- sr-users - kamailio.org

Kamailio as redirect
by Mike Claudi Pedersen 24 Feb '14

24 Feb '14

i want to setup a test environment to see how kamailio handles redirection. i want to be able to reroute calls between servers, where do i insert the configuration for this? can i append a list of user and destination to make the configuration go through, to see where the call is going? im new at kamailio, and im having a really hard time finding the right documentation to support a redirect server. could somone please give me an example? lets say i have 3 user 1000 1001 1002 and they are … [View More]

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Configuring Kamailio as an upstream proxy for FreeSwitch and which RTP proxy to choose
by Sean Kemball 24 Feb '14

24 Feb '14

Hi New to Kamailio and FreeSwitch, loosely familiar with SIP mechanics, and not a complete network idiot... but please be gentle. :) We're trying to get Kamailio set up in front of a FreeSwitch-based SIP application server, to do some simple policy controls and, using one of the RTP proxy modules, to help handle the media side of things. Our SIP trunks come from our provider on a VLAN addressed as 10.x, and they provide an upstream proxy/media server (10.y address). The link also carries … [View More]Internet traffic so we're running it through a Cisco ASA which breaks out the VLANs and static NATs the SIP stuff to our DMZ (10.z address). This is where we want to put the Kamailio box, where it should receive the calls and route them back through the ASA to the internal address (10.w range - all 10.w/x/y/z are separate, non-overlapping networks). Outbound calls from the application server (there are very few) should pass back up to Kamailio which will validate the numbers against an approved list (the app should only dial a subset of numbers) and pass them to the upstream proxy if necessary. We need to ensure that inbound calls from outside trombone through our application server so that the caller doesn't get billed for any calls, but this should be the default I think as that side of things is handled by FS (the app stands up a new call and bridges it to the incoming). This topology should allow us to use Kamailio to hide details of the FS app from the outside world, and make life easier for the app developer who should just send/receive to the Kamailio box with no further thought or complexity. If we put the FreeSwitch box in the DMZ where we want to put Kamailio, and then turn on SIP packet inspection in the ASA, calls flow but quality is poor for some callers. If we turn off SIP packet inspection, we get no audio - I can't find any clear Cisco documentation for this but I think the SIP inspection stuff in the ASA seems to handle RTP NAT fixups and the like too. But with no docs it may as well be magic, and I want to remove it if possible. With Kamailio in the DMZ and FS internal, we loosely followed http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc and hacked out the voicemail and Lua stuff, but couldn't get calls to flow, whether or not we used the SIP inspection on the ASA. We were confused why this example isn't define as WITH_NAT and doesn't use a local RTP proxy. Surely that would be needed? Questions: 1. Should the proposed topology, with Kamailio + an RTP proxy behind a firewall, relaying to FS on an inside interface, work? (Can't see why not) 2. Does it need a local RTP proxy on the Kamailio box, particularly if we turn off the ASA SIP inspect stuff? 3. Can you recommend which RTP proxy to use? There seem to be at least 3 that work with Kamailio. The box is CentOS 6.5, and it would be nice to use known-to-work packages rather than compile from source. (But eh, if I haveta). 4. Can anyone point me to some docs to explain what ports need to be open between the Kamailio box and my upstream proxy/media server? I can be more liberal between inside and DMZ I guess. 5. Is static NAT in this environment going to bite me, or should it be OK? 6. Is there any better documentation that we should be using to make this easier, or should I just man up and try harder? TIA Sean Volunteer for our Red Puppy street appeal to help puppies like Gordy become guide dogs. ############################################################################# This email, including any attachments, is intended solely for the addressee(s) It is confidential and may be legally privileged. If you are not the intended recipient, you must not copy, disclose, distribute or otherwise use it or the information in it. Please notify the sender at once and delete it from your system immediately. Any views or opinions expressed are solely those of the author and do not necessarily represent those of Blind Foundation. The Blind Foundation does not accept responsibility for any viruses or other malicious code that may be transmitted with this email. ############################################################################# [View Less]

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Change from_uri
by arun Jayaprakash 24 Feb '14

24 Feb '14

Hello, I have a problem that my calls get rejected by he pstn gateway as my gateway is expecting a DID number instead of a 4 digit extension as the 'from user". Let me explain my situation: 1. I have set callfwd_busy in my user preference table. 2. In my failure route I check to see if callfwd_busy has been set, if it has then I execute the following script: if (t_check_status("486|408")) { $avp(dst_number) = $rU; &… [View More]nbsp; if (avp_db_load("$avp(dst_number)", "$avp(callfwd_busy)")) { xlog("LOG: avp(callfwd_busy)=$avp(callfwd_busy), avp(dst_number)=$avp(dst_number)\n"); } xlog("LOG: Failure route with $rU $avp(dst_number) \n"); $rU = $avp(callfwd_busy); $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; } When I call from extension 7004 to ext 7002 ( this has callfwd_busy set to a DID number) the trace looks as follows: U 172.10.30.15:5080 -> 172.10.30.15:5060 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP 54.200.xx.yy:5060;branch=z9hG4bKee75.ab791c5.0;received=172.10.30.15. Via: SIP/2.0/UDP 192.168.1.8:5060;received=71.252.219.63;branch=z9hG4bK209939211;rport=1025. From: "7004" <sip:7004@ajfmc1.myDomain.net>;tag=1830267367. To: <sip:7002@ajfmc1.myDomain.net>;tag=as17cd0b40. Call-ID: 876294364-5060-74(a)BJC.BGI.B.I. CSeq: 731 INVITE. Server: Asterisk PBX 1.8.17.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0. U 172.10.30.15:5060 -> 172.10.30.15:5080 INVITE sip:18455981234@64.136.xx.yy:5060 SIP/2.0. Record-Route: <sip:54.200.xx.xx;lr=on;ftag=1830267367;nat=yes>. Via: SIP/2.0/UDP 54.200.xx.xx:5060;branch=z9hG4bKee75.ab791c5.1. Via: SIP/2.0/UDP 192.168.1.8:5060;received=71.252.xx.yy;branch=z9hG4bK209939211;rport=1025. From: "7004" <sip:7004@ajfmc1.myDomain.net>;tag=1830267367. To: <sip:7002@ajfmc1.myDomain.net>. Call-ID: 876294364-5060-74(a)BJC.BGI.B.I. CSeq: 731 INVITE. Contact: "7004" <sip:7004@71.252.xx.yy:1025>. Max-Forwards: 16. User-Agent: Grandstream HT701 1.0.4.8. Privacy: none. P-Preferred-Identity: "7004" <sip:7004@ajfmc1.myDomain.net>. Supported: replaces, path, timer, eventlist. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE. Content-Type: application/sdp. Accept: application/sdp, application/dtmf-relay. Content-Length: 453. . v=0. o=7004 8000 8000 IN IP4 172.10.30.1554.200.xx.xx. s=SIP Call. c=IN IP4 172.10.30.1554.200.xx.xx. t=0 0. m=audio 6248262482 RTP/AVP 0 18 4 8 2 97 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16,32-36,54. a=nortpproxy:yes. a=nortpproxy:yes. # U 172.10.30.15:5080 -> 172.10.30.15:5060 SIP/2.0 503 Unavailable. Via: SIP/2.0/UDP 54.200.xx.xx:5060;branch=z9hG4bKee75.ab791c5.1;received=172.10.30.15. Via: SIP/2.0/UDP 192.168.1.8:5060;received=71.252.xx.yy;branch=z9hG4bK209939211;rport=1025. From: "7004" <sip:7004@ajfmc1.myDomain.net>;tag=1830267367. To: <sip:7002@ajfmc1.myDomain.net>;tag=as17cd0b40. Call-ID: 876294364-5060-74(a)BJC.BGI.B.I. CSeq: 731 INVITE. Server: Asterisk PBX 1.8.17.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0. It looks like my gateway is expecting to see a DID number in the from header. Can someone let me know how I can dot this? Thank you. Arun [View Less]

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Query about RTP flow
by Nandini madhu 23 Feb '14

23 Feb '14

Dear All, I have read about SIP and RTP packet flows but ended with a question about RTP packets flow logic between client-to client when the SIP session starts. I have a running Kamailio (V 4.0.4) server on ubuntu (12.04) platform which is also configured fro DNS and DHCP servers. And following are my test bed set-ups: Kamailio server (DNS+ DHCP+Kamailio) | Router <--------------------… [View More]

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HTABLE update from DB
by Uri Shacked 23 Feb '14

23 Feb '14

It does not work...I tried with autoexpire=0, I tried with no autoexpires define, I Tried defining a number and see if the value is updated.... this is my table definition: modparam("htable", "htable", "A=>size=8,dbtable=aaaaaa;initial=0,dbmobe=0;") I remind you i need the memory to be updated from the DB every interval, and not the DB form the memory. thanks. >Declare the table without autoexpire. Check again the module README. >Regards, >Ovidiu Sas >On Feb 23, 2014 11:… [View More]

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RTPProxy/Mediaproxy issue
by Wingsravi R 23 Feb '14

23 Feb '14

Dear Kamailio'ns, I have configured kamailio (V 4.0) with RTPproxy server. Audio/Video call establishments are just fine. With this i have some unknown behaviour in my set-up is that When i am running kamailio server with RTPproxy instance, i am experiencing Jitter, Latency, pixelled audio/video calls. But when i run only Kamailio server without RTPproxy service running, audio/video calls are just fine (no any jitter, latency kind of issues). I also tried with Mediaproxy (AG projects) server … [View More]

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HTABLE update from DB
by Uri Shacked 23 Feb '14

23 Feb '14

Hi, Following my issue with reloading data. I am thinking of a way to keep data updated in memory. I read the HTABLE module again and notice the "db_expires" option. It does not work for me... I try to set it so an item will expire in 60 sec, and when expired, it will be reloaded from DB and not deleted. Any ideas? Uri

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Using Kamailio as a websocket LB
by Luis Silva 22 Feb '14

22 Feb '14

Hi guys, Is it possible to use Kamailio as a Websocket LoadBalancer (transparent to the Websocket content, rather it's SIP or other protocol)? Many thanks, Luís

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parallel forking and 30x
by Andreas Granig 21 Feb '14

21 Feb '14

Hi, What is the intended behaviour from a general point of view if you get a 1xx on one branch and a 30x on another branch for a parallel fork? >From my understanding, the 30x is a final reply, so it's kept in tm until all other branches are finished, then it enters failure-route with the highest error code (e.g. a 487 if you cancel the other branch). What I actually want is intercepting the 30x and send another INVITE to the Contact of the 30x in the failure route. This works fine if you … [View More]

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Kamailio not starting
by Owais ul Haq 21 Feb '14

21 Feb '14

Hello, I have deployed Kamailio-3.1 on a fedora machine. And my database is placed on another windows 2008 server situated on the local network. Problem is when I run kamailio. I get the following error from logs. <core> [cfg.y:3416] : parse error in config gile /usr/local/kamailio-3.1/etc/kamailio//kamailio.cfg, line 311, column 31-35 : syntax error <core> [cfg.y:3416] : parse error in config gile /usr/local/kamailio-3.1/etc/kamailio//kamailio.cfg, line 311, column 31-35 : … [View More]

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Information
by hfermi 21 Feb '14

21 Feb '14

Good morning I would like to receive more information about your platform web based voip. Many Thanks and Regards Henry Landi

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Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Shankar 21 Feb '14

21 Feb '14

Hi, We have found the root cause for the problem that was reported (refer below mail for details) in async module. Below is the brief description, - async_route("Resume", "1") - At time t, the async records are stored at slot 't+1' of async list. - Every second, async records from the async list are processed. - In function 'async_timer_exec', slot = ticks % ASYNC_RING_SIZE; since slot is based on the ticks and if the previous invocation of '… [View More]async_timer_exec' did not finish in time, the next tick is missed. Subsequent call to 'async_timer_exec' will be with t+2. Here slot t+1 will be missed and the same will be processed only during next cycle. That means the async records will be stored for the next 100 seconds before they are actually processed. TM module will drop those timed out transactions and hence the error "t_continue: transaction not found" Now we have modified the code in such a way that slot is incremented sequentially irrespective of the ticks that is sent to the function 'async_timer_exec' function. This way we do not see any call failures and all our load runs are successful. If someone is interested we can share the code as well. Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Thursday, January 23, 2014 12:18 PM To: 'Jason Penton'; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi, >From our repeated load tests what we can conclude irrespective of the number of simultaneous calls, there is always this error "t_continue: transaction not found" occurring. If I run say 20 cps, then after running 5000 calls, we observe exactly 20 calls failing with the above error. We doubt that there is something happening during a particular point in time (for a second) which impacts the saving of those new transactions into shared memory. For 10 cps run, we observe exactly 10 call failures. We repeated with different cps and found that error is exactly equal to the cps being run. Any configuration we are missing. Anyone can help? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 3:09 PM To: 'Jason Penton'; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: FW: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, Below is our config, route[LOCATION] { if(is_method("INVITE")) { if(!route(FROMCSCF)) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); dlg_manage(); dlg_setflag("4"); async_route("RESUME", "1"); exit; } } } route[RESUME] { route(TO_LOCATION); // here t_relay to REGISTRAR is done for user lookup. exit; } Regards, Shankar Date: Tue, 21 Jan 2014 11:14:21 +0200 From: Jason Penton <jason.penton(a)smilecoms.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Message-ID: <CAE=KcrghqJHgnGDxqS1fYvUzM=HqRAcKWfEAsNJjm8xUDCq68w(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We use it heavily, but not using the async module - we use it directly from the IMS code. Can you please provide your config (or a relevant snippet) file so I can see what exactly you are testing/trying to do Cheers jason From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 2:25 PM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi, Anyone who had used t_suspend() and t_continue() can share the performance details? I tried async module with one sec sleep time. I tried only 5 calls per second but still it was not successful. After sometime I see below logs, Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm [t_suspend.c:128]: t_continue(): ERROR: t_continue: transaction not found Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times Any configuration changes can help here? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Wednesday, January 15, 2014 1:26 PM To: 'Jason Penton' Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, I am using 4.0.2 Regards, Shankar From: Jason Penton [mailto:jason.penton@smilecoms.com] Sent: Wednesday, January 15, 2014 1:21 PM To: Shankar Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Shankar, What version of Kamailio are you running? Kamailio -V Cheers Jason On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: Hi Jason, Please find below my response inline, I have some questions for you as we have used suspend/continue quite a lot in the IMS code and don't have any leaks. Firstly, why are you using pkg_mem for your hash_id and label? Remember that you will be in 2 different processes in the suspend and continue portions of the code... so pkg_mem will not work - you should use shm_mem instead. [Shankar] We use pkg_mem because we are invoking t_continue from the same process ( using thread ). Secondly, how are you using top to tell that you have a leak? Kamailio's memory is internally managed. [Shankar] After running for say 20minutes, we get out of shared memory error. Also in top output we observed incremental increase in the shared usage of shared memory for the process. Cheers Jason On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > Re-sending without the attachment. > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, January 13, 2014 4:57 PM > *To:* 'sr-users(a)lists.sip-router.org' > *Subject:* Regd. t_suspend() and t_continue() > > > > Hi, > > > > We are trying out the t_suspend() and t_continue() in our test setup. > We are facing memory leak ( both shm and pkg as per top command results). > > > > Please find below the scenario, > > > > 1) Do a t_newtran() > > 2) Allocate pkg memory for hashid and label. > > 3) Call t_suspend() > > 4) Do t_continue() when async result is available > > 5) De-allocate pkg memory reserved for hashid and label > > 6) Do a t_relay() which forwards the sip message to another sip node. > > > > In the step (6) above, we see t_newtran() allocates one more time > shared memory for the same transaction. > > > > We tried t_release() after step (4) to release the transaction as > t_relay() anyways allocates new shared memory. Nothing helped. > > > > Please let me know what are the logs you would require to debug the same. > I am attaching syslog for this run. > > > > Regards, > > Shankar -- Jason Penton Senior Manager: Applications and Services Smile Communications Pty (Ltd) Mobile: +27 (0) 83 283 7000 Skype: jason.barry.penton <mailto:name.surname@smilecoms.com> jason.penton(a)smilecoms.com <http://www.smilecoms.com/> www.smilecoms.com <http://196.33.227.129/~smlcoms/sigs/pty/images/smile_signature_07_09.jpg> This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer [View Less]

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Call Control for Credit and Rate
by Raj Roy Ghandhi 21 Feb '14

21 Feb '14

Hi, I want to implement the call control by 1. Validating the CLI and get the credit from DB 2. Check the Rate for the dialed number 3. set the call timeout for the call duration. ( credit/rate) 4. pass the call to remote PSTN GW Please guide me on this. Regards, Roy.

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Debian packages not updated since august 2013?
by Johan Wilfer 21 Feb '14

21 Feb '14

Hi, Looking at http://www.kamailio.org/wiki/packages/debs the suggestion is to use http://deb.kamailio.org/ for debian packages. But the last version of kamailio 4.0 here is 4.0.3, and 4.1 doesn't exist. The most files are last modified in 2013-08.. Here for example: http://deb.kamailio.org/kamailio/dists/wheezy/main/binary-amd64/ Is there a new server to get the deb-packages from or should I build from source from now on? /Johan

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Performance impact with AVP and VAR's
by Jijo 21 Feb '14

21 Feb '14

Hi All, I have around 300 AVP's and quite amount of VAR's are used in the config file? Does that impact performance?, If so how can i improve it? Thanks Jijo

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Re: [SR-Users] Issue with Async module
by Daniel-Constantin Mierla 20 Feb '14

20 Feb '14

Hello, thanks for reporting, I pushed a slightly different patch in master, 4.1 and 4.0 branches -- free is done even when ai->act is null. Cheers, Daniel On 20/02/14 15:13, Surendra wrote: > Hi Daniel, > > > I am using async module to continue the transaction in > asynchronously. But I am observing a leak in shared memory. Then i checked > the code, we are not freeing the async_item_t after calling t_continue. > Same this has checked with MEMDBG=1. > > … [View More]

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Re: [SR-Users] NAT traversal problem with Kamailio when trying to communicate between two different LANs
by Neo Quartz 20 Feb '14

20 Feb '14

Hi All, I installed Kamailio server on my ubuntu machine. And installed rtpproxy also in the same machine. I didm't make many changes to default configuration files, kamctlrc and kamailio.cfg. Now I am able to communicate between two SIP clients, if both sip clients are in the same LAN. But I am not able to communicate between one sip client in one LAN and the other SIP client in other LAN. Sip Call handling and ringing etc. is ok but there is no voice for this case. I don't have much … [View More]

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understanding the sqlops module and kamailio children
by Uri Shacked 20 Feb '14

20 Feb '14

Hi, Following my issue with the above.... 1. Is there a way to see if a worker (child) is processing a transaction for too long (let's say 2-3 sec)? 2. Is there an option to see if the DB connection is alive? (I am familiar with db_mysql options) 3. If all workers are stuck on this "long" transaction, is there a Q for the new INVITES? Can I see how many are there? What I was experiencing is that when my DB was very slow, kamailio crashed after a while.... I will configure … [View More]

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Asterisk Ignore "UNREGISTER" because retransmit
by pavel＠eremina.net 20 Feb '14

20 Feb '14

Hi, All! Try to register user in Kamailio and then relay REG massage to Asterisk. Asterisk register user and looks good. But when I send Unreg message to Kamailio, it unregister user successfully but Asterisk say: Message REGISTER Ignoring because retransmit (CSeq 1234 our CSeq 1234). Did someone know how solve this problem: i think that Asterisk need different CSeq for first REGISTER and Second after 401 code. dump my conf: At route[REGISTER]: if (!save("location")) { … [View More]

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Issue with Async module
by Surendra 20 Feb '14

20 Feb '14

Hi Daniel, I am using async module to continue the transaction in asynchronously. But I am observing a leak in shared memory. Then i checked the code, we are not freeing the async_item_t after calling t_continue. Same this has checked with MEMDBG=1. Original code: while(1) { lock_get(&_async_list_head->ring[slot].lock); ai = _async_list_head->ring[slot].lstart; if(ai!=NULL) _async_list_head->ring[slot].lstart = ai->next; … [View More]

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Problem with special sip notify (trying to make central provisioning for endpoints)
by Pedro Niño 20 Feb '14

20 Feb '14

Hi, how are you all I have a lab setup for using Kamailio and asterisk, with realtime, and currently I am facing 2 problems. The first one, is when trying to get the phones to reboot. Asterisk send a notify with the message, reboot. But the phone answers with a 401, and asks for a digest. In a normal asterisk environment it would not be a problem, asterisk tries to answer with the user and password cipher with the digest. But because of using this design, asterisk sends the notify with the … [View More]

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8 digit restriction on dialplan
by phillman25 20 Feb '14

20 Feb '14

Dear List I am trying to restrict 8 digit dial on my dialplan after the prefix 6859A , could someone help me with the correct syntax? Please see below what i have: Dpid: 1 Pr: 1 Match Op: 1 Match Exp: sip:6859A(only 8 additional digits allowed).* Match Len: 0 Subst Exp: sip:6859A(only 8 additional digits allowed).* Repl Exp: 3005A1883(a)xx.xx.xx.xx So basically any 8 digits after the prefix 6859A is allowed otherwise there shouldn't be a match, how could i go about doing this? Thanks in advance Phillip

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Error in carrierroute in kamailio 4.1
by Eduardo Alonso 20 Feb '14

20 Feb '14

Hi all: It's my first mail to the community and I know that will be a great community to share knowledge. OS Debian 6.4 Kamailio 4.1 I have the same problem that Oliver Roth describes in http://lists.sip-router.org/pipermail/sr-users/2013-October/080096.html, but in Kamailio 4.1: Module carrierroute not working starting from version kamailio 4.1 trying to activate carrierroute in kamailio.cfg loadmodule "carrierroute.so" i get the following error when starting kamailio (5794) ERROR: <… [View More]

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Issues with keepalive_timeout
by Kristian Kielhofner 20 Feb '14

20 Feb '14

Hello All, I'm trying to use keepalive_timeout with 4.1.1 as described here: http://lists.sip-router.org/pipermail/sr-users/2012-August/074337.html I believe I've followed these configuration steps and those defined in the nathelper module. Here are the relevant portions of my config: modparam("nathelper", "natping_interval", 60) modparam("nathelper", "keepalive_timeout", 180) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "sipping_from", "sip:pinger@domain.com") … [View More]modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", 9) And of course a setting the correct branch flag (verified) during REGISTER handling: setbflag(9); My goal is to use SIP OPTIONS messages to verify the reachability of all of my endpoints (not just NATd ones). I would just lower my re-registration timeout but there are other issues associated with that kind of change. With my current configuration (after a successful registration) a SIP OPTIONS message is sent and replied to: U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-9627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-93072f43-7c4cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . U 4.2.2.2:5060 -> 4.2.2.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 4.2.2.1:5060;received=4.2.2.1;branch=0. Call-ID: 8af79586-93072f43-7c4cf55(a)4.2.2.1. From: <sip:pinger@domain.com>;tag=uloc-5303810e-7c78-1-2273f0eb-9627292. To: <sip:user372@4.2.2.2:5060>;tag=0;ob. CSeq: 1 OPTIONS. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain. Supported: replaces, 100rel, timer, norefersub. Allow-Events: presence, message-summary, refer. User-Agent: My_UA. Content-Length: 0. . If I kill the registered UA (making it unavailable) Kamailio endlessly sends OPTIONS messages, never receives a reply, and never removes the registration from usrloc: U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-b627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-b3072f43-f35cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-c627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-c3072f43-b75cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-d627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-d3072f43-7b5cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-e627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-e3072f43-3f5cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . (and on, and on, and...) Eventually the registration timeout kicks in, the usrloc entry is removed, and Kamailio stops sending OPTIONS to the (long) unreachable endpoint. Is this expected behavior? What am I doing wrong? Thanks! -- Kristian Kielhofner [View Less]

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Re: [SR-Users] Regd. MAX_LDG_LOCKS
by Daniel-Constantin Mierla 20 Feb '14

20 Feb '14

Hello, setting the hash size to 32MB is way too much, are you having like 100 000 000 active calls at the same time? The size of the hash is used for computing the number of slots in the hash table, not the capacity in terms of memory size of all items stored in hash table. Cheers, Daniel On 03/02/14 07:33, Shankar wrote: > > Hi Alex, > > We set the hash size to 32 MB and found that the hash size is not the > same as configured one. At the max it was taking 64KB. > … [View More]

1 0

Re: [SR-Users] Regd. MAX_LDG_LOCKS
by Daniel-Constantin Mierla 20 Feb '14

20 Feb '14

Hello, it is not a really necessary to increase the value of the locks. For better performances you can do it, but again, it is not a must. Cheers, Daniel On 04/02/14 06:50, Shankar wrote: > > Hi, > > Can someone clarify here please? > > Regards, > > Shankar > > *From:*Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, February 03, 2014 10:53 AM > *To:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > *… [View More]

1 0

Kamailio 4.1 Installation from Source
by Francesco Maria Magnini 19 Feb '14

19 Feb '14

Hi, following the documentation, the following lines need to be updated on the init.d script. DAEMON=/usr/local/sbin/kamailio CFGFILE=/usr/local/etc/kamailio/kamailio.cfg Anyway, there is no CFGFILE already setted. Is it correct to add CFGFILE along with DAEMON? On top of this, in the check_fork () procedure, there’s a reference to a cfg_file that is nonexistent. In my installation i changed it to the real path of the kamailio.cfg file generated during installation. check_fork () { if … [View More]

2 2

Kamailio World - Presentations
by Daniel-Constantin Mierla 19 Feb '14

19 Feb '14

Only very few presentations accepted at Kamailio World Conference & Exhibition are not yet published. In a matters of days we should have the draft of the schedule published. It is again a good variety of topics around Kamailio and other open source VoIP applications, presenting experiences from the field, showing new product and solutions or open debate and demos panels. You can lean about various ways of deploying scalable platforms with Open Stack, Erlang or asynchronous, carrier … [View More]

1 0

uac_reg_request_to
by Isaac McDonald 19 Feb '14

19 Feb '14

Hello All, I need some clarification on the function uac_reg_request_to. The doc says "This function can be used to send an authenticated request to a remote user in the uac registrations table" Does this mean I can forward a SUBSCRIBE request, or any request for that matter, to an upstream server that's Kamailio is registered to via the UAC module? Secondly, I'm having trouble matching the user on this function. The error is: "ac [uac_reg.c:996]: uac_reg_request_to(): no user: 2503" … [View More]

1 0

Memory leak in http_query() function (utils module)
by Guillaume Bour 18 Feb '14

18 Feb '14

Hi folks, We identified a memory leak in *utils* module, when calling the http_query() function. a memory piece is allocated with malloc() to store http request result string (functions.c:54), but this memory is never freed. There is also a possible crash issue, because curl_easy_getinfo() is called *after* curl_easy_cleanup(), which must not be done (see http://curl.haxx.se/libcurl/c/curl_easy_cleanup.html) I attached patches to fix those issues (applies to 4.0 branch) Regards, Guillaume Bour

2 5

error using siremis
by Rubi Mazaki 18 Feb '14

18 Feb '14

Hi I successfully install siremis 4.0.0 +kamailio 4.1 on my ubuntu (32 bit) server. Kamailio was installed successfully and I can now run "kamctl moni" and see a nice status screen. I had to setup again siremis (by deleting the install.lock and opening the localhost/siremis setup again) but one that was achieved I can now login in to the siremis site without errors. Now to the problem: Once I enter "groups" and press "add" I get the following error displayed: [2014-02-12 20:46:33 (GMT)] An … [View More]exception occurred while executing this script: Error message: #8192, preg_replace(): The /e modifier is deprecated, use preg_replace_callback instead Script name and line number of error: /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty_Compiler.class.php:270 *function:* errorHandler ( 8192, "preg_replace(): The /e modifier is deprecated, use preg_replace_...", "/home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty_Compi...", 270, Array(8) ) @ /home/rubi/siremis-4.0.0/openbiz/bin/sysheader.inc 117 *function:* userErrorHandler ( 8192, "preg_replace(): The /e modifier is deprecated, use preg_replace_...", "/home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty_Compi...", 270, Array(8) ) @ *function:* preg_replace ( "~\{\*(.*?)\*\}|\{\s*literal\s*\}(.*?)\{\s*/literal\s*\}|\{\s*php...", "'{php' . str_repeat(" ", substr_count('\0', " ")) .'}'", "<form id="{$ form.name}" name="{$form.name}"> <div style="paddin..." ) @ /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty_Compiler.class.php 270 *function:* _compile_file ( "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail....", "<form id="{$form.name}" name="{$form.name}"> <div style="paddin...", Null ) @ /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty.class.php 1484 *function:* _compile_source ( "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail....", "<form id="{$form.name}" name="{$form.name}"> <div style="paddin...", Null, "/home/rubi/siremis-4.0.0/siremis/themes/default/template/cpl/%%2..." ) @ /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty.class.php 1417 *function:* _compile_resource ( "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail....", "/home/rubi/siremis-4.0.0/siremis/themes/default/template/cpl/%%2..." ) @ /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty.class.php 1256 *function:* fetch ( "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail...." ) @ /home/rubi/siremis-4.0.0/openbiz/bin/easy/FormRenderer.php 95 *function:* renderSmarty ( Object(EasyForm), "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail...." ) @ /home/rubi/siremis-4.0.0/openbiz/bin/easy/FormRenderer.php 45 *function:* render ( Object(EasyForm) ) @ /home/rubi/siremis-4.0.0/openbiz/bin/easy/EasyForm.php 1648 *function:* renderHTML ( ) @ /home/rubi/siremis-4.0.0/openbiz/bin/easy/EasyForm.php 1538 Can anyone tell me what I did wrong, or how can I re-install completely siremis? Hopefully that's not a bug :/ With Regards, Rubi Mazaki Email: rubimazaki(a)gmail.com [View Less]

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One more question 'bout db_mode
by Victor V. Kustov 18 Feb '14

18 Feb '14

Hi all! my usrloc database use postgress. In config I had: modparam("usrloc", "db_mode", 1) after day of work I see strange thing: # kamctl ul show --brief | grep -c AOR 26 psql -U kamailio -c "select * from location;" ... (1 row) 26 registered users in inner kamailio's memory and only 1 in database. See in doc: 1 - Write-Through scheme. All changes to usrloc are immediately reflected in database too. This is very slow, but very reliable. Use this scheme if speed is not your … [View More]

1 0

Change table type...
by arun Jayaprakash 18 Feb '14

18 Feb '14

Hello, I am planning to change the table type in the Kamailio Mysql db from MyISAM to Innodb. Could anyone let me know if this will be an issue? Thank you. Arun

2 2

modify To header, prefix + ip, using dispacher module.
by Helena Garcia-Nieto 17 Feb '14

17 Feb '14

Hey, Thank you in advanced for the time taken on my question! I have an environment phone---sbc---kamailio---gw1, gw2 I am using kamailio 4.2.0, quite basic config, using dispatcher module to route to the gw and to monitor them with options. I have a file dispatcher.list with the two gw ips: # $Id$ # dispatcher destination sets # # line format # setit(int) destination(sip uri) flags(int,opt) priority(int,opt) attributes(str,opt) # gateways 1 sip:10.X.X.38:5060 IP 1 sip:10.X.X.39:5060 AP I … [View More]

4 12

deb.kamailio.org down?
by Paul Belanger 17 Feb '14

17 Feb '14

Anybody know why deb.kamailio.org is down? Is there a mirror setup some place else? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

3 2

Re: [SR-Users] Msilo message body type
by sipatse＠in.gr 16 Feb '14

16 Feb '14

2 1

Kamailio call forward help
by Sajan Thomas 16 Feb '14

16 Feb '14

Hi sir, I have 2 server running with kamailio.I have tested in both with creating two users and chat using jitsi client. Please help me with 1.How can i implement kamailio call forward? suppose if make achat or call to kamailio server on user shoud can access the same from kamailio server 2. 2.Is there any possible modules are available for kamailio Please help me i am really got stuck here. Expecting your kind reply.

1 1

Error with ACK from SIPML5
by Jayesh Nambiar 14 Feb '14

14 Feb '14

Hello All, I have a setup which is SIPML5 -> Kamailio(Websocket) -> Freeswitch. All I am trying to do is initiate a call from SIPML5 and Play a Music file from freeswitch. This works, but the call hangs up after 30 seconds due to ACK timeout from from the SIPML5. What I can see is the SIPML5 from Chrome does send an ACK on websocket, but I see the following error in kamailio syslog: *via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found* *ERROR: <core> [… [View More]

4 11

websocket.so module not generated
by Alvaro Urionabarrenetxea Gomez 14 Feb '14

14 Feb '14

Hi. I have a problem when I'm trying to start with my Kamailio server. I have an error about not existing "websocket.so" module. Here is the error: root@kamailio:# /etc/init.d/kamailio start ** Not starting Kamailio SIP server: invalid configuration file!* *** ** 0 (8327) ERROR: <core> [sr_module.c:587]: load_module(): ERROR: load_module: could not find module <websocket> in </usr/local/lib/kamailio/modules/>* *0(8327) : <core> [cfg.y:3408]: yyerror_at(): parse_error … [View More]

1 0

FW: Client site NAT negotiation with no STUN.
by Rob Moore 14 Feb '14

14 Feb '14

Hi there. So I'm fighting what must be a bit of a common problem here but the usual googling isn't really helping me out. A bit of background: Classic Nat scenario of a sip handset behind a NAT firewall over the internet to VoIP SBC setup using Kamailio 3.3 and RTPProxy and Freeswitch as the registrar / calls server behind kamailio. (please excuse the beautiful ASCI art) ------… [View More]

2 3

SSL connection issue
by Owais ul Haq 14 Feb '14

14 Feb '14

Hello, I have installed kamailio server. Now I am trying to use SSL for my server - client connections. I have made certificates succesfully. Now when my client connects, I get the following error. [tcp_read.c:882] : ERROR : TLS accept : error : 1408F10B : SSL routines:SSL3_GET_RECORD:wrong version number I have tried finding solution online. I am asked to change the version number. Can some one please tell me where I have to change the version number ?? And change it to what ?? Thanks. Cheers, Owais

2 2

Problem with SSL version number
by Owais ul Haq 14 Feb '14

14 Feb '14

Hi kamailians, I am trying to connect kamailio and use TLS between kamailio and client. I could not get any tls_method to work. Openser error messages for each method are listed below: with tls_method=SSLv2: " tls_accept: Error in SSL: 3(8332) tls_error: error:140EC0AF:SSL routines:SSL2_READ_INTERNAL:non sslv2 initial packet " with tls_method=SSLv3 or tls_method=TLSv1: " 3(8386) tls_accept: Error in SSL: 3(8386) tls_error: error:1408F10B:SSL routines:SSL3_GET_RECORD:wrong version number "… [View More]

2 1

Re: [SR-Users] Suggestion for VOIP Solution
by DanB 14 Feb '14

14 Feb '14

That is not entirely true. We use MySQL without any kind of issues via ODBC driver. DanB On 14.02.2014 10:18, sr-users-request(a)lists.sip-router.org wrote: > Asterisk would be suitable solution, FREESwitch is also fine, but with > FREESwitch latest version uses postgreSQL, and removed support for MySQL, > if you are comfortable with postgreSQL for reporting stuffs like that then > you can proceed with FREESwitch. > > Regards > > > > Thank you with regards, > … [View More]

1 0

Suggestion for VOIP Solution
by Vijay Thakur 14 Feb '14

14 Feb '14

Hi All List Members, I have worked on kamailio for one year. It is a great open source Skype like solution. Now i have to deploy a new solution with new requirements. Kindly suggest that what will be the best open source application will fulfill my requirements. I need the following facilities: (1) Call Recording (Outbound/Inbound) (2) On Demand Recording (3) Dictation Service (4) Complex Inbound Routing (5) IVR (Interactive Voice Response) (6) Auto Attendant (7) Client Recordings Mgmt … [View More]

5 4

understanding the sqlops module and kamailio children.
by Uri Shacked 13 Feb '14

13 Feb '14

Hi, Lately i am experiencing some issues with the sqlops module. let's say, for example, that i use sqlops to query some information about the subscriber when an INVITE arrives. I use kamailio 3.3.2, MySQL 5.6 and the DB server is remote and not local. Usually, the DB result is very fast and the call continues. But, when the DB is unavailable, or slow, the result is delayed or lost. In this case: 1. What is the query time out in kamailio? 2. If the database is available but slow, does … [View More]

2 2

How do I allow any user to register? And why doesn't a call reach the other end in my case?
by Victor Sergienko 13 Feb '14

13 Feb '14

Hi, I'm pretty ignorant about Kamailio, sorry for it and possible terminology mistakes. I'd like to allow any user to register. Ideally, I'd like to make some script (Lua/whatever) function to check some data encoded in a username and allow registering based on that username only. I undefined WITH_AUTH in kamailio.cfg, and it used to work for me; but sometimes I'm starting to get "SIP Progress: 100 trying -- your call is important to us ()" and the call never reaches the other party. Here's … [View More]

1 0

$var binary operation
by Jijo 13 Feb '14

13 Feb '14

Hi I tried the binary operation using $var as per the documentation and its seems not working. Can somebody help me to understand it. http://kamailio.org/dokuwiki/doku.php/pseudovariables:3.1.x $var(a) = 3 + (7&(~2)); if( [ $var(a) & 4 ] ) { xlog("var a has third bit set\n"); } The format with square bracket in the conditional statement is showing parse error when i load the cfg. So it tried like below and that too not working $var(a) = 8; if( $var(a) & 4 ) { xlog("var a … [View More]

2 2

core generated - 502 Bad gateway
by Tuan Viet Nguyen 13 Feb '14

13 Feb '14

Hello, Upon receipt an 502 Bad gateway, I have the following error then Kamailio crashed Feb 11 11:49:57 kamailio23 /usr/local/sbin/kamailio[28895]: WARNING: tm [t_lookup.c:1564]: t_unref(): WARNING: script writer didn't release transaction Feb 11 11:50:03 kamailio23 /usr/local/sbin/kamailio[28926]: : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 31 Feb 11 11:50:03 kamailio23 /usr/local/sbin/kamailio[28876]: ALERT: <core> [main.c:788]: handle_sigs(): child process … [View More]28903 exited by a signal 11 Feb 11 11:50:03 kamailio23 /usr/local/sbin/kamailio[28876]: ALERT: <core> [main.c:791]: handle_sigs(): core was generated (gdb) bt #0 0x000000000053a81d in qm_free (qm=0x7fd6f962a000, p=0x75d63ff100000000) at mem/q_malloc.c:476 #1 0x00007fd700c5581c in free_cell (dead_cell=0x7fd6f9777bf0) at h_table.c:162 #2 0x00007fd700c56783 in free_hash_table () at h_table.c:443 #3 0x00007fd700c6981b in tm_shutdown () at t_funcs.c:122 #4 0x00000000004ee6f0 in destroy_modules () at sr_module.c:790 #5 0x0000000000463baa in cleanup (show_status=1) at main.c:573 #6 0x0000000000464c32 in shutdown_children (sig=15, show_status=1) at main.c:715 #7 0x0000000000466223 in handle_sigs () at main.c:806 #8 0x000000000046a459 in main_loop () at main.c:1767 #9 0x000000000046ca85 in main (argc=5, argv=0x7fff2f13ec38) at main.c:2566 (gdb) bt full #0 0x000000000053a81d in qm_free (qm=0x7fd6f962a000, p=0x75d63ff100000000) at mem/q_malloc.c:476 f = 0x75d63ff0fffffff0 size = 4880 next = 0x7fd6f98c5380 prev = 0x7fd6f9634840 __FUNCTION__ = "qm_free" #1 0x00007fd700c5581c in free_cell (dead_cell=0x7fd6f9777bf0) at h_table.c:162 b = 0x0 i = 1 rpl = 0x0 tt = 0x0 foo = 0x59e92da4 cbs = 0x0 cbs_tmp = 0x7fd6f98c5370 __FUNCTION__ = "free_cell" #2 0x00007fd700c56783 in free_hash_table () at h_table.c:443 p_cell = 0x7fd6f9777bf0 tmp_cell = 0x7fd6f98fb9c0 i = 3084 #3 0x00007fd700c6981b in tm_shutdown () at t_funcs.c:122 __FUNCTION__ = "tm_shutdown" #4 0x00000000004ee6f0 in destroy_modules () at sr_module.c:790 t = 0x7fd7022322f0 foo = 0x7fd702231cc0 #5 0x0000000000463baa in cleanup (show_status=1) at main.c:573 memlog = 32727 __FUNCTION__ = "cleanup" #6 0x0000000000464c32 in shutdown_children (sig=15, show_status=1) at main.c:715 __FUNCTION__ = "shutdown_children" #7 0x0000000000466223 in handle_sigs () at main.c:806 chld = 0 chld_status = 139 memlog = 0 __FUNCTION__ = "handle_sigs" #8 0x000000000046a459 in main_loop () at main.c:1767 i = 8 pid = 28926 si = 0x0 si_desc = "udp receiver child=7 sock=91.213.79.31:5060\000\326\177", '\000' <repeats 18 times>, "@\352\023/\377\177\000\000\244-\351Y\000\000\000\000\240mA\000\000\000\000\000\060\354\023/\377\177", '\000' <repeats 18 times>, "p\352\023/\377\177\000\000\030>K\000\000\000\000" nrprocs = 8 __FUNCTION__ = "main_loop" #9 0x000000000046ca85 in main (argc=5, argv=0x7fff2f13ec38) at main.c:2566 cfg_stream = 0xc36010 c = -1 r = 0 tmp = 0x7fff2f13ec68 "\233\356\023/\377\177" tmp_len = 0 port = 5 proto = 0 options = 0x5c7c18 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 3628627122 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xbf p = 0x4162c9 "H\203\304\b\303" <Address 0x4162ce out of bounds> __FUNCTION__ = "main" (gdb) Regards, [View Less]

2 2

Call forwarding from external did
by arun Jayaprakash 12 Feb '14

12 Feb '14

Hello, I have set up call forwarding in Kamailio using user_preference table. When I make a call from a local extension the callfwd funcition works. The call gets forwarded to an external did number. The problem happens when and external call ( from a DID) comes to this extension the call does not get forwarded. I had to comment out the following lines in the config file to make it work: # only local users allowed to call # if((from_uri!=myself)) { # sl_send_reply("403", "Not … [View More]

2 3

ice problems
by jaflong jaflong 12 Feb '14

12 Feb '14

I am having problems with calls from webrtc to kamailio forwarded to Asterisk These are snippet of the debug logs Asterisk CSeq: 4910 BYE Reason: SIP ;cause=488; text="Not Acceptable Here" Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0 Content-Length: 0 Jssip Cause: Bad Media Description Origin: remote Searching on google I get some indication this is to do with ice config? Please can some one suggest if this is so. In my scenerio the webrt clients will only call to the … [View More]

3 2

About a bug when registering a Tandberg (Cisco) E20 to Kamailio - Crash Segfault at 18 IP - error 4 in auth_db.so
by Pedro Niño 12 Feb '14

12 Feb '14

Kamailio is having a crash (Segfault at 18 IP - error 4 in auth_db.so) when registering a cisco E20, via SIP Here is a full dbg backtrack, and the core dump, but have no ideas about how to proceed. Im using Kamailio 4.1.1: Got this Error at /var/log: [14629359.586980] kamailio[982]: segfault at 18 ip 00007f034db8dcfd sp 00007fff34702950 error 4 in auth_db.so[7f034db85000+c000] Kamailio Version: ---------- version: kamailio 4.1.1 (x86_64/linux) d6ffa1 flags: STATS: Off, USE_TCP, USE_TLS, … [View More]

2 2

reSIProcate v1.9.0 released
by Daniel Pocock 11 Feb '14

11 Feb '14

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 The v1.9.0 release has been tagged (r10958) and made available for download: https://www.resiprocate.org/files/pub/reSIProcate/releases/ The release info page is here: http://www.resiprocate.org/ReSIProcate_1.9_Release and a direct link to the ChangeLog in SVN: https://svn.resiprocate.org/rep/resiprocate/tags/resiprocate-1.9.0/ChangeLog Debian, Fedora and Ubuntu packages have been updated, automated builds are in progress and they … [View More]

1 0

Kamailio World Updates
by Daniel-Constantin Mierla 11 Feb '14

11 Feb '14

From the first edition, Kamailio World tried to connect people and companies that can help each other or complement. If anyone is looking for new challenges in their career, don't hesitate to approach us. You can see more details at: - http://conference.kamailio.com/k02/2014/02/career-jobs/ Check also the Speakers <http://conference.kamailio.com/k02/speakers/> [1] and Schedule <http://conference.kamailio.com/k02/schedule/> [2] pages for new information about who is going … [View More]

1 0

Problem with Port=5061
by Owais ul Haq 10 Feb '14

10 Feb '14

Hello, I have configured TLS on kamailio server. I am done with all procedure of creation and installation of CA and site certificates. Problem is that in kamailio.cfg, when I use command port =5061. Kamailio doesnot start and error given in logs is >> ERROR: tcp_init : bind(a, 0x9007f4, 16) on MyIP:5061 : Adress already in use >> ERROR: Error while initializing TCP port of TLS Socket MyIP:5061 Point to note is that kamailio works fine with port=5062. Can somebody please help me … [View More]

4 4

Determining size of INVITE
by Daniel Tryba 10 Feb '14

10 Feb '14

How can I get the determine the total size of an outgoing INVITE? The size of the body is $bs, but is there a way to get the size of all headers? Based on the size of the total request I have to chose between UDP and TCP, upstream provider requires IP packets>1300 to be transmitted over TCP. Is there an easy way (like $bs) or should I just call msg_apply_changes() and loop through all headers and sum the lengths of them? -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 … [View More]

2 3

Get information in 'Reason' header field
by Tuan Viet Nguyen 10 Feb '14

10 Feb '14

Hello, How to get ISUP cause in the 'Reason' header field of SIP error reply ? I've tried a few ways but it did not work @message.header.Reason.params["cause"] or $hdr(Reason) gives null Thanks,

3 4

SNMP Monitoring
by Volkan Oransoy 09 Feb '14

09 Feb '14

Hi all, I have activated snmpstats module for my sip proxy. What I wonder is, which sensors are important to monitor a registrar and proxy? Which sensors do you use to monitor production servers? Thanks. /Volkan

1 0

Kamailio Admin Book
by omid mohajerani 09 Feb '14

09 Feb '14

Dear All , Would you mind tell me when is the Kamailio admin Book release time ? I want to start working with kamailio but it is very hard to start . Im sure this book will help me a lot . Regards Omid

2 2

Re: [SR-Users] Dialog
by Daniel-Constantin Mierla 09 Feb '14

09 Feb '14

Hello, do you need it in kamailio config or from an external application? From external application the mi/rpc command has to be used. Cheers, Daniel On 06/02/14 21:45, John Murray wrote: > > Hi, > > I need to get call-id and from-tag from a call in ringing state (2). > > If I use dlg_manage() and put the call into a profile using > set_dlg_profile("usr",”1"). > > Then do a avp_db_query() I don’t see the call until it is connected. > > I am using db_mode 1. … [View More]

3 7

"Locating SIP-Servers" problem
by Franz Edler 09 Feb '14

09 Feb '14

Dear all, I have a problem to verify the correct RFC 3263 behavior of Kamailio SIP Server v4.1. I have a basic configuration of two SIP servers one domains "net1.test" and the other for domain "net2.test". When I now setup a session between both domains (from sip:alice@net1.test to sip:bob@net2.test) I expect a NAPTR-, SRV- and A-query before forwarding the INVITE to the (inbound) proxy server of net2.test. But I instead get an error: 0(3994) ERROR: <core> [resolve.c:1733]: … [View More]

2 2

Troubles with PERL module in Kamailio 3.3.5 and 4.0.3
by neumann 09 Feb '14

09 Feb '14

Hi everybody. Sorry for my bad english. Then i trying to use OpenSER module in my perl script i got the error: Can't locate object method "bootstrap" via package "OpenSER" at /usr/lib64/kamailio/perl//OpenSER/Message.pm line 33. Compilation failed in require at /usr/lib64/kamailio/perl//OpenSER.pm line 34. BEGIN failed--compilation aborted at /usr/lib64/kamailio/perl//OpenSER.pm line 34. Compilation failed in require at ./qutest.pl line 4. BEGIN failed--compilation aborted at ./qutest.pl line … [View More]

2 1

Increasing MAX_RECV_BUFFER_SIZE
by Wingsravi R 09 Feb '14

09 Feb '14

Dear Kamailio'ns I am working on Kamailio (V 4.0)+ Mediaproxy server (2.5.2) running on Ubuntu 12.04. I am experiencing some Latency, jitter, pixelled, choppy video errors in my set-up. So i wanted to increase buffer sizes of my kamailio server , Mediaproxy server and Ubuntu also. In that regards i have increased Shared memory value to 512 MB and Package memory value 16 MB. BUFFER SIZE value to 8 MB in *config.h* file, But where can i change the value for 'MAX_RECV_BUFFER_SIZE' of kamailio? … [View More]

4 4

kamailo uses wrong ip address
by jaflong jaflong 07 Feb '14

07 Feb '14

I have this in my config $ru = "sip:"+$rU+"10.31.2.101:5090;transport=tls"; if (!t_relay_to_tls()) { sl_reply_error(); return; } My ip address dest is 10.31.2.101. Why has 130.31.2.101 been used? 13(10882) DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 130.31.2.101 4(10883) ERROR: <core> [tcp_main.c:4422]: tcpconn_main_timeout(): connect 130.31.2.101:5090 failed (timeout)

2 1

My Kamailio + Asterisk != NOT realtime integration: TRANSFERERNAME and BLINDTRANSFER
by Alexandr Usov 07 Feb '14

07 Feb '14

Integration - works. Problem - dialing peer to peer via Kamailio OK but with missing VARs and extension number, on dialing/transferring. Maybe you know other way to configure Asterisk dialplan for users, registered on kamailio and alowing dial as SIP/UserNumber insted SIP/ UserNumber(a)kamailio.host.name What we have. My Kamailio config: *http://pastebin.com/p7YxsFaw <http://pastebin.com/p7YxsFaw>* Asterisk's user (peer) - registered on kamailio: [3] host=192.168.144.212 qualify=… [View More]

1 1

CONFIRMED_NA dialog state timeout
by Alex Balashov 07 Feb '14

07 Feb '14

Hi, We frequently run into the situation where a call is simultaneously CANCEL'd by the caller and answered (2xx) by the callee. This results in the caller not sending an e2e ACK, since it CANCEL'd the branch. It results in retransmission of the 200 OK, since the proxy can't CANCEL the branch anymore. It leaves the dialog in state 3 #define DLG_STATE_CONFIRMED_NA 3 /*!< confirmed dialog without a ACK yet */ (dlg_hash.h) for up to 'dialog_timeout' time. Is there a … [View More]

1 0

ssl error
by jaflong jaflong 06 Feb '14

06 Feb '14

This is my tls.cfg for server [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /etc/asterisk/certs/proxy.key certificate = /etc/asterisk/certs/proxy.crt As far as I understand (verify_certificate = no), and (require_certificate = no) should allow a client connecting without certicates. Can anyone understand what this debug indicates What is causes this error tls_read_f(): TLS read:error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown … [View More]ca 7(14841) DEBUG: tls [tls_server.c:178]: tls_complete_init(): Using TLS domain TLSs<default> 7(14841) DEBUG: tls [tls_domain.c:609]: sr_ssl_ctx_info_callback(): SSL handshake started 7(14841) DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... 7(14841) DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7eff24e41260 n=940 fd=5 7(14841) DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf= 7(14841) DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x8df1e0, 5, 2, 0x7eff24e41260), fd_no=1 7(14841) DEBUG: tls [tls_domain.c:621]: sr_ssl_ctx_info_callback(): SSL handshake done 7(14841) DEBUG: tls [tls_domain.c:624]: sr_ssl_ctx_info_callback(): SSL disable renegotiation 7(14841) DEBUG: tls [tls_server.c:348]: tls_accept(): TLS accept successful 7(14841) DEBUG: tls [tls_server.c:355]: tls_accept(): tls_accept: new connection from 10.1.2.3:35573 using TLSv1/SSLv3 AES256-SHA 256 7(14841) DEBUG: tls [tls_server.c:358]: tls_accept(): tls_accept: local socket: 10.1.2.4:6443 7(14841) DEBUG: tls [tls_server.c:369]: tls_accept(): tls_accept: client did not present a certificate 7(14841) DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... 7(14841) DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7eff24e41260 n=250 fd=5 7(14841) DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf= 7(14841) DEBUG: <core> [tcp_read.c:296]: tcp_read_data(): EOF on 0x7eff24e41260, FD 5 7(14841) ERROR: tls [tls_server.c:1186]: tls_read_f(): TLS read:error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca 7(14841) ERROR: <core> [tcp_read.c:1281]: tcp_read_req(): ERROR: tcp_read_req: error reading 7(14841) DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x8df1e0, 5, -1, 0x10) fd_no=2 called 7(14841) DEBUG: <core> [tcp_read.c:1437]: release_tcpconn(): releasing con 0x7eff24e41260, state -2, fd=5, id=2 7(14841) DEBUG: <core> [tcp_read.c:1438]: release_tcpconn(): extra_data 0x7eff24e3c200 10(14844) DEBUG: <core> [tcp_main.c:3385]: handle_tcp_child(): handle_tcp_child: reader response= 7eff24e41260, -2 from 1 10(14844) DEBUG: tls [tls_server.c:597]: tls_h_close(): Closing SSL connection 0x7eff24e3c200 [View Less]

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Dialog
by John Murray 06 Feb '14

06 Feb '14

Hi, I need to get call-id and from-tag from a call in ringing state (2). If I use dlg_manage() and put the call into a profile using set_dlg_profile("usr","1"). Then do a avp_db_query() I don't see the call until it is connected. I am using db_mode 1. Yet if I do a sercmd dlg.dlg_list I see the call in ringing state and when it is connected. How do I get the call-id when it is in ringing state? Regards John

1 0

Change of source ip address
by Diego Alejandro Ozuna Escalada 06 Feb '14

06 Feb '14

Dear list, I'm having a problem. I have an SBC server with two network interfaces, one external IP (X.X.248.194) and other internal IP (X.X.248.21). Messages are sent to the proxy sip server through internal ip with this configuration: route [sip-serverr] { rewritehostport ("ip_sip_proxy: 5060"); force_send_socket (X.X.248.21: 5060); t_relay (); exit; } When a UA sends an INVITE to the SBC, the responses (back) to UA are being sent with the correct … [View More]

2 2

FS#380 - fixes for memory leak in TLS module
by Ding Ma 06 Feb '14

06 Feb '14

Daniel, Klaus, Sorry to reply late to your comments, was busy doing some other things until now. Just provided some comments in https://sip-router.org/tracker/index.php?do=details&task_id=380 Please take a look and let me know if you agree with my assessment. Thanks, Ding

1 0

bug in authdb_mod ??
by jay binks 06 Feb '14

06 Feb '14

So when Im trying to use is_subscriber & load_credentials together, It seems I have either hit a bug or just a bad config ( but I dont think its the latter ) My symptom is that only the field on the far right of "load_credentials" ever returns the correct value, the other fields are always null. My config has this setting : modparam("auth_db", "load_credentials", "field1;field2") and I can see the following debug when running kamailio -ddddd 0(14174) DEBUG: auth_db [authdb_mod.c:291]: … [View More]

2 5

Handling RFC3578 (Overlap Dialing) in Kamailio/ Asynchronious transaction handling
by Moritz Graf 06 Feb '14

06 Feb '14

Hoho, i want to implement RFC3578 (transaction oriented) in Kamailio. Therefore I have to correlate two different Invites with same Call-ID. I see various ways of accomplishing this, but I am stuck at all of my solutions. Find the algorithm I want to implement attached. Shortly explained what RFC3578 is: In a open numbering plan you never know if the INVITE you received is already complete, or if there are more numbers coming in. One way of accomplishing this is to set up a timer. If the … [View More]timer elapses you assume the number is complete. If not, you are receiving a new INVITE with one digit more. Now you have to close the old transaction with a "484 - Address Incomplete"-response and start the timer again. (Find the algorithm I want to implement attached) The following lines, I'm going to explain my thoughts. For reference I'm calling "Thread A" the first INVITE coming in and "Thread B" the second. ####################### ### using Module ASYNC * Thread A -> async_route("handle_overlap","<timer t99 value>"); * Thread B -> if new incoming transaction, ** set avp("<call-id>")= <digitlength of called number> and ** call async_route("handle_overlap","<timer t99 value>"); * in route[handle_overlap] ** if avp("call-id") greater than "<digitlength of called number>" *** cancel transaction with "484 -Address Incomplete" ** if avp("call-id") equals "<digitlength of called number>" *** handle call => the number is complete PROBLEM: The old transaction is not canceled immediately, but after t99 elapsed... ####################### ### using Module TMX using t_suspend() and t_continue(), but here i see the following problems: * Thread A calls "t_suspend()" ** How is it possible to cancel "Thread A"'s transaction from Thread B? ** If there is no Thread B, so the numbers were complete, how can I get Thread A to continue processing?? * General TMX-Question: Are there any examples available? ####################### ### using simple sleep() and shv()/avp() Sounds like dirty hacking... I'd appriciate a solution with one of aboves modules. But if there is a solution available, I'd try this. Thx in advise Moritz -- Moritz Graf, B.Sc. Betrieb NGN-Plattform G-FIT GmbH & Co. KG Greflingerstr. 26, 93055 Regensburg Telefon +49 (9 41) 69 85 - 1 86 Telefax +49 (9 41) 69 85 - 2 86 mailto:moritz.graf@g-fit.de http://www.g-fit.de G-FIT Gesellschaft für innovative Telekommunikationsdienste mbH & Co. KG, Kommanditgesellschaft, Sitz Regensburg, Registergericht Regensburg, HRA 7626; Geschäftsführer: Dipl.Inf. (FH) Alfred Rauscher [View Less]

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Kamailio + CDR-stats
by davy 06 Feb '14

06 Feb '14

Hi, I’ve recently tried out CDR-stats, and after a bumpy ride to get it going ;) , I’m happily surprised with it. Especially the fraud detection mechanisms (which could use some extra’s), and the eye-candy makes it attractive. It is based on a mongodb which has to be filled with CDRs. As there is no mongodb module in Kamailio, I was thinking of using the jsonrpc module to push cdrs from Kamailio to some Python thing which pushes it into MongoDB. Perhaps by using the async module. … [View More]

3 2

dialog module - dialogs not deleted on BYE
by jay binks 06 Feb '14

06 Feb '14

Im fairly use to the dialog module, and I can not seem to get the dialogs to be removed from the dialogs table on hangup ( BYE ). What am I missing here.. this seems very basic and is essential to using dialog module for for channel limiting. any help would be great. maybe there is an example config for this somewhere ? -- Sincerely Jay

2 3

MI commands ?
by arun Jayaprakash 06 Feb '14

06 Feb '14

Hello, what do I need to do to be able to run MI commands on Kamailio 4.0? When I run MI commands from Siremis no results come back. Thank you. Arun

2 4

kamailio as sip router and rtp proxy without reg on kamailio
by *sm1Ly 05 Feb '14

05 Feb '14

hello all again. trying the third day... my cfg is: http://pastebin.com/vMHd7kki I need scheme like: any_sip_device_on_internet_nated_and_not => kamailio ext ip (194.190.8.171) | kamailio int ip (50.0.0.1) => asterisk (50.0.0.10) a do not need realtime configuration of asterisk. I only need forward everything to asterisk. and after signalling will work, I need rtp too... but its the long way I think. so how I must config kamailio that it doesnt answer 200 when I send register? it … [View More]looks like sip device send register, kamailio fwd it, but answer to device 200. and than my device doesnt resend register normal. some dump: 20:40:44.649660 IP (tos 0x0, ttl 121, id 14950, offset 0, flags [none], proto UDP (17), length 583) 77.37.241.151.5060 > 194.190.8.171.5060: SIP, length: 555 REGISTER sip:194.190.8.171 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.119:5060 ;branch=z9hG4bK009318ba288ce311b95a985a85ad1a82;rport From: <sip:1001@194.190.8.171>;tag=3041452802 To: <sip:1001@194.190.8.171> Call-ID: 00C48A3A-288C-E311-B956-985A85AD1A82(a)192.168.1.119 CSeq: 4 REGISTER Contact: <sip:1001@192.168.1.119:5060 >;Expires=0;+sip.instance="<urn:uuid:8012D316-1188-E311-99FF-536675A7CDE6>" Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 User-Agent: SIPPER for PhonerLite Expires: 0 Content-Length: 0 20:40:44.651668 IP (tos 0x10, ttl 64, id 33739, offset 0, flags [none], proto UDP (17), length 422) 194.190.8.171.5060 > 77.37.241.151.5060: SIP, length: 394 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.119:5060 ;branch=z9hG4bK009318ba288ce311b95a985a85ad1a82;rport=5060;received=77.37.241.151 From: <sip:1001@194.190.8.171>;tag=3041452802 To: <sip:1001@194.190.8.171 >;tag=c5aea4efd01978ba46557f38f3d08cfc.63a1 Call-ID: 00C48A3A-288C-E311-B956-985A85AD1A82(a)192.168.1.119 CSeq: 4 REGISTER Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 20:40:44.653916 IP (tos 0x10, ttl 64, id 33740, offset 0, flags [none], proto UDP (17), length 582) 194.190.8.171.5060 > 77.37.241.151.5060: SIP, length: 554 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.119:5060 ;received=77.37.241.151;branch=z9hG4bK009318ba288ce311b95a985a85ad1a82;rport=5060 From: <sip:1001@194.190.8.171>;tag=3041452802 To: <sip:1001@194.190.8.171>;tag=as40c64a63 Call-ID: 00C48A3A-288C-E311-B956-985A85AD1A82(a)192.168.1.119 CSeq: 4 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4cffbc95" Content-Length: 0 20:40:45.073854 IP (tos 0x0, ttl 121, id 14953, offset 0, flags [none], proto UDP (17), length 575) 77.37.241.151.5060 > 194.190.8.171.5060: SIP, length: 547 REGISTER sip:194.190.8.171 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.119:5060 ;branch=z9hG4bK009318ba288ce311b1713a64da149374;rport From: <sip:1001@194.190.8.171>;tag=3123771917 To: <sip:1001@194.190.8.171> Call-ID: 009318BA-288C-E311-B16F-3A64DA149374(a)192.168.1.119 CSeq: 1 REGISTER Contact: <sip:1001@192.168.1.119:5060 >;+sip.instance="<urn:uuid:8012D316-1188-E311-99FF-536675A7CDE6>" Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 User-Agent: SIPPER for PhonerLite Expires: 900 Content-Length: 0 20:40:45.076169 IP (tos 0x10, ttl 64, id 33741, offset 0, flags [none], proto UDP (17), length 572) 194.190.8.171.5060 > 77.37.241.151.5060: SIP, length: 544 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.119:5060 ;branch=z9hG4bK009318ba288ce311b1713a64da149374;rport=5060;received=77.37.241.151 From: <sip:1001@194.190.8.171>;tag=3123771917 To: <sip:1001@194.190.8.171 >;tag=c5aea4efd01978ba46557f38f3d08cfc.795d Call-ID: 009318BA-288C-E311-B16F-3A64DA149374(a)192.168.1.119 CSeq: 1 REGISTER Contact: <sip:1001@192.168.1.119:5060>;expires=900;received="sip: 77.37.241.151:5060 ";+sip.instance="<urn:uuid:8012D316-1188-E311-99FF-536675A7CDE6>" Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 -- yours respectfully, Alexander Vasin. 8 926 1437200 icq: 9906064 [View Less]

2 1

NAT helper frustrations with multiple kamailios
by Daniel Tryba 05 Feb '14

05 Feb '14

I'm experiencing some problems getting 2 Kamailios to work together. A clients - A kamailio - B kamailio - B clients | the outside world A and B client may or may not be NATed to their own kamailio instance. A and B kamailio (with forced rtpproxies) are in the same public network. A-kamailio and its clients are essentially a reseller of my services on the B side. The problem arises when A or B incorrectly flag a message as needing NAT help … [View More]

3 3

Re: [SR-Users] Kamailio not mdifying Contact in multi-homed (*sm1Ly)
by *sm1Ly 05 Feb '14

05 Feb '14

I got really modify my config, and now it looks like: http://pastebin.com/0M7938Mm but now I got loopback problems. kamailio just asking itself. ngrep -d lo port 5060 talks me: U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;branch=z9hG4bKc2bb.e627f45ee16a4b45ca99321b3bc8832c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.328c8c23df8e916010525f7f786cbc49.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060 ;branch=… [View More]z9hG4bKc2bb.b8ec8f173657be182ad714f59d37d4a8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.7bec0e7c4a5962e90811bf082f0e6635.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6bbb824e92f328ce4053167b2e2c596a.0..V ia: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6f732fece1d8956bf5b075118b3dcb2f.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b4ab8b8de819be6fff9203bfdf4e758c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2 bb.977524b1111849dd2cd8020d4c09090c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 1 94.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a 7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rpor t=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060 ;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988 c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x 86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.328c8c23df8e916010525f7f786cbc49.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b8ec8f173657be182ad714f59d37d4a8.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.7bec0e7c4a5962e90811bf082f0e6635.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6bbb824e92f328ce4053167b2e2c596a.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6f732fece1d8956bf5b075118b 3dcb2f.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b4ab8b8de819be6fff9203bfdf4e758c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.977524b1111849dd2cd8020d4c09090c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SI P/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01d cad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190 .8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001 @194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b8ec8f173657be182ad714f59d37d4a8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.7bec0e7c4a5962e90811bf082f0e6635.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.6bbb824e92f328ce4053167b2e2c596a.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6f732fece1d8956bf5b075118b3dcb2f.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b4ab8b8de819be6fff9203bfdf 4e758c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.977524b1111849dd2cd8020d4c09090c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SI P/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946 cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168 .1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757D AE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.7bec0e7c4a5962e90811bf082f0e6635.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6bbb824e92f328ce4053167b2e2c596a.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.6f732fece1d8956bf5b075118b3dcb2f.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b4ab8b8de819be6fff9203bfdf4e758c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.977524b1111849dd2cd8020d4c 09090c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SI P/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0ea f0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060 ;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..Fro m: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length : 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6bbb824e92f328ce4053167b2e2c596a.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6f732fece1d8956bf5b075118b3dcb2f.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.b4ab8b8de819be6fff9203bfdf4e758c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.977524b1111849dd2cd8020d4c09090c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b 43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SI P/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d 28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171 >;tag=c5aea4efd01978ba4655 7f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.6f732fece1d8956bf5b075118b3dcb2f.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b4ab8b8de819be6fff9203bfdf4e758c.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.977524b1111849dd2cd8020d4c09090c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0 c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SI P/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branc h=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: < sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE ..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b4ab8b8de819be6fff9203bfdf4e758c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.977524b1111849dd2cd8020d4c09090c.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e 520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SI P/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=12744772 58..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.977524b1111849dd2cd8020d4c09090c.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf7 3e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SI P/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558 B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.cb7d178bf3fdb6596399beaf5b43e66d.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c 7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060 ;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14df b;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: < sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux) )..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.bd5d0e524121d2fd121fa6e8f0c69cbf.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66 b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branc h=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171 >;tag=c5a ea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.98352f1121cb37b6feb7814a0e520e4e.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173 b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77 .37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB(a)192.168.1.119.. CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.b670b117fb4d3a40912cb25cf73e9e32.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc28 04cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8. 171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.ac9e718c5fd57a7f73adc7761c7ba31b.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d 2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171 >;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call -ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.01dcad6cb7e8257bcbe8ad7b66b7b095.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060 ;received=77.37.241.151;branch=z9hG4bK00c46d3a558be3 11988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1 .1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.946cc81be473148e1a6dc5a173b2cfb1.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171 ;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.1 90.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.0eaf0b545ac8239ad5e105fc2804cae8.0..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119 :5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <sip:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DF B@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0.... # U 194.190.8.171:5060 -> 194.190.8.171:5060 SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 194.190.8.171;rport=5060;branch=z9hG4bKc2bb.c1d28c8a5ba1eeda8f8b296f6d2232e8.0..Via: SIP/2.0/UDP 192.168.1.119:5060;received=77.37.241.151;branch=z9hG4bK00c46d3a558be311988c4757dae14dfb;rport=5060..From: <si p:1001@194.190.8.171>;tag=1274477258..To: <sip:1001@194.190.8.171>;tag=c5aea4efd01978ba46557f38f3d08cfc.2793..Call-ID: 00C46D3A-558B-E311-988A-4757DAE14DFB@192.168.1.119..CSeq: 2 SUBSCRIBE..Server: kamailio (4.1.1 (x86_64/linux))..Content-Length: 0... . and how looks dump 1. from sip device to kamailio [root@kamaz sm1ly]# tcpdump -nn -i eth0 host 77.37.241.151 -v tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes 19:54:51.205942 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 421) 77.37.241.151.57288 > 194.190.8.171.5060: SIP, length: 393 REGISTER sip:194.190.8.171 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5569 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bKe5bd1a6ceba7f0a0f4e7487cb17ca365353236;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Content-Length: 0 19:54:51.210060 IP (tos 0x10, ttl 64, id 35861, offset 0, flags [none], proto UDP (17), length 603) 194.190.8.171.5060 > 77.37.241.151.57288: SIP, length: 575 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKe5bd1a6ceba7f0a0f4e7487cb17ca365353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as518dbea9 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5569 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31daf0f8" Content-Length: 0 19:54:51.257552 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 586) 77.37.241.151.57288 > 194.190.8.171.5060: SIP, length: 558 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5570 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bK440337c23337c9913220fd825f4d9ded353236;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Authorization: Digest username="1002",realm="asterisk",nonce="31daf0f8",uri="sip: 194.190.8.171:5060 ",response="2b0222b07da7af1c3b72c2eb36241f5f",algorithm=MD5 Content-Length: 0 19:54:51.261209 IP (tos 0x10, ttl 64, id 35862, offset 0, flags [none], proto UDP (17), length 603) 194.190.8.171.5060 > 77.37.241.151.57288: SIP, length: 575 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK440337c23337c9913220fd825f4d9ded353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as0716dea5 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5570 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f70c759" Content-Length: 0 19:54:51.290458 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 586) 77.37.241.151.57288 > 194.190.8.171.5060: SIP, length: 558 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5571 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bK30fe030eee49e2eb50c619074c0f057b353236;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Authorization: Digest username="1002",realm="asterisk",nonce="2f70c759",uri="sip: 194.190.8.171:5060 ",response="9eb7d157cfd0d8e8575b59c94c5f82f6",algorithm=MD5 Content-Length: 0 19:54:51.295285 IP (tos 0x10, ttl 64, id 35863, offset 0, flags [none], proto UDP (17), length 566) 194.190.8.171.5060 > 77.37.241.151.57288: SIP, length: 538 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK30fe030eee49e2eb50c619074c0f057b353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as0716dea5 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5571 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 0 Date: Mon, 03 Feb 2014 15:54:51 GMT Content-Length: 0 19:54:51.314824 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 473) 77.37.241.151.57288 > 194.190.8.171.5060: SIP, length: 445 REGISTER sip:194.190.8.171 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2404 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bKdeb946998664720c2c93538625cc649e353236;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Content-Length: 0 19:54:51.319697 IP (tos 0x10, ttl 64, id 35864, offset 0, flags [none], proto UDP (17), length 602) 194.190.8.171.5060 > 77.37.241.151.57288: SIP, length: 574 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKdeb946998664720c2c93538625cc649e353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as0bba2324 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2404 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="023deb34" Content-Length: 0 19:54:51.344080 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 638) 77.37.241.151.57288 > 194.190.8.171.5060: SIP, length: 610 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2405 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bK60a6798f72024444a9870698b766a04c353236;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Authorization: Digest username="1002",realm="asterisk",nonce="023deb34",uri="sip: 194.190.8.171:5060 ",response="7fa5be98c7c0c395ed41f9e7da095629",algorithm=MD5 Content-Length: 0 19:54:51.350423 IP (tos 0x10, ttl 64, id 35865, offset 0, flags [none], proto UDP (17), length 602) 194.190.8.171.5060 > 77.37.241.151.57288: SIP, length: 574 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK60a6798f72024444a9870698b766a04c353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as065abeba Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2405 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26886c3d" Content-Length: 0 19:54:51.370976 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 638) 77.37.241.151.57288 > 194.190.8.171.5060: SIP, length: 610 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2406 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bKf4d831e0612e1074bf49397d92907c80353236;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Authorization: Digest username="1002",realm="asterisk",nonce="26886c3d",uri="sip: 194.190.8.171:5060 ",response="cc17aae74d45f06e4712aa66b08e68df",algorithm=MD5 Content-Length: 0 19:54:51.380962 IP (tos 0x10, ttl 64, id 35866, offset 0, flags [none], proto UDP (17), length 636) 194.190.8.171.5060 > 77.37.241.151.57288: SIP, length: 608 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKf4d831e0612e1074bf49397d92907c80353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as065abeba Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2406 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: <sip:1002@192.168.1.103:57288;transport=udp>;expires=3600 Date: Mon, 03 Feb 2014 15:54:51 GMT Content-Length: 0 19:54:51.406160 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 395) 77.37.241.151.57288 > 194.190.8.171.5060: SIP, length: 367 OPTIONS sip:194.190.8.171 SIP/2.0 Call-ID: 28b382a24bc3e426f030593cec5f852a(a)192.168.1.103 CSeq: 1287 OPTIONS From: "1002" <sip:1002@194.190.8.171>;tag=3866552602 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bK9bddae72a5ccaf0df80be00ba2882dfd353236;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Content-Length: 0 19:54:51.407799 IP (tos 0x10, ttl 64, id 35867, offset 0, flags [none], proto UDP (17), length 458) 194.190.8.171.5060 > 77.37.241.151.57288: SIP, length: 430 SIP/2.0 484 Address Incomplete Call-ID: 28b382a24bc3e426f030593cec5f852a(a)192.168.1.103 CSeq: 1287 OPTIONS From: "1002" <sip:1002@194.190.8.171>;tag=3866552602 To: "1002" <sip:1002@194.190.8.171 >;tag=c5aea4efd01978ba46557f38f3d08cfc.5162 Via: SIP/2.0/UDP 192.168.1.103:57288 ;branch=z9hG4bK9bddae72a5ccaf0df80be00ba2882dfd353236;rport=57288;received=77.37.241.151 Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 2. from kamailio to asterisk. [root@kamaz sm1ly]# tcpdump -nn -i eth1 host 50.0.0.10 -v and port 5060 tcpdump: listening on eth1, link-type EN10MB (Ethernet), capture size 65535 bytes 19:54:41.906391 IP (tos 0x0, ttl 64, id 5618, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:34713;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:42.906385 IP (tos 0x0, ttl 64, id 5619, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:34713;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:43.906565 IP (tos 0x0, ttl 64, id 5620, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:34713;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:44.904726 IP (tos 0x10, ttl 64, id 34743, offset 0, flags [none], proto UDP (17), length 413) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 385 SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport=5060 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713 ;transport=udp>;tag=6f6209fafb0c4bc816e1e7aa854ee2dd-8578 Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 19:54:51.207422 IP (tos 0x10, ttl 64, id 34744, offset 0, flags [none], proto UDP (17), length 531) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 503 REGISTER sip:194.190.8.171 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5569 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bKd8a1.094de0504827adb22c7b874f77dfd698.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKe5bd1a6ceba7f0a0f4e7487cb17ca365353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Content-Length: 0 19:54:51.209052 IP (tos 0x0, ttl 64, id 5621, offset 0, flags [none], proto UDP (17), length 713) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 685 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bKd8a1.094de0504827adb22c7b874f77dfd698.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKe5bd1a6ceba7f0a0f4e7487cb17ca365353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as518dbea9 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5569 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31daf0f8" Content-Length: 0 19:54:51.258759 IP (tos 0x10, ttl 64, id 34745, offset 0, flags [none], proto UDP (17), length 696) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 668 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5570 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK56b1.d46d0ab6448c5edd99f24aa7ba8a2881.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK440337c23337c9913220fd825f4d9ded353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Authorization: Digest username="1002",realm="asterisk",nonce="31daf0f8",uri="sip: 194.190.8.171:5060 ",response="2b0222b07da7af1c3b72c2eb36241f5f",algorithm=MD5 Content-Length: 0 19:54:51.259902 IP (tos 0x0, ttl 64, id 5622, offset 0, flags [none], proto UDP (17), length 713) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 685 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK56b1.d46d0ab6448c5edd99f24aa7ba8a2881.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK440337c23337c9913220fd825f4d9ded353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as0716dea5 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5570 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f70c759" Content-Length: 0 19:54:51.292190 IP (tos 0x10, ttl 64, id 34746, offset 0, flags [none], proto UDP (17), length 696) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 668 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5571 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK66b1.0173e2524e6deda0f27b83591a62730f.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK30fe030eee49e2eb50c619074c0f057b353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Authorization: Digest username="1002",realm="asterisk",nonce="2f70c759",uri="sip: 194.190.8.171:5060 ",response="9eb7d157cfd0d8e8575b59c94c5f82f6",algorithm=MD5 Content-Length: 0 19:54:51.293302 IP (tos 0x0, ttl 64, id 5623, offset 0, flags [none], proto UDP (17), length 676) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 648 SIP/2.0 200 OK Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK66b1.0173e2524e6deda0f27b83591a62730f.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK30fe030eee49e2eb50c619074c0f057b353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as0716dea5 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5571 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 0 Date: Mon, 03 Feb 2014 15:54:51 GMT Content-Length: 0 19:54:51.317242 IP (tos 0x10, ttl 64, id 34747, offset 0, flags [none], proto UDP (17), length 583) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 555 REGISTER sip:194.190.8.171 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2404 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK2624.44980ae0ad3ba5026d6f044c24770539.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKdeb946998664720c2c93538625cc649e353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Content-Length: 0 19:54:51.318123 IP (tos 0x0, ttl 64, id 5624, offset 0, flags [none], proto UDP (17), length 712) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 684 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK2624.44980ae0ad3ba5026d6f044c24770539.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKdeb946998664720c2c93538625cc649e353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as0bba2324 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2404 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="023deb34" Content-Length: 0 19:54:51.346522 IP (tos 0x10, ttl 64, id 34748, offset 0, flags [none], proto UDP (17), length 748) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 720 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2405 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK3624.173cef757dc8f7da8b864ee1ab1bfe50.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK60a6798f72024444a9870698b766a04c353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Authorization: Digest username="1002",realm="asterisk",nonce="023deb34",uri="sip: 194.190.8.171:5060 ",response="7fa5be98c7c0c395ed41f9e7da095629",algorithm=MD5 Content-Length: 0 19:54:51.348056 IP (tos 0x0, ttl 64, id 5625, offset 0, flags [none], proto UDP (17), length 712) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 684 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK3624.173cef757dc8f7da8b864ee1ab1bfe50.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK60a6798f72024444a9870698b766a04c353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as065abeba Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2405 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26886c3d" Content-Length: 0 19:54:51.373298 IP (tos 0x10, ttl 64, id 34749, offset 0, flags [none], proto UDP (17), length 748) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 720 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2406 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK0624.82ac15efd62f203c1b009e3ab3c711bc.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKf4d831e0612e1074bf49397d92907c80353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Authorization: Digest username="1002",realm="asterisk",nonce="26886c3d",uri="sip: 194.190.8.171:5060 ",response="cc17aae74d45f06e4712aa66b08e68df",algorithm=MD5 Content-Length: 0 19:54:51.374893 IP (tos 0x0, ttl 64, id 5626, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:57288;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:51.374906 IP (tos 0x0, ttl 64, id 5627, offset 0, flags [none], proto UDP (17), length 746) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 718 SIP/2.0 200 OK Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK0624.82ac15efd62f203c1b009e3ab3c711bc.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKf4d831e0612e1074bf49397d92907c80353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as065abeba Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2406 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: <sip:1002@192.168.1.103:57288;transport=udp>;expires=3600 Date: Mon, 03 Feb 2014 15:54:51 GMT Content-Length: 0 19:54:52.375165 IP (tos 0x0, ttl 64, id 5628, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:57288;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:53.375333 IP (tos 0x0, ttl 64, id 5629, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:57288;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:54.342025 IP (tos 0x10, ttl 64, id 34750, offset 0, flags [none], proto UDP (17), length 413) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 385 SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport=5060 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288 ;transport=udp>;tag=6f6209fafb0c4bc816e1e7aa854ee2dd-1f6e Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 3. on asterisk to kamailio [root@astkamaz sm1ly]# tcpdump -nn -i eth0 -v host 50.0.0.1 and port 5060 tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes 19:54:41.906401 IP (tos 0x0, ttl 64, id 5618, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:34713;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:42.906581 IP (tos 0x0, ttl 64, id 5619, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:34713;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:43.906740 IP (tos 0x0, ttl 64, id 5620, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:34713;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:44.905366 IP (tos 0x10, ttl 64, id 34743, offset 0, flags [none], proto UDP (17), length 413) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 385 SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK75f42160;rport=5060 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40fc69ea To: <sip:1002@192.168.1.103:34713 ;transport=udp>;tag=6f6209fafb0c4bc816e1e7aa854ee2dd-8578 Call-ID: 7bacfa62676409266240568c5c9d055b@50.0.0.10:5060 CSeq: 102 OPTIONS Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 19:54:51.208464 IP (tos 0x10, ttl 64, id 34744, offset 0, flags [none], proto UDP (17), length 531) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 503 REGISTER sip:194.190.8.171 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5569 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bKd8a1.094de0504827adb22c7b874f77dfd698.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKe5bd1a6ceba7f0a0f4e7487cb17ca365353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Content-Length: 0 19:54:51.209176 IP (tos 0x0, ttl 64, id 5621, offset 0, flags [none], proto UDP (17), length 713) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 685 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bKd8a1.094de0504827adb22c7b874f77dfd698.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKe5bd1a6ceba7f0a0f4e7487cb17ca365353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as518dbea9 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5569 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31daf0f8" Content-Length: 0 19:54:51.259436 IP (tos 0x10, ttl 64, id 34745, offset 0, flags [none], proto UDP (17), length 696) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 668 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5570 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK56b1.d46d0ab6448c5edd99f24aa7ba8a2881.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK440337c23337c9913220fd825f4d9ded353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Authorization: Digest username="1002",realm="asterisk",nonce="31daf0f8",uri="sip: 194.190.8.171:5060 ",response="2b0222b07da7af1c3b72c2eb36241f5f",algorithm=MD5 Content-Length: 0 19:54:51.260101 IP (tos 0x0, ttl 64, id 5622, offset 0, flags [none], proto UDP (17), length 713) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 685 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK56b1.d46d0ab6448c5edd99f24aa7ba8a2881.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK440337c23337c9913220fd825f4d9ded353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as0716dea5 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5570 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f70c759" Content-Length: 0 19:54:51.292708 IP (tos 0x10, ttl 64, id 34746, offset 0, flags [none], proto UDP (17), length 696) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 668 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5571 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK66b1.0173e2524e6deda0f27b83591a62730f.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK30fe030eee49e2eb50c619074c0f057b353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: * Expires: 0 Authorization: Digest username="1002",realm="asterisk",nonce="2f70c759",uri="sip: 194.190.8.171:5060 ",response="9eb7d157cfd0d8e8575b59c94c5f82f6",algorithm=MD5 Content-Length: 0 19:54:51.293541 IP (tos 0x0, ttl 64, id 5623, offset 0, flags [none], proto UDP (17), length 676) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 648 SIP/2.0 200 OK Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK66b1.0173e2524e6deda0f27b83591a62730f.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK30fe030eee49e2eb50c619074c0f057b353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=2801080754 To: "1002" <sip:1002@194.190.8.171>;tag=as0716dea5 Call-ID: 676c32ab221a665a9bc9875ffb74930a(a)192.168.1.103 CSeq: 5571 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 0 Date: Mon, 03 Feb 2014 15:54:51 GMT Content-Length: 0 19:54:51.317704 IP (tos 0x10, ttl 64, id 34747, offset 0, flags [none], proto UDP (17), length 583) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 555 REGISTER sip:194.190.8.171 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2404 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK2624.44980ae0ad3ba5026d6f044c24770539.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKdeb946998664720c2c93538625cc649e353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Content-Length: 0 19:54:51.318355 IP (tos 0x0, ttl 64, id 5624, offset 0, flags [none], proto UDP (17), length 712) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 684 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK2624.44980ae0ad3ba5026d6f044c24770539.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKdeb946998664720c2c93538625cc649e353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as0bba2324 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2404 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="023deb34" Content-Length: 0 19:54:51.347429 IP (tos 0x10, ttl 64, id 34748, offset 0, flags [none], proto UDP (17), length 748) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 720 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2405 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK3624.173cef757dc8f7da8b864ee1ab1bfe50.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK60a6798f72024444a9870698b766a04c353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Authorization: Digest username="1002",realm="asterisk",nonce="023deb34",uri="sip: 194.190.8.171:5060 ",response="7fa5be98c7c0c395ed41f9e7da095629",algorithm=MD5 Content-Length: 0 19:54:51.348212 IP (tos 0x0, ttl 64, id 5625, offset 0, flags [none], proto UDP (17), length 712) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 684 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK3624.173cef757dc8f7da8b864ee1ab1bfe50.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bK60a6798f72024444a9870698b766a04c353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as065abeba Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2405 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26886c3d" Content-Length: 0 19:54:51.373777 IP (tos 0x10, ttl 64, id 34749, offset 0, flags [none], proto UDP (17), length 748) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 720 REGISTER sip:194.190.8.171:5060 SIP/2.0 Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2406 REGISTER From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171> Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK0624.82ac15efd62f203c1b009e3ab3c711bc.0 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKf4d831e0612e1074bf49397d92907c80353236;rport=57288 Max-Forwards: 16 User-Agent: SIPAUA/0.1.001 Contact: "1002" <sip:1002@192.168.1.103:57288;transport=udp> Expires: 3600 Authorization: Digest username="1002",realm="asterisk",nonce="26886c3d",uri="sip: 194.190.8.171:5060 ",response="cc17aae74d45f06e4712aa66b08e68df",algorithm=MD5 Content-Length: 0 19:54:51.374731 IP (tos 0x0, ttl 64, id 5626, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:57288;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:51.374804 IP (tos 0x0, ttl 64, id 5627, offset 0, flags [none], proto UDP (17), length 746) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 718 SIP/2.0 200 OK Via: SIP/2.0/UDP 50.0.0.1;branch=z9hG4bK0624.82ac15efd62f203c1b009e3ab3c711bc.0;received=50.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.103:57288 ;received=77.37.241.151;branch=z9hG4bKf4d831e0612e1074bf49397d92907c80353236;rport=57288 From: "1002" <sip:1002@194.190.8.171>;tag=134783744 To: "1002" <sip:1002@194.190.8.171>;tag=as065abeba Call-ID: fffb7a09e489ce1cad6783709652b6f9(a)192.168.1.103 CSeq: 2406 REGISTER Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: <sip:1002@192.168.1.103:57288;transport=udp>;expires=3600 Date: Mon, 03 Feb 2014 15:54:51 GMT Content-Length: 0 19:54:52.375348 IP (tos 0x0, ttl 64, id 5628, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:57288;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:53.375562 IP (tos 0x0, ttl 64, id 5629, offset 0, flags [none], proto UDP (17), length 597) 50.0.0.10.5060 > 50.0.0.1.5060: SIP, length: 569 OPTIONS sip:1002@192.168.1.103:57288;transport=udp SIP/2.0 Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288;transport=udp> Contact: <sip:asterisk@50.0.0.10:5060> Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.20.0 Date: Mon, 03 Feb 2014 15:54:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 19:54:54.342525 IP (tos 0x10, ttl 64, id 34750, offset 0, flags [none], proto UDP (17), length 413) 50.0.0.1.5060 > 50.0.0.10.5060: SIP, length: 385 SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 50.0.0.10:5060;branch=z9hG4bK129c83a7;rport=5060 From: "asterisk" <sip:asterisk@50.0.0.10>;tag=as40306b56 To: <sip:1002@192.168.1.103:57288 ;transport=udp>;tag=6f6209fafb0c4bc816e1e7aa854ee2dd-1f6e Call-ID: 0e6d3cec7afd68275843c14303988ff3@50.0.0.10:5060 CSeq: 102 OPTIONS Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 ####### Defined Values ######### # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### # Enable debugging only when kamailio is mading boo-boo ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR debug=3 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=8 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ #alias="kamaz.ikatele.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:50.0.0.1:5060 advertise 194.190.8.171:5060 #listen=udp:194.190.8.171:5060 advertise 50.0.0.1:5060 listen=udp:194.190.8.171 listen=udp:50.0.0.1 dns=no rev_dns=no check_via=yes /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 mhomed=1 ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) mpath="/usr/lib64/kamailio/modules/" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "nathelper.so" loadmodule "rtpproxy.so" #loadmodule "topoh.so" #modparam("topoh", "mask_key", "balalayka") #modparam("topoh", "mask_ip", "50.0.0.1") # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 3000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 12000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@194.190.8.171") # params needed for NAT traversal in other modules modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } route { # per request initial checks route(REQINIT); # NAT detection xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) Main Route before ---NAT---\n"); route(NATDETECT); xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in Route[NATDETECT] fix_nat-register\n"); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } route(RELAY); } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("20")) { xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in MAX FDW HEADERS ALARMO\n"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Caller NAT detection route route[NATDETECT] { force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } return; } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { xlog("L_NOTICE","ACK without matching transaction ... ignore and discard\n"); # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { # if(isflagset(FLT_NATS)) # { # setbflag(FLB_NATB); # # uncomment next line to do SIP NAT pinging # xlog("L_NOTICE","--------------------- SIP NAT PINGING --------------------------------------\n"); # setbflag(FLB_NATSIPPING); # } # if (!save("location")) # sl_reply_error(); # xlog("L_NOTICE","--------------------- SL REPLY ERROR --------------------------------------\n"); if( t_check_trans() ) { xlog("L_NOTICE","Transaction already exist, forwarding\n"); } else { xlog("L_NOTICE","New transaction upcoming, creating\n"); # forward("50.0.0.10"); # save("sip:50.0.0.10","0x02"); t_relay_to_udp("50.0.0.10", "5060"); exit; } exit; } } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. xlog("L_NOTICE","Default routing block raised!\n"); if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # RTPProxy control route[NATMANAGE] { if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xlog("L_NOTICE","------------------------------------ SETBFLAG ----------------------------\n"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xlog("L_NOTICE","------------------------------------ ADD RR PARAM NAT YES ----------------------------\n"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) xlog("L_NOTICE","------------------------------------ IS FIRST HOP SRT CONACT ALIAS ----------------------------\n"); set_contact_alias(); } xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in route[NATMANAGE] RTPproxy with EI Flags\n"); } return; } # URI update for dialog requests route[DLGURI] { if(!isdsturiset()) { handle_ruri_alias(); xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in route[DLGURI] RTPproxy with EI Flags\n"); } return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in MANAGE_BRANCH\n"); xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in MANAGE_REPLY\n"); xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in MANAGE_FAILURE\n"); route(NATMANAGE); if (t_is_canceled()) { exit; } } route[DEBUG] { if (method==("PUBLISH|SUBSCRIBE|REGISTER|OPTIONS")) { return; } xlog("[$mi] Received SIP Message (method: $rm) ($ml bytes) to $Ri:$Rp from $si:$sp:\n$mb\n"); } > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 3 Feb 2014 15:14:19 +0400 > From: "*sm1Ly" <st.uzver(a)gmail.com> > To: dipak.biswas(a)mezocliq.com, sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Kamailio not mdifying Contact in multi-homed > setup > Message-ID: > < > CABCivXiED_CDQiJtgQ-p_XQ1h8p3rbkE3fnxnS2iuUH+kdHZvA(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > hello, I got the same issue. my sip device getting ok from kamailio, but > asterisk reply 401. > but. > this is my config: http://pastebin.com/jGCak01E > > and I changed regfwd route to > $du = "sip:50.0.0.10:5060;transport=udp"; > forward(); > > and now I see this in logs: > > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:633]: parse_msg(): SIP Reply (status): > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:635]: parse_msg(): version: <SIP/2.0> > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:637]: parse_msg(): status: <200> > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:639]: parse_msg(): reason: <OK> > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, > <branch> = <0>; state=16 > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, > flags=2 > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the > first via > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [receive.c:152]: receive_msg(): After parse_msg... > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm > [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=25 global > id=24 T start=(nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: > tag=006cfccc318be31188fc19977b3a5651 > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/parse_addr_spec.c:885]: parse_addr_spec(): end of header reached, > state=29 > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> > [63]; uri=[sip:77.37.241.151:5060] > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip: > 77.37.241.151:5060>] > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <1> > <OPTIONS> > *Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm > [t_lookup.c:1045]: t_reply_matching(): DEBUG: t_reply_matching: failure to > match a transaction* > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm > [t_lookup.c:1140]: t_check_msg(): DEBUG: t_check_msg: msg id=25 global > id=25 T end=(nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : > content_length=0 > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [parser/msg_parser.c:106]: get_hdr_field(): found end of header > *Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [forward.c:784]: do_forward_reply(): reply cannot be forwarded - no 2nd > via* > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying > list (nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying > list (nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying > list (nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying > list (nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying > list (nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying > list (nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) > Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> > [receive.c:296]: receive_msg(): receive_msg: cleaning up > > > any ideas? > and show me ur conf please. > > > -- > > Hi All, > > I'm running into an issue, I'm not sure whether any of you seen this > yourselves and resolved it. Please share some pointers. My network is: > > clients <--> Public IP(Kamailio/RTPProxy)10.1.128.11 <--> 10.1.128.34 > (Freeswitch) > > The 200 OK response from Freeswitch (on the way back from called party to > caller) to Kamailio is shown below. Notice the Contact header URI host part > contains Freeswitch Private IP (10.1.128.34). Kamailio suppose to change > that to Public IP before forwarding the 200 OK (copied below) to Caller in > public domain. But, it's not. As a result, ACK from Caller is not reaching > back to Kamailio. > > How did you or anybody out there using Kamailio resolve this problem? If > needed, I can copy/paste my kamailio.cfg. > > -- > yours respectfully, Alexander Vasin. > > 8 926 1437200 > icq: 9906064 > [View Less]

2 1

Re: [SR-Users] reg event support in subscribe
by Andrew Pogrebennyk 05 Feb '14

05 Feb '14

Hi, See presence_reginfo - Extension to Presence server for registration info replication (RFC3680) http://kamailio.org/docs/modules/stable/modules/presence_reginfo.html On 02/05/2014 07:56 AM, Premchandiran wrote: > Hi All, > > May I know whether kamailio supports event header with reg (EVENT:reg) > (rfc 3680) , if kamailio supports may I know the module whether it is in > presence or presence_xml or any other module? > > > > Regards, > > *Prem Chandiran M*** >

1 0

Problem with response of http_query of the moudule utils
by Aman Agarwal 05 Feb '14

05 Feb '14

Hi, I am using http_query to make some HTTP API calls. I have found that the response of the HTTP call always has the initial 1062 characters missing. The HTTP call is sending strings of different lengths but the initial 1062 characters are always missing. Can anybody help me with this issue or has any idea why this might be happening? Thanks Aman Agarwal

1 0

Msilo message body type
by sipatse＠in.gr 05 Feb '14

05 Feb '14

1 0

websocket relay
by jaflong jaflong 05 Feb '14

05 Feb '14

Hi Everyone, I want kamailio to do a single task, that is relay incoming (webrtc sip over websockets) connections to asterisk. All sip handling is to be done is asterisk. I have configured asterisk with tls settings. My sample conf file is below At the moment I am getting these errors ERROR: tm [ut.h:343]: uri2dst2(): no corresponding socket for af 2 ERROR: tm [t_fwd.c:520]: prepare_new_uac(): ERROR: can't fwd to af 2, proto 3 (no corresponding listening socket) Please can anyone provide … [View More]any advise to complete this configuration listen=tcp:10.2.3.4:5080 enable_tls=yes tcp_connection_lifetime=3605 tcp_accept_no_cl=yes ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) mpath="/usr/local/lib64/kamailio/modules/" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "sdpops.so" loadmodule "nathelper.so" loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "tls.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") ####### Routing Logic ######## event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); xlog("L_INFO", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm =~"GET") { if (ws_handle_handshake()) { exit; } } xhttp_reply("404", "Not Found", "", ""); } request_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP request received on $Rp\n"); sl_send_reply("403", "Forbidden"); exit; } route(REQINIT); route(NATDETECT); route(RELAY); } route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } route[NATDETECT] { if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) fix_nated_register(); else { if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } } } route[RELAY] { xlog("L_INFO", "Forwarding Request\n"); $ru = "sip:"+$rU+"10.2.3.5:5080;transport=tcp"; if (!t_relay_to_tls()) { sl_reply_error(); return; } exit; } [View Less]

2 1

reg event support in subscribe
by Premchandiran 05 Feb '14

05 Feb '14

Hi All, May I know whether kamailio supports event header with reg (EVENT:reg) (rfc 3680) , if kamailio supports may I know the module whether it is in presence or presence_xml or any other module? Regards, Prem Chandiran M

1 0

GSM PDU Decoding
by Ramya Ramamurthy 04 Feb '14

04 Feb '14

Dear SR-Users, I am using kamailio for decoding of SIP Packets with Message Body as a GSM PDU. But I am not sure of the modules which i need to use for the same. Some help on this would be greatly appreciated. Thanks, ~Ramya.

5 7

Change the name of the location table for the usrloc module
by Alejandro Recarey 04 Feb '14

04 Feb '14

Hi everybody, I have been trying to setup a new installation of kamailio, and I need the user location table to have a specific name. This will allow two kamailio servers to use the same database for their registry information. I cannot find any modparam in the documentation that allows me to set the table name for the usrloc. Is there any way to do this? It seems like an oversight, every other module I have been able to change the table name without a problem. Thank you, Alex

2 2

Cluster: check/mark servers in cluster
by Victor V. Kustov 04 Feb '14

04 Feb '14

Hi, all! I had in config: #!define KAMDB "cluster://postcash" loadmodule "db_cluster.so" modparam("db_cluster", "connection","con1=>postgres://user1:password1@host1/kamailio") modparam("db_cluster", "connection","con2=>postgres://user2:password2@host2/kamailio") modparam("db_cluster", "cluster", "postcash=>con1=9s8p;con2=9s8p") modparam("usrloc", "db_url", KAMDB) and save registered user's locations in cluster after some server's restarts second base has empty set. How can I see … [View More]

1 0

Hello from newbie
by Alex Lake 04 Feb '14

04 Feb '14

Just trying to get to grips with Kamailio and thought I'd say hello to you all... Not sure if this is of any interest, but I've found it quite hard to get into Kamailio compared to (say) Freeswitch. This is not meant as criticism at all, but if you're looking to make it easier, I may be able to help! Installation was more or less OK, although a couple of slightly blind alleys and lack of concrete examples of config files. I believe I have a working server, but where to go next has left … [View More]

2 5

Asterisk re-INVITE race condition, error 500.
by David K 04 Feb '14

04 Feb '14

Hello all, So I have three machines, we don't care about audio for this problem, so everything I mention here is SIP related. Freeswitch <--> Kamailio 3.3.2 <--> Asterisk 1. Asterisk sends an INVITE to Freeswitch through the Kamailio proxy. 2. Kamailio replies 100 Trying and forwards to Freeswitch 3. Freeswitch replies 100 Trying 4. Freeswitch replies 180 Ringing to Kamailio 5. Kamailio routes the answer to Asterisk 6. Freeswitch replies 200 OK to Kamailio 7. Kamailio replies … [View More]

6 11

Kamailio World 2014 - New speakers and presentations
by Daniel-Constantin Mierla 04 Feb '14

04 Feb '14

Slightly less than two months to the start of Kamailio World 2014 <http://www.kamailioworld.com> [1] the organization of the event is getting accelerated. We received plenty of interesting proposals so far, we expect to end the Call for Speakers very soon, earlier than previously announced. If you have plans to submit, you should really hurry up. The profiles of about 20 speakers have been published in Speakers page <http://conference.kamailio.com/k02/speakers/> [2], few … [View More]

1 0

Proxied authentication
by Pete Ashdown 04 Feb '14

04 Feb '14

I'm having trouble divining the proper way to authenticate calls to remote destinations that require it. I'm hoping someone can clear it up. Freeswitch client server A is authenticated to Kamailio. Freeswitch test server B is authenticated to Kamailio. Calls from A -> Kamailio -> B fail with rejection of the authentication information that A is using to Kamailio. As far as I understand, I need to rewrite the authentication information with UAC? However, UAC's uacreg table seems to … [View More]

2 1

Re: [SR-Users] Regd. MAX_LDG_LOCKS
by Shankar 04 Feb '14

04 Feb '14

Hi, Can someone clarify here please? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Monday, February 03, 2014 10:53 AM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: Regd. MAX_LDG_LOCKS Hello, Can anyone brief about MAX_LDG_LOCKS in dlg_hash.c? Should I increase the value of MAX_LDG_LOCKS (hardcoded to 2048) if the hash size is increased? Regards, Shankar

1 0

Media-proxy Query
by Nandini madhu 04 Feb '14

04 Feb '14

Dear All, Greetings, I have configured Kamailio (4.0.3) with mediaproxy server, with this set-up audio/video calls are going good and hence NAT issue has solved. I have used two soft clients: 1) IMSDroid 2) Bria. But i have a query about media-proxy statistics i.e in the syslog, the info about media-proxy activity is shown as below: *Case 1*: when Call between two IMSDroid clients. caller_codec : 'Unknown(73)' callee_codec : 'Unknown(73)' callee_ua : 'unknown agent'. *Case 2* : When call … [View More]

2 3

Buffer Size
by Wingsravi R 04 Feb '14

04 Feb '14

Dear Kamailio'ns I am working on Kamailio (V 4.0)+ Mediaproxy server (2.5.2) running on Ubuntu 12.04. I am experiencing Latency, jitter, pixel led, choppy video errors in my set-up. So i wanted increase buffer sizes in my kamailio server , Mediaproxy server and Ubuntu also. In that regards i have increased Shared memory value to 512 MB and Package memory value 16 MB. BUFFER SIZE value to 8 MB in *config.h* file, But where can i change the value for 'MAX_RECV_BUFFER_SIZE' ? And are there any … [View More]

1 0

Re: [SR-Users] dialog list query
by John Murray 03 Feb '14

03 Feb '14

From: Charles Chance <charles.chance(a)sipcentric.com> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] dialog list query Message-ID: Charles hi, >What db_mode are you using for dialog module? Mode 5. I Don't need to persist the dialog state so means table maintained in memory and not flushed to disk until shutdown but I can access it through normal avp_db_query right? &… [View More]

2 4

Re: [SR-Users] Kamailio not mdifying Contact in multi-homed setup
by *sm1Ly 03 Feb '14

03 Feb '14

hello, I got the same issue. my sip device getting ok from kamailio, but asterisk reply 401. but. this is my config: http://pastebin.com/jGCak01E and I changed regfwd route to $du = "sip:50.0.0.10:5060;transport=udp"; forward(); and now I see this in logs: Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:633]: parse_msg(): SIP Reply (status): Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:635]… [View More]: parse_msg(): version: <SIP/2.0> Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:637]: parse_msg(): status: <200> Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:639]: parse_msg(): reason: <OK> Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <0>; state=16 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [receive.c:152]: receive_msg(): After parse_msg... Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=25 global id=24 T start=(nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=006cfccc318be31188fc19977b3a5651 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/parse_addr_spec.c:885]: parse_addr_spec(): end of header reached, state=29 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [63]; uri=[sip:77.37.241.151:5060] Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip: 77.37.241.151:5060>] Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <1> <OPTIONS> *Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm [t_lookup.c:1045]: t_reply_matching(): DEBUG: t_reply_matching: failure to match a transaction* Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm [t_lookup.c:1140]: t_check_msg(): DEBUG: t_check_msg: msg id=25 global id=25 T end=(nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header *Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [forward.c:784]: do_forward_reply(): reply cannot be forwarded - no 2nd via* Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [receive.c:296]: receive_msg(): receive_msg: cleaning up any ideas? and show me ur conf please. -- Hi All, I'm running into an issue, I'm not sure whether any of you seen this yourselves and resolved it. Please share some pointers. My network is: clients <--> Public IP(Kamailio/RTPProxy)10.1.128.11 <--> 10.1.128.34 (Freeswitch) The 200 OK response from Freeswitch (on the way back from called party to caller) to Kamailio is shown below. Notice the Contact header URI host part contains Freeswitch Private IP (10.1.128.34). Kamailio suppose to change that to Public IP before forwarding the 200 OK (copied below) to Caller in public domain. But, it's not. As a result, ACK from Caller is not reaching back to Kamailio. How did you or anybody out there using Kamailio resolve this problem? If needed, I can copy/paste my kamailio.cfg. -- yours respectfully, Alexander Vasin. 8 926 1437200 icq: 9906064 [View Less]

2 1

Change of source ip address
by Diego Alejandro Ozuna Escalada 03 Feb '14

03 Feb '14

Dear list, I'm having a problem. I have an SBC server with two network interfaces, one external IP (X.X.248.194) and other internal IP (X.X.248.21). Messages are sent to the proxy sip server through internal ip with this configuration: route [sip-serverr] { rewritehostport ("ip_sip_proxy: 5060"); force_send_socket (X.X.248.21: 5060); t_relay (); exit; } When a UA sends an INVITE to the SBC, the responses (back) to UA are being sent with the correct … [View More]

1 0

Security hygiene for Kamailio
by davy 03 Feb '14

03 Feb '14

Hi all, we all enjoy our FAIL2BAN and snippets of our Kamailio config when we see it successfully fight off the "friendly-scanner", and multiple futile attempts to fool our systems. But it got me thinking… What is a sufficient level of security on our Kamailio machinery… ? Are we all just doing whatever, or is the nature of the beast, that every setup is different? Eventually while having a beer, we will end up in the discussion Kamailio is as good (and even much better) as most of the … [View More]

6 12

Carrierroute - scan_prefix
by Oliver Roth 03 Feb '14

03 Feb '14

Hi all I need a solution to get the used scan_prefix in carrierroute module. We have different carriers for our destinations, sometimes split up 50%/50% or other splittings - all routes with failure routes. Now if one destination is not reachable, then I would like to get this information and reroute to another carrier. As we use the carrierroute module, we use the "description" to get the carriername (for billing purposes). Is there a solution to get the used scan_prefix from the … [View More]

2 2

kamailio answers instead asterisk
by *sm1Ly 03 Feb '14

03 Feb '14

hello all. this is my kamailio conf: http://pastebin.com/jGCak01E when I try to register any sip device. it looks like. device -> ext IP kamailio -> int IP kamailio -> asterisk. so I trying to nat something to lan. but on asterisk I see that register, it answer 401, than register again and it again answer 401. but on my sip device I see register and than kamailio answer 200 !! why ??? 1. #!KAMAILIO 2. # 3. # Kamailio (OpenSER) SIP Server v4.1 - default configuration script … [View More] 4. # - web: http://www.kamailio.org 5. # - git: http://sip-router.org 6. # 7. # Direct your questions about this file to: < sr-users(a)lists.sip-router.org> 8. # 9. # Refer to the Core CookBook at http://www.kamailio.org/wiki/ 10. # for an explanation of possible statements, functions and parameters. 11. # 12. # Several features can be enabled using '#!define WITH_FEATURE' directives: 13. # 14. # *** To run in debug mode: 15. # - define WITH_DEBUG 16. # 17. # *** To enable mysql: 18. # - define WITH_MYSQL 19. # 20. # *** To enable authentication execute: 21. # - enable mysql 22. # - define WITH_AUTH 23. # - add users using 'kamctl' 24. # 25. # *** To enable IP authentication execute: 26. # - enable mysql 27. # - enable authentication 28. # - define WITH_IPAUTH 29. # - add IP addresses with group id '1' to 'address' table 30. # 31. # *** To enable persistent user location execute: 32. # - enable mysql 33. # - define WITH_USRLOCDB 34. # 35. # *** To enable presence server execute: 36. # - enable mysql 37. # - define WITH_PRESENCE 38. # 39. # *** To enable nat traversal execute: 40. # - define WITH_NAT 41. # - install RTPProxy: http://www.rtpproxy.org 42. # - start RTPProxy: 43. # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 44. # 45. # *** To enable PSTN gateway routing execute: 46. # - define WITH_PSTN 47. # - set the value of pstn.gw_ip 48. # - check route[PSTN] for regexp routing condition 49. # 50. # *** To enable database aliases lookup execute: 51. # - enable mysql 52. # - define WITH_ALIASDB 53. # 54. # *** To enable speed dial lookup execute: 55. # - enable mysql 56. # - define WITH_SPEEDDIAL 57. # 58. # *** To enable multi-domain support execute: 59. # - enable mysql 60. # - define WITH_MULTIDOMAIN 61. # 62. # *** To enable TLS support execute: 63. # - adjust CFGDIR/tls.cfg as needed 64. # - define WITH_TLS 65. # 66. # *** To enable XMLRPC support execute: 67. # - define WITH_XMLRPC 68. # - adjust route[XMLRPC] for access policy 69. # 70. # *** To enable anti-flood detection execute: 71. # - adjust pike and htable=>ipban settings as needed (default is 72. # block if more than 16 requests in 2 seconds and ban for 300 seconds) 73. # - define WITH_ANTIFLOOD 74. # 75. # *** To block 3XX redirect replies execute: 76. # - define WITH_BLOCK3XX 77. # 78. # *** To enable VoiceMail routing execute: 79. # - define WITH_VOICEMAIL 80. # - set the value of voicemail.srv_ip 81. # - adjust the value of voicemail.srv_port 82. # 83. # *** To enhance accounting execute: 84. # - enable mysql 85. # - define WITH_ACCDB 86. # - add following columns to database 87. #!ifdef ACCDB_COMMENT 88. ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; 89. ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; 90. ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; 91. ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; 92. ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; 93. ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; 94. ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; 95. ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; 96. ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; 97. ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; 98. ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; 99. ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; 100. #!endif 101. 102. # #!define WITH_MYSQL 103. # #!define WITH_AUTH 104. # #!define WITH_USRLOCDB 105. #!define WITH_NAT 106. #!define WITH_ASTERISK 107. # #!define WITH_DEBUG 108. 109. ####### Include Local Config If Exists ######### 110. import_file "kamailio-local.cfg" 111. 112. ####### Defined Values ######### 113. 114. # *** Value defines - IDs used later in config 115. #!ifdef WITH_MYSQL 116. # - database URL - used to connect to database server by modules such 117. # as: auth_db, acc, usrloc, a.s.o. 118. #!ifndef DBURL 119. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" 120. #!endif 121. #!endif 122. #!ifdef WITH_MULTIDOMAIN 123. # - the value for 'use_domain' parameters 124. #!define MULTIDOMAIN 1 125. #!else 126. #!define MULTIDOMAIN 0 127. #!endif 128. 129. # - flags 130. # FLT_ - per transaction (message) flags 131. # FLB_ - per branch flags 132. #!define FLT_ACC 1 133. #!define FLT_ACCMISSED 2 134. #!define FLT_ACCFAILED 3 135. #!define FLT_NATS 5 136. 137. #!define FLB_NATB 6 138. #!define FLB_NATSIPPING 7 139. 140. ####### Global Parameters ######### 141. 142. ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR 143. #!ifdef WITH_DEBUG 144. debug=4 145. log_stderror=yes 146. #!else 147. debug=3 148. log_stderror=no 149. #!endif 150. 151. memdbg=5 152. memlog=5 153. 154. log_facility=LOG_LOCAL0 155. 156. fork=yes 157. children=4 158. 159. /* uncomment the next line to disable TCP (default on) */ 160. #disable_tcp=yes 161. 162. /* uncomment the next line to disable the auto discovery of local aliases 163. based on reverse DNS on IPs (default on) */ 164. #auto_aliases=no 165. 166. /* add local domain aliases */ 167. #alias="sip.mydomain.com" 168. 169. /* uncomment and configure the following line if you want Kamailio to 170. bind on a specific interface/port/proto (default bind on all available) */ 171. 172. #listen=udp:50.0.0.1:5060 advertise 194.190.8.171:5060 173. listen=udp:194.190.8.171:5060 advertise 50.0.0.1:5060 174. 175. /* port to listen to 176. * - can be specified more than once if needed to listen on many ports */ 177. port=5060 178. 179. mhomed=1 180. 181. #!ifdef WITH_TLS 182. enable_tls=yes 183. #!endif 184. 185. # life time of TCP connection when there is no traffic 186. # - a bit higher than registration expires to cope with UA behind NAT 187. tcp_connection_lifetime=3605 188. 189. ####### Custom Parameters ######### 190. 191. # These parameters can be modified runtime via RPC interface 192. # - see the documentation of 'cfg_rpc' module. 193. # 194. # Format: group.id = value 'desc' description 195. # Access: $sel(cfg_get.group.id) or @cfg_get.group.id 196. # 197. 198. #!ifdef WITH_PSTN 199. # PSTN GW Routing 200. # 201. # - pstn.gw_ip: valid IP or hostname as string value, example: 202. # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" 203. # 204. # - by default is empty to avoid misrouting 205. #pstn.gw_ip = "" desc "PSTN GW Address" 206. #pstn.gw_port = "" desc "PSTN GW Port" 207. #!endif 208. 209. #!ifdef WITH_VOICEMAIL 210. # VoiceMail Routing on offline, busy or no answer 211. # 212. # - by default Voicemail server IP is empty to avoid misrouting 213. #voicemail.srv_ip = "" desc "VoiceMail IP Address" 214. #voicemail.srv_port = "5060" desc "VoiceMail Port" 215. #!endif 216. 217. #!ifdef WITH_ASTERISK 218. asterisk.bindip = "50.0.0.10" desc "Asterisk IP Address" 219. asterisk.bindport = "5060" desc "Asterisk Port" 220. kamailio.bindip = "50.0.0.1" desc "Kamailio IP Address" 221. kamailio.bindport = "5060" desc "Kamailio Port" 222. #!endif 223. 224. ####### Modules Section ######## 225. 226. # set paths to location of modules (to sources or installation folders) 227. #!ifdef WITH_SRCPATH 228. mpath="modules/" 229. #!else 230. mpath="/usr/lib64/kamailio/modules/" 231. #!endif 232. 233. #!ifdef WITH_MYSQL 234. loadmodule "db_mysql.so" 235. #!endif 236. 237. loadmodule "mi_fifo.so" 238. loadmodule "kex.so" 239. loadmodule "tm.so" 240. loadmodule "tmx.so" 241. loadmodule "sl.so" 242. loadmodule "rr.so" 243. loadmodule "pv.so" 244. loadmodule "maxfwd.so" 245. loadmodule "usrloc.so" 246. loadmodule "registrar.so" 247. loadmodule "textops.so" 248. loadmodule "siputils.so" 249. loadmodule "xlog.so" 250. loadmodule "sanity.so" 251. loadmodule "ctl.so" 252. loadmodule "cfg_rpc.so" 253. loadmodule "mi_rpc.so" 254. loadmodule "acc.so" 255. 256. #!ifdef WITH_AUTH 257. loadmodule "auth.so" 258. loadmodule "auth_db.so" 259. #!ifdef WITH_IPAUTH 260. loadmodule "permissions.so" 261. #!endif 262. #!endif 263. 264. #!ifdef WITH_ALIASDB 265. loadmodule "alias_db.so" 266. #!endif 267. 268. #!ifdef WITH_SPEEDDIAL 269. loadmodule "speeddial.so" 270. #!endif 271. 272. #!ifdef WITH_MULTIDOMAIN 273. loadmodule "domain.so" 274. #!endif 275. 276. #!ifdef WITH_PRESENCE 277. loadmodule "presence.so" 278. loadmodule "presence_xml.so" 279. #!endif 280. 281. #!ifdef WITH_NAT 282. loadmodule "nathelper.so" 283. loadmodule "rtpproxy.so" 284. #!endif 285. 286. #!ifdef WITH_TLS 287. loadmodule "tls.so" 288. #!endif 289. 290. #!ifdef WITH_ANTIFLOOD 291. loadmodule "htable.so" 292. loadmodule "pike.so" 293. #!endif 294. 295. #!ifdef WITH_XMLRPC 296. loadmodule "xmlrpc.so" 297. #!endif 298. 299. #!ifdef WITH_DEBUG 300. loadmodule "debugger.so" 301. #!endif 302. 303. #!ifdef WITH_ASTERISK 304. loadmodule "uac.so" 305. #!endif 306. 307. 308. #loadmodule "topoh.so" 309. #modparam("topoh", "mask_key", "balalayka") 310. #modparam("topoh", "mask_ip", "50.0.0.1") 311. 312. # ----------------- setting module-specific parameters --------------- 313. 314. 315. # ----- mi_fifo params ----- 316. modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") 317. 318. 319. # ----- tm params ----- 320. # auto-discard branches from previous serial forking leg 321. modparam("tm", "failure_reply_mode", 3) 322. # default retransmission timeout: 30sec 323. modparam("tm", "fr_timer", 30000) 324. # default invite retransmission timeout after 1xx: 120sec 325. modparam("tm", "fr_inv_timer", 120000) 326. 327. 328. # ----- rr params ----- 329. # add value to ;lr param to cope with most of the UAs 330. modparam("rr", "enable_full_lr", 1) 331. # do not append from tag to the RR (no need for this script) 332. #!ifdef WITH_ASTERISK 333. modparam("rr", "append_fromtag", 1) 334. #!else 335. modparam("rr", "append_fromtag", 0) 336. #!endif 337. 338. 339. # ----- registrar params ----- 340. modparam("registrar", "method_filtering", 1) 341. /* uncomment the next line to disable parallel forking via location */ 342. # modparam("registrar", "append_branches", 0) 343. /* uncomment the next line not to allow more than 10 contacts per AOR */ 344. #modparam("registrar", "max_contacts", 10) 345. # max value for expires of registrations 346. modparam("registrar", "max_expires", 3600) 347. # set it to 1 to enable GRUU 348. modparam("registrar", "gruu_enabled", 0) 349. 350. 351. # ----- acc params ----- 352. /* what special events should be accounted ? */ 353. modparam("acc", "early_media", 0) 354. modparam("acc", "report_ack", 0) 355. modparam("acc", "report_cancels", 0) 356. /* by default ww do not adjust the direct of the sequential requests. 357. if you enable this parameter, be sure the enable "append_fromtag" 358. in "rr" module */ 359. modparam("acc", "detect_direction", 0) 360. /* account triggers (flags) */ 361. modparam("acc", "log_flag", FLT_ACC) 362. modparam("acc", "log_missed_flag", FLT_ACCMISSED) 363. modparam("acc", "log_extra", 364. "src_user=$fU;src_domain=$fd;src_ip=$si;" 365. "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") 366. modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) 367. /* enhanced DB accounting */ 368. #!ifdef WITH_ACCDB 369. modparam("acc", "db_flag", FLT_ACC) 370. modparam("acc", "db_missed_flag", FLT_ACCMISSED) 371. modparam("acc", "db_url", DBURL) 372. modparam("acc", "db_extra", 373. "src_user=$fU;src_domain=$fd;src_ip=$si;" 374. "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") 375. #!endif 376. 377. 378. # ----- usrloc params ----- 379. /* enable DB persistency for location entries */ 380. #!ifdef WITH_USRLOCDB 381. modparam("usrloc", "db_url", DBURL) 382. modparam("usrloc", "db_mode", 2) 383. modparam("usrloc", "use_domain", MULTIDOMAIN) 384. #!endif 385. 386. 387. # ----- auth_db params ----- 388. #!ifdef WITH_AUTH 389. modparam("auth_db", "db_url", DBURL) 390. modparam("auth_db", "calculate_ha1", yes) 391. modparam("auth_db", "password_column", "password") 392. modparam("auth_db", "load_credentials", "") 393. modparam("auth_db", "use_domain", MULTIDOMAIN) 394. 395. # ----- permissions params ----- 396. #!ifdef WITH_IPAUTH 397. modparam("permissions", "db_url", DBURL) 398. modparam("permissions", "db_mode", 1) 399. #!endif 400. 401. #!endif 402. 403. 404. # ----- alias_db params ----- 405. #!ifdef WITH_ALIASDB 406. modparam("alias_db", "db_url", DBURL) 407. modparam("alias_db", "use_domain", MULTIDOMAIN) 408. #!endif 409. 410. 411. # ----- speeddial params ----- 412. #!ifdef WITH_SPEEDDIAL 413. modparam("speeddial", "db_url", DBURL) 414. modparam("speeddial", "use_domain", MULTIDOMAIN) 415. #!endif 416. 417. 418. # ----- domain params ----- 419. #!ifdef WITH_MULTIDOMAIN 420. modparam("domain", "db_url", DBURL) 421. # register callback to match myself condition with domains list 422. modparam("domain", "register_myself", 1) 423. #!endif 424. 425. 426. #!ifdef WITH_PRESENCE 427. # ----- presence params ----- 428. modparam("presence", "db_url", DBURL) 429. 430. # ----- presence_xml params ----- 431. modparam("presence_xml", "db_url", DBURL) 432. modparam("presence_xml", "force_active", 1) 433. #!endif 434. 435. 436. #!ifdef WITH_NAT 437. # ----- rtpproxy params ----- 438. modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") 439. 440. # ----- nathelper params ----- 441. modparam("nathelper", "natping_interval", 30) 442. modparam("nathelper", "ping_nated_only", 1) 443. modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) 444. modparam("nathelper", "sipping_from", "sip:pinger@kamaz") 445. 446. # params needed for NAT traversal in other modules 447. modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") 448. modparam("usrloc", "nat_bflag", FLB_NATB) 449. #!endif 450. 451. 452. #!ifdef WITH_TLS 453. # ----- tls params ----- 454. modparam("tls", "config", "//etc/kamailio/tls.cfg") 455. #!endif 456. 457. #!ifdef WITH_ANTIFLOOD 458. # ----- pike params ----- 459. modparam("pike", "sampling_time_unit", 2) 460. modparam("pike", "reqs_density_per_unit", 16) 461. modparam("pike", "remove_latency", 4) 462. 463. # ----- htable params ----- 464. # ip ban htable with autoexpire after 5 minutes 465. modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") 466. #!endif 467. 468. #!ifdef WITH_XMLRPC 469. # ----- xmlrpc params ----- 470. modparam("xmlrpc", "route", "XMLRPC"); 471. modparam("xmlrpc", "url_match", "^/RPC") 472. #!endif 473. 474. #!ifdef WITH_DEBUG 475. # ----- debugger params ----- 476. modparam("debugger", "cfgtrace", 1) 477. #!endif 478. 479. ####### Routing Logic ######## 480. 481. 482. # Main SIP request routing logic 483. # - processing of any incoming SIP request starts with this route 484. # - note: this is the same as route { ... } 485. request_route { 486. 487. # per request initial checks 488. route(REQINIT); 489. 490. xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) Main Route before ---NAT---\n"); 491. route(DEBUG); 492. # NAT detection 493. route(NATDETECT); 494. 495. xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in Route[NATDETECT] fix_nat-register\n"); 496. route(DEBUG); 497. 498. # CANCEL processing 499. if (is_method("CANCEL")) 500. { 501. if (t_check_trans()) { 502. route(RELAY); 503. } 504. exit; 505. } 506. 507. route(DEBUG); 508. 509. # handle requests within SIP dialogs 510. route(WITHINDLG); 511. 512. route(DEBUG); 513. 514. ### only initial requests (no To tag) 515. 516. t_check_trans(); 517. 518. route(DEBUG); 519. 520. # authentication 521. route(AUTH); 522. 523. route(DEBUG); 524. 525. # record routing for dialog forming requests (in case they are routed) 526. # - remove preloaded route headers 527. remove_hf("Route"); 528. if (is_method("INVITE|SUBSCRIBE")) 529. record_route(); 530. 531. # account only INVITEs 532. if (is_method("INVITE")) 533. { 534. setflag(FLT_ACC); # do accounting 535. } 536. 537. # dispatch requests to foreign domains 538. route(SIPOUT); 539. 540. ### requests for my local domains 541. 542. # handle presence related requests 543. route(PRESENCE); 544. 545. route(DEBUG); 546. 547. # handle registrations 548. route(REGISTRAR); 549. 550. route(DEBUG); 551. 552. if ($rU==$null) 553. { 554. # request with no Username in RURI 555. sl_send_reply("484","Address Incomplete"); 556. exit; 557. } 558. 559. # dispatch destinations to PSTN 560. # route(PSTN); 561. 562. route(DEBUG); 563. 564. # user location service 565. route(LOCATION); 566. 567. route(DEBUG); 568. 569. route(RELAY); 570. 571. } 572. 573. 574. route[RELAY] { 575. 576. # enable additional event routes for forwarded requests 577. # - serial forking, RTP relaying handling, a.s.o. 578. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { 579. if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); 580. } 581. if (is_method("INVITE|SUBSCRIBE|UPDATE")) { 582. if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); 583. } 584. if (is_method("INVITE")) { 585. if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); 586. } 587. 588. if (!t_relay()) { 589. sl_reply_error(); 590. } 591. exit; 592. } 593. 594. # Per SIP request initial checks 595. route[REQINIT] { 596. #!ifdef WITH_ANTIFLOOD 597. # flood dection from same IP and traffic ban for a while 598. # be sure you exclude checking trusted peers, such as pstn gateways 599. # - local host excluded (e.g., loop to self) 600. if(src_ip!=myself) 601. { 602. if($sht(ipban=>$si)!=$null) 603. { 604. # ip is already blocked 605. xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); 606. exit; 607. } 608. if (!pike_check_req()) 609. { 610. xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); 611. $sht(ipban=>$si) = 1; 612. exit; 613. } 614. } 615. #!endif 616. 617. if (!mf_process_maxfwd_header("10")) { 618. sl_send_reply("483","Too Many Hops"); 619. exit; 620. } 621. 622. if(!sanity_check("1511", "7")) 623. { 624. xlog("Malformed SIP message from $si:$sp\n"); 625. exit; 626. } 627. } 628. 629. # Handle requests within SIP dialogs 630. route[WITHINDLG] { 631. if (has_totag()) { 632. # sequential request withing a dialog should 633. # take the path determined by record-routing 634. if (loose_route()) { 635. route(DLGURI); 636. if (is_method("BYE")) { 637. setflag(FLT_ACC); # do accounting ... 638. setflag(FLT_ACCFAILED); # ... even if the transaction fails 639. } 640. else if ( is_method("ACK") ) { 641. # ACK is forwarded statelessy 642. route(NATMANAGE); 643. } 644. else if ( is_method("NOTIFY") ) { 645. # Add Record-Route for in-dialog NOTIFY as per RFC 6665. 646. record_route(); 647. } 648. route(RELAY); 649. } else { 650. if (is_method("SUBSCRIBE") && uri == myself) { 651. # in-dialog subscribe requests 652. route(PRESENCE); 653. exit; 654. } 655. if ( is_method("ACK") ) { 656. if ( t_check_trans() ) { 657. # no loose-route, but stateful ACK; 658. # must be an ACK after a 487 659. # or e.g. 404 from upstream server 660. route(RELAY); 661. exit; 662. } else { 663. # ACK without matching transaction ... ignore and discard 664. exit; 665. } 666. } 667. sl_send_reply("404","Not here"); 668. } 669. exit; 670. } 671. } 672. 673. # Handle SIP registrations 674. route[REGISTRAR] { 675. if (is_method("REGISTER")) 676. { 677. if(isflagset(FLT_NATS)) 678. { 679. setbflag(FLB_NATB); 680. # uncomment next line to do SIP NAT pinging 681. setbflag(FLB_NATSIPPING); 682. } 683. if (!save("location")) 684. sl_reply_error(); 685. 686. 687. #!ifdef WITH_ASTERISK 688. route(REGFWD); 689. #!endif 690. 691. exit; 692. } 693. } 694. 695. 696. #!ifdef WITH_ASTERISK 697. # Test if coming from Asterisk 698. route[FROMASTERISK] { 699. if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; 700. return -1; 701. } 702. 703. 704. # Send to Asterisk 705. route[TOASTERISK] { 706. $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); 707. xlog("L_INFO","[$fU@$si:$sp]{$rm} From Outside World to Asterisk Box $du\n"); 708. route(RELAY); 709. exit; 710. } 711. 712. # Forward REGISTER to Asterisk 713. route[REGFWD] { 714. if(!is_method("REGISTER")) 715. { 716. return; 717. } 718. 719. xlog("[$mi] PRINT AU: $au PRINT fU: $fU Received SIP Message (method: $rm) ($ml bytes) to $Ri:$Rp from $si:$sp:\n$mb\n"); 720. 721. t_relay_to_udp("50.0.0.10", "5060"); 722. 723. xlog("[$mi] PRINT AU: $au PRINT fU: $fU Sended SIP Message (method: $rm) ($ml bytes) to $Ri:$Rp from $si:$sp:\n$mb\n"); 724. 725. } 726. #!endif 727. 728. # USER location service 729. route[LOCATION] { 730. 731. #!ifdef WITH_SPEEDDIAL 732. # search for short dialing - 2-digit extension 733. if($rU=~"^[0-9][0-9]$") 734. if(sd_lookup("speed_dial")) 735. route(SIPOUT); 736. #!endif 737. 738. #!ifdef WITH_ALIASDB 739. # search in DB-based aliases 740. if(alias_db_lookup("dbaliases")) 741. route(SIPOUT); 742. #!endif 743. 744. #!ifdef WITH_ASTERISK 745. if(is_method("INVITE") && (!route(FROMASTERISK))) { 746. # if new call from out there - send to Asterisk 747. # - non-INVITE request are routed directly by Kamailio 748. # - traffic from Asterisk is routed also directy by Kamailio 749. route(TOASTERISK); 750. exit; 751. } 752. #!endif 753. 754. $avp(oexten) = $rU; 755. if (!lookup("location")) { 756. $var(rc) = $rc; 757. route(TOVOICEMAIL); 758. t_newtran(); 759. switch ($var(rc)) { 760. case -1: 761. case -3: 762. send_reply("404", "Not Found"); 763. exit; 764. case -2: 765. send_reply("405", "Method Not Allowed"); 766. exit; 767. } 768. } 769. 770. # when routing via usrloc, log the missed calls also 771. if (is_method("INVITE")) 772. { 773. setflag(FLT_ACCMISSED); 774. } 775. 776. route(RELAY); 777. exit; 778. } 779. 780. # Presence server route 781. route[PRESENCE] { 782. if(!is_method("PUBLISH|SUBSCRIBE")) 783. return; 784. 785. if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { 786. route(TOVOICEMAIL); 787. # returns here if no voicemail server is configured 788. sl_send_reply("404", "No voicemail service"); 789. exit; 790. } 791. 792. #!ifdef WITH_PRESENCE 793. if (!t_newtran()) 794. { 795. sl_reply_error(); 796. exit; 797. } 798. 799. if(is_method("PUBLISH")) 800. { 801. handle_publish(); 802. t_release(); 803. } else if(is_method("SUBSCRIBE")) { 804. handle_subscribe(); 805. t_release(); 806. } 807. exit; 808. #!endif 809. 810. # if presence enabled, this part will not be executed 811. if (is_method("PUBLISH") || $rU==$null) 812. { 813. sl_send_reply("404", "Not here"); 814. exit; 815. } 816. return; 817. } 818. 819. # Authentication route 820. route[AUTH] { 821. 822. #!ifdef WITH_ASTERISK 823. # do not auth traffic from Asterisk - trusted! 824. if(route(FROMASTERISK)) 825. return; 826. #!endif 827. 828. 829. #!ifdef WITH_AUTH 830. 831. #!ifdef WITH_IPAUTH 832. if((!is_method("REGISTER")) && allow_source_address()) 833. { 834. # source IP allowed 835. return; 836. } 837. #!endif 838. 839. if (is_method("REGISTER") || from_uri==myself) 840. { 841. # authenticate requests 842. if (!auth_check("$fd", "subscriber", "1")) { 843. auth_challenge("$fd", "0"); 844. exit; 845. } 846. # user authenticated - remove auth header 847. if(!is_method("REGISTER|PUBLISH")) 848. consume_credentials(); 849. } 850. # if caller is not local subscriber, then check if it calls 851. # a local destination, otherwise deny, not an open relay here 852. if (from_uri!=myself && uri!=myself) 853. { 854. sl_send_reply("403","Not relaying"); 855. exit; 856. } 857. 858. #!endif 859. return; 860. } 861. 862. # Caller NAT detection route 863. route[NATDETECT] { 864. #!ifdef WITH_NAT 865. force_rport(); 866. if (nat_uac_test("19")) { 867. if (is_method("REGISTER")) { 868. fix_nated_register(); 869. } else { 870. if(is_first_hop()) 871. set_contact_alias(); 872. } 873. setflag(FLT_NATS); 874. 875. } 876. #!endif 877. return; 878. } 879. 880. # RTPProxy control 881. route[NATMANAGE] { 882. #!ifdef WITH_NAT 883. if (is_request()) { 884. if(has_totag()) { 885. if(check_route_param("nat=yes")) { 886. setbflag(FLB_NATB); 887. } 888. } 889. } 890. if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) 891. return; 892. 893. rtpproxy_manage(); 894. 895. if (is_request()) { 896. if (!has_totag()) { 897. if(t_is_branch_route()) { 898. add_rr_param(";nat=yes"); 899. } 900. } 901. } 902. if (is_reply()) { 903. if(isbflagset(FLB_NATB)) { 904. if(is_first_hop()) 905. set_contact_alias(); 906. } 907. 908. xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in route[NATMANAGE] RTPproxy with EI Flags\n"); 909. 910. } 911. #!endif 912. return; 913. } 914. 915. # URI update for dialog requests 916. route[DLGURI] { 917. #!ifdef WITH_NAT 918. if(!isdsturiset()) { 919. handle_ruri_alias(); 920. 921. xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in route[DLGURI] RTPproxy with EI Flags\n"); 922. 923. } 924. #!endif 925. return; 926. } 927. 928. # Routing to foreign domains 929. route[SIPOUT] { 930. if (!uri==myself) 931. { 932. append_hf("P-hint: outbound\r\n"); 933. route(RELAY); 934. } 935. } 936. 937. # PSTN GW routing 938. route[PSTN] { 939. #!ifdef WITH_PSTN 940. # check if PSTN GW IP is defined 941. if (strempty($sel(cfg_get.pstn.gw_ip))) { 942. xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); 943. return; 944. } 945. 946. # route to PSTN dialed numbers starting with '+' or '00' 947. # (international format) 948. # - update the condition to match your dialing rules for PSTN routing 949. if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) 950. return; 951. 952. # only local users allowed to call 953. if(from_uri!=myself) { 954. sl_send_reply("403", "Not Allowed"); 955. exit; 956. } 957. 958. if (strempty($sel(cfg_get.pstn.gw_port))) { 959. $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); 960. } else { 961. $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" 962. + $sel(cfg_get.pstn.gw_port); 963. } 964. 965. route(RELAY); 966. exit; 967. #!endif 968. 969. return; 970. } 971. 972. # XMLRPC routing 973. #!ifdef WITH_XMLRPC 974. route[XMLRPC] { 975. # allow XMLRPC from localhost 976. if ((method=="POST" || method=="GET") 977. && (src_ip==127.0.0.1)) { 978. # close connection only for xmlrpclib user agents (there is a bug in 979. # xmlrpclib: it waits for EOF before interpreting the response). 980. if ($hdr(User-Agent) =~ "xmlrpclib") 981. set_reply_close(); 982. set_reply_no_connect(); 983. dispatch_rpc(); 984. exit; 985. } 986. send_reply("403", "Forbidden"); 987. exit; 988. } 989. #!endif 990. 991. # route to voicemail server 992. route[TOVOICEMAIL] { 993. #!ifdef WITH_VOICEMAIL 994. if(!is_method("INVITE|SUBSCRIBE")) 995. return; 996. 997. # check if VoiceMail server IP is defined 998. if (strempty($sel(cfg_get.voicemail.srv_ip))) { 999. xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); 1000. return; 1001. } 1002. if(is_method("INVITE")) { 1003. if($avp(oexten)==$null) 1004. return; 1005. $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) 1006. + ":" + $sel(cfg_get.voicemail.srv_port); 1007. } else { 1008. if($rU==$null) 1009. return; 1010. $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) 1011. + ":" + $sel(cfg_get.voicemail.srv_port); 1012. } 1013. route(RELAY); 1014. exit; 1015. #!endif 1016. 1017. return; 1018. } 1019. 1020. # manage outgoing branches 1021. branch_route[MANAGE_BRANCH] { 1022. xdbg("new branch [$T_branch_idx] to $ru\n"); 1023. route(NATMANAGE); 1024. } 1025. 1026. # manage incoming replies 1027. onreply_route[MANAGE_REPLY] { 1028. xdbg("incoming reply\n"); 1029. if(status=~"[12][0-9][0-9]") 1030. route(NATMANAGE); 1031. } 1032. 1033. # manage failure routing cases 1034. failure_route[MANAGE_FAILURE] { 1035. route(NATMANAGE); 1036. 1037. if (t_is_canceled()) { 1038. exit; 1039. } 1040. 1041. #!ifdef WITH_BLOCK3XX 1042. # block call redirect based on 3xx replies. 1043. if (t_check_status("3[0-9][0-9]")) { 1044. t_reply("404","Not found"); 1045. exit; 1046. } 1047. #!endif 1048. 1049. #!ifdef WITH_VOICEMAIL 1050. # serial forking 1051. # - route to voicemail on busy or no answer (timeout) 1052. if (t_check_status("486|408")) { 1053. $du = $null; 1054. route(TOVOICEMAIL); 1055. exit; 1056. } 1057. #!endif 1058. } 1059. 1060. 1061. route[DEBUG] { 1062. if (method==("PUBLISH|SUBSCRIBE|REGISTER|OPTIONS")) { 1063. return; 1064. } 1065. xlog("[$mi] Received SIP Message (method: $rm) ($ml bytes) to $Ri:$Rp from $si:$sp:\n$mb\n"); 1066. } -- С уважением, Александр Васин. 8 926 1437200 icq: 9906064 [View Less]

1 0

ngcp-mediaproxy-ng media address
by Kelvin Chua 03 Feb '14

03 Feb '14

somehow, mediaproxy is ignoring the media address set via rtpproxy_manage("co","10.17.0.102"); anybody tried this before? Jan 31 09:03:29 kam2 mediaproxy-ng[20455]: Got valid command from 127.0.0.1:46077: answer - { "sdp": "v=0#015#012o=root 345956998 345956998 IN IP4 10.17.0.105#015#012s=Asterisk PBX 11.7.0#015#012c=IN IP4 10.17.0.105#015#012t=0 0#015#012m=audio 15712 RTP/AVP 97 3 0 8 101#015#012a=rtpmap:97 speex/8000#015#012a=rtpmap:3 GSM/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 … [View More]

2 4

Re: [SR-Users] Regd. MAX_LDG_LOCKS
by Shankar 03 Feb '14

03 Feb '14

Hi Alex, We set the hash size to 32 MB and found that the hash size is not the same as configured one. At the max it was taking 64KB. Regards, Shankar Date: Mon, 03 Feb 2014 00:27:29 -0500 From: Alex Balashov <abalashov(a)evaristesys.com> To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Regd. MAX_LDG_LOCKS Message-ID: <52EF28C1.5010204(a)evaristesys.com> Content-Type: text/plain; charset=windows-1252; format=flowed On 02/03/2014 12:22 AM, Shankar … [View More]

1 0

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