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Re: [Kamailio-Users] Kamailio and SIP <-> XMPP gateway
by Daniel-Constantin Mierla 03 Mar '10

03 Mar '10

On 03/02/2010 11:25 AM, Even André Fiskvik wrote: > Thanks for the answers Daniel. > > On 1. mars 2010, at 12.23, Daniel-Constantin Mierla wrote: > > >> Hello, >> >> On 02/28/2010 06:43 PM, Even André Fiskvik wrote: >> >>> Reading through various earlier discussion though, I've found an issue that I wonder wether has been solved or not. >>> In some mailing-lists I saw that only text/plain MESSAGEs was supported through the xmpp gateway, is this still true? >>> >>> >> yes and 'no' :-). Yes because nothing was done in this respect to the module itself. 'No' because now (kamailio 3.0) you can use transformations to get content of xml documents and if the body is xhtml. >> > That's good to know. > > >>> If that's the case, is there any way to get CounterPath eyeBeam/Bria to work in the decribed scenario? >>> As to my knowledge it can only send text/html ? >>> >>> >> You can build a new message with uac module or update the existing SIP message (content type, body), do msg_apply_changes(): >> http://kamailio.org/docs/modules/3.0.x/modules_k/textops.html#id2513318 >> >> and they use xmpp functions. >> > I did a "quick" hack to support relying the MESSAGE by just setting the content-type to text/plain for now to do some testing. > The error about unsupported content-type thus went away. > > >> You can use ngrep to sniff the network to see connection between kamailio (openser) and xmpp server as well as run kamailio in debug mode in order to get more verbose log messages for each operation. >> > I've tried sniffing traffic on the server, and it doesn't seem like kamailio tries to connect to the XMPP server at all :/ > Any clues what might be causing this? Did you run in debug mode? Can you post the log messages? > I've attached my kamailio config-file as well if anyone would take You sent the email only to me. Did you intend it for mailing list? I recommend to keep sending to mailing list apart of messages with private data. In this way many can reply and other can learn from discussion. > a look at it (the main config is in xmpp.cfg) and also a diagram showing how the components are realated to each other. > I will try to look over configs when I get some time -- because of a trip ahead I am pretty constrained today. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] preparing for kamailio 3.0.1
by Daniel-Constantin Mierla 03 Mar '10

03 Mar '10

Hello, I think it is time to prepare for first minor release (3.0.1) in 3.0 series. We are approaching two months since kamailio 3.0.0, period within several issues were reported and fixed. Please report anything you are aware is not currently working in 3.0 to sr-dev(a)lists.sip-router.org or bug tracker: http://sip-router.org/tracker/ As for date, end of this week or beginning of next week would be a good time? Other opinions? Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Serusers] Need a way to substitute SIP Response Code 504 for something else
by Frank Durda IV 03 Mar '10

03 Mar '10

There is a commercial switch vendor who has recently decided to make two changes that combined are quite evil: 1, They decided to devivate from RFC 3398 and when a given call INVITE timer expire (any of the SS7 ones, plus the SIP ones, including any limit specified in the INVITE itself), instead of translating the ISUP Release Cause 102 into a SIP Response Code of 504 (Gateway Timeout), they send out a 503 (Service Unavailable) instead. (503 is commonly used in low-cost carriers and the commercial equipment they use to trigger a route-advance, where the call doesn't end but is relaunched via another slightly more expensive route/carrier who perhaps can get the call to the destination. Some of these low-cost outfits may retry the same call with five or six carriers before reluctantly sending to the local access tandem which is usually the most expensive choice, but almost certain to work. This probably isn't what the inventors of the 503 code had in mind, but that is how it is used. A minority use 404 the same way.) 2. The same vendor implemented an automatic "Address Reach" testing mechanism for outbound calls. This thing means that when sending a call to a distant switch, if a single or small number of calls are returned from that distant switch with a SIP Response Code of 504 from the destination, the sending switch will take that destination switch completely out of route ("blacklist") and not send any calls at all to that destination until the blacklist is manually reset. So a result that applies to one call is fatal to all subsequent calls. This also means that if you sent the INVITE Expires timer to a value less than the number of seconds the calling party is willing to let the called number ring (called party doesn't have voice mail or it doesn't pick up for several rings), you trigger a 504 and bingo, you have killed the route between those two switches. So your own choices for how long to wait on ring-back for a call dictate how quickly it could kill the entire route for that call. Now, the two changes taken together are also somewhat convienent because this equipment maker just happened to make it impossible for their own equipment to ever return a 504 and do can never trip off their own booby-trap. So in a way what these two changes effectively create a not-our-hardware-detector, triggering route failures only when in contact with RFC-compliant equipment made by somebody else. I think that behavior is anti-competitive, and probably illegal in quite a few countries (notably the EU), but that is what exists at the moment. This creation started getting deployed in the past few months. So, to get around this nonsense, I need a way in SER to detect that a 504 has come in and change it to something else before sending it on. I won't quibble about the "proxies can't forward a 503", so I won't try that. I would like to turn the 504 into a 404, which is better than having route get pulled all over the place. Now, the last time (a year or so ago) I asked about trying to change a SIP Response Code in SER, the suggested ser.cfg "fix" created all sorts of strange warnings in the logs about timers expiring and I was told to not worry about that. Well, it seemed to cause a lot more problems than just warnings like memory leaks or cores caused by something, so I ended up having to hard-code the translation I wanted into the SER source code and not use rules in ser.cfg to do this. I'm writing now to (A) alert people about what this major brand switch maker is doing with regard to 504s so you will recognize it if you encounter it, but also (B) to see if maybe has a more elegant way to change SIP Responses via the ser.cfg file has emerged. The obvious things like subst subst_uri don't appear to work, obviously busily editing the non-existent INVITE method message and ignoring the reply message. Using a reply/drop combination to generate a new reply and kill the real one doesn't seem to work either. As I recall it ends up sending the response specified in the reply, and then passing the original response as well. "drop" apparently doesn't kill it completely and I suspect reply is really meant to be used only during handling of the method messages, and not the response messages. Suggestions appreciated.

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[Kamailio-Users] kam 1.5/asterisk 1.6 hints/blf?
by Robert McGilvray 02 Mar '10

02 Mar '10

Please excuse the noob questions, I'm not very familiar with presence. Users are registered to kamailio, authenticated against radius with LDAP backend. I'd like to setup some queues on asterisk for helpdesk call center but in order to do it correctly asterisk needs the device state (ringing, in use etc). I assume I'll need to use realtime ldap and setup device entries for all my phones, then use hints on the extensions as if they were local. If I understand correctly this will make asterisk subscribe to kamailio (I will use kams IP in host field), kam then has the responsibility of handling the subscribe/publish/notify stuff. What needs to be done on the kam side? Thanks! Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

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[Kamailio-Users] Kamailio (1.5 rev5834) and/or rtpproxy gets frozen
by Iñaki Baz Castillo 01 Mar '10

01 Mar '10

Hi, I'm experimenting an annoying issue in a Debian Lenny 32 bits under a DELL 850 (2 cores). Same occured before with Debian Etch 32 and 64 bits in other host (also DELL 850). Kamailio (1.5 rev 5834) behaves as a load balancer in front of 2 PSTN gateways. There are ~200 calls and for all of them rtpproxy is applied. Sometimes, with no reason (not just in the moments of highest traffic) kamailio gets totally frozen, this is, it doesn't reply to SIP messages, neither relays them. Then Kamailio cannot be killed (just with -9). Sometimes it occurs when performing a fifo command (i.e. "kamctl fifo address_reload"). After killing it, kamailio cannot be started again. It starts and logs the usual logs (to syslog) but after "rtp proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled" there is no more logs. Then if I do a "ps aux | grep kamailio" I just see a single process, it's like the children fail when connecting to rtpproxy and die but the master process doesn't realize of it and remains alive showing no error at all. So I suspect from rtpproxy as even when kamailio was killed rtpproxy process consumes lot of CPU. I'll try with rtpproxy-1.2.1 (the latest stable version) as before I was using a old "truk" version (unfortunatelly I've no idea about how to know the CVS revision of a CVS working directory...). Anyhow I also suspect that the problem could be in the server itself, as sometimes the above problem occurs when reloading iptables and so on, very very strange, no a definitive cause or reason, very annoying. So nothing is clear for me. But I know that the problem occurs: - In two DELL 850, one with 32 and the other with 64 bits. - In both Debian Leeny and Etch. - With kamailio 1.5 rev 5834. - With rtpproxy rev XXX (how to get the last modification date in a CVS working dirrectory?) So, the fact is that I'm completely lost, no idea of what is happening. Any suggestion? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] Project Update, London Meeting, March 9
by Daniel-Constantin Mierla 01 Mar '10

01 Mar '10

Hello, Tuesday, March 9, I am going to be in London and plan to organize a social networking event for folks around SIP/VoIP, Kamailio (OpenSER) & SIP Router. One is going to be the usual evening dinner/pub meeting, see pics from some past events: http://www.asipto.com/gallery/v/social_meeting_barcelona2008/ http://www.asipto.com/gallery/v/FOSDEM2009/ I addition, if there are people interested, I would like to have a session in the afternoon (2-3 hours) to give an update about Kamailio (OpenSER) 3.0, present and future development of SIP Router. Time is rather short, but I encourage attendees to prepare short presentations as well, about their products and services, to have more information exchanged there. Let me know if you wish to attend such event in order to find a proper place to host it (drop me an email). Participation is free for everybody, at the dinner, the classic rule, everyone is going to pay for himself/herself. Looking forward to meeting many of you in London. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] Kamailio and SIP <-> XMPP gateway
by Even André Fiskvik 01 Mar '10

01 Mar '10

Hi. I've just started looking at kamailio, and I'm experimenting a bit with configuring a SIP <-> XMPP gateway with it using the pua_xmpp module. The configuration part hasn't beed straight-forward, but I think I'm starting to get somewhere. Both kamailio and XMPP/Jabber are new technologies for me serverside, so I don't have a great deal of experience with either. In short, what I'm trying to accomplish at this stage is: - Have CounterPath eyeBeam/Bria to register to my kamailio server - Have Aastra 6755i register to my kamailio server - Connect an XMPP client to my XMPP server - The CounterPath eyeBeam/Bria, Aastra and the XMPP client should all be able to see each others presence - The CounterPath eyeBeam/Bria should be able to IM other eyeBeams/Brias - The CounterPath eyeBeam/bria should be able to IM the XMPP clients and the other way around Reading through various earlier discussion though, I've found an issue that I wonder wether has been solved or not. In some mailing-lists I saw that only text/plain MESSAGEs was supported through the xmpp gateway, is this still true? If that's the case, is there any way to get CounterPath eyeBeam/Bria to work in the decribed scenario? As to my knowledge it can only send text/html ? Could anyone with a good knowledge of Kamailio and the various modules pua_xmpp, xmpp, pua, pua_bla comment on this kind of setup is at all possible with kamailio and it's current state? Is there any way I can verify that opensips connects to the XMPP server correctly? I tried connecting FreeSWITCH to my XMPP server (Openfire) in component mode, and it would show up connected in Openfire's administration GUI, however I can not see the same when I try to configure kamailio. If anyone has experience with this setup and have another XMPP server they have tested it with, I would also want to try to swap Openfire with that (given they also have some guidance on XMPP server configuration). Best regards, Even André Fiskvik

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[Kamailio-Users] remove parameter
by Elgar Onel 26 Feb '10

26 Feb '10

Dear friends, how to remove parameters from INVITE sip address? tia, e.

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[Kamailio-Users] ratelimit enhancements
by Daniel-Constantin Mierla 26 Feb '10

26 Feb '10

Hi Marius, since you did some updates to this module, I am opening for debate some needed enhancements I did during 3.0 testing phase and want to get opinions how to get in the code repo. Practically is a new module I named for now ratelimit2 and my last idea is to get it named pipelimit in the trunk. The reason for a new module are some major changes. The module uses the same algorithm but its core is overhaul. - definitions of pipes are loaded from database - there can be unlimited number of pipes - pipes are identified by string names - should be possible to reload pipes at runtime (iirc, not yet in) - new pipes can be added at runtime - functions accept variables to identify the pipe Since I never used queues from this module and haven't spent time to understand the concept behind, this functionality is completely missing. The old module might be good to keep in place, probably many people are using it in this form. So, proposals? What is the way to go on? Common code (algorithms) can be made lib at some point. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] tracking new features
by Daniel-Constantin Mierla 25 Feb '10

25 Feb '10

Hello, a reminder for developers to write a short note for each new feature either to NEWS file in source tree or to wiki page: http://sip-router.org/wiki/features/new-in-devel Makes life much easier when the next major release is going to happen and help people to spot quickly what new features are already available in devel branch, hopefully attracting some of them to go for it and test. Now the wiki page is updated with most of NEWS content for 3.1. Thanks, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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