June 2012 - sr-users - kamailio.org

system call
by Kelvin Chua 28 Jun '12

28 Jun '12

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

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Question about git update
by Ricardo Martinez 28 Jun '12

28 Jun '12

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

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

28 Jun '12

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

28 Jun '12

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

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

28 Jun '12

Hello all of you, I tried to install kamallio 3.3 version released a couple days ago . I followed the steps mentioned here https://www.kamailio.org/wiki/install/3.3.x/git Everything works fine until I have to create mysql database . Everything goes fine until I have to install mysql database (step 7) : I run "sudo /usr/local/sbin/kamdbctl create" for database creation but I get this : INFO: test server charset /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 105: mysql: command not found /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 106: mysql: command not found Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 71: mysql: command not found ERROR: Creating core database and grant privileges failed! It seems that mysql -serverside- is not installed (my opinion ...) .What do you think? Many Thanks, Aurel ps: please excuse me if this is not right forum for this question but I did not find the right one ....

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

28 Jun '12

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

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dlg_refer
by Uri Shacked 28 Jun '12

28 Jun '12

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

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

27 Jun '12

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

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default kamailio configuration
by Vitaliy Aleksandrov 27 Jun '12

27 Jun '12

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

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

27 Jun '12

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

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

26 Jun '12

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

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

26 Jun '12

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

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

25 Jun '12

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

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

25 Jun '12

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

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

25 Jun '12

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

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

25 Jun '12

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

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

22 Jun '12

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

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

22 Jun '12

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

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

22 Jun '12

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

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

22 Jun '12

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

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

21 Jun '12

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

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$200 bounty
by copycall 21 Jun '12

21 Jun '12

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

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Bug with server originated call
by David 20 Jun '12

20 Jun '12

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

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

20 Jun '12

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

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

20 Jun '12

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

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sr-users June 2012