outbound_proxy = 0x322f5049, extra_headers = 0x552f302e, record_route = { s = 0x31205044 <Address 0x31205044 out of bounds>, len = 825111097}, remote_contact = {s = 0x312e3836 <Address 0x312e3836 out of bounds>, len = 976498224}, contact = { s = 0x36373135 <Address 0x36373135 out of bounds>, len = 1701985073}} Yes, it is a test server so very happy to arrange for remote access if required. In the meantime, I will do a little more digging to try to find out why entity is null. Cheers, Charles -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2176 / Virus Database: 2425/5004 - Release Date: 05/16/12 -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

Conditionnal hunting with usrloc?
by Igor Potjevlesch 15 Jun '12

15 Jun '12

Hi all, I'm facing a problem with multiple contact in 'location' table with the same 'username'. The idea is to rewrite R-URI with 1 contact and if there is no answer or SIP 5xx, try with the second. Is that make sense? I use SER0.9.9. Regards, Igor.

3 5

Architecture
by Uriel Rozenbaum 15 Jun '12

15 Jun '12

Hey guys, I'm trying to plan a deployment for a couple thousands of users. As we are facing now some issues for scaling, I'd like to make a modular architecture in order to add a server when needed. I was thinking about N+1 Kamailio Proxies to Register customers and handle the border of the network, 1+1 Redirect servers so internal Network Elements can "find" the right Proxy to terminate a call. Customers would use DNS SRV to decide which Proxy to send a REGISTER and then send INVITES to that same proxy. As we have a private core network addressing, I'm thinking I'll have an Asterisk farm in order to perform media handling. Do you have some recommendation or guidelines for this deployment? Thanks! Uriel

3 3

AMR and AMR-WB support
by Jakson Kalsson 15 Jun '12

15 Jun '12

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the SIPX support these codecs and RFC4867 ? Also, any other Server/PBX which support AMR, AMR-WB recommended are welcome. Best regards,

3 3

add user/delete/add user again on kamailio 3.3 presence server
by Min Wang 15 Jun '12

15 Jun '12

HI I am testing the following case in the kamailio 3.3 presence server, all are in 192.168.122.0 network 101 (bria, ip:.224) ----- kamailio ps( ip:.32) --- 103 ( jitsi, ip:.1) on 101 I added 103 in the contact list, 101 <-> 103 can see each other, Then on 101, I deleted 103, thus on 103, 101 became grey, so far so good. Now I tried to add 103 back to contact list, on 101 I can see 103's status (online), but on jitsi 103, I can not see 101's status. ( grey) The watcher table showed that status terminated. +----+------------------------+------------------+----------------+----------+--------+------------+---------------+ | id | presentity_uri | watcher_username | watcher_domain | event | status | reason | inserted_time | +----+------------------------+------------------+----------------+----------+--------+------------+---------------+ | 1 | sip:103@192.168.122.32 | 101 | 192.168.122.32 | presence | 1 | | 1339752028 | | 2 | sip:101@192.168.122.32 | 103 | 192.168.122.32 | presence | 3 | terminated | 1339752031 | +----+------------------------+------------------+----------------+----------+--------+------------+---------------+ I guess it should be status=1 in this case, but somehow kamailio seems does not update the table? I have a complete packet-trace file if needed. thanks. min

1 0

drouting module: can I use non-digit prefix?
by Yufei Tao 14 Jun '12

14 Jun '12

Hi I'm using drouting module and seems that it only allows digit-only prefixes. Also all the number of digits to be stripped ('strip' defined in dr_gateways) must be all digits too. When I used non-digits in these, Kamailio crashed. Is there any way in drouting module that I can use prefixes with alphabet and dots, e.g. "gateway1."? Thanks very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

2 7

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Daniel-Constantin Mierla 14 Jun '12

14 Jun '12

Hello, finally had a chance to look over it. I guess it is some race, resulting in a memory space being overwritten. The 200ok for publish with ringing state is processed more or less at the time when 200ok for invite comes and dialog changes state. Can you re-compile with MEMDBG=1 (edit the Makefile.defs or give it in the command line parameter) and try again? Maybe the memory manager can catch the fault operation and provide more details. Cheers, Daniel On 5/17/12 6:58 PM, Charles Chance wrote: > > Hi Daniel, > > I have sent the messages over to you. > > I look forward to hear what you can find. > > Best, > > Charles > > ------------------------------------------------------------------------ > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* 17 May 2012 15:06 > *To:* Charles Chance > *Cc:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) > -Users Mailing List' > *Subject:* Re: FW: (Devel) Segmentation fault using pua_dialoginfo > > Hello, > > the content of ps->param is pretty much all invalid. So looks like a > callback on a freed/invalid parameter, could be a double-callback > execution. > > Set debug=3 and send all the messages for such case -- it might be > quite a lot of them, so send them to my email address, the mailing > list has a limit of message size. > > Cheers, > Daniel > > On 5/17/12 2:13 PM, Charles Chance wrote: > >> Hi, >> >> Requested output as follows: >> >> (gdb) frame 1 >> >> #1 publ_cback_func (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) >> >> at send_publish.c:246 >> >> 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); >> >> (gdb) p *ps >> >> $2 = {req = 0x0, rpl = 0xb7c30340, param = 0xb3dc3ff4, code = 412, >> flags = 0, >> >> branch = 0, t_rbuf = 0x0, dst = 0x0, send_buf = {s = 0x0, len = 0}} >> >> (gdb) p *(ua_pres_t*)(*ps->param) >> >> $3 = {id = {s = 0x20455942 <Address 0x20455942 out of bounds>, >> >> len = 980445555}, pres_uri = 0x40323531, event = 875444279, >> >> expires = 875703856, desired_expires = 858861105, flag = 808794676, >> >> db_flag = 1394618421, cb_param = 0x322f5049, next = 0xa0d302e, >> >> ua_flag = 979462486, etag = { >> >> s = 0x50495320 <Address 0x50495320 out of bounds>, len = 808333871}, >> >> tuple_id = {s = 0x5044552f <Address 0x5044552f out of bounds>, >> >> len = 775436064}, body = 0x322e3034, content_type = { >> >> s = 0x312e3134 <Address 0x312e3134 out of bounds>, len = 1648047155}, >> >> watcher_uri = 0x636e6172, call_id = { >> >> s = 0x397a3d68 <Address 0x397a3d68 out of bounds>, len = 1647593320}, >> >> to_tag = {s = 0x3435634b <Address 0x3435634b out of bounds>, >> >> len = 842149473}, from_tag = { >> >> s = 0x63306461 <Address 0x63306461 out of bounds>, len = 808334648}, >> >> cseq = 1767246349, version = 1394621025, outbound_proxy = 0x322f5049, >> >> extra_headers = 0x552f302e, record_route = { >> >> s = 0x31205044 <Address 0x31205044 out of bounds>, len = 825111097}, >> >> remote_contact = {s = 0x312e3836 <Address 0x312e3836 out of bounds>, >> >> len = 976498224}, contact = { >> >> s = 0x36373135 <Address 0x36373135 out of bounds>, len = 1701985073}} >> >> Yes, it is a test server so very happy to arrange for remote access >> if required. In the meantime, I will do a little more digging to try >> to find out why entity is null. >> >> Cheers, >> >> Charles >> > > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > > ------------------------------------------------------------------------ > > No virus found in this message. > Checked by AVG - www.avg.com <http://www.avg.com> > Version: 2012.0.2176 / Virus Database: 2425/5004 - Release Date: 05/16/12 > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu

1 1

Making dialog module work for me !!
by SamyGo 14 Jun '12

14 Jun '12

Hello I've two problems regarding dialog module- used for profiling users that are in-a-call. My topology is quiet simple. SIP-USER-71141---->Kamailio------------->Asterisk[AGI]|| CALLEE<-----------------Kamailio<--------------------------------|| 1- I wish to store my custom SIP header value in profiles. I'm having trouble in figuring out on how to use Dialog_vars table in my kamailio server. I'm using dialog module to profile SIP users that are in call or making a call. It works well and the dialog maintains an entry when the call is established. Now, the difficult part is that I want my custom header to be inserted into either dialog table or dialog_vars table such that when I do a query "*is_in_profile("$avp(my_value)")*" it returns me a result. 2- Alternative to 1, I can log A,B leg separately but it doesn't erase all data on call hangup. Asterisk AGI changes the caller-id of SIP-USERA such that if it was 71141 it would be something like "007" on its way out to destination via kamailio. Now, at this point in kamailio when the call is answered by destination the dialog module saves the call data with values corresponding to "007" and not "71141". I have to disable the "dialog", "detect_spirals" in order to count this one call as two separate entities BUT problem occurs when on hangup the B-Leg gets erased from the table but the A-leg remains in DB for as long as infinity. Please help me in this. Let me know if more details are required from me. Regards, Sammy G.

2 9

Kamailio debian repository changes
by Jon Bonilla 14 Jun '12

14 Jun '12

Hi All With the upcoming release of version 3.3.0, we have updated our kamailio debian repositories. * Debian 5.0 "lenny" will no longer be supported. This means that no more nightly builds for this distribution will be triggered. We'll provide builds for stable releases of 3.0 ,3.1 and 3.2 branches though. * Ubuntu 12.04 "Precise" is now supported for nightly builds and 3.2+ versions. Next time a 3.2.x version is tagged we'll provide the build. Meanwhile you can download the nightly build of 3.2 branch for this distro. * There will be no more nightly builds of 3.1 branch. We'll provide stable builds in case another version is released though. * We now support nightly builds of branch 3.3 (same as 3.2 and master). You can test latest git versions of branch 3.3 until a stable release is built using these repositories: deb http://deb.kamailio.org/kamailio33-nightly squeeze main deb http://deb.kamailio.org/kamailio33-nightly wheezy main deb http://deb.kamailio.org/kamailio33-nightly lucid main deb http://deb.kamailio.org/kamailio33-nightly precise main All the information and repositories can be found in kamailio wiki as usual: http://www.kamailio.org/dokuwiki/doku.php/packages:debs http://www.kamailio.org/wiki/packages/debs

1 0

Dispatcher alg:7 not working accordingly
by SamyGo 14 Jun '12

14 Jun '12

Hello, I'm having trouble using algorithm 7 in dispatcher module. Here is my kamailio version. The problem is that despite multiple dispatcher entries corresponding to one set all calls go only to only one destination. [root@control1 ~]# kamailio -V version: kamailio 3.2.3 (x86_64/linux) 59f87e flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, 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, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 59f87e compiled on 18:04:08 Apr 19 2012 with gcc 4.1.2 [root@control1 ~]# Here is the dipacther module params. # ------- Load-balancer params ------ modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:proxy@109.XXX.2XX.XX") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") modparam("dispatcher", "hash_pvar", "$avp(i:274)") modparam("dispatcher", "ds_hash_size", 9) modparam("dispatcher", "ds_hash_initexpire", 60) Below is the function call based on avp(i:274) avp_db_query("select uid from uid_did_map where accountcode = '$fU'","$avp(i:274)"); xlog("L_INFO","$rm from $fu (IP:$si:$sp) --------- UID:'$avp(i:274)-----------'\n"); ds_load_update(); if(!ds_select_dst("1","7")) { sl_send_reply("500", "No destination available"); xlog("route[MAIN] : $rm : No destinations available for $rd"); exit; } I'm getting two different values for avp(i:274) {26,29} but the destination are not hashed on its base. Please help. Regards, Sammy G.

3 15

Alias possible for presence...
by Gertjan Wolzak 14 Jun '12

14 Jun '12

Hello All, I have the following challenge. We are using kamailio 3.2 and have enabled presence. Which works fine. But we would like to be able to work with aliases, so that the account name and "presentation" name can be different. Any Ideas. Or did I miss the documentation on that? Regards, Gertjan Wolzak

1 0

Lua script calling rewriteuri
by Sharif Tanvir Rahman 14 Jun '12

14 Jun '12

Hello, We plan to use Kamailio as redirect server. Corresponding contact list is to be fetched by a Lua script from DB. The problem is about calling the required 'rewriteuri' function: 1. From lua script. both 'sr.rewriteuri()' and 'sr.modf("rewriteuri", ...)" fails to write the 'contact' field. 2. I tried an alternative way; pushing the value from lua (e.g. sr.pv.sets("contact", "sip:...")) and got the value in the Kamailio.cfg ($var(contact)). But can't call the rewriteuri function with this variable's value as argument. This seems to be a general problem of calling a function from .cfg with a variable's value as parameter. Any help much appreciated. -- Sharif

2 3

topoh - capturing encrypted values
by Alex Balashov 13 Jun '12

13 Jun '12

Given the transparency with which 'topoh' operates, is there any way to capture the new, fake values that it has substituted? In other words, can I, for example, xlog the new Call-ID produced by it? Or does that all take place on a level of abstraction that cannot be intercepted in script? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 1

About xcap xuid in kamailio
by Min Wang 13 Jun '12

13 Jun '12

Hi I am trying to use kamailio xcap server, saw the sample code: if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; why should the xuid have sip:xx@domain.com format? The xcap standard seems does not enforce the format on the xui part? http://xcap.example.com/root/resource-lists/users/joe |-------------v-------------||------v------||---v---| document root AUID XUI thanks. min

4 14

AVPs - how many values on stack
by Olle E. Johansson 13 Jun '12

13 Jun '12

I got this question today that I can't find the answer on: How many values are there in an avp? If I have pushed X values into the AVP - is there a function that counts them for me? /O

2 1

release of v3.3.0 scheduled for Monday, June 18, 2012
by Daniel-Constantin Mierla 13 Jun '12

13 Jun '12

Hello, as branch 3.3 was created and several web pages updated/created already for v3.3.0, the plan is to make the full release next Monday, on June 18, 2012. Packaging specs look very good thanks to Jon and Peter. The few issues left open should be worked meanwhile. For other opinions, reply with your comments here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

About sdpops
by Grégoire Vandendeurpel 13 Jun '12

13 Jun '12

Hello, I'm using Kamailio 3.2.3 and SDPOPS module. I just have a question about the configuration: What do you mean by "ANY_ROUTE" in configuration ?? Cause I don't know where in the configuration , I have to use the differents functions of SDPOPS. If I do something like this: request_route { if ($fU == "103") { sdp_remove_codecs_by_id("0,3,8"); #sdp_remove_media("audio"); } It does not work . So please help me about where I have to put this in?? Thanks in advance -- *Grégoire Vandendeurpel, * * * *IT Sector*

2 8

sipcapture logging
by Konstantin M. 13 Jun '12

13 Jun '12

Hello, Is it possible to log SIP sessions using sipcapture module to log files by log format instead of database ? Thanks!

3 6

Re: [SR-Users] Kamailio Unexpectedly Terminating
by Nathaniel L. Keeling III 12 Jun '12

12 Jun '12

Yes, This is a 64bit system T-Mobile. America’s First Nationwide 4G Network Jason Penton <jason.penton(a)smilecoms.com> wrote: >from the logs he has sent the mem addresses all look 64 bit. Sparc AFAIK is >only 64 bit anyway > >Cheers >Jason > >On Tue, Jun 12, 2012 at 8:50 AM, Daniel-Constantin Mierla <miconda(a)gmail.com >> wrote: > >> Hello, >> >> is it 64bit architecture? >> >> Cheers, >> Daniel >> >> >> On 6/11/12 9:12 PM, Akan wrote: >> >>> Here is the information requested: >>> >>> System = SunOS >>> Node = -f >>> Release = 5.10 >>> KernelID = Generic_141444-09 >>> Machine = sun4u >>> BusType = <unknown> >>> Serial = <unknown> >>> Users = <unknown> >>> OEM# = 0 >>> Origin# = 1 >>> NumCPU = 1 >>> >>> One machine is a Sun Fire V120, 2g memory, UltraSPARC-IIe 650MHz, >>> UltraAX-i2 >>> The other machine is a Sun Netra T1 200, 1g memory, UltraSparc-IIe >>> 500MHz, UltraAX-i2 >>> >>> Thanks >>> >>> Nathaniel L Keeling >>> >>> >>> >>> On 6/11/2012 8:33 AM, Daniel-Constantin Mierla wrote: >>> >>>> Hello, >>>> >>>> I committed a patch that should make it work when realm_prefix is not >>>> set for registrar module: >>>> >>>> http://git.sip-router.org/cgi-**bin/gitweb.cgi/sip-router/?a=**commit;h= >>>> **d42379da90f2ec87cb5dbb00ebb563**c7528ec910<http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=d42379d…> >>>> >>>> For the moment I have no idea why SIGBUS is triggered, i tried to access >>>> the structure directly, not via defines, and there is still a problem when >>>> realm_prefix is set. It's ok accessing the .len field but not .s of the >>>> same structure. The field .s is the first, making me think it's not >>>> alignment issue, or it is something very specific to this system. >>>> >>>> Overall, I am not familiar with sparc/solaris systems, maybe if you send >>>> here full details about OS version, architecture and CPU type, others using >>>> similar systems can give hints. >>>> >>>> Jason reported that on solaris with x86, this problem was not >>>> encountered. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> On 6/4/12 8:57 AM, Daniel-Constantin Mierla wrote: >>>> >>>>> Hello, >>>>> >>>>> still strange with the crash on that if condition. One of your previous >>>>> reports showed a different line, when assigning a pointer. Also, I kind of >>>>> understood that save() was ok, but lookup() not. Is not the case anymore as >>>>> the backtrace shows. >>>>> >>>>> SIGBUS can occur because of alignment in memory which is thrown in >>>>> solaris, does not happen in linux flavours. I am thinking of putting the >>>>> prefix field first so the access to it is at beginning of the straucture, >>>>> just to see if that is the problem, although is should not be. >>>>> >>>>> I will add a patch soon, otherwise I am out of ideas for the moment >>>>> with the SIGBUS, I would need access to the machine to do some testing by >>>>> myself. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>> On 6/2/12 10:36 PM, Akan wrote: >>>>> >>>>>> Ok, I reloaded the servers with v3.2 from git without performing the >>>>>> checkout and with just the master branch. Reran my tests on 2 servers and >>>>>> Kamailio terminated with a core dump. I have included the full trace of one >>>>>> of the servers. The other trace has the same results. >>>>>> >>>>>> Thanks >>>>>> >>>>>> Nathaniel >>>>>> >>>>>> On 5/30/2012 2:25 PM, Daniel-Constantin Mierla wrote: >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> On 5/30/12 9:10 PM, Akan wrote: >>>>>>> >>>>>>>> This was a re-pull from the git master. I had deleted all of the old >>>>>>>> files and issued these commands: >>>>>>>> >>>>>>>> git clone --depth 1 git://git.sip-router.org/sip-**router<http://git.sip-router.org/sip-router>kamailio >>>>>>>> cd kamailio >>>>>>>> git checkout -b 3.2 origin/3.2 >>>>>>>> >>>>>>>> This is the source that I got. Could this be a problem within git? >>>>>>>> >>>>>>> do not checkout branch 3.2, just try with the master branch, as said, >>>>>>> not all patches are backported to stable 3.2. >>>>>>> >>>>>>> For sake of completeness, here are some guidelines: >>>>>>> http://www.kamailio.org/wiki/**install/devel/git<http://www.kamailio.org/wiki/install/devel/git> >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> >>>>>> >>>>>> ______________________________**_________________ >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>>>> sr-users(a)lists.sip-router.org >>>>>> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>>>>> >>>>> >>>>> -- >>>>> Daniel-Constantin Mierla -http://www.asipto.com >>>>> http://twitter.com/#!/miconda -http://www.linkedin.com/in/**miconda<http://www.linkedin.com/in/miconda> >>>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - >>>>> http://asipto.com/u/katu >>>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - >>>>> http://asipto.com/u/kpw >>>>> >>>>> >>>>> >>>> -- >>>> Daniel-Constantin Mierla -http://www.asipto.com >>>> http://twitter.com/#!/miconda -http://www.linkedin.com/in/**miconda<http://www.linkedin.com/in/miconda> >>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - >>>> http://asipto.com/u/katu >>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - >>>> http://asipto.com/u/kpw >>>> >>>> >>>> No virus found in this message. >>>> Checked by AVG - www.avg.com <http://www.avg.com> >>>> Version: 2012.0.2178 / Virus Database: 2433/5061 - Release Date: 06/10/12 >>>> >>>> >>> >>> >>> ______________________________**_________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.com >> http://twitter.com/#!/miconda - http://www.linkedin.com/in/**miconda<http://www.linkedin.com/in/miconda> >> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - >> http://asipto.com/u/katu >> >> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - >> http://asipto.com/u/kpw >> >> >> >> >> ______________________________**_________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >> > >This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer > >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

High availability design with Kamailio
by Reinhard, Carel (GD) 12 Jun '12

12 Jun '12

Dear all, hope to get some information from you guys! For a redundant service we have to build up SIP proxy/registrar server in a high available scenario. Therefore, we will have 2 SIP proxy/registrar on two completely independent servers. They should be SIP dialog stateful and replicate all the session/call states between each other. If one server fails, the other should have all the SIP dialog information for call handling and accounting. How can this done in a stable and reliable way? Is this feature mature enough to support enterprise requirements? The only information I have is to build it up with the DISPATCHER module, but it seems more to be a stateless load-balancer instead of a HA module. Who has experience with HA failover designs with Kamailio and can give me some hints? Thank you very much in advance Kind regards Carel Carel Reinhard Security and VoIP Engineer SRG SSR Switzerland Telefon direkt +41 31 350 94 07 Carel.Reinhard(a)srgssr.ch<mailto:Carel.Reinhard@srgssr.ch>

5 10

Shared memory fragmentation
by Morten Isaksen 12 Jun '12

12 Jun '12

Hi, We have 2 Kamailio 3.0.3 servers that has been running with carrierroute for about 2 years without any problems. They have 128 MB shared memory and modparam("carrierroute", "config_source", "db") modparam("carrierroute", "db_url", "<DBURL>") modparam("carrierroute", "fetch_rows", 500) The carrierroute table is about 91K lines and have been growing slowly. Suddenly we get this ERROR: carrierroute [cr_data.c:585]: could not allocate shared memory from available pool after a few "kamctl cr reload". I increased the shared memory to 256 MB but with the same result. I have now increased it to 512 MB and it seems to work better now. I have noticed this. After a restart the shmem counters is like this: shmem:total_size = 536870912 shmem:used_size = 28486752 shmem:real_used_size = 40147128 shmem:max_used_size = 41135424 shmem:free_size = 496723784 shmem:fragments = 555 And after the first "kamctl cr reload" it is like this: shmem:total_size = 536870912 shmem:used_size = 28619016 shmem:real_used_size = 51842768 shmem:max_used_size = 76993616 shmem:free_size = 485028144 shmem:fragments = 722063 Notice the increase in fragments. Sequentials "kamctl cr reload" does not change the fragments allot. Any ideas? -- Morten Isaksen

3 4

maddr parameter in contact header and fix_nated_contact
by Morten Isaksen 12 Jun '12

12 Jun '12

Hi, I have this setup: Mediation server with internal ip -- NAT firewall -- Kamailio proxy -- PSTN gateway The mediation server sends a contact header with maddr=<internal IP> fix_nated_contact only changes the host part and leaves the maddr intact. When the PSTN gateway sends a PRACK or a BYE it puts the <internal IP> in the RURI, and kamailio then try to forward the message directly to the <internal IP> and not the NAT ip. >From RFC 3261 19.1.1: The maddr parameter indicates the server address to be contacted for this user, overriding any address derived from the host field. When an maddr parameter is present, the port and transport components of the URI apply to the address indicated in the maddr parameter value. [4] describes the proper interpretation of the transport, maddr, and hostport in order to obtain the destination address, port, and transport for sending a request. I can see that OpenSIPS has fixed this. http://sourceforge.net/tracker/?func=detail&atid=1086410&aid=3312423&group_… Would it be possible to get this patch into Kamailio also? -- Morten Isaksen

2 5

Install Kamailio v3.3.x from GIT
by Daniel-Constantin Mierla 12 Jun '12

12 Jun '12

Hello, now that the branch 3.3 is available, I created the tutorial about how to install Kamailio v3.3.x from GIT: * http://www.kamailio.org/wiki/install/3.3.x/git Feedback and improvements are welcome! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

git branch 3.3 created
by Daniel-Constantin Mierla 11 Jun '12

11 Jun '12

Hello, GIT branch 3.3 was created to hold the release series 3.3.x -- version 3.3.0 will be out in max. 1 week, a decision after collecting eventual issues still open to be reported today. Here are few details about how to use it. Download and use branch 3.3 from git: git clone --depth 1 git://git.sip-router.org/sip-router kamailio cd kamailio git checkout -b 3.3 origin/3.3 Pushing patches: - only bug fixes or documentation improvements have to be committed to branch 3.3 - first commit to master branch (if the bug applies there as well), cherry-pick the commit to branch 3.3 and push to remote repository Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

development for future version 3.4.0 is open
by Daniel-Constantin Mierla 11 Jun '12

11 Jun '12

Hello, short note to let everyone know that new features can be committed again to the master branch -- they will be part of future version 3.4.0 (if no other numbering release will be decided). Testing phase of version 3.3.0 within master is ended, a dedicated branch being created for it (details in a separate message). Happy coding, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

Capturing a users call (SIP+RTP+T.38)
by Klaus Darilion 11 Jun '12

11 Jun '12

Hi all! Do you know any comfortable tools to filter out a certain users call? e.g. searching in SIP packets for the user pattern, get the media port out of SDP and capture also RTP+T.38 I found pcapsipdump but have not tried it yet. Any suggestions or do I have to write my own tool? thanks Klaus

4 6

branching 3.3 - to be used for 3.3.x releases
by Daniel-Constantin Mierla 11 Jun '12

11 Jun '12

Hello, I am going to create the GIT branch 3.3 in short time -- this branch will be used for releasing 3.3.x versions. After that master can take again commits of new features, to be part of next major release 3.4.x. Branch 3.3 will take only fixes. From my point of view, I have two issues that I am trying to solve and the 3.3.0 is ready to go out, namely: - a solaris sigbus - pua_dialoginfo issue reported on mailing list Otherwise, nothing else critcal is out there, to my knowledge. Report them if you have other ones. Packaging specs were updated, so we are pretty much ready to release when no relevant issues is open. The wiki page with what is new in this release was updated a bit as well, add if you find something missing at: - http://www.kamailio.org/wiki/features/new-in-devel Also, contribute to the upgrade guidelines: - http://www.kamailio.org/wiki/install/upgrade/3.2.x-to-3.3.0 Cheer, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

2 2

About kamailio 3.3 msrp module and sylkserver
by Min Wang 11 Jun '12

11 Jun '12

Hi I am wondering what are the major differences between kamailio 3.3 msrp module and sylkserver (http://sylkserver.com/features.phtml ) ? thanks. min

2 2

Dispatcher PSTN
by Kr0m 11 Jun '12

11 Jun '12

Hello I am not able to dial to pstn with kamailio, the call is routed to my pstn-gw(asterisk), but the final phone rings 4 or 5 seconds and then it is hanged up. My outbound route is: route[PSTN] { if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } route(TOASTERISK); exit; return; } route[TOASTERISK] { sl_send_reply("100","Trying"); uac_replace_from("$fn","sip:$fn@$fd"); route(NATMANAGE); ds_select_dst("1","4"); t_on_failure("1"); t_relay(); } failure_route[1] { ds_mark_dst("i"); if (!ds_next_dst()) { t_reply("503", "Service unavailable: no more dst"); exit; } route(TOASTERISK); } With a traffic capture i can see the traffic returning to my kamailio server. Any suggestion will be appreciated.

2 1

Kamailio and Nonce Count
by patrice.bodeven＠orange.com 11 Jun '12

11 Jun '12

Hello, I try to understand the nonce count handling. By parameter "modparam("auth", "nonce_expire", 21600)" I can define the validity of the Nonce in the time. But I want also to define the validity of the Nonce in the number of use. For exemple, I want to limit the reuse of the nonce at 64 times (including register, Invite) But I can see the definition of the nonce_count for all the SIP users on Kamailio and not user per user. If I understand when I use modparam("auth", "nonce_count", 1) # enable nonce_count support modparam("auth", "nc_array_order", 20) # 1M in-flight nonces, using 1Mb memory modparam("auth", "nid_pool_no", 4) I define a memory to store the nonce values for all the SIP users. So when User A use nonce x for REGISTER and nonce y for INVITE, nonce x is still valid ! My understanding is it correct ? Is anyway to define a nonce validity in number of times per user ? Patrice B _________________________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, France Telecom - Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, France Telecom - Orange is not liable for messages that have been modified, changed or falsified. Thank you.

2 1

Kamailio 3.2 + RTP proxy - no errors - no media
by Lukas Lani 10 Jun '12

10 Jun '12

Hi, I am trying to use RTPproxy 1.2.1 in bridge mode between IPv4 and IPv6 together with Kamailio 3.2. I configured and ran it on server together with Kamailio. I start it with parameters: -l 192.168.6.2 -6 /2001:6666:6666:6666::2 -s udp:localhost:7722 SIP messages are correctly exchanged, SDP looks to be translated, but when RTP session starts, I see RTP flow only from UC1 to RTPproxy and nothing more. On RTPproxy server I see 4 open ports for IPv4 address but no one for IPv6 address. When I tried to change start parameters to -6 2001:6666:6666:6666::2 -l /192.168.6.2 -s udp:localhost:7722 I could see 4 open ports for IPv6 address but no one for IPv4 address. Can anybody help? do you know about any known error? can it be related to Kamailio or RTPproxy? thank you, Lukas Lani

2 2

Looking for RTP Proxy in TCP
by Austin Einter 08 Jun '12

08 Jun '12

Hi All Now I am using Kamailio 3.1.5 and RTP proxy 1.1. Looks both are compatible and working fine. The RTP Proxy basically sends/receives RTP packets over UDP. Is there any RTP Proxy available that does send/receive of RTP packets over TCP and also should be compatible with Kamailio 3.1.5. If you have any information in this regard, kindly share. Thanks a lot. Austin

6 7

lcr from_uri not matching
by Ben WIlliams 08 Jun '12

08 Jun '12

Has anyone experienced this problem where lcr only works when from_uri is null. When I change it to .* the match fails. Thanks Ben

3 8

Re: [SR-Users] Issue using Kamailio as a proxy in front of Asterisk
by Daniel-Constantin Mierla 07 Jun '12

07 Jun '12

Do not write private emails, keep the mailing list cc-ed for the topics started there. Cheers, Daniel On 6/7/12 12:22 PM, Dominik Mauritz wrote: > Daniel, Stoyna, > > the solutions you described look very interesting to me. I will have a > look at both and pick the one that fits best. > > Thanks guys. > > Rgs, > Dominik > > > > Am 07.06.12 10:57, schrieb Daniel-Constantin Mierla: >> Hello, >> >> On 6/6/12 9:57 PM, Stoyan Mihaylov wrote: >>> We use Jitsi as SIP client, and openxcap along with camailio to >>> handle presence. Then jitsi know if account is online or offline. >>> Our Asterisk dont know nothing about accounts (it accepts all calls >>> from kamailio). There I run AGI scripts, which can check kamailio >>> tables - and I can know if account is online or offline. Of course >>> this do not work if account is "forced" to offline. >>> But may be there is better solution. >>> >>> On Wed, Jun 6, 2012 at 10:04 PM, Dominik Mauritz >>> <dominik.mauritz(a)web.de <mailto:dominik.mauritz@web.de>> wrote: >>> >>> I have already tried that. I defined SIP-Accounts in Asterisk >>> with host=<Kamailio-IP> (instead of host=dynamic). This solves the >>> described problem but it also has side effects: >>> >>> - You don't have the correct presence status on your phone (e. >>> g. xlite) indicating wether the account is online or offline >>> - Asterisk always sends invites to Kamailio on incoming calls >>> even if there is no phone registered to the account >>> >>> These are not the worst things in the world, but it is maybe not >>> the best solution possible. >>> >>> >>> >>> Am 06.06.12 20:41, schrieb Stoyan Mihaylov: >>> >>> We use also Kamailio in front of Asterisk - but I forward >>> only calls to Asterisk - register/unregister stay in Kamailio. >>> Asterisk dont know which device is registered, and which is not. >>> >>> >>> On Wed, Jun 6, 2012 at 8:20 PM, Dominik Mauritz >>> <dominik.mauritz(a)web.de <mailto:dominik.mauritz@web.de> >>> <mailto:dominik.mauritz@web.de <mailto:dominik.mauritz@web.de>>> wrote: >>> >>> Hi All, >>> >>> some days ago I installed Kamailio as a front end for >>> Asterisk following this tutorial: >>> >>> http://kb.asipto.com/asterisk:__realtime:kamailio-3.1.x-__asterisk-1.6.2-as… >>> <http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb> >>> >>> >>> >>> I added Multi Domain support and adjusted some other things >>> to fit my environment. Almost everything is working perfectly now. >>> One small issue is left: >>> >>> With Kamailio in front of Asterisk I have one feature (next >>> to other cool things) I was missing for years. I am now able to >>> register more than one device on the same SIP account. This is >>> nothing new for Kamailio users but Asterisk doesn’t support this. >>> >>> If I register two phones on one SIP account with Kamailio >>> everything is fine. They are able to make outgoing calls and both >>> ring on incoming calls. But if one phone unregisters Kamailio >>> forwards the unregister request to Asterisk and Asterisk sets this >>> account to offline. Now a call comes in but Asterisk is not sending >>> out an invite because for Asterisk the phone is offline. >>> >>> I wonder if it is possible to forward an unregister request >>> from Kamailio to Asterisk only if the last endpoint registered with >>> one SIP account unregisters. If there is more than one endpoint >>> using the same Account Kamailio should not forward the unregister >>> request to Asterisk. >>> >>> Any idea? >>> >> you can use reg_fetch_contacts(...) in your config to find out how >> many contacts are for an user and based on that do not send the >> un-register to asterisk, see: >> >> http://kamailio.org/docs/modules/stable/modules_k/registrar.html#id2498205 >> >> http://kamailio.org/docs/modules/stable/modules_k/registrar.html#id2498441 >> >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla -http://www.asipto.com >> http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 >> -http://asipto.com/u/katu >> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 >> -http://asipto.com/u/kpw >> >> >> > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

2 1

record calls audio and video
by Saul Waizer 07 Jun '12

07 Jun '12

Hello List, I am looking for a way to record the audio and video of a call. I am using Kamailio (latest) on Ubuntu. Does anyone know if there's any existing modules that allow for recording or any other solution that can provide recording of calls on Kamailio? Thank you!

4 6

Siremis v3.2.1 Released
by Elena-Ramona Modroiu 07 Jun '12

07 Jun '12

Hi, Siremis v3.2.1 is out, the web management interface for Kamailio SIP Server. It is the last release with new features planned to be compatible with Kamailio v3.2.x. Next major releases for Siremis will target upcoming Kamailio v3.3.x. This time new features relate to web pages that allow management of own profile by SIP users, such as changing SIP password, management of speed dials or views for accounting, cdrs and location records. There was also added a form for doing user registrations (e.g., can be used for allowing public registrations). You can read more details about the release as well as details about installation at: * http://siremis.asipto.com/ Regards, Ramona

1 0

Limiting calls from users in prepaid scenerio
by Aft nix 07 Jun '12

07 Jun '12

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, We are thinking about launching a different "usage policy". The usage policy will be prepaid in some sense. but instead of limiting "number of minutes" we are thinking about limiting "number of calls". That means if you pay this X amount of doller, you can make Y amount of calls. calls can last whatever the user wants given that its under 1 hr. I've seen how to limit concurrent calls using SER, but the usage i have described, is not seen by me. So please help me out with this thing. My initial understanding about how it can be done : 1) when a dialog is confirmed with ACK(we have both callid, from tag, to tag), we can make an entry to the database by increasing "number of calls" field by one. 2) A field in database will be used as flag for usage limit. when "number of calls" exceeds the usage limit, it will flag it as "finished usage limit". 3) when a INVITE arrives, a check will be made with $fU against DB. With this framework in mind, I have read "db_mysql" module. And to my surprise, no direct the database functions are not exported to the cfg. I don't know if any module in kamailio exports mysql_queury() to cfg, which i need for my purpose. And may be there is some built in module which provides a higher level layer which will enable myself achieving my goal. If in the end i need "exported" mysql functions, then i can develop it myself. So which way i should go? Thanks in advance - -- - -aft -----BEGIN PGP SIGNATURE----- Version: OpenPGP.js v0.1 Comment: http://openpgpjs.org wsBcBAEBAgAQBQJPzc6dCRCJVJ6A/SK8awAARAkH/R5rqfiTPqBP+zZr8rgM 0JE42l9jxYq5OteD13U4CvcrUklXZlgDf2jwr1F7ndd1rlV3yKXAT2u61Bz5 LwNneRZydl3xnPmt2+TV6Qkcgy8RXgyRMntP9iLeQOXWJNW7D637h7tLL11d O50s86ofgJT+dfTICbomNZjLZRr1DvYOdcByG3LHcYx033G6bIFlMqJ0ZO+G cBdgtJcLrwo6iPn8Gv+5aJDy2kRUphrF7cVCzFzLPnNVZpvXNNcv7LB9bkUH cClcLDVAbsdw0TPd9Z6DNJSxFlHy2uzwNWB9mhpVJD1toT8Geb4iCOi+ZG8P WT5fgNwSoZlEm0BBBHwMUdk= =fcdD -----END PGP SIGNATURE-----

2 2

Issue using Kamailio as a proxy in front of Asterisk
by Dominik Mauritz 07 Jun '12

07 Jun '12

Hi All, some days ago I installed Kamailio as a front end for Asterisk following this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb I added Multi Domain support and adjusted some other things to fit my environment. Almost everything is working perfectly now. One small issue is left: With Kamailio in front of Asterisk I have one feature (next to other cool things) I was missing for years. I am now able to register more than one device on the same SIP account. This is nothing new for Kamailio users but Asterisk doesn’t support this. If I register two phones on one SIP account with Kamailio everything is fine. They are able to make outgoing calls and both ring on incoming calls. But if one phone unregisters Kamailio forwards the unregister request to Asterisk and Asterisk sets this account to offline. Now a call comes in but Asterisk is not sending out an invite because for Asterisk the phone is offline. I wonder if it is possible to forward an unregister request from Kamailio to Asterisk only if the last endpoint registered with one SIP account unregisters. If there is more than one endpoint using the same Account Kamailio should not forward the unregister request to Asterisk. Any idea? Thanks, Dominik

3 4

kamailio as SBC
by Mino Haluz 06 Jun '12

06 Jun '12

Hi, I know that kamailio is SIP proxy, but is there any way how to implement kamailio as SBC like OpenSIPS with B2BUA module ? I tried OpenSIPS with this module, but it does not work with mediaproxy module. Mino

5 4

Re: [SR-Users] : Capturing a users call (SIP+RTP+T.38)
by Anton Kvashenkin 06 Jun '12

06 Jun '12

You can try homer sip capture server, just google it. 06.06.2012 15:08 пользователь "Klaus Darilion" <klaus.mailinglists(a)pernau.at> написал: > Hi all! > > Do you know any comfortable tools to filter out a certain users call? e.g. > searching in SIP packets for the user pattern, get the media port out of > SDP and capture also RTP+T.38 > > I found pcapsipdump but have not tried it yet. > > Any suggestions or do I have to write my own tool? > > thanks > Klaus > > ______________________________**_________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >

1 0

Kamailio and Netann
by patrice.bodeven＠orange.com 05 Jun '12

05 Jun '12

Hello, I'am working on Kamailio version 3.2.2. I'm doing some test in interaction with Application server and media server. The applicationServer sends a INVITE sip to Media Server via the Kamailio. The R-URI has the format sip:annc@cl1.oms.ser.v0.pftest.net;play=http://1xx.1xx.1xx.1xx:8080/errfile/err.wav;repeat=forever But the parsing done by Kamailio of $ru is trunked to sip:annc@cl1.oms.ser.v0.pftest.net;play=http 1) Could you confirm if Kamailio supports Netann protocol ? 2) If not, how to route the INVITE with the correct R-URI format ? Thanks in advance. PatriceB _________________________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, France Telecom - Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, France Telecom - Orange is not liable for messages that have been modified, changed or falsified. Thank you.

3 3

Test SIP Servers for Simultaneous calls
by Vineet Menon 05 Jun '12

05 Jun '12

Hi, I want to test kamailio or any other server for it's simultaneous call rates, form which I mean how much calls it can support simultaneously?? I know about SIPp, but that won't fulfill my purpose, because SIPp doesn't offer simultaneity; rather it issues calls serially. BTW, is it possible to do multiple calls from same machine, since ports are consumed in the previous call?? I think the answer is negative.... I think, to do this kind of testing, we need multiple PCs..which call simultaneously... Any light on this?? Regards, Vineet Menon

3 4

Single-quoted vs double-quoted params in Path/Route
by Andreas Granig 04 Jun '12

04 Jun '12

Hi guys, When I implemented the Path module 6 years ago, I chose to enclose the received-URI in double-quotes, like this: Path: <sip:1.2.3.4:5060;received="sip:1.2.3.5:1234;transport=tcp"> This Path value is stored in usrloc by registrar module, and is converted to a Route in a subsequent request. This worked fine so far, because there's the logic in parser/parse_param.c to handle double-quoted parameters. Now while doing some interop tests with another platform, it turned out that double-quotes are not allowed in URI params (see https://lists.cs.columbia.edu/pipermail/sip-implementors/2008-May/019335.ht…, and in the grammar the allowed chars in the "unreserved" definition are "alphanum" and "mark", where "mark" is only "-" / "_" / "." / "!" / "~" / "*" / "'" / "(" / ")" ). Anyways, to fully comply with RFC3261, I'd like to push a bugfix to path module to use single-quotes instead, however it also needs a change in the parser to use parse_quoted_param() also in case of single-quoted params (which hasn't been handled at all until now). My question now is if there are any concerns on your side with handling single-quoted URI params the way we handle double-quoted ones? If we were strict, we'd actually need to remove the handling for double-quoted ones, but for backwards-compatibility I'd rather keep it there. Comments? Andreas

2 2

FS#213 - pua_reginfo: registration state may be incorrect when unregistering single contact
by Gnaneshwar Gatla 01 Jun '12

01 Jun '12

I'm working with pua_reginfo module, I thought everything was working fine. But FS#213, is not resolved yet. @@ -118,9 +111,10 @@ str* build_reginfo_full(urecord_t * record, str uri, ucontact_t* c, int type) { xmlNewProp(contact_node, BAD_CAST "id", BAD_CAST buf); /* Check, if this is the modified contact: */ if (ptr == c) { - if ((type & UL_CONTACT_INSERT) || (type & UL_CONTACT_UPDATE)) + if ((type & UL_CONTACT_INSERT) || (type & UL_CONTACT_UPDATE)) { + reg_active = 1; xmlNewProp(contact_node, BAD_CAST "state", BAD_CAST "active"); - else + } else xmlNewProp(contact_node, BAD_CAST "state", BAD_CAST "terminated"); if (type & UL_CONTACT_INSERT) xmlNewProp(contact_node, BAD_CAST "event", BAD_CAST "created"); else if (type & UL_CONTACT_UPDATE) xmlNewProp(contact_node, BAD_CAST "event", BAD_CAST "refreshed"); from the above code in pua_reginfo_fix_registration_state.patch If(ptr == c) returns false and the server keeps the state to "active" and the reg_info is never set to refreshed/terminated. I was wondering if I could use the following instead. If(strcmp(ptr->c, c->c)), which would compare the contact string, if equal the reg_info is update in the presentity. Regards Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

1 0

Re: [SR-Users] SIP Client for iphone
by Daniel-Constantin Mierla 01 Jun '12

01 Jun '12

Please keep the mailing list cc-ed -- private messages from mailing list are going to be ignored. If you go to jitsi.org, in the download page there are references to the mac os x builds. Cheers, Daniel On 6/1/12 9:38 AM, Vijay Thakur wrote: > Thanks for prompt reply. > Please suggest me the link from where i can download jitsi for Mac OS X. > > On Friday 01 June 2012 01:06 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> On 6/1/12 9:06 AM, Vijay Thakur wrote: >>> Dear All, >>> >>> Kindly suggest me the compatible client for iphone and Mac OS Leopard. >>> I have found that Jisti is working very fine for all features on >>> Linux and MS Windows. >>> >> I use jitsi for Mac OS X, iirc, it worked fine on Leopard, now I >> upgraded the OS. For iPhone and even the other OSes, check Linphone. >> >> Cheers, >> Daniel >> > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

multiple t_on_branch[]
by Vitaliy Aleksandrov 01 Jun '12

01 Jun '12

Hi all, Does kamailio allow to configure more than one branch_route for an initial INVITE ? I have tried to add: t_on_branch(branch_r1); t_on_branch(branch_r2); before t_relay(). Mentioned branch routes write debug to the kamailio.log and it looks like that only the last branch_route[branch_r2] works.

2 1

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