- sr-users - kamailio.org

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

7 6

$200 bounty
by copycall 21 Jun '12

21 Jun '12

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

10 25

Bug with server originated call
by David 20 Jun '12

20 Jun '12

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

2 2

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

20 Jun '12

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

4 5

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

20 Jun '12

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

2 3

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

20 Jun '12

4 7

rtimer question
by Alex Balashov 20 Jun '12

20 Jun '12

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

2 1

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

20 Jun '12

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

4 7

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

20 Jun '12

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

2 2

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

20 Jun '12

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

2 1

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

19 Jun '12

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

1 0

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

19 Jun '12

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

2 1

IP ToS
by Olle E. Johansson 19 Jun '12

19 Jun '12

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

2 1

install problem
by copycall 19 Jun '12

19 Jun '12

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

2 1

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

18 Jun '12

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

2 1

Sipcapture and Siptrace Compile Error
by Akan 18 Jun '12

18 Jun '12

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

2 4

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

18 Jun '12

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

1 0

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

15 Jun '12

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

3 5

Architecture
by Uriel Rozenbaum 15 Jun '12

15 Jun '12

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

3 3

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

15 Jun '12

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

3 3

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

15 Jun '12

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

1 0

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

14 Jun '12

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

2 7

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

14 Jun '12

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

1 1

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

14 Jun '12

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

2 9

Kamailio debian repository changes
by Jon Bonilla 14 Jun '12

14 Jun '12

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

1 0

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

14 Jun '12

Hello, I'm having trouble using algorithm 7 in dispatcher module. Here is my kamailio version. The problem is that despite multiple dispatcher entries corresponding to one set all calls go only to only one destination. [root@control1 ~]# kamailio -V version: kamailio 3.2.3 (x86_64/linux) 59f87e flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 59f87e compiled on 18:04:08 Apr 19 2012 with gcc 4.1.2 [root@control1 ~]# Here is the dipacther module params. # ------- Load-balancer params ------ modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:proxy@109.XXX.2XX.XX") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") modparam("dispatcher", "hash_pvar", "$avp(i:274)") modparam("dispatcher", "ds_hash_size", 9) modparam("dispatcher", "ds_hash_initexpire", 60) Below is the function call based on avp(i:274) avp_db_query("select uid from uid_did_map where accountcode = '$fU'","$avp(i:274)"); xlog("L_INFO","$rm from $fu (IP:$si:$sp) --------- UID:'$avp(i:274)-----------'\n"); ds_load_update(); if(!ds_select_dst("1","7")) { sl_send_reply("500", "No destination available"); xlog("route[MAIN] : $rm : No destinations available for $rd"); exit; } I'm getting two different values for avp(i:274) {26,29} but the destination are not hashed on its base. Please help. Regards, Sammy G.

3 15

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

14 Jun '12

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

1 0

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

14 Jun '12

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

2 3

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

13 Jun '12

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

2 1

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

13 Jun '12

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

4 14

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

13 Jun '12

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

2 1

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

13 Jun '12

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

1 0

About sdpops
by Grégoire Vandendeurpel 13 Jun '12

13 Jun '12

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

2 8

sipcapture logging
by Konstantin M. 13 Jun '12

13 Jun '12

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

3 6

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

12 Jun '12

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

1 0

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

12 Jun '12

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

5 10

Shared memory fragmentation
by Morten Isaksen 12 Jun '12

12 Jun '12

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

3 4

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

12 Jun '12

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

2 5

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

12 Jun '12

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

1 0

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

11 Jun '12

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

1 0

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

11 Jun '12

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

1 0

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

11 Jun '12

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

4 6

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

11 Jun '12

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

2 2

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

11 Jun '12

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

2 2

Dispatcher PSTN
by Kr0m 11 Jun '12

11 Jun '12

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

2 1

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

11 Jun '12

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

2 1

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

10 Jun '12

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

2 2

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

08 Jun '12

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

6 7

lcr from_uri not matching
by Ben WIlliams 08 Jun '12

08 Jun '12

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

3 8

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

07 Jun '12

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

2 1

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

07 Jun '12

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

4 6

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

07 Jun '12

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

1 0

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

07 Jun '12

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

2 2

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

07 Jun '12

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

3 4

kamailio as SBC
by Mino Haluz 06 Jun '12

06 Jun '12

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

5 4

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

06 Jun '12

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

1 0

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

05 Jun '12

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

3 3

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

05 Jun '12

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

3 4

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

04 Jun '12

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

2 2

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

01 Jun '12

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

1 0

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

01 Jun '12

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

1 0

multiple t_on_branch[]
by Vitaliy Aleksandrov 01 Jun '12

01 Jun '12

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

2 1

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.

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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 !

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