[SR-Users] Kamailio 6.0 architecture

[Olle E. Johansson]

Dear Kamailians!

We got news for you! The Kamailio 6.0 architecture is finalised! It's something 
we've been working on for a long time in the project team and you've already 
seen part of it in the recent Kamailio 5 release - the KEMI interface for LUA.

We've realised that we have too many modules, especially all those IMS* 
modules. And the core is getting slimmer and slimmer. This is what we're going 
to do:

* All RPC, EVENTAPI,AUTH, JSON, TM, HTTP_CLIENT and TLS modules will be 
integrated back into the core. 
* The EVENT API will be the core API for external modules
* The rest of the functionality in the modules will have to be rewritten in LUA 
and maintained as separate GITHUB projects outside of the core

After successfully integrating LUA into the heart of Kamailio and exposing the 
Kamailio API we have decided to make life more simple for the core maintainers. 
Maintaining well over 100 modules took too much time, and who cares about some 
of the stuff? Billing, least cost routing and all the other crazy stuff in 
modules are going away anyhow as all services will be free. Calls really wants 
to be free. And presence is no longer needed, we'll just tap into the mobile 
phones from the core to know where everyone is.

The removal of the modules will take place during the month of April 2017 and a 
new core will raise like a Fenix from the ashes, a much larger, monolithic and 
complex core with new functionality that will take Kamailio to new areas, like 
a message bus for games, an API server for mobile yoga apps and a control plane 
for embedded IoT devices. "Kamailio everywhere" is the new slogan for the 
Kamailio project.

- "We will all remember April 1st 2017 as a new start of the project" says 
Olle-Carsten Mierla, the new elected project leader. "From now on, Kamailio 
aims to be a standard part of every operating system distribution, defining the 
new core of the Internet.”

We hope all of you join us in this effort and refrain from reporting any bugs 
or other disturbing issues while we rebuild the new Kamailio.

/O
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Re: [SR-Users] [sr-dev] Planning Kamailio v5.0.1

[Olle E. Johansson]

> On 28 Mar 2017, at 09:40, Daniel-Constantin Mierla  wrote:
> 
> Hello,
> 
> I am considering to release Kamailio v5.0.1 sometime next week, likely
> on Wednesday, April 5, 2017. Should anyone be aware of issues not listed
> yet on bug tracker, report them there as soon as possible to try to fix.
> 
> Soon after, we should release a new version from branch 4.4 and the last
> one from branch 4.3.

I personally have no issues. I just want to say Thank You Daniel for all your
hard work getting releases out of the door! 

/O
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Re: [SR-Users] Transaction-persistent data and uac_req_send()

[Olle E. Johansson]

> On 20 Feb 2017, at 20:40, Alex Balashov  wrote:
> 
> Hello,
> 
> I am using uac_req_send() to send INVITEs to a redirect server. However,
> upon receiving the redirect, I need access to some state information
> that was last available at the time of sending the request. 
> 
> Does uac_req_send() allow me to persist AVPs like a normal TM
> transaction? If not, what are my options, other than building my own
> state using htable or what have you? I would really rather not
> maintain—and be responsible for garbage-collecting—externally
> constructed state like that if possible.

If you configure yourself as an outbound proxy you can start a transaction as
you would normally do.

/O
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Re: [SR-Users] Event when acc is written?

[Olle E. Johansson]

You could check the dialog module that already has an event_route executed when 
a call ends.

/O

> On 25 Jan 2017, at 15:45, Tobias  wrote:
> 
> Hi Daniel,
> 
> Thanks. Can you point me to a good example of how this is implemented in 
> another module, perhaps I could then add it myself to acc?
> 
> /Tobias
> 
> From: Daniel-Constantin Mierla >
> Sent: Wednesday, January 25, 2017 3:20 PM
> To: Tobias; Kamailio (SER) - Users Mailing List
> Subject: Re: [SR-Users] Event when acc is written?
>  
> Hello,
> 
> On 25/01/2017 09:44, Tobias wrote:
>> Hi Daniel,
>> 
>> Thanks for your reply.
>> 
>> Disregarding the MySQL ID, would it be possible to get the callid of the 
>> call back from the acc module once a write has been made?
> to my knowledge, there is no cfg event_route executed at that moment. It 
> should not be something complex to add, but requires c coding in acc module.
> 
> Cheers,
> Daniel
> -- 
> Daniel-Constantin Mierla
> www.twitter.com/miconda  -- 
> www.linkedin.com/in/miconda 
> Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - 
> www.asipto.com 
> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com 
> 
> Kamailio World - Our site will be launched soon 
> 
> www.kamailioworld.com 
> the 5th edition May 8-10, 2017 - Berlin, Germany. Website of the event and 
> more details will be available very soon!
> 
> Daniel-Constantin Mierla | LinkedIn 
> www.linkedin.com 
> View Daniel-Constantin Mierla’s professional profile on LinkedIn. LinkedIn is 
> the world's largest business network, helping professionals like 
> Daniel-Constantin Mierla discover inside connections to recommended job 
> candidates, industry experts, and business partners.
> 
> miconda (@miconda) | Twitter 
> www.twitter.com 
> The latest Tweets from miconda (@miconda). Co-founder and leader of Kamailio 
> SIP Server project (former OpenSER). C Dev. Open Source RTC advocate. SIP, 
> VoIP, VoLTE and WebRTC consultancy at Asipto. Berlin, Germany
> 
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Re: [SR-Users] FOSDEM 2017

[Olle E. Johansson]

> On 25 Jan 2017, at 12:52, Victor Seva  
> wrote:
> 
> El 24 ene. 2017 15:05, "Daniel-Constantin Mierla"  > escribió:
> Hello,
> 
> anyone else interested in having a dinner event at Fosdem?
> 
> 
> I'll be there and I would love to join.
Sounds like a great idea!

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Re: [SR-Users] Kamailio cancel branch only after receiving replies

[Olle E. Johansson]

> On 03 Jan 2017, at 09:20, Daniel-Constantin Mierla  wrote:
> 
> Hello,
> this is the behaviour required by rfc, to send cancel only after receiving a 
> provisional reply 1xx.
Which is the only way - before that point we don’t know if any SIP-capable 
software has gotten the INVITE and
the INVITE will be retransmitted still. There’s simply no point in starting to 
send any CANCEL requests at that time.
The big question is really why there was no “100 trying” from the phone before 
the ringing…

/O
> 
> The cancel_b_method parameter is there to control this behaviour, but it may 
> apply only to the case when the cancel is received and needs to be forwarded, 
> so this needs to be investigated for t_cancel_branches("others").
> 
> Perhaps the best is to open an item on issue tracker from 
> github.com/kamailio/kamailio not to forget about it -- these days are rather 
> busy, with the freezing of the release in few days.
> 
> Cheers,
> Daniel
> 
> On 02/01/2017 21:11, Aqs Younas wrote:
>> Greetings list, 
>> I am forking a call to multiple destinations and want to keep the only 
>> branch which sends quicker first 180/183 reply and cancel the remaining 
>> branches. Below is my related cfg snippet.
>> 
>> 
>> modparam("tm", "failure_reply_mode", 3)
>> modparam("tm", "fr_timer", 3)
>> modparam("tm", "fr_inv_timer", 12)
>> modparam("tm", "cancel_b_method", 2)
>> ...
>> route[SIPOUT] {
>> if (uri==myself) return;
>> 
>> append_hf("P-hint: outbound\r\n");
>> append_branch();
>> append_branch();
>> route(RELAY);
>> exit;
>> }
>> onreply_route[MANAGE_REPLY] {
>> xdbg("incoming reply\n");
>> if(status=~"[12][0-9][0-9]") {
>> xlog("L_INFO","Received $rs (IP:$si:$sp)\n");
>> if(status=~"18[03]"){
>> t_cancel_branches("others");
>> xlog("L_INFO","cancelled all other branches\n");
>> }
>> route(NATMANAGE);
>> }
>> }
>> ...
>> 
>> But I see kamailio does not instantly send CANCEL to other branches after it 
>> has received 180/180 from any branch. 
>> Kamailio is sending CANCEL to other branches after they start sending 
>> 180/183 one by one. How can I cancel all other branches instantly?
>> 
>> I am sure there is something wrongly configured in my configuration.  
>> 
>> Any suggestion is much appreciated. 
>> Best Regards.
>> 
>> 
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> 
> -- 
> Daniel-Constantin Mierla
> www.twitter.com/miconda  -- 
> www.linkedin.com/in/miconda 
> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com 
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[SR-Users] Merry Christmas to all Kamailians

[Olle E. Johansson]

Friends,

Trying to calm down and prepare for Christmas with the family.

It’s been another great year for Kamailio and I’m proud to be part of the 
Kamailio development community. 
We’ve made great releases, had a great conference and overall done good stuff 
:-)

For all of you that celebrate Christmas - a Merry Christmas! To the rest of 
you: Happy Holidays!

A big Thank You to all developers and a special giga-Thank You to Daniel for 
all the time you spend
working with the code, helping out in discussions on the lists and in the bug 
tracker. 

And to all of you in the community - Thank You for being part of this! 

Greetings from a cold Sweden, currently without snow in my area just north of 
Stockholm. We celebrate
on Christmas Eve (tomorrow) so there’s a lot of food being prepared right now 
and the last minute
shopping is just about to start...

Keep your Kamailio running!

/Olle


PS. If you are not currently following the project on Twitter - it’s time!
Find us as @kamailioproject


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Re: [SR-Users] DTMF Using INFO

[Olle E. Johansson]

> On 11 Dec 2016, at 08:20, Nahum Nir  wrote:
> 
> :) Thanks, any hints/examples?
The standard kamailio konfiguration supports it by default.

/O
> 
> On Sat, Dec 10, 2016 at 10:41 PM, Alex Balashov  > wrote:
> On Sat, Dec 10, 2016 at 09:07:09PM +0200, Nahum Nir wrote:
> 
> > I tried Googling but didn'tfind anything.
> > Is it possible to support DTMF using INFO?
> 
> Sure, absolutely. :-)
> 
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 
> Tel: +1-706-510-6800  (direct) / +1-800-250-5920 
>  (toll-free)
> Web: http://www.evaristesys.com/ , 
> http://www.csrpswitch.com/ 
> 
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Re: [SR-Users] Out of memory in UB 210: OOM killed process 12261 (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB

[Olle E. Johansson]

I think we need to update the Make system instead of the docs to avoid this 
version.
Or implement some code like we have for OpenSSL that outputs warnings at 
runtime.

/O
> On 07 Nov 2016, at 10:42, Jurijs Ivolga  wrote:
> 
> Hi Daniel,
> 
> I found modules what are impacted by this leak:
> 
> http_client
> utils
> xcap
> http_async_client
> auth_identity
> xcap_client
> 
> I would like to update documentations, but this is first time when I'm 
> updating documentation.
> 
> I believe I need to update xml file for module. Should I then regenerate 
> readme file as stated here:
> 
> https://www.kamailio.org/wiki/devel/how-to/module-readme 
> 
> 
> Or not?
> 
> Then I believe to create pull request with my changes. Correct?
> 
> Maybe you have manual for this?
> 
> With kind regards,
> 
> Jurijs
> 
> On Wed, Oct 12, 2016 at 4:30 PM, Daniel-Constantin Mierla  > wrote:
> Hello,
> 
> ok, so that was...
> 
> Maybe it would be good to add a note to the docs of the module about this 
> issue so people become aware of it. I guess the other http_* modules are 
> affected. Pull requests or other suggestions are welcome, of course!
> 
> Cheers,
> Daniel
> 
> On 12/10/16 15:04, Jurijs Ivolga wrote:
>> Hi Daniel,
>> 
>> Thank you a lot, it looks that issue is solved, after updating libcurl.
>> 
>> I was using following manual for updating libcurl, in case if somebody will 
>> have same issue.
>> 
>> https://www.digitalocean.com/community/questions/how-to-upgrade-curl-in-centos6
>>  
>> 
>> 
>> With kind regards,
>> 
>> Jurijs
>> 
>> On Tue, Oct 11, 2016 at 2:43 PM, Jurijs Ivolga > > wrote:
>> Hi Daniel,
>> 
>> You are correct we are using heavily http_query.
>> 
>> I found following bug report:
>> 
>> https://bugs.centos.org/view.php?id=9391 
>> 
>> 
>> I will try to update to libcurl 7.44 and check if this help.
>> 
>> Thank you a lot Daniel!
>> 
>> With kind regards,
>> 
>> Jurijs
>> 
>> On Tue, Oct 11, 2016 at 10:55 AM, Daniel-Constantin Mierla 
>> > wrote:
>> Hello,
>> 
>> from the logs, it seems to be related to curl library, I see many reports 
>> like:
>> 
>> ==16459== 189,318 bytes in 167 blocks are possibly lost in loss record 681 
>> of 683
>> ==16459==at 0x4C26FEF: calloc (vg_replace_malloc.c:711)
>> ==16459==by 0x104BB699: ??? (in /usr/lib64/libnsspem.so)
>> ==16459==by 0x104AA537: ??? (in /usr/lib64/libnsspem.so)
>> ==16459==by 0x104AB81E: ??? (in /usr/lib64/libnsspem.so)
>> ==16459==by 0x104B0B88: ??? (in /usr/lib64/libnsspem.so)
>> ==16459==by 0x104B77E1: ??? (in /usr/lib64/libnsspem.so)
>> ==16459==by 0xB71ABC9: ??? (in /usr/lib64/libnss3.so)
>> ==16459==by 0xB71AE62: PK11_CreateGenericObject (in 
>> /usr/lib64/libnss3.so)
>> ==16459==by 0xA0674DF: ??? (in /usr/lib64/libcurl.so.4.1.1)
>> ==16459==by 0xA067666: ??? (in /usr/lib64/libcurl.so.4.1.1)
>> ==16459==by 0xA069141: ??? (in /usr/lib64/libcurl.so.4.1.1)
>> ==16459==by 0xA0601C4: Curl_ssl_connect (in /usr/lib64/libcurl.so.4.1.1)
>> That's like almost 200KB lost in this report.
>> From the list of the modules, I see you have utils and I guess you use http 
>> query function from there, is it?
>> 
>> Cheers,
>> Daniel
>> 
>> On 10/10/16 12:06, Jurijs Ivolga wrote:
>>> Hi Daniel,
>>> 
>>> I left valgrind running for little while, not sure if this will be enough.
>>> 
>>> Please find attached log file.
>>> 
>>> Thank you a lot for your help!
>>> 
>>> With kind regards,
>>> 
>>> Jurijs
>>> 
>>> On Fri, Oct 7, 2016 at 7:15 PM, Daniel-Constantin Mierla >> > wrote:
>>> Hello,
>>> 
>>> that's the way it was done for older versions of kamailio.
>>> 
>>> In master and 4.4 the memory debugging is turned on and it is reflected by 
>>> the presence of DBG_SR_MEMORY in the output of 'kamailio -v'.
>>> 
>>> Anyhow, what you reported is not a leak inside kamailio memory manager, but 
>>> a leak of using system memory, so it is not affected by DBG_SR_MEMORY and 
>>> cannot be troubleshooted using the mechanisms for pkg and shm managers.
>>> 
>>> Cheers,
>>> Daniel
>>> 
>> 
>> -- 
>> Daniel-Constantin Mierla
>> http://twitter.com/#!/miconda  - 
>> http://www.linkedin.com/in/miconda 
>> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com 
>> -- 
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda  - 
> http://www.linkedin.com/in/miconda 
> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com 
> 

Re: [SR-Users] Kamailio script config - Generate Documentation

[Olle E. Johansson]

I imagine that you could use tools like Doxygen for this. Doxygen is something 
we use in
Kamailio to create developer docs - web pages that explains variables, 
functions and
takes data from comments with a special syntax.

Here’s one example:
http://rpm.kamailio.org/doxygen/sip-router/branch/master/index.html

There are several similar tools, like jdoc and perldoc.


/O

> On 20 Oct 2016, at 10:33, José Seabra  wrote:
> 
> Hello Daniel,
> Thank you for your reply,
> Can you advice one of these tools please? i will try it :)
> 
> BR
> José Seabra
> 
> 2016-10-20 10:18 GMT+01:00 Daniel-Constantin Mierla  >:
> Hello,
> 
> not aware of any such effort, but maybe tools that can be used to extract the 
> documentation from a c code can be reused to some extent -- inside 
> kamailio.cfg we support the c-style comments, both:
> 
> //
> 
> and
> 
> /*  */
> 
> If anyone tries, would be great to know the results.
> Cheers,
> Daniel
> 
> On 18/10/16 11:22, José Seabra wrote:
>> Hello there,
>> I'm sending this email in order to know if anyone had work or has any idea 
>> how we can generate Documentation from kamailio script using Comments?
>> 
>> This would be great because we could write comments explaining the Route 
>> method during  script development  and use it to generate documentation 
>> about our kamailio script, saving time and leave our deployment documented.
>> 
>> 
>> Thank you
>> --
>> Best Regards
>> José Seabra
>> 
>> 
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> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda  - 
> http://www.linkedin.com/in/miconda 
> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com 
> 
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> 
> 
> 
> 
> --
> Cumprimentos
> José Seabra
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Re: [SR-Users] Kamailio - DMQ dns srv Question

[Olle E. Johansson]

> On 11 Oct 2016, at 09:46, José Seabra  wrote:
> 
> Hi Charles,
> Sorry for my late reply.
> I have tried that parameter but seems that for the dmq FQDN kamailio doesn't 
> send a NAPTR query.
> Even for the SRV query, the kamailio makes it without service 
> associated(_sip._udp).
> 
We should propably define a SRV service tag for DMQ.

/O
> Thank you.
> 
> Best regards
> José Seabra
> 
> 2016-09-26 13:07 GMT+01:00 Charles Chance  >:
> Hi José,
> 
> On 21 September 2016 at 18:28, José Seabra  > wrote:
> Hello,
> I have a doubt related with DMQ dns behavior, I noticed that when kamailio 
> starts, it tries to resolve DMQ name configured on parameter 
> notification_address as the following sequence:
> SRV 
> A
> 
> 
> Isn't supposed kamailio  try first resolve the  NAPTR DMQ name, and then SRV?
> 
> 
> Can you confirm you set dns_try_naptr = yes?
> 
> https://www.kamailio.org/wiki/cookbooks/4.4.x/core#dns_try_naptr 
> 
> 
> https://raw.githubusercontent.com/kamailio/kamailio/master/doc/dns.txt 
> 
> 
> Cheers,
> Charles
> 
> 
> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered 
> office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, Birmingham 
> Science Park, Birmingham B7 4BB.
> 
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> 
> 
> 
> 
> -- 
> Cumprimentos
> José Seabra
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Re: [SR-Users] REGISTER between Kamailio 4.4 and Asterisk 13

[Olle E. Johansson]

> On 26 Sep 2016, at 15:53, Grant Bagdasarian  wrote:
> 
> Hello, 
>  
> We’re trying to get our Asterisk test server to REGISTER with Kamailio.
> The first attempts are ok, but once the nonce timer is exceeded Kamailio 
> starts rejecting the REGISTER requests from Asterisk, due to the nonce being 
> expired.
> The auth_check function is the one returning error -4.
>  
> Is this an issue in Asterisk or Kamailio? Which component is responsible for 
> generating a new nonce value?
> I’m a bit confused regarding the flow of SIP REGISTER. 
In that case, Asterisk should get a new challenge from Kamailio and asterisk 
should send a new register with the
new nonce. Asterisk chan_sip will retry with the old nonce but if the server it 
registers to challenges again,
then it will retry auth automatically.

/O
>  
> I hope someone could help me out on this. 
>  
> Thanks!
>  
> Regards,
>  
> Grant Bagdasarian
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Re: [SR-Users] Load balancing with Kamailio

[Olle E. Johansson]

> On 26 Sep 2016, at 12:15, NITESH BANSAL  wrote:
> 
> 
> 
> 
> 
>  
> Hello,
> 
> I'm planning on doing some smart load balancing with Kamailio.
> We have a distributed network, with multiple Kamailio boxes in different 
> locations serving as Ingress SBC, 
> these Kamailio boxes are the entry point for a SIP call and then they route 
> the call to a pre-configured Asterisk boxes.
> 
> I want to move away from this, I would like these Kamailios to be able to 
> distribute the traffic to Asterisk boxes based on the
> actual load on these boxes, the goal is to be more dynamic?

> 
> Is there any Kamailio module which could do that? Do I need to integrate some 
> other tool(Homer etc) with Kamailio to achieve this?
The dispatcher module is built for this.

Cheers,
/O

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Re: [SR-Users] Kamailio fails to start when MySQL down

[Olle E. Johansson]

> On 16 Sep 2016, at 11:46, Daniel Pocock  wrote:
> 
> 
> 
> I recently set up a Kamailio instance using the default configuration
> for HOMER with MySQL[1]
> 
> The database was not running and Kamailio refused to start.
> 
> Would it be better for Kamailio to start anyway and go into a loop
> trying to connect to the database, just as if auto_reconnect was set?
> 
> This is useful for people who start Kamailio from their init script and
> can't be sure if their database host is up before their Kamailio host boots
> 
> 
Check if you can do that using db_cluster.
In most cases I don’t want Kamailio to start with no database, so in my case 
it’s a good thing (TM) :-)

/O
> 
> 
> 
> 1.
> https://github.com/sipcapture/homer-api/blob/master/examples/sipcapture/sipcapture.kamailio
> 
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Re: [SR-Users] deb.kamailio.org down?

[Olle E. Johansson]

> On 06 Sep 2016, at 13:56, Victor Seva  
> wrote:
> 
> deb.kamailio.org is back online

Victor - I’ll take this opportunity to say Thank You for your work with this 
server!

/O
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Re: [SR-Users] Stress Testing

[Olle E. Johansson]

> On 16 Aug 2016, at 16:26, Jack Stevens  wrote:
> 
> Hi Guys,
>  
> I have been stress testing my Kamailio box but I am unable to get it upto 
> 2000 concurrent calls it starts to fall over at 1300 have you got any ideas 
> on how I can increase the performance of kamilio btw I am also using rtpengine
Can you describe “fall over”

As Kamailio doesn’t handle media it’s likely an issue with rtpengine and the 
developers there needs to respond,
but some more facts would be good. :-)

/O
>  
> Kind Regards
> 
> 
> 
> 
> 
> 
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Re: [SR-Users] t_set_fr() different behavior whether transport is udp or tcp

[Olle E. Johansson]

> On 16 Aug 2016, at 11:41, Daniel Tryba  wrote:
> 
> I'm seeing a different behavior of t_set_fr depending on transports.
Which is correct. UDP is connectionless and thus SIP has timers for 
retransmits and fails when there’s no response. With connection-oriented
protocols, the failure happens when the connection fails, which in many
cases is much faster and not based on any retransmission timers.

In fact - failures in TCP is quite operating system dependent if the URI
results in an IP address. There are some interesting papers on this topic
which for SIP lead to the development of solutions like SIP outbound.

/O

> Scenario is that a endpoint has a failover defined in the registrat
> after 10s (t_set_fr(1) and handling the locally generated 408 to the
> failover destination). This works fine when the request and response
> where delivered over UDP. When the Path is TCP the failover happends
> after 30s (even when using a different time t_set_fr(2) so it is not
> a factor 3 or something like that).
> 
> Setup:
> OK, 10s:
> Orig->UDP->loadbalancer->UDP->registrar->UDP->loadbalancer->TCP->Term
> Fail, 30s:
> Orig->UDP->loadbalancer->TCP->registrar->TCP->loadbalancer->TCP->Term
> 
> Loadbalancer and registrar are kamailio machines (4.3.6). Communication
> between lb and registrar is based on dispatcher and path modules. 
> 1 sip:registrar:5060;transport=udp 8 0
> or via tcp:
> 1 sip:registrar:5060;transport=tcp 8 0
> 
> In the location database of the registrar the difference between the
> cases is:
> socket:
> udp:registrar:5060
> or via tcp
> tcp:registrar:5060
> path:
> 
> or via tcp
> 
> 
> Looking at debug(=3) nothing happens between the initial INVITE and the
> local 408 as far as I can see. 
> 
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Re: [SR-Users] Checking for 200ok Response to a REGISTER request kamailio-Asterisk

[Olle E. Johansson]

(Shooting from the hip, but let’s brainstorm just for fun :-) )

I would consider not saving the incoming REGISTER in the Kamailio location 
database,
notfork it but replicate or forward twice to the ASterisk servers. Keep a 
counter - maybe in a hash table - 
and when the first 200 ok come in, raise the counter and then drop it.

When the second response comes in, save to kamailio location database (yes, it 
works on a response) and
then forward the response to the client. Note that you may end up in trouble 
unless you are really sure
that all servers have exactly the same expiry time.

Whatever you do you will have a UA that retransmits unless you respond with 
some 1xx code and
a situation where you may timeout the UA before you time out on Asterisk - so 
trim the Kamailio
transmit timer to be very short, much shorter than the UA so you make sure that 
Kamailio times out
way ahead of your UA.

Now, you can have a timeout on htable so that you catch the situation where you 
don’t get any
response from Asterisk and do something about it - tell the UA to come back in 
a while
with a retry-after or something else.

I am pretty sure I am missing something here, but it may give you some ideas to 
test out.

Cheers,
/O

> On 27 Jul 2016, at 14:38, Jonathan Hunter  wrote:
> 
> Hello,
> 
> Thanks for the response.
> 
> I appreciate your comments and agree, however the architecture cannot be 
> changed currently so in the meantime its looking to apply a fix to allow for 
> stability in the short term.
> 
> I have built/designed other platforms and registrations don't go anywhere 
> near the Media servers,  so it is a case of working with what we have for the 
> short term due to a number of reasons I wont go into. :)
> 
> Understand where your coming from however.
> 
> Jon
> 
> 
> 
> From: o...@edvina.net 
> Date: Wed, 27 Jul 2016 14:17:14 +0200
> To: sr-users@lists.sip-router.org 
> Subject: Re: [SR-Users] Checking for 200ok Response to a REGISTER request 
> kamailio-Asterisk
> 
> 
> On 27 Jul 2016, at 14:01, Jonathan Hunter  > wrote:
> 
> 
> Hi Guys,
> 
> So currently on our network we have a kamailio server which users register 
> against, we then replicate the register messages to 2 Asterisk boxes sat 
> behind it so that all entities are aware of the registration state of the 
> users.
> 
> REGISTER--->KAMAILIO>ASTERISK A
> >ASTERISK B
> 
> With a REGISTER---200OK exchange between Kamailio and Asterisk.
> 
> We have an issue where at some points the Asterisk servers when under load 
> dont respond with a 200 ok(something being investigated)  to the register 
> messages sent to kamailio, so I am just working on some logic for the 
> register message to be resent using the t_replicate and t_set_fr functions.
> 
> This works well should both Asterisk servers not respond, however, as I am 
> using replicate and it is parallel forking, if say Asterisk A answers first 
> and is available with a 200ok then that in turn cancels the register message 
> branch being sent to Asterisk B(which I know is fine), however  there could 
> be a scenario where Asterisk B doesnt respond, and we wont know about it to 
> try and resend the Register message, as the branch is cancelled.
> 
> Hope that makes sense?
> 
> I am looking at checking that both the Asterisk servers have responded and 
> sent a 200ok, which I can grab in an onreply route but Im just wondering if 
> someone has done something similar or has any suggestions as it is tricky to 
> achieve currently.
> 
> I have also thought about stateless working but I really need kamailio to 
> keep retransmitting the register until it gets a response.
> 
> Many thanks
> 
> In my view you are making a very complex solution. Why do you need to store 
> the same registration in so many places? That’s 
> indicating a problem in the architecture.
> 
> /O
> 
> 
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Re: [SR-Users] Checking for 200ok Response to a REGISTER request kamailio-Asterisk

[Olle E. Johansson]

> On 27 Jul 2016, at 14:01, Jonathan Hunter  wrote:
> 
> 
> Hi Guys,
> 
> So currently on our network we have a kamailio server which users register 
> against, we then replicate the register messages to 2 Asterisk boxes sat 
> behind it so that all entities are aware of the registration state of the 
> users.
> 
> REGISTER--->KAMAILIO>ASTERISK A
> >ASTERISK B
> 
> With a REGISTER---200OK exchange between Kamailio and Asterisk.
> 
> We have an issue where at some points the Asterisk servers when under load 
> dont respond with a 200 ok(something being investigated)  to the register 
> messages sent to kamailio, so I am just working on some logic for the 
> register message to be resent using the t_replicate and t_set_fr functions.
> 
> This works well should both Asterisk servers not respond, however, as I am 
> using replicate and it is parallel forking, if say Asterisk A answers first 
> and is available with a 200ok then that in turn cancels the register message 
> branch being sent to Asterisk B(which I know is fine), however  there could 
> be a scenario where Asterisk B doesnt respond, and we wont know about it to 
> try and resend the Register message, as the branch is cancelled.
> 
> Hope that makes sense?
> 
> I am looking at checking that both the Asterisk servers have responded and 
> sent a 200ok, which I can grab in an onreply route but Im just wondering if 
> someone has done something similar or has any suggestions as it is tricky to 
> achieve currently.
> 
> I have also thought about stateless working but I really need kamailio to 
> keep retransmitting the register until it gets a response.
> 
> Many thanks

In my view you are making a very complex solution. Why do you need to store the 
same registration in so many places? That’s 
indicating a problem in the architecture.

/O

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Re: [SR-Users] Detect 200 OK for ReINVITE

[Olle E. Johansson]

> On 12 Jul 2016, at 15:18, Grant Bagdasarian  wrote:
> 
> Hi,
>  
> Is it possible in Kamailio, or SIP in general, to detect if a 200 OK is for a 
> ReINVITE?
>  
Not from the 200 OK, but the matching INVITE you can check if there’s a to-tag 
in Kamailio and SIP.

In Kamailio, if you are stateful, you can use the TMX module pseudovariables to 
check the request
matching the response and check if there’s a to-tag. Check the cookbook for 
$T_req(pv)

http://www.kamailio.org/wiki/cookbooks/4.4.x/pseudovariables

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Re: [SR-Users] Kamailio compact headers ?!

[Olle E. Johansson]

> On 06 Jul 2016, at 11:38, Alex Balashov  wrote:
> 
> Stefan, 
> 
> The construction of headers is generally the province of the UAs/endpoints of 
> the call. Kamailio understands compact headers, as all compliant SIP stacks 
> do, but it can't singlehandedly reformat them. 
I think the question was about the headers added by Kamailio - Via and 
Record-Route. Which makes the question perfectly valid :-)
I can’t remember such a setting anywhere and am curious on why it’s needed.
/O

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Re: [SR-Users] Kamailio compact headers ?!

[Olle E. Johansson]

> On 06 Jul 2016, at 11:36, Anonim Stefan  wrote:
> 
> Hi,
> 
> Is there a parameter to be enabled such that Kamailio will append headers in 
> compacted form? (I'm thinking of Via: and Record-Route:)


Just curious - why would you need that?

/O
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[SR-Users] Kamailio summer-of-documentation - join the effort

[Olle E. Johansson]

Hi!


Looking for a summer-of-documentation project? Which README is the best? Which 
one is so full of Swedish that you totally fail to parse it? Check our 
documentation pages, suggest improvements on the mailing lists (or here on 
Facebook) and give us feedback! 

This is something you can spend your lazy summer days on - a nice cool drink, 
cool sunshades and Kamailio documentation!

We constantly work to update our README files for every module and the 
cookbooks for the Kamailio core, the transformations and pseudovariables. But 
we’re developers, not casual administrators and propably fail to see some 
obvious things missing. 
Please tell us (feel free to use the mailing lists).

And if you can - but not required at all - you can suggest rewrites in the XML 
source and upload to github.

Let’s keep our docs up to date and make sure they really assist our users!

http://kamailio.org/docs/modules/devel/

Have a great summer!

Regards,
/O
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Re: [SR-Users] Simple Kamailio configuration for Asterisk load balancing?

[Olle E. Johansson]

> On 05 Jul 2016, at 19:36, Tickling Contest  wrote:
> 
> Hello,
> 
> I am beginning to front my Asterisk cluster with Kamailio and so far my 
> biggest issue is the complete lack of quick-start-like documentation for 
> this. Is there any place I can get a very simple HA configuration (telling me 
> where the config files are, for starters, is a good thing) for Kamailio with 
> the following features:
> 
> (a) Support an arbitrarily large number of Asterisk servers (say, upto 10).
> (b) Offload SIP registration to a realtime/mysql DB used by the PBXs.
> 
> Please also let me know if I should have to change my Asterisk PBX config in 
> any way for this to happen.
> 
> On a very related note, while I appreciate the fact that you need to _really_ 
> understand SIP to configure Kamailio, it should be possible to get this setup 
> by just running a script or GUI. It is difficult to find information on how 
> to load balance multiple PBXs behind Kamailio.

While I haven’t got full configurations - I do have some notes published on 
things you need to consider
when doing this - especially the offloading of registrations.

An old but still valid presentation (from 2010):
http://www.slideshare.net/oej/astricon-2010-scaling-asterisk-installations

General presentations:
http://www.slideshare.net/oej/why-is-kamailio-so-different-an-introduction 

http://www.slideshare.net/oej/kamailio-a-quick-introduction 


You have a lot of examples in the dispatcher module README that will help you.

Regards,
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Re: [SR-Users] Kamailio Iterate All Headers

[Olle E. Johansson]

> On 26 Jun 2016, at 22:29, Colin Morelli  wrote:
> 
> Hey all,
> 
> Back with more questions.
> 
> I'm using Kamailio to make an HTTP call to my API to perform authentication 
> and message routing. Currently, I'm trying to build up the post body that I 
> send to my API to make those decisions.
> 
> I've cherry picked a few of the headers that are important in my routing 
> decisions. But, ideally, I'd like to just iterate over all the SIP message 
> headers and append them as request parameters to my API call. Is this 
> possible with Kamailio? I've been looking through the docs and can't seem to 
> find a function to iterate the full list of headers.
> 
> Thanks in advance.

$mb 

will give you the full SIP message.

/O


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Re: [SR-Users] Kamailio ERROR "Address already in use" in tcp_init().

[Olle E. Johansson]

> On 20 Jun 2016, at 04:32, joey@yulong.com wrote:
> 
> 0(3834) ERROR:  [tcp_main.c:2790]: tcp_init(): bind(a, 0x2b7d245f0f6c, 
> 16) on 127.0.0.1:5060 : Address already in use

Kamailio could not listen to the localhost socket - are you running some other 
SIP application on the same machine?

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Re: [SR-Users] http_client module configuring User-Agent

[Olle E. Johansson]

> On 16 Jun 2016, at 12:05, Jurijs Ivolga  wrote:
> 
> Hi All,
> 
> I tried today new module http_client. I just made GET request using 
> http_client_query in following way:
> 
> http_client_query("http://cool.api.com ", 
> "$var(result)"); 
> 
> It is working, but I can't see User-Agent in request. Below you can find my 
> module options:
> 
> modparam("http_client", "useragent", "kamailio")
> modparam("http_client", "connection_timeout", 1)
> 
> Is it a bug, or am I doing something wrong?

That function works like the old http_query in utils. I should propably 
document that it’s limited.
Use the functions that use the connection structure to get full functionality.

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Re: [SR-Users] Kamailio consult CNAM server through SIP Subscribe.

[Olle E. Johansson]

> On 14 Jun 2016, at 15:45, José Seabra  wrote:
> 
> Hello,
> 
> Regarding to this implementation, I'm sending SIP Subscribe message from 
> uac_req_send function to CNAM server but it is unavailable and Kamailio 
> doesn't receive any reply from CNAM server, What I'm noticing is that 
> kamailio is retransmitting the SIP Subscribe msg until get "408 request time 
> out".
> 
> In order to implement a failure_route to this SIP Subscribe msg sent from 
> uac_req_send I put kamailio send this SIP msg first to itself then kamailio 
> set the failure_route and send SIP Subscribe msg to CNAM server using t_relay 
> from script configuration file, but kamailio still not entering on 
> failure_route block.
I am doing the same with SIP REGISTER and get into the failure route with a lot 
of different issues. Haven’t specifically tried with a local TM timeout, a 
locally generated 408 though. 
> 
> How can I set  failure_route to this SIP Subscribe msg sent from 
> uac_req_send()
SHould work with a normal failure route trigger in the script. 

Sounds very strange ...

/O
> 
> Regards
> 
> 2016-06-07 22:06 GMT+01:00 José Seabra  >:
> Hello, 
> 
> Thank you for the feedback.
> 
> BR
> 
> 2016-06-07 8:38 GMT+01:00 Pavel Eremin  >:
> Hi,
> 
> We using PERL moodule and PERL script to do this.
> Don't sure but, Kamailio will stop anyway when it sends www query and wating 
> answer.
> 
> 
> 2016-06-06 19:56 GMT+05:00 José Seabra  >:
> Hello there,
> I need to use kamailio to consult an CNAM server in order to get the caller 
> id name and number to the call, the diagram bellow represents what i need 
> implement:
> 
> 
> ​
> Can I  implement this behavior through the following kamailio modules:
> tm and tmx module (call the function t_suspend and t_continue in order to 
> stop INVITE transaction and resume it).
> UAC module (generate the SUBSCRIBE sip message)
> Some kamailio scripting logic to manage this behavior.
> what are your advices to implement this scenario?
> 
> Thank you for your help
> 
> Regards
> 
> 
> 
> -- 
> Cumprimentos
> José Seabra
> 
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> 
> 
> 
> 
> -- 
> Cumprimentos
> José Seabra
> 
> 
> 
> -- 
> Cumprimentos
> José Seabra
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Re: [SR-Users] forward calls from kamailio(RTPengine) to FREESWITCH

[Olle E. Johansson]

> On 09 Jun 2016, at 13:27, Ankitt Sharma  wrote:
> 
> hello
> 
> i am using using kamailio + RTPengine in my setup and i want to forward every 
> single call to freeswitch(for recording purpose),
In that case you can remove RTPengine from the mix, as I am pretty sure 
FreeSwitch can handle symmetric media.
Unless RTPproxy helps you with something else.

/O
> 
> all three are running on the same machine (testing in the local network)
> my requirement for media flow is:
> 
> UA1-->RTPproxy-->>freeswitch-->RTPproxy-->UA2
> 
> 
> my user agents are using webrtc media profile so my rtp engine flags are :
> 
> rtpengine_manage("trust-address  replace-origin replace-session-connection 
> RTP/SAVPF");
> 
> i've attached my configuration file 
> 
> i've configured freeswitch for media_webrtc=true ,still 
> when i try calling the freeswitch send 488 without sending call to other end 
> 
> should media-address flag be used to forward media through, if yes how it 
> should be done ??
> 
> 
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Re: [SR-Users] Nonce extract from SIP Meesage

[Olle E. Johansson]

> On 30 May 2016, at 15:40, Sergio Serrano  wrote:
> 
> Hi all,
> 
>   I just try integrate Kamailio with external software and We
> want taht kamailio doesn't authenticate users. I want that the
> autehtication process would be done by external software.  I want To
> receive a REGISTER for a user. Then I send a 407 and then when I
> receive all paramters for autehtincation(nonce, response, digestURI,
> etc, etc etc) send that data to external software and receive if data
> is correct, then sent 200 OK and save location in external software. 
> 
> Bu t I can see any pseudovariable or enything else to obtain that dato
> to provide to external software. Anyone could think that this process
> could be done?

This is one of the reasons I built  http_client - we fetch the data with
http_client and feed it to the pvar-based authentication.

It’s tricky if you can’t use that since the 200 OK is special and
needs to include all current contacts, not just the recently registred
one. 

So fetch username and pwd from external system or feed external
system with all data to confirm.

/O


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Re: [SR-Users] Kamailio not sending reply

[Olle E. Johansson]

> On 26 May 2016, at 18:47, Jeremy Betts  wrote:
> 
> Hello,
> 
> I'm having an issue where kamailio is not sending the failure reply as I 
> would expect when receiving calls from a certain provider. Kamailio sends the 
> 403 response on calls from other providers. The big difference I see is that 
> the "problem" provider is using compact headers. I am trying to force the 403 
> response to be sent for testing purposes but I just can't get it to reply to 
> this one provider. 
> 
> I've included traces and configuration excerpts below. Any help is much 
> appreciated!
> 
> Problem Call (No Reply Sent):
There must be something missing here. Without a response, there would be no 
ACK. 

/O
> 
> U 63.79.178.192:5060 -> 184.171.164.100:5060
> INVITE sip:+19727289377@184.171.164.100;transport=UDP;user=phone 
>  SIP/2.0.
> v: SIP/2.0/UDP 
> 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013.
> Record-Route:  .
> f:  
> ;tag=-45026-41c7bce-729616d4-41c7bce.
> t:  
> .
> i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719.
> CSeq: 1 INVITE.
> Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK.
> v: SIP/2.0/UDP 
> SCR9:5060;maddr=199.173.94.144;branch=z9hG4bK-41c7bce-f439f11-21b78f96;received=199.173.94.144.
> Max-Forwards: 27.
> m:  
> .
> k: 100rel, resource-priority, replaces.
> c: application/sdp.
> l: 235.
> P-Asserted-Identity:  
> .
> Privacy: none.
> .
> v=0.
> o=PVG 1464277233580 1464277233580 IN IP4 199.173.68.106.
> s=-.
> p=+1 613555.
> c=IN IP4 199.173.68.106.
> t=0 0.
> m=audio 55380 RTP/AVP 18 0 8 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=fmtp:18 annexb=no.
> 
> #
> U 63.79.178.192:5060 -> 184.171.164.100:5060
> ACK sip:+19727289377@184.171.164.100;transport=UDP;user=phone 
>  SIP/2.0.
> v: SIP/2.0/UDP 
> 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013.
> f:  
> ;tag=-45026-41c7bce-729616d4-41c7bce.
> t:  
> ;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.1aac.
> i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719.
> CSeq: 1 ACK.
> l: 0.
> Max-Forwards: 27.
> .
> 
> Successful Call (Reply Sent):
> 
> #
> U 64.136.173.31:5060 -> 184.171.164.100:5066
> INVITE sip:7146466334@184.171.164.100:5066 
>  SIP/2.0.
> Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345.
> Record-Route: 
>  
> .
> To:  .
> From:  
> ;tag=sansay3408006464rdb14345.
> Call-ID: 1089333243-0-3123809354@64.136.173.226 
> .
> CSeq: 1 INVITE.
> Contact:  
> .
> Supported: timer.
> Session-Expires: 1800;refresher=uac.
> Min-SE: 90.
> P-Asserted-Identity:  
> .
> Privacy: none.
> Expires: 120.
> Max-Forwards: 67.
> Content-Type: application/sdp.
> Content-Length: 274.
> .
> v=0.
> o=Sansay-VSXi 188 1 IN IP4 64.136.173.31.
> s=Session Controller.
> c=IN IP4 69.85.185.142.
> t=0 0.
> m=audio 37970 RTP/AVP 0 18 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=sendrecv.
> a=ptime:20.
> 
> 
> #
> U 184.171.164.100:5066 -> 64.136.173.31:5060
> SIP/2.0 403 DID Lookup Failed.
> Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345.
> To:  
> ;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec.
> From:  
> ;tag=sansay3408006464rdb14345.
> Call-ID: 1089333243-0-3123809354@64.136.173.226 
> .
> CSeq: 1 INVITE.
> Server: x-Freevoice SIP Proxy 4.21.
> Content-Length: 0.
> .
> 
> #
> U 64.136.173.31:5060 -> 184.171.164.100:5066
> ACK sip:7146466334@184.171.164.100:5066  
> SIP/2.0.
> Via: 

Re: [SR-Users] Kamailio Client-Server

[Olle E. Johansson]

> On 02 May 2016, at 17:01, Shiv Patidar  wrote:
> 
> hiii
> 1.We are writing our own SIP Client to test my Client i am using Kamailio 
> server
> 2. So i am trying to map the identities used between Client and Server
> 3. If can possible can u please help me to the steps has to be follower to 
> set identities common between Client and Server.
> 4. Or if u have some test code for client can u share the SIP message dump 
> for test code so i can re-use the same for configuration my client code.

The default configuration file should be a good start. You just invent an 
account in your client and register it - and it will become your AOR.
Start there, and then add complexity.

/O


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Re: [SR-Users] Multiple SIP-servers with SRV-records and authentication secrets

[Olle E. Johansson]

> On 03 Apr 2016, at 18:09, Alfred E. Heggestad  wrote:
> 
> Dear SIP-experts and DNS-SRV gurus;
> 
> 
> I have some questions to the deployers of SER/Kamailio and
> best current practice for multiple SIP-servers with SRV-records
> and authentication. This is not a question about Kamailio itself
> but rather experience with deployment of it in the field.
> 
> 
> The current usecase is:
> 
> 1. Multiple SIP-servers are deployed for the same domain
> 
> 2. The DNS is configured with SRV-records for load balancing,
>   example: (lets call the domain "example.com")
> 
>   $ host -t SRV _sip._udp.example.com
>   _sip._udp.example.com has SRV record 20 0 5080 alpha1.example.com.
>   _sip._udp.example.com has SRV record 20 0 5080 alpha2.example.com.
> 
> 3. when a SIP client registers, it resolves the domain using RFC3263 [1]
>   and the first REGISTER request is sent to SIP-Server #1
> 
> 4. SIP-server #1 replies with 401 containing the authentication challenge
> 
> 5. The SIP Client adds the authentication header to the REGISTER
>   request and re-sends it, but this time also using RFC 3263, and due
>   to DNS rotation the request is sent to SIP-Server #2
> 
> 6. Now, because the SIP-Servers are configured with _different_
>   secrets in the "auth" module [2], the REGISTER request
>   fails with authentication error.
> 
> 
> 
> Now, I know that it is common for SIP user-agents to send both requests
> to the same SIP-server instance. Baresip [3] is not doing that, it does
> a new RFC 3263 lookup for all requests (except e2e ACK/CANCEL).
> 
> 
> so here are my questions:
> 
> - What is common practice in the field, to configure auth module
>  with the same "secret" or different "secret" values?
Within the same realm the user has the same credentials. As long as
you have the same realm, one user name should NOT have different
secrets. If the two servers have different secrets for the same user, the
realm should be different.
I am not sure if there’s a document stating that every server in the same
realm has to handle the same set of nonces, i.e. accept an authentication
based on a nonce from another server, but in my world that follows
the idea of a realm.

> 
> - Do you know if there is any reference to IETF documents about how
>  this should be handled? RFC 3263 says that every request should be
>  resolved, except:
> 
>  "The procedures here MUST be done exactly once per transaction, where
>   transaction is as defined in [1].  That is, once a SIP server has
>   successfully been contacted (success is defined below), all
>   retransmissions of the SIP request and the ACK for non-2xx SIP
>   responses to INVITE MUST be sent to the same host.  Furthermore, a
>   CANCEL for a particular SIP request MUST be sent to the same SIP
>   server that the SIP request was delivered to.”

I think many developers are confused about REGISTER. It does not
create a dialog but some software handles it like one. It’s easy to
fall into a trap thinking about REGISTER as a dialog.

> 
> - What is common practice for SIP user-agents to do in this case?
Not many does what baresip does, which I think is correct. In Kamailio
multiple servers can handle the same nonce, so this behaviour works
very well with multiple Kamailios .

/O
> 
> 
> 
> 
> 
> 
> /alfred
> 
> [1] https://tools.ietf.org/html/rfc3263#section-4.4
> 
> [2] http://www.kamailio.org/docs/modules/3.4.x/modules/auth.html#auth.secret
> 
> [3] https://github.com/alfredh/baresip/issues/39
> 
> 
> 
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Re: [SR-Users] Kamailio 4.3.4: receive_msg(): no via found in reply

[Olle E. Johansson]

> On 04 Apr 2016, at 10:02, Grant Bagdasarian  wrote:
> 
> Hi Daniel,
>  
> You mentioned below about this issue not being complaint with the RFC.
> Our supplier is telling us this is normal behavior and that they went through 
> the RFC and found this was normal behavior.
>  
> If it’s not too much work, could you tell me in which part of the RFC this is 
> described?
>  
8.2.6.2 Headers and Tags


   The From field of the response MUST equal the From header field of
   the request.  The Call-ID header field of the response MUST equal the
   Call-ID header field of the request.  The CSeq header field of the
   response MUST equal the CSeq field of the request.  The Via header
   field values in the response MUST equal the Via header field values
   in the request and MUST maintain the same ordering.



RFC 3261 basics. The UAS copies ALL Via:s from the request into the response. 
This is how we route responses through
a SIP network, so without all headers, communication will likely be very broken.

/O
 
> 
> 

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Re: [SR-Users] Multiple Registrations - Overwrite oldest contact

[Olle E. Johansson]

> On 01 Apr 2016, at 22:37, Bruno Emer  wrote:
> 
> Now, I have just one more question about my scenario: when I register a user, 
> is there a way to create something like a "custom field" on the location? To 
> explain better: If possible, I can add something like a custom header that 
> says if the user is registered from a phone device or the web interface, then 
> I can perform the loop and replace only the oldest contact using that 
> device...
> 
Specific devices should have a unique instance-ID in the contact header so you 
can separate them. otherwise it’s guesswork.
This is part of many SIP standards and used by more and more SIP stacks.

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Re: [SR-Users] Multiple Registrations - Overwrite oldest contact

[Olle E. Johansson]

That requires a bit more of reading the same doc page, but I still think you 
can do it.

Run reg_fetch_contacts and check how many you have. Loop through them and 
unregister by the ruri (id) of the one that expires first
if you have too many.

We’ve made this function to make the location database handling very 
transparent to the script writer that wants to go beyond 
the ordinary and do some fun scripting :-)

http://kamailio.org/docs/modules/devel/modules/registrar.html#registrar.f.reg_fetch_contacts
http://kamailio.org/docs/modules/devel/modules/registrar.html#registrar.f.unregister

Kamailio is cool, isn’t it? :-)

/O

> On 01 Apr 2016, at 22:18, Bruno Emer  wrote:
> 
> This is my problem... I want to have 2 contacts for AOR, and not only one. 
> 
> When the 3rd arrives, it must remove the oldest and continue with 2...
> 
> If I am using the 0x04 parameter, it will save just one contact, and the user 
> will not be able to get the Invite both, in the mobile device and web 
> interface.
> 
> Bruno Emer
> 
> Mobile: +55 11 96540-0044  
> email: brunoe...@gmail.com 
>    
>   
>   
>   
>   
> 2016-04-01 17:07 GMT-03:00 Bruno Emer  >:
> Hello, Olle!
> 
> Thanks for your help in this case, and don't worry about the time (actually I 
> wasn't even expecting to get an answer today).
> 
> I know the max_contacts parameter. But the problem is that if I set the 
> max_contacts to 2, when the user tries to register again, he gets an error 
> message 503. The point is that I want to get the user registered, and replace 
> the oldest location entry for the newest one.
> 
> 
> 
> Bruno Emer
> 
> Mobile: +55 11 96540-0044  
> email: brunoe...@gmail.com 
> 
> 2016-04-01 16:31 GMT-03:00 Bruno Emer  >:
> Hello all.
> 
> I have problem here and I tried to find a solution and search over internet, 
> but without success.
> 
> My scenario is the following: I have an application that must be registered 
> in Kamailio when a user logs in the web interface, so he can get calls 
> (something like a web softphone using webrtc). At this point, we are OK, and 
> everything is working fine.
> 
> To get these register functions I am using the parameter "save("location", 
> "0x04")" as described in the REGISTRAR module documentation, so if a user 
> logs in another web browser or computer, only the last one will continue 
> registered and all calls will be forwarded to him.
> 
> The point is that now we are creating a phone app that will do almost the 
> same thing as the web interface, allowing users to receive calls using the 
> mobile device, and here is my problem: I want to allow my users to be 
> registered on two devices at the same time, but if a user logs into another 
> device, I don't want to reply with a 503. I want to allow the user to 
> register again, deregistering the oldest contact.
> 
> I saw that there is a module named "ims_usrloc_scscf" and on its description 
> it says "implemented overwrite oldest contact behaviour", but I couldn't find 
> any documentation about it.
> 
> So, is there a way to get this working today?
> 
> 
> Bruno Emer
> 
> Mobile: +55 11 96540-0044  
> email: brunoe...@gmail.com 
> 
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Re: [SR-Users] Multiple Registrations - Overwrite oldest contact

[Olle E. Johansson]

> On 01 Apr 2016, at 21:59, Olle E. Johansson <o...@edvina.net> wrote:
> 
> 
>> On 01 Apr 2016, at 21:31, Bruno Emer <brunoe...@gmail.com 
>> <mailto:brunoe...@gmail.com>> wrote:
>> 
>> Hello all.
>> 
>> I have problem here and I tried to find a solution and search over internet, 
>> but without success.
>> 
>> My scenario is the following: I have an application that must be registered 
>> in Kamailio when a user logs in the web interface, so he can get calls 
>> (something like a web softphone using webrtc). At this point, we are OK, and 
>> everything is working fine.
>> 
>> To get these register functions I am using the parameter "save("location", 
>> "0x04")" as described in the REGISTRAR module documentation, so if a user 
>> logs in another web browser or computer, only the last one will continue 
>> registered and all calls will be forwarded to him.
>> 
>> The point is that now we are creating a phone app that will do almost the 
>> same thing as the web interface, allowing users to receive calls using the 
>> mobile device, and here is my problem: I want to allow my users to be 
>> registered on two devices at the same time, but if a user logs into another 
>> device, I don't want to reply with a 503. I want to allow the user to 
>> register again, deregistering the oldest contact.
>> 
>> I saw that there is a module named "ims_usrloc_scscf" and on its description 
>> it says "implemented overwrite oldest contact behaviour", but I couldn't 
>> find any documentation about it.
>> 
>> So, is there a way to get this working today?
> 
> There is a modparam to register the amount of contacts per AOR - which would 
> do what you need.
Sorry - late evening… It’s within the normal registrar module, not in ims.
http://kamailio.org/docs/modules/4.4.x/modules/registrar.html#registrar.p.max_contacts

/O

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Re: [SR-Users] Multiple Registrations - Overwrite oldest contact

[Olle E. Johansson]

> On 01 Apr 2016, at 21:31, Bruno Emer  wrote:
> 
> Hello all.
> 
> I have problem here and I tried to find a solution and search over internet, 
> but without success.
> 
> My scenario is the following: I have an application that must be registered 
> in Kamailio when a user logs in the web interface, so he can get calls 
> (something like a web softphone using webrtc). At this point, we are OK, and 
> everything is working fine.
> 
> To get these register functions I am using the parameter "save("location", 
> "0x04")" as described in the REGISTRAR module documentation, so if a user 
> logs in another web browser or computer, only the last one will continue 
> registered and all calls will be forwarded to him.
> 
> The point is that now we are creating a phone app that will do almost the 
> same thing as the web interface, allowing users to receive calls using the 
> mobile device, and here is my problem: I want to allow my users to be 
> registered on two devices at the same time, but if a user logs into another 
> device, I don't want to reply with a 503. I want to allow the user to 
> register again, deregistering the oldest contact.
> 
> I saw that there is a module named "ims_usrloc_scscf" and on its description 
> it says "implemented overwrite oldest contact behaviour", but I couldn't find 
> any documentation about it.
> 
> So, is there a way to get this working today?

There is a modparam to register the amount of contacts per AOR - which would do 
what you need.

/O

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Re: [SR-Users] change the to_tag on send_reply() or sl_send_reply()

[Olle E. Johansson]

> On 01 Mar 2016, at 16:29, Uri Shacked  wrote:
> 
> Hi,
> 
> Is i possible to control the to_tag when i send my own stateless reply from 
> kamailio.

Why would you want to do that? Curious.

I did however lack the capability of accessing the Kamailioi defined to-tag in 
the scripts. Maybe we should make
it available in a pseudovariable somewhere.

/O



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Re: [SR-Users] db_cluster.so module and kamctlrc DBHOST= field

[Olle E. Johansson]

> On 15 Feb 2016, at 22:32, Derek Bolichowski  wrote:
> 
> Hi there,
>  
> New user to Kamailio here. We currently have it up and running in a 
> virtualized environment with 1 Kamailio sever, 1 Asterisk server and 1 MySQL 
> server.
>  
> I’m currently writing install scripts to make deploying new nodes/servers 
> easy and to keep settings the same across the board.  I’ve chosed to load the 
> db_cluster.so module in kamailio.cfg, as we will have 2x MySQL servers in 
> master-master replication which will contain the ‘kamailio’ and ‘asterisk’ 
> tables.
>  
> I’ve just hit a stumbling block – in `kamctlrc`, there is a field called 
> `DBHOST=`.  How can I reference my cluster here?
>  
> In kamailio.cfg, I simply define DBURL as “cluster//”.  What is 
> the syntax for ‘DBHOST=’ in ‘kamctlrc’?  Can I reference the cluster? Can I 
> have 2 separate DBHOST= lines?
>  
> Looking for some guidance on this one.

kamctlrc controls Kamctl that is an outside script that one can decide to use 
to manage Kamailio. It has it’s own set of database connection methods and 
can’t use the cluster module in Kamailio.

There are two cases where kamctl and kamdbctl (the sister script) use database 
- to create the actual database and to search/update. With a master/master you 
decide if you create the tables on one side and let mysql replicate or if you 
create it on both sides before you initialize replication.

For other queries you can choose to simply refer to one of the servers in 
kamctlrc - it’s usually
not mission critical. If that server is down, your query will fail but the 
Kamailio service will run thanks to cluster logic in the db_cluster module.

If you want a similar functionality for kamctl you may want to install the 
mysql proxy that can do failover in a way similar to db_cluster. 

Cheers,
/O



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Re: [SR-Users] Reg: REGISTER request don't have username in request line

[Olle E. Johansson]

> On 11 Feb 2016, at 13:22,  
>  wrote:
> 
> One more thing observed from logs here is REGISTER request don’t have 
> username (username is not there in request line), 
> 

Please check RFC 3261. A register r-uri is a domain or a server, not a username 
based URI.

/O

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Re: [SR-Users] Kamailio Social Event at Fosdem 2016

[Olle E. Johansson]

> On 27 Jan 2016, at 10:47, Daniel-Constantin Mierla  wrote:
> 
> Hello,
> 
> only few days left, so we need to nail the location down for Kamailio
> dinner at Fosdem.
Which day do you suggest?
> 
> One suggestion in walking distance is:
> 
>   - http://restaurant-italien-picotin-bruxelles.be/en
> 
> I understood that it was used in the past also for Fosdem Speaker's
> Dinner, so it might be busy and we can't get the seats. So new
> suggestions are more than welcome in order to have some backups to try!
> 
> If there is no strong preference for something else by end of today,
> tomorrow we should attempt to do the reservation at this place.
/O
> 
> Cheers,
> Daniel
> 
> -- 
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Book: SIP Routing With Kamailio - http://www.asipto.com
> http://miconda.eu
> 
> 
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Re: [SR-Users] [sr-dev] RFC: removing (disabling) fork=no

[Olle E. Johansson]

> On 20 Jan 2016, at 09:19, Daniel-Constantin Mierla  wrote:
> 
> Hello,
> 
> wondering if anyone is using fork=no -- some old docs suggest it is
> suitable for debugging, but actually kamailio doesn't work properly in
> this mode, leading to more troubles than benefits (e.g., having reports
> of invalid issues, like tcp not working in this mode).
> 
> In first phase I would disable setting this value, with a warning if set
> to no, because most of the configs out there have fork=yes. Removing it
> could be considered in the future.
> 
> Note that this fork=no is different than don't daemonize controlled with
> -D, which will stay being useful for some init.d systems.
What is the difference?

I have been using fork=no a lot in test scripts, but could possibly move to
-D. Having a config file parameter is easier though, said the lazy man.

/O
> 
> Comments or other suggestions?
> 
> Cheers,
> Daniel
> 
> -- 
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Book: SIP Routing With Kamailio - http://www.asipto.com
> http://miconda.eu
> 
> 
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Re: [SR-Users] Configure Kamailio to generate OPTIONS

[Olle E. Johansson]

> On 28 Dec 2015, at 09:51, Mititelu Stefan  wrote:
> 
> Hello,
> 
> What is the simplest way to configure Kamailio to generate periodical OPTIONS 
> to the caller, for an established dialog?
> 
> Can you give me some starting references?
> 
> 
To be SIP correct, a proxy may not insert any transactions in a dialog between 
two endpoints. The endpoints have the cseq counter and inserting anything will 
mess with the CSEQ.

Having said that, the dialog module has a keep-alive function that does this. 
It may generate strange error messages which is fine. You want to test the 
behaviour on a per-device basis.

Cheers,
/O___
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Re: [SR-Users] crash on using http_query when response is 200

[Olle E. Johansson]

Please post this on the issue tracker. I also get crashes, but they’re not like 
this. 

/Olle

> On 07 Dec 2015, at 15:54, Jayesh Nambiar  wrote:
> 
> Hi,
> I'm using http_query to send a post request on a https URL. The weird part is 
> it crashes with segmentation fault when the response from the server is 200. 
> If the response was a 500, it works and continues processing.
> Here's the function:
> http_query("https://notify.abcd.com/onnet/call 
> ", "$avp(devices)", "Content-Type: 
> application/json" "$var(response)");
> xlog("L_INFO", "Response is $var(response)");
> 
> And here's the backtrace on crash:
> 0x7f62f2bf8e38 in ?? ()
> #1  0x7f62ea22251b in http_query (_m=0x7f62f2c3a038, _url=0x7f62f2bf3548 
> "\330\064\277\362b\177", _dst=0x7f62f2bf9250 "\030\215\277\362b\177", 
> _post=0x7f62f2bf35c8 "\200r\277\362b\177", 
> _hdr=0x7f62f2bf91d0 "h\221\277\362b\177") at functions.c:226
> #2  0x7f62ea229e0b in w_http_query_post_hdr (_m=0x7f62f2c3a038, 
> _url=0x7f62f2bf3548 "\330\064\277\362b\177", _post=0x7f62f2bf35c8 
> "\200r\277\362b\177", _hdr=0x7f62f2bf91d0 "h\221\277\362b\177", 
> _result=0x7f62f2bf9250 "\030\215\277\362b\177") at utils.c:483
> #3  0x0041f6b5 in do_action (h=0x7ffeb128c8c0, a=0x7f62f2bf8a58, 
> msg=0x7f62f2c3a038) at action.c:1079
> #4  0x0042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2bf6fc0, 
> msg=0x7f62f2c3a038) at action.c:1549
> #5  0x0041f3ce in do_action (h=0x7ffeb128c8c0, a=0x7f62f2bfb9b0, 
> msg=0x7f62f2c3a038) at action.c:1045
> #6  0x0042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2bb9388, 
> msg=0x7f62f2c3a038) at action.c:1549
> #7  0x0041bc28 in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b8f248, 
> msg=0x7f62f2c3a038) at action.c:678
> #8  0x0042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b8e268, 
> msg=0x7f62f2c3a038) at action.c:1549
> #9  0x0041f3ce in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b91040, 
> msg=0x7f62f2c3a038) at action.c:1045
> #10 0x0042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b8b7a8, 
> msg=0x7f62f2c3a038) at action.c:1549
> #11 0x0041bc28 in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b3a2c0, 
> msg=0x7f62f2c3a038) at action.c:678
> #12 0x0042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b3a2c0, 
> msg=0x7f62f2c3a038) at action.c:1549
> #13 0x0041f3ce in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b47f00, 
> msg=0x7f62f2c3a038) at action.c:1045
> #14 0x0042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b37e80, 
> msg=0x7f62f2c3a038) at action.c:1549
> #15 0x0041bc28 in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b1c9c8, 
> msg=0x7f62f2c3a038) at action.c:678
> #16 0x0042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b1c9c8, 
> msg=0x7f62f2c3a038) at action.c:1549
> #17 0x0041f3ce in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b1cc28, 
> msg=0x7f62f2c3a038) at action.c:1045
> #18 0x0042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b170a0, 
> msg=0x7f62f2c3a038) at action.c:1549
> #19 0x0042ca93 in run_top_route (a=0x7f62f2b170a0, 
> msg=0x7f62f2c3a038, c=0x0) at action.c:1635
> #20 0x0051c115 in receive_msg (
> buf=0xab7520  "INVITE 
> sip:bdbbf60001b5b91...@devsip.abcd.com:8321 
>  
> SIP/2.0\r\nRecord-Route: 
> \r\nRecord-Route: 
> \r\nVia: SIP/2.0/UDP"..., 
> len=1901, rcv_info=0x7ffeb128cc10) at receive.c:196
> #21 0x0062d427 in udp_rcv_loop () at udp_server.c:495
> #22 0x004b28c1 in main_loop () at main.c:1593
> #23 0x004b9aa5 in main (argc=5, argv=0x7ffeb128d118) at main.c:2597
> 
> Can someone please confirm the crash so that I'll go ahead and report this as 
> an issue on the tracker. Basically when it crashes, kamailio doesn't even 
> execute the xlog function after the http_query call.
> P.S. abcd.com  is a changed domain for confidential reasons.
> 
> Thanks,
> 
> - Jayesh
> 
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Re: [SR-Users] SNMPstats dialog 0

[Olle E. Johansson]

Must be something in how the SNMP module reads the data from the statistics 
counters.
Please open a bug report and we’ll look into this.

/O

> On 30 Nov 2015, at 10:25, Igor Potjevlesch  wrote:
> 
> Hello,
> 
> Do you know what could be the solution?
> 
> Thanks,
> 
> Regards,
> 
> 2015-11-25 10:14 GMT+01:00 Igor Potjevlesch  >:
> Hello,
> 
> # kamctl stats dialog
> dialog:active_dialogs = 498
> dialog:early_dialogs = 57
> dialog:expired_dialogs = 24
> dialog:failed_dialogs = 73222
> dialog:processed_dialogs = 539449
> 
> Regards,
> 
> 2015-11-25 10:04 GMT+01:00 Daniel-Constantin Mierla  >:
> Hello,
> 
> if you do 'kamctl stats dialog', what is printed?
> 
> Cheers,
> Daniel
> 
> 
> On 25/11/15 09:46, Igor Potjevlesch wrote:
>> Hello,
>> 
>> I have rebooted my kamailio machine and now the value of kamailio dialogs is 
>> always 0:
>> 
>> KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0
>> KAMAILIO-MIB::kamailioCurNumDialogsInProgress.0 = Gauge32: 0
>> KAMAILIO-MIB::kamailioCurNumDialogsInSetup.0 = Gauge32: 0
>> KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 = Counter32: 0
>> 
>> And also the status is:
>> 
>> KAMAILIO-MIB::kamailioDialogUsageState.0 = INTEGER: idle(0)
>> 
>> How can I change it to active(1)? I have dialog.so loaded as well as 
>> modparam("dialog", "dlg_flag", 4) and dlg_manage().
>> 
>> Thank you.
>> 
>> Regards,
>> 
>> 
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>> 
> 
> -- 
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda  - 
> http://www.linkedin.com/in/miconda 
> Book: SIP Routing With Kamailio - http://www.asipto.com 
> 
> Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat 
> 
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> 
> 
> 
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Re: [SR-Users] q-value on unsupported clients

[Olle E. Johansson]

> On 05 Nov 2015, at 10:15, Daniel Tryba  wrote:
> 
>> Based on the xlog output, the following doesn't appear to be working -
>> remove_hf("Contact");
>> append_hf("Contact: $var(newct)\r\n");
>> 
>> I've actually had a similar issue before replacing a hdr so I'm wondering if
>> I'm doing something wrong?
> 
> Header manipulation is doesn't update the already parsed message that is 
> being 
> used for $hdr etc.
This is an important fact for all routing scripts. You have one message in 
memory that you
read from and one copy that you apply changes on. The changes are applied when 
you send
the message or when you run the function below.

> You have to apply the changes with
> http://kamailio.org/docs/modules/stable/modules/textopsx.html#textopsx.f.msg_apply_changes

Not that if you run this function you owerwrite the incoming message and have 
no way of going
back to the original incoming message. 

/O
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Re: [SR-Users] Avoid multiple registration

[Olle E. Johansson]

> On 03 Nov 2015, at 11:29, Marino Mileti  wrote:
> 
> I would like to avoid multiple registration with the same AOR...
> I know that there's a parameter for this (max_contacts) but if I set it to 1, 
> all REGISTER overwrite location info.
>  
> Instead i would like to reply with error if a REGISTER arrive with the same 
> AOR just registered on location server.
>  
> Is that possible? Any help is appreciated J
>  
A simple glance at the documentation page for the REGISTRAR shows me a function 
called “registred”.
http://kamailio.org/docs/modules/4.3.x/modules/registrar.html#registrar.f.registered

"The function returns true if the AOR in the URI is registered, false 
otherwise. “

If you use that one and refuse a registration if true, you will get exactly 
what you want.

Reading available documentation is a good thing (TM) :-)

Regards,
/Olle___
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Re: [SR-Users] Implementation of RFC 5393

[Olle E. Johansson]

> On 21 Oct 2015, at 14:51, Guillaume  wrote:
> 
> But why don't you implement this feature after your demo at kamailio world? 
> Do you think it's useless at the end?
I have it implemented, but in a routing script. It’s not useless, but not the 
full function. You need to be able to reuse 
breadth and a script can’t easily do that - only TM knows if there’s a branch 
failure and can restart another branch that was previously blocked. Move to 
kind of semi-serial forking based on available resources instead of going 
parallell.
> 
> And how your script was working with kamailio ?
Just fine :-)
But I guess you want another answer. I calculated the number of branches in 
each fork and added the required headers when sending downstream. Without it 
Kamailio would eat up my laptop and eventually explode and crash.
With it, a lot of branches was blocked and the network (and my laptop) saved.

I will have to dig up the scripts, written on the flights back home from SIPit, 
to be able to remember exactly how I did it.

I do believe we will have to do something to TM so that TM knows the allowed 
number of branches and keeps control of it. We have some hooks for branch 
failures now that may be used to improve my script - so it may be easier to get 
it done properly without source code changes today.

Max-breadth is critical to avoid flooding of a network  when forking.

/O
> 
> 
> Thanks for your response
> 
> 
> Guillaume
> 
> From: o...@edvina.net 
> Date: Wed, 21 Oct 2015 14:15:43 +0200
> To: mico...@gmail.com 
> CC: sr-users@lists.sip-router.org 
> Subject: Re: [SR-Users] Implementation of RFC 5393
> 
> 
> On 21 Oct 2015, at 14:09, Daniel-Constantin Mierla  > wrote:
> 
> Hello,
> 
> checking the IP in the Via headers can be done in config file using a while 
> loop:
> 
> $var(i) = 0;
> 
> while($(hdr(Via)[$var(i)])!=$null) {
># use transformations to extract the IP in $(hdr(Via)[$var(i)]) and test 
> it against $Ri
>...
>$var(i) = $var(i)  + 1;
> }
> 
> Also, checking the max-breadth should be possible in config file -- iirc, 
> Olle played with it at one of the SIPit events I attended, maybe he can add 
> more details here. I haven't read the RFC 5393 to be able to provide an 
> example here.
> I have a kind-of working solution in script, that I used in the Dangerous 
> Demos at kamailio world.
> 
> 
> If someone wants to add a module to simplify the config, he/she is welcome to 
> do it.
> :-)
> 
> I think it needs to have hooks into tm.
> 
> /O
> 
> Cheers,
> Daniel
> 
> On 21/10/15 10:35, Guillaume wrote:
> Hi guys,
> 
> What do you think about the RFC 5393 on loop detection and amplification 
> attack protection? 
> 
> The RFC is short and still a proposed standard but don't you think it could 
> be useful to prevent loop and amplification attack? Because even if the 
> max-forward field reduces the loop to ~70 hosts (in most cases) with some 
> techniques we could fork the message up to 2^70 messages (as described in the 
> RFC) to crash the servers.
> 
> Basically the server has to do 2 things:
> * check if it is not already in the via of the message
> * the previous check is not enough as a B2BUA could have replace the via 
> headers, so the RFC introduces a new field called max-breadth to limit the 
> forking.
> 
> I have not seen a lot of implementation of this RFC on the free SIP software 
> and I think it could be a good way to improve kamailio making a module for it 
> (the easier way to implement this feature I think).
> 
> In fact I'm in a research internship about VoIP security and I have time to 
> develop such a module for kamailio if you think it's a good idea (I'm looking 
> for some security improvements in free software solutions so if you have 
> other idea don't hesitate to tell me).
> 
> Cheers,
> 
> 
> Tetram
> 
> 
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> -- 
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> http://twitter.com/#!/miconda  - 
> http://www.linkedin.com/in/miconda 
> Book: SIP Routing With Kamailio - http://www.asipto.com 
> 
> 
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Re: [SR-Users] Sending ReINVITE from Kamailio

[Olle E. Johansson]

> On 21 Oct 2015, at 08:31, Grant Bagdasarian  wrote:
> 
> Hello,
>  
> Is it possible to have Kamailio send a ReINVITE every X minutes to determine 
> if a session is still active?
> I know it’s a proxy and doesn’t have B2BUA capabilities, but there was a 
> module which allowed Kamailio to generate SIP messages, but I can’t find it 
> anymore.
> If Kamailio is not the place to do this, which component in the voip network 
> should be responsible for this? Are there perhaps other ways to poll for 
> session state in Kamailio?

The dialog module has in-dialog keepalives. Not Re-INVITE, but at least a 
message. 

The best way is to use SIP Session Timers in both or at least one user agent.

/O

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Re: [SR-Users] Implementation of RFC 5393

[Olle E. Johansson]

> On 21 Oct 2015, at 14:09, Daniel-Constantin Mierla  wrote:
> 
> Hello,
> 
> checking the IP in the Via headers can be done in config file using a while 
> loop:
> 
> $var(i) = 0;
> 
> while($(hdr(Via)[$var(i)])!=$null) {
># use transformations to extract the IP in $(hdr(Via)[$var(i)]) and test 
> it against $Ri
>...
>$var(i) = $var(i)  + 1;
> }
> 
> Also, checking the max-breadth should be possible in config file -- iirc, 
> Olle played with it at one of the SIPit events I attended, maybe he can add 
> more details here. I haven't read the RFC 5393 to be able to provide an 
> example here.
I have a kind-of working solution in script, that I used in the Dangerous Demos 
at kamailio world.

> 
> If someone wants to add a module to simplify the config, he/she is welcome to 
> do it.
:-)

I think it needs to have hooks into tm.

/O
> 
> Cheers,
> Daniel
> 
> On 21/10/15 10:35, Guillaume wrote:
>> Hi guys,
>> 
>> What do you think about the RFC 5393 on loop detection and amplification 
>> attack protection? 
>> 
>> The RFC is short and still a proposed standard but don't you think it could 
>> be useful to prevent loop and amplification attack? Because even if the 
>> max-forward field reduces the loop to ~70 hosts (in most cases) with some 
>> techniques we could fork the message up to 2^70 messages (as described in 
>> the RFC) to crash the servers.
>> 
>> Basically the server has to do 2 things:
>> * check if it is not already in the via of the message
>> * the previous check is not enough as a B2BUA could have replace the via 
>> headers, so the RFC introduces a new field called max-breadth to limit the 
>> forking.
>> 
>> I have not seen a lot of implementation of this RFC on the free SIP software 
>> and I think it could be a good way to improve kamailio making a module for 
>> it (the easier way to implement this feature I think).
>> 
>> In fact I'm in a research internship about VoIP security and I have time to 
>> develop such a module for kamailio if you think it's a good idea (I'm 
>> looking for some security improvements in free software solutions so if you 
>> have other idea don't hesitate to tell me).
>> 
>> Cheers,
>> 
>> 
>> Tetram
>> 
>> 
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>> 
> 
> -- 
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda  - 
> http://www.linkedin.com/in/miconda 
> Book: SIP Routing With Kamailio - http://www.asipto.com 
> 
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Re: [SR-Users] Sending ReINVITE from Kamailio

[Olle E. Johansson]

> On 21 Oct 2015, at 09:27, ycaner  wrote:
> 
> Hello;
> I think Dialog module can do it with ka_timer. take a look please.
> in addition , if you want to know call is still up , check the RTP session.
> if there isn't Rtp  session , so call is  hung up. Asterisk can listen rtp
> packet and then in silence it can close session. 
> 
> have a look "rtptimeout" parameter
> 
This doesn’t always apply either - if the call is on hold there’s no RTP
but it should not be hung up. Asterisk handles this, but for other
proxys it’s hard to know the state of the media session.

/O
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Re: [SR-Users] kamailio cfg file with user db authentication and rtpproxy

[Olle E. Johansson]

> On 18 Oct 2015, at 12:53, amar Smart Telecom  wrote:
> 
> Dear all,
>  
> Can you pls mail me a working Kamailio cfg file which has:
> “Sip user db authentication” and “rtpproxy for all calls”
>  
You can look for consultants that will do this for pay on the web site or ask 
on the business mailing list.
In the default configuration example, I believe you have just this, but if you 
need help these 
are the resources available.

You will find both the business mailing list and business consultants on this 
link:
http://www.kamailio.org/w/business/

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Re: [SR-Users] Familiar topic about Kamailio with Asterisk behind NAT

[Olle E. Johansson]

> On 30 Sep 2015, at 11:10, Daniel Tryba  wrote:
> 
> On Wednesday 30 September 2015 10:22:51 Dirk Teurlings - SIGNET B.V. wrote:
>> CLIENTS <-> (NAT) <-> INTERNET <-> KAMAILIO(4.2.5) with 
>> RTPPROXY(v1) <-> PRIVATE LAN <-> ASTERISK (v1.8)
> 
>> I'm kind of stuck as to where I need to fix this. I tried using the 
>> externaddr option in Asterisk to solve it on that end. But that didn't 
>> help anything. The NAT options in Kamailio are not really suited for 
>> this, as they tend to fix client NAT problems.
> 
> The asterisk doesn't need to know the network topology beyond its connection 
> with kamailio, so you need to fix it at the place of traversal between public 
> and private: kamailio.
> 

Actually, if you turn on NAT support in Asterisk you should not need RTPproxy 
between clients and Asterisk. Only if two clients talk with each other without 
Asterisk RTPproxy is needed in the call. Adding RTPproxy in order to be able to 
move Asterisk to the private LAN seems like a good way to add latency to the 
call, which is usually not what you want. If Asterisk needs to be there for 
other reasons, you will need to check that the internal addresses are not sent 
out on the other side of Kamailio in the signalling.

/O



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Re: [SR-Users] Wrong Call-ID and tags in response to OPTIONS

[Olle E. Johansson]

 On 10 Aug 2015, at 11:19, Jean-Marie Baran jean-marie.ba...@ama.bzh wrote:
 
 By the way, is it normal that I had to change the code to have Kamailio 
 accept OPTIONS messages, or did I miss a config somewhere ?
It’s the normal way. You are in full control of handling of SIP messages - 
requests and responses. Nothing should happen magically in the background.

/O___
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Re: [SR-Users] VIA header question.

[Olle E. Johansson]

 On 04 Aug 2015, at 13:14, Chad ccolu...@hotmail.com wrote:
 
 Hi list,
 I need a little help, I am a business owner trying to get Kamailio up and 
 running as a SIP load balancer.
 I hired a Kamailio consultant to help me do so, but Kamailio is not working 
 and I am getting conflicting information.
 
 My Kamailio consultant and my VOIP provider are telling me 2 different things 
 and I don't know which one is right.
 
 Kamailio sends SIP traffic to the VOIP provider with 2 VIA headers like this 
 (in this order):
 Via: SIP/2.0/UDP 
 10.10.10.254;branch=z9hG4bK7291.6a0bbd2e8fd639a47d7d2de606779e47.0.
 Via: SIP/2.0/UDP 
 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060.
 
 The VOIP provider says that is incorrect because they are supposed to reply 
 back to the topmost VIA header so they reply to the 10.10.10.254 IP (which is 
 not public) and the call ends.
 The VOIP provider says Kamailio should send the VIA headers like this instead:
 Via: SIP/2.0/UDP 
 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060.
 Via: SIP/2.0/UDP 
 10.10.10.254;branch=z9hG4bK7291.6a0bbd2e8fd639a47d7d2de606779e47.0.
 
 My Kamailio consultant says the way we are sending it is right and that the 
 VOIP provider is processing the call incorrectly.
 
 I read that the SIP proxy is supposed to remove the internal header from the 
 1st example above based on this RFC:
 https://tools.ietf.org/html/rfc3261#section-16.7
 Item: 3. Via
 The proxy removes the topmost Via header field value from the response.”
That applies to response forwarding.
 
 If that applies to this situation (which I don't know if it does) then 
 Kamailio should be removing the 10.10.10.254 VIA line and only sending 1 VIA 
 header like this:
 Via: SIP/2.0/UDP 
 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060.
 
 Which would sort of make the VOIP provider right in that the topmost VIA line 
 would then be the external IP, but how they said to fix it (reversing the VIA 
 lines) is wrong.
The response is sent to the topmost, leftmost VIA header value. The proxy adds 
it’s own address on TOP of other values when forwarding a request. That’s the 
address used for sending responses. 
See section 18.2.2 “Sending Responses” of RFC 3261.

What I don’t understand is how the via’s got in a different order. Kamailio 
does this right by default. It’s a very basic operation, but could be something 
related to the handling of public/private IP addresses that got wrong.

/O
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Re: [SR-Users] Dispatcher module number of calls

[Olle E. Johansson]

The best way is to let the server you send the calls to decide when it has too 
many calls. It has all the states
needed and should be able to block the call with an error response, preferrably 
in the 5xx range. 

Asterisk chan_sip will do this if you set maxcalls in asterisk.conf. I don’t 
remember the error code used and won’t check right now, as I’m on holiday :-)

Having the same state in two different servers is looking for trouble.

/O

 On 23 Jul 2015, at 11:16, Alberto Sagredo alberto.sagr...@avanzada7.com 
 wrote:
 
 Sorry :)
 
 I found two scripts, the one on other mail worked fine
 
 per user limit is this one
 
 https://www.mail-archive.com/sr-users@lists.sip-router.org/msg07072.html 
 https://www.mail-archive.com/sr-users@lists.sip-router.org/msg07072.html
 
 Best Regards :)
 
 Any comment to improve it is welcome
 
 2015-07-23 9:48 GMT+02:00 Alberto Sagredo alberto.sagr...@avanzada7.com 
 mailto:alberto.sagr...@avanzada7.com:
 Hi
 
 Hi have read documentation but it seems dispatcher does not keep how many 
 calls has been dispatched or currently are in any of dispatcher destinations
 
 I have take a look to code on:
 
 http://lists.sip-router.org/pipermail/sr-users/2012-July/073919.html 
 http://lists.sip-router.org/pipermail/sr-users/2012-July/073919.html
 
 But it seems to use calls limit per user, 
 
 I would need to establish a limit per dispatcher destination i order to do 
 not send more calls.
 
 Any idea where to look for?
 
 I have take a look to DMQ module, that maybe i could to use to spread 
 information of how many calls to every destination but do not know how to 
 extract calls from kamailio that have been dispatched to this destination 
 using disptcher module.
 
 Best Regards
 
 Alberto
 
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Re: [SR-Users] Nat Detect does not work with GS Phones

[Olle E. Johansson]

 On 14 Jul 2015, at 11:56, Alberto Sagredo alberto.sagr...@avanzada7.com 
 wrote:
 
 I have found an issue detecting NAT of GS 1.4.23 Phones when using Stun on 
 this phones
If they activate STUN they signal with a public IP. If they signal with a 
public IP, they tell us they 
can handle NAT by themself and require no help with NAT traversal from 
Kamailio. 

Turn off STUN to get help from Kamailio.

/O
 
 Usual nat_uac_test with numbers 19 and 3 does not seems to detect is behind 
 nat so NATMANAGE is not called
 
 
 Here its some trace. ANy clue how to handle this phones if they activate STUN 
 for example
 
 U 80.26.x.x:52768 - 192.168.0.170:8002 http://192.168.0.170:8002/
 INVITE sip:2@x.x.x.x:8002 SIP/2.0.
 
 Via: SIP/2.0/UDP 80.26.x.x:52768;branch=z9hG4bK358742535;rport.
 
 From: Anonymous sip:anonymous@anonymous.invalid;tag=147856.
 
 To: sip:2@x.x.x.x:8002.
 
 Call-ID: 244257786-52768...@ia.cg.bie.bch.
 
 CSeq: 550 INVITE.
 
 Contact: Anonymous sip:212@80.26.x.x:52768.
 
 X-Grandstream-PBX: true.
 
 Max-Forwards: 70.
 
 User-Agent: Grandstream GXP2140 1.0.4.23.
 
 Privacy: id.
 
 P-Preferred-Identity: sip:212@x.x.x.x:8002.
 
 Supported: replaces, path, timer.
 
 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, 
 UPDATE, MESSAGE.
 
 Content-Type: application/sdp.
 
 Accept: application/sdp, application/dtmf-relay.
 
 Content-Length:   335.
 
 .
 
 v=0.
 
 o=212 8000 8000 IN IP4 80.26.x.x.
 
 s=SIP Call.
 
 c=IN IP4 80.26.x.x.
 
 t=0 0.
 
 m=audio 55422 RTP/AVP 0 8 18 9 2 101.
 
 a=sendrecv.
 
 a=rtpmap:0 PCMU/8000.
 
 a=ptime:20.
 
 a=rtpmap:8 PCMA/8000.
 
 a=rtpmap:18 G729/8000.
 
 a=fmtp:18 annexb=no.
 
 a=rtpmap:9 G722/8000.
 
 a=rtpmap:2 G726-32/8000.
 
 a=rtpmap:101 telephone-event/8000.
 
 a=fmtp:101 0-15.
 
 
 # Caller NAT detection route
 
 route[NATDETECT] {
 
 #!ifdef WITH_NAT
 
 force_rport();
 
 if (nat_uac_test(19)) {
 
 if (is_method(REGISTER)) {
 
 fix_nated_register();
 
 } else {
 
 fix_nated_contact();
 
 }
 
 setflag(FLT_NATS);
 
 }
 
 #!endif
 
 return;
 
 }
 
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Re: [SR-Users] Kamailio multiple registration

[Olle E. Johansson]

I quite frequently use good ol' Asterisk to generate - and maintain - a large 
number of registrations.

This is quite useful to be able to test overflow handling and other peaks in 
traffic.

/O

On 06 Jul 2015, at 09:31, Alberto Sagredo alberto.sagr...@avanzada7.com wrote:

 Hi Fadi
 
 As Daniel and Loic commented .
 
 Create users using kamctl or adding with a script to database mysql (if you 
 are using it) and you could use SIPP 
 
 Take a look to : 
 http://www.sipfish.com/blog/generating-voip-traffic-with-sipp/
 
 BR
 
 2015-07-06 9:28 GMT+02:00 Loic Chabert chabert.loic...@gmail.com:
 Hello,
 
 Yes you can use Sipp and a template for registration. You can find a list of 
 scenario on : https://github.com/saghul/sipp-scenarios
 
 Just launch the command with a CSV (as parameter) and your user/password 
 inside.
 
 Regards,
 Loic. 
 
 2015-07-06 8:48 GMT+02:00 Daniel-Constantin Mierla mico...@gmail.com:
 Hello,
 
 perhaps you can use tools like sipp or sipsak.
 
 There is also a command 'kamctl ul add ...'.
 
 Cheers,
 Daniel
 
 On 04/07/15 22:45, Fadi Hawari wrote:
  Dear Sir ,
  i need to register 10k users in sequance could you please let me know how 
  to do it .
  Regards
 
 
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 --
 Daniel-Constantin Mierla
 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
 Book: SIP Routing With Kamailio - http://www.asipto.com
 
 
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Re: [SR-Users] modules.lst

[Olle E. Johansson]

On 05 Jul 2015, at 21:24, Joseph Zimmer webproductions@gmail.com wrote:

 /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 71: mysql: command not 
 found

If you haven't got mysql installed, it's hard to create a database.

/O___
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Re: [SR-Users] Measuring subscriber latency

[Olle E. Johansson]

On 29 Apr 2015, at 10:04, Daniel-Constantin Mierla mico...@gmail.com wrote:

 Hello,
 
 On 28/04/15 20:42, Jon Bonilla (Manwe) wrote:
 Hi all
 
 I'm replacing an Asterisk based system with a kamailio based one. One of the
 features the legacy system has is showing the subscriber the latency obtained
 from the qualify option of sip.conf
 
 Now, I'd like to measure the latency but I'm not sure how to do it. AFAIK
 nathelper module sends the OPTIONS keepalive messages stateless mode and
 there's no information there. 
 
 I was thinking on triggering a route send_options via the timer module, save
 the timestamp of the relay and the timestamp of the response in onreply_route
 but it doesn't look elegant. Creating my own daemon in an external server and
 reading the info from the location module seems to be another option.
 
 I guess I'm not the first one with this need so I wonder if there's an 
 already
 existing solution or an elegant way of dealing with it. 
 
 
 any ideas?
 
 it looks like you are the first wanting this, or at least the first that
 has expressed it.
As Jon said, this is a feature that has been in Asterisk for a very long time 
and 
we need in Kamailio. I think many of us has looked for it, but never mailed 
about
it since we still have Asterisk in there. Since Kamailio has grown so much and
we now can build Asterisk-free solutions, I think this would be a valuable 
feature
both for dispatcher and for usrloc.

/O
 
 It might not be hard to code it in c, it will require to extend the
 usrloc structure to have with two timestamps, one to be set when sending
 the options and one when receiving the reply. At this moment there is
 only one timestamp when the reply is received, see the function
 ul_refresh_keepalive() from usrloc, the field in ucontact_t structure is
 last_keepalive. So half is done more or less.
 
 You can open a feature request on tracker and probably will get into
 4.4. The development for 4.3 is frozen.
 
 Cheers,
 Daniel
 
 -- 
 Daniel-Constantin Mierla
 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
 Kamailio World Conference, May 27-29, 2015
 Berlin, Germany - http://www.kamailioworld.com
 
 
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Re: [SR-Users] Shared location database used by independend kamailios

[Olle E. Johansson]

On 14 Apr 2015, at 17:06, Daniel Tryba d.tr...@pocos.nl wrote:

 Is there an easy way to figure out on which server a uac is registered when 
 using a shared database (modparam(usrloc, db_mode, 3)).
 
 When uac1 is registered on server1 (dns srv loadbalancing) and uac2 is on 
 server2. A call from uac1 to uac2 with a simple lookup(location) will 
 result 
 in server1 directly trying to connect to uac2. If this device is natted (or 
 behind a stateful firewall) this will fail.
 
 In the location table the column socket contains the server which received 
 the 
 request. So if I know this value I can have server2 route the call to this 
 value/server and have the call delivered to uac1 via an existing connection.
 
 The idea was to use reg_fetch_contacts to fetch this/these socket values in 
 $ulc and in case they are not local redirect/branch to those servers. But if 
 the value of socket isn't local the $ulc for socket is null.
 
 I could use sqlops to fetch this manually, but is there an easier way I am 
 missing?
 
Use the path header?

/O
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Re: [SR-Users] Kamailio LDAP authentication

[Olle E. Johansson]

On 10 Apr 2015, at 09:23, Marek Moravcik marekmorav...@imafex.sk wrote:

 Hello,
 
 I'd like to authenticate Kamailio users in LDAP. But it looks like
 Kamailio need to download password from LDAP and authenticate
 user on it's own. Is there any possibility to send password to LDAP
 and let LDAP to say, if the user can be sign in?

For MD5 Digest challenge-response authentication the cleartext password is 
needed.
We do not get any cleartext password from the client, so the SIP auth server
needs to calculate a hash based on the nonce (the challenge), the authentication
realm and the secret. This hash is compared with the hash we get from the 
client.

This is a good reason to run LDAP over TLS.

/O
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Re: [SR-Users] [sr-dev] Announcement: Kamailio is now systemd-rtc-server

[Olle E. Johansson]

On 01 Apr 2015, at 10:31, Daniel-Constantin Mierla mico...@gmail.com wrote:

 Given it is a rather technical community around here, I would expected a
 bit of engineering approach when announcing the achievement. Shortly,
 here are some of the facts.
 
 Any interaction with or inside systemd is now *simple*, using the well
 know publish-subscribe-notify mechanism, glued with xcap. If you want to
 restart a daemon, you have to subscribe to its state, publish the fact
 you want to restart, and systemd will notify you if the operations is
 done or not according to permissions rules in xcap.
 
 Worth to mention that the real reason of forking linux kernel by systemd
 (see http://distrowatch.com/weekly.php?issue=20150330#community) is to
 *simplefy* it by migrating to publish-subscribe-notify-xcap for
 everything that requires real time interaction. Forget about the complex
 file permissions strange 3-4 digits which are not in e164 format, thus
 hard to remember! Do you want to read a file? Just subscribe to it, xcap
 knows who you are and what you can do or not, notifying you promptly
 with the content from the file or /dev/null.
In addition this also totally removes the need for tools like Icinga, Nagios 
and Monit.
If something happens, your systemd kernel will simply call you.

/O
 
 *Simplefying* everything is the future!
 
 Cheers,
 Daniel
 
 On 01/04/15 06:11, Alex Balashov wrote:
 For immediate release:
 
 ATLANTA, GA (1 April 2015)--Evariste Systems LLC, an Atlanta-based
 software
 vendor specialising in Kamailio-based service delivery solutions for the
 VoIP ITSP market, is pleased to announce that it, in collaboration with
 Red Hat Software and Ringfree Communications, has finalised the
 absorption of the Kamailio SIP Server into the 'systemd' system
 management
 platform for Linux. The new component shall be called
 'systemd-rtc-server',
 or 'Systemd Real-Time Communication Server'.
 
 Alex Balashov, principal of Evariste and leader of the tri-vendor
 collaboration effort, will officially announce the handover of the reigns
 of the Kamailio project to the personal leadership of Lennart Poettering
 at the upcoming Systemd Real Time Communications World conference, to be
 held in Berlin on 27-29 May of this year.
 
 John Knight, Director of GNOME 3 Integration and part-time usability
 consultant at Ringfree Communications, based in Hendersonville, North
 Carolina,was quick to summarise the triumphs of the long-standing
 integration effort.
 
 Remarked Knight:
 
 The industry has recognised for years that a SIP proxy is a basic
 building
 block in the 'init' subsystem of any Linux host. In this age of
 multimedia
 communication with voice and video, it was a travesty that systemd
 handled
 time synchronisation, network configuration, login management, logging,
 and console, but not SIP message routing.
 
 Sean McCord, a veteran partner at Atlanta-based integrator CyCORE 
 Docker,
 was quick to concur:
 
 SIP calls are much easier to troubleshoot with binary logs. Combined
 with packet captures of TLS-encrypted WebRTC calls, systemd-journald
 is the ultimate call setup troubleshooting methodology of the responsive,
 kinetic enterprise.
 
 To support the integration of Kamailio into the ecosystem of every major
 Linux distribution, Evariste has released new 'dbus_api' and 'pulseaudio'
 modules for the project.
 
 Balashov stated, We fully expect to use the D-Bus API to achieve
 gnome-session integration with systemd-rtc-server-usrloc, but we aren't
 going to leave Windows users behind; KamailioSvcHost.exe will support
 Domain Controller policies for G.722 in Active Directory forests.
 
 Despite an aggressive delivery timeline by the tri-vendor consortium
 behind
 systemd-rtc-server, industry commentators have widely lambasted the fact
 that it took so long for Kamailio to become integrated into systemd. Fred
 Posner, solutions architect at The Palner Group in Fort Lauderdale,
 Florida,
 recently wrote in a widely-publicised blog post:
 
 sr-dev have been keeping their heads in the sand for too long. For years
 now, it has been completely obvious and self-evident to anyone with half
 a brain that all kinds of VoIP software should be included in systemd.
 It's a basic building block of the whole OS, having absorbed
 functionality
 previously provided by all kinds of packages like util-linux and
 wireless-tools.
 
 John Knight of Ringfree accepted the criticism readily, but advocated a
 forward-thinking orientation focused on breaking with the uncertainty of
 the past:
 
 In the absence of a SIP component for routing calls to the PSTN, some
 people thought, 'systemd has no clear direction apart from the whims
 of its
 developers, and is a perpetually moving goal post.' Well, a SIP server
 should
 put an end to that whole discussion; that's exactly what was missing,
 and now
 that we have systemd-rtc-server, we've eliminated all doubts about the
 coherence, conceptual integrity and finality of systemd.
 
 

Re: [SR-Users] keep Kamailio running

[Olle E. Johansson]

On 31 Mar 2015, at 11:11, Mihaly Zachar zmih...@gmail.com wrote:

 Hi All,
 
 
 As far as I can see if a child dies due to segmentation fault Kamailio
 stops all the childs and it exits.
 
 Is there a possibility to change this behaviour to a kind of respawn
 the child way ?
 
 What do you think is the best way to keep Kamailio running on Ubuntu ?
 On RHEL ?
 

In my view a software that gets segementation faults needs to be fixed
so that it does not happen. Anything else is workarounds that may
hide the fact that you have a serious bug. 

Tell us more about the segfault and help us fix it.

Thank you,
/Olle

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Re: [SR-Users] Reload htable

[Olle E. Johansson]

On 27 Mar 2015, at 14:50, Alex Balashov abalas...@evaristesys.com wrote:

 This command is not exposed within the route script. This is a fairly common 
 situation with modules; some meta functionality like this that is available 
 externally (via MI, RPC, etc.) is not available within the route script.
http://kamailio.org/docs/modules/4.2.x/modules/jsonrpc-s.html#jsonrpc-s.f.jsonrpc_exec

In 4.2 the jsonrpc-s module introduced a way to execute RPC commands from 
within the routing script. Awsome!

Example 1.3. jsonrpc_exec usage

...
jsonrpc_exec({jsonrpc: 2.0, method: dispatcher.reload, id: 1}');
...



/O
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Re: [SR-Users] Kamailio BLF Issue TCP vs UDP

[Olle E. Johansson]

On 26 Mar 2015, at 23:33, Fred Posner f...@palner.com wrote:

 Subscription-State: terminated;reason=noresource.

THis header in the TCP example should tell you something. FreeSwitch is telling 
you that something has gone wrong with your subscription request.

/O
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Re: [SR-Users] kamailio asterisk NOTIFY

[Olle E. Johansson]

On 23 Mar 2015, at 23:42, Alex Balashov abalas...@evaristesys.com wrote:

 Anthony,
 
 The Contact presented by the subscriber in the initial subscription is:
 
   m: Test User 
 sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84OC82LK-2ehwl7NYDdNYDxNYlFUAm6UYzm0gme;+sip.ice
 
 And the subsequent NOTIFY is correctly targeted to this RURI:
 
   NOTIFY 
 sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84OC82LK-2ehwl7NYDdNYDxNYlFUAm6UYzm0gme
  SIP/2.0
 
 The subscriber returns this Contact in the 200 OK response for the NOTIFY:
 
   m: Test User 
 sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84O;+sip.ice
 
 And subsequent NOTIFYs appear to use this RURI instead of the original 
 Contact of the subscriber:
 
   NOTIFY 
 sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84O
  SIP/2.0
 
   NOTIFY 
 sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84O
  SIP/2.0
 
 etc.
 
 A SUBSCRIBE is a dialog-creating event, so in-dialog messages (e.g. NOTIFYs) 
 should be targeted at a Request URI corresponding to the Contact URI of the 
 SUBSCRIBE-er. It seems to me that this is the problem here, rather than 
 truncation.
 
I think that changed recently. The NOTIFY is a dialog-creating event, since a 
SUBSCRIBE can fork to many destinations.

/O


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Re: [SR-Users] What is the best SIP trunk authentication strategy

[Olle E. Johansson]

On 19 Mar 2015, at 18:38, canuck15 canuc...@hotmail.com wrote:

 It looks like auth_check() will work. It seems intelligent enough to scan all 
 instances of the same domain as long as the username is unique so that should 
 get things working.
 
 The problem here is that there is a fundamental difference between Asterisk 
 and Kamailio authentication.  Asterisk authentication works with FQDN or IP.  
 However, Kamailio is not designed to authenticate anything with FQDN unless 
 it is also a realm and identified as such by the UA.  I believe that is the 
 main issue here.  SIP trunks typically do not use or care about realm.  So 
 after the initial invite response from Kamailio the SIP trunk provider 
 typically responds with the IP address as the realm.
Asterisk authentication is kind of broken - it disregards the domain and is 
based on the user name or only use IP/port. Many years ago I worked on adding
multiple domain support in asterisk - part of the code is still there. Then the 
project leader added a huge patch for single-domain TLS and I gave up that
work. 

Kamailio is much more flexible. While the auth module only handles realm, you 
can easily connect the account to a set of specific From: SIP URI's and do a 
full authentication
and authorization scheme that works as you want. You can build in a number of 
ways - which makes it very mush more SIP-compliant and flexible.


 
 It does almost seem like there should be a special module to deal with this 
 sort of thing.  None of the existing modules seem to be the right fit.
Kamailio is a toolkit. Don't take a single module as the only solution. It's 
like linux, you combine a set of small functions and build solutions.
Very different from Asterisk. 

I don't think we need a new module. You can already build stuff like this by 
combining functionality in different modules.

/O


 
 
 On 3/18/2015 9:03 AM, Daniel Tryba wrote:
 On Wednesday 18 March 2015 08:32:10 canuck15 wrote:
 I can run a cron job every hour to DNS lookup and update the ip_addr
 table as needed so I think this is a satisfactory solution for IP
 authentication.
 Is there a mechanism to identify all originating servers for a
 hostname/domain? If the answer is no (and AFAIK is it) then this solution
 doesn't work.
 
 I used this in the past, a subscriber has a userpref with ip/port combo. But
 this ins't an answer for subaccounts on trunks (unles you can get the sender
 to actually use different ports). 3 is the whitelist for ip adresses on
 record. I abandoned this due to to much problems with trunks, they just have
 to authenticate or go elsewere.
 
 BTW only for tcp since udp sources can be spoofed. I guess the best way is to
 use tls with certificate verification (good luck getting the trunks to
 implement this :)
 
 route[AUTHENTICATE]
 {
 if(!is_method(REGISTER)  allow_address(3, $si, $sp) 
 $proto==tcp)
 {
 if(!avp_db_query(select username from usr_preferences where
 attribute='ip_authentication' and domain='$td' and (value='$si:$sp' or value
 like '$si:%') order by length(value) limit 1))
 {
 xlog(L_ALERT,ACL: $rm from $fu (IP:$si:$sp)\n);
 sl_send_reply(403, Not Allowed by AUTHENTICATE
 ACL);
 exit;
 }
 
 $avp(au)=$avp(i:1);
 }
 else
 {
 $var(authenticated)=www_authenticate($td, subscriber);
 
 if (!www_authenticate($td, subscriber)) {
 xlog(L_ALERT,AUTHENTICATE: $rm from $fu to $tu 
 (IP:
 $si:$sp)\n);
 www_challenge($td, 1);
 exit;
 }
 
 $avp(au)=$au;
 
 consume_credentials();
 }
 
 
 
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Re: [SR-Users] Hardware config to run 1000 Kamailio TLS users

[Olle E. Johansson]

This is an interesting discussion. 

   • on a 32-bit machine with 4GB of memory and with 2.5GB reserved for 
SIP server, the server could support 43 000 simultaneous TLS connections – 
consumed energy 209W

http://www.kamailio.org/w/2011/05/green-voip-energy-efficiency-and-performaces-of-v3-0/

That paper by long-time contributor Jan Janak is a good one to return to now 
and then.

/O
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Re: [SR-Users] PKG and Shared Memory statistics

[Olle E. Johansson]

On 16 Mar 2015, at 16:37, Dirk Teurlings Signet B.V. engineer...@signet.nl 
wrote:

 Hi,
 
 We'd like to be able to get statistics of the used amount of memory in 
 Kamailio.
 
 # kamctl stats shmem
 
 The above command only tells us what the shared memory is doing. Is there any 
 way to get the committed size of memory for each of the PKG_MEMORY pools?
 
 If not, could this be added at some point?

Use kamcmd to check and you will find both shared memory and package memory 
statistics.

Cheers,
/O
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Re: [SR-Users] Redirect server for load balancing and HA

[Olle E. Johansson]

On 13 Mar 2015, at 20:34, Daniel Tryba d.tr...@pocos.nl wrote:

 On Fri, Mar 13, 2015 at 08:16:50PM +0100, Olle E. Johansson wrote:
 Cumbersome, why not just use some custom headers (stored in 
 usr_preferences)?
 There are settings you can't set from the dialplan, but when you can this is 
 a good
 idea.
 
 With modern Asterisks is has been a long time since I needed anything
 that wasn't available from the dialplan (can't think of anything).
 
Good. Then we've done a good job with Asterisk. :-)

Thank you for the feedback.

/O
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Re: [SR-Users] Redirect server for load balancing and HA

[Olle E. Johansson]

On 13 Mar 2015, at 18:45, Daniel Tryba d.tr...@pocos.nl wrote:

 On Friday 13 March 2015 17:14:31 Olle E. Johansson wrote:
 I'd love to find out why this is the case. I would have estimated 9 out of
 10 people need to fine-tune DTMF + codec settings in Asterisk.
 The most common solution I've seen is to have kamailio use 10 different
 ports and have 10 different matching peers in the Asterisk SIP config to
 select settings.
 
 Cumbersome, why not just use some custom headers (stored in usr_preferences)?
There are settings you can't set from the dialplan, but when you can this is a 
good
idea.

/O
 
 kamailio stuff:
if($avp(src_dtmfmode)!=)
{
append_hf(X-DTMFMode: $avp(src_dtmfmode)\r\n);
}
 
 asterisk stuff:
 exten = _[+X].,n,ExecIf($[${LEN(${SIP_HEADER(X-DTMFMode)})}  
 0]?SIPDtmfMode(${SIP_HEADER(X-DTMFMode)}))
 
 The other way around is bit more tricky. YOu have to dial using the M option 
 (i'm using AGI instead of RT):
 M(setdtmftx^.$row['dtmftx'].)
 
 [macro-setdtmftx]
 exten = s,1,NoOp()
 exten = s,n,SIPDtmfMode(${ARG1})
 exten = s,n,MacroExit
 
 
 -- 
 
 Telefoon: 088 0100 700
 Sales: sa...@pocos.nl | Service: serviced...@pocos.nl
 http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel 
 17097024
 


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Re: [SR-Users] Redirect server for load balancing and HA

[Olle E. Johansson]

On 13 Mar 2015, at 16:39, Markus unive...@truemetal.org wrote:

 Hi Daniel,
 
 Am 13.03.2015 um 09:47 schrieb Daniel-Constantin Mierla:
 Can asterisk make some decision on headers? IIRC, someone was mentioning
 at some point to use Remote-Party-ID (could be now P-Asserted-Identity)
 header to match on a specific profile.
 
 I don't know. I'll ask in the Asterisk list. But based on your answer it 
 really seems this is a scenario that not many people need or ask for.
 
 I'd love to find out why this is the case. I would have estimated 9 out of 10 
 people need to fine-tune DTMF + codec settings in Asterisk.
The most common solution I've seen is to have kamailio use 10 different ports 
and have 10 different matching peers in the Asterisk SIP config to select 
settings.


/O
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Re: [SR-Users] Regd. Kamailio GPL

[Olle E. Johansson]

On 11 Mar 2015, at 10:13, Daniel Tryba d.tr...@pocos.nl wrote:

 On Tuesday 10 March 2015 14:52:22 Shankar wrote:
 We are exploring kamailio source for use in our VOIP solution. This is the
 first time we are looking at open source. We have few doubts in using
 kamailio for providing our VOIP service. Under GPL any customization
 (customization mainly with respect to interacting with our proprietary
 applications e.g. Billing server)  we do to kamailio also has to be provided
 to the end user. Our doubt is whether we can procure commercial rights to
 our customised code?
 
 You should really contact a lawyer specialised in IP to get good advice. But 
 gnu.org points out that above assumption isn't always true:
 http://www.gnu.org/licenses/old-licenses/gpl-2.0-faq.html#GPLRequireSourcePostedPublic
 
 The GPL does not require you to release your modified version. You are free 
 to make modifications and use them privately, without ever releasing them. 
 This applies to organizations (including companies), too; an organization can 
 make a modified version and use it internally without ever releasing it 
 outside the organization.
 
 But if you release the modified version to the public in some way, the GPL 
 requires you to make the modified source code available to the program's 
 users, under the GPL.
 
 So if you don't ship a binary to endusers, you don't have to release source 
 (including modifications). It all depends on what you are supplying to your 
 endusers (a service or a (complete) software product).
 

There is a big difference between GPLv2 and GPLv3 here.

For GPL2, providing services is not seen as distribution of the source, and the
commercial code can stay in the company. For GPLv3 it's different as far as I
understand.

Kamailio is GPLv2 and Siremis - the web interface by Asipto - is GPLv3.

/O
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Re: [SR-Users] Help with 407 Proxy Auth. Required

[Olle E. Johansson]

On 09 Mar 2015, at 15:07, Agiftel agif...@gmail.com wrote:

 Hi all, i cannot understand where is the problem with this transaction:
 
 Kamailio ask for Proxy authorization and in the second INVITE credentials
 are present.
 Can you help me understand?
This is typical - it happens when the password is wrong either in the server or 
the client.

/O
 
 Regards
 
 U 2015/03/09 15:03:42.191831 10.160.21.51:5060 - 10.160.20.18:5060
 INVITE sip:1...@sip.longwave.labo;user=phone SIP/2.0.
 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, NOTIFY, UPDATE.
 Supported: 100rel,from-change,timer,histinfo.
 User-Agent: OXO_GW_820/092.001.
 Session-Expires: 43200.
 P-Asserted-Identity: PIPPO sip:2...@oxo.longwave.labo;user=phone.
 History-Info: sip:1...@sip.longwave.labo;user=phone;index=1.
 To: sip:1...@sip.longwave.labo;user=phone.
 From: PIPPO
 sip:2...@oxo.longwave.labo;user=phone;tag=11056089a66d6a43ec04cd21b78039c5.
 Contact: PIPPO sip:2000@10.160.21.51;transport=UDP;user=phone.
 Content-Type: application/sdp.
 Call-ID: 346eded1901cb5b60b9d97a14f196e29@10.160.21.51.
 CSeq: 178945190 INVITE.
 Via: SIP/2.0/UDP
 10.160.21.51;rport;branch=z9hG4bK1fd5f032d2296189ba85c9bf8553ff28.
 Max-Forwards: 70.
 Content-Length: 208.
 .
 v=0.
 o=default 1425909984 1425909984 IN IP4 10.160.21.51.
 s=-.
 c=IN IP4 10.160.21.51.
 t=0 0.
 m=audio 32000 RTP/AVP 18 106 4 8 0.
 a=sendrecv.
 a=rtpmap:106 telephone-event/8000.
 a=fmtp:106 0-15.
 a=maxptime:90.
 
 
 U 2015/03/09 15:03:42.193299 10.160.20.18:5060 - 10.160.21.51:5060
 SIP/2.0 407 Proxy Authentication Required.
 To:
 sip:1...@sip.longwave.labo;user=phone;tag=b27e1a1d33761e85846fc98f5f3a7e58.aa4d.
 From: PIPPO
 sip:2...@oxo.longwave.labo;user=phone;tag=11056089a66d6a43ec04cd21b78039c5.
 Call-ID: 346eded1901cb5b60b9d97a14f196e29@10.160.21.51.
 CSeq: 178945190 INVITE.
 Via: SIP/2.0/UDP
 10.160.21.51;rport=5060;branch=z9hG4bK1fd5f032d2296189ba85c9bf8553ff28;received=10.160.21.51.
 Proxy-Authenticate: Digest realm=oxo.longwave.labo,
 nonce=VP2palT9qD7LOD/nl465ujUSUW0NiDo/.
 Server: lw-sipproxysrv.
 Content-Length: 0.
 .
 
 
 U 2015/03/09 15:03:42.195800 10.160.21.51:5060 - 10.160.20.18:5060
 ACK sip:1...@sip.longwave.labo;user=phone SIP/2.0.
 Call-ID: 346eded1901cb5b60b9d97a14f196e29@10.160.21.51.
 From: PIPPO
 sip:2...@oxo.longwave.labo;user=phone;tag=11056089a66d6a43ec04cd21b78039c5.
 To:
 sip:1...@sip.longwave.labo;user=phone;tag=b27e1a1d33761e85846fc98f5f3a7e58.aa4d.
 Via: SIP/2.0/UDP
 10.160.21.51;rport;branch=z9hG4bK1fd5f032d2296189ba85c9bf8553ff28.
 CSeq: 178945190 ACK.
 Content-Length: 0.
 .
 
 
 U 2015/03/09 15:03:42.199176 10.160.21.51:5060 - 10.160.20.18:5060
 INVITE sip:1...@sip.longwave.labo;user=phone SIP/2.0.
 Proxy-Authorization: Digest
 username=oxo,realm=oxo.longwave.labo,nonce=VP2palT9qD7LOD/nl465ujUSUW0NiDo/,uri=sip:1...@sip.longwave.labo;user=phone,response=263ed33da80fb520af0fb0b2f246d310.
 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, NOTIFY, UPDATE.
 Supported: 100rel,from-change,timer,histinfo.
 User-Agent: OXO_GW_820/092.001.
 Session-Expires: 43200.
 P-Asserted-Identity: PIPPO sip:2...@oxo.longwave.labo;user=phone.
 History-Info: sip:1...@sip.longwave.labo;user=phone;index=1.
 To: sip:1...@sip.longwave.labo;user=phone.
 From: PIPPO
 sip:2...@oxo.longwave.labo;user=phone;tag=11056089a66d6a43ec04cd21b78039c5.
 Contact: PIPPO sip:2000@10.160.21.51;transport=UDP;user=phone.
 Content-Type: application/sdp.
 Call-ID: 346eded1901cb5b60b9d97a14f196e29@10.160.21.51.
 CSeq: 178945191 INVITE.
 Max-Forwards: 70.
 Via: SIP/2.0/UDP
 10.160.21.51;rport;branch=z9hG4bK503a3569cf5b057bd47d1154c5ba0842.
 Content-Length: 208.
 .
 v=0.
 o=default 1425909984 1425909984 IN IP4 10.160.21.51.
 s=-.
 c=IN IP4 10.160.21.51.
 t=0 0.
 m=audio 32000 RTP/AVP 18 106 4 8 0.
 a=sendrecv.
 a=rtpmap:106 telephone-event/8000.
 a=fmtp:106 0-15.
 a=maxptime:90.
 
 
 U 2015/03/09 15:03:42.200387 10.160.20.18:5060 - 10.160.21.51:5060
 SIP/2.0 407 Proxy Authentication Required.
 To:
 sip:1...@sip.longwave.labo;user=phone;tag=b27e1a1d33761e85846fc98f5f3a7e58.d413.
 From: PIPPO
 sip:2...@oxo.longwave.labo;user=phone;tag=11056089a66d6a43ec04cd21b78039c5.
 Call-ID: 346eded1901cb5b60b9d97a14f196e29@10.160.21.51.
 CSeq: 178945191 INVITE.
 Via: SIP/2.0/UDP
 10.160.21.51;rport=5060;branch=z9hG4bK503a3569cf5b057bd47d1154c5ba0842;received=10.160.21.51.
 Proxy-Authenticate: Digest realm=oxo.longwave.labo,
 nonce=VP2palT9qD7LOD/nl465ujUSUW0NiDo/.
 Server: lw-sipproxysrv.
 Content-Length: 0.
 .
 
 
 U 2015/03/09 15:03:42.203016 10.160.21.51:5060 - 10.160.20.18:5060
 ACK sip:1...@sip.longwave.labo;user=phone SIP/2.0.
 Call-ID: 346eded1901cb5b60b9d97a14f196e29@10.160.21.51.
 From: PIPPO
 sip:2...@oxo.longwave.labo;user=phone;tag=11056089a66d6a43ec04cd21b78039c5.
 To:
 sip:1...@sip.longwave.labo;user=phone;tag=b27e1a1d33761e85846fc98f5f3a7e58.d413.
 Via: SIP/2.0/UDP
 10.160.21.51;rport;branch=z9hG4bK503a3569cf5b057bd47d1154c5ba0842.
 CSeq: 178945191 ACK.
 Content-Length: 0.
 . 

Re: [SR-Users] kamailio 4.2.3 Segmentation fault when using db_cluster module

[Olle E. Johansson]

On 09 Mar 2015, at 17:16, Jan Hazenberg je...@cyberchaos.nl wrote:

 Hello,
 
 I have a issue on kamailio 4.2.3 when using the db_cluster module. I have the 
 following config:
 
 #!define DBURL cluster://cls1
 
 # - db_cluster params -
 modparam(db_cluster, connection,
 con1=mysql://kamailio:kamailiorw@10.121.0.120/kamailio)
 modparam(db_cluster, connection,
 con2=mysql://kamailio:kamailiorw@10.121.0.121/kamailio)
 modparam(db_cluster, connection,
 con3=mysql://kamailio:kamailiorw@10.121.0.122/kamailio)
 modparam(db_cluster, cluster, cls1=con1=9r8r;con2=9r8r;con3=9r8r)
 modparam(db_cluster, inactive_interval, 180)
 modparam(db_cluster, max_query_length, 5)
 
 # - sqlops params -
 modparam(sqlops,sqlcon,ca=cluster://cls1)
 
 
 When i start kamailio it fails with:
 
 Mar  9 17:11:49 localhost kamailio[31063]: INFO: rr [../outbound/api.h:54]: 
 ob_load_api(): Failed to import bind_ob
 Mar  9 17:11:49 localhost kamailio[31063]: INFO: rr [rr_mod.c:160]: 
 mod_init(): outbound module not available
 Mar  9 17:11:49 localhost kamailio[31063]: INFO: usrloc [hslot.c:53]: 
 ul_init_locks(): locks array size 1024
 Mar  9 17:11:49 localhost kamailio[31063]: INFO: permissions 
 [parse_config.c:251]: parse_config_file(): file not found: 
 /usr/local/etc/kamailio/permissions.allow
 Mar  9 17:11:49 localhost kamailio[31063]: INFO: permissions 
 [permissions.c:608]: mod_init(): default allow file 
 (/usr/local/etc/kamailio/permissions.allow) not found = empty rule set
 Mar  9 17:11:49 localhost kamailio[31063]: INFO: permissions 
 [parse_config.c:251]: parse_config_file(): file not found: 
 /usr/local/etc/kamailio/permissions.deny
 Mar  9 17:11:49 localhost kamailio[31063]: INFO: permissions 
 [permissions.c:617]: mod_init(): default deny file 
 (/usr/local/etc/kamailio/permissions.deny) not found = empty rule set
 Mar  9 17:11:49 localhost kernel: kamailio[31063]: segfault at 80 ip 
 7f7f4f9ffddc sp 7fffde2f88a0 error 4 in 
 db_cluster.so[7f7f4f9ec000+28000]
 Mar  9 17:11:50 localhost kamailio: ERROR: core [daemonize.c:315]: 
 daemonize(): Main process exited before writing to pipe
 
 
 Could this be a bug or is this a configuration issue? I tested the dbnodes 
 and they seem to respond fine to any query's i send.

Configurations should never create segmentation faults, so it's clearly a bug. 
Can you find the core dump file and produce a backtrace?

/O
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Re: [SR-Users] Group call advice: remove calling branch after alias_db_lookup/lookup_branches

[Olle E. Johansson]

On 07 Mar 2015, at 15:20, Anthony Messina amess...@messinet.com wrote:

 So far, I've been trying out the following which seems to work OK for online
 subscribers, but not as well when a group call member is a DAHDI phone
 routed from/to Asterisk.
Yeah, the asterisk usage of contacts is not a good thing here.

/O


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Re: [SR-Users] Group call advice: remove calling branch after alias_db_lookup/lookup_branches

[Olle E. Johansson]

On 07 Mar 2015, at 01:32, Anthony Messina amess...@messinet.com wrote:

 In terms of implementing group calling via the append_branch feature of 
 both 
 alias_db_lookup followed by lookup_branches, I'm looking for a reliable way 
 to 
 ensure that if the caller happens to be a member of the group (list of 
 branches), the branch that's created to the original caller is dropped.
 
 Originally, I was thinking of comparing $rU and $fU in branch_route, but this 
 would limit the ability for one contact of an AOR to call another contact of 
 the same AOR.
 
 Can anyone offer an example of an efficient method to accomplish this?

From reading your e-mail it seems that you propose a possible idea
- why not compare contact URI's?

If they have +sip.instance, just compare that. If they have not,
try to compare the full SIP URI's in the contacts.

Just brainstorming.
/O

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Re: [SR-Users] Keeping Registrations alive in IPv6 scenarios

[Olle E. Johansson]

On 27 Feb 2015, at 09:04, Sebastian Damm d...@sipgate.de wrote:

 Hi,
 
 while testing IPv6 with customers, we fell over quite a few cases, where 
 customers aren't reachable on inbound calls most of the time. And digging 
 into this, we found the home router firewall as the cause for those problems.
 
 Normally, you would think, all the NAT problems cease when switching to IPv6. 
 But actually, right now I don't know how to fix that problem.
 
 In IPv4 NAT scenarios, we would flag the customer during the registration, 
 and Kamailio would send NAT pings (those 4 bytes of UDP junk) every few 
 seconds to keep the firewall in the NAT router open. And that worked pretty 
 great.
 
 Now we have IPv6. We don't have NAT. But we still have a home router in front 
 of SIP devices, with a firewall. And this firewall will allow outbound 
 traffic. But after a few seconds it won't allow incoming connections anymore. 
 And the routers I have seen so far don't have a configurable firewall where 
 you could allow inbound traffic from our server.
 
 Unfortunately, only our load balancer is IPv6, our registrar is still IPv4 
 only. And the loadbalancer doesn't know anything about registrations and 
 which customer needs an IPv6 keepalive. 
 
 Does anyone have a hint, how to keep the IPv6 registrations alive? Thanks in 
 advance.
 

AARGH. Now I see. Thank you. (Disregard earlier e-mail).

In all modern standards, like SIP Outbound, the need to send keep-alives has 
been pushed to the client. They need to open a network flow and keep it open - 
UDP or TCP or TCP/TLS.

As we can't trust device developers to stay up to date, we may need IPv6 
keepalives. 

/O


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Re: [SR-Users] Keeping Registrations alive in IPv6 scenarios

[Olle E. Johansson]

On 27 Feb 2015, at 09:07, Juha Heinanen j...@tutpro.com wrote:

 Sebastian Damm writes:
 
 Does anyone have a hint, how to keep the IPv6 registrations alive? Thanks
 in advance.
 
 use tcp.

What is the problem? There is no need for any keep-alived in IPv6.

/O

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Re: [SR-Users] Registrations/TCP (new topic)

[Olle E. Johansson]

On 27 Feb 2015, at 09:53, Sebastian Damm d...@sipgate.de wrote:

 Actually, it's the latter. Our current high availability setup reilies on 
 anycast. And with TCP, this would mean a huge change in our setup.
 
That is in fact an interesting topic. Can you please elaborate a bit more on 
this  as I would like to see what we can
do in the software to make things easier.

I had a similar discussion a while ago and it seems like failover handling is 
easier in UDP and we will need to fix this in order to be able to migrate more 
users to TLS.

I haven't tested how different clients behave in regards of TCP if the server 
close a connection.

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Re: [SR-Users] kamailio asterisk

[Olle E. Johansson]

On 25 Feb 2015, at 19:24, Slava Bendersky volga...@networklab.ca wrote:

 Feb 25 13:13:00 canlvprx01 kamailio: 3(5051) DEBUG: core 
 [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body 
 [sip:sips:10101@public_ip:5066
 
 
This is clearly not a valid sip uri.

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[SR-Users] Kamailio Security Policy - How to handle vulnerability reports

[Olle E. Johansson]

Hello Kamailians!

During our last developer meeting, we had a discussion about implementing a 
security policy for the project. I drafted a proposal that seemed fine with the 
developer team. At this point, I'm looking for your feedback.

The proposal is short and brief at this point, we'll learn as we go. Much of it 
is inspired by the policy of the Asterisk project.
You can find it here:
http://www.kamailio.org/wiki/securitypolicy


We encourage your feedback!

- Is this a good thing for the project?

- Do you have any changes to the policy to suggest?

At this point, we're not looking for support systems for this, or any software 
platform - we're focusing on getting the policy right, then we're going to look 
on how to implement it.

Looking forward to your responses!

Best regards,
/Olle


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Re: [SR-Users] Kamailio Security Policy - How to handle vulnerability reports

[Olle E. Johansson]

On 25 Feb 2015, at 17:24, Daniel Tryba d.tr...@pocos.nl wrote:

 On Wednesday 25 February 2015 16:14:43 Olle E. Johansson wrote:
 http://www.kamailio.org/wiki/securitypolicy
 
 
 We encourage your feedback!
 
 - Is this a good thing for the project?
 
 Yes
 
 - Do you have any changes to the policy to suggest?
 
 Yes:
 
 secur...@kamailio.org
 This address should have a PGP key associated, used by the security officers.
 
 This is a security nightmare (a (for all purposes) shared private key).
 
 You might want to look at the Debian security announces, there the individuals
 key is used for signing and the list filters on valid keys from individuals.
 https://www.debian.org/security/faq#signature
 This makes it a little more difficult to check if an announcement is actually
 from the list:
 -get key for fingerprint in mail
 -check key with currect securitylist member
Thank you for the feedback!

 
 But I fail to see how a pgp key for security is really important. Is there a
 PKI for kamailio releases? http://www.kamailio.org/pub/kamailio/latest/src/
 contains the latest version, but there is no way to verify if this is really
 the latest release. No ssl, no dnssec, no signed checksums. These should be
 considered also.

I would love seeing signatures on releases. I think there's a key for the RPM
packages somewhere.

/O


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Re: [SR-Users] Kamailio Security Policy - How to handle vulnerability reports

[Olle E. Johansson]

On 25 Feb 2015, at 18:56, Daniel Tryba d.tr...@pocos.nl wrote:

 On Wednesday 25 February 2015 18:14:06 Olle E. Johansson wrote:
 Thank you for the feedback!
 
 BTW the Yes to is this a good thing ment: this is a really good idea to have
 in writing. But you still have to rely on the bugfinders to realize the
 impact/need to secrecy.
+1000 - this was discussed during the dev meeting.

 
 But I fail to see how a pgp key for security is really important. Is
 there a PKI for kamailio releases?
 http://www.kamailio.org/pub/kamailio/latest/src/ contains the latest
 version, but there is no way to verify if this is really the latest
 release. No ssl, no dnssec, no signed checksums. These should be
 considered also.
 
 I would love seeing signatures
 
 This needs some release management, this needs to be discussed with Daniel(-
 Constantin) as manager of the project and with the builders of packages.

Agree fully. It's currently out of scope for this document.

/O


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Re: [SR-Users] kamailio asterisk

[Olle E. Johansson]

We also need to check the core file from the crash.

/O
On 19 Feb 2015, at 09:30, Daniel-Constantin Mierla mico...@gmail.com wrote:

 Hello,
 
 can you send the REGISTER request received by kamailio and your config to me?
 
 As you receive it over TLS, you can get the register by adding the next line 
 in kamailio.cfg at the beginning of request_route:
 
 xlog(received request: [[$mb]]\n);
 
 I will like to double check if the issue is still present.
 
 You should upgrade to 4.2.3, because it is the latest stable, you have 4.2.1 
 and there were many fixes meanwhile.
 
 If you preserve sips as uri schema, then you force tls further for 
 forwarding. You should change that to sip:domain...
 
 Cheers,
 Daniel
 
 On 18/02/15 00:37, Slava Bendersky wrote:
 Hello Everyone,
 I have standard case where kamailio play role of proxy for asterisk servers.
 Kamailio configured use TLS transport on public side and on private side UDP 
 5060.
 When client (SIP soft phone) connect to TLS socket everything goes well 
 until kamailio trying forward request. Kamailio tries DNS resolve tls 
 transport srv records instead of udp then it just crashed when no tls 
 configured on private side of kamailio.
 
 Do I need manually fix sips in URI ? Or some different miss configuration ?
 
 
 [root@canlvprx01 kamailio]# rpm -qa | grep kamail
 kamailio-carrierroute-4.2.1-4.2.fc21.x86_64
 kamailio-mysql-4.2.1-4.2.fc21.x86_64
 kamailio-outbound-4.2.1-4.2.fc21.x86_64
 kamailio-4.2.1-4.2.fc21.x86_64
 kamailio-tls-4.2.1-4.2.fc21.x86_64
 
 
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: core 
 [parser/msg_parser.c:625]: parse_msg():  method:  REGISTER
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: core 
 [parser/msg_parser.c:627]: parse_msg():  uri: sips:domain.org  --- 
 Client come with TLS transport
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: core 
 [parser/msg_parser.c:629]: parse_msg():  version: SIP/2.0
 
 
 
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: core 
 [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if 
 host==us: 13==12  [domain.org] == [10.18.130.46]
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: core 
 [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 
 5060 (advertise 0) matches port 5060
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: core 
 [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if 
 host==us: 13==11  [domain.org] == [67.34.12.56]
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: core 
 [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 
 5081 (advertise 0) matches port 5060
 Feb 17 11:13:49 canlvprx01 kernel: [4130713.518667] kamailio[22484]: 
 segfault at 88 ip 004bd30c sp 7fffa2f73a20 error 4 in 
 kamailio[40+3b8000]
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: core 
 [forward.c:448]: check_self(): check_self: host != me
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: *** 
 cfgtrace:request_route=[SIPOUT] c=[/etc/kamailio/kamailio-asterisk.cfg] 
 l=850 a=25 n=append_hf
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: *** 
 cfgtrace:request_route=[SIPOUT] c=[/etc/kamailio/kamailio-asterisk.cfg] 
 l=851 a=5 n=route
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: *** 
 cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio-asterisk.cfg] l=567 
 a=16 n=if
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: *** 
 cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio-asterisk.cfg] l=563 
 a=25 n=is_method
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: *** 
 cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio-asterisk.cfg] l=571 
 a=16 n=if
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: *** 
 cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio-asterisk.cfg] l=567 
 a=25 n=is_method
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: *** 
 cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio-asterisk.cfg] l=574 
 a=16 n=if
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: *** 
 cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio-asterisk.cfg] l=571 
 a=24 n=t_relay
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tm [t_lookup.c:1373]: 
 t_newtran(): DEBUG: t_newtran: msg id=1 , global msg id=1 , T on 
 entrance=(nil)
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tm [t_lookup.c:527]: 
 t_lookup_request(): t_lookup_request: start searching: hash=48550, isACK=0
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tm [t_lookup.c:485]: 
 matching_3261(): DEBUG: RFC3261 transaction matching failed
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tm [t_lookup.c:709]: 
 t_lookup_request(): DEBUG: t_lookup_request: no transaction found
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tm [t_hooks.c:380]: 
 run_reqin_callbacks_internal(): DBG: trans=0x7f598a9ced40, callback type 1, 
 id 0 entered
 Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tm 

Re: [SR-Users] Re-invites from carrier breaks the call

[Olle E. Johansson]

On 19 Feb 2015, at 15:44, Michael Young myo...@redmonsters.net wrote:

 Some carriers require session timers -- even if you refuse them in
 Asterisk\Freeswitch\etc, the call will disconnect when the timer fires off.
 I wouldn't rely on this as a possible solution in this situation.
 
The standard is written in a way that even if the other party does not
use SIP Session timers, you can still do it one-sided. It's a feature,
not a bug. I am more concerned over the phones that doesn't
accept a plain re-invite.

/O
 Michael
 
 -Original Message-
 From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
 Eric Koome
 Sent: Thursday, February 19, 2015 8:36 AM
 To: Kamailio (SER) - Users Mailing List
 Subject: Re: [SR-Users] Re-invites from carrier breaks the call
 
 If session timers, you can accept or refuse in UAS. Eg asterisk peer
 settings : session-timer = accept | refuse | originate.
 
 
 
 On 19 Feb 2015, at 14:26, Daniel Tryba d.tr...@pocos.nl wrote:
 
 On Thursday 19 February 2015 03:44:25 Will Ferrer wrote:
 Hopefully there is something clever we could do to correct the 
 problem, it is preventing us from using alot of our carriers since 
 the re-invite breaks our clients softphones.
 
 What kind of reInvites are this? If they are session timer related, 
 you could try to remove any support for them in kamailio if disabling 
 them on the client doesn't work.
 
 --
 
 Telefoon: 088 0100 700
 Sales: sa...@pocos.nl | Service: serviced...@pocos.nl 
 http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van 
 Koophandel
 17097024
 
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Re: [SR-Users] Kamailio and rtpengine

[Olle E. Johansson]

On 13 Feb 2015, at 20:07, Richard Fuchs rfu...@sipwise.com wrote:

 Load balancing is achieved by running a hash over the call-id and using
 the hash value to determine which RTP proxy from the selected set to
 use. The hash ensures that everything related to the same call ends up
 on the same RTP proxy, which is a requirement for proper operation.
 
 If weighting is used, then RTP proxies with a higher weight will
 accordingly get a larger fraction of the calls. If two RTP proxies are
 defined, the first with a weight of 1 (which is the default) and the
 second with the weight of 2, then the second will get twice as many
 calls as the first one.
 
 Syntax for the config is like this:
 
 modparam(rtpengine, rtpengine_sock,
udp:localhost:2223 udp:localhost:2224=2)

Thank you for this explanation. This text could be committed directly
to the XML doc file in my opinion.

/O

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Re: [SR-Users] Unable to route outbound with multiple VRRP IP's

[Olle E. Johansson]

On 13 Feb 2015, at 09:48, m...@brightvoip.co.uk m...@brightvoip.co.uk wrote:

 Hi,
  
 I am trying to install a Kamailio server in a HA configuration, with IP's for 
 3 attached networks.  I am using Keepalived/VRRP to manage the VIP's.  I have 
 tested v 4.1.6 and latest 4.2.2
  
 I have 3 x listen entries, one for each network and mhomed=1.  Also, for 
 info, we have net.ipv4.ip_nonlocal_bind = 1  in sysctl.conf
  
 listen=udp:my.public.net.46:5060
 listen=udp:192.168.106.46:5060
 listen=udp:192.168.116.46:5060

On a Linux system, kamailio can not listen to an address not assigned to the 
system, without a specific sysctl.
If the 3 virtual IPs are independently moving between the servers, and you have 
fixed the sysctl (which I
can not remember right now) you will always have kamailio's that can't reach 
the network since the address
is not there. You may want to start Kamailio that listens to one IP when that 
IP is assigned by Keepalived.
Keepalived has scripts executed when an IP is assigned or lost on the system.

Hope this helps.
/O

 mhomed=1
 I have dispatcher set up to load balance over two seperate groups of Asterisk 
 servers, one on each private network.  Dispatcher does probing to know which 
 Asterisks are 'alive'.
  
 When I ran the system, Kamailio/Dispatcher was not able to send any probing 
 OPTIONS to any Asterisk (normal PINGS work fine in all networks in/out of 
 Asterisk/Kamailio)
  
 Looking at the log, I see these entries for all 3 networks:
 Feb 12 17:28:19 app-srv-dev-1-01 /usr/local/sbin/kamailio[2992]: ERROR: 
 core [forward.c:218]: get_out_socket(): no socket found
 Feb 12 17:28:19 app-srv-dev-1-01 /usr/local/sbin/kamailio[2992]: ERROR: 
 core [forward.c:220]: get_out_socket(): no corresponding socket found for(u
 dp:192.168.116.38:5060)
 Feb 12 17:28:19 app-srv-dev-1-01 /usr/local/sbin/kamailio[2992]: ERROR: tm 
 [ut.h:345]: uri2dst2(): no corresponding socket found for 192.168.116.38
  af 2 (udp:192.168.116.38:5060)
 Feb 12 17:28:19 app-srv-dev-1-01 /usr/local/sbin/kamailio[2992]: ERROR: tm 
 [uac.c:307]: t_uac_prepare(): t_uac: no socket found
 Feb 12 17:28:19 app-srv-dev-1-01 /usr/local/sbin/kamailio[2992]: ERROR: 
 dispatcher [dispatch.c:2564]: ds_check_timer(): unable to ping [sip:192.168.1
 16.38:5060]
 I notice from other logging that Kamailio is receiving and processing 
 incoming OPTIONS messages, and sending appropriate replies.
  
 Now, this might not be a Kamailio issue, as when I run the same config with 3 
 IP's, but NOT using VRRP/Keepalived/Aliased IP's, everything works normally.
  
 However, I do need to run this setup in HA, so would welcome any suggestions 
 as to how I might resolve this issue.
  
 Kind regards,
 Mark Hall
  
  
  
  
  
  
  
  
  
  
  
  
 
 
   
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Re: [SR-Users] Facing kamailio issue

[Olle E. Johansson]

On 09 Feb 2015, at 17:36, taha zafar tahaza...@eocean.com.pk wrote:

 Dear Kamailo team,
 
 I am facing problem in connecting kamailio on TLS, even kamailio is listening 
 perfectly. TLS certificate are also tested.
To get help more quickly, it would really help if you pointed out the problem 
better.
facing problem is too inexact for us to look at it say yes, I remember that 
and respond with an
answer. 

If you say My softphone doesn't connect, complains over certificate or The 
phone refuses to connect
and wireshark shows this it would help.

Also, if there is something special in your logs that tell you more, copy that 
into the e-mail instead
of sending an attachment. That is easier for us all to read.

In order to get help on a public mailing list, the challenge is to feed us 
details so we can spot the
issue. Try not to be to generic, instead be as specific as you can. 

/O
 
 Kindly find attached logs and kamailio configuration file and give me your 
 valuable feedback.
  
 Eocean Pvt. Ltd.
 Taha Zafar
 VoIP Engineer
 Mobile: 03318244432
  
  
 
 Disclaimer:
 This email and any attachments to it may be confidential and are intended 
 solely for the use of the individual to whom it is addressed. Any views or 
 opinions expressed are solely those of the author and do not necessarily 
 represent those of Eocean (Pvt) Ltd. If you are not the intended recipient of 
 this email, you must neither take any action based upon its contents, nor 
 copy or show it to anyone. Please contact the sender if you believe you have 
 received this email in error.
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Re: [SR-Users] [sr-dev] RTPEngine - No support for kernel packet forwarding available

[Olle E. Johansson]

On 05 Feb 2015, at 08:27, Muhammad Shahzad shaherya...@gmail.com wrote:

 I post it on both since i was a bit confused about which list is appropriate 
 for this question.
 
 Anyways, i will toss a coin next time. ;-)
Please always start with sr-users. Most developers are on that list and will 
try to help you,
as well as other users.

sr-dev is for discussion of code issues and development, not for second-level 
support
when somebody have problems. It is an important tool for developers working 
with 
code, bugs and documentation.

So in the end it's quite easy - always start with -users. If you get a 
suggestion that
you should mail -dev, then do.

Best regards,
/Olle

 
 Thank you.
 
 
 
 On Thu, Feb 5, 2015 at 1:16 AM, Juha Heinanen j...@tutpro.com wrote:
 Muhammad Shahzad writes:
 
  I have latest stable release of RTPEngine deployed in a virtual machine
  (KVM) along with Kamailio v4.2. All is working fine except i see this
  message in RTPEngine logs,
 
 Please do not post the same question to two lists.
 
 -- Juha
 
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Re: [SR-Users] How to install kamailio-4.2.2 in DEBIAN -7.1.0

[Olle E. Johansson]

On 06 Feb 2015, at 00:33, Fred Posner f...@palner.com wrote:

 On 02/05/2015 06:18 PM, Yanko Marín Muro wrote:
 Hello:
 
 How to install kamailio-4.2.2 in DEBIAN -7.1.0-amd64?
 
 This URL is the last version:
 http://www.kamailio.org/pub/kamailio/4.2.2/src/kamailio-4.2.2_src.tar.gz
 
 Best regards,
 
 Yanko Antonio Marín Muro
 
 
 Hi Yanko,
 
 This url will give you step-by-step instructions for installing via git, 
 which is the recommended method.
 
 http://www.kamailio.org/wiki/install/4.2.x/git
 
 You can copy / paste most of these commands to do a fairly quick install.

If you want to install with packages, here's the URL:
http://deb.kamailio.org

/O
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Re: [SR-Users] Lint and syntax check

[Olle E. Johansson]

On 05 Feb 2015, at 12:33, Manuel Rubio man...@altenwald.com wrote:

 Hi,
 
 I want to know if someone is using something similar to lint-like 
 behaviour[1] and syntax highlighting in editors like Sublime, TextMate, 
 emacs, ... for Kamailio configuration files.
 
 [1] http://en.wikipedia.org/wiki/Lint_%28software%29

https://github.com/miconda/vim-extensions

Is one that Daniel created. I've seen something similar for Emacs.

/O
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Re: [SR-Users] Linking Kamailio to Asterisk

[Olle E. Johansson]

On 04 Feb 2015, at 15:53, Brian Echesa bech...@gmail.com wrote:

 Hi,
 
 Kindly assist.
 I have successfully installed Kamailio and Asterisk.  Asterisk is acting as 
 an PSTN in my case.  The trunk between Kamailio and Asterisk is up. Issue is 
 that i can not route calls to the PSTN. I am using the default configuration 
 
 version: kamailio 4.2.2 (i386/linux)
 
 OS: Debian wheezy
 
If you want to get help, you need to provide more details than this. 

If you need commercial support to fix your issues - use the -business list and 
ask for help or look for companies
on the Kamailio.org web site.

On this list you need to do a bit more work to get help. Where does the call 
come from, how is the message routed to Kamailio and from Kamailio to Asterisk. 
Do you see the message in asterisk at all or doesn't it reach asterisk? Does it 
reach asterisk, but asterisk refuses to handle it and sends an error message?

Both Kamailio and Asterisk can be used to debug messages, so you can follow the 
request through your platform and see what is happening. 

When you have found out where it goes wrong, then try mailing again adding more 
specific information that makes it easier for us to help you. Many on this list 
wants to help. Making it easier to pinpoint an issue by providing more details 
will help you get good answers.

Best regards,
/Olle




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Re: [SR-Users] [sr-dev] Next IRC development meeting

[Olle E. Johansson]

On 03 Feb 2015, at 14:23, Daniel-Constantin Mierla mico...@gmail.com wrote:

 Hello,
 
 I thought to propose organizing the next IRC meeting to discuss and sync
 each other about Kamailio development and surrounding ecosystem.
 
 First date proposal is next week, Wednesday, 15:00GMT,  Feb 11, 2015, on
Works fine for me.

/O
 the usual #kamailio channel at freenode.net. The date can be changed,
 based on feedback from the people that want to attends.
 
 I made a wiki page for the event at:
 
  - http://www.kamailio.org/wiki/devel/irc-meetings/2015a
 
 Feel free to propose topics to the agenda and/or new date.
 
 Cheers,
 Daniel
 
 -- 
 Daniel-Constantin Mierla
 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
 
 
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Re: [SR-Users] [sr-dev] [OT] ssl certificates updated for kamailio.org

[Olle E. Johansson]

On 03 Feb 2015, at 13:55, Daniel-Constantin Mierla mico...@gmail.com wrote:

 Hello,
 
 short note to say that I updated the ssl certificates for kamailio.org
 web site. They are issued by cacert.org, which is probably not trusted
 by most of the browsers, but they are well known in the open source
 environment.
 
 If you get a warning about certificate change and the issuer is
 cacert.org, then you should be safe to login over https (e.g., to edit
 the wiki pages).

You can download the CAcert root cert here:
https://www.cacert.org/index.php?id=3

Install it in your browser and you won't get any warnings. If you trust it - if 
not, please don't install it.

/O
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Re: [SR-Users] Registrar module and ipv6 addresses

[Olle E. Johansson]

On 03 Feb 2015, at 18:17, Sergey Okhapkin s...@sokhapkin.dyndns.org wrote:

 Registrar module save() function stores received field in wrong format like
 
 sip:2601:3:8805:107:41E6:C7A1:7724:CF01:5062
This is an invalid URI.
The URI needs to have [ and ] surrounding the IPv6 address.

Please open a bug report.

/O
 
 When lookup() config function is executed for ipv6 client, kamailio log shows 
 error
 
 
 Feb  3 12:07:34 east /usr/local/sbin/kamailio[16069]: ERROR: tm [ut.h:254]: 
 uri2dst2(): ERROR: uri2dst: bad_uri: sip:2601:3:8
 805:107:41E6:C7A1:7724:CF01:5062
 Feb  3 12:07:34 east /usr/local/sbin/kamailio[16069]: ERROR: tm 
 [t_fwd.c:1709]: t_forward_nonack(): ERROR: t_forward_nonack: 
 failure to add branches
 
 Is it kamailio bug or a problem with my config?
 
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Re: [SR-Users] Video Key-Frame Request using RTCP FIR or SIP INFO message

[Olle E. Johansson]

On 29 Jan 2015, at 23:56, Muhammad Shahzad shaherya...@gmail.com wrote:

 Hi,
 
 This may be a bit out of focus topic for this forum but i am posting it here 
 anyway with hope that some guru would shed some light on it and point me to 
 right direction.
 
 The problem is that i want to establish video call between a webrtc and a sip 
 client using kamailio (for signalling) and RTPEngine (for media relay). Both 
 signalling and the audio stream seems to work perfectly fine The remote video 
 on webrtc client side (i.e. video stream from sip client) takes about 20-30 
 seconds to establish but once it starts it works fine. However, the remote 
 video on sip client side (i.e. video stream from webrtc client) starts almost 
 immediately (within 3-5 seconds) but it gets stuck after 1 or 2 seconds, then 
 it goes blank after about 30 seconds.
 
 After a long discussion with sip client developer, we now understand the fact 
 that sip client sends a request for so called key-frame, which is ignored by 
 webrtc client. This request is sent through both RTCP stream and SIP INFO 
 message.
 
 The SIP INFO message seems to be pointless as media is internally managed by 
 chrome/firefox and these browsers don't give us such sophisticated access and 
 control over media streams. Please let me know if this assumption is wrong.
 
 For the RTCP stream based request (RTCP-FIR), i only see Invalid RTCP packet 
 type error message in RTPEngine logs (not sure if it drops this packet or 
 relay it anyway).
 
 Does anyone has any idea on how can we either,
 
 1. Force WebRTC client (running on Chrome / Firefox) to honor SIP INFO 
 message and issue a key-frame in RTP video stream in response to this SIP 
 request?
Talk with the SIP stack developer. I don't know if it's possible at all and I 
think using SIP info for this is 
more or less the old way. Sending it in the actual media stream feels like a 
more modern and better way.
 
 OR
 
 2. Force RTPEngine to accept RTCP-FIR and issue key-frame in RTP video stream 
 on webrtc client's behalf?
File a bug report with the RTPengine team. It's clearly something they need to 
support.

/O


 
 If there is any other solution to this, please feel free to share.
 
 
 Thank you.
 
 
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Re: [SR-Users] Can kamailio generate a PRACK

[Olle E. Johansson]

On 30 Jan 2015, at 07:58, Rahul MathuR rahul.ultim...@gmail.com wrote:

 Hello,
 
 I was wondering whether Kamailio (as proxy) can generate a PRACK on its own ( 
 since one of the custom written dialer is not sending PRACK) ?
 Is there any way I can achieve this ?
Kamailio can not participate in a dialog between two UAs like that. The CSEQ 
would be out of sync
and it would disrupt further messaging in the dialog.

You need a back2back user agent like Asterisk or FreeSwitch to do that. There 
are patches
for Asterisk that implements PRACK (which I've coded).

/O


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