- sr-users - kamailio.org

SIP Client for iphone
by Vijay Thakur 01 Jun '12

01 Jun '12

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. -- 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. -----------------------------------------------------------------------------------------------------------------

4 6

CFGUtils module question
by Fabian Borot 31 May '12

31 May '12

Hello Can somebody help me understand the example given with MI command "is_gflag"? ... $ kamctl fifo set_gflag 1024 $ kamctl fifo is_gflag 1024 TRUE $ kamctl fifo is_gflag 1025 TRUE $ kamctl fifo is_gflag 1023 FALSE $ kamctl fifo set_gflag 0x10 $ kamctl fifo is_gflag 1023 TRUE $ kamctl fifo is_gflag 1007 FALSE $ kamctl fifo is_gflag 16 TRUE This is how I understand it. There are 32 posible flags (0..31), each one can be set independently. The commands "is_flag" and "set_gflag" expect a bitmask as value (decimal or hex) If I set "kamctl fifo set_gflag 1024", this means "100 0000 0000", I expect that the only flag that has been set to 1 is the position 10. How come the command "kamctl fifo is_gflag 1025" which is "100 0000 0001" can yield TRUE?. I expect it to be TRUE only if flags 0 and 10 are both set to 1. Same thing with setting the flag to 0x10 (16 in decimal, "1 0000" in binary), how come the "is_gflag 1023 and is_gflag 1007" can return TRUE? tx a lot fborot

2 2

Fwd: [Solved]Creating RURI ($ru) from Contact ($ct)
by Aft nix 31 May '12

31 May '12

Hi all, I'm posting the script which worked. In future if anybody faces the same problem can get a solution out of this. In request_route: #!ifdef WITH_HASH if (is_method("INVITE") && !has_totag()){ xdbg ("Contact [$ct] with Callid [$ci]"); $sht(a=>$ci) = $ct; } #!endif in route[WITHINDLG] #!ifdef WITH_HASH xdbg ("ruri [$ru]"); if ($(sht(a=>$ci){nameaddr.uri}) != $ru){ xdbg ("hash table entry [$sht(a=>$ci)]"); $ru = $(sht(a=>$ci){nameaddr.uri}); sht_rm_name_re("ha=>$ci"); xdbg ("rewriting ruri to [$ru]"); } #!endif Thanks everyone for the help. I'm marking this thread as solved. -- -aft -- -aft

3 7

SIP-XMPP gateway and not existent recipient
by intel＠intrans.baku.az 31 May '12

31 May '12

I set up sip-xmpp gateway using xmpp module(in component mode). messages from sip client to xmpp delivered, but I have a problem. If recipient is not existent, the sip client don't know nothing about it. How can I inform sip client about such condition? The xmpp server(ejabberd) inform it's component about this error(I've checked it using tcpdump), but kamailio seems ignore this error completely. What can I do in this situation? -- Homo Homini domini est!

1 0

Max registered users
by Vitaliy Aleksandrov 30 May '12

30 May '12

Hi all, I'm trying to find a way to get a number of the registered user locations and if it is bigger that some limit then kamailio will not save new registrations. reg_fetch_contacts() from registrar module doesn't help as it can fetch contacts only for one user. max_contacts parameter of the registrar module limits the number of contacts per AOR, but i need to limit max number of all contacts that proxy will keep in usrloc.

3 3

acc: can't get code for the Source-IP attribute
by Rabary 30 May '12

30 May '12

Hi all, When I start kamailio, I have theses errors in syslog and kamailio failed to start: May 30 12:02:11 kamailio /usr/sbin/kamailio[5652]: INFO: usrloc [hslot.c:53]: locks array size 512 May 30 12:02:11 kamailio /usr/sbin/kamailio[5652]: INFO: auth [auth_mod.c:312]: auth: qop set, but nonce-count (nc_enabled) support disabled May 30 12:02:11 kamailio /usr/sbin/kamailio[5652]: ERROR: acc_radius [acc_radius_mod.c:276]: acc: can't get code for the Source-IP attribute May 30 12:02:11 kamailio /usr/sbin/kamailio[5652]: ERROR: acc_radius [acc_radius_mod.c:289]: failed to init radius May 30 12:02:11 kamailio /usr/sbin/kamailio[5652]: ERROR: acc [acc_mod.c:758]: failed to initialize extra acc engine May 30 12:02:11 kamailio /usr/sbin/kamailio[5652]: ERROR: acc_radius [acc_radius_mod.c:154]: cannot register ACC RADIUS engine May 30 12:02:11 kamailio /usr/sbin/kamailio[5652]: ERROR: <core> [sr_module.c:932]: init_mod(): Error while initializing module acc_radius (/usr/lib/kamailio/modules_k/acc_radius.so) In radiusclient dictionary.sip, I have this: teddy@kamailio:~$ sudo grep -r -i source-ip /etc/radiusclient-ng/* /etc/radiusclient-ng/dictionary.sip:ATTRIBUTE Sip-Source-IP-Address 108 ipaddr And in dictionary file I include this dictionary.sip like this at the end of the file: $INCLUDE /etc/radiusclient-ng/dictionary.sip There are kamailio and libradiusclient-ng2 version installed: ii kamailio 3.2.3+squeeze1 very fast and configurable SIP proxy ii kamailio-mysql-modules 3.2.3+squeeze1 MySQL database connectivity module for Kamailio ii kamailio-radius-modules 3.2.3+squeeze1 radius modules for Kamailio ii libradiusclient-ng-dev 0.5.6-1.1 Enhanced RADIUS client library development files ii libradiusclient-ng2 0.5.6-1.1 Enhanced RADIUS client library So I don't know where I'm doing wrong, can anyone tell me what am I missing ? Thanks. -- Rabary Teddy Inutile d'imprimer ce mail

1 1

PUA_RegInfo patch
by Gnaneshwar Gatla 30 May '12

30 May '12

Hey, I have fixed the incorrect uri's being generated by this module while publishing user location. This patch also includes some more user record data to the xml during publish. I'm not sure if I have followed the naming convention right. Please find the patch file attached. Regards, Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

2 2

xhttp README
by Klaus Darilion 29 May '12

29 May '12

Hi! The readme should mention that core setting tcp_accept_no_cl=yes is required (at least for Firefox and Chrome) regards Klaus

3 5

Using Sip-Router Logging & Register Problem Problem
by Faisal Yousuf 29 May '12

29 May '12

Hi Experts, I am a newbee with SIP-Router/Kamailio - I installed Kamailio on a CentOS 5.5 VM - following the instructions from the following link http://kb.asipto.com/kamailio:install:3.0.x-from-git-centos5x Kamailio is installed & I can start kamailio deamon from service kamailio start However I also created two subscribers - but when I try to register my SIP Phone (by setting Kamailio as the domain to register to) - I can see my sip-phone sending the Register request to Kamailio (I put a tcpdump on the kamailio machine to see sip-messages reaching the host) but I dont get any response from Kamailio - so my sip phone reaches the 408 Registeration Timeout state. Now, I dont get any logs from Kamailio to know what is happening while my phone is sending the register request - I tried to configure logging following instructions from the following link http://www.kamailio.org/dokuwiki/doku.php/utils:basic-syslog-configuration But I dont get any logs in the /var/log/kamailio log file. To assist in understanding the issue - I am attaching the syslog config , kamailio config file & the kamailio service script in the init.d folder. My first objective will be to enable logging for kamailio & the next objective is to get the basic registration & routing between endpoints - my ultimate objective is to configure SER/Kamailio as a load-balancer for our B2BUA & Media Servers (I found the dispatcher module can help in accomplishing this) - Can anybody help me in identifying why I am not getting any logs ? I have already seen the wiki & have developed some understanding regarding how Kamailio is configured but am I seeking more information regarding the route language - (the state-model + reference of the functions that can be used in the route script etc.) - in this regards if I can get any documentation pointers from the user community that can help me gain a better understanding the use of Kamailio for above it will be great Regards, Faisal Regards, Faisal

4 9

Logging peer status
by Niklas Larsson 29 May '12

29 May '12

Hi, I'm coming from an asterisk background and are now using kamailio to handle registration and load balancing. In the asterisk world I added qualify=yes and listening on the AMI for events when peer is registering / unregistering / peer goes unreachable and reachable - then storing this info in a MySQL db to be displayed on a web page for support etc. This works really nicely and I'm trying to figure out how to do the same with kamailio and are able to generate an event when a peer is registering (in the registry route using mqueue and some lua magic). But can't find a way to get an event when an option ping (nathelper module) is unsuccessful or when an location record has expired(http://lists.kamailio.org/pipermail/users/2009-May/023270.html). So my question is - how are you guys dealing with this? What have I missed? /niklas

1 0

Validate domains, expose sip_hostport2su or similar
by Øyvind Kolbu 29 May '12

29 May '12

Hi, is there a way to validate a domain, e.g. it resolves to something, without calling t_relay() or forward() and checking the return code? In my case I'd like to validate enum results before dispatching it to our outbound proxies. If not valid I must use the pstn servers. In the logs I can see a function named sip_hostport2su does some validation, but it seems to be a c-function only. From the log: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "example.org" ERROR: tm [ut.h:327]: failed to resolve "example.org" -- Øyvind Kolbu

2 2

Address of Record (AOR) parameter
by Bruno Bresciani 29 May '12

29 May '12

Hi all, I can't understand how obtain address of record (aor) to functions lock_udomain and unlock_udomain... What is the meaning of this parameter? In other modules this parameter is obtained by R-URI on SIP messages struct (registrar) or by next value of node.kids mi_root struct (usrloc). Best Regards

1 0

planning the release of v3.3.0
by Daniel-Constantin Mierla 29 May '12

29 May '12

Hello, looking at the volume of reported issues on mailing lists and tracker, upcoming v3.3.0 seems to go well. I have few items on my to-do reported on the mailing list. Otherwise, in my side, I have 3.3.0 running in a production-like environment for about 4 weeks without problems and another one in a pre-production phase. Having in consideration all these, I propose to branch 3.3 on June 11. If everything looks very good at that time (or actually few days before), I think we can do the release also by that date. If not, we can do it a week later. Branching will allow new features development on git master, there are already several contributed patches with new features flowing around on mailing lists and tracker. Therefore, if you are aware of any issue, please report asap on mailing list (sr-dev) or tracker (preferable - http://sip-router.org/tracker/). Other suggestions about planning the release 3.3.0 are welcome, 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

1 0

Creating RURI ($ru) from Contact ($ct)
by Aft nix 28 May '12

28 May '12

On Mon, May 28, 2012 at 4:52 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > On 5/27/12 12:46 AM, Aft nix wrote: >> >> [...] >> >> Hi Aleksandrov >> Thanks for the reply. >> >> Yes my checks should be more specific. >> >> On the side how , i'm facing difficulties to print script variables >> and hashtable entries. >> >> xdbg("my var [$var(temp)]") >> xdbg("my hash entries [$sht(a=>$ci)"]) >> >> Both statements shows parse errors. What's the correct syntax to print >> script >> variables and hash table entries? > > if you copy&paste the lines from your config, then the errors are: > - missing ; at the end of lines > - second xdbg has the double quotes before ], making it invalid syntax Hi Daniel, Thanks, other than the typo, its ok to print stuffs like this, then. > > 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 > > > -- -aft -- -aft

1 0

Installation of Kamailio, Asterisk, siremis in Debain Squeeze
by Vijay Thakur 28 May '12

28 May '12

Dear Experts, I want to install Kamailio, Asterisk, siremis in Debian 6.0 Squeeze. For this purpose i have followed the following document: (1) http://nil.uniza.sk/sip/kamailio/installing-siremis-20-kamailio-web-managem… - Siremis Installation (2) http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour - Kamailio Installation (3) http://nil.uniza.sk/sip/kamailio/installing-siremis-20-kamailio-web-managem… - Siremis Installation I need the following features: 1 User Management 2 Contacts 3 Instant Messaging 4 My Presence (Presence Status) 5 Voice Call (SIP) 6 Conference (SIP Audio Calls) 7 Video calls 8 SMS 9 Outbound Audio Call 10 File Transfer 11 Video calls 12 Share Link 13 Face book Integration 14 Push Notifications (Broadcast Message) According to the the documents, some features are not working, or working features are not working sometime as supposed to work. Kindly suggest me straight forward document or any help to achieve it. -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail :vijay.thakur@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

Sending MSRP message to kamailio/msrprelay causes 501 error
by Reichert Alexander 28 May '12

28 May '12

Hi, We are developing a Java based SIP gateway implementation and are performing some integration tests on a kamailio/msrprelay server (plugin from www.msrprelay.org). The SIP signaling part works so far, but when sending a MSRP message, the client receives a 501 "Unknown method" error. However, it seems the request does not reach the MSRP module on the server as there is no corresponding trace log on the server side. On Protocol level, everything looks fine so far, the sequence executed is as follows: -->SIP REGISTER <-- SIP/2.0 401 Unauthorized -->SIP REGISTER <--SIP/2.0 200 OK -->INVITE <--SIP/2.0 407 Proxy Authentication Required -->ACK -->INVITE <--SIP/2.0 180 Ringing <--SIP/2.0 180 Ringing <--SIP/2.0 200 OK -->ACK The INVITE is processed with the following SDP packet in the request: v=0 o=- 1337788359935 1337788359935 IN IP4 localhost s=MSRP Session c=IN IP4 localhost t=0 0 m=message 2855 TCP/MSRP * a=path:msrp://localhost:2855/U3FuvW00m62UY0020H71;tcp a=accept-types:message/cpim text/* application/im-iscomposing+xml a=accept-wrapped-types:* a=setup:active And response: v=0 o=- 3546777215 3546777216 IN IP4 46.105.78.68 s=sipsimple 0.20.0 c=IN IP4 46.105.78.68 t=0 0 m=message 57233 TCP/MSRP * a=path:msrp://msrphost:2855/5AUNTOt7XlFsK7VuT4SzETEzMzc3ODg0MTUuMDcyOjQ2LjEwNS43OC42OA==;tcp msrp://msrphost:57233/6bf6dd73b9e4587b65b6;tcp a=accept-types:message/cpim text/* application/im-iscomposing+xml a=accept-wrapped-types:* a=setup:passive But when sending the MSRP message, 501 is returned: MSRP f0CHA1TD SEND To-Path: msrp://msrphost:57233/6bf6dd73b9e4587b65b6;tcp From-Path: msrp://localhost:52673/b10XadK9;tcp Message-ID: 13377883623135c260a3568cc Byte-Range: 1-*/14 Hello Francois -------f0CHA1TD$ MSRP f0CHA1TD 501 Unknown method: SEND To-Path: msrp://localhost:52673/b10XadK9;tcp From-Path: msrp://msrphost:57233/6bf6dd73b9e4587b65b6;tcp -------f0CHA1TD$ When testing with two sip-simple client endpoints, MSRP messaging works fine, but on protocol traces we cannot identify any differences between sip-simple communication flows and the ones we are implementing. Does anyone have a hint on what is wrong in this message flow? Any advice is appreciated. Thanks, Alex

2 3

Creating RURI ($ru) from Contact ($ct)
by Aft nix 28 May '12

28 May '12

Hi, I'm trying to create a ruri from contact header like following : In request route I've saved contact header in a hashtable using htable module. request_route{ --------------------- --------------------- #!ifdef WITH_HASH if (is_method("INVITE") && !has_totag()){ xdbg ("Contact [$ct] with Callid [$ci]"); $sht(a=>$ci) = $ct; } #!endif ---------------- } And in route[WITHINDLG] route[WITHINDLG] { --------------------------- --------------------------- #!ifdef WITH_HASH xdbg ("ruri [$ru]"); if ($sht(a=>$ci) != $ru){ $ru = ($sht(a=>$ci)); xdbg ("rewriting ruri to [$ru]"); } #!endif But it log shows that its unable to parse the uri : [4039]: DEBUG: <core> [parser/parse_uri.c:1277]: parse_uri: bad uri, state 0 parsed: <<sip> (4) / <<sip:2020@103.23.169.2:64358>> (29) : DEBUG: <core> [parser/parse_uri.c:1327]: ERROR: parse_sip_msg_uri: bad uri <<sip:2020@103.23.169.2:64358>> [4039]: ERROR: pv [pv_core.c:232]: failed to parse the R-URI [4039]: DEBUG: <script>: rewriting ruri to [<null>] I'm not understanding what's wrong with that. May be the port part should be discarded, but how i'm gonna do that? -- -aft

3 3

SER configuration File build system
by G Madhavi 28 May '12

28 May '12

Dear sir, We are searching for *ser.buildsystem.latest.tar.gz* in the internet throughly. But we are unable to find this particular file.so please send me the file or the web links which we can be able to download and configure our IMS network in proper. * * * * Thank you Best Regards, madhavi -- * *

1 0

any documentation around?
by Sherif Omran 26 May '12

26 May '12

Hi guys, is there any documentation for Siremis around? thanks S.

1 0

TLS config file
by Bruno Bresciani 25 May '12

25 May '12

The parameter [server:default] of TLS config file must be declared? I want specify two different interfaces (WAN and LAN) and I don't know which interface will be used as default by TLS module... Someone know some information about my question? Best Regards

2 2

[OT] sems [and kamailio] in risk-of-life service
by Daniel-Constantin Mierla 25 May '12

25 May '12

just forwarding from SEMS mailing list an interesting story of using kamailio and sems in emergency services... -------- Original Message -------- Subject: [Sems] sems in risk-of-life service Date: Thu, 24 May 2012 15:29:38 +0800 From: Jeremy A <jeremy(a)electrosilk.net> To: sems(a)lists.iptel.org Hi, This is a belated report on the use of SEMS in a risk of life service. The system uses kamailio in a distributed architecture of dozens of Fire & Rescue stations. This is heavily based on distributed and replicated DNS. A single '911' style headquarters has duplicate hot swap-over control rooms at other locations. The headquarter and alternate posts have servers that service HQ operator positions with SIP phones. These provides sidecar indication of F&R Station state for up to 64 F&R stations - using BLF. These phones are hooked into an integrated analogue audio management system. Each Fire and Rescue station has an embedded SIP based controller that runs Kamailio and proprietary software to control the F&R station electrical and safety systems as well as provide public address functions to alert the F&R staff of a new emergency. These PA announcements are SIP based using a DSL network and are live from the HQ positions, plus computer synthesized voice, as well as alerting tones. Each station also has multiple SIP phones for in-station and station to station calling. The network is decentralized so failure of the central control system still allows point to point communications between Fire and Rescue stations. The headquarter systems uses SEMS as the primary operator manager to perform multiple simultaneous deployment calls to remote Fire and Rescue stations. SEMS is used to create a dynamic conference between an operator and multiple Fire & Rescue stations. These are automatically initiated by SEMS and answered by the F&R embedded systems. This means an operator can broadcast a deployment message and initiate station control activities at up to five stations (fifth alarm) This is only constrained by the bandwidth available at the headquarters. Our SEMS packages have been designed to handle non-answered calls to the conference and provide operator indication by 'SMS' messages to the handsets and audio feedback. The system provides full forensic recording by using rtpproxy at all locations. These recordings are archived by an out-of-band process. Control of the system is purely SIP based - so every item in the system is a SIP based entity. This includes servers, embedded systems, and phones. The phones are physically integrated into operator positions that also handle PSTN, PBX, and radio traffic. The interface is purely keyboard on the operator phones. Options for integration of the SIP system into CAD (Computer Aided Dispatch) are obvious. The only drawback is the rusty and ancient systems and the unbelievable process required to get approval to integrate. The system as provided provides at least 5 nines reliability. Probably a lot better. The only downside is the DSL network (provided by others at amazing expense) that provides a system with a lousy 2 nines reliability. We are in the process of developing an offering using redundant DLS/3G routing to improve this. The field stations are a hybrid Centos 5/Slax system running out of flash. The HQ systems are straight Centos 5 systems running off disk or off flash. Future versions will be pure Centos out of flash with no fancy memory overlay - flash is well good enough. The system has been live for over a year with no major issues. I can't say how many lives have been saved, but certainly quite a few. At least we haven't been sued yet :-) Cheers.

1 0

Hop by hop CANCEL, wrong to header
by Vitaliy Aleksandrov 25 May '12

25 May '12

Hi all, I have a question about CANCEL message processing. My call sceraio: When an INVITE request comes I need to rewrite a domain part of the several headers(to, from, contact, SDP ip) according to the outgoing interface. I can do it before t_relay(), but when destination user has more then one locations and they are reachable through different interfaces all forked INVITEs will have the same domain. To avoid that problem i have tried to move rewriting part(subst from the textops module) to the branch_route. Unfortunately that solution didn't helped me, because "To" header of the outgoing CANCEL messages is wrong (unchanged. as it was at the moment when transaction was created by the t_relay() ). As i understood from the documentation and mailing list kamailio builds CANCEL based on outgoing INVITE. I did a small research and found that Kamailio really takes outgoing INVITE from each branch (invite_transaction_cell -> uac[b_id].request.buffer) and builds CANCEL (build_local_reparse() from tm/t_msgbuilder.c) based on it. But kamailio does an exception for the To header which is described below: ***cancel_branch* function (from t_cancel.c) calls *build_local_reparse*() and fills one of the parameters with a pointer to unmodified To header. * build_local_reparse*() uses received To header to construct outgoing CANCEL. I have changed: cancel = build_local_reparse(t, branch, &len, CANCEL, CANCEL_LEN, &t->to, reason); to: cancel = build_local_reparse(t, branch, &len, CANCEL, CANCEL_LEN, NULL, reason); and it works for me now. It looks like if E2E_CANCEL_HOP_BY_HOP *e2e_cancel()* (t_fwd.c) will call *e2e_cancel_branch()* which works as i want instead of *cancel_branch()*, but it is just my assumption. Why does kamailio generate CANCEL requests in such a way ? Did i miss something from the RFC3261 or kamailio documentation ? Thanks in advance for any help !

2 2

rtpproxy and timeout notification
by Laurent Schweizer 24 May '12

24 May '12

Hello all, I see the new timeout notification feature from the rtpproxy when no voice traffic is detected. I have 2 questions regarding this timeout notification: 1) When a timeout is detected and kamailio send the BYE is that possible to manipulate the BYE before he is sent to add maybe an extra header that indicate that the call was stopped due to a timeout issue ? 2) If a user put the call on hold (with a=recvonly or maybe old way IP=0.0.0.0.0) what will happens (the music on hold is not generated by rtpproxy but by another server) ? he will detect this as a timeout ? Is that possible to indicate to rtpproxy that for a moment stream will be only in one way ? Best regards Laurent

2 1

Question about more complicated rtpproxy scenario
by Lukas Macura 24 May '12

24 May '12

Hi to all, please could somevody help me with more complicated rtpproxy scenario? I still cannot understand how to make it working. We have SBC based on kamailio 3.2.3. Everything seems to work until I want to translate all SDP messages by SBC IP address (for both sides). Moreover, I need to translate ipv4 to ipv6 and vice versa too. I found some example script how to achieve ipv4 to ipv6 and vice versa and I am tryinf to use it. It is relatively easy to translate all RFC addresses to one public IP, but I need to translate even our public IPS into SBC IP. I made some rtpproxy sets: modparam("rtpproxy", "rtpproxy_sock", "1 == udp:127.0.0.1:7722") (EXTRA_OPTS="-l SBC_IP") for internal net to external modparam("rtpproxy", "rtpproxy_sock", "2 == udp:127.0.0.1:7723") (EXTRA_OPTS="-l /SBC_IP -6 SBC_IP6") for ipv4 to ipv6 modparam("rtpproxy", "rtpproxy_sock", "3 == udp:127.0.0.1:7724") (EXTRA_OPTS="-l SBC_IP -6 /SBC_IP6") for ipv6 to ipv4 modparam("rtpproxy", "rtpproxy_sock", "4 == udp:127.0.0.1:7725") (EXTRA_OPTS="-l SBC_IP/SBC_IP) for external to internal but I cannot make it working. Either there are problems with "incorrect port 0 in reply from rtp proxy" or SDP gets malformed because of more times rtpproxy_manage(). Please can you help me with some basic hint, where into routing logic put rtpproxy_manage() and set_rtpproxy_set() ? Moreover, how to check if rtpproxy was already forced? I could not find any example of forcing RTP proxy for all calls for both sides. I need this because entire RTP traffic has to go across SBC. And my internal IPs are not private. Thank you for any hint, Lukas Macura

2 1

vsf value in Record-Route header
by Robert R 24 May '12

24 May '12

Hi, * * * * We are facing a problem in our opener proxy receiving: *"Message too big" error.* Looking at the INVITE message sent by opener I notice the following in the record-route header: Record-Route: <sip:10.10.10.10;lr=on;ftag=5114A38-F80;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA-> Is this a valid value for vsd? Can we correct the value? Thanks, R

2 1

LinuxTag Today: Workshop build your own UC service with Kamailio and Jitsi
by Daniel-Constantin Mierla 24 May '12

24 May '12

Hello, short message to let you know that today at 12:00 I will do a practical workshop at LinuxTag in Berlin, about how to build your own unified communication service using kamailio as server and jitsi as client. If you visit the event, you can join the workshop in room New York 2.1, more details at: http://www.linuxtag.org/2012/en/program/project-workshops/details.html?no_c… Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

allow_address failed initialize permissions
by Grégoire Vandendeurpel 23 May '12

23 May '12

Hi, I'm using kamailio 3.2.3 and I use permissions.so But when I reload Kamailio , error: failed to initialize the allow_address function 0(3815) ERROR: <core> [sr_module.c:932]: init_mod(): Error while initializing module permissions (/usr/lib/kamailio/modules_k/permissions.so) ERROR: error while initializing modules I don't use this function in kamailio.cfg.. just allow_trusted !!! Any idea why this error still appears? Best regards -- *Grégoire Vandendeurpel, * * * *IT Sector*

2 1

Detect and Correct wrong BYEs from proxy.
by Aft nix 23 May '12

23 May '12

Hi all, When record routing is enabled, that means, the proxy will act statefull mode, There has to be a mechanism for storing dialogue states. Like the location of the peers (Contact header filed of initial INVITE and 200 OK). And i suspect these states are stored in variables which has a lifespan same as the dialogue itself. I guess important variables (Contact header, local and remote tag etc) are exported to configuration file also. I'm aware of the dialogue module but i'm kind of convinced there has to be a simpler way for achieving my goal. Our Sip gateway provider sending a wrong BYE when PSTN side phone hangs up. Instead of using Contact Header of the initial INVITE, its sending BYEs with its IP as ruri. So the BYE message is looping between proxy and gateway instead of relayed to UAC. What i'm trying to achieve is following : route[RELAY] { if(is_method('BYE')) { <rewrite ruri with contact field of the initial invite> t_relay() } } -- -aft

1 0

Siremis 3.2.1 PDO Issue.
by Jeremy Ardley 23 May '12

23 May '12

Hi, I'm installing a brand-new Kamailio / Siremis system. I have an issue with Siremis failing to load PDO extensions. I have other systems working O.K. on Centos 5.x This is my configuration: Centos 6.2 php-5.3.3-3.el6_2.8.i686.rpm php-cli-5.3.3-3.el6_2.8.i686.rpm php-common-5.3.3-3.el6_2.8.i686.rpm php-mysql-5.3.3-3.el6_2.8.i686.rpm php-pdo-5.3.3-3.el6_2.8.i686.rpm Siremis 3.2.1 php -m [PHP Modules] bz2 calendar Core ctype curl date ereg exif fileinfo filter ftp gettext gmp hash iconv json libxml mysql mysqli openssl pcntl pcre PDO pdo_mysql pdo_sqlite Phar readline Reflection session shmop SimpleXML sockets SPL sqlite3 standard tokenizer xml zip zlib [Zend Modules] The area where it screws up is in util.php while I am setting up Siremis via web-page $status[5]['item'] = 'PDO extensions'; $pdos = array(); if (extension_loaded('pdo')) $pdos[] = "pdo"; if (extension_loaded('pdo_mysql')) $pdos[] = "pdo_mysql"; if (extension_loaded('pdo_mssql')) $pdos[] = "pdo_mssql"; if (extension_loaded('pdo_oci')) $pdos[] = "pdo_oci"; if (extension_loaded('pdo_pgsql')) $pdos[] = "pdo_pgsql"; $status[5]['value'] = implode(", ", $pdos); $status[5]['status'] = $pdos[0]=='pdo' ? 'OK' : 'FAIL - PDO extensions are required.'; I have commented out the test but it fails later saying the PDO extensions are not loaded. I note that the php modules report gives upper case but the test is lower case. (I understand the extension_loaded function is case insensitive) Any assistance appreciated.

1 0

Rtpproxy and UPDATEs
by Spencer Thomason 22 May '12

22 May '12

Hello, I'm working on a residential type application where we are using Kamailio for NAT traversal and Freeswitch as a voicemail and media server. When a UA that is behind NAT sends an INVITE to check voicemail everything works correctly until the user listens to the message. The sdp in the initial INVITE is rewritten and rtp proxy is working but Freeswitch (on a public IP) then sends an UPDATE to display the caller name of the person who left the message. The problem is that the UAC (in this case a Polycom phone) then responds with its private IP in the SDP. Is there a was to handle these UPDATEs? I'm using Kamailio 3.2.3 with a fairly stock config. This is an excerpt of the config file with the NAT handling route: # RTPProxy control route[NATMANAGE] { if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } return; } Thanks, Spencer

3 7

kamailio debug send messages
by Bruno Bresciani 22 May '12

22 May '12

Hi all, Why kamailio send to syslog only received SIP messages? I want see the messages that kamailio send to the UAs, it's possible? On TCP/TLS SIP messages there is a degub of buffer with the message that will be send, but a strange content has been shown... following the log: *tcp_send: buf= ^W^C^A^B0HFCÃ°g^W^R Â£MÃ<8f>C^U<8f>ÃQB^^,Â¼8ÃÃ^KZÃÃ|<97>Â¼zÃÃ¶<82>Ã¬Ã±Ã%sÃÃ¢Â³gÂ¿Ã SÂ©<97>}y<91>.<94>Ã°Â¯Ã´<8f>YÂ¿Â¥Âµf<9d>Ã«<8f><9d>* My kamailio is version 3.1.2 and run with global parameter 'debug=3' and send the syslog messages. Best Regards

7 15

SQLite datatypes
by Sebastian Ferguson 22 May '12

22 May '12

Hi: I'm trying to configure a Kamailio 3.2 server as a Redirect Server. A softswitch we own will send a SIP INVITE for some calls like this "INVITE sip:102@sipwise.telecentro.net.ar SIP/2.0". Once detected the B number "102" (in this example), the kamailio server will search in a sqlite3 database the transltated number the call should be directed. And then the server will send a 300 Redirect with the new request uri. Like this "SIP/2.0 300 Redirect /.../ Contact: sip:63793266@voip.telecentro.net.ar". And the softswitch will let the call go to the right number. The translation desition is made based on the geographic area where the A party is located. If the translated number is like 63793266 or 1557311721 the system is working ok. So the .cfg file seems to be OK. But if the translated number is like 8003330303 the variable holding the SQL return is set to "-486604289". I think this is related to data types. In the logs I can see that kamailio is defaulting datatypes to INT (I've tested varchar, int, and bigint in sqlite3), but I don't know how large could an integer be in Kamailio. According to sqlite3 documentation the number 8003330303 is not long enough and in fact the number is in the database. Has anybody seen something like this? Can anyone help? I can provide the hole script and database if this helps. Regards, Sebastian Ferguson

3 8

Free Tickets for LinuxTag 2012
by Carsten Bock 22 May '12

22 May '12

Hi, we've got some free tickets for LinuxTag 2012, we want to share. If you like to receive one, just send me a short reply. They are given away on a first come, first served basis. Meet us at Hall 7.2b, Booth 278! So long, Carsten P.S.: I'll give a presentation about Kamailio, IMS and Rich Communications on Wednesday, May 23rd, 12:45 - 13:30 (Room Berlin I). -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ -- Meet ng-voice at LinuxTag 2012 in Berlin - May 23rd - 26th, 2012. Save the date!

1 0

Priority of targets when using uac_redirect
by Stefan Prelle 21 May '12

21 May '12

Hi, I am using kamailio 3.2.2 and the uac_redirect module to evaluate the response from a SIP redirect server. My problem is, that kamailio does not pick the target with the highest priority, but the one with the lowest. In the 302 the kamailio is receiving, there is a contact header like this: Contact:<sip:+49xxx@a.a.a.a:5060;user=phone>;q=0.5,<sip: +49xxx@b.b.b.b:5060;user=phone>;q=0.25. In my failure_route block I basically use the following commands get_redirects("1"); t_relay(); The call then gets redirected to b.b.b.b whereas I would expect it to go to a.a.a.a When I change my script to get_redirects("*"); the call is forked to both IPs in parallel (starting with a.a.a.a) Is this a bug in uac_redirect or am I using it in a wrong way? Regards, Stefan

1 0

installing the kamailio-devel problem with db_mysql
by Uri Shacked 21 May '12

21 May '12

Hi, i am trying to install the devel version using git. did the changes on the modules,lst file to include "db_mysql" i have mysql5.5 already installed with the shared components and client... i have no mysql.h file... so, the "make all" of kamailio failes... any ideas ? BR, Uri

3 3

Presence dialoginfo expiry
by Tais Plougmann Hansen 18 May '12

18 May '12

Hi, I have presence/pua_dialoginfo set up and working well with 3.2.3. The only problem I'm facing right now is dialog event expiry from presentity. If there's no call activity for a while, the dialog event will expire and should a client reboot after this, they'll start up with unknown state in BLF subscriptions. I.e. question marks in the display on Aastra phones and will stay that way until there's new call activity. Is there a way to keep dialog events fresh even without call activity? I've tried setting override_lifetime very high (604800) but the events still managed to get removed after an hour or so. -- Tais Plougmann Hansen OSD Consulting ApS

1 0

Re: [SR-Users] Change the Server field on the SIP dialog
by Charles Chance 18 May '12

18 May '12

Actually, for your version the documentation is at http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#server_header but it's the same thing :) Regards, Charles http://www.sipcentric.com -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Charles Chance Sent: 18 May 2012 11:12 To: sal(a)kom-1.com; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER)-Users Mailing List' Subject: Re: [SR-Users] Change the Server field on the SIP dialog Hi, Have you tried setting server_header? See here for more info: http://www.kamailio.org/wiki/cookbooks/3.2.x/core#server_header Regards, Charles http://www.sipcentric.com ________________________________________ From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Sal Aguilar Sent: 17 May 2012 19:00 To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Change the Server field on the SIP dialog Hi Kamailio/SIP Router List: I turn to you guys since I could find a straight forward answer (it seems my google skills have dimished) on the wiki. Yesterday evening, my boss asked me about the possibility of replacing the value of the Server field, that our Kamailio 3.1 SIP Dispatcher is sending to customers. I want to know if this can be accomplished by adding a statement into the kamailio.cfg or is there something else needed. Below is a sample of the SIP dialog that we are currently sending to customers. Call-ID: 0a63682b2d021cbf4ddeac4833fe1b10(a)192.168.1.2. CSeq: 102 INVITE. Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. Your feedback would be greatly appreciated. Salvador Aguilar Email: sal(a)kom-1.com USA: +1.305.523.9581 Colombia: +57.1.381.9533 Mobile: +505.8672.4725 KOM 1 Communication ~more than just talk. www.kom-1.com ________________________________________ No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2176 / Virus Database: 2425/5006 - Release Date: 05/17/12 _______________________________________________ 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 ----- No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2176 / Virus Database: 2425/5006 - Release Date: 05/17/12

1 0

Change the Server field on the SIP dialog
by Sal Aguilar 18 May '12

18 May '12

Hi Kamailio/SIP Router List: I turn to you guys since I could find a straight forward answer (it seems my google skills have dimished) on the wiki. Yesterday evening, my boss asked me about the possibility of replacing the value of the Server field, that our Kamailio 3.1 SIP Dispatcher is sending to customers. I want to know if this can be accomplished by adding a statement into the kamailio.cfg or is there something else needed. Below is a sample of the SIP dialog that we are currently sending to customers. Call-ID: 0a63682b2d021cbf4ddeac4833fe1b10(a)192.168.1.2. CSeq: 102 INVITE. Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. Your feedback would be greatly appreciated. Salvador Aguilar Email: <mailto:sal@kom-1.com> sal(a)kom-1.com USA: +1.305.523.9581 Colombia: +57.1.381.9533 Mobile: +505.8672.4725 KOM 1 Communication ~more than just talk. www.kom-1.com <http://www.kom-1.com/>

2 1

sdpops command not working
by Grégoire Vandendeurpel 18 May '12

18 May '12

Hello, I use Kamailio 3.2.3 and I used the command from the sdpops module " sdp_remove_media("video");" But when i restarted kamailio, this error appeared: 0(2623) ERROR: <core> [cfg.y:3393]: cfg. parser: failed to find command sdp_remove_media 0(2623) : <core> [cfg.y:3532]: parse error in config file /etc/kamailio/kamailio.cfg, line 721, column 27: unknown command, missing loadmodule? Do You know why ? Cause I loeded the sdpops module. Thanks in advance for your help -- *Grégoire Vandendeurpel, * * * *IT Sector*

2 13

Question about LCR ping option
by Fabian Borot 17 May '12

17 May '12

Hello I am using kamailio 1.5.2. Our kamailio server has multiple IPs, and 1 of them happens to be a private IP for management purposes. When enabling the "ping" feature of the LCR module the machine sends the OPTIONS messages from the private IP. I need it to use at least any of the public IPs and if I can specify it it would be even better. I tried using the directive "ping from" with one of the public IPs but it only uses it to populate the "From" header, not to send the message from. One option could be to configure the configuration file with the "listen" directive including only the public IPs. However, there are more than 50 IPs, is there a limit to the number of IPs? I tried with a subnet and it is not allowed. Is there any other work around? txs a lot fborot

1 0

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Daniel-Constantin Mierla 17 May '12

17 May '12

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

2 1

dlz-ldap-enum - expose LDAP data to Kamailio via ENUM
by Daniel Pocock 17 May '12

17 May '12

I've recently released a dlz ENUM module for the bind9 nameserver: http://www.opentelecoms.org/dlz-ldap-enum Basically, it handles ENUM queries from Kamailio, Asterisk, FreeSWITCH, repro, Lumicall, searches for the phone number in LDAP, and if found, returns the email address as both a SIP address and Jabber address This should make it even easier than ever before to get federated VoIP up and running using email addresses interchangeably with phone numbers. If the data already exists in LDAP as an address book, then just install bind9, install the module and you're up and running. Regards, Daniel

1 0

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Daniel-Constantin Mierla 17 May '12

17 May '12

Hello, can you print in frame one: p *ps p *(ua_pres_t*)(*ps->param) I wonder why entity is null Is this a test system where eventually one can get remote access for live troubleshooting while you are testing? Cheers, Daniel On 5/16/12 2:02 PM, Charles Chance wrote: > > Hi Daniel, > > It happens every time but I just can't seem to pinpoint the cause. > Those parameters were initially unset and it still happened -- I added > them later in an attempt to diagnose. > > I removed them and re-ran, which produces a slightly different > backtrace... > > Program terminated with signal 11, Segmentation fault. > > #0 core_hash (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) at > ../../hashes.h:277 > > 277 end=s1->s+s1->len; > > (gdb) bt > > #0 core_hash (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) at > ../../hashes.h:277 > > #1 publ_cback_func (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) at > send_publish.c:246 > > #2 0x005041d6 in run_trans_callbacks_internal (cb_lst=0xb3dc8e78, > type=1024, trans=0xb3dc8e38, params=0xbfcdd5d8) at t_hooks.c:290 > > #3 0x005044d6 in run_trans_callbacks (type=1024, trans=0x200, > req=0x0, rpl=0xb7c30340, code=412) at t_hooks.c:317 > > #4 0x0052db28 in local_reply (t=0xb3dc8e38, p_msg=0xb7c30340, > branch=0, msg_status=412, cancel_data=0xbfcdd86c) at t_reply.c:2001 > > #5 0x00530b45 in reply_received (p_msg=0xb7c30340) at t_reply.c:2350 > > #6 0x0809d5f4 in forward_reply (msg=0xb7c30340) at forward.c:790 > > #7 0x080e11a9 in receive_msg ( > > buf=0x82c4fa0 "SIP/2.0 412 Conditional request failed\r\nVia: > SIP/2.0/UDP xx.xx.xx.xxx;branch=z9hG4bK7058.f06a8876.0\r\nTo: > sip:152@xx.xx.xx.xxx:5060;tag=03373b1b433b65504ea28bd3d9e6db76.331b\r\nFrom: > sip:152@xx.xx.xx."..., len=374, rcv_info=0xbfcdda58) at receive.c:270 > > #8 0x0817801b in udp_rcv_loop () at udp_server.c:544 > > #9 0x080b0426 in main_loop () at main.c:1633 > > #10 0x080b3d02 in main (argc=11, argv=0xbfcddd34) at main.c:2546 > > ...although the root cause appears to be the same: > > (gdb) frame 0 > > #0 core_hash (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) at > ../../hashes.h:277 > > 277 end=s1->s+s1->len; > > (gdb) p *s1 > > Cannot access memory at address 0x40323531 > > (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 *hentity > > Cannot access memory at address 0x0 > > The PUBLISH seems to be sent ok for both caller/callee for states > trying/confirmed. On tear-down however, it appears to crash somewhere > after sending state terminated for the callee as demonstrated by the > log... > > May 16 11:48:57 ws1 /usr/local/kamailio-devel/sbin/kamailio[31954]: > INFO: pua_dialoginfo [dialog_publish.c:386]: CALLING dialog_publish > for URI sip:000091@xx.xx.xx.xxx:5060 > > May 16 11:48:57 ws1 /usr/local/kamailio-devel/sbin/kamailio[31953]: > INFO: pua_dialoginfo [dialog_publish.c:386]: CALLING dialog_publish > for URI sip:000091@ xx.xx.xx.xxx:5060 > > May 16 11:48:57 ws1 /usr/local/kamailio-devel/sbin/kamailio[31953]: > INFO: pua_dialoginfo [dialog_publish.c:386]: CALLING dialog_publish > for URI sip:152@ xx.xx.xx.xxx:5060 > > May 16 11:48:57 ws1 /usr/local/kamailio-devel/sbin/kamailio[31953]: > INFO: <core> [mem/f_malloc.c:529]: freeing a free fragment > (0xb3dc7c18/0xb3dc7c20) - ignore > > May 16 11:49:08 ws1 /usr/local/kamailio-devel/sbin/kamailio[31958]: > INFO: pua_dialoginfo [dialog_publish.c:386]: CALLING dialog_publish > for URI sip:000091@ xx.xx.xx.xxx:5060 > > May 16 11:49:08 ws1 /usr/local/kamailio-devel/sbin/kamailio[31958]: > INFO: pua_dialoginfo [dialog_publish.c:386]: CALLING dialog_publish > for URI sip:152@ xx.xx.xx.xxx:5060 > > May 16 11:49:08 ws1 /usr/local/kamailio-devel/sbin/kamailio[31953]: > ERROR: presence [presentity.c:784]: No E_Tag match a.1337165329.31958.1.0 > > May 16 11:49:09 ws1 /usr/local/kamailio-devel/sbin/kamailio[31941]: > ALERT: <core> [main.c:785]: child process 31958 exited by a signal 11 > > I've attached the records from the presentity table also, as the > presence module doesn't appear to be matching the PUBLISHes with > existing records. > > Thanks again for your time, > > Charles > > ------------------------------------------------------------------------ > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* 16 May 2012 08:15 > *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, > > hmm, hentity is null even a check for its value is done above ... > > Is this reproducible or happens from time to time? What are the values > you set for $avp(s:puburis_caller) and $avp(s:puburis_callee)? > > Can you try without parameters: > > modparam("pua_dialoginfo", "use_pubruri_avps", 1) > modparam("pua_dialoginfo", "pubruri_caller_avp", "$avp(s:puburis_caller)") > modparam("pua_dialoginfo", "pubruri_callee_avp", "$avp(s:puburis_callee)") > > These were added by a recent contribution, I want to see if they are > the reason. > > Cheers, > Daniel > > On 5/16/12 12:00 AM, Charles Chance wrote: > >> Hi Daniel, >> >> I tried to check that already, but it's not there: >> >> (gdb) frame 1 >> >> #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) >> >> at send_publish.c:246 >> >> 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); >> >> (gdb) p *hentity >> >> Cannot access memory at address 0x0 >> >> Config parameters are: >> >> modparam("pua", "db_url", "mysql://user:pass@xx.xx.xx.xxx/dbname" >> <mailto:mysql://user:pass@xx.xx.xx.xxx/dbname>) >> >> modparam("pua", "db_table", "kam_pua") >> >> modparam("pua_dialoginfo", "caller_confirmed", 1) >> >> modparam("pua_dialoginfo", "include_localremote", 0) >> >> modparam("pua_dialoginfo", "override_lifetime", 300) >> >> modparam("pua_dialoginfo", "use_pubruri_avps", 1) >> >> modparam("pua_dialoginfo", "pubruri_caller_avp", >> "$avp(s:puburis_caller)") >> >> modparam("pua_dialoginfo", "pubruri_callee_avp", >> "$avp(s:puburis_callee)") >> >> Cheers, >> >> Charles >> >> ------------------------------------------------------------------------ >> >> *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> *Sent:* 15 May 2012 21:58 >> *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, >> >> interesting that the output of bt shows inline core_hash function >> prototype with same parameters as publ_cback_func, not being able to >> spot eventual wrong parameter values. >> >> Anyhow, in the gdb, can you do the commands: >> >> frame 1 >> p *hentity >> >> and send the output? Let's see the value of the pub uri for which >> hashing function crashes. >> >> What are the parameters for pua and pua_dialog info you set in the >> config file? >> >> Cheers, >> Daniel >> >> On 5/15/12 11:53 AM, Charles Chance wrote: >> >>> Hi Daniel, >>> >>> As per your previous reply (sorry, I deleted it accidentally), >>> please find attached output of 'bt full' as requested. >>> >>> Best regards and thanks again, >>> >>> Charles >>> >>> ------------------------------------------------------------------------ >>> >>> *From:*Charles Chance [mailto:charles.chance@sipcentric.com] >>> *Sent:* 13 May 2012 11:21 >>> *To:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) >>> -Users Mailing List' >>> *Subject:* (Devel) Segmentation fault using pua_dialoginfo >>> >>> Hi, >>> >>> Currently testing devel version and experiencing the following when >>> using pua_dialoginfo: >>> >>> ------------------------------------- >>> >>> Program terminated with signal 11, Segmentation fault. >>> >>> #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >>> ../../hashes.h:279 >>> >>> 279 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; >>> >>> (gdb) bt >>> >>> #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >>> ../../hashes.h:279 >>> >>> #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >>> send_publish.c:246 >>> >>> #2 0x002861d6 in run_trans_callbacks_internal (cb_lst=0xb3d7e370, >>> type=1024, trans=0xb3d7e330, params=0xbfcc31c8) at t_hooks.c:290 >>> >>> #3 0x002864d6 in run_trans_callbacks (type=1024, trans=0x200, >>> req=0x0, rpl=0xb7bea008, code=200) at t_hooks.c:317 >>> >>> #4 0x002afb28 in local_reply (t=0xb3d7e330, p_msg=0xb7bea008, >>> branch=0, msg_status=200, cancel_data=0xbfcc345c) at t_reply.c:2001 >>> >>> #5 0x002b2b45 in reply_received (p_msg=0xb7bea008) at t_reply.c:2350 >>> >>> #6 0x0809d5f4 in forward_reply (msg=0xb7bea008) at forward.c:790 >>> >>> #7 0x080e11a9 in receive_msg ( >>> >>> buf=0x82c4fa0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP >>> xx.xx.xx.xxx;branch=z9hG4bKc13b.25313b6.0\r\nTo: >>> sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2\r\nFrom >>> <sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2%5Cr%5CnFrom>: >>> sip:000094@xx.xx.xx.xxx;tag=4cc17c2c75cb82fa"..., len=393, >>> rcv_info=0xbfcc3648) at receive.c:270 >>> >>> #8 0x0817801b in udp_rcv_loop () at udp_server.c:544 >>> >>> #9 0x080b0426 in main_loop () at main.c:1633 >>> >>> #10 0x080b3d02 in main (argc=11, argv=0xbfcc3924) at main.c:2546 >>> >>> (gdb) up >>> >>> #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >>> send_publish.c:246 >>> >>> 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); >>> >>> (gdb) p hentity->pres_uri >>> >>> Cannot access memory at address 0x8 >>> >>> (gdb) p *hentity >>> >>> Cannot access memory at address 0x0 >>> >>> ------------------------------------- >>> >>> Your time is greatly appreciated... >>> >>> 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: 2012.0.2171 / Virus Database: 2425/5000 - Release Date: 05/15/12 >> > > > -- > 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: 2425/5001 - Release Date: 05/15/12 > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

un-suscribe
by Davis Doe 17 May '12

17 May '12

Please i would not like to receive mail from the kamailio group thanks davis

1 0

Setting up siremis 2.0
by pranjal deka 17 May '12

17 May '12

Hi, I have a sip registrar.(kamailio server installed in one server) I want to use the siremis 2.0 to collect management information from the sip registrar. Can you guide me how to make the connection between the sip registrar and the siremis so that the stats can be collected? Thanks

1 0

How to use Async module of Kamailio
by 罗俊明 17 May '12

17 May '12

1 0

Re: [SR-Users] How does RLS work with kamailio.
by Gnaneshwar Gatla 17 May '12

17 May '12

Hello, I have partly fixed this, I was using an incorrect xml format and missing usage of pua module. It took a little more than debugging to find this out. Now I have run into this new issue. I have implemented a presence + rls + xcap server: When client subscribes for RLS, kamailio Notifies with resourcelist for the client. The pua within the server does backend subscribtions. These backend subscribtions are responded with 202 Accepted replies. When the resources in the contact list change their presence information the pua generates NOTIFIY with subscription state: pending. Rls module does not reply back. Kamailio does not return from the following command. if(is_method("NOTIFY")) { xlog("handle notify"); rls_handle_notify(); xlog("notify was handled"); exit; }; Thank you for looking into this. Regards Gnani From: Gnaneshwar Gatla Sent: Thursday, April 19, 2012 4:53 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Cody Herzog Subject: How does RLS work with kamailio. Hello, I'm trying to implement RLS with integrated XCAP server. I have successfully run the Kamailio instance with the following snippet. Since I had no RLS clients available, I have tried to test this with SIPp. The attached pcap trace is the result. Test: 1. Send an XCAP signal over http to Kamailio. 2. Register and subscribe Kamailio. The test does store the xml received over http in the xcap table in the database. But rls module does not store the data neither in rls_presentity or rls_watcher. When I sent the Subscribe with sipp, the server throws error that subscription was not found in the database. I would like to know if I'm doing something wrong to implement this service. Is there any brief tutorial on how to implement this? Kamailio.cfg snippet: ##Presence modparam("presence", "server_address", PRESENCE_IP) modparam("presence", "clean_period", 65) modparam("presence", "db_update_period", 60) modparam("presence", "max_expires", 75) modparam("presence", "fallback2db", 1) modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 0) modparam("presence_xml", "integrated_xcap_server", 1) ##rls modparams modparam("rls", "db_url", DBURL) modparam("rls", "waitn_time", 10) modparam("rls", "integrated_xcap_server", 1) modparam("rls", "max_notify_body_length", 32000) modparam("rls", "to_presence_code", 10) modparam("rls", "server_address", RLS_URI) modparam("rls", "outbound_proxy", PUA_OUTBOUND_PROXY) modparam("rls", "xcap_root", XCAP_ROOT) ##XCAP modparams modparam("xcap_server", "db_url", DBURL) modparam("xcap_client", "db_url", DBURL) modparam("xcap_client", "query_period", 50) route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { $var(ret_code)= rls_handle_subscribe(); xlog("Subscribe: $var(ret_code)"); if($var(ret_code)== 10) handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } In the [xhttp:request]: switch($rm) { case "PUT": xcaps_put("$var(uri)", "$hu", "$rb"); if($xcapuri(u=>auid)=~"pres-rules") { xlog("===== xhttp put: refreshing watchers for $var(uri)\n"); pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } else if ($xcapuri(u=>auid)=~"resource-lists" || $xcapuri(u=>auid)=~"rls-services") { xlog("==== xhttp put: refreshing rls watchers $var(uri)\n"); rls_update_subs("$var(uri)", "presence"); } exit; break; Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

1 0

Re: [SR-Users] kamailio Crashes with rls module
by Gnaneshwar Gatla 17 May '12

17 May '12

Oppsie.. forgot to attach the document. Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199 From: Gnaneshwar Gatla Sent: Thursday, April 26, 2012 2:14 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: kamailio Crashes with rls module Hello, I'm experimenting the usage of rls module with Kamailio. I have been able to successfully update xcap documents on Kamailio XCAP server. I have tried to use SIPp to get a NOTIFY for subscribing a rls-service list. Kamailio does get the xml from the xcap table and tries to parse it and crashes. I'm not sure if this is because the format of the XML, I derived the xml format from the RFC 4662 and 4826. I have attached the Kamailio log. I need help with this. Any help is appreciated, thank you very much. Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

2 6

TLS performance testing
by Kristian Kielhofner 16 May '12

16 May '12

Hello everyone, I'm doing some TLS performance testing on Kamailio 3.2.1. Here's my setup: kamailio -V version: kamailio 3.2.1 (x86_64/linux) 31c991 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, USE_STUN, 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: 31c991 compiled on 19:38:03 Dec 20 2011 with gcc 4.4.6 uname -a Linux null.null.com 2.6.32-220.el6.x86_64 #1 SMP Tue Dec 6 19:48:22 GMT 2011 x86_64 x86_64 x86_64 GNU/Linux CentOS 6.2 on a Dell PowerEdge R610 with 24 Intel X5650 Cores at 2.67GHz and 12GB of RAM (I could use more). Kamailio is the default config with a few changes: - WITH_TLS defined - TLS is using self-generated CAs/certs (essentially openvpn easy-rsa) with 1024 bit key size - TLS is *not* configured to verify client OR server certs by default - I'm using TLS v1 (SSL 3.1) - TLS cipher suites are set to any (although my simulated UAs only offer AES 256+SHA) - Various changes to Kamailio children (up to 256 at times) and memory sizes (up to 2048mb and even 4096mb at times) - One DNS based alias added - Maximum TCP connections increased to 65000 - Kamailio is configured to only listen on tested IP (UDP, TCP, TLS sockets active) - Syslog has been configured to log local0 (Kamailio) asynchronously My test rig/call generator is an Ixia Xcellon-Ultra NP load module with IXLoad. My call scenario does the following: - Registers two simulated user agents (100000, 200000) to Kamailio with TLS - Places call from 100000 to 200000 via Kamailio with TLS - Increments both user agents by 1 and continues as quickly (cps) as I like up to a channel limit (also configurable) - The Ixia generates a valid SDP but no RTP is generated (although that's certainly possible at these call levels) Two 1 gig ports on the Ixia are connected to the Broadcom NICs on the Dell R610 via a Cisco Catalyst 4948 switch. One port on the Ixia emulates the 100000 agents (A leg) and the other emulates the 200000 agents (B leg). Of course I can provide more information if needed. Here are some test numbers: With TLS at 20cps, 120 sec calls, up to a total of 2470 calls (4940 registrations) life is good. Very good - call setup time averages 23ms, the cps rate holds indefinitely, and not a single call or registration fails over long term tests. UDP and TCP numbers are excellent (bordering ridiculous) - usually around 500cps with practically no reasonable upper limit on simultaneous calls. This doesn't need any further discussion :). The TLS numbers start falling apart pretty quickly after 20cps, however. If I change the TLS test to 40cps, 120 sec calls, up to a total of 4940 calls (9,880 registrations) Kamailio starts to (seriously) struggle. The rate starts fluctuating all over the place, call setup time averages jump to 8000ms (or more) and things just generally get ugly. Interestingly enough all of the user agents are able to register, the logs look fine (to my eye at this log level) and the system (CPU, network, etc) doesn't appear to be under stress at all. I have a few questions: 1) Is there something obviously wrong or stupid I'm doing here? 2) Why are the TLS tests so much worse than TCP and UDP? Am I missing something here? Thanks (in advance) for any advice anyone might be able to offer! -- Kristian Kielhofner

3 10

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Marius Zbihlei 16 May '12

16 May '12

On 05/16/2012 03:02 PM, Charles Chance wrote: >>> >>> Currently testing devel version and experiencing the following when >>> using pua_dialoginfo: >>> >>> ------------------------------------- >>> >>> Program terminated with signal 11, Segmentation fault. >>> >>> #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >>> ../../hashes.h:279 >>> >>> 279 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; >>> >>> Hello, Don't know if this is related, but the statement above hides a nasty bug: p[1] is a char (as for the description), so is signed, shifting to the left 16 bits is ok, as an integral promotion takes place. The problem is that if p[1] is negative (-1 is 0xff), then the integral promotion is signed as well and is complementary of 2 (or whatever the implementation considers), and the shift will return another value as expected. For example : #include <stdio.h> int main(){ char b = 0xff; unsigned h = (b<<16); printf("%x\n", h); return 0; } will print ffff0000 instead of ff0000 as expected. Haven't calculated the implication of this related to the bug. An fix would be to cast p[1] to (unsigned char) Cheers, Marius -- Zbihlei Marius Head of Linux Development Services Romania 1&1 Internet Development srl Tel KA: 754-9152 Str Mircea Eliade 18 Tel RO: +40-31-223-9152 Sect 1, Bucuresti mailto: marius.zbihlei(a)1and1.ro 71295, Romania

2 1

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Daniel-Constantin Mierla 16 May '12

16 May '12

Hello, hmm, hentity is null even a check for its value is done above ... Is this reproducible or happens from time to time? What are the values you set for $avp(s:puburis_caller) and $avp(s:puburis_callee)? Can you try without parameters: modparam("pua_dialoginfo", "use_pubruri_avps", 1) modparam("pua_dialoginfo", "pubruri_caller_avp", "$avp(s:puburis_caller)") modparam("pua_dialoginfo", "pubruri_callee_avp", "$avp(s:puburis_callee)") These were added by a recent contribution, I want to see if they are the reason. Cheers, Daniel On 5/16/12 12:00 AM, Charles Chance wrote: > > Hi Daniel, > > I tried to check that already, but it's not there: > > (gdb) frame 1 > > #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) > > at send_publish.c:246 > > 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); > > (gdb) p *hentity > > Cannot access memory at address 0x0 > > Config parameters are: > > modparam("pua", "db_url", "mysql://user:pass@xx.xx.xx.xxx/dbname") > > modparam("pua", "db_table", "kam_pua") > > modparam("pua_dialoginfo", "caller_confirmed", 1) > > modparam("pua_dialoginfo", "include_localremote", 0) > > modparam("pua_dialoginfo", "override_lifetime", 300) > > modparam("pua_dialoginfo", "use_pubruri_avps", 1) > > modparam("pua_dialoginfo", "pubruri_caller_avp", "$avp(s:puburis_caller)") > > modparam("pua_dialoginfo", "pubruri_callee_avp", "$avp(s:puburis_callee)") > > Cheers, > > Charles > > ------------------------------------------------------------------------ > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* 15 May 2012 21:58 > *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, > > interesting that the output of bt shows inline core_hash function > prototype with same parameters as publ_cback_func, not being able to > spot eventual wrong parameter values. > > Anyhow, in the gdb, can you do the commands: > > frame 1 > p *hentity > > and send the output? Let's see the value of the pub uri for which > hashing function crashes. > > What are the parameters for pua and pua_dialog info you set in the > config file? > > Cheers, > Daniel > > On 5/15/12 11:53 AM, Charles Chance wrote: > >> Hi Daniel, >> >> As per your previous reply (sorry, I deleted it accidentally), please >> find attached output of 'bt full' as requested. >> >> Best regards and thanks again, >> >> Charles >> >> ------------------------------------------------------------------------ >> >> *From:*Charles Chance [mailto:charles.chance@sipcentric.com] >> *Sent:* 13 May 2012 11:21 >> *To:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) >> -Users Mailing List' >> *Subject:* (Devel) Segmentation fault using pua_dialoginfo >> >> Hi, >> >> Currently testing devel version and experiencing the following when >> using pua_dialoginfo: >> >> ------------------------------------- >> >> Program terminated with signal 11, Segmentation fault. >> >> #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >> ../../hashes.h:279 >> >> 279 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; >> >> (gdb) bt >> >> #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >> ../../hashes.h:279 >> >> #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >> send_publish.c:246 >> >> #2 0x002861d6 in run_trans_callbacks_internal (cb_lst=0xb3d7e370, >> type=1024, trans=0xb3d7e330, params=0xbfcc31c8) at t_hooks.c:290 >> >> #3 0x002864d6 in run_trans_callbacks (type=1024, trans=0x200, >> req=0x0, rpl=0xb7bea008, code=200) at t_hooks.c:317 >> >> #4 0x002afb28 in local_reply (t=0xb3d7e330, p_msg=0xb7bea008, >> branch=0, msg_status=200, cancel_data=0xbfcc345c) at t_reply.c:2001 >> >> #5 0x002b2b45 in reply_received (p_msg=0xb7bea008) at t_reply.c:2350 >> >> #6 0x0809d5f4 in forward_reply (msg=0xb7bea008) at forward.c:790 >> >> #7 0x080e11a9 in receive_msg ( >> >> buf=0x82c4fa0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP >> xx.xx.xx.xxx;branch=z9hG4bKc13b.25313b6.0\r\nTo: >> sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2\r\nFrom >> <sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2%5Cr%5CnFrom>: >> sip:000094@xx.xx.xx.xxx;tag=4cc17c2c75cb82fa"..., len=393, >> rcv_info=0xbfcc3648) at receive.c:270 >> >> #8 0x0817801b in udp_rcv_loop () at udp_server.c:544 >> >> #9 0x080b0426 in main_loop () at main.c:1633 >> >> #10 0x080b3d02 in main (argc=11, argv=0xbfcc3924) at main.c:2546 >> >> (gdb) up >> >> #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >> send_publish.c:246 >> >> 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); >> >> (gdb) p hentity->pres_uri >> >> Cannot access memory at address 0x8 >> >> (gdb) p *hentity >> >> Cannot access memory at address 0x0 >> >> ------------------------------------- >> >> Your time is greatly appreciated... >> >> 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: 2012.0.2171 / Virus Database: 2425/5000 - Release Date: 05/15/12 > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

db_cluster module
by Uri Shacked 16 May '12

16 May '12

Hi, can the db_cluster module be used to all DB connections? if today i set for example "modparam("dialplan", db_url", DBURL)" i will be able to use "modparam("dialplan", db_url", CLUSTER)" ? still, on the #!define DBURL "mysql://.........." i can not use clustering right? BR, Uri

2 1

Re: [SR-Users] Siremis install fail at last page of web install--Solved
by Magnus 16 May '12

16 May '12

Just in case anyone comes up with same issue - post solving the directory/file ownership and repeating the import of db files - Siremis came up as it should Sorry for noise. -----Original Message----- From: Magnus Sent: Tuesday, May 15, 2012 4:35 PM To: sr-users(a)lists.sip-router.org Subject: RE: Siremis install fail at last page of web install "can't open cmp file to write " in the httpd error log I get two lines " mkdir(): No such file or directory in /var/www/html/siremis-3.2.0/openbiz/bin/Resource.php on line 408" and next line of "fopen(/var/www/html/siremis-3.2.0/siremis/files/cache/metadata/CB3F6C48_log Service.cmp): failed to open stream"

2 1

Siremis install fail at last page of web install
by Magnus 16 May '12

16 May '12

Hello all, I have an issue with final stage of installing siremis - at the last screen of the install I get an error in browser of "can't open cmp file to write" in the httpd log I get two lines " mkdir(): No such file or directory in /var/www/html/siremis-3.2.0/openbiz/bin/Resource.php on line 408" and next line of "fopen(/var/www/html/siremis-3.2.0/siremis/files/cache/metadata/CB3F6C48_log Service.cmp): failed to open stream" I have down a chown on the siremis dir to make the web process the owner, and this allowed me to get to the last page of the install, all shows as successful expect instead of a login page I get the error. The server platform is centos if makes any difference. Thanks Magnus

2 2

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Daniel-Constantin Mierla 15 May '12

15 May '12

Hello, interesting that the output of bt shows inline core_hash function prototype with same parameters as publ_cback_func, not being able to spot eventual wrong parameter values. Anyhow, in the gdb, can you do the commands: frame 1 p *hentity and send the output? Let's see the value of the pub uri for which hashing function crashes. What are the parameters for pua and pua_dialog info you set in the config file? Cheers, Daniel On 5/15/12 11:53 AM, Charles Chance wrote: > > Hi Daniel, > > As per your previous reply (sorry, I deleted it accidentally), please > find attached output of 'bt full' as requested. > > Best regards and thanks again, > > Charles > > ------------------------------------------------------------------------ > > *From:*Charles Chance [mailto:charles.chance@sipcentric.com] > *Sent:* 13 May 2012 11:21 > *To:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) > -Users Mailing List' > *Subject:* (Devel) Segmentation fault using pua_dialoginfo > > Hi, > > Currently testing devel version and experiencing the following when > using pua_dialoginfo: > > ------------------------------------- > > Program terminated with signal 11, Segmentation fault. > > #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > ../../hashes.h:279 > > 279 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; > > (gdb) bt > > #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > ../../hashes.h:279 > > #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > send_publish.c:246 > > #2 0x002861d6 in run_trans_callbacks_internal (cb_lst=0xb3d7e370, > type=1024, trans=0xb3d7e330, params=0xbfcc31c8) at t_hooks.c:290 > > #3 0x002864d6 in run_trans_callbacks (type=1024, trans=0x200, > req=0x0, rpl=0xb7bea008, code=200) at t_hooks.c:317 > > #4 0x002afb28 in local_reply (t=0xb3d7e330, p_msg=0xb7bea008, > branch=0, msg_status=200, cancel_data=0xbfcc345c) at t_reply.c:2001 > > #5 0x002b2b45 in reply_received (p_msg=0xb7bea008) at t_reply.c:2350 > > #6 0x0809d5f4 in forward_reply (msg=0xb7bea008) at forward.c:790 > > #7 0x080e11a9 in receive_msg ( > > buf=0x82c4fa0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP > xx.xx.xx.xxx;branch=z9hG4bKc13b.25313b6.0\r\nTo: > sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2\r\nFrom: > sip:000094@xx.xx.xx.xxx;tag=4cc17c2c75cb82fa"..., len=393, > rcv_info=0xbfcc3648) at receive.c:270 > > #8 0x0817801b in udp_rcv_loop () at udp_server.c:544 > > #9 0x080b0426 in main_loop () at main.c:1633 > > #10 0x080b3d02 in main (argc=11, argv=0xbfcc3924) at main.c:2546 > > (gdb) up > > #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > send_publish.c:246 > > 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); > > (gdb) p hentity->pres_uri > > Cannot access memory at address 0x8 > > (gdb) p *hentity > > Cannot access memory at address 0x0 > > ------------------------------------- > > Your time is greatly appreciated... > > Charles > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

kamdbctl create changing mysql root password
by Tim King 15 May '12

15 May '12

This is not the first time I have had this issue during installation, but this time around I am having difficulty figuring out what I am doing wrong. Mysql is up and working, and client works. I can connect to it with the client and run queries without issue. When I run kamdbctl create I get the following. INFO: test server charset INFO: creating database openser ... ERROR 1045 (28000): Access denied for user 'root'@'localhost' (using password: YES) ERROR: Creating core tables failed! It creates the database but than fails to create tables. From this point I can no longer connect to mysql with what was my root password. I have to go through the process of resetting my root password before I can log back into mysql. How is this script chaning or corrupting my mysql root password? This is the relevant content of my kamctlrc file: # $Id$ # # The Kamailio configuration file for the control tools. # # Here you can set variables used in the kamctl and kamdbctl setup # scripts. Per default all variables here are commented out, the control tools # will use their internal default values. ## your SIP domain SIP_DOMAIN=sbc.mydomain.net ## chrooted directory # $CHROOT_DIR="/path/to/ chrooted/directory" ## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, DBTEXT, or SQLITE # by default none is loaded # # If you want to setup a database with kamdbctl, you must at least specify # this parameter. DBENGINE=MYSQL ## database host DBHOST=localhost ## database name (for ORACLE this is TNS name) DBNAME=openser # database path used by dbtext, db_berkeley or sqlite # DB_PATH="/usr/local/etc/kamailio/dbtext" ## database read/write user DBRWUSER=root ## password for database read/write user DBRWPW="mypassword" ## database read only user DBROUSER=root ## password for database read only user DBROPW="mypassword" ## database super user (for ORACLE this is 'scheme-creator' user) DBROOTUSER="root"

2 1

kamdbctl create changing mysql root password
by Tim King 15 May '12

15 May '12

This is not the first time I have had this issue during installation, but this time around I am having difficulty figuring out what I am doing wrong. Mysql is up and working, and client works. I can connect to it with the client and run queries without issue. When I run kamdbctl create I get the following. INFO: test server charset INFO: creating database openser ... ERROR 1045 (28000): Access denied for user 'root'@'localhost' (using password: YES) ERROR: Creating core tables failed! It creates the database but than fails. From this point I can no longer connect to mysql with what was my root password. I have to go through the process of resetting my root password before I can log back into mysql. How is this script chaning or corrupting my mysql root password? This is the relevant content of my kamctlrc file: # $Id$ # # The Kamailio configuration file for the control tools. # # Here you can set variables used in the kamctl and kamdbctl setup # scripts. Per default all variables here are commented out, the control tools # will use their internal default values. ## your SIP domain SIP_DOMAIN=sbc.mydomain.net ## chrooted directory # $CHROOT_DIR="/path/to/chrooted/directory" ## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, DBTEXT, or SQLITE # by default none is loaded # # If you want to setup a database with kamdbctl, you must at least specify # this parameter. DBENGINE=MYSQL ## database host DBHOST=localhost ## database name (for ORACLE this is TNS name) DBNAME=openser # database path used by dbtext, db_berkeley or sqlite # DB_PATH="/usr/local/etc/kamailio/dbtext" ## database read/write user DBRWUSER=root ## password for database read/write user DBRWPW="mypassword" ## database read only user DBROUSER=root ## password for database read only user DBROPW="mypassword" ## database super user (for ORACLE this is 'scheme-creator' user) DBROOTUSER="root"

1 0

Kamailio CS_Dirty
by Albert Petit 15 May '12

15 May '12

Hello, On a Kamailio deployment it seems some users get to a wrong state and permanently keeping in: CS_DIRTY Expires==0 for some long times (even half an hour). When they're in this status seems system cannot reach them. Normally , when working on USRLoc mode 2 if the registration expires usually what I observed is that it just not appears on 'kamctl ul show' The affected users are seen like this: AOR:: blue4 Contact:: sip:Blue4@192.168.101.6:5700 Q= Expires:: deleted Callid:: efc560dc2dbd248d4d1f9d39633773df Cseq:: 49995 User-agent:: PoC-client/OMA1.0 Genaker_X-Platform_IMS_SDK/v2.0 State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp:10.133.21.141:6666 Methods:: 8143 The refresh from the clients seem to come intime however some users (and not all) got to this CS_DIRTY state instead of being removed or refreshed. The client refresh looks to arrive in-time so it should be refreshed. Which could be a common cause (misconfiguration?) for this type of problem? If not, could enable debug mode for usrloc or I need to enable for full kamailio? Thanks Albert

1 0

Question about dialog module (state 1)
by Ricardo Martinez 15 May '12

15 May '12

Hello. I’m using the dialog module to keep control of simultaneous calls. In some cases, and I’m still trying to find why this happens, the dialog stays in “STATE:: 1”, which according to the docs is a dialog which no provisional response has been sent yet. Is there a way to eliminate this kind of dialogs??? When I list the dialogs I can see : dialog:: hash=2523:2040584107 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: 25b8e5354ca0850a4d0bd9c43eeb617a(a)pxext.redvoiss.net from_uri:: sip:557100052213@pxext.redvoiss.net from_tag:: as10dc7c85 caller_contact:: sip:557100052213@200.68.19.44:5060 caller_cseq:: 103 caller_route_set:: caller_bind_addr:: udp:64.76.154.110:5060 callee_bind_addr:: to_uri:: sip:0981985480@pxext.redvoiss.net to_tag:: callee_contact:: callee_cseq:: callee_route_set:: when I try to eliminate this dialog : kamctl fifo dlg_terminate_dlg 25b8e5354ca0850a4d0bd9c43eeb617a(a)pxext.redvoiss.net 500 command 'dlg_terminate_dlg' failed Thanks in advance, Regards, Ricardo Martinez.-

2 4

[SIREMIS] send MI Commands to multiple kamailio servers
by SamyGo 15 May '12

15 May '12

Hello all, Following through the manual at : http://kb.asipto.com/siremis:install32x:mi-commands I'm able to send MI commands on external/remote server But next hurdle is how can I define multiple remote servers i.e <Remote name="remote0" address="127.0.0.1" port="8033"/> <Remote name="remote1" address="192.168.2.156" port="8033"/> <Remote name="remote2" address=" 192.168.2.150" port="8033"/> Is there any already working configuration else I think I may need to start a surgery on siremisMICommands.php file and make it loop over the remote tag and disptach commands until loop ends. is there any "MI-proxy" sort of thing available !!? Regards, Sammy.

3 5

Pickup group implementation in Kamailio
by M.C. 15 May '12

15 May '12

Hi all, I want to implement a "group pickup" (asterisk like) with Kamailio and Snom phones. I found in the changelog this comment: "- an useful application - call pickup: - INVITE comes in, callee is in a pickup group - store callid, cseq, etc in database or notify the other phones in the group (easy to do for snom phones and extra programmable buttons) via uac_req_send() - send a remote control command telling which call to pickup (callid, cseq) and where to redirect. In config, call t_cancel_callid(). in failure route for INVITE catch the cancelled transaction (flag is set), get the new destination (e.g., stored in htable) and forward the invite there"" regarding new functions in tmx module, but I have found no examples in kamailio documentation. Is this the way to implement a group pickup in kamailio? Regards, Matteo

2 2

FW: (Devel) Segmentation fault using pua_dialoginfo
by Charles Chance 15 May '12

15 May '12

Hi Daniel, As per your previous reply (sorry, I deleted it accidentally), please find attached output of 'bt full' as requested. Best regards and thanks again, Charles _____ From: Charles Chance [mailto:charles.chance@sipcentric.com] Sent: 13 May 2012 11:21 To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List' Subject: (Devel) Segmentation fault using pua_dialoginfo Hi, Currently testing devel version and experiencing the following when using pua_dialoginfo: ------------------------------------- Program terminated with signal 11, Segmentation fault. #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at ../../hashes.h:279 279 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; (gdb) bt #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at ../../hashes.h:279 #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at send_publish.c:246 #2 0x002861d6 in run_trans_callbacks_internal (cb_lst=0xb3d7e370, type=1024, trans=0xb3d7e330, params=0xbfcc31c8) at t_hooks.c:290 #3 0x002864d6 in run_trans_callbacks (type=1024, trans=0x200, req=0x0, rpl=0xb7bea008, code=200) at t_hooks.c:317 #4 0x002afb28 in local_reply (t=0xb3d7e330, p_msg=0xb7bea008, branch=0, msg_status=200, cancel_data=0xbfcc345c) at t_reply.c:2001 #5 0x002b2b45 in reply_received (p_msg=0xb7bea008) at t_reply.c:2350 #6 0x0809d5f4 in forward_reply (msg=0xb7bea008) at forward.c:790 #7 0x080e11a9 in receive_msg ( buf=0x82c4fa0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP xx.xx.xx.xxx;branch=z9hG4bKc13b.25313b6.0\r\nTo: sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2\r\nFrom: sip:000094@xx.xx.xx.xxx;tag=4cc17c2c75cb82fa"..., len=393, rcv_info=0xbfcc3648) at receive.c:270 #8 0x0817801b in udp_rcv_loop () at udp_server.c:544 #9 0x080b0426 in main_loop () at main.c:1633 #10 0x080b3d02 in main (argc=11, argv=0xbfcc3924) at main.c:2546 (gdb) up #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at send_publish.c:246 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); (gdb) p hentity->pres_uri Cannot access memory at address 0x8 (gdb) p *hentity Cannot access memory at address 0x0 ------------------------------------- Your time is greatly appreciated. Charles

1 0

modify connection information field from a 200 OK response
by Openser Kamailio 15 May '12

15 May '12

hi, I try to change the IP address in connection information field from a 200 OK request passing through Kamailio. I want the IP address of the Kamailio in connection information. So, I've tried rtpproxy_answer("","@IP kamailio"), rtppproxy_offer("","@IP kamailio), rtpproxy_manage("","@IP kamailio"). It didn't change anything. Thanks for Help!

2 4

why is tcp_send failing?
by Juha Heinanen 15 May '12

15 May '12

i have been wondering why kamailio refuses to forward bye to $du sip:192.26.99.39:8000;transport=tcp;lr;ftag=a8266461a81c723o0 wireshark shows that it is not even trying to set up tcp connection to the destination and with telnet i can verify that tcp connection can be set up. debug at level 3 is below. -- juha May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: rr [loose.c:776]: URI to be processed: 'sip:192.26.99.39:8000;transport=tcp;lr;ftag=a8266461a81c723o0' May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: rr [loose.c:785]: Next URI is a loose router ... May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=5 , global msg id=5 , T on entrance=(nil) May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_lookup.c:527]: t_lookup_request: start searching: hash=30919, isACK=0 May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb486de58, callback type 1, id 0 entered May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: <core> [md5utils.c:67]: DEBUG: MD5 calculated: d28c87700c1c48b71a3b7f63090bf160 May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: <core> [dns_cache.c:3400]: dns_sip_resolve(192.26.99.39, 0, 0), ip, ret=0 May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: <core> [msg_translator.c:204]: check_via_address(192.98.100.135, 192.168.0.145, 0) May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: ERROR: tm [../../forward.h:171]: msg_send: ERROR: tcp_send failed May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_fwd.c:1365]: t_send_branch: send to 192.26.99.39:8000 (2) failed May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: ERROR: tm [t_fwd.c:1383]: ERROR: t_send_branch: sending request on branch 0 failed May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_funcs.c:361]: ERROR:tm:t_relay_to: t_forward_nonack returned error May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_funcs.c:369]: -477 error reply generation delayed

2 5

Re: [SR-Users] (Devel) Segmentation fault using pua_dialoginfo
by Daniel-Constantin Mierla 15 May '12

15 May '12

Hello, to get some info until I get the time to look more deeper, can you send also the output of 'bt full'? It will print also other variables that may help troubleshooting faster. Cheers, Daniel On 5/13/12 12:21 PM, Charles Chance wrote: > > Hi, > > Currently testing devel version and experiencing the following when > using pua_dialoginfo: > > ------------------------------------- > > Program terminated with signal 11, Segmentation fault. > > #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > ../../hashes.h:279 > > 279 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; > > (gdb) bt > > #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > ../../hashes.h:279 > > #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > send_publish.c:246 > > #2 0x002861d6 in run_trans_callbacks_internal (cb_lst=0xb3d7e370, > type=1024, trans=0xb3d7e330, params=0xbfcc31c8) at t_hooks.c:290 > > #3 0x002864d6 in run_trans_callbacks (type=1024, trans=0x200, > req=0x0, rpl=0xb7bea008, code=200) at t_hooks.c:317 > > #4 0x002afb28 in local_reply (t=0xb3d7e330, p_msg=0xb7bea008, > branch=0, msg_status=200, cancel_data=0xbfcc345c) at t_reply.c:2001 > > #5 0x002b2b45 in reply_received (p_msg=0xb7bea008) at t_reply.c:2350 > > #6 0x0809d5f4 in forward_reply (msg=0xb7bea008) at forward.c:790 > > #7 0x080e11a9 in receive_msg ( > > buf=0x82c4fa0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP > xx.xx.xx.xxx;branch=z9hG4bKc13b.25313b6.0\r\nTo: > sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2\r\nFrom: > sip:000094@xx.xx.xx.xxx;tag=4cc17c2c75cb82fa"..., len=393, > rcv_info=0xbfcc3648) at receive.c:270 > > #8 0x0817801b in udp_rcv_loop () at udp_server.c:544 > > #9 0x080b0426 in main_loop () at main.c:1633 > > #10 0x080b3d02 in main (argc=11, argv=0xbfcc3924) at main.c:2546 > > (gdb) up > > #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > send_publish.c:246 > > 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); > > (gdb) p hentity->pres_uri > > Cannot access memory at address 0x8 > > (gdb) p *hentity > > Cannot access memory at address 0x0 > > ------------------------------------- > > Your time is greatly appreciated... > > Charles > > > > _______________________________________________ > 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

1 0

Extending app_python (patch)
by Konstantin M. 15 May '12

15 May '12

Hi, As I see, app_python module has almost no logging functionality except LM_ERR. As a python developer and user I need a more python functionality. did a patch (attached) against kamailio-3.2.3 for adding an additional logging and I'd like to share it. Testcase: 0(5403) INFO: app_python [python_iface.c:41]: Loggers.py: LM_GEN1: msg: Test Message 0(5403) WARNING: app_python [python_iface.c:60]: Loggers.py: LM_GEN2: msg: Test Message 0(5403) ALERT: app_python [python_iface.c:76]: Loggers.py: LM_ALERT: msg: Test Message 0(5403) CRITICAL: app_python [python_iface.c:92]: Loggers.py: LM_CRIT: msg: Test Message 0(5403) ERROR: app_python [python_iface.c:108]: Loggers.py: LM_ERR: msg: Test Message 0(5403) WARNING: app_python [python_iface.c:124]: Loggers.py: LM_WARN: msg: Test Message 0(5403) NOTICE: app_python [python_iface.c:140]: Loggers.py: LM_NOTICE: msg: Test Message 0(5403) INFO: app_python [python_iface.c:156]: Loggers.py: LM_INFO: msg: Test Message

2 1

Re: [SR-Users] pua_dialoginfo - publish for caller or callee
by charles.chance＠sipcentric.com 15 May '12

15 May '12

2 1

Re: [SR-Users] pua_dialoginfo - publish for caller or callee
by Daniel-Constantin Mierla 14 May '12

14 May '12

Hello, the upcoming 3.3 has some options to set the URIs for which to send the PUBLISH, see: http://kamailio.org/docs/modules/devel/modules_k/pua_dialoginfo.html#id2534… You can decide in config file how many publishes and where to be sent. But these are additions after 3.2. Cheers, Daniel On 5/12/12 3:38 PM, Charles Chance wrote: > > Hi, > > Quick question relating to pua_dialoginfo (3.2) - is there a way to > specify for which side (caller/callee) a PUBLISH is generated, or must > it always be for both? > > Cheers, > > Charles > > > > _______________________________________________ > 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

1 0

kamdbctl error in Ubuntu 12.04
by Ryan Gholam 14 May '12

14 May '12

I am trying to install Kamailio with ( Sudo apt-get install ...) without recompiling etc .... , i reach a point where :> /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL > ERROR: database engine not loaded - tried 'MYSQL' any ideas how to solve this prob ? Thank you

2 4

TM Callback Error
by Reda Aouad 14 May '12

14 May '12

Hi, After upgrading from 3.2 to 3.3, I am getting the following error: ERROR: dialog [dlg_handlers.c:937]: registering TMCB to wait for negative ACK This is my version: kamailio -V version: kamailio 3.3.0-pre1 (i386/linux) 275c8a 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: 275c8a compiled on 13:14:05 May 6 2012 with gcc 4.5.2 And idea? Thanks Reda

3 3

Consulting needed
by Pete Ashdown 14 May '12

14 May '12

I have been trying to accomplish a couple tasks with Kamailio over the past month with no luck. What I need is a bit of one-on-one training with someone who knows the lay of the land. If you do this kind of consulting and can use Skype with possibly a shared-screen terminal, please drop me an email with your rate.

4 5

Kamailio SIP server and Cisco 7971G-GE
by KA Veelenturf 14 May '12

14 May '12

Hello everyone, I am trying to get a new SIP firmware version on our Cisco 7971G-GE. Now the problem is that I don't really know which configuration files have to be included to get everything working. For now on I have the new SIP firmware and SEP<mac>cnf.xml and the XMLDefault.cnf.xml files. Does anybody have a template for the .xml files to get a new firmware version working on our Cisco and also to get it work with Kamailio? With kind regards, Koen.

2 1

(Devel) Segmentation fault using pua_dialoginfo
by Charles Chance 13 May '12

13 May '12

Hi, Currently testing devel version and experiencing the following when using pua_dialoginfo: ------------------------------------- Program terminated with signal 11, Segmentation fault. #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at ../../hashes.h:279 279 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; (gdb) bt #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at ../../hashes.h:279 #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at send_publish.c:246 #2 0x002861d6 in run_trans_callbacks_internal (cb_lst=0xb3d7e370, type=1024, trans=0xb3d7e330, params=0xbfcc31c8) at t_hooks.c:290 #3 0x002864d6 in run_trans_callbacks (type=1024, trans=0x200, req=0x0, rpl=0xb7bea008, code=200) at t_hooks.c:317 #4 0x002afb28 in local_reply (t=0xb3d7e330, p_msg=0xb7bea008, branch=0, msg_status=200, cancel_data=0xbfcc345c) at t_reply.c:2001 #5 0x002b2b45 in reply_received (p_msg=0xb7bea008) at t_reply.c:2350 #6 0x0809d5f4 in forward_reply (msg=0xb7bea008) at forward.c:790 #7 0x080e11a9 in receive_msg ( buf=0x82c4fa0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP xx.xx.xx.xxx;branch=z9hG4bKc13b.25313b6.0\r\nTo: sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2\r\nFrom: sip:000094@xx.xx.xx.xxx;tag=4cc17c2c75cb82fa"..., len=393, rcv_info=0xbfcc3648) at receive.c:270 #8 0x0817801b in udp_rcv_loop () at udp_server.c:544 #9 0x080b0426 in main_loop () at main.c:1633 #10 0x080b3d02 in main (argc=11, argv=0xbfcc3924) at main.c:2546 (gdb) up #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at send_publish.c:246 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); (gdb) p hentity->pres_uri Cannot access memory at address 0x8 (gdb) p *hentity Cannot access memory at address 0x0 ------------------------------------- Your time is greatly appreciated. Charles

1 0

pua_dialoginfo - publish for caller or callee
by Charles Chance 12 May '12

12 May '12

Hi, Quick question relating to pua_dialoginfo (3.2) - is there a way to specify for which side (caller/callee) a PUBLISH is generated, or must it always be for both? Cheers, Charles

1 0

change/replace anything in body message
by Konstantin M. 12 May '12

12 May '12

Hi, Is there a way to change/replace anything in body message ? Used: replace_body_atonce, replace_body_all, replace_all, subst_body. Nothing is working. Thanks,

2 2

(no subject)
by Albert Petit 11 May '12

11 May '12

Hello, On a Kamailio deployment it seems some users get to a wrong state and permanently keeping in: CS_DIRTY Expires==0 for some long times (even half an hour). When they're in this status seems system cannot reach them. Normally , when working on USRLoc mode 2 if the registration expires usually what I observed is that it just not appears on 'kamctl ul show' The affected users are seen like this: AOR:: blue4 Contact:: sip:Blue4@192.168.101.6:5700 Q= Expires:: deleted Callid:: efc560dc2dbd248d4d1f9d39633773df Cseq:: 49995 User-agent:: PoC-client/OMA1.0 Genaker_X-Platform_IMS_SDK/v2.0 State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp:10.133.21.141:6666 Methods:: 8143 The refresh from the clients seem to come intime however some users (and not all) got to this CS_DIRTY state instead of being removed or refreshed. The client refresh looks to arrive in-time so it should be refreshed. Which could be a common cause (misconfiguration?) for this type of problem? If not, could enable debug mode for usrloc or I need to enable for full kamailio? Thanks Albert

1 0

Kamailio ignores some ACK
by Efelin Novak 11 May '12

11 May '12

Hi folks, I have a strange problem when Kamailio ignores ACKs in a specific scenario. The call flow is as follows: A -> INVITE -> kamailio -> INVITE -> B [omitting 100 and 180] A <- 200 OK <- kamailio <- 200 OK <- B A -> ACK -> kamailio There are INVITE Xlogs, Reply ROUTE xlogs and media-proxy logs in the syslog. However there is no information about these ACKs. No XLOGs are printed even if there is one on the top of the main route. "tcpdump -A -s0 -i any -n port 5060" receives this message correctly: 14:47:01.246153 IP 111.111.11.11.5060 > 80.80.80.80.60442: SIP, length: 915 SIP/2.0 200 OK Via: SIP/2.0/UDP 111.111.11.11:5060;rport=60442;x-route-tag="tgrp:A";branch=z9hG4bK1634E6A88 Record-Route: <sip:111.111.11.11;lr;ftag=599248D4-260;vsf=AAAAAAAAAAAAAAAAAAAAW0FVT0ZWHF1aNy4xGzA-;nat=yes;did=3bb.327c47e6> Contact: <sip:80.80.80.80:65002;transport=udp> To: "test_account"<sip:bob@server.com>;tag=cb7dd641 From: <sip:alice@111.111.11.50>;tag=599248D4-260 Call-ID: 9AFCFC51.11.50 CSeq: 101 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE Content-Type: application/sdp Content-Length:263 v=0 o=- 492575093 492575093 IN IP4 111.111.11.60 s=test_device i=(o=IN IP4 192.168.1.10) c=IN IP4 111.111.11.71 t=0 0 m=audio 16416 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 14:47:01.254511 IP 111.111.11.50.60442 > 111.111.11.11.5060: SIP, length: 521 ACK sip:80.80.80.80:65002;transport=udp SIP/2.0 Via: SIP/2.0/UDP 111.111.11.50:5060;x-route-tag="tgrp:A";branch=z9hG4bK1634E7DE8 From: <sip:alice@111.111.11.50>;tag=599248D4-260 To: "test_account"<sip:bob@server.com>;tag=cb7dd641 Call-ID: 9AFCFC51.11.50 Route: <sip:111.111.11.11;lr;ftag=599248D4-260;vsf=AAAAAAAAAAAAAAAAAAAAW0FVT0ZWHF1aNy4xGzA-;nat=yes;did=3bb.327c47e6> Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 My Kamailio version is kamailio 3.1.0 (i386/linux) 1e204f. Does anybody knows where can be a problem? How can I check whether Kamailio receives something? ... Jan

5 6

error searching pvar "sp."
by Konstantin M. 11 May '12

11 May '12

Hello, When I have the following code: xlog("Dropped INVITE from unregistered location: $si:$sp.\n"); -- I'm getting a parse error: 0(3735) ERROR: <core> [pvapi.c:708]: error searching pvar "sp." 0(3735) ERROR: <core> [pvapi.c:912]: wrong char [./46] in [$sp. ] at [3 (0)] 0(3735) ERROR: xlog [xlog.c:314]: wrong format[Dropped INVITE from unregistered location: $si:$sp. ] When I changed to (removed a point after $sp): xlog("Dropped INVITE from unregistered location: $si:$sp\n"); -- No errors. What I'm doing wrong ? Or it's a parser bug ? Thanks!

2 1

t_continue usage
by Vineet Menon 11 May '12

11 May '12

Hi, the module doc page http://www.kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2525893says t_continue() has 3rd parameter which says which route block to execute. SO does that mean t_continue() should be present before routing logic(s)?? Where should be t_continue() be placed in a config?? Regards, Vineet Menon

2 1

Community dinner: Barcelona June 13th
by Olle E. Johansson 10 May '12

10 May '12

Hello! I will be running an Asterisk SIP Masterclass - the last one - in Barcelona in June. During this week, I will organize a dinner for everyone working with or interested in Asterisk, Kamailio and other Open Source platforms for realtime communication. It's Wednesday, June 13th somewhere in Barcelona - location will be announced later. You pay our own dinner (unless we can find sponsors) and enjoy the geeky company for free! To join the event, use this Facebook event https://www.facebook.com/events/307548349321608/ See you in Barcelona! /O ------ http://edvina.net - Open Unified Communication - training, consulting, workshops

1 0

kamailio - mysql master-slave connection question
by Uri Shacked 10 May '12

10 May '12

Hi, In my system configuration, i use 2 DB servers. One will be the master and the other is the slave. In kamailio cfg, I set the mysql server as a host name (not ip). On my DNS server the ip behind the name is the master`s address. I wonder, if I stop the mysqld on the master, and change the ip address in the DNS to the slave`s address. Than i do "kamctl dispatcher reload"... Will kamailio go to the DNS, find the address and connect to the slave? BR, Uri

1 0

[SIREMIS] Multiple kamailio management commands
by SamyGo 10 May '12

10 May '12

Hello list, I'm trying to have my Siremis interface send MI commands to multiple kamailio servers i.e reload dispatcher of all the kamailio servers when I reload from Siremis interface. The issue Im facing is that the commands dont get executed on any other server except localhost/ as mentioned in URL: http://kb.asipto.com/siremis:install32x:mi-commands I've edited the <Remote name="remote" address="127.0.0.1" port="8033"/> to : <Remote name="remote" address="192.168.2.156" port="8033"/> Please guide on how to have multiple kamailio receive command from single Siremis interface. Regards, Sammy.

2 8

Re: [SR-Users] Bug: Pua_reginfo
by Gnaneshwar Gatla 10 May '12

10 May '12

Hello, I have noticed that user records were not being correctly saved in the location table. I have tried to fix this will the attached code. This would fix reginfo_handle_notify() to store the user-record in the right format. This fixes part of the problem, I'm still trying to fix the incorrect uri in the PUBLISH message generated by this module. Regards, Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199 From: Gnaneshwar Gatla Sent: Tuesday, May 08, 2012 4:58 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Carsten Bock (carsten(a)ng-voice.com) Subject: Bug: Pua_reginfo Hello all, There seems to be an deadlock between registrar and pua_reginfo module whenever a SIP client refreshes its registration. 1. The pua_reginfo does not publish Event:reg when a sip client registers for the first time. 2. The pua_reginfo module does not use ul.lock_udomain() to lock the ul table to get the record. 3. The pua_reginfo does not release the record that it accessed from ul.get_urecord() Since the record is not released, when the client re-registers, the registrar module is unable to query for the userlocation and sends a 200 OK response with 0 bindings. Which leads to no contact header in the response message and eventually deletion of the record in the location table(expires). I have tried to fix this using lock_udomain, release_urecord and unlock_udomain, but failed to rectify the error. The deadlock happens in modules_k/usrloc/udomain.c (db_load_urecord()). if (ul_dbf.query(_c, keys, 0, vals, columns, (use_domain)?2:1, 13, order, &res) < 0) My Question: 1. is pua_reginfo supposed to delete record from location database and send the same data to a presence server? a. If above is true, then should the registrar module insert a new record in the location table? b. If above is false, the registrar module should be able to refresh/update the record in the location table. This error is also impacts pua_usrloc module too as it cannot access the record and sends creates an incorrect uri. Help: I'm new and learning how Kamailio works, is there a clean way that pua_reginfo module gets the record? Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

1 0

Problem with radius_www_authorize
by Ricardo Martinez 09 May '12

09 May '12

Hello. We have detected a behavior not specified in the docs. I ask for support to solve this issue. We are using a Radius system to authorize our clients, so, for REGISTER we use the command radius_www_authorize(“mydomain.com”) According to the documentation the possible answers are : Negative result codes may be interpreted as follows: -7 (internal error) - some internal error occurred (see syslog); -6 (nonce reused) - nonce is used more than once; -5 (no credentials) - credentials were not found in request; -4 (stale nonce) - stale nonce; -3 (authorization failed) - RADIUS responded with Access Reject which may be, for example, due to user not found or wrong password; -1 (error) - some error occured during authorization (see syslog); Despite of this, for BadPassword or UserNotExist from the radius answer the radius_www_authorize command return “-2” as a retcode. So, is the documentation missing something? Or I’m doing something wrong? Thanks Ricardo Martinez.-

2 2

Config include
by Konstantin M. 09 May '12

09 May '12

Hi, I would like (and a many people here I believe) to have a functional of including a multiple config files like (foe example asterisk's #include "path/to/some/config.conf"). Is it possible to implement a such feature ? Thanks!

4 7

Rtpproxy issue with connection information
by Openser Kamailio 09 May '12

09 May '12

Hi, i'm currently working with kamailio 3.2 and rtpproxy 1.2.1. Both are set up on the same computer. When rtpproxy adds an SDP to an Invite, it adds two IPv4 addresses in owner/creator session and connection information field with an error, i.e: *Owner/Connection Information (o)*: doubango 1983 678901 IN IP4 * 172.27.170.984* 172.27.170.98 *Connection Information (c)*: IN IP4 *172.27.170.984* 172.27.170.98 the correct ones should be : *Owner/Connection Information (o)*: doubango 1983 678901 IN IP4 172.27.170.98 *Connection Information (c)*: IN IP4 172.27.170.98 Thanks for help!!

3 3

Alias configuration with multidomain
by Arjan Kuiken 09 May '12

09 May '12

Hello SR-users, I want to configure a kamailio 3.2.3. We want to use it to connect multiple SIP servers as users and let the Kamailio work as a Trunk/ Proxy connected to a PSTN gateway. We have multiple domains and aliases per SIP servers. When I register a SIP server to Kamailio everything goes well, but when I try to call an dbalias it doesn't work. It gives 2 errors (trace included). I think something is done wrong in my kamailio.cfg but I don't have a clue anymore. I included the cfg file as attachement. Can you please have a look? Thanks in advance! Kind regards, Arjan Kuiken Technical Specialist [intercity-zakelijk-logo-handetekening (2)] Kruisweg 659, 2132 NC, Hoofddorp T: 020 - 655 3000 F: 020 - 6531534 E: Arjan.kuiken(a)intercityzakelijk.nl<mailto:arjan.kuiken@intercityzakelijk.nl> I: www.intercityzakelijk.nl<http://www.intercityzakelijk.nl/>

2 2

Accounting '180 Ringing' without SDP
by Stephen Dodge (Bistech) 09 May '12

09 May '12

Hello, We are running Kamailio 3.1.5 and using the acc module to generate CDRs and provide reporting information. We have a requirement to report ring time and we do this by accounting on early media for 183 provisional responses. Unfortunately we have a couple of customer PBX's that cannot provide a SDP with the ringing response. i.e they just respond with a '180 ringing'. My understanding of the module is that the "early_media" parameter is looking for a SDP in either a 180 or 183 response before accounting. Although it may be possible to account the 180 without SDP using the acc module I cannot find anything obvious in the documentation. If it is possible could someone point me in the right direction or suggest a creative solution, otherwise can I ask that this functionality be introduced. Many Thanks. Steve Dodge. ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

2 1

detect retransmission messages...
by Vineet Menon 09 May '12

09 May '12

Hi, How do we detect a message as retransmitted message, in a transaction oriented proxy? Regards, Vineet Menon

2 1

Re: [SR-Users] detect retransmission messages...
by Alex Balashov 09 May '12

09 May '12

t_check_trans() generally does the trick. -- 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 Vineet Menon <mvineetmenon(a)gmail.com> wrote: >Hi, > >How do we detect a message as retransmitted message, in a transaction >oriented proxy? > >Regards, > >Vineet Menon > >_______________________________________________ >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

kamdbctl
by Klaus Darilion 09 May '12

09 May '12

Hi! I wonder what is the status of kamdbctl tool for 'migrate' command - should it work? The description says: kamdbctl migrate <old_db> <new_db> .....................(migrates DB from 1.2 to 1.3, not implemented yet!) This seems a bit outdated. regards Klaus

3 3

kamdbctl error
by Vineet Menon 09 May '12

09 May '12

Hi, I am compiling sources of kamailio 3.2.3 on ubuntu 12.04. I have used make command > `make group_include="standard mysql" all` > to make the sources. and then > `make install`..... > Now when i do kamdbctl it throws an error, $ kamdbctl create > ERROR: could not load the script in > /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL > ERROR: database engine not loaded - tried 'MYSQL' > Any guess what is the problem??? Regards, Vineet Menon

3 7

Kamailio [No Audio]
by Fred Flintsone 09 May '12

09 May '12

I am attempting to route local registered users to local registered users without going to the media server. I have a media server [*] for PSTN. But it doesn't support all the CODECS i wan't to use. Signalling seems fine, but i get NO AUDIO on either call leg. I have the RTP captures from the local side making the call and the media packets seem to be going exactly where the SDP dictates. So i'm a little confused. Is this where an RTP Proxy would come in handy? I haven't been able to get the audio to work without going to the media server. If an RTP Proxy is the answer, how much overhead does the proxy add to the Kamailio server? Is it something that you don't run on the same machine and use a distributed environment of RTP proxies on other servers? Or is this something that should be fine and working without a media proxy? I really appreciate any help! Thank you, Fred

2 1

Bug: Pua_reginfo
by Gnaneshwar Gatla 08 May '12

08 May '12

Hello all, There seems to be an deadlock between registrar and pua_reginfo module whenever a SIP client refreshes its registration. 1. The pua_reginfo does not publish Event:reg when a sip client registers for the first time. 2. The pua_reginfo module does not use ul.lock_udomain() to lock the ul table to get the record. 3. The pua_reginfo does not release the record that it accessed from ul.get_urecord() Since the record is not released, when the client re-registers, the registrar module is unable to query for the userlocation and sends a 200 OK response with 0 bindings. Which leads to no contact header in the response message and eventually deletion of the record in the location table(expires). I have tried to fix this using lock_udomain, release_urecord and unlock_udomain, but failed to rectify the error. The deadlock happens in modules_k/usrloc/udomain.c (db_load_urecord()). if (ul_dbf.query(_c, keys, 0, vals, columns, (use_domain)?2:1, 13, order, &res) < 0) My Question: 1. is pua_reginfo supposed to delete record from location database and send the same data to a presence server? a. If above is true, then should the registrar module insert a new record in the location table? b. If above is false, the registrar module should be able to refresh/update the record in the location table. This error is also impacts pua_usrloc module too as it cannot access the record and sends creates an incorrect uri. Help: I'm new and learning how Kamailio works, is there a clean way that pua_reginfo module gets the record? Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

1 0

SDPOPS in siremis how to?
by Grégoire Vandendeurpel 08 May '12

08 May '12

Hello, Thx already for your first explanation. I'm going to describe more precisaly what is my purpose and what I need: About my final project: On one physical machine, I use 4 virtual machines, which are : one with kamailio 3.2.3, asterisk 1.6.13, odbc and Siremis 3.2 An other one is a first client in an other virtual subnet as kamailio.. which reprisents the "public network" ! The two others virtual machines are in the same subnet as Kamailio which represent the "private network" ( like a company and the public network) So, everything in the private network works well, with TLS, Siremis and asterisk, perfect communication! I still need to make the route between the two subnet but this shoud be easy and what I need precisly is to make a filter on SDP media managed by Siremis like: This is why I need to use the sdpops module of Kamailio, of course. I need to make the possibilities in Siremis to set things like : this subnet or this address -> remove this media like application, video , only audio can be allow ! This is the filter I need to make it work. Of course in Siremis because , I m simulating calls from public network and into the private network of a company . And it as to be user friendly, not spacially for a computer scientist. This is why, I need to make this simple filter on the media of sdp, and user friendly in Siremis. Thank u in advance for your help Best regards -- *Grégoire Vandendeurpel, * * * *IT Sector*

1 0

routing logic
by Vineet Menon 08 May '12

08 May '12

Hi, Where can i find a nice tutorial for routing logic in cfg file for kamailio? i want help on the basic functions like record_route() , sl_send_reply(), t_relay() etc... Regards, Vineet Menon

2 3

Re: [SR-Users] Kamailio LDAP integration
by Reda Aouad 07 May '12

07 May '12

Sorry didn't reply to mailing list before. Emails are below. SHA1 encryption may not encrypt the same way as HA1 (HA1 = MD5 of realm + username + password), so the problem may be here. I suggest you store your passwords as clear text in LDAP for testing first. Reda On Fri, May 4, 2012 at 11:14 PM, Saul Waizer <saulwaizer(a)gmail.com> wrote: > with the variations I get different results: > 4(24126) ERROR: <script>: Password={SHA}v/m3IZiuy+VVizqnt56e2baZsT8= > 4(24126) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=760 a=17 n=if > 4(24126) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=756 a=28 > n=pv_www_authenticate > 4(24126) DEBUG: auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1 > 4(24126) DEBUG: auth [api.c:210]: check_response: Our result = > '3839aa4cae572f5f8b23601a2bb1178f' > 4(24126) DEBUG: auth [api.c:220]: check_response: Authorization failed > > On Fri, May 4, 2012 at 3:11 PM, Saul Waizer <saulwaizer(a)gmail.com> wrote: > >> Also: i used xlog to print out the password and I get the same exact >> password I have on my LDAP server, so it seems something with the decoding >> >> >> On Fri, May 4, 2012 at 3:01 PM, Saul Waizer <saulwaizer(a)gmail.com> wrote: >> >>> Now i got it down to this: >>> >>> 2(23003) INFO: <script>: ldap_search: found [1] entries for >>> (uid=mmiller) 2(23003) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] >>> l=759 a=17 n=if >>> 2(23003) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=755 a=28 >>> n=pv_www_authenticate >>> 2(23003) DEBUG: auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1 >>> 2(23003) DEBUG: auth [auth_mod.c:455]: HA1 string calculated: >>> c69622bbd922ec9321ab1293c226b703 >>> 2(23003) DEBUG: auth [api.c:210]: check_response: Our result = >>> '939676a5591165f1da8ba04562d446b2' >>> 2(23003) DEBUG: auth [api.c:220]: check_response: Authorization failed >>> 2(23003) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=756 a=27 >>> n=www_challenge >>> 2(23003) DEBUG: auth [challenge.c:102]: build_challenge_hf: >>> realm='23.22.35.43' >>> 2(23003) DEBUG: auth [challenge.c:244]: auth: 'WWW-Authenticate: Digest >>> realm="23.22.35.43", nonce="T6Qn/E+kJtAU7IvGh4OLivg7ptLbdida" >>> >>> I have changed the values of: >>> >>> if (!pv_www_authenticate("$td", "$avp(password)", "0")) { >>> www_challenge("$td", "0"); >>> >>> to: >>> >>> if (!pv_www_authenticate("$td", "$avp(password)", "1")) { >>> www_challenge("$td", "0"); >>> >>> because of the password in LDAP is stored as SHA1, and according to >>> the docs, it should be 1. I'm so close it seems :) >>> >>> *flags* - the value of this parameter can be a bitmask of following: >>> >>> - >>> >>> *1* - the value of password parameter is HA1 format >>> >>> >>> >>> On Fri, May 4, 2012 at 2:47 PM, Reda Aouad <reda.aouad(a)gmail.com> wrote: >>> >>>> can you also print the avp(s:password) to log to see what its value is? >>>> use: >>>> xlog('Password=$avp(s:password)') >>>> after ldap_search and you'll see its output in the log file >>>> maybe you're not correctly getting the password from the ldap search >>>> url, avp(s:password) is then null and you get the error that it can't be >>>> converted to string >>>> >>>> Reda >>>> >>>> >>>> >>>> On Fri, May 4, 2012 at 8:40 PM, Reda Aouad <reda.aouad(a)gmail.com>wrote: >>>> >>>>> in the line >>>>> if (!pv_www_authenticate("$td", "$avp(password)", "0")) { >>>>> >>>>> write avp(s:password) instead of avp(password) >>>>> not sure it will solve it though.. if it doesn't, maybe others can >>>>> help you more on this. >>>>> >>>>> Reda >>>>> >>>>> >>>>> >>>>> On Fri, May 4, 2012 at 5:50 PM, Saul Waizer <saulwaizer(a)gmail.com>wrote: >>>>> >>>>>> Hello Reda, >>>>>> >>>>>> Thank you for your feedback, after some further research and testing >>>>>> I got the LDAP search working, I am just having one issue with the password >>>>>> variable: >>>>>> >>>>>> 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=755 >>>>>> a=28 n=pv_www_authenticate >>>>>> 3(22487) ERROR: <core> [sr_module.c:1613]: Could not convert PV to >>>>>> str >>>>>> 3(22487) ERROR: auth [auth_mod.c:569]: failed to get passwd value >>>>>> >>>>>> My relevant configuration: >>>>>> >>>>>> route[AUTH] { >>>>>> #!ifdef WITH_AUTH >>>>>> if (is_method("REGISTER")) >>>>>> { >>>>>> if(is_present_hf("Authorization")) >>>>>> { >>>>>> # ldap search >>>>>> >>>>>> if >>>>>> (!ldap_search("ldap://demo/ou=demo,dc=mydomain,dc=com?uid,userPassword?")) >>>>>> >>>>>> { >>>>>> switch ($retcode) >>>>>> { >>>>>> case -1: >>>>>> # no LDAP entry found >>>>>> sl_send_reply("404", "User Not Found"); >>>>>> exit; >>>>>> case -2: >>>>>> # internal error >>>>>> sl_send_reply("500", "Internal server error"); >>>>>> exit; >>>>>> default: >>>>>> exit; >>>>>> } >>>>>> } >>>>>> ldap_result("uid/$avp(s:username)"); >>>>>> ldap_result("userPassword/$avp(s:password)"); >>>>>> xlog("L_INFO", "ldap_search: found [$retcode] entries for >>>>>> (uid=$fU)"); >>>>>> if (!pv_www_authenticate("$td", "$avp(password)", "0")) { >>>>>> www_challenge("$td", "1"); >>>>>> exit; >>>>>> } >>>>>> sl_send_reply("200", "ok"); >>>>>> exit; >>>>>> } else { >>>>>> www_challenge("$td", "1"); >>>>>> exit; >>>>>> } >>>>>> } else { >>>>>> >>>>>> And the error message: >>>>>> >>>>>> 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=735 >>>>>> a=26 n=ldap_search >>>>>> 3(22487) DEBUG: ldap [ldap_api_fn.c:273]: LDAP URL parsed into >>>>>> session_name [demo], base [ou=demo,dc=mydomain,dc=com], scope [0], filter [] >>>>>> 3(22487) DEBUG: ldap [ldap_api_fn.c:433]: [demo]: performing LDAP >>>>>> search: dn [ou=demo,dc=mydomain,dc=com], scope [0], filter [(null)], >>>>>> client_timeout [5000000] usecs >>>>>> 3(22487) DEBUG: ldap [ldap_api_fn.c:240]: [demo]: [1] LDAP entries >>>>>> found >>>>>> 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=752 >>>>>> a=26 n=ldap_result >>>>>> 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=753 >>>>>> a=26 n=ldap_result >>>>>> 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=754 >>>>>> a=27 n=xlog >>>>>> 3(22487) INFO: <script>: ldap_search: found [-1] entries for >>>>>> (uid=mmiller) 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] >>>>>> l=759 a=17 n=if >>>>>> 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=755 >>>>>> a=28 n=pv_www_authenticate >>>>>> 3(22487) ERROR: <core> [sr_module.c:1613]: Could not convert PV to >>>>>> str >>>>>> 3(22487) ERROR: auth [auth_mod.c:569]: failed to get passwd value >>>>>> 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=756 >>>>>> a=27 n=www_challenge >>>>>> 3(22487) DEBUG: auth [challenge.c:102]: build_challenge_hf: >>>>>> realm='ip.of.sip.server' >>>>>> 3(22487) DEBUG: auth [challenge.c:113]: build_challenge_hf: >>>>>> qop='auth' >>>>>> 3(22487) DEBUG: auth [challenge.c:244]: auth: 'WWW-Authenticate: >>>>>> Digest realm="ip.of.sip.server", nonce="T6P5yU+j+J23OE93mPaektZpJszGpt/l", >>>>>> qop="auth" >>>>>> >>>>>> Any help is greatly appreciated! >>>>>> Thanks >>>>>> >>>>>> >>>>>> >>>>>> On Thu, May 3, 2012 at 4:22 PM, Reda Aouad <reda.aouad(a)gmail.com>wrote: >>>>>> >>>>>>> Hi Saul, >>>>>>> >>>>>>> username_avp_spec was previously a AUTH module parameter to specify >>>>>>> a variable that was passed to pv_www_authorize implicitly (the function >>>>>>> doesn't take arguments). Now you should use the new >>>>>>> pv_www_authenticate and pass to it explicitly the credentials as arguments. >>>>>>> >>>>>>> So forget about username_avp_spec since it doesn't exist as module >>>>>>> param anymore (this is why you are getting the error). Store the result of >>>>>>> ldap_search in the avps as in the tutorial using ldap_result, and pass them >>>>>>> to pv_www_authenticate as parameters. pv_www_authenticate takes the >>>>>>> following arguments: >>>>>>> - realm: which you can get from "to domain" using $td >>>>>>> - password: $avp(s:password) >>>>>>> - flag: set it to 0 as a first test >>>>>>> >>>>>>> example: >>>>>>> pv_www_authorize("$td", "$avp(s:password)", 0) >>>>>>> >>>>>>> This function takes the username from the authentication header, so >>>>>>> no need to pass it anymore as argument. >>>>>>> >>>>>>> Reda >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Thu, May 3, 2012 at 8:47 PM, Saul Waizer <saulwaizer(a)gmail.com>wrote: >>>>>>> >>>>>>>> Hello List, >>>>>>>> >>>>>>>> I am trying to incorporate an existing LDAP directory with our >>>>>>>> Kamailio installation for SIP authentication. A good friend suggested to >>>>>>>> checkout this tutorial and adapt it to fit my needs (and current version) >>>>>>>> >>>>>>>> >>>>>>>> http://www.kamailio.org/dokuwiki/doku.php/tutorials:openser-auth-ldap >>>>>>>> >>>>>>>> It seems like the AUTH module does not contain the function >>>>>>>> username_spec (which I believe is not used anymore) but the >>>>>>>> username_avp_spec which is not part of the AUTH module but the H350 module >>>>>>>> http://kamailio.org/docs/modules/3.2.x/modules_k/h350.html >>>>>>>> >>>>>>>> I enabled the h350 module and tried setting the params as described >>>>>>>> in the documentation: >>>>>>>> >>>>>>>> modparam("auth", "username_spec", "$avp(s:username)") >>>>>>>> modparam("auth", "password_spec", "$avp(s:password)") >>>>>>>> modparam("auth", "calculate_ha1", 1) >>>>>>>> >>>>>>>> I got the following error after checking the configuration: >>>>>>>> >>>>>>>> ERROR: <core> [modparam.c:151]: set_mod_param_regex: parameter >>>>>>>> <username_spec> not found in module <auth> >>>>>>>> >>>>>>>> I am running kamailio 3.2.3 (i386/linux) Ubuntu >>>>>>>> >>>>>>>> Thank you in advance! >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> 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] Remove_hf doesn't work
by Daniel-Constantin Mierla 07 May '12

07 May '12

Hello, you have fix_nated_contact() which does an update to contact address: May 7 15:46:58 localhost /usr/local/sbin/kamailio[15742]: ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio//kamailio.cfg] l=889 a=25 n=fix_nated_contact If you want to remove the contact, then do not call that function, or use the msg_apply_changes() as mentioned previously. Cheers, Daniel On 5/7/12 4:23 PM, Openser Kamailio wrote: > Hi Daniel, > > Please find my log messages in attachments. > Open my log file with Notepad, it's better. > > Thanks! > > On Mon, May 7, 2012 at 2:41 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > do you use any other function that updates the address in contact > header? > > If you load debugger module, enable cfgtrace option and send the > execution log messages over here, we can tell if there is another > function that may alter contact header. > > One option is to use msg_apply_changes() before and after > remove_hf(), but it is not the optimal way. > > Cheers, > Daniel > > > On 5/7/12 2:13 PM, Openser Kamailio wrote: >> Hi, >> >> I'm currently working with kamailio 3.2. I try to remove the >> Contact Header field with remove_hf("Contact") through the file >> named kamailio.cfg. >> When i apply remove_hf("Contact"), it only removes the word >> Contact, but the contact address is still present. Let me show you: >> >> Without remove_hf: >> Contact: sip:<20505@sipserver.com:5060 >> <http://20505@sipserver.com:5060>>;......... >> >> After remove_hf: >> sip:<20502@sipserver.com:5060 >> <http://20502@sipserver.com:5060>>;............ >> >> I want to remove the whole contact address. >> Thanks for help! >> >> >> >> _______________________________________________ >> 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 > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Problem with adding extra field in acc table.
by Gary Chen 07 May '12

07 May '12

I using Mysql DB. I need to add extra fields in acc table. The field having problem is the last one 'account' Here is my code in kamailio.cfg: ... modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;user_agent=$hdr(User-Agent);account=$var(s:account_number)") ... route { .... dp_translate("3", "$fU/$avp(s:dst)") $v(s:account_number) = $avp(s:dst); .... } By looking at the xlog message, $v(s:account_number) has correct value. But the value in acc table is not always correct. Sometime account field populated with 0 and sometime it populated with wrong account number. It seems that $v(s:account_number) sometime corrupted. Am I doing something wrong? Gary

2 1

Remove_hf doesn't work
by Openser Kamailio 07 May '12

07 May '12

Hi, I'm currently working with kamailio 3.2. I try to remove the Contact Header field with remove_hf("Contact") through the file named kamailio.cfg. When i apply remove_hf("Contact"), it only removes the word Contact, but the contact address is still present. Let me show you: Without remove_hf: Contact: sip:<20505@sipserver.com:5060>;......... After remove_hf: sip:<20502@sipserver.com:5060>;............ I want to remove the whole contact address. Thanks for help!

2 1

Re: [SR-Users] json and jsonrpc-c
by Ron McLeod 06 May '12

06 May '12

I set the PKG_CONFIG_PATH environment variable to /usr/local/lib/pkgconfig . After that it built fine. From: Ron McLeod [mailto:ron.kamailio@mcleodnet.com] Sent: Saturday, May 05, 2012 9:24 PM To: 'sr-users(a)lists.sip-router.org' Subject: json and jsonrpc-c I am trying to build json and jsonrpc-c for kamailio-3.2.3 from a fresh download of the source tarball but not having success. I modified the Makefile to include the two modules in the standard package, and remove the line which excluded them from building. I see the code for the modules in the source. Also, I have already built and installed the two dependencies libevent and libjson. Could someone take a look at the make out and see if there is any indication of what I might be doing wrong? http://www.siplabs.net/tracebin/make_and_install_output.txt Thanks Ron

2 2

siptrace port number error
by Reda Aouad 06 May '12

06 May '12

Hi, I'm trying to setup Kamailio as a Homer capture server itself. Kamailio listens on 2 ports : 53 for SIP users, and 9060 for homer SIP capture. siptrace module duplicates packets to sipcapture module on port 9060. I'm getting the following error : ERROR: siptrace [siptrace.c:1669]: invalid port number; must be in [1024,65536] This is my conf ----------------------- listen=udp:10.238.235.173:53 listen=udp:127.0.0.1:9060 #!ifdef WITH_SIPTRACE modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", FLD_TRACE) modparam("siptrace", "duplicate_uri","sip:127.0.0.1:9060") modparam("siptrace", "hep_mode_on",1); modparam("siptrace", "trace_to_database", 0); #!endif Any help? Thanks Reda

3 4

json and jsonrpc-c
by Ron McLeod 06 May '12

06 May '12

I am trying to build json and jsonrpc-c for kamailio-3.2.3 from a fresh download of the source tarball but not having success. I modified the Makefile to include the two modules in the standard package, and remove the line which excluded them from building. I see the code for the modules in the source. Also, I have already built and installed the two dependencies - libevent and libjson. Could someone take a look at the make out and see if there is any indication of what I might be doing wrong? http://www.siplabs.net/tracebin/make_and_install_output.txt Thanks - Ron

1 0

json and jsonrpc-c
by Ron McLeod 06 May '12

06 May '12

I am trying to build json and jsonrpc-c for kamailio-3.2.3 from a fresh download of the source tarball but not having success. I modified the Makefile to include the two modules in the standard package, and remove the line which excluded them from building. I see the code for the modules in the source. Also, I have already built and installed the two dependencies - libevent and libjson. Could someone take a look at the make out and see if there is any indication of what I might be doing wrong? http://www.siplabs.net/tracebin/make_and_install_output.txt Thanks - Ron

1 0

domain kaobona.com
by Thulani Ngwenya 05 May '12

05 May '12

Hi, I registered my SIP domain kaobona.com to be hosted on iptel.org, how do I get users to register accounts using my domain? Kind regards, Thulani

1 0

pua_reginfo: no Username in PUBLISH message
by Gnaneshwar Gatla 04 May '12

04 May '12

Hello, I was looking at the pua_reginfo to make multiple instances of Kamailio be aware of registered users on the network. I'm experimenting with pua_reginfo with Kamailio 3.2.2. I've used the configuration as described in http://kamailio.org/docs/modules/3.2.x/modules_k/pua_reginfo.html#id2551427 I tried an trial run of the Kamailio instance and it gives the following output PUBLISH sip:@64.64.203.108 SIP/2.0 Via: SIP/2.0/UDP 64.64.203.108;branch=z9hG4bKb62c.b8e3d5b5.0 To: sip:@64.64.203.108 From: sip:@64.64.203.108;tag=533cb9e91f4b999cf76861cbb9ed54ed-e085 CSeq: 10 PUBLISH Call-ID: 17dedfff-14418(a)127.0.0.1 Content-Length: 320 Max-Forwards: 70 Event: reg Expires: 3601 Content-Type: application/reginfo+xml <reginfo xmlns="urn:ietf:params:xml:ns:reginfo" version="0" state="full"> <registration aor="sip:@64.64.203.108" id="0xb6ef7740" state="active"> <contact id="0xb4d1f820" state="active" event="registered" expires="60" callid="ZmRjZDg3MmIxMjI5NmU1OWU0YjJlYWUzYzhjZDViM2I." cseq="2"> <uri>sip:ggatla01@192.168.128.108:23840;rinstance=27b18eed9568795e;transport=udp</uri> </contact> </registration> </reginfo> The PUA module does not insert username in the Request, To and From headers while publishing for Event: reg. The xml payload is right but the "aor" does not have the username. Is there anything missing that I need to configure? Lastly, the presence module in the same Kamailio instance is generating "478 Unresolvable destination" for this message. Regards Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

2 2

Kamailio LDAP integration
by Saul Waizer 04 May '12

04 May '12

Hello List, I am trying to incorporate an existing LDAP directory with our Kamailio installation for SIP authentication. A good friend suggested to checkout this tutorial and adapt it to fit my needs (and current version) http://www.kamailio.org/dokuwiki/doku.php/tutorials:openser-auth-ldap It seems like the AUTH module does not contain the function username_spec (which I believe is not used anymore) but the username_avp_spec which is not part of the AUTH module but the H350 module http://kamailio.org/docs/modules/3.2.x/modules_k/h350.html I enabled the h350 module and tried setting the params as described in the documentation: modparam("auth", "username_spec", "$avp(s:username)") modparam("auth", "password_spec", "$avp(s:password)") modparam("auth", "calculate_ha1", 1) I got the following error after checking the configuration: ERROR: <core> [modparam.c:151]: set_mod_param_regex: parameter <username_spec> not found in module <auth> I am running kamailio 3.2.3 (i386/linux) Ubuntu Thank you in advance!

2 3

sdp parsing error
by RUSHI NAYAK 04 May '12

04 May '12

for the following sdp content i'm getting error. v=0 o=rushi 602301 603670 IN IP4 192.168.1.125 s=- c=IN IPV4 192.168.1.125 t=0 0 m=audio 8192 RTP/AVP 5 0 3 can you give correction or sample sdp string.

2 1

Sending NOTIFY using t_uac_dlg over xmlrpc
by Krishna Kurapati 04 May '12

04 May '12

Hi, I was able to send MESSAGE using xmlrpc (mi and t_uac_dlg) I got the 200 OK response to the message. However, when I try to send NOTIFY, I receive HTTP response but with empty methodResponse. Is NOTIFY not suppored by t_uac_dlg? Thanks Krish Kura POST /RPC2 HTTP/1.0 User-Agent: ruby-sip-client Host: 10.1.0.6 Content-Type: text/xml Connection: keep-alive Content-Length: 650 <?xml version="1.0" ?><methodCall><methodName>mi</methodName><params><param><value><string>t_uac_dlg</string></value></param><param><value><string>NOTIFY</string></value></param><param><value><string> sip:kkurapati_staff_test@10.1.0.6</string></value></param><param><value><string>.</string></value></param><param><value><string>.</string></value></param><param><value><string>From: <sip:kkurapati_provider_test@10.1.0.6> To: <sip:kkurapati_staff_test@10.1.0.6> Max-Forwards: 70 Call-ID: e3f15b5dc200234cff2c7b648d79c49f(a)10.1.0.3 </string></value></param><param><value><string>User added</string></value></param></params></methodCall> ## T 2012/04/27 14:39:11.579281 10.1.0.6:5060 -> 10.1.0.3:44817 [AP] HTTP/1.0 200 OK. Via: SIP/2.0/TCP 10.1.0.3:44817. Server: kamailio (3.3.0-dev7 (x86_64/linux)). Content-Length: 137. . <?xml version="1.0"?> <methodResponse> <params> <param> <value><array><data> </data></array></value> </param> </params> </methodResponse>

2 3

sdpops kamailio ans Siremis
by Grégoire Vandendeurpel 04 May '12

04 May '12

Hi, I'm using Siremis 3.2 ans Kamailio 3.2.3 for my final project of my bachloor degree I would like to know how Siremis and Kamailio work together? Like: If I decide to set a protocol in the acl trusted in Siremis like TLS, ok, it works well !! But I need to know the different links between siremis and kamailio to working well.. Why? Cause I need to implement in Siremis a filter form on the sdpops module, and it has to work well of course with the database of ser! So.. What are the different steps to do this? - Create new table sdpops in ser - create the form in siremis and view .. - what in kamailio.cfg?? I found this: http://kb.asipto.com/siremis:install32x:new-views OK, but in my case, I need to do that for SDPOPS, which is a module of kamailio, whithout any table and components about! I mean, there is no openser.sdpops in the database of kamailio, so create the table ok.. but what are the columns, datas, to set in ?? cause I tried php gen_meta.php Serdb sdpops ser.acl.sdpops of course it does not work --> no openser.sdpops ...... thx in advance for your help Best regards -- *Grégoire Vandendeurpel, * * * *IT Sector*

2 2

SEMS 1.4.3 released
by Stefan Sayer 04 May '12

04 May '12

Hello, the SIP Express Media Server project happily announces the availability of the SEMS 1.4.3 release. This is a SEMS 1.4 series bugfix release and should be a drop-in replacement to 1.4 installations. For details of the changes see the changelog at http://ftp.iptel.org/pub/sems/1.4/1.4.3/CHANGELOG Please find sources at http://ftp.iptel.org/pub/sems/sems-1.4.3.tar.gz binary packages in the usual locations soon, and all documentation and links at http://iptel.org/sems If you are using custom applications which create outbound calls, please specifically test the behavior with the commit 26fdaa8e b/f: stopping failed outbound calls Many thanks to everyone who contributed with issue reports and patches! Best Regards Stefan Sayer SEMS - E stands for Express!

1 0

Need Help please
by Rudi Kusnaka 04 May '12

04 May '12

Hello there, I've installing the Siremis 3.2 on a linux that installed Kamailio 3.2. When i'll try to access everything in SER Menu, i'm getting a error messages that tell me about the table in OPENSER database is missing or can't located. But this is a fresh installed Siremis system. Where i can found the openser database stucture so i can build them again? Best Regards, Rudi Kusnaka

2 1

Siremis on Mysql and Kamailio on Postgresql
by Jonny Klompas 04 May '12

04 May '12

I've setup a siremis installation using Mysql for the siremis tables. Kamailio is installed with postgresql on another machine. I selected Postgresql in seremis setup under SIP DB and correctly inserted the right username/password/host. Connection was successful. I can login to Siremis (ver 3.2.0) and access the Admin menu successfully. However when I access the Ser Menu, all the options give errors: For example, trying to to select Server Services, Domain List gives the following errors: [2012-04-05 18:11:36 (GMT)] An exception occurred while executing this script: Error message: #256, Unable to location template file system_internal_error.tpl. Script name and line number of error: /usr/local/siremis-3.2.0/openbiz/bin/Resource.php:283 *function:* errorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/usr/local/siremis-3.2.0/openbiz/bin/Resource.php", 283, Array(11) ) @ /usr/local/siremis-3.2.0/openbiz/bin/sysheader.inc 117 *function:* userErrorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/usr/local/siremis-3.2.0/openbiz/bin/Resource.php", 283, Array(11) ) @ *function:* trigger_error ( "Unable to location template file system_internal_error.tpl.", 256 ) @ /usr/local/siremis-3.2.0/openbiz/bin/Resource.php 283 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /usr/local/siremis-3.2.0/openbiz/bin/BizSystem.php 544 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /usr/local/siremis-3.2.0/siremis/modules/service/userEmailService.php 108 *function:* SystemInternalErrorEmail ( Array(2), Null ) @ /usr/local/siremis-3.2.0/siremis/modules/common/form/ErrorForm.php 34 *function:* Report ( ) @ /usr/local/siremis-3.2.0/openbiz/bin/BizController.php 310 *function:* invoke ( ) @ /usr/local/siremis-3.2.0/openbiz/bin/BizController.php 110 *function:* dispatchRequest ( ) @ /usr/local/siremis-3.2.0/openbiz/bin/BizController.php 32 *function:* include_once ( "/usr/local/siremis-3.2.0/openbiz/bin/BizController.php" ) @ /usr/local/siremis-3.2.0/siremis/bin/controller.php 6 ------------------------------ * * Similarly. the all the other optiones under the Ser Menu give errors. My Config.xml (passwords and IPs modified) follows: <?xml version="1.0" standalone="no"?> <Config> <DataSource> <Database Name="Default" Driver="Pdo_Mysql" Server="localhost" Port="3306" DB Name="siremis" User="siremis" Password="siremisrw" Charset="utf8"/> <Database Name="Serdb" Driver="Pdo_Pgsql" Server="x.x.x.x" Port="5432" DBName="dbname" User="dbuser" Password="123456" Charset="utf8"/> </DataSource> </Config> Please advise. Many thanks Jonny * *

4 11

Siremis upload file
by Jesús Pérez Rubio 03 May '12

03 May '12

Hi, I have a problem making some changes on siremis front-end everything was ok except when I try to add a field to upload a file I got this error: #2, fopen(ejemplo.csv): failed to open stream: No such file or directory could anybody help me please? :) Mi configuration: DO: <BizField Name="batch_file" Column="batch_file" Length="64" Required="N" Type="Blob"/> Insert: <Element Name="fld_batch_file" Class="File" FieldName="batch_file" Label="File (.csv)" AllowURLParam="N"/> .php: public function InsertRecord($recArr) { return parent::InsertRecord($recArr); } SQL: mysql> desc batch_provision; +------------+---------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +------------+---------+------+-----+---------+----------------+ | id | int(11) | NO | PRI | NULL | auto_increment | | batch_file | blob | YES | | NULL | | +------------+---------+------+-----+---------+----------------+ 2 rows in set (0.02 sec) Thanks in advance. -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com

2 1

Routing based on gruu
by Krishna Kurapati 03 May '12

03 May '12

Hi, I am trying to use GRUU for routing MESSAGEs. For example if I want to route to specific instance of the registered user, what do I need to send in the MESSAGE? If I know pub-gruu of the instance, where do I insert it so that kamailio can correctly process it and send it to the right instance? Thanks Krish Kura

2 1

kamctl trouble
by Vineet Menon 03 May '12

03 May '12

Hi, I have a kamailio installation running, but in kamctl the moni command works, but stop, restart, start command doesn't work... why is it so?? Regards, Vineet Menon

3 3

AVP assignment question
by Klaus Darilion 03 May '12

03 May '12

Hi! For example: $avp(i:test) = 1; $avp(i:test) = 2; # now: test[0]=2, test[1]=1 I would think that assigning or deleting using an index would overwrite the respective element in the AVP list, but assigning $null always deletes [0] and assigning a value always add another AVP to the top, e.g: $(avp(i:test)[1]) = 3; # I would expect: test[0]=2, test[1]=3 # but it is: test[0]=3, test[1]=2, test[2]=1 $(avp(i:test)[2]) = $null; # I would expect: test[0]=3, test[1]=2 # but it is: test[0]=2, test[1]=1 So, is the current behavior really the correct behavior? It is not what I would expect when using indexes. If it is correct, I will improve the documentation accordingly. regards Klaus

2 2

dialog - keep alives with lower cseq
by Daniel-Constantin Mierla 03 May '12

03 May '12

Hello, several days ago, just before freezing the development for v3.3.0, I added to dialog module the option to send keep alives for ongoing calls in order to detect if caller/callee is gone. SIP specs require to increment the CSeq for requests within dialog, but because the keep alives are sent from the sip server, it would results de-synchronization of the CSeq values hold in phones themselves (e.g., a BYE created by caller/callee after a keep alive will be with lower cseq than the other side would expect and accept). One solution would be to update cseq when BYE passes the server to the right value. This implies more processing, as a call can have many re-INVITEs or other requests within dialog sent by caller/callee, including periodic updates to database to store cseq numbers. So I went for a different approach, like stated in subject -- let's see your opinion if you think is going to work. Practically the keep alives will be OPTIONS with CSeq equal or lower than the last valid value (e.g., the cseq of the INVITE creating the call). If the caller/callee is gone, that is simply, the OPTION will be timed out and dialog module will send BYE. If the caller/callee are reachable, but for some reason the call was destroyed (e.g., a crash and restart meanwhile), since there is a To-tag, the OPTIONS should get a 481 Call/Leg transaction does not exists. Again, a case when kamailio will end the dialog. The one to be discussed here would be caller/callee are still on the call. Based on RFC (and the feedback from people on another thread here), a requests coming within a dialog with lower CSEq should be replies with 500. CSeq numbers are still ok in both sides (note that requests with lower CSeq can happen in reality, like two fast re-INVITEs sent over UDP, the second arriving first due to network transmission). I tested with snom phones and jitsi so far, seems to be a working solution (well, jitsi was replying 500 after 20secs to keep alive OPTIONS request, not sure for what reason, just reported back to the project). Is anyone here seeing any possible issues with the approach? Call are ended by dialog, with proper CSeq in the BYE, after 10seconds from the moment 408 or 481 is received, no other replies are taken in consideration for ending the dialog from server side. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 5

Can't load perl module
by Ladislav Jurák 02 May '12

02 May '12

Hello, I have problem with loading perl module on Kamailio v3.2 installed from repository on Debian sqeeze amd64. Perl modules was also installed from repository package kamailio-perl-modules. I have also installed all perl dependences listed in perl module documentary. Into Kamailio config file I added: loadmodule "perl.so" and modparam("perl", "filename", "/etc/kamailio/skript.pl") Now when I'm trying to start Kamailio I'll get following: May 1 23:13:46 server kamailio: DEBUG: <core> [cfg.y:1712]: loading module perl.so May 1 23:13:46 server kamailio: DEBUG: <core> [sr_module.c:557]: load_module: trying to load </usr/lib64/kamailio/modules_k/perl.so> May 1 23:13:46 server kamailio: WARNING: <core> [sr_module.c:620]: /usr/lib64/kamailio/modules_k/perl.so: exports dlflags interface is deprecated and it will not be supported in newer versions; consider using mod_register() instead May 1 23:13:46 server kamailio: DEBUG: <core> [modparam.c:96]: set_mod_param_regex: 'perl' matches module 'perl' May 1 23:13:46 server kamailio: DEBUG: <core> [sr_module.c:763]: find_param_export: found <filename> in module perl [/usr/lib64/kamailio/modules_k/perl.so] May 1 23:13:46 server kamailio: DEBUG: <core> [modparam.c:113]: set_mod_param_regex: found <filename> in module perl [/usr/lib64/kamailio/modules_k/perl.so] May 1 23:13:46 server /usr/sbin/kamailio[31192]: DEBUG: <core> [sr_module.c:928]: DEBUG: init_mod: perl May 1 23:13:46 server /usr/sbin/kamailio[31192]: ERROR: perl [perl.c:238]: failed to load perl file "/etc/kamailio/skript.pl". May 1 23:13:46 server /usr/sbin/kamailio[31192]: ERROR: <core> [sr_module.c:932]: init_mod(): Error while initializing module perl (/usr/lib64/kamailio/modules_k/perl.so) I am confused about the WARNIG too, can it have some influence to the errors? File skript.pl is in the specified directory. I also changed permissions to 777. It's content is from sample "branches.pl". I have similar problem with module app_python.so. Can you please give me some advice? It's important for me to get this module working. Thank you. -- Regards Ladislav Jurak

2 6

sdpops Siremis
by Grégoire Vandendeurpel 02 May '12

02 May '12

Hi, I'm using Siremis 3.2 ans Kamailio 3.2.3 for my final project of my bachloor degree I would like to know how Siremis and Kamailio work together? Like: If I decide to set a protocol in the acl trusted in Siremis like TLS, ok, it works well !! But I need to know the different links between siremis and kamailio to working well.. Why? Cause I need to implement in Siremis a choise form about the sdpops module, and it has to work well of course with the database of ser! So.. What are the different steps to do this? - Create new table sdpops in ser - create the form in siremis and view .. - what in kamailio.cfg?? I found this: http://kb.asipto.com/siremis:install32x:new-views OK, but in my case, I need to do that for SDPOPS, which is a module of kamailio, whithout any table and components about! I mean, there is no openser.sdpops in the database of kamailio, so create the table ok.. but what are the columns, datas, to set in ?? cause I tried php gen_meta.php Serdb sdpops ser.acl.sdpops of course it does not work --> no openser.sdpops ...... thx in advance for your help Best regards -- *Grégoire Vandendeurpel, * * * *IT Sector*

1 0

Problem with Route header
by Geoffrey Mina 02 May '12

02 May '12

Greetings, I am confused at some functionality I am seeing with Kamailio 1.5.4. I know this is an old version, but I don't have the time to go through a lengthy upgrade process right now. The issue I am seeing is that the server is inserting a Route header with it's own IP address for an unknown reason. Here is the initial invite (removed SDP for simplicity): INVITE sip:13@67.207.130.146:5060 SIP/2.0 Via: SIP/2.0/UDP 68.64.220.108:5060;branch=z9hG4bK78dd33c6;rport From: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com >;tag=as1cad6370 To: <sip:13@67.207.130.146:5060> Contact: <sip:9546496707@68.64.220.108> Call-ID: 43134ece101abfca6ecab20212295909(a)dev-asterisk.mydomain.com CSeq: 102 INVITE User-Agent: G-Tel v1.0 Max-Forwards: 70 Remote-Party-ID: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com >;privacy=off;screen=no Date: Tue, 01 May 2012 18:17:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Route: <sip:13@boulder-voip.mydomain.com> P-Account-ID: 99990023 P-Proxy-Route: Yes Content-Type: application/sdp Content-Length: 240 The basics of what happen next are: t_check_trans(); record_route(); remove_hf("P-Proxy-Route"); if(loose_route()){ route(3); } route[3]{ t_on_reply("1"); if(!t_relay()){ sl_reply_error(); } } The INVITE that goes out has the funky Route: header with the Kamailio IP in there. This is causing problems for some of the upstream proxy servers (obviously). INVITE sip:13@boulder-voip.mydomain.com SIP/2.0 Record-Route: <sip:67.207.130.146;lr;ftag=as1cad6370> Via: SIP/2.0/UDP 67.207.130.146;branch=z9hG4bKf183.456d51e1.0 Via: SIP/2.0/UDP 68.64.220.108:5060 ;received=68.64.220.108;branch=z9hG4bK78dd33c6;rport=5060 From: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com >;tag=as1cad6370 To: <sip:13@67.207.130.146:5060> Contact: <sip:9546496707@68.64.220.108> Call-ID: 43134ece101abfca6ecab20212295909(a)dev-asterisk.mydomain.com CSeq: 102 INVITE User-Agent: G-Tel v1.0 Max-Forwards: 69 Remote-Party-ID: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com >;privacy=off;screen=no Date: Tue, 01 May 2012 18:17:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces P-Account-ID: 99990023 Content-Type: application/sdp Content-Length: 240 Route: <sip:13@67.207.130.146:5060> Any idea what may be causing this to happen and how I could prevent it? I have tried removing the Route header using the remove_hf("Route") before doing the t_relay, but that doesn't seem to help. Thanks, Geoff

2 2

kamailio and openser
by Vineet Menon 01 May '12

01 May '12

Hi, I have a ubuntu 12.04 installation with me, and want to install kamailio in it... I have added repo from kamailio.ord site, but they are pretty old and require libmysqlclient16, which is no longer available. Ubunut 12.04 has a canonical repo for openser.... are they both interchangeble?? can i install openser and feel the same as kamailio?? Regards, Vineet Menon

3 3

LCR module from_uri module how to ?
by Sébastien Cramatte 01 May '12

01 May '12

Hello, Does anyone can give me a proper example how setup LCR to be able to use prefix and from_uri field. Does not a route should be preferred if prefix or from_uri is more accurate ? I mean that if you st from_uri value instead of default NULL route should be considered as better ? Thank you for you help

2 1

OpenXcap always returns full resource list
by Fabian Bernhard 01 May '12

01 May '12

Dear list, we are running Kamailio 3.2 together with the included OpenXcap server. I try to retrieve only part of a certain resource list using XPath, but the server always returns the full list, see example below. I was so far not able to retrieve only part of a resource list, independent of the XPath expression I'm using. Do I make something wrong or is this a bug? Thanks and regards, Fabian fubeh@orion:~$ curl --digest -u test4:XXXX http://192.168.1.134:5060/xcap-root/resource-lists/users/sip:test4@192.168.… <?xml version="1.0"?> <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list name="Business"> <entry uri="sip:test1@192.168.1.134"> <display-name>Test 1</display-name> </entry> <entry uri="sip:test122@192.168.1.134"> <display-name>Test 1</display-name> </entry> </list> <list name="Family"> <entry uri="sip:test2@192.168.1.134"> <display-name>Test 2</display-name> </entry> <entry uri="sip:test222@192.168.1.134"> <display-name>Test 2</display-name> </entry> </list> <list name="Friends"> <entry uri="sip:test3@192.168.1.134"> <display-name>Test 3</display-name> </entry> <entry uri="sip:test322@192.168.1.134"> <display-name>Test 3</display-name> </entry> </list> <list name="Blocked" /> </resource-lists>

2 4

Re: [SR-Users] testing the value of $si
by Alex Balashov 27 Apr '12

27 Apr '12

It should work. What are your symptoms that it doesn't? -- 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 x-kamailio(a)sidell.org wrote: >How do I compare $si to a particular IP address value? This doesn't >seem to work: > >if( $si == "123.123.123.123") > >_______________________________________________ >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

testing the value of $si
by x-kamailio＠sidell.org 27 Apr '12

27 Apr '12

How do I compare $si to a particular IP address value? This doesn't seem to work: if( $si == "123.123.123.123")

1 0

Re: [SR-Users] dispatcher and call transfer
by charles.chance＠sipcentric.com 27 Apr '12

27 Apr '12

1 0

Re: [SR-Users] dispatcher and call transfer
by charles.chance＠sipcentric.com 27 Apr '12

27 Apr '12

2 1

Re: [SR-Users] dispatcher and call transfer
by charles.chance＠sipcentric.com 27 Apr '12

27 Apr '12

1 0

Re: [SR-Users] dispatcher and call transfer
by Asgaroth 27 Apr '12

27 Apr '12

Hi All, Just resending this as my previous message was too big (bigger than 60k) ---. Thanks all for the suggestions. @Charles Yes that is correct, that is the scenario that is occurring, we are using asterisk, when a call is initiated via asterisk 1, the second call then routes through to asterisk 2, asterisk 1 then gets the REFER but doesn't know about the 2nd call. @Reda I like this option of using the hash table (I'd like to exclude a database lookup for each initial invite), however, would I be forced to use a database in the case of 2 loadbalancers, surely some shared state would need to exist between the loadbalancers to make this decision on where to route the call if it already exists. I suspect the hashtable suggestion may work perfectly in a single loadbalancer scenario, unless I'm missing something. For example, the initialy call may go via loadbalancer 1 however the second call may go through loadbalancer 2, that way loadbalancer 2 may not know of an existing call already been setup. In keeping with the hashtable train of thought, is it really neccessary to keep a reference to the callid, as the second call in the transfer scenario described by Charles, the callid of the second call is different to the original call. Just as an FYI, I am currently using dispatcher algorithm 0 which is a hash over callid. Thanks again all for the suggestions/tips

1 0

Re: [SR-Users] dispatcher and call transfer
by Asgaroth 27 Apr '12

27 Apr '12

Thanks all for the suggestions. @Charles Yes that is correct, that is the scenario that is occurring, we are using asterisk, when a call is initiated via asterisk 1, the second call then routes through to asterisk 2, asterisk 1 then gets the REFER but doesn't know about the 2nd call. @Reda I like this option of using the hash table (I'd like to exclude a database lookup for each initial invite), however, would I be forced to use a database in the case of 2 loadbalancers, surely some shared state would need to exist between the loadbalancers to make this decision on where to route the call if it already exists. I suspect the hashtable suggestion may work perfectly in a single loadbalancer scenario, unless I'm missing something. For example, the initialy call may go via loadbalancer 1 however the second call may go through loadbalancer 2, that way loadbalancer 2 may not know of an existing call already been setup. In keeping with the hashtable train of thought, is it really neccessary to keep a reference to the callid, as the second call in the transfer scenario described by Charles, the callid of the second call is different to the original call. Just as an FYI, I am currently using dispatcher algorithm 0 which is a hash over callid. Thanks again all for the suggestions/tips On 26/04/12 16:12, Charles Chance wrote: > > Yes, I realise Asterisk is not a media relay, but I don't think the OP > is using a pure media relay, otherwise the REFER message would not be > sent to it, would it? > > > > Thank you for the hash table suggestion by the way. > > > > Charles > > > > ------------------------------------------------------------------------ > > *From:*sr-users-bounces@lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org] *On Behalf Of *Reda Aouad > *Sent:* 26 April 2012 15:15 > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > UsersMailing List > *Subject:* Re: [SR-Users] dispatcher and call transfer > > > > Asterisk is not really a media relay. Asterisk establishes 2 legs for > each call, and I'm not sure what happens in this case. > > An improvement you can make is to use a hash table (in-memory) to > store the information you need (call-id, from, to) then lookup in > the htable for existing calls for same users. That should relieve your > database from a query on every invite and increase performance if you > have a large number of calls. > > > > Reda > > > > > > On Thu, Apr 26, 2012 at 15:52, Charles Chance > <charles.chance(a)sipcentric.com <mailto:charles.chance@sipcentric.com>> > wrote: > > Hi Reda, > > > > Sorry, I should have been more specific -- I am referring to instances > where the media server is for example Asterisk. If first call goes > through Asterisk 1, second call goes through Asterisk 2, when Asterisk > 2 receives the REFER it does not know of initial call on Asterisk 1. > The only way we have found for it to work is to ensure the second call > is dispatched to the same Asterisk box as the first. > > > > I would be pleased to hear of an alternative method. > > > > Regards, > > > > Charles > > ------------------------------------------------------------------------ > > *From:*sr-users-bounces@lists.sip-router.org > <mailto:sr-users-bounces@lists.sip-router.org> > [mailto:sr-users-bounces@lists.sip-router.org > <mailto:sr-users-bounces@lists.sip-router.org>] *On Behalf Of *Reda Aouad > *Sent:* 26 April 2012 14:34 > > > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > UsersMailing List > *Subject:* Re: [SR-Users] dispatcher and call transfer > > > > Hi, > > @Carsten > Dispatcher algorithm 0 based on call-id should do it in your case of > re-invite within dialog with same call-id. > > @Charles > In the case of attended transfers, shouldn't both media servers be > relaying media between them? I didn't understand why your are obliged > to dispatch to the same media server since they are 2 different calls > with different call-ids. > > > > Reda > > > > > > On Thu, Apr 26, 2012 at 14:30, Charles Chance > <charles.chance(a)sipcentric.com <mailto:charles.chance@sipcentric.com>> > wrote: > > Hi, > > Actually, this won't help for attended transfers where another call is > initiated first then the two are joined together by REFER. In this > case, the > second INVITE must be routed to the same media server as the existing call > for the transfer to work. > > What we do is store the dialogs in DB, then when a new call comes in, > prior > to doing ds_select_dst we query DB for existing call involving same > user. If > we find one, we simply replace destination host with that from the contact > (to/from depending on direction of call). > > It may not be the most elegant way but it works for us :) > > Charles > > > > -----Original Message----- > From: sr-users-bounces(a)lists.sip-router.org > <mailto:sr-users-bounces@lists.sip-router.org> > [mailto:sr-users-bounces@lists.sip-router.org > <mailto:sr-users-bounces@lists.sip-router.org>] On Behalf Of Carsten Bock > Sent: 26 April 2012 13:25 > To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > UsersMailing List > Subject: Re: [SR-Users] dispatcher and call transfer > > Hi, > > if you look at the docs of the dispatcher module, you'll find this: > > alg - the algorithm used to select the destination address. The > parameter can be an integer or a variable holding an interger. > - "0" - hash over callid > (http://kamailio.org/docs/modules/devel/modules_k/dispatcher.html#id2498492) > > But probably you should look into record/loose_route for your setup. > Since REFER is normally an in-dialog request (belongs to another > voice-session), it should take the same route as the initial INVITE. > This is normally achieved by the record/loose-route mechanisms > described in RFC3261. > In the example config of Kamailio you find an configuration example > (below). > > It is not a bug in the dispatcher module, it's how you use it. > > So long, > Carsten > > 455 # handle requests within SIP dialogs > 456 route(WITHINDLG); > > [...] > > 473 # record routing for dialog forming requests (in case > they are routed) > 474 # - remove preloaded route headers > 475 remove_hf("Route"); > 476 if (is_method("INVITE|SUBSCRIBE")) > 477 record_route(); > > and the relevent parts in the "WITHINDLG" route: > > 566 # Handle requests within SIP dialogs > 567 route[WITHINDLG] { > 568 if (has_totag()) { > 569 # sequential request withing a dialog should > 570 # take the path determined by record-routing > 571 if (loose_route()) { > [...] > 580 route(RELAY); > 581 } else { > [...] > 587 if ( is_method("ACK") ) { > 588 if ( t_check_trans() ) { > 589 # no loose-route, but > stateful > ACK; > 590 # must be an ACK after a 487 > 591 # or e.g. 404 from upstream > server > 592 t_relay(); > 593 exit; > 594 } else { > 595 # ACK without matching > transaction ... ignore and discard > 596 exit; > 597 } > 598 } > 599 sl_send_reply("404","Not here"); > 600 } > 601 exit; > 602 } > 603 } > > 2012/4/26 Asgaroth <00asgaroth00(a)gmail.com > <mailto:00asgaroth00@gmail.com>>: > > Hi All, > > > > Currently we are running kamailio in a loadbalanced fashion whereby calls > > come in via the loadbalancers and distribute calls accross 2 media > servers. > > We have come accross and issue whereby call transfers may be distributed > > accross two media servers and when the REFER message comes along to > transfer > > the call, in some cases (if we're lucky) the message arrives at the wrong > > media server (transaction leg doesnt exist). > > > > Some googling later and it appears that dispatcher doesnt play nice when > it > > comes to this scenario. Some suggestions popped up in my previous > searches > > saying that a potential work around is to use the dialog module to check > if > > a call is eastablished and then to send all calls to the same media > server > > based on the dialog already being established. > > > > I'd appreciate some input from the guru's out there that have come > accross > > this same issue and, if possible, some suggestions on how to work around > the > > problem, does the dispatcher module have a hashing algorithm that can be > > suited for this particular scenario? > > > > Thanks in advance for any tips or sugestions. > > > > _______________________________________________ > > 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 > > > > > > -- > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 Hamburg / Germany > > http://www.ng-voice.com > mailto:carsten@ng-voice.com <mailto:carsten@ng-voice.com> > > Mobile +49 179 2021244 <tel:%2B49%20179%202021244> > Office +49 40 34927219 <tel:%2B49%2040%2034927219> > Fax +49 40 34927220 <tel:%2B49%2040%2034927220> > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/ > > -- > Meet ng-voice at LinuxTag 2012 in Berlin - May 23rd - 26th, 2012. Save the > date! > > _______________________________________________ > 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 > > ----- > No virus found in this message. > Checked by AVG - www.avg.com <http://www.avg.com> > Version: 2012.0.1913 / Virus Database: 2411/4959 - Release Date: 04/25/12 > > > > _______________________________________________ > 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 > > > > ------------------------------------------------------------------------ > > No virus found in this message. > Checked by AVG - www.avg.com <http://www.avg.com> > Version: 2012.0.1913 / Virus Database: 2411/4959 - Release Date: 04/25/12 > > > _______________________________________________ > 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 > > > > ------------------------------------------------------------------------ > > No virus found in this message. > Checked by AVG - www.avg.com <http://www.avg.com> > Version: 2012.0.1913 / Virus Database: 2411/4959 - Release Date: 04/25/12 > > > > _______________________________________________ > 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

1 0

Copy reason field from 503 to 500.
by Vitaliy Aleksandrov 27 Apr '12

27 Apr '12

Hi all, I have Kamailio connected to PSTN gateway. Sometimes gateway replies with "503 temporary unavailable". PSTN gateway adds a "reason" header with ISUP code. But according to RFC 3261 kamailio doesn't forward 503 replies transparently. When 503 reply is received Kamailio replaces 503 with a 500 which is created from original request (build_res_buf_from_sip_req). I have tried append_hf from textops module, but it's not working with a locally generated replies. How can i copy Reason header from 503 to 500 ? I will appreciate your help with this situation. Best regards, Vitaliy Aleksandrov.

2 4

Gentoo ebuild v3.2.3
by caio 26 Apr '12

26 Apr '12

Good afternoon, It's ready the Gentoo ebuild for 3.2.3. You can get it from the Git repository or from the Gentoo Portage overlays: http://goo.gl/35uFX http://goo.gl/fmFIU Best regards, Caio

1 0

kamailio Crashes with rls module
by Gnaneshwar Gatla 26 Apr '12

26 Apr '12

Hello, I'm experimenting the usage of rls module with Kamailio. I have been able to successfully update xcap documents on Kamailio XCAP server. I have tried to use SIPp to get a NOTIFY for subscribing a rls-service list. Kamailio does get the xml from the xcap table and tries to parse it and crashes. I'm not sure if this is because the format of the XML, I derived the xml format from the RFC 4662 and 4826. I have attached the Kamailio log. I need help with this. Any help is appreciated, thank you very much. Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

1 0

Re: [SR-Users] dispatcher and call transfer
by Reda Aouad 26 Apr '12

26 Apr '12

Asterisk is not really a media relay. Asterisk establishes 2 legs for each call, and I'm not sure what happens in this case. An improvement you can make is to use a hash table (in-memory) to store the information you need (call-id, from, to) then lookup in the htable for existing calls for same users. That should relieve your database from a query on every invite and increase performance if you have a large number of calls. Reda On Thu, Apr 26, 2012 at 15:52, Charles Chance <charles.chance(a)sipcentric.com > wrote: > ** ** ** ** > > Hi Reda,**** > > ** ** > > Sorry, I should have been more specific – I am referring to instances > where the media server is for example Asterisk. If first call goes through > Asterisk 1, second call goes through Asterisk 2, when Asterisk 2 receives > the REFER it does not know of initial call on Asterisk 1. The only way we > have found for it to work is to ensure the second call is dispatched to the > same Asterisk box as the first.**** > > ** ** > > I would be pleased to hear of an alternative method.**** > > ** ** > > Regards,**** > > ** ** > > Charles**** > ------------------------------ > > *From:* sr-users-bounces(a)lists.sip-router.org [mailto: > sr-users-bounces(a)lists.sip-router.org] *On Behalf Of *Reda Aouad > *Sent:* 26 April 2012 14:34 > > *To:* **SIP Router - Kamailio** (OpenSER) and SIP Express Router (SER) - > UsersMailing List > *Subject:* Re: [SR-Users] dispatcher and call transfer > **** > > ** ** > > Hi, > > @Carsten > Dispatcher algorithm 0 based on call-id should do it in your case of > re-invite within dialog with same call-id. > > @Charles > In the case of attended transfers, shouldn't both media servers be > relaying media between them? I didn't understand why your are obliged to > dispatch to the same media server since they are 2 different calls with > different call-ids.**** > > ** ** > > Reda**** > > ** ** > > ** ** > > On Thu, Apr 26, 2012 at 14:30, Charles Chance < > charles.chance(a)sipcentric.com> wrote:**** > > Hi, > > Actually, this won't help for attended transfers where another call is > initiated first then the two are joined together by REFER. In this case, > the > second INVITE must be routed to the same media server as the existing call > for the transfer to work. > > What we do is store the dialogs in DB, then when a new call comes in, prior > to doing ds_select_dst we query DB for existing call involving same user. > If > we find one, we simply replace destination host with that from the contact > (to/from depending on direction of call). > > It may not be the most elegant way but it works for us :) > > Charles**** > > > > -----Original Message----- > From: sr-users-bounces(a)lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Carsten Bock > Sent: 26 April 2012 13:25 > To: **SIP Router - Kamailio** (OpenSER) and SIP Express Router (SER) - > UsersMailing List > Subject: Re: [SR-Users] dispatcher and call transfer > > Hi, > > if you look at the docs of the dispatcher module, you'll find this: > > alg - the algorithm used to select the destination address. The > parameter can be an integer or a variable holding an interger. > - “0” - hash over callid > ( > http://kamailio.org/docs/modules/devel/modules_k/dispatcher.html#id2498492 > ) > > But probably you should look into record/loose_route for your setup. > Since REFER is normally an in-dialog request (belongs to another > voice-session), it should take the same route as the initial INVITE. > This is normally achieved by the record/loose-route mechanisms > described in RFC3261. > In the example config of Kamailio you find an configuration example > (below). > > It is not a bug in the dispatcher module, it's how you use it. > > So long, > Carsten > > 455 # handle requests within SIP dialogs > 456 route(WITHINDLG); > > [...] > > 473 # record routing for dialog forming requests (in case > they are routed) > 474 # - remove preloaded route headers > 475 remove_hf("Route"); > 476 if (is_method("INVITE|SUBSCRIBE")) > 477 record_route(); > > and the relevent parts in the "WITHINDLG" route: > > 566 # Handle requests within SIP dialogs > 567 route[WITHINDLG] { > 568 if (has_totag()) { > 569 # sequential request withing a dialog should > 570 # take the path determined by record-routing > 571 if (loose_route()) { > [...] > 580 route(RELAY); > 581 } else { > [...] > 587 if ( is_method("ACK") ) { > 588 if ( t_check_trans() ) { > 589 # no loose-route, but stateful > ACK; > 590 # must be an ACK after a 487 > 591 # or e.g. 404 from upstream > server > 592 t_relay(); > 593 exit; > 594 } else { > 595 # ACK without matching > transaction ... ignore and discard > 596 exit; > 597 } > 598 } > 599 sl_send_reply("404","Not here"); > 600 } > 601 exit; > 602 } > 603 } > > 2012/4/26 Asgaroth <00asgaroth00(a)gmail.com>: > > Hi All, > > > > Currently we are running kamailio in a loadbalanced fashion whereby calls > > come in via the loadbalancers and distribute calls accross 2 media > servers. > > We have come accross and issue whereby call transfers may be distributed > > accross two media servers and when the REFER message comes along to > transfer > > the call, in some cases (if we're lucky) the message arrives at the wrong > > media server (transaction leg doesnt exist). > > > > Some googling later and it appears that dispatcher doesnt play nice when > it > > comes to this scenario. Some suggestions popped up in my previous > searches > > saying that a potential work around is to use the dialog module to check > if > > a call is eastablished and then to send all calls to the same media > server > > based on the dialog already being established. > > > > I'd appreciate some input from the guru's out there that have come > accross > > this same issue and, if possible, some suggestions on how to work around > the > > problem, does the dispatcher module have a hashing algorithm that can be > > suited for this particular scenario? > > > > Thanks in advance for any tips or sugestions. > > > > _______________________________________________ > > 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 > > > > > > -- > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 **Hamburg** / ****Germany**** > > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > Mobile +49 179 2021244 <%2B49%20179%202021244> > Office +49 40 34927219 <%2B49%2040%2034927219> > Fax +49 40 34927220 > > Sitz der Gesellschaft: ****Hamburg**** > Registergericht: Amtsgericht ****Hamburg****, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/ > > -- > Meet ng-voice at LinuxTag 2012 in ****Berlin**** - May 23rd - 26th, 2012. > Save the > date! > > _______________________________________________ > 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**** > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1913 / Virus Database: 2411/4959 - Release Date: 04/25/12* > *** > > > > _______________________________________________ > 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**** > > ** ** > ------------------------------ > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1913 / Virus Database: 2411/4959 - Release Date: 04/25/12* > *** > > _______________________________________________ > 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 > >

2 1

Re: [SR-Users] About MSRP relay
by 罗俊明 26 Apr '12

26 Apr '12

3 2

Re: [SR-Users] dispatcher and call transfer
by Reda Aouad 26 Apr '12

26 Apr '12

Hi, @Carsten Dispatcher algorithm 0 based on call-id should do it in your case of re-invite within dialog with same call-id. @Charles In the case of attended transfers, shouldn't both media servers be relaying media between them? I didn't understand why your are obliged to dispatch to the same media server since they are 2 different calls with different call-ids. Reda On Thu, Apr 26, 2012 at 14:30, Charles Chance <charles.chance(a)sipcentric.com > wrote: > Hi, > > Actually, this won't help for attended transfers where another call is > initiated first then the two are joined together by REFER. In this case, > the > second INVITE must be routed to the same media server as the existing call > for the transfer to work. > > What we do is store the dialogs in DB, then when a new call comes in, prior > to doing ds_select_dst we query DB for existing call involving same user. > If > we find one, we simply replace destination host with that from the contact > (to/from depending on direction of call). > > It may not be the most elegant way but it works for us :) > > Charles > > > -----Original Message----- > From: sr-users-bounces(a)lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Carsten Bock > Sent: 26 April 2012 13:25 > To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > UsersMailing List > Subject: Re: [SR-Users] dispatcher and call transfer > > Hi, > > if you look at the docs of the dispatcher module, you'll find this: > > alg - the algorithm used to select the destination address. The > parameter can be an integer or a variable holding an interger. > - “0” - hash over callid > ( > http://kamailio.org/docs/modules/devel/modules_k/dispatcher.html#id2498492 > ) > > But probably you should look into record/loose_route for your setup. > Since REFER is normally an in-dialog request (belongs to another > voice-session), it should take the same route as the initial INVITE. > This is normally achieved by the record/loose-route mechanisms > described in RFC3261. > In the example config of Kamailio you find an configuration example > (below). > > It is not a bug in the dispatcher module, it's how you use it. > > So long, > Carsten > > 455 # handle requests within SIP dialogs > 456 route(WITHINDLG); > > [...] > > 473 # record routing for dialog forming requests (in case > they are routed) > 474 # - remove preloaded route headers > 475 remove_hf("Route"); > 476 if (is_method("INVITE|SUBSCRIBE")) > 477 record_route(); > > and the relevent parts in the "WITHINDLG" route: > > 566 # Handle requests within SIP dialogs > 567 route[WITHINDLG] { > 568 if (has_totag()) { > 569 # sequential request withing a dialog should > 570 # take the path determined by record-routing > 571 if (loose_route()) { > [...] > 580 route(RELAY); > 581 } else { > [...] > 587 if ( is_method("ACK") ) { > 588 if ( t_check_trans() ) { > 589 # no loose-route, but stateful > ACK; > 590 # must be an ACK after a 487 > 591 # or e.g. 404 from upstream > server > 592 t_relay(); > 593 exit; > 594 } else { > 595 # ACK without matching > transaction ... ignore and discard > 596 exit; > 597 } > 598 } > 599 sl_send_reply("404","Not here"); > 600 } > 601 exit; > 602 } > 603 } > > 2012/4/26 Asgaroth <00asgaroth00(a)gmail.com>: > > Hi All, > > > > Currently we are running kamailio in a loadbalanced fashion whereby calls > > come in via the loadbalancers and distribute calls accross 2 media > servers. > > We have come accross and issue whereby call transfers may be distributed > > accross two media servers and when the REFER message comes along to > transfer > > the call, in some cases (if we're lucky) the message arrives at the wrong > > media server (transaction leg doesnt exist). > > > > Some googling later and it appears that dispatcher doesnt play nice when > it > > comes to this scenario. Some suggestions popped up in my previous > searches > > saying that a potential work around is to use the dialog module to check > if > > a call is eastablished and then to send all calls to the same media > server > > based on the dialog already being established. > > > > I'd appreciate some input from the guru's out there that have come > accross > > this same issue and, if possible, some suggestions on how to work around > the > > problem, does the dispatcher module have a hashing algorithm that can be > > suited for this particular scenario? > > > > Thanks in advance for any tips or sugestions. > > > > _______________________________________________ > > 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 > > > > > > -- > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 Hamburg / Germany > > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > Mobile +49 179 2021244 > Office +49 40 34927219 > Fax +49 40 34927220 > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/ > > -- > Meet ng-voice at LinuxTag 2012 in Berlin - May 23rd - 26th, 2012. Save the > date! > > _______________________________________________ > 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 > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1913 / Virus Database: 2411/4959 - Release Date: 04/25/12 > > > _______________________________________________ > 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 >

2 1

dispatcher and call transfer
by Asgaroth 26 Apr '12

26 Apr '12

Hi All, Currently we are running kamailio in a loadbalanced fashion whereby calls come in via the loadbalancers and distribute calls accross 2 media servers. We have come accross and issue whereby call transfers may be distributed accross two media servers and when the REFER message comes along to transfer the call, in some cases (if we're lucky) the message arrives at the wrong media server (transaction leg doesnt exist). Some googling later and it appears that dispatcher doesnt play nice when it comes to this scenario. Some suggestions popped up in my previous searches saying that a potential work around is to use the dialog module to check if a call is eastablished and then to send all calls to the same media server based on the dialog already being established. I'd appreciate some input from the guru's out there that have come accross this same issue and, if possible, some suggestions on how to work around the problem, does the dispatcher module have a hashing algorithm that can be suited for this particular scenario? Thanks in advance for any tips or sugestions.

3 2

generating cdrs
by Pirjo Ahvenainen 26 Apr '12

26 Apr '12

Hi all, I'm currently playing with accounting and cdrs, and meaning to test cdr record generation. Using Kamailio 3.2.3 on Ubuntu. My question is how to enable Kamailio to save cdr data into database (MySql in my case)? I have loaded the dialog module and enabled the cdr generation: modparam("acc", "cdr_enable", 1) ... but no cdrs in db, what else do I need to do? The script is mostly default config, any logic I need to add etc? I'll be fine with the most basic cdrs first and will expand from there. I do get entries in syslog for invite and bye of the transaction. I assume the cdr recodrs should be visible in cdrs table, but have nothing there. According to the acc module documentation: "In order to set up CDR generation, you must enable the CDR switch and load the dialog module", does cdr switch here mean the cdr_enable module parameter of the acc module? cheers, Pirjo -- Pirjo Ahvenainen +35844 559 7729

3 9

reply code for lower cseq
by Daniel-Constantin Mierla 25 Apr '12

25 Apr '12

Hello, I was wondering if someone here can point quickly where specs mention what is the right reply code to send when a request within dialog is received with lower cseq value than the previous request. I couldn't spot the part in the RFC yet, if any related exists. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 7

Redirection fails
by Pepe 25 Apr '12

25 Apr '12

Hi, since few days ago I have many problems with the redirections in kamailio (3.1.1). The blind forwarding works fine, but the redirection don't. Here's my code: failure_route[FAIL_ONE] { xlog("L_INFO", "entering failure_route for reply code $T_reply_code\n"); if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } if (t_is_canceled()) { exit; } if (t_check_status("486|408|302")) { xlog("L_NOTICE", "r[FAIL] / &ru TimeOut or Busy\n"); if(avp_db_load("$ru/username","$avp(s:callbusy)")) { xlog("L_NOTICE", "Callbusy activado!\n"); avp_pushto ("$ru","$avp(s:callbusy)"); xlog("Redireccionando a $ru"); km_append_branch(); route(CONSISTENCIA); } t_relay(); } } route[CONSISTENCIA] { xlog("L_NOTICE", "KAM-INFO: r[CONSISTENCIA] / - CONSISTENCY FOR FORWARDINGS \n"); route(NAT); route(ALIAS); } When I make a call to a contact that is using the phone in this moment, i receive a 302 status code, instead of 486. The big problem is this, I make the call, the forwarding works well and call rings in the right destination but i can't hear anything until the primary destination terminates the call. Any help will be apreciated. Best regards.

2 2

Kamailio fails with public ip address for authentication
by Ryan Gholam 25 Apr '12

25 Apr '12

Hello there , I am trying now my kamailio on a real public ip , i am trying to connect from a client from a different public ip , but i always get from kamailio unauthorized eventho the username and password are correct and when i connect to the kamailio localy everything works

2 1

kamailio-sipcapture-modules package?
by Skyler 25 Apr '12

25 Apr '12

Hi all, Is there any kamailio-sipcapture-modules or kamailio-homer-modules package available for amd64? Skyler

3 3

xhttp and xmlrpc
by Krishna Kurapati 24 Apr '12

24 Apr '12

Hi, I have two routes event_route[xhttp:request] { } route[XMLRPC]{ } Both are using HTTP as transport. When I tried to send a XMLRPC POST command, the event_route is getting triggered. Do I need to handle the XMLRPC inside the event_route? Since I am using XMLRPC for Click2Dial kind of feature, I am wondering how the async responses be handled if I include in event_route. Krish Kura

2 2

Re: [SR-Users] RTPproxy and kamailio
by Ryan Gholam 24 Apr '12

24 Apr '12

So let me get this straight , so if i have a client behind a NAT , the client should be able to connect to the stun server ( which has the real ip for example : 192.168.10.1 ) , and by default i should be able to make the stun server connect to the kamailio (which also have a real ip for example : 192.168.10.2 ) . Thank you On 4/24/12, Pavel Segeč <Pavel.Segec(a)fri.uniza.sk> wrote: > HI, > > stun is working completely out of kamailio, it deal only witch SIP clients. > Older fashioned stun server is included for example inside of regular linux > repository, for example for debian > > palo73 > >> -----Original Message----- >> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >> bounces(a)lists.sip-router.org] On Behalf Of Ryan Gholam >> Sent: Tuesday, April 24, 2012 9:02 AM >> To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users >> Mailing List >> Subject: Re: [SR-Users] RTPproxy and kamailio >> >> Thank you it worked well enough , but do you have an idea if there any >> configuration for Stun server ? , or i just install stun server and link >> it up to kamailio ? >> >> On 4/23/12, Jan Klepal <jan(a)klepal.cz> wrote: >> > Hello Ryan, >> > >> > kamailio default config is already prepared for rtpproxy. Do exactly >> > what is noted in kamailio.cfg :-) >> > >> > - download and install rtpproxy from http://www.rtpproxy.org >> > - start rtpproxy: rtpproxy -l your.public.ip.address -s >> > udp:127.0.0.1:7722 >> > - start kamailio: kamailio -A WITH_NAT >> > >> > Function which controls rtpproxy is rtpproxy_manage(), documented >> here: >> > http://kamailio.org/docs/modules/3.2.x/modules/rtpproxy.html >> > >> > Check flags you can pass to rtpproxy_manage(), especially o and c. >> > >> > Cheers, >> > Jan >> > >> > On Monday 23 of April 2012 11:35:15 Ryan Gholam wrote: >> >> I would appreciate if someone can guide me through the setup to >> >> configure Rtpproxy with kamailio , and what changes should be made in >> >> the kamailio.cfg file . >> > >> > >> > _______________________________________________ >> > 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 >> > >> >> _______________________________________________ >> 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 > > > _______________________________________________ > 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 >

2 1

call for participation: kamailio project at LinuxTag 2012
by Carsten Bock 24 Apr '12

24 Apr '12

Hello, in about four weeks the kamailio project will (again) be present on the LinuxTag fair, "the most important place for Linux and open source software in Europe" - http://www.linuxtag.org/2012/en.html . If you are in Berlin during this time, plan to visit the LinuxTag or just like to help your favourite SIP server project - please consider supporting us in running our fair booth! Even showing up for one or two hours helps a lot. We need also support to properly present our project, like flyers, a banner, maybe some live demonstrations.. So even if you can't participate in person, there are other ways to help us! Please contact me per mail if you could help at the booth or have some ideas about our project presentation. The actual state of preparations can be found in our wiki: http://sip-router.org/wiki/meetings/berlin-2012 Thanks in advance, Carsten -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ -- Meet ng-voice at LinuxTag 2012 in Berlin - May 23rd - 26th, 2012. Save the date!

1 0

RTPproxy and kamailio
by Ryan Gholam 24 Apr '12

24 Apr '12

I would appreciate if someone can guide me through the setup to configure Rtpproxy with kamailio , and what changes should be made in the kamailio.cfg file . Thank you .

3 3

Development frozen for v3.3.0
by Daniel-Constantin Mierla 24 Apr '12

24 Apr '12

Hello, just short message to mark the freezing of the new features development for v3.3.0. For about 1-1.5 months, git master branch should be used to commit only fixes to existing features, improvements to documentation and eventually merges of the duplicated modules without affecting existing database structure. Of course, developers can use personal branches to add new features, which can be merged later in master, once branch 3.3 will be created. Database structure should be in its final format for 3.3.0. The main focus should be now on testing master branch to be sure all goes ok. From my point of view, I would like to ask special attention to registrar/usrloc modules that introduced support for SIP outbound and GRUU, having the location table structure updated. Also add to the wiki pages details about migration from 3.2.x to 3.3.0: * http://www.kamailio.org/wiki/install/upgrade/3.2.x-to-3.3.0 If you discover new features not listed there, add to the page: * http://www.kamailio.org/wiki/features/new-in-devel Cheers, Danie -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

kamailio 3.2.2 - uac calls from sipgate.de - rr [loose.c:103]: failed to parse Route HF
by Karsten Horsmann 23 Apr '12

23 Apr '12

Hello List, after my multihomed setup seems to be fine i have an question about this log entries: Apr 18 10:02:28 kamailio1 rtpproxy[15286]: INFO:handle_command: delete request failed: session 153a2e592314d58d51897ed10187f0da(a)sipgate.de, tags as11f6a774/ds-32764e7d-e76ac287 not found Apr 18 10:02:28 kamailio1 /usr/local/sbin/kamailio[27071]: DEBUG: <script>: BYE OR CANCEL. Dialog lifetime: <null> Status: <null> Apr 18 10:02:28 kamailio1 /usr/local/sbin/kamailio[27071]: DEBUG: <core> [parser/parse_rr.c:62]: parse_rr_body(): No body for record-route Apr 18 10:02:28 kamailio1 /usr/local/sbin/kamailio[27071]: ERROR: rr [loose.c:103]: failed to parse Route HF Apr 18 10:02:28 kamailio1 /usr/local/sbin/kamailio[27071]: ERROR: rr [loose.c:103]: failed to parse Route HF Apr 18 10:02:28 kamailio1 /usr/local/sbin/kamailio[27071]: DEBUG: rr [loose.c:838]: There is no Route HF Any hints to fix that? -- Mit freundlichen Grüßen *Karsten Horsmann*

4 22

Freezing development today for release 3.3.0
by Daniel-Constantin Mierla 23 Apr '12

23 Apr '12

Hello, quick reminder for everyone that by end of today is planned the freezing for the development of the new features to be in version 3.3.0. I know there are going to be few hammerings to newly added features, when discovering issues, that's the testing phase for. But try to get at least the database structure settled by today, as most of the users need to adapt management interfaces to start using the new version. Cheers, Daniel

1 0

replace body content
by Konstantin M. 22 Apr '12

22 Apr '12

Hi, Is there a way to change/replace anything in body message ? Used: replace_body_atonce, replace_body_all, replace_all, subst_body. Nothing is working. Thanks,

1 0

Core Dump from select stmt
by Akan Technology 22 Apr '12

22 Apr '12

hello, I am running kamailio 3.2.2 on Solaris 10 64 bit and I am getting a core dump on this statement: $ru = "sip:" + $rU + "@" + $sel(cfg_get.registrar1.server_ip) + $sel(cfg_get.registrar1.server_port); Here is how the variable is defined: registrar1.server_ip = "xxx.xxx.xxx.xxx" desc "Registrar server IP address" registrar1.server_port = "5080" desc "Registrar server Port" Here is the output from the logs with debug: 10(25373) ERROR: *** cfgtrace: c=[/opt/kamailio-3.2/etc/kamailio/kamailio.cfg] l=1072 a=65 n=assign 10(25373) DEBUG: <core> [select.c:424]: Calling SELECT 1004b2bc0 10(25373) DEBUG: <core> [select.c:424]: Calling SELECT 1004b4da0 10(25373) ERROR: *** cfgtrace:14(25377) : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 22 14(25377) DEBUG: <core> [tcp_main.c:3555]: DBG: handle_ser_child: dead child 10, pid 25373 (shutting down?) 14(25377) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (1003738b8, 22, 0, 0x0) fd_no=18 called 14(25377) DEBUG: <core> [tcp_main.c:3316]: DBG: handle_tcp_child: dead tcp child 0 (pid 25373, no 10) (shutting down?) 14(25377) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (1003738b8, 24, 1, 0x0) fd_no=17 called 0(25363) ALERT: <core> [main.c:751]: child process 25373 exited by a signal 11 0(25363) ALERT: <core> [main.c:754]: core was generated 0(25363) INFO: <core> [main.c:766]: INFO: terminating due to SIGCHLD 14(25377) INFO: <core> [main.c:817]: INFO: signal 15 received Here is the full route code: route[TO_REGISTRAR] { # check if Registrar server's IP is defined if (strempty($sel(cfg_get.registrar1.server_ip))) { xlog("SCRIPT: REGISTRAR - registrar1.server_ip not defined\n"); return; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.registrar1.server_ip) + ":" + $sel(cfg_get.registrar1.server_port); route(RELAY); exit; } # End of TO_Registrar Route I am trying to setup the proxy server to route all registers requests to another kamailio server Thanks Nathaniel

4 10

reminder - freezing development for next major release v3.3.0
by Daniel-Constantin Mierla 20 Apr '12

20 Apr '12

Hello, a short reminder about freezing the development for next major release v3.3.0. It is planned for April 23, less than two weeks away. If you have new features to add for v3.3.0, commit them or send the patches before the date. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

3 4

htable feature request
by Andreas Granig 20 Apr '12

20 Apr '12

Hi, I'd like to ask for a feature request for htable. Would it be possible to add an MI function to delete specific entries from an htable, e.g. with "sercmd htable.remove <table> <key>" before freezing 3.3? That'd be handy for example if you hold blocked IPs in htable, but want to unblock some of them manually. At the moment, I only see the method of restarting kamailio to whipe the whole table (and it'd only work if dbmode=0). Thanks in advance, Andreas

2 2

Help about VoIP
by Zuzu 20 Apr '12

20 Apr '12

I got my sip account ipadvoip(a)iptel.org what is next step VoIP to complete plz reply thank you my email yahya_satofa(a)yahoo.com

1 0

SIP Attack
by Ricardo Martinez 20 Apr '12

20 Apr '12

Hello. I was wondering if someone could help me here. From time to time I stat to receive from the internet this SIP message : U 190.22.140.170:51316 -> 64.76.154.110:5060 SIP/2.0 400 BadRequest. Via: . From: . To: . Call-ID: . CSeq: . User-Agent: AddPac SIP Gateway. Content-Length: 0. . At burst rate of 124 pps (packets per second), this meesage is entering to Kamailio routine and generating a lot of ERROR logs lie these : Apr 1 03:32:19 kmborde /usr/local/sbin/kamailio[2311]: ERROR: <core> [parser/msg_parser.c:179]: ERROR: get_hdr_field: bad to header Apr 1 03:32:19 kmborde /usr/local/sbin/kamailio[2311]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [To: <sip:Re gister=>5] Apr 1 03:32:19 kmborde /usr/local/sbin/kamailio[2311]: ERROR: <core> [parser/msg_parser.c:179]: ERROR: get_hdr_field: bad to header Apr 1 03:32:19 kmborde /usr/local/sbin/kamailio[2311]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [To: <sip:Re gister=>5] Apr 1 03:32:19 kmborde /usr/local/sbin/kamailio[2311]: ERROR: <core> [parser/msg_parser.c:179]: ERROR: get_hdr_field: bad to header Apr 1 03:32:19 kmborde /usr/local/sbin/kamailio[2311]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [To: <sip:Re gister=>5] Apr 1 03:32:19 kmborde /usr/local/sbin/kamailio[2311]: ERROR: <core> [msg_translator.c:1943]: ERROR: build_res_buf_from_sip_req: al as, parse_headers failed Apr 1 03:32:19 kmborde /usr/local/sbin/kamailio[2311]: WARNING: sanity [sanity.c:254]: sanity_check(): check_required_headers(): fa iled to send 400 via sl reply Apr 1 03:32:20 kmborde /usr/local/sbin/kamailio[2301]: ERROR: <core> [parser/msg_parser.c:179]: ERROR: get_hdr_field: bad to header Apr 1 03:32:20 kmborde /usr/local/sbin/kamailio[2301]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [To: <sip:Re gister=>5] Apr 1 03:32:20 kmborde /usr/local/sbin/kamailio[2301]: ERROR: <core> [parser/msg_parser.c:179]: ERROR: get_hdr_field: bad to header Apr 1 03:32:20 kmborde /usr/local/sbin/kamailio[2301]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [To: <sip:Re gister=>5] Apr 1 03:32:20 kmborde /usr/local/sbin/kamailio[2301]: ERROR: <core> [parser/msg_parser.c:179]: ERROR: get_hdr_field: bad to header Apr 1 03:32:20 kmborde /usr/local/sbin/kamailio[2301]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [To: <sip:Re gister=>5] Apr 1 03:32:20 kmborde /usr/local/sbin/kamailio[2301]: ERROR: <core> [msg_translator.c:1943]: ERROR: build_res_buf_from_sip_req: al as, parse_headers failed Apr 1 03:32:20 kmborde /usr/local/sbin/kamailio[2301]: WARNING: sanity [sanity.c:254]: sanity_check(): check_required_headers(): fa iled to send 400 via sl reply Apr 1 03:32:23 kmborde /usr/local/sbin/kamailio[2320]: ERROR: <core> [parser/msg_parser.c:179]: ERROR: get_hdr_field: bad to header Apr 1 03:32:23 kmborde /usr/local/sbin/kamailio[2320]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [To: <sip:Re gister=>5] Apr 1 03:32:23 kmborde /usr/local/sbin/kamailio[2320]: ERROR: <core> [parser/msg_parser.c:179]: ERROR: get_hdr_field: bad to header Apr 1 03:32:23 kmborde /usr/local/sbin/kamailio[2320]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [To: <sip:Re gister=>5] Apr 1 03:32:23 kmborde /usr/local/sbin/kamailio[2320]: ERROR: <core> [parser/msg_parser.c:179]: ERROR: get_hdr_field: bad to header Apr 1 03:32:23 kmborde /usr/local/sbin/kamailio[2320]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [To: <sip:Re gister=>5] Apr 1 03:32:23 kmborde /usr/local/sbin/kamailio[2320]: ERROR: <core> [msg_translator.c:1943]: ERROR: build_res_buf_from_sip_req: al as, parse_headers failed Apr 1 03:32:23 kmborde /usr/local/sbin/kamailio[2320]: WARNING: sanity [sanity.c:254]: sanity_check(): check_required_headers(): fa iled to send 400 via sl reply The only way that I have now for blocking this packet to hit the Kamailio server is via iptables : iptables -A INPUT -s 190.22.140.170 -p udp --dport 5060 --jump REJECT Is there a better way to do this?! Thanks in advance, * * *Ricardo Martinez.-*

5 5

Re: [SR-Users] sr-users Digest, Vol 83, Issue 77
by Ali Wazni 20 Apr '12

20 Apr '12

Chefta men chway. Jaydeeeeee 7adret el consultant!!!!! Ali Wazni On 20 avr. 2012, at 08:41, 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. Kamailio v3.2.3 Released (Daniel-Constantin Mierla) > 2. Re: [sr-dev] reminder - freezing development for next major > release v3.3.0 (Peter Dunkley) > 3. Stateless Proxy (Akan) > 4. How does RLS work with kamailio. (Gnaneshwar Gatla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 19 Apr 2012 19:15:29 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: [SR-Users] Kamailio v3.2.3 Released > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org>, business(a)lists.kamailio.org > Message-ID: <4F904831.1030503(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello, > > Kamailio SIP Server v3.2.3 stable release is out. > > This is a maintenance release of the latest stable branch, 3.2, that > includes fixes since release of v3.2.2. There is no change to database > schema or configuration language structure that you have to do on > installations of v3.2.0, v3.2.1 or v3.2.2. Deployments running previous > v3.2.x versions are strongly recommended to be upgraded to v3.2.3. > > For more details about version 3.2.3 (including links and hints to > download the tarball or from GIT repository), visit: > > * http://www.kamailio.org/w/2012/04/kamailio-v3-2-3-released/ > > RPMs and Debian/Ubuntu packages will be available soon. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany > http://www.asipto.com/index.php/kamailio-advanced-training/ > > > > > ------------------------------ > > Message: 2 > Date: Thu, 19 Apr 2012 18:33:49 +0100 > From: Peter Dunkley <peter.dunkley(a)crocodile-rcs.com> > Subject: Re: [SR-Users] [sr-dev] reminder - freezing development for > next major release v3.3.0 > To: miconda(a)gmail.com, Development mailing list of the sip-router > project <sr-dev(a)lists.sip-router.org> > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <1334856829.2795.29.camel@pd-laptop-linux> > Content-Type: text/plain; charset="utf-8" > > Hi, > > One of the developments I am currently working on is to add notifier > processes to presence. I have already done this for RLS. This is > needed not just for performance, but because there are race conditions > (I see them under load) where you get multiple NOTIFY requests > outstanding on the same dialog and, when in DB only mode, they often > have the same CSeq (despite being different requests). > > Unfortunately, I don't believe I will get this finished by the end of > Monday (April 23). I still expect to complete this development next > week. > > As the CSeq issue is quite clearly a problem I would like to get this > development into Kamailio 3.3. Will it be OK for me to put this > development into master next week after the freeze? > > Thanks, > > Peter > > On Wed, 2012-04-11 at 11:34 +0200, Daniel-Constantin Mierla wrote: > >> Hello, >> >> a short reminder about freezing the development for next major release >> v3.3.0. It is planned for April 23, less than two weeks away. If you >> have new features to add for v3.3.0, commit them or send the patches >> before the date. >> >> Cheers, >> Daniel >> > > > -- > Peter Dunkley > Technical Director > Crocodile RCS Ltd >

1 0

XCAP_Client version error
by Gnaneshwar Gatla 20 Apr '12

20 Apr '12

Hey, In Kamailio 3.2.2: I was working with XCAP and RLS today and found that the XCAP client has not been updated to the latest version of the xcap table. This resulted in error while starting Kamailio. Currently in modules_k/xcap_client/xcap_client.c : #define XCAP_TABLE_VERSION 3 I have set this macro to 4 and started the Kamailio instance. This is in regard to the followup email for rls service, does this have any impact to the rls module? Regards Gnani Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

2 1

Stateless Proxy
by Akan 20 Apr '12

20 Apr '12

Hello, I am trying to setup a stateless proxy where certain requests are forwarded to another sip server that will handle those requests specifically. I have the request message forwarding to the other sip server but the reply is being sent back to the original sip server then to the client. When I look at the SIP request and reply, I see a new "Via" header line with the IP address of the original sip proxy server. What I am trying to setup is a stateless proxy which just forwards the original request to another sip server and then this second sip server will handle everything from there. Basically offloading the traffic to other servers. Currently i am rewriting the ruri host and port as done in the sample config and issuing the forward() function. I have tried t_relay() as also in the sample config but I get the same results. Is there something else that I need to do before forwarding to the other sip server? How can I get rid of the new "Via" line? I believe this is from record_route() but I am not issue this before I perform the relay. I have Kamailio 3.2.3 loaded and used the sample config as my base config. Thanks Nathaniel

1 0

How does RLS work with kamailio.
by Gnaneshwar Gatla 19 Apr '12

19 Apr '12

Hello, I'm trying to implement RLS with integrated XCAP server. I have successfully run the Kamailio instance with the following snippet. Since I had no RLS clients available, I have tried to test this with SIPp. The attached pcap trace is the result. Test: 1. Send an XCAP signal over http to Kamailio. 2. Register and subscribe Kamailio. The test does store the xml received over http in the xcap table in the database. But rls module does not store the data neither in rls_presentity or rls_watcher. When I sent the Subscribe with sipp, the server throws error that subscription was not found in the database. I would like to know if I'm doing something wrong to implement this service. Is there any brief tutorial on how to implement this? Kamailio.cfg snippet: ##Presence modparam("presence", "server_address", PRESENCE_IP) modparam("presence", "clean_period", 65) modparam("presence", "db_update_period", 60) modparam("presence", "max_expires", 75) modparam("presence", "fallback2db", 1) modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 0) modparam("presence_xml", "integrated_xcap_server", 1) ##rls modparams modparam("rls", "db_url", DBURL) modparam("rls", "waitn_time", 10) modparam("rls", "integrated_xcap_server", 1) modparam("rls", "max_notify_body_length", 32000) modparam("rls", "to_presence_code", 10) modparam("rls", "server_address", RLS_URI) modparam("rls", "outbound_proxy", PUA_OUTBOUND_PROXY) modparam("rls", "xcap_root", XCAP_ROOT) ##XCAP modparams modparam("xcap_server", "db_url", DBURL) modparam("xcap_client", "db_url", DBURL) modparam("xcap_client", "query_period", 50) route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { $var(ret_code)= rls_handle_subscribe(); xlog("Subscribe: $var(ret_code)"); if($var(ret_code)== 10) handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } In the [xhttp:request]: switch($rm) { case "PUT": xcaps_put("$var(uri)", "$hu", "$rb"); if($xcapuri(u=>auid)=~"pres-rules") { xlog("===== xhttp put: refreshing watchers for $var(uri)\n"); pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } else if ($xcapuri(u=>auid)=~"resource-lists" || $xcapuri(u=>auid)=~"rls-services") { xlog("==== xhttp put: refreshing rls watchers $var(uri)\n"); rls_update_subs("$var(uri)", "presence"); } exit; break; Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

1 0

Kamailio v3.2.3 Released
by Daniel-Constantin Mierla 19 Apr '12

19 Apr '12

Hello, Kamailio SIP Server v3.2.3 stable release is out. This is a maintenance release of the latest stable branch, 3.2, that includes fixes since release of v3.2.2. There is no change to database schema or configuration language structure that you have to do on installations of v3.2.0, v3.2.1 or v3.2.2. Deployments running previous v3.2.x versions are strongly recommended to be upgraded to v3.2.3. For more details about version 3.2.3 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/04/kamailio-v3-2-3-released/ RPMs and Debian/Ubuntu packages will be available soon. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

Distributed subscribers Kamailio
by Juanjo 19 Apr '12

19 Apr '12

Hello everybody I have a question about clustering kamailio, i want to register subscribers to two diferent kamailios, and the subscribers from one kamailio should be able to communicate with the subscribers registered in the other kamailio How can i do it? Is there something like DUNDI in the Asterisk world? Thank you in advance.

2 1

uac module incoming INVITE become wrong dst_ip address
by Karsten Horsmann 19 Apr '12

19 Apr '12

Hello List, i am nearly finishing my multihomed config (public ip/internal ip) with kamailio. For incoming calls i want to use the uac module too. Then i want to locate an permanet entry and dispatch it to ivr system behind kamailio. i tried something like this: if (lookup("location") && (dst_ip==222.222.222.222)) { if(ds_select_domain("1", "4")) { Okay for registerd UAC this works perfect - but if made an call to the uac registerd number i got as $Ri the 172.20.100.74 - and thats is the private ip address. Why that??? Any hints to check "if its internal or external" call??? -- Mit freundlichen Grüßen *Karsten Horsmann*

2 3

sqlite -> usrloc tables deleted after 1 minute
by Pedro Antonio Vico Solano 19 Apr '12

19 Apr '12

2 6

Releasing v3.2.3
by Daniel-Constantin Mierla 19 Apr '12

19 Apr '12

Hello, last patch release (v3.2.2) was done quite some time ago, therefore I am thinking of releasing v3.2.3 sometime next week, maybe Wednesday or Thursday. If there are patches in master branch to be backported to 3.2, please start doing them. Also, if anyone is aware of issues in 3.2 branch not reported to the tracker, announce them asap. Thanks, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

1 2

Stress test
by Lucas Alvarez 18 Apr '12

18 Apr '12

Can someone recommend a good stress tool for testing kamailio? Or can point through the right path? Thanks in advance. Lucas

3 2

implementing Kamailio with FreeSwitch.
by Mehdi Badiezadegan 18 Apr '12

18 Apr '12

Hello and thanks for your helping,excuse me I'm newbie in mailing list. I have a question about implementing Kamailio with FreeSwitch. Our plan is a configuration like this at the follow: 1.Kamailio as registering and LoadBalancer ,response to chat and call(voice/video) in p2p 2.we have 2 FreeSwitch media engine just for Conference case(Voice/Video) . How we can implementing above kamailio as loadBalancer for 2 FreeSwitchs in conference case , And at the same time,It answer as voip/chat server for p2p case ? we have success by implementing Kamailio with one FreeSwitch, (FreeSwitch as media Engine just for conference and Kamailio for p2p call) But if we have 2 FreeSwitch with same activity,How we can implementing... And So I had Efforts with dispatcher module, but it was good only for route/Loadbalancer not for answering p2p call. Thanks a lot Mehdi Badiezadegan

2 1

loose_route() for ACK message
by Laurent Schweizer 18 Apr '12

18 Apr '12

Hello all, I have a script working with openser1.2.1 and I try to move them to kamailio 3.2.1 I have actually an issue with the loose_route() function at least for ACK message. When I get the ACK the loose_route function is changing the URI with the value of the route header of the ACK message . The problem is that this value is the local proxy himself so the t_relay() function send the ACK on the loopback interface. Any Idea ? Regards Laurent Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: START REQ:[ACK] Tran:[7070] URI:[sip:peoplefone@95.128.80.93:7070] from:[sip:41763770377@sip.call.lt;user=phone] to:[sip:37052077800@81.16.238.195;user=phone] CallID:[25694-UI-05aa645a-7000c6ef6@sip.call.lt] Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: SANITY OK, REQ:[ACK] Tran:[7070] URI:[sip:peoplefone@95.128.80.93:7070] from:[sip:41763770377@sip.call.lt;user=phone] to:[sip:37052077800@81.16.238.195;user=phone] CallID:[25694-UI-05aa645a-7000c6ef6@sip.call.lt] Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: DIALOG1 REQ:[ACK] Tran:[7070] URI:[sip:peoplefone@95.128.80.93:7070] from:[sip:41763770377@sip.call.lt;user=phone] to:[sip:37052077800@81.16.238.195;user=phone] CallID:[25694-UI-05aa645a-7000c6ef6@sip.call.lt] If (loose_route()) is called Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: DIALOG1ACK REQ:[ACK] Tran:[5060] URI:[sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2] from:[sip:41763770377@sip.call.lt;user=phone] to:[sip:37052077800@81.16.238.195;user=phone] CallID:[25694-UI-05aa645a-7000c6ef6@sip.call.lt] Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: NATMANAGE REQ:[ACK] Tran:[5060] URI:[sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2] from:[sip:41763770377@sip.call.lt;user=phone] to:[sip:37052077800@81.16.238.195;user=phone] CallID:[25694-UI-05aa645a-7000c6ef6@sip.call.lt] Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: DIAL_RELAY1 REQ:[ACK] Tran:[5060] URI:[sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2] from:[sip:41763770377@sip.call.lt;user=phone] to:[sip:37052077800@81.16.238.195;user=phone] CallID:[25694-UI-05aa645a-7000c6ef6@sip.call.lt] Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: T_RELAY REQ:[ACK] Tran:[5060] URI:[sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2] from:[sip:41763770377@sip.call.lt;user=phone] to:[sip:37052077800@81.16.238.195;user=phone] CallID:[25694-UI-05aa645a-7000c6ef6@sip.call.lt] U 81.16.238.195:5060 -> 87.110.233.1:5060 SIP/2.0 200 OK. To: <sip:37052077800@81.16.238.195;user=phone>;tag=e868748c. Via: SIP/2.0/UDP 87.110.233.1:5060;rport=5060;branch=z9hG4bK-26E6-2AB3E9F. Record-Route: <sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2>. CSeq: 79950814 INVITE. Call-ID: 25694-UI-05aa645a-7000c6ef6(a)sip.call.lt. From: <sip:41763770377@sip.call.lt;user=phone>;tag=25694-BR-05aa645b-0f448aeb2. Content-Type: application/sdp. Contact: <sip:peoplefone@95.128.80.93:7070>. Content-Length: 205. . v=0. o=90706832381 5012 11 IN IP4 81.16.238.195. s=Mapping. c=IN IP4 81.16.238.195. t=0 0. m=audio 8170 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=sendrecv. U 87.110.233.1:5060 -> 81.16.238.195:5060 ACK sip:peoplefone@95.128.80.93:7070 SIP/2.0. Call-ID: 25694-UI-05aa645a-7000c6ef6(a)sip.call.lt. Contact: <sip:87.110.233.1:5060>. CSeq: 79950814 ACK. From: <sip:41763770377@sip.call.lt;user=phone>;tag=25694-BR-05aa645b-0f448aeb2. Max-Forwards: 30. Route: <sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2>. To: <sip:37052077800@81.16.238.195;user=phone>;tag=e868748c. User-Agent: Cirpack/v4.42o (gw_sip). Via: SIP/2.0/UDP 87.110.233.1:5060;branch=z9hG4bK-388F-2AB3F03. Content-Length: 0. .

2 1

dropped calls after 38 seconds openSER
by Saul Waizer 18 Apr '12

18 Apr '12

Greetings list, I am experiencing a strange behavior with openser 1.3.2 running on ubuntu 10. I have a basic configuration (see bellow) and i am using Linphone for iPad as my client. I have 2 users registered and I am able to place calls no problem. The problem is that the calls (audio or A/V) drop after 38 seconds exactly, this behavior is pretty consistent, 38 seconds is all I can get. There is no firewall in front of the clients. Here is my configuration, ip addresses changed to protect the innocent: http://pastie.org/private/x1ck8rxjcxv6hl44hrmqg You can see the logs of the call here (the majority): http://pastie.org/private/4fj5efpbsrxan8plzqvfza Am I missing something or is there anything that needs to be changed in the routing/configuration to achieve basic functionality? Thank you in advance!

5 15

Re: [SR-Users] CallControl and MediaProxy
by Daniel-Constantin Mierla 18 Apr '12

18 Apr '12

Hello, I will patch and backport during next days -- I was mostly out of the office during past weeks. Cheers, Daniel On 3/12/12 11:30 AM, Reda Aouad wrote: > Hi Daniel, > > Any plans to backport this to 3.2 ? > I could still do the changes manually before compilation if you don't > have time to do it. > > Thank you again > Reda > > > > On Wed, Feb 29, 2012 at 10:19, Reda Aouad <reda.aouad(a)gmail.com > <mailto:reda.aouad@gmail.com>> wrote: > > Daniel, It works with flag 28. > Can you confirm that flag 28 isn't used by another module? > If so, can you patch it? When is the next release scheduled for? > > The following are the changes I made: > > modules_k/call_control.c: flag changed to 28 > ------------------------------------------- > #define FL_USE_CALL_CONTROL (1<<28) // use call control for > a dialog > > parser/msg_parser.h: warning added > ------------------------------------------- > /* WARNING: Value (1 << 28) is temporarily reserved for use in > kamailio call_control > * module (flag FL_USE_CALL_CONTROL )! */ > > > Thank you :) > Reda > > > > On Wed, Feb 29, 2012 at 09:58, Reda Aouad <reda.aouad(a)gmail.com > <mailto:reda.aouad@gmail.com>> wrote: > > A quick grep on flags FL_* in the sources shows the following : > Flag 29 is used by acc module, 31 by nat_traversal, 0 to 12 in > the parser. > Thus I assume that it is safe to test using flag 28. > I'll keep you posted on the test result. > > You'll also find below warnings in msg_parser.h for the used > flags. Since flag 30 is declared for mediaproxy in > msg_parser.h, I'll change the flag of callcontrol to 28. > > ----------------------------------------------------------------- > /* WARNING: Value (1 << 29) is temporarily reserved for use in > kamailio acc > * module (flag FL_REQ_UPSTREAM)! */ > > /* WARNING: Value (1 << 30) is temporarily reserved for use in > kamailio > * media proxy module (flag FL_USE_MEDIA_PROXY)! */ > > /* WARNING: Value (1 << 31) is temporarily reserved for use in > kamailio > * nat_traversal module (flag FL_DO_KEEPALIVE)! */ > ----------------------------------------------------------------- > > $ grep -R 'define FL.* (1' src/kamailio/kamailio-3.2.0 > > modules_k/call_control/call_control.c:#define > FL_USE_CALL_CONTROL (1<<30) // use call control for a dialog > modules_k/nat_traversal/nat_traversal.c:#define > FL_DO_KEEPALIVE (1<<31) > modules_k/acc/acc.h:#define FL_REQ_UPSTREAM (1<<29) > parser/msg_parser.h:#define FL_FORCE_RPORT (1 << 0) /*!< force > rport */ > parser/msg_parser.h:#define FL_FORCE_ACTIVE (1 << 1) /*!< > force active SDP */ > parser/msg_parser.h:#define FL_SDP_IP_AFS (1 << 2) /*!< SDP > IP rewritten */ > parser/msg_parser.h:#define FL_SDP_PORT_AFS (1 << 3) /*!< SDP > port rewritten */ > parser/msg_parser.h:#define FL_SHM_CLONE (1 << 4) /*!< msg > cloned in SHM as a single chunk */ > parser/msg_parser.h:#define FL_TIMEOUT (1 << 5) /*!< > message belongs to an "expired" branch > parser/msg_parser.h:#define FL_REPLIED (1 << 6) /*!< > message branch received at least one reply > parser/msg_parser.h:#define FL_HASH_INDEX (1 << 7) /*!< > msg->hash_index contains a valid value (tm use)*/ > parser/msg_parser.h:#define FL_MTU_TCP_FB (1 << 8) > parser/msg_parser.h:#define FL_MTU_TLS_FB (1 << 9) > parser/msg_parser.h:#define FL_MTU_SCTP_FB (1 << 10) > parser/msg_parser.h:#define FL_ADD_LOCAL_RPORT (1 << 11) /*!< > add 'rport' to local via hdr */ > parser/msg_parser.h:#define FL_SDP_BODY (1 << 12) /*!< > msg has SDP in body */ > modules/mediaproxy/mediaproxy.c:#define FL_USE_MEDIA_PROXY (1<<30) > > ----------------------------------------------------------------- > > > Reda > > > > On Wed, Feb 29, 2012 at 00:18, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > That should be. Try changing one of them to (1<<29) and > see if all works fine. > > On another hand, defining and using core msg flags in a > module is a risk, a different solution has to be done, a > simple one is to move the definition of these flags in the > core, so there will be no overlap in the future. > > Cheers, > Daniel > > > On 2/27/12 9:32 PM, Reda Aouad wrote: >> I looked into mediaproxy.c and found the following : >> >> ------------------------------------------------------- >> #define FL_USE_MEDIA_PROXY (1<<30) >> >> ... >> >> # dialog callback >> >> __dialog_created (...) { >> .... >> if ((request->msg_flags & FL_USE_MEDIA_PROXY) == 0) >> return; >> .... >> use_media_proxy (...); >> } >> ------------------------------------------------------- >> >> >> I also found this in call_control.c >> ------------------------------------------------------- >> #define FL_USE_CALL_CONTROL (1<<30) >> >> # Public API >> CallControl (...) { >> ... >> msg->msg_flags |= FL_USE_CALL_CONTROL; >> ... >> } >> ------------------------------------------------------- >> >> So I suspect that since the call_control module uses the >> same flag as the mediaproxy module, call_control function >> is used, flag 30 is set, and the following condition in >> the __dialog_created callback function above is never met >> >> (request->msg_flags & FL_USE_MEDIA_PROXY) == 0 >> >> so the callback function continues until executing its >> last line : use_media_proxy (...) >> which is called on every call to call_control ( ) function.. >> >> Reda >> >> >> >> On Mon, Feb 27, 2012 at 18:39, Reda Aouad >> <reda.aouad(a)gmail.com <mailto:reda.aouad@gmail.com>> wrote: >> >> Ok thanks Daniel. >> >> I'll do what you suggested and we'll see how to proceed. >> >> Thanks again >> Reda >> > > > -- > Daniel-Constantin Mierla --http://www.asipto.com > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

2 2

SIP Status: 401 Unauthorized (0 bindings)
by Saul Waizer 17 Apr '12

17 Apr '12

Hello List, I have a fresh install kamailio 3.2.0 running on ubuntu. Everything seems to work fine according to the logs, I have enabled support for the following: - enabled support to connect to MySQL server - enabled user authentication - enabled persistent user location service - enabled NAT traversal and updated RTPProxy control socket parameter I added a few users with kamctl add user pass but when I try to register with my client I get the following: 8.257745 x.x.x.x -> 10.62.99.103 SIP Request: REGISTER sip:x.x.x.x 8.258462 10.62.99.103 -> x.x.x.x SIP Status: 401 Unauthorized (0 bindings) Here are some detailed logs for the reg process: T x.x.x.x:63171 -> 10.62.99.103:5060 [AP] REGISTER sip:x.x.x.x SIP/2.0. Via: SIP/2.0/TCP 192.168.126.19:4369;rport;branch=z9hG4bK239028812. From: <sip:daniel@x.x.x.x>;tag=303867081. To: <sip:daniel@x.x.x.x>. Call-ID: 1300614569. CSeq: 7 REGISTER. Contact: <sip:daniel@x.x.x.x:63171;transport=tcp;line=5a9552052fb65c8>. Max-Forwards: 70. User-Agent: Linphone/3.5.0 (eXosip2/3.6.0). Expires: 600. Content-Length: 0. . T 10.62.99.103:5060 -> x.x.x.x:63171 [AP] SIP/2.0 401 Unauthorized. Via: SIP/2.0/TCP 192.168.126.19:4369 ;rport=63171;branch=z9hG4bK239028812;received=x.x.x.x. From: <sip:daniel@x.x.x.x>;tag=303867081. To: <sip:daniel@x.x.x.x>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b260. Call-ID: 1300614569. CSeq: 7 REGISTER. WWW-Authenticate: Digest realm="x.x.x.x", nonce="T42z9U+NssnZP82+sdkwPgl1nRk61QDm". Server: kamailio (3.2.0 (i386/linux)). Content-Length: 0. Any help is greatly appreciated.

2 3

How to configure Kamailio routing logic based on mobile phone numbers
by Tam Doan 17 Apr '12

17 Apr '12

Hi there, Now I could configure Kamailio routing logic based on service's short code (i.e route calls to 200, 300 to separated service's appliction (so call, app serv 1 and app serv 2). My problem is how can i configure Kamailio so that it can route the call from mobile numbers, such as calls from mobile number starting with 0988 xxxxx, 0978 xxxxx will be routed to app serv1 and the calls of 0976 xxxxx to app serv 2. So do you guys have an answer for me ? I would really appreciate your help!!!

1 0

registrar/usrloc updates - gruu support (rfc 5627)
by Daniel-Constantin Mierla 17 Apr '12

17 Apr '12

Hello, I just pushed GRUU support in git master branch. That implied adding two new columns to location table, therefore if you use devel version, **you have to update your db schema**. Perhaps a bit of tuning will follow in the next days, but GRUU support should be fully functional in terms of routing. Testing and feedback will be very appreciated as well as hints on what SIP hard/phones support GRUU properly. Be also aware that now you may have to do lookup location even for within dialog requests, if the r-uri is a local GRUU. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

1 1

dispatcher didnt use the gateway
by Karsten Horsmann 17 Apr '12

17 Apr '12

Hi List! I try to use the dispatcher module to route calls to my ivr systems. But it didnt load the gateway. I didnt see my xlog message after the ds_select_dst() Any hints? Gateway is Active and Probing. #!ifdef WITH_DISPATCHER if (!lookup("location")) { if(ds_select_dst("1", "4")) xlog("DISPATCHER ds_select_dst\n"); #forward(); } #!endif [root@kamailio1 log]# kamctl fifo ds_list SET_NO:: 1 SET:: 1 URI:: sip:172.20.100.61:5060 flags=AP priority=0 attrs= -- Mit freundlichen Grüßen *Karsten Horsmann*

4 10

PDD Timer (Branch Level)
by Orlando F. Guitian 16 Apr '12

16 Apr '12

Recently, I have used TM's fr_timer() parameter to control PDD, non-responsive gateways, etc. However, I would like to know if it is possible to set the fr_timer(), or something similar, on a per branch level. For example, in comparison to our US-based gateways, some of our international gateways may require a higher PDD value. Is it possible to assign a branch-specific timeout value? Thank you, Orlando -- Orlando F. Guitian OrlandoG(a)gmail.com <OrlandoG(a)gmail.com>www.AsteriskCertified.com<http://www.asteriskcertified.com/>

2 1

kamctl ps in 1.5.0 equivallent in 3.1.2
by Renan Capaverde 16 Apr '12

16 Apr '12

Hello list, In kamailio 1.5.0 there is one command called 'ps'. Which when executed produces an output similar to: Process:: ID=0 PID=29533 Type=attendant Process:: ID=1 PID=29537 Type=SIP receiver udp:192.168.166.75:5060 Process:: ID=2 PID=29538 Type=SIP receiver udp:192.168.166.75:5060 Process:: ID=3 PID=29539 Type=SIP receiver udp:192.168.166.75:5060 Process:: ID=4 PID=29540 Type=SIP receiver udp:192.168.166.75:5060 Process:: ID=5 PID=29541 Type=timer Process:: ID=6 PID=29542 Type=timer Process:: ID=7 PID=29543 Type=MI FIFO Process:: ID=8 PID=29544 Type=TCP receiver Process:: ID=9 PID=29545 Type=TCP receiver Process:: ID=10 PID=29546 Type=TCP receiver Process:: ID=11 PID=29547 Type=TCP receiver Process:: ID=12 PID=29548 Type=TCP main In kamailio 3.1.2, I get the following output: *500 command 'ps' not available *Is there any equivalent for *kamctl ps* in version 3.1.2? Regards, -- *Renan Capaverde* /Estagiário Nível Superior/ /DDT - STE - Soluções em Tecnologia e Embarcados/ *DÍGITRO TECNOLOGIA* *E-mail:* renan.capaverde(a)digitro.com.br <mailto:renan.capaverde@digitro.com.br> *Fone:* +55 48 3281-7000*Ramal:* 8132 *Fax:* +55 48 3281-7299 *Site:* www.digitro.com <http://www.digitro.com> /"Antes de imprimir, pense na sua responsabilidade e no seu compromisso com o meio ambiente"/ Esta mensagem, incluindo seus anexos, é reservada somente à Dígitro e ao destinatário da mensagem. Caso você tenha recebido esta mensagem por engano, queira por favor, retorná-la ao remetente e apagá-la de seus arquivos.

2 4

Kamailio and Oracle
by Orlando F. Guitian 16 Apr '12

16 Apr '12

Hi. I am attempting to install Kamailio with the DB_ORACLE module, but I am running into an error. Basically, I am running a CentOS 64-bit and I have also installed the instant client SDK and header files (v10.2) from Oracle. However, the installation is erroring out when trying to find the Oracle XE Client directory (v10.2). However, I am unable to find an Oracle 10g XE Client download in 64 bit version. Does anybody have a work around? Or is there another way to install Kamailio with Oracle? Thank you, Orlando -- Orlando F. Guitian OrlandoG(a)gmail.com <OrlandoG(a)gmail.com>www.AsteriskCertified.com<http://www.asteriskcertified.com/>

2 2

Failed shared memory allocation even though there's enough free memory
by Alex Hermann 16 Apr '12

16 Apr '12

Hello, i'm seeing a lot of failed memory allocations for shared memory after the shared memory has been fully used once (see shmem:max_used_size below). Used_size keeps increasing fast on a heavy loaded server, until it reaches the total_size. From that moment memory allocations start failing and eventually used_size starts dropping. After while, even when there's a lot of free memory, new allocations keep failing. ERROR:tm:build_uac_req: no more share memory (19852) ERROR:tm:t_uac: failed to build message ERROR:presence:send_notify_request: in function tmb.t_request_within ERROR:presence:notify: sending Notify not successful ERROR:presence:publ_notify: Could not send notify for dialog When i request the shmem statistics at such time, there's lots of free memory: shmem:total_size = 268435456 shmem:used_size = 3749696 shmem:real_used_size = 4822288 shmem:max_used_size = 268048584 shmem:free_size = 263613168 shmem:fragments = 62156 This is on 1.5.x, any suggestion on why it keeps failing and how to prevent this error? -- Greetings, Alex Hermann

2 1

Question about processing of packet
by Fabian Borot 15 Apr '12

15 Apr '12

greetings, I have a customer that is sending the REGISTER with 2 ports in the Contact header: Contact: <sip:44435661000448181@201.xxx.xxx.xxx:19778:5060> After some troubleshooting we found that the wireless router is the one inserting the 2nd port. It is not the SIP UA itself. Is it possible to "sanitize" the REGISTER at the beginning of the configuration logic with the TEXTOPS module so that we can leave one port and then process the REGISTERwith the typical: if (!www_authorize("$fd", "subscriber")){...} Or will this only applies to messages that are proxied/forwarded (example: received malformed INVITE, before calling route[1] to send it to the destination sanitize it with the TEXTOPS functions)I guess my question is: after removing the extra port at the begining, when I call the "if (!www_authorize("$fd", "subscriber"))",will it process the modified version or the original one? txs a lot fborot

3 3

Re: [SR-Users] Visualizing SIP Trace
by Norm 15 Apr '12

15 Apr '12

Take a look at Homer. Reda Aouad <reda.aouad(a)gmail.com> wrote: >Hello, > >Anyone knows a tool/software (NOT CDRTool) to graphically visualize the SIP >traces recorded in database by Kamailio's siptrace module? > >Thanks >Reda > >_______________________________________________ >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

2 1

Visualizing SIP Trace
by Reda Aouad 15 Apr '12

15 Apr '12

Hello, Anyone knows a tool/software (NOT CDRTool) to graphically visualize the SIP traces recorded in database by Kamailio's siptrace module? Thanks Reda

1 0

RFC 5626 Outbound/ sip.instance in Contact header?
by Daniel Pocock 15 Apr '12

15 Apr '12

Hi, I'm just wondering if there has been any type of work to implement any part of RFC 5626, particularly the registrar server code? Lumicall now generates a UUID at install time, and sends that as sip.instance in the Contact header. It always sends reg-id=1 (only trying one proxy at the moment). Although this is not a full implementation of the spec, it seems to be sufficient for eliminating extra Contact records in the repro registration server (from reSIProcate). However, I've noticed Kamailio doesn't seem to understand this parameter yet. Regards, Daniel

2 1

Core Dump from select stmt
by Akan Technology 14 Apr '12

14 Apr '12

hello, I am running kamailio 3.2.2 on Solaris 10 64 bit and I am getting a core dump on this statement: $ru = "sip:" + $rU + "@" + $sel(cfg_get.registrar1.server_ip) + $sel(cfg_get.registrar1.server_port); Here is how the variable is defined: registrar1.server_ip = "xxx.xxx.xxx.xxx" desc "Registrar server IP address" registrar1.server_port = "5080" desc "Registrar server Port" Here is the output from the logs with debug: 10(25373) ERROR: *** cfgtrace: c=[/opt/kamailio-3.2/etc/kamailio/kamailio.cfg] l=1072 a=65 n=assign 10(25373) DEBUG: <core> [select.c:424]: Calling SELECT 1004b2bc0 10(25373) DEBUG: <core> [select.c:424]: Calling SELECT 1004b4da0 10(25373) ERROR: *** cfgtrace:14(25377) : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 22 14(25377) DEBUG: <core> [tcp_main.c:3555]: DBG: handle_ser_child: dead child 10, pid 25373 (shutting down?) 14(25377) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (1003738b8, 22, 0, 0x0) fd_no=18 called 14(25377) DEBUG: <core> [tcp_main.c:3316]: DBG: handle_tcp_child: dead tcp child 0 (pid 25373, no 10) (shutting down?) 14(25377) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (1003738b8, 24, 1, 0x0) fd_no=17 called 0(25363) ALERT: <core> [main.c:751]: child process 25373 exited by a signal 11 0(25363) ALERT: <core> [main.c:754]: core was generated 0(25363) INFO: <core> [main.c:766]: INFO: terminating due to SIGCHLD 14(25377) INFO: <core> [main.c:817]: INFO: signal 15 received Here is the full route code: route[TO_REGISTRAR] { # check if Registrar server's IP is defined if (strempty($sel(cfg_get.registrar1.server_ip))) { xlog("SCRIPT: REGISTRAR - registrar1.server_ip not defined\n"); return; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.registrar1.server_ip) + ":" + $sel(cfg_get.registrar1.server_port); route(RELAY); exit; } # End of TO_Registrar Route I am trying to setup the proxy server to route all registers requests to another kamailio server Thanks Nathaniel

1 0

200 OK with SDP got wrong Contact HF after fix_nated_contact()
by Karsten Horsmann 13 Apr '12

13 Apr '12

Hello! I got the following problem. My 200 OK with SDP sends the private ip of the B-Leg party. Kamailio is multihomed (pulic/private address) in front of an ivr system. 222.222.222.222 public ip kamailio 172.20.100.74 private ip kamailio 172.20.100.61 private ip ivr 217.217.217.217 public ip uac 172.20.100.203 private ip uac U 2012/04/13 13:56:57.307314 222.222.222.222:5060 -> 217.217.217.217:42885 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 172.20.100.103:28472;rport=42885;branch=z9hG4bK-d8754z-824b81ca678a7de4-1---d8754z-;received=217.217.217.217. From: <sip:4horsmann@222.222.222.222>;tag=e9d1124c. To: <sip:kalkbrenner@222.222.222.222>;tag=ds-1c78ac-2ca01e29. Call-ID: NjQ5YWU0NGYxZjM3YWJhYmE5ZWYxYmQzZjYzYzQyMzI.. CSeq: 2 INVITE. Content-Length: 202. Content-Type: application/sdp. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=e9d1124c;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=e9d1124c;nat=yes>. Supported: replaces. Supported: 100rel. Allow: INVITE. Allow: ACK. Allow: BYE. Allow: CANCEL. Allow: OPTIONS. Allow: NOTIFY. Allow: REFER. Allow: PRACK. Allow: INFO. Allow: UPDATE. Allow: MESSAGE. Contact: <sip:172.20.100.61:5060>. P-hint: NATMANAGE WITH SDP. . v=0. o=aculab-04246C01 2060618394 2060618394 IN IP4 212.59.42.189. s=-. c=IN IP4 222.222.222.222. t=0 0. m=audio 40390 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=nortpproxy:yes. ############################################################## # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT xlog("NATMANAGE M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(has_body("application/sdp")) { xlog("NATMANAGE M=$rm with SDP\n"); append_hf("P-hint: NATMANAGE WITH SDP\r\n"); fix_nated_contact(); } if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (isbflagset(FLB_NATB)) return; if (dst_ip == 212.59.42.189) { xlog("rtpproxy_manage extern zu intern\n"); #rtpproxy_manage("OCFEI","172.20.10.74"); rtpproxy_manage("FWCEI","172.20.10.74"); #xlog("e->i Body vor fix: $rb\n"); #fix_nated_contact(); #xlog("e->i Body nach fix: $rb\n"); } else { xlog("rtpproxy_manage intern zu extern\n"); #rtpproxy_manage("OCFIE","222.222.222.222"); rtpproxy_manage("FWCIE","222.222.222.222"); xlog("i->e Body vor fix: $rb\n"); fix_nated_contact(); xlog("i->e Body nach fix: $rb\n"); } if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } -- Mit freundlichen Grüßen *Karsten Horsmann*

1 0

200 OK with SDP gets private ip in contact field
by Karsten Horsmann 13 Apr '12

13 Apr '12

Hello! I get step by step to my multihomed setup and have now problems that kamailio didnt rewrite the 200 OK to my public ip address. I have attached an ngrep trace and my kamailio 3.2.2 config. Would be great if someone can give me an hint! 222.222.222.222 public ip-address from kamailio 172.20.100.74 private ip-address from kamailio 217.777.777.777 public ip-address from the UAC 172.20.100.103 private ip-address from the UAC 172.20.100.61 sip ip-address from the IVR behind kamailio 172.20.100.71 rtp ip-address from the IVR behind kamailio -- Mit freundlichen Grüßen *Karsten Horsmann*

2 2

LCR routing and From Uri
by Alexey Mechanoshin 13 Apr '12

13 Apr '12

Hello all. In my config added LCR routing by this manual: http://www.amitnepal.com/using-lcr-with-kamailio/ and if added, to 'lcr_rule' table, uri to 'from_uri' column - kamailio always reply 503 - 'No More Gateways' :-( If remove data - routing work. A'm using last kamailio from git ... Please help me change my config to working routing by checking 'from_uri' . Need check routing prefix 5499 only from IP 192.xxx.xxx.59 , as example. In Log: Apr 10 17:01:23 r59vredfs /usr/sbin/kamailio[29582]: INFO: <script>: New request - M=INVITE RURI=sip:5499@192.xxx.xxx.50:5070 F=sip:5491@192.xxx.xxx.59 T=sip:5499@192.xxx.xxx.50:5070 IP=192.xxx.xxx.59 ID=6c518b61-fd9f-122f-91b7-002655e85348 Apr 10 17:01:23 r59vredfs /usr/sbin/kamailio[29582]: INFO: <script>: Entering route[LCR] for method: INVITE <sip:5499@192.xxx.xxx.50:5070>, and rU=5499, caller_uri='0' Apr 10 17:01:23 r59vredfs /usr/sbin/kamailio[29582]: INFO: <script>: No More Gateways … Apr 10 17:01:23 r59vredfs /usr/sbin/kamailio[29582]: INFO: <script>: Send back 503 code … MySQL data: mysql> select * from lcr_rule; +----+--------+--------+----------+---------+---------+ | id | lcr_id | prefix | from_uri | stopper | enabled | +----+--------+--------+----------+---------+---------+ | 1 | 1 | 5498 | | 0 | 1 | | 2 | 1 | 57 | NULL | 0 | 1 | | 3 | 1 | 58 | NULL | 0 | 1 | | 4 | 1 | 541 | NULL | 0 | 1 | | 5 | 1 | 5499 | sip:192.xxx.xxx.59 | 1 | 1 | +----+--------+--------+----------+---------+---------+ 5 rows in set (0.00 sec) mysql> select * from lcr_gw; +----+--------+---------+----------------+----------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | prefix | tag | flags | defunct | +----+--------+---------+----------------+----------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | 1 | 1 | route1 | 213.xxx.xxx.26 | NULL | 5060 | NULL | 1 | 1 | NULL | | NULL | 1 | NULL | | 2 | 1 | routemy | 192.xxx.xxx.59 | NULL | 5070 | NULL | 1 | 1 | NULL | NULL | NULL | 1 | NULL | | 3 | 1 | route1 | 213.xxx.xxx.27 | NULL | 5060 | NULL | 1 | 1 | NULL | NULL | NULL | 1 | NULL | +----+--------+---------+----------------+----------+------+--------+------------+-----------+-------+--------+------+-------+---------+ 3 rows in set (0.00 sec) mysql> select * from lcr_rule_target; +----+--------+---------+-------+----------+--------+ | id | lcr_id | rule_id | gw_id | priority | weight | +----+--------+---------+-------+----------+--------+ | 1 | 1 | 1 | 1 | 10 | 1 | | 2 | 1 | 2 | 2 | 10 | 1 | | 3 | 1 | 3 | 2 | 10 | 1 | | 4 | 1 | 4 | 2 | 10 | 1 | | 5 | 1 | 5 | 1 | 10 | 1 | | 6 | 1 | 5 | 3 | 20 | 1 | +----+--------+---------+-------+----------+--------+ 6 rows in set (0.00 sec) Config: #!define WITH_LCR ... #!ifdef WITH_LCR loadmodule "avpops.so" loadmodule "lcr.so" #!endif ... #!ifdef WITH_LCR modparam("lcr", "db_url",DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "lcr_id_avp", "$avp(s:lcr_id_avp)") #!endif ... # dispatch destinations to PSTN route(PSTN); # LCR route route(LCR); # user location service #route(LOCATION); ... # --=== LCR routing ===-- # route[LCR] { xlog("L_INFO","Entering route[LCR] for method: $rm <$ru>, and rU=$rU, caller_uri='$var(caller_uri)'\n"); #!ifdef WITH_LCR ### if (!load_gws(1)) { ### if (!load_gws(1, $rU)) { if (!load_gws(1)) { sl_send_reply("500", "Server Internal Error - Cannot load gateways"); exit; } else { $var(i) = 0; while(is_avp_set("$(avp(i:709)[$var(i)])")) { xlog("L_INFO", "loading gw_uri_avp[$var(i)]='$(avp(i:709)[$var(i)])'\n"); $var(i) = $var(i) + 1; } if(is_avp_set("$avp(i:709)")) { xlog("L_INFO", "Trying gateway '$avp(i:709)'\n"); } else { xlog("L_INFO", "No More Gateways …\n"); } # try the first matched gateway if (next_gw()) { xlog("L_INFO", "ruri_user_avp='$avp(i:500)'\n"); # Print AVP - for debug #avp_print(); # Route to failure for failover route(RELAY); } else { avp_print(); xlog("L_INFO", "Send back 503 code …\n"); sl_send_reply("503", "No available gateways"); exit; } } #!endif #sl_send_reply("503", "Service Unavailable"); #exit; return; } PS Sorry for my bad English please. -- "В связи с отсутствием интереса общественности, конец света отменяется" (с) ? С Уважением, Механошин Алексей

2 8

rtpproxy 1.2.1 segfaults with kamailio 3.2.2 in bridging mode
by Karsten Horsmann 12 Apr '12

12 Apr '12

Hi all, i try to register at the rtpproxy mailinglist, but its still pending. Maybe someone here had the same issue. In my multihomed setup (with mhomed=1 in kamailio.cfg) i had segfaults from rtpproxy if i called rtpproxy_manage with params. Notice that i must run rtpproxy as root to get core files. Any hints? Maybe my kamailio config isnt right? ulimit -c unlimited ; /usr/local/bin/rtpproxy -u root -s /home/sys/bin/rtpproxy.sock -l <PUBLIC-IP> 172.20.100.74 -d DBUG:LOG_LOCAL5 Apr 10 15:13:40 kamailio1 rtpproxy[19728]: DBUG:handle_command: received command "V" Apr 10 15:13:40 kamailio1 rtpproxy[19728]: DBUG:doreply: sending reply "20040107#012" Apr 10 15:13:40 kamailio1 rtpproxy[19728]: DBUG:handle_command: received command "VF 20050322" Apr 10 15:13:40 kamailio1 rtpproxy[19728]: DBUG:doreply: sending reply "1#012" Apr 10 15:13:40 kamailio1 rtpproxy[19728]: DBUG:handle_command: received command "VF 20071116" Apr 10 15:13:40 kamailio1 rtpproxy[19728]: DBUG:doreply: sending reply "1#012" Apr 10 15:13:40 kamailio1 rtpproxy[19728]: DBUG:handle_command: received command "VF 20081102" Apr 10 15:13:40 kamailio1 rtpproxy[19728]: DBUG:doreply: sending reply "1#012" Apr 10 15:13:40 kamailio1 rtpproxy[19728]: DBUG:handle_command: received command "UEIc107,0,8,101 NTgxNzRkMmJmYzFlYjg2NTdiN2Q3MzJkYjkzMzI3Yjk. 217.XXX.XXX.XX 49814 00671612;1" Apr 10 15:13:40 kamailio1 rtpproxy[19728]: INFO:handle_command: new session NTgxNzRkMmJmYzFlYjg2NTdiN2Q3MzJkYjkzMzI3Yjk., tag 00671612;1 requested, type strong Apr 10 15:13:40 kamailio1 rtpproxy[19728]: INFO:handle_command: new session on a port 36156 created, tag 00671612;1 Apr 10 15:13:40 kamailio1 rtpproxy[19728]: INFO:handle_command: pre-filling caller's address with 217.XXX.XX.XX:49814 Apr 10 15:13:40 kamailio1 rtpproxy[19728]: DBUG:doreply: sending reply "36156 212.XX.XX.XXX#012" (gdb) bt #0 0x0804ccb8 in ishostseq (ia1=0x99c54c8, ia2=0x0) at rtpp_util.c:55 #1 0x080514ba in handle_command (cf=0xbffb9d8c, controlfd=5, dtime=1334063620.8220949) at rtpp_command.c:761 #2 0x0804afb7 in process_commands (argc=9, argv=0xbffda634) at main.c:752 #3 main (argc=9, argv=0xbffda634) at main.c:856 (gdb) p ia1->sa_family $1 = 2 (gdb) p ia2->sa_family Cannot access memory at address 0x0 # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT xlog("NATMANAGE M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (isbflagset(FLB_NATB)) return; if (dst_ip == PUBLIC_IP ) { xlog("rtpproxy_manage extern zu intern\n"); rtpproxy_manage("OCFEI","172.20.10.74"); } else { xlog("rtpproxy_manage intern zu extern\n"); rtpproxy_manage("OCFIE","PUBLIC_IP"); } if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } -- Mit freundlichen Grüßen *Karsten Horsmann*

3 8

Kamailio Failed to have audo converstaion with RTP
by Ryan Gholam 12 Apr '12

12 Apr '12

Dear Daniel , I thank you for your reply , I have a server having the Astersisk ip address (192.168.10.15) , and rtp + kamailio is installed on an another pc have the following ip (192.168.10.17) which is linked to the Astersik , and on the same pc , another network card exists having the ip address (192.168.20.3 ) which is linked to a client pc having the ip address ( 192.168.20.4) . I tracked the call and i can see SIP ACK nd BYE between 20.3 and 20.4 but there is no audio conversation this is my configuration file for kamailio attached above . P.S : testing without NATING as described in the above setup . I thank you alot again for all your help .

2 1

Variable $fu Kamailio 3.2
by Gilbert T. Gutierrez, Jr. 11 Apr '12

11 Apr '12

I wrote the code below to rewrite an extension to a phone number (It is called in route[LOCATION]). This code works fine with MULTIDOMAIN enabled but when I run it as a single domain the line $fu=$var(b); does not seem to work.. You can see that I have several xlog lines outputting the variable values into my /var/log/messages. $var(b) has the correct value in it, so I do not know what is happening. I have also included the /var/log/messages output which I redacted the phone number, domain, and extension for security purposes. I am sure I am going about this backwards, but can someone provide me some guidance as to what is going wrong? Thank you, Gilbert # Rewrite - Gilbert route[REWRITE] { # This section rewrites the outbound calling number so that caller id works correctly. #!ifdef WITH_REWRITE # lookup an outbound number to replace the extension with $var(b)="NO REV"; sql_xquery("ca","select number from pioutalias where username='$fU'","ra"); # determine if a outbound number exists if ($xavp(ra=>number)) { $var(b)="sip:" + $xavp(ra=>number) + "@" + $fd; xlog("L_INFO","var(b): '$var(b)'"); xlog("L_INFO","fu: '$fu'"); # Assign the outbound calling number $fu=$var(b); xlog("L_INFO","New fu: '$fu'"); } sql_result_free("ra"); # see if it found a number and log if it did not if ($var(b)=="NO REV") xlog("L_INFO","No number found for extension: '$fu'"); #!endif } /var/log/messages output Apr 9 14:38:38 tempfax /usr/sbin/kamailio[6088]: INFO: <script>: var(b): 'sip:602XXXXXXX@xxxx.com' Apr 9 14:38:38 tempfax /usr/sbin/kamailio[6088]: INFO: <script>: fu: 'sip:30XXXXX@xxxx.com' Apr 9 14:38:38 tempfax /usr/sbin/kamailio[6088]: INFO: <script>: New fu: 'sip:30XXXXX@xxxx.com'

2 1

dropped calls after 38 seconds openSER
by Saul Waizer 11 Apr '12

11 Apr '12

Greetings list, I am experiencing a strange behavior with openser 1.3.2 running on ubuntu 10. I have a basic configuration (see bellow) and i am using Linphone for iPad as my client. I have 2 users registered and I am able to place calls no problem. The problem is that the calls (audio or A/V) drop after 38 seconds exactly, this behavior is pretty consistent, 38 seconds is all I can get. There is no firewall in front of the clients. Here is my configuration, ip addresses changed to protect the innocent: http://pastie.org/private/x1ck8rxjcxv6hl44hrmqg You can see the logs of the call here (the majority): http://pastie.org/private/4fj5efpbsrxan8plzqvfza Am I missing something or is there anything that needs to be changed in the routing/configuration to achieve basic functionality? Thank you in advance!

2 1

Re: [SR-Users] Variable $fu Kamailio 3.2
by Alex Balashov 10 Apr '12

10 Apr '12

$fu is not mutable. -- 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 "Gilbert T. Gutierrez, Jr." <mailing-lists(a)phoenixinternet.net> wrote: >I wrote the code below to rewrite an extension to a phone number (It is >called in route[LOCATION]). This code works fine with MULTIDOMAIN >enabled but when I run it as a single domain the line $fu=$var(b); does >not seem to work.. You can see that I have several xlog lines >outputting the variable values into my /var/log/messages. $var(b) has >the correct value in it, so I do not know what is happening. I have also >included the /var/log/messages output which I redacted the phone number, >domain, and extension for security purposes. I am sure I am going about >this backwards, but can someone provide me some guidance as to what is >going wrong? > >Thank you, >Gilbert > ># Rewrite - Gilbert >route[REWRITE] { ># This section rewrites the outbound calling number so that caller id >works correctly. >#!ifdef WITH_REWRITE > # lookup an outbound number to replace the extension with > $var(b)="NO REV"; > sql_xquery("ca","select number from pioutalias where >username='$fU'","ra"); > # determine if a outbound number exists > if ($xavp(ra=>number)) { > $var(b)="sip:" + $xavp(ra=>number) + "@" + $fd; > xlog("L_INFO","var(b): '$var(b)'"); > xlog("L_INFO","fu: '$fu'"); > # Assign the outbound calling number > $fu=$var(b); > xlog("L_INFO","New fu: '$fu'"); > } > sql_result_free("ra"); > # see if it found a number and log if it did not > if ($var(b)=="NO REV") > xlog("L_INFO","No number found for extension: '$fu'"); >#!endif >} > > >/var/log/messages output > >Apr 9 14:38:38 tempfax /usr/sbin/kamailio[6088]: INFO: <script>: >var(b): 'sip:602XXXXXXX@xxxx.com' >Apr 9 14:38:38 tempfax /usr/sbin/kamailio[6088]: INFO: <script>: fu: >'sip:30XXXXX@xxxx.com' >Apr 9 14:38:38 tempfax /usr/sbin/kamailio[6088]: INFO: <script>: New >fu: 'sip:30XXXXX@xxxx.com' > > > >_______________________________________________ >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

3 5

[OT] sip:provider CE v2.5 is out!
by Andreas Granig 10 Apr '12

10 Apr '12

Hi all, I'm excited to announce the release of the Sipwise sip:provider Community Edition v2.5, a free and open source turn-key platform, which uses Kamailio, Sems and Asterisk in its core to provide a full-blown and feature-rich VoIP soft-switch. http://www.sipwise.com/news/announcements/spce-v2_5-release/ New core features are IPv6 and video support, serial call hunting and time based routing, beside lots of other fixes and enhancements. Check our VM images to get started within a couple of minutes and learn what can be built with the power of Kamailio as a stateless load-balancer and a stateful proxy/registrar, and Sems as an SBC and application server. Thanks a lot to all developers, contributors and members of this mailing list to make this possible and for helping us out! Andreas

1 0

ClueCon 2012: Call For Speakers
by Michael Collins 10 Apr '12

10 Apr '12

[image: Inline image 2] ClueCon 2012 - Call For Speakers ClueCon - the open source telephony conference by developers, for developers - would like to announce that we are having an open call for speaking proposals for this year's event. If you have an idea for a technical presentation for ClueCon 2012 then we would like to hear about. What makes a great ClueCon presentation? The tech savvy crowd that attends ClueCon loves *technical* presentations. In general, the more technical the presentation, the better. If you are thinking about a presentation then consider these points: - ClueCon talks are 30 minutes in length, including Q&A time with the audience - ClueCon has a special focus on open source VoIP and telephony projects like FreeSWITCH, Asterisk, OpenSIPS, and Kamailio - Attendees enjoy hearing about projects built with open source tools - Highly technical discussions that show the nuts and bolts are especially well-liked - The audience appreciates seeing and participating in live demonstrations Please send your proposals to marketing(a)cluecon.com. Be sure to include a working title, description of the talk, and name of the presenter. Don't delay! There are a limited number of openings. ClueCon 2012 Registration Information ClueCon 2012 registration is now open. Visit the registration page<http://www.cluecon.com/register?cc12cfs>for details. Be sure to book your room at the Wyndham <http://www.cluecon.com/hotels?cc12cfs> and qualify for the $300 discount. As always, feel free to call us at 877.742.CLUE (877.742.2583) if you have any questions about ClueCon 2012. Also, keep in mind that the FreeSWITCH community has a conference call<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12cfs>each Wednesday at 1PM Eastern time. This is a great opportunity to talk about open source telephony and get to know a number folks who will be at ClueCon 2012. Stay tuned for more news about ClueCon speakers, sponsors, and related events! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12cfs2

1 0

Can rtpproxy be used withou NAT ?
by Rabary 10 Apr '12

10 Apr '12

Hi mailing, I installed kamailio 3.2, rtpproxy 1.2.1, callcontrol 2.0.15 on a vz, and cdrtool 8.2.5 and freeradius 2.1.10 on another vz. The 2 vz container have public ip address, and the UAC have private ip address. I want to use rtpproxy, and the following are what ps and netstat command returns about rtpproxy: teddy@kamailio:~$ ps aux | grep rtpproxy teddy 22866 0.0 0.0 3312 800 pts/0 S+ 11:08 0:00 grep rtpproxy teddy 31326 0.0 0.0 11360 804 ? Ssl Apr06 0:02 /usr/sbin/rtpproxy -F -l our_public_ip -s udp:localhost 22222 teddy@kamailio:~$ sudo netstat -pln | grep rtp udp 0 0 127.0.0.1:22222 0.0.0.0:* 31326/rtpproxy And my problem is: when I make calls, for example the UAC1 calls UAC2, with ngrep I see the UAC1 calls UAC1 and not UAC2, and I don't know why. In kamailio config file, there are directive WITH_NAT for everything related to rtpproxy (loadmodule, routing logic, etc) as you can see in the attached conf file. I try disable the use of this directive WITH_NAT so I disable the use of rtpproxy in kamailio and it works: when UAC1 calls UAC2, the UAC2 is ringing. Please tell me what am I doing wrong, or to use rtpproxy without NAT is not possible ? In attached file the kamailio config file, the diff between rtpproxy enable and rtpproxy disable, and the result of ngrep and what is in syslog when I made calls. Thanks in advance. -- Rabary Teddy Inutile d'imprimer ce mail

2 2

ACK to 200 OK discarded by loose_route()
by SIP Mailing-list 10 Apr '12

10 Apr '12

Hi guys, Hope someone can help me with this. The background: I'm just trying to implement a simple load balancer with one source and multiple destinations using the call load balancer (ALG10) in dispatcher. kamailio version 3.2.0 The symptoms: I have a simple script working in one test environment (csipsimple -> kamailaio -> Twinkle) but can't get it to work in another (Dialogic HMP 3.0 -> kamailio -> Dialogic HMP 3.0) The details: I've done a debug trace (below) in both environments and can't see the problem. For some reason the ACK is destroyed before it is forwarded even though the log says "[forward.c:601]: Sending:#012ACK..." Any help would be much appreciated! TIA, Richard kamailio.cfg: ... #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # RAC: Uses dispatcher algorithm 10 to keep track of load distribution # # Use: kamctl start/stop/restart ####### Global Parameters ######### # Output goes to /var/log/syslog # 5 gives too much output # 3 is DEBUG, even this can go crazy # 2 is INFO # 1 is NOTICE debug=3 #log_stderror=yes memdbg=5 memlog=5 mem_summary=2 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ alias=10.130.11.121:5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:10.130.11.121:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ #port=5062 # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif # For the kamctl interface loadmodule "mi_fifo.so" #loadmodule "kex.so" # TM module is required for t_relay() loadmodule "tm.so" #loadmodule "tmx.so" loadmodule "sl.so" # RR module is required for loose_route() and record_route() loadmodule "rr.so" # PV is required for avp's for ALG10 loadmodule "pv.so" # Used for mf_process_maxfwd_header() loadmodule "maxfwd.so" #loadmodule "usrloc.so" #loadmodule "registrar.so" # textops is required for is_method() loadmodule "textops.so" # For has_totag() loadmodule "siputils.so" # xlog is required for user xlog() formatted messages. Output goes to /var/log/messages loadmodule "xlog.so" # For sanity_check() loadmodule "sanity.so" #loadmodule "ctl.so" #loadmodule "cfg_rpc.so" #loadmodule "mi_rpc.so" #loadmodule "acc.so" # Added by RAC 2012-04-04: required for ds_select_dst() etc loadmodule "dispatcher.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # Added by RAC 2012-04-04 # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 0) modparam("dispatcher", "use_default", 1) modparam("dispatcher", "ds_ping_method", "OPTIONS") # Only set ping reply codes that are acceptable. CE should only reply "200OK" if OK, "486 Busy Here" if not modparam("dispatcher", "ds_ping_reply_codes", "class=2") modparam("dispatcher", "ds_ping_from", "sip:proxy@10.130.11.121") modparam("dispatcher", "ds_ping_interval", 20) # In this mode only inactive destinations marked as probe-able will be pinged modparam("dispatcher", "ds_probing_mode", 2) modparam("dispatcher", "ds_probing_threshhold", 1) modparam("dispatcher", "ds_hash_expire", 3600) # Flag "2" MUST be set to use the AVPs listed below. If not set these will be <null> modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") # For algorithm 10, call load distribution duid (in list_file), dstid_avp and ds_hash_size are required modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") # Appears to be the size (2^ds_hash_size) of a hashtable for storing the state of individual sessions # Presumably needs to be bigger than the number of concurrent sessions expected # 2^12 = 4096 modparam("dispatcher", "ds_hash_size", 12) ####### Routing Logic ######## # ds_load_update(): # if it is a BYE or CANCEL - remove the load from destination address used to forward the INVITE # if it is a reply to INVITE - set internal state to confirmed for call load structure when reply code is 2xx. # # ds_load_unset(): Remove the call load for the destination that routed the call. # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } # Added by RAC 2012-04-04 # Do not run ds_load_update() here; the reply to the orginal invite is sufficient to release the load # loose_route() performs routing of in-dialog requests if(has_totag()) { if (loose_route()) { # t_relay() relays a message statefully; forward() (stateless) easily gets in a muddle if(!t_relay()) { if (method == "ACK") xlog("L_ERR", "cid:$ci [$rm] [200-OK] t_relay failed\n"); else xlog("L_WARN", "cid:$ci Existing Transaction relay failed: [$rm] from($ft) [$fu] to($tt) [$tu]\n"); sl_reply_error(); } else xlog("L_INFO", "cid:$ci Existing Transaction: [$rm] from($ft) [$fu] to($tt) [$tu]\n"); } else if (method == "ACK") { if (t_check_trans()) { # stateful ACK after 487 or 404 t_relay(); } else { # ACK without transaction, ignore and discard xlog("L_WARN", "cid:$ci Dropping ACK: from($ft) [$fu] to($tt) [$tu]\n"); } } else { sl_send_reply("404","Not here"); } exit; } # has_totag() # dest set1, ALG10: call load distribution: least number of calls receives the next call # Keeps a hash table of calls based on Call-ID to track call load per destination if(ds_select_dst("1", "10")) xlog("L_NOTICE", "cid:$ci Destination Selected: [$rm] from($ft) [$fu] routed to [sip:$tU@$dd:$dp] ($avp(dsdstid))\n"); else { xlog("L_ERR", "No destinations available: [$rm] from [$fu]\n"); sl_send_reply("500","No destinations available"); drop(); exit(); } # record_route() causes ALL messages to go through the proxy including ACKs and BYEs. Results in 16 SIP packets # Without record_route(), after 200OK, the ACK and BYE is sent direct. Results in 11 SIP packets # Must use record_route() with call load distribution, otherwise we will not know when a call is released # and cannot account for it record_route(); # With t_relay(), the proxy creates its own messages like 100 trying, instead of forwarding the original # forward() is not recommended t_relay(); } event_route[dispatcher:dst-down] { xlog("L_ERR", "Destination down: [$rm] $ru ($du)\n"); } event_route[dispatcher:dst-up] { xlog("L_NOTICE", "Destination up: [$rm] $ru\n"); } # Added by RAC 2012-04-04: Update call loads on replies for ALG10. # Default action is to reply upstream statelessly using Via headers onreply_route { xlog("L_INFO", "cid:$ci Reply: [$rm] [$rs-$rr] from($ft) [$fu] to($tt) [$tu]\n"); if(is_method("INVITE")) { if(status=~"2[0-9][0-9]") ds_load_update(); else if(status=~"[3-7][0-9][0-9]") ds_load_unset(); } else if(is_method("BYE")) { ds_load_update(); } } ... Bad debug trace (CE to IVRSim): NOTE: Phone numbers have been changed ... Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/msg_parser.c:628]: SIP Request: Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/msg_parser.c:630]: method: <ACK> Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/msg_parser.c:632]: uri: <sip:18000005835@10.130.11.121> Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/msg_parser.c:634]: version: <SIP/2.0> Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=f8a6698-0-13c4-50022-16d0d4-103b30a3-16d0d4 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_field: <To> [81]; uri=[ sip:18000005835@10.130.11.121] Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/msg_parser.c:189]: DEBUG: to body [<sip:18000005835@10.130.11.121>] Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <1> <ACK> Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/parse_via.c:1288]: Found param type 232, <branch> = <z9hG4bK-182890-5e5e75ec-235d6d1e>; state=16 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/parse_via.c:2563]: end of header reached, state=5 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [receive.c:146]: After parse_msg... Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [receive.c:187]: preparing to run routing scripts... Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: sl [sl_funcs.c:307]: DEBUG : sl_filter_ACK: to late to be a local ACK! Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=774ee38-650b820a-13c4-50022-182890-4de156df-182890 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: sanity [mod_sanity.c:251]: sanity checks result: 1 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: siputils [checks.c:107]: totag found Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: rr [loose.c:85]: is_preloaded: No Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [socket_info.c:502]: grep_sock_info - checking if host==us: 13==13 && [10.130.11.121] == [10.130.11.121] Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [socket_info.c:505]: grep_sock_info - checking if port 5060 matches port 5060 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: rr [loose.c:594]: Next hop: 'sip:10.130.11.121;lr;ftag=774ee38-650b820a-13c4-50022-182890-4de156df-182890' is loose router Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: rr [loose.c:644]: The last route URI: 'sip:10.130.11.121;lr;ftag=774ee38-650b820a-13c4-50022-182890-4de156df-182890' Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=6 , global msg id=5 , T on entrance=(nil) Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: tm [t_lookup.c:527]: t_lookup_request: start searching: hash=37904, isACK=1 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: tm [t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: tm [t_funcs.c:315]: SER: forwarding ACK statelessly Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [msg_translator.c:204]: check_via_address(10.130.11.101, 10.130.11.101, 0) Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [forward.c:601]: Sending:#012ACK sip:10.130.11.121;lr;ftag=774ee38-650b820a-13c4-50022-182890-4de156df-182890 SIP/2.0#015#012From: <sip:441865000101@10.130.11.101>;tag=774ee38-650b820a-13c4-50022-182890-4de156df-182890#015#012To: <sip:18000005835@10.130.11.121>;tag=f8a6698-0-13c4-50022-16d0d4-103b30a3-16d0d4#015#012Call-ID: fb041312-da6d-4fb7-bfee-32677fbab37b#015#012CSeq: 1 ACK#015#012Via: SIP/2.0/UDP 10.130.11.121;branch=z9hG4bKcydzigwkX#015#012Via: SIP/2.0/UDP 10.130.11.101:5060;branch=z9hG4bK-182890-5e5e75ec-235d6d1e#015#012Max-Forwards: 69#015#012Contact: <sip:441865000101@10.130.11.101>#015#012Allow-Events: refer#015#012Content-Length: 0#015#012#015#012. Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [forward.c:603]: orig. len=550, new_len=566, proto=1 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/msg_parser.c:628]: SIP Request: Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/msg_parser.c:630]: method: <ACK> Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/msg_parser.c:632]: uri: <sip:10.130.11.121;lr;ftag=774ee38-650b820a-13c4-50022-182890-4de156df-182890> Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/msg_parser.c:634]: version: <SIP/2.0> Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=f8a6698-0-13c4-50022-16d0d4-103b30a3-16d0d4 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_field: <To> [81]; uri=[ sip:18000005835@10.130.11.121] Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28277]: DEBUG: <core> [receive.c:290]: receive_msg: cleaning up Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/msg_parser.c:189]: DEBUG: to body [<sip:18000005835@10.130.11.121>] Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <1> <ACK> Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/parse_via.c:1288]: Found param type 232, <branch> = <z9hG4bKcydzigwkX>; state=16 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/parse_via.c:2563]: end of header reached, state=5 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [receive.c:146]: After parse_msg... Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [receive.c:187]: preparing to run routing scripts... Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: sl [sl_funcs.c:307]: DEBUG : sl_filter_ACK: to late to be a local ACK! Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/parse_via.c:1288]: Found param type 232, <branch> = <z9hG4bK-182890-5e5e75ec-235d6d1e>; state=16 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/parse_via.c:2563]: end of header reached, state=5 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=100 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/msg_parser.c:528]: parse_headers: this is the second via Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: maxfwd [mf_funcs.c:85]: value = 69 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=774ee38-650b820a-13c4-50022-182890-4de156df-182890 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: sanity [mod_sanity.c:251]: sanity checks result: 1 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: siputils [checks.c:107]: totag found Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: rr [loose.c:108]: No Route headers found Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: rr [loose.c:838]: There is no Route HF Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=6 global id=5 T start=(nil) Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: tm [t_lookup.c:527]: t_lookup_request: start searching: hash=37904, isACK=1 Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: tm [t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: tm [t_lookup.c:1148]: DEBUG: t_check_msg: msg id=6 global id=6 T end=(nil) Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: WARNING: <script>: cid:fb041312-da6d-4fb7-bfee-32677fbab37b Dropping ACK: from(774ee38-650b820a-13c4-50022-182890-4de156df-182890) [ sip:441865000101@10.130.11.101] to(f8a6698-0-13c4-50022-16d0d4-103b30a3-16d0d4) [ sip:18000005835@10.130.11.121] Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: last message repeated 5 times Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Apr 6 14:10:29 UKCY-VM-LINUX199 /usr/sbin/kamailio[28276]: DEBUG: <core> [receive.c:290]: receive_msg: cleaning up ... Bad SIP ACK: ... Session Initiation Protocol Request-Line: ACK sip:18000005835@10.130.11.121 SIP/2.0 Method: ACK Request-URI: sip:18000005835@10.130.11.121 Request-URI User Part: 18000005835 Request-URI Host Part: 10.130.11.121 [Resent Packet: False] [Request Frame: 7] [Response Time (ms): 29] Message Header From: <sip:441865000101@10.130.11.101 >;tag=774ee38-650b820a-13c4-50022-182890-4de156df-182890 SIP from address: sip:441865000101@10.130.11.101 SIP from address User Part: 441865000101 SIP from address Host Part: 10.130.11.101 SIP tag: 774ee38-650b820a-13c4-50022-182890-4de156df-182890 To: <sip:18000005835@10.130.11.121 >;tag=f8a6698-0-13c4-50022-16d0d4-103b30a3-16d0d4 SIP to address: sip:18000005835@10.130.11.121 SIP to address User Part: 18000005835 SIP to address Host Part: 10.130.11.121 SIP tag: f8a6698-0-13c4-50022-16d0d4-103b30a3-16d0d4 Call-ID: fb041312-da6d-4fb7-bfee-32677fbab37b CSeq: 1 ACK Sequence Number: 1 Method: ACK Via: SIP/2.0/UDP 10.130.11.101:5060 ;branch=z9hG4bK-182890-5e5e75ec-235d6d1e Transport: UDP Sent-by Address: 10.130.11.101 Sent-by port: 5060 Branch: z9hG4bK-182890-5e5e75ec-235d6d1e Max-Forwards: 70 Route: <sip:10.130.11.121;lr;ftag=774ee38-650b820a-13c4-50022-182890-4de156df-182890> Contact: <sip:441865000101@10.130.11.101> Contact Binding: <sip:441865000101@10.130.11.101> URI: <sip:441865000101@10.130.11.101> SIP contact address: sip:441865000101@10.130.11.101 Allow-Events: refer Content-Length: 0 ... Good debug trace (Richard laptop, Twinkle local and csipsimple on wifi): ... Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/msg_parser.c:628]: SIP Request: Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/msg_parser.c:630]: method: <ACK> Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/msg_parser.c:632]: uri: <sip:123@192.168.0.104;transport=udp> Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/msg_parser.c:634]: version: <SIP/2.0> Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/parse_via.c:1288]: Found param type 235, <rport> = <n/a>; state=6 Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/parse_via.c:1288]: Found param type 232, <branch> = <z9hG4bKPjWvIFG5BzVRtUe1jnpRa1ilioYiezOrTz>; state=16 Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/parse_via.c:2563]: end of header reached, state=5 Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [receive.c:146]: After parse_msg... Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [receive.c:187]: preparing to run routing scripts... Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: sl [sl_funcs.c:307]: DEBUG : sl_filter_ACK: to late to be a local ACK! Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=ggpxl Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_field: <To> [35]; uri=[ sip:123@192.168.0.104] Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/msg_parser.c:189]: DEBUG: to body [<sip:123@192.168.0.104>] Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <10807> <ACK> Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: rr [loose.c:85]: is_preloaded: No Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [socket_info.c:502]: grep_sock_info - checking if host==us: 13==13 && [192.168.0.104] == [192.168.0.104] Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [socket_info.c:505]: grep_sock_info - checking if port 5062 matches port 5060 Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [forward.c:448]: check_self: host != me Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [socket_info.c:502]: grep_sock_info - checking if host==us: 13==13 && [192.168.0.104] == [192.168.0.104] Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [socket_info.c:505]: grep_sock_info - checking if port 5062 matches port 5062 Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: rr [loose.c:709]: Topmost route URI: 'sip:192.168.0.104:5062;lr;ftag=gYRr.jGE57ZMeLuxYrVmKDuHlnX964Fa' is me Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: rr [loose.c:173]: No next Route HF found Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: rr [loose.c:728]: No next URI found Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=2 , global msg id=1 , T on entrance=0xffffffffffffffff Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: tm [t_lookup.c:527]: t_lookup_request: start searching: hash=17962, isACK=1 Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: tm [t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: tm [t_funcs.c:315]: SER: forwarding ACK statelessly Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [msg_translator.c:204]: check_via_address(192.168.0.3, 192.168.0.3, 0) Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [forward.c:601]: Sending:#012ACK sip:123@192.168.0.104;transport=udp SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.0.104:5062;branch=z9hG4bKcydzigwkX#015#012Via: SIP/2.0/UDP 192.168.0.3:51351;rport=51351;branch=z9hG4bKPjWvIFG5BzVRtUe1jnpRa1ilioYiezOrTz#015#012Max-Forwards: 70#015#012From: <sip:racitup@mobile>;tag=gYRr.jGE57ZMeLuxYrVmKDuHlnX964Fa#015#012To: <sip:123@192.168.0.104>;tag=ggpxl#015#012Call-ID: zH5ub5H6ooVWQmBzZyswgMSLn7rb2xJl#015#012CSeq: 10807 ACK#015#012Content-Length: 0#015#012#015#012. Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [forward.c:603]: orig. len=418, new_len=411, proto=1 Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: last message repeated 5 times Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Apr 6 14:24:30 adventure /usr/sbin/kamailio[8545]: DEBUG: <core> [receive.c:290]: receive_msg: cleaning up ... Good ACK: ... Session Initiation Protocol Request-Line: ACK sip:123@192.168.0.104;transport=udp SIP/2.0 Method: ACK Request-URI: sip:123@192.168.0.104;transport=udp Request-URI User Part: 123 Request-URI Host Part: 192.168.0.104 [Resent Packet: False] [Request Frame: 13] [Response Time (ms): 6489] Message Header Via: SIP/2.0/UDP 192.168.0.3:38653 ;rport;branch=z9hG4bKPjyL0GhUP12E0Dkn2CQ5BR9aQm67DdxKfE Transport: UDP Sent-by Address: 192.168.0.3 Sent-by port: 38653 RPort: rport Branch: z9hG4bKPjyL0GhUP12E0Dkn2CQ5BR9aQm67DdxKfE Max-Forwards: 70 From: <sip:racitup@mobile>;tag=6UtDyO9vBPzP46w7K8AGOfL6qQbFciij SIP from address: sip:racitup@mobile SIP from address User Part: racitup SIP from address Host Part: mobile SIP tag: 6UtDyO9vBPzP46w7K8AGOfL6qQbFciij To: <sip:123@192.168.0.104>;tag=lsboj SIP to address: sip:123@192.168.0.104 SIP to address User Part: 123 SIP to address Host Part: 192.168.0.104 SIP tag: lsboj Call-ID: 1jtxqtjWsDqEL3C2tsrhS3WtRirBFcB7 CSeq: 3674 ACK Sequence Number: 3674 Method: ACK Route: <sip:192.168.0.104:5062 ;lr;ftag=6UtDyO9vBPzP46w7K8AGOfL6qQbFciij> Content-Length: 0 ...

2 2

SER or Kamailio
by Daniel Gonzalez 10 Apr '12

10 Apr '12

Hello, I am starting to deploy a SIP router, and after reading the documentation in http://sip-router.org I am a bit confused. I am planning to integrate the SIP router with an asterisk PBX. Which of the available projects is recommended to get in touch with the technology, SER or Kamailio? Which are most of you in the list using? Is there a particular use case where one or the other is more appropriate? Thanks, Daniel Gonzalez

4 5

RTPPROXY not working with kamailio .
by Ryan Gholam 10 Apr '12

10 Apr '12

Hello I need help concerning this issue , i am creating a testing scenario to make the RTPPROXY work , where i have a server having installed on it Asterisk , running on (192.168.10.15), and another pc connected to the Asterisk server having the ip (192.168.10.17) , on this same pc i've installed RTPPROXY and managed to insert another Ethernet card to connect to a client on ip (192.168.20.3) . After connecting any client on this ip 192.168.20.* , i manage to make a call i hear the ring but no audio conversation . My RTPPROXY command is : /usr/sbin/rtpproxy -F -s udp:192.168.10.17:22222 -l 192.168.20.3/192.168.10.17 -d DBUG:LOG_LOCAL0 Thank you

2 1

status 407 proxy authentication required
by Marwan Idriss 10 Apr '12

10 Apr '12

Greeting, while Registering with correct authentication I am getting : status 407 proxy authentication required even though the registration is successfull, do you know how I can get rid of this error Regards Marwan

1 0

Re: [SR-Users] SER or Kamailio
by Alex Balashov 10 Apr '12

10 Apr '12

My perception is that SER is something like a kernel, but Kamailio a user-oriented distribution and ecosystem wrapper around it that serves to "practicalise" it in an applied way. -- 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 Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >Hello, > >if you install from sources, there is practically everything: ser + >kamailio together, it is up to you what modules you want to use. > >If you install from kamailio packages (debian, centos), you get only >kamailio specific modules (what is located in modules/ and modules_k/). > >Maturity of the code is the same, I am more into kamailio specific >modules, though, since I use them. Kamailio is traditionally more >interactive in terms of public relations, having a clear release policy, >with packages built every time. > >Also, kamailio specific modules have more into SIMPLE extensions (I mean >here presence, xcap, ...), but this is not a problem since you can use >the modules with the ones developed in the past by SER. So there is >nothing different between Kamailio and SER if you install from sources. >As mentioned in the link, there are just some inter-module dependencies, >so when using a module, you have have to use another specific one. This >is actually valid in the same group of modules (like in modules_k), one >module requires another one. > >What really make the difference in a deployment is the database >structure used behind. If you start from scratch, it does not matter >probably, but people upgrading from older versions, tend to stick to >what they have, so they continue using one or the other. > > From my observations on this mailing list, it is more likely you get >people answering more often for kamailio than ser. Also, there are more >tutorials showing how to use kamailio, including integration with >asterisk, if that is main concern, for example: > > * http://kb.asipto.com/asterisk:index > >Cheers, >Daniel > > > >On 4/9/12 9:43 PM, Daniel Gonzalez wrote: >> Thanks for the link. >> >> If I understand correctly, both projects share the same source code, >> and implement more or less the same functionality. >> Is there a list of specific features which are only available in SER >> or in Kamailio? >> >> Which is the most widely deployed / documented option? Which is more >> mature / stable? Which is the option which interoperates easier with >> Asterisk? >> >> Sorry for the beginner questions, but I have found no place where this >> questions are adressed. >> >> On Mon, Apr 9, 2012 at 9:17 PM, SamyGo <govoiper(a)gmail.com >> <mailto:govoiper@gmail.com>> wrote: >> >> :-| >> >> http://www.kamailio.org/w/sip-router-releases/ >> >> >> >> On Mon, Apr 9, 2012 at 11:06 PM, Daniel Gonzalez >> <gonvaled(a)gonvaled.com <mailto:gonvaled@gonvaled.com>> wrote: >> >> Hello, >> >> I am starting to deploy a SIP router, and after reading the >> documentation in http://sip-router.org I am a bit confused. I am >> planning to integrate the SIP router with an asterisk PBX. >> Which of >> the available projects is recommended to get in touch with the >> technology, SER or Kamailio? Which are most of you in the list >> using? >> Is there a particular use case where one or the other is more >> appropriate? >> >> Thanks, >> Daniel Gonzalez >> >> _______________________________________________ >> 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 >> >> >> >> _______________________________________________ >> 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 >> >> >> >> >> _______________________________________________ >> 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 >Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany >http://www.asipto.com/index.php/kamailio-advanced-training/ > > >_______________________________________________ >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

Re: [SR-Users] Installing in a non-standard path
by Alex Balashov 10 Apr '12

10 Apr '12

The prefix only specifies the destination for the build targets. As far as things like module paths, they are not affected. You will have to change them manually in kamailio.cfg. However, the good news is that there aren't a whole lot of paths to change; you really only need to change it in kamailio,cfg, and in any init scripts. -- 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 Daniel Gonzalez <gonvaled(a)gonvaled.com> wrote: >Hello, > >This is my first message to this list. > >I have installed sip-router as follows (my system is Ubuntu 10.04): > > cd /tmp > git clone --depth 1 git://git.sip-router.org/sip-router sip-router > cd sip-router > git checkout -b 3.1 origin/3.1 > local prefix=/tmp/my/very/strange/path > mkdir -p $prefix > sudo apt-get install -y flex > sudo apt-get install -y docbook2x > make proper > make cfg prefix=$prefix > make all prefix=$prefix > make install prefix=$prefix > >And this has worked as expected. > >Now I try to start the daemon: > > $ /tmp/my/very/strange/path/sbin/ser > loading modules under /usr/lib/ser/modules:/usr/lib/ser/modules_s > >What can I do so that my non-standard path is fully recognized by all >sip router tools? > >Thanks, >Daniel Gonzalez > >_______________________________________________ >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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