- sr-users - kamailio.org

proposed patch for the registrar module - crash fix
by Dragos Oancea 28 Feb '14

28 Feb '14

Hi We think that we found and fix a bug in the registrar module. The bug is hard to reproduce, and it crashes our Kamailio from time to time (once at 2-3 weeks for us) . In save.c, function update_contacts() there are two places where we free() a pointer and then we reuse it, line 700 and line 730: while(ptr){ ptr0 = ptr; if(ptr!=c) ul.delete_ucontact(_r, ptr); ptr=ptr0->next; } And then from inside delete_ucontact(_r,ptr) we call mem_delete_ucontact(_r, _c) which calls free_ucontact(_c) , which calls shm_free(_c) . _c is actually our ptr. If another process writes at the location pointed by ptr during the "while" loop , the current process will crash. The bug affects 4.0.4, 4.0.5 and maybe older versions. We keep the location table in memory, no database backend, and we do alot of REGISTER/un-REGISTER in our environment, and maybe that's why this bug was not spotted by the community before. Just to clarify, this fix is in relation with this post: http://lists.sip-router.org/pipermail/sr-dev/2014-February/022934.html Please take into consideration the attached patch provided by the Libon Voice Team. Regards, Dragos Oancea

2 1

Mobile 3G Block from provider
by Rene Montilva 28 Feb '14

28 Feb '14

Hi List I'm trying to call over 3G networks, but doesn't work, i think my mobile provider block voip traffic, is there any way to hide voip traffic or jump the provider block rule ??

2 1

keep buddy list - best way
by Mihai Marin 27 Feb '14

27 Feb '14

Hello Sirs, I'm a little confused getting a decision about implementing buddy list (step: presence & IM). What is the best way for keeping the contacts list: xmpp or xcap? You will say probably xcap server but even if I intend to use in the future an xmpp gateway in order to be able to communicate sip with with xmpp servers like google - google talk? And why xcap? Because most of the clients has the possibility to connect to a xcap server to get the buddy list? From web perspective I think both are OK because of the xhttp module power in kamailio to get the buddy list xml. Thank you. Best regards, Mihai M

1 0

kamailio error, failed to start.
by Arya Farzan 27 Feb '14

27 Feb '14

I just installed kamailio on Debian 7 and when I start it with init.d I get the following messages. root@voip:/etc/kamailio# /etc/init.d/kamailio start Starting Kamailio: loading modules under /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: MYSERVERIP:5060 tcp: 127.0.0.1:5060 tcp: MYSERVERIP:5060 tls: 127.0.0.1:5061 tls: MYSERVERIP:5061 Aliases: tls: voip:5061 tls: localhost:5061 tcp: voip:5060 tcp: localhost:5060 udp: voip:5060 udp: localhost:5060 kamailio error, failed to start. root@voip:/etc/kamailio# How can I find out what the problem is?

3 4

Re: [SR-Users] Replacing an ACME Packet Net-Net SBC
by Raúl Alexis Betancor Santana 27 Feb '14

27 Feb '14

Sorry ... I disagree at all .... All that features you mention, could be done with kamalio ... BUT you have to work on your cfg to get them. You don't get then out of the box. Kamailio it's a powerfull tool ... It's not a final product as the AP Net SBC Best regards Luis Silva <luisfilsilva(a)gmail.com> wrote: >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

kamailio with mediaproxy-ng, 488 Not Acceptable Here
by Mihai Marin 26 Feb '14

26 Feb '14

Hello Sirs, I have a problem configuring kamailio with mediaproxy-ng and I'm asking for help. I managed to build everything, kamailio find support for mediaproxy-ng using rtpproxy-ng. When I'm trying to make a call from Web using my phone's internet provider to my computer's web I get 488 Not Acceptable Here. Swithing the caller I get no video. I have used the kamailio-advanced.cfg generated and added websocket support. The call in my network is working perfect. Can you help me with this? mediaproxy-ng log: mediaproxy-ng[14896]: Returning to SIP proxy: d7:createdi1390864117e7:streamslld3:tag10:trhh9viigs6:status34:known but unconfirmed peer address5:statsd3:rtpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv47:address13:213.233.85.554:porti48279ee23:advertised peer addressd6:family4:IPv47:address13:213.233.85.554:porti48279ee10:local porti30008ee4:rtcpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv47:address13:213.233.85.554:porti48280ee23:advertised peer addressd6:family4:IPv47:address13:213.233.85.554:porti48280ee10:local porti30009eeeed3:tag0:6:status20:unknown peer address5:statsd3:rtpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv67:address2:::4:porti0ee23:advertised peer addressd6:family4:IPv67:address2:::4:porti0ee10:local porti30010ee4:rtcpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv67:address2:::4:porti0ee23:advertised peer addressd6:family4:IPv67:address2:::4:porti0ee10:local porti30011eeeeee6:totalsd5:inputd3:rtpd7:packetsi0e5:bytesi0e6:errorsi0ee4:rtcpd7:packetsi0e5:bytesi0e6:errorsi0eee6:outputd3:rtpd7:packetsi0e5:bytesi0e6:errorsi0ee4:rtcpd7:packetsi0e5:bytesi0e6:errorsi0eeee6:result2:oke kamailio: 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=860 a=24 n=rtpproxy_manage 11(14926) DEBUG: rtpproxy-ng [rtpproxy.c:1333]: rtpp_function_call(): proxy reply: d7:createdi1390864117e7:streamslld3:tag10:trhh9viigs6:status34:known but unconfirmed peer address5:statsd3:rtpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv47:address13:213.233.85.554:porti48279ee23:advertised peer addressd6:family4:IPv47:address13:213.233.85.554:porti48279ee10:local porti30008ee4:rtcpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv47:address13:213.233.85.554:porti48280ee23:advertised peer addressd6:family4:IPv47:address13:213.233.85.554:porti48280ee10:local porti30009eeeed3:tag0:6:status20:unknown peer address5:statsd3:rtpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv67:address2:::4:porti0ee23:advertised peer addressd6:family4:IPv67:address2:::4:porti0ee10:local porti30010ee4:rtcpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv67:address2:::4:porti0ee23:advertised peer addressd6:family4:IPv67:address2:::4:porti0ee10:local porti30011eeeeee6:totalsd5:inputd3:rtpd7:packetsi0e5:bytesi0e6:errorsi0ee4:rtcpd7:packetsi0e5:bytesi0e6:errorsi0eee6:outputd3:rtpd7:packetsi0e5:bytesi0e6:errorsi0ee4:rtcpd7:packetsi0e5:bytesi0e6:errorsi0eeee6:result2:oke 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=867 a=16 n=if 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=862 a=24 n=is_request 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=866 a=16 n=if 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=863 a=24 n=has_totag 11(14926) DEBUG: siputils [checks.c:103]: has_totag(): no totag 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=864 a=25 n=add_rr_param 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=873 a=16 n=if 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=867 a=24 n=is_reply 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=873 a=2 n=return 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=1009 a=16 n=if 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=989 a=24 n=t_is_canceled 11(14926) DEBUG: tm [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T start=0x7f62174df4e0 11(14926) DEBUG: tm [t_lookup.c:1143]: t_check_msg(): DEBUG: t_check_msg: T already found! 11(14926) DEBUG: tm [t_reply.c:1827]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 11(14926) DEBUG: <core> [msg_translator.c:2007]: generate_res_buf_from_sip_res(): old size: 397, new size: 305 11(14926) DEBUG: <core> [msg_translator.c:2025]: generate_res_buf_from_sip_res(): copied size: orig:125, new: 33, rest: 272 msg= SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/WS ebhg3v0qb6fm.invalid;rport=1034;received=213.233.85.55;branch=z9hG4bK8048296 To: <sip:bob@93.187.138.214>;tag=r6dc2287g9 From: "Alice Test" <sip:alice@93.187.138.214>;tag=trhh9viigs Call-ID: fvgga4ikm8vrvuji0g0n CSeq: 6228 INVITE Content-Length: 0 11(14926) DEBUG: websocket [ws_frame.c:713]: ws_frame_transmit(): Tx message: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/WS ebhg3v0qb6fm.invalid;rport=1034;received=213.233.85.55;branch=z9hG4bK8048296 To: <sip:bob@93.187.138.214>;tag=r6dc2287g9 From: "Alice Test" <sip:alice@93.187.138.214>;tag=trhh9viigs Call-ID: fvgga4ikm8vrvuji0g0n CSeq: 6228 INVITE Content-Length: 0 Thank you. Best regards, Mihai M

8 27

coredump on using jsonprc_request
by davy van de moere 26 Feb '14

26 Feb '14

Gents, I’ve stumbled upon a coredump when using jsonrpc_request. I might be expecting the wrong things from jsonrpc_request. I was hoping to use it in combination with CNXCC, and have jsonrpc_request get a maximum duration for every call, as a fraud avoidance mechanism. I tried this with the branch 4.1 and master. Both behave identical. So from the kamailio-basic.cfg example, I just added this line around line 350 (just above route(SIPOUT)): jsonrpc_request("bar.sendsms", "{'ru':'$rU','rb':'$rb'}", "JSONRESPONSE", "JSONERROR", "$var(result)"); Doing a ngrep on the netstrings I have: interface: venet0 (127.0.0.2/255.255.255.255) filter: (ip or ip6) and ( port 7080 ) # T 10.66.8.42:44995 -> 192.168.60.45:7080 [AP] 94:{ "id": 2, "jsonrpc": "2.0", "method": "bar.sendsms", "params": { "ru": "", "rb": "" } }, ## T 192.168.60.45:7080 -> 10.66.8.42:44995 [AP] 43:{"jsonrpc": "2.0", "result": "12", "id": 2}, ####^Cexit 7 received, 0 dropped I do get some results from the JSONRESPONSE route. But when launching a call, or when sending just a register packet, kamailio gives up with: Feb 26 06:29:11 SBC01 kamailio[18043]: WARNING: tm [t_lookup.c:1537]: t_unref(): WARNING: script writer didn't release transaction Feb 26 06:29:12 SBC01 kamailio[18033]: : [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0xb4c01624), called from tm: h_table.c: free_cell(157), first free tm: h_table.c: free_cell(157) - aborting Feb 26 06:29:12 SBC01 kamailio[18023]: ALERT: [main.c:775]: handle_sigs(): child process 18033 exited by a signal 6 Feb 26 06:29:12 SBC01 kamailio[18023]: ALERT: [main.c:778]: handle_sigs(): core was generated Feb 26 06:29:12 SBC01 kamailio[18023]: INFO: [main.c:790]: handle_sigs(): INFO: terminating due to SIGCHLD Feb 26 06:29:12 SBC01 kamailio[18041]: INFO: [main.c:841]: sig_usr(): INFO: signal 15 received Feb 26 06:29:12 SBC01 kamailio[18047]: INFO: [main.c:841]: sig_usr(): INFO: signal 15 received Feb 26 06:29:12 SBC01 kamailio[18043]: INFO: [main.c:841]: sig_usr(): INFO: signal 15 received Feb 26 06:29:12 SBC01 kamailio[18048]: INFO: [main.c:841]: sig_usr(): INFO: signal 15 received Feb 26 06:29:12 SBC01 kamailio[18023]: : [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0xb4c01624), called from tm: h_table.c: free_cell(157), first free tm: h_table.c: free_cell(157) - aborting Using the jsonrpc_notification works like a charm. I got coredumps of it, if someone is interested. Side questions are offcourse, am I aiming to use jsonrpc_request for the correct purpose? Are there any other good alternatives? (next to using plain db queries?) Grtz, Davy -- davy van de moere

1 0

Kamailio World 2014 - Agenda
by Daniel-Constantin Mierla 26 Feb '14

26 Feb '14

The first version of the schedule during Kamailio World 2014 <http://www.kamailioworld.com> [1] has been published. Over 30 sessions grouped in two days of conference and pre-conference technical tutorials, spanning from various use cases for Kamailio to presentations approaching related VoIP applications such as Jitsi, Sems, Asterisk or FreeSwitch. Among the topics: WebRTC, IMS or VoIP security and fraud detection, tools for testing and troubleshooting VoIP systems, scalability with OpenStack and asynchronous processing, real time charging. First version of agenda has been published at: - http://conference.kamailio.com/k02/schedule/ Information about speakers can be found at: - http://conference.kamailio.com/k02/speakers/ Many thanks to the sponsors [2], they are making possible this event: FhG Fokus <http://www.fokus.fraunhofer.de/en/fokus/index.html>, Asipto <http://www.asipto.com>, Sipwise <http://www.sipwise.com>, Sipgate <http://www.sipgate.de>, IT Center <http://www.itcenter.com.pt/>, Simwood <http://www.simwood.com>, and NG Voice <http://www.ng-voice.com/>. Should you consider sponsoring, contact us via email at <conference [at] kamailio.com>**and we can present you more details. If you plan to attend, do not delay the registration, the number of seats is limited – see more details in Registration page <http://conference.kamailio.com/k02/registration/> [3]. /Looking forward to meeting many of you in Berlin! / Daniel/ / [1] - http://conference.kamailio.com [2] - http://conference.kamailio.com/k02/sponsors/ [3] - http://conference.kamailio.com/k02/registration/ -- Daniel-Constantin Mierla http://www.asipto.com

1 0

debian packages from opensuse.org
by Jöran Vinzens 26 Feb '14

26 Feb '14

Hi All, we are using Kamailio 4.1.0 packages from opensuse: http://download.opensuse.org/repositories/home:/kamailio:/telephony-debs/De… During installation it seems, the packages are malformed. Installation of kamailio-mysql-modules and kamailio-xml-modules stops during unpacking with failure: "trying to overwrite '/usr/lib64/kamailio/libsrdb2.so.1.0', which is also in package kamailio 4.1.0" it seem this one file is part of all three packages. There is a way to enforce the installation using dpkg -i --force-all.. but it seems not the right way. Does anyone have an idea how to solve without building own packages? Maybe this is an bug in kamailio Project? Many thanks for all proposals. BR Jöran -- Jöran Vinzens sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk

2 1

Re: [SR-Users] RTPProxy/Mediaproxy issue
by Daniel Grotti 25 Feb '14

25 Feb '14

Hi, Yes packet loss is very high. Please investigate around that. Daniel Ravi <wingsravi777(a)gmail.com> wrote: >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 5

Problems while reading from location and parsing expires
by Jöran Vinzens 25 Feb '14

25 Feb '14

Hi All, we are upgrading from kamailio 3.1 to 4.1 and we got stuck at a Problem with usrloc module. While starting up kamailio reads location from DB and try to parse expires date from DB. this is the Problem: Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: <core> [db_ut.c:90]: db_str2int(): Unexpected characters: [-02-25 11:27:08] Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: <core> [db_val.c:75]: db_str2val(): error while converting integer value from string Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: db_mysql[km_row.c:68]: db_ mysql_convert_row(): failed to convert value Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: db_mysql [km_dbase.c:375]: db_mysql_fetch_result(): error while converting row #0 Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: usrloc [udomain.c:420]: preload_udomain(): fetching rows failed usrloc Module is configured with db_mode "1" so it saves it's entries by itself. When i have a look on the database it looks good and the expires field is an datetime. From logs it looks like it try to parse a unixtimestamp instead of the datetime. when i truncate the location table it starts perfectly, since there are no entries stored where it could fail parsing expires date. does anybody has similar issue or any suggestions how to solve? found in Version: 4.1.0 many thanks! BR Jöran -- Jöran Vinzens sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk

2 4

Re: [SR-Users] Kamailio as redirect
by Daniel Tryba 24 Feb '14

24 Feb '14

On Monday 24 February 2014 14:45:18 Mike Claudi Pedersen wrote: > okay i somewhat get that.. but cant i use something like a lookup in a > textfile instead of a db? You can. alias_db also accepts dbtext as backend: http://kamailio.org/docs/modules/stable/modules/db_text.html But TIMTOWTDI, alias_db was just a suggestion. Please keep the mailing list included. -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

1 0

Re: [SR-Users] Configuring Kamailio as an upstream proxy for FreeSwitch and which RTP proxy to choose
by Fred Posner 24 Feb '14

24 Feb '14

On Mon Feb 24 10:50:07 CET 2014 Sean Kemball wrote: > New to Kamailio and FreeSwitch, loosely familiar with SIP mechanics, > and not a complete network idiot... but please be gentle. :) Welcome!s, > 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) Yes, you said that your upstream is on the same private network. So it should be pretty straight forward. > 2. Does it need a local RTP proxy on the Kamailio box, particularly > if we turn off the ASA SIP inspect stuff? If you are all on the same private network, I would let FreeSWITCH handle the RTP, but you can do this a variety of ways. > 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). On your scenario, I'd just use FreeSWITCH for the media proxy. Again, many different ways to go here. > 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. Your upstream provider would generally tell you which rtp ports they would want opened. > 5. Is static NAT in this environment going to bite me, or should > it be OK? I've never had an upstream provider communicate with me on private nat. > 6. Is there any better documentation that we should be using to > make this easier, or should I just man up and try harder? Man up. =) Practive makes perfect. -- Fred Posner The Palner Group, Inc.

1 0

pcscf - removing registration entries from hash table
by Daniel Ciprus 24 Feb '14

24 Feb '14

Hello, Our current setup is Kamailio as IMS core and we're observing following behavior: Clients register themselves over TCP so pcscf creates hash table which keeps these information in memory. Once tcp reset occurs for any reason, registration entry stays in the memory and client re-registers himself with new pair ip:port. This kind of behavior creates problem because outbound SIP messages are being forked and requests are sent to non-existent tcp sessions which creates a lot of timeouts and confusion for AS. We're using ims_usrloc_pcscf module to keep reg entries in and as I noticed, with usrloc module, there was an option which automatically removed entry from hash table once tcp session was closed which is not the case in this module. Any suggestions how to get out of this ? thanks for any hints/suggestions. -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

1 0

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 at 3 different servers eg. 1000(a)192.168.1.10 1001(a)192.168.1.11 1002(a)192.168.1.12 then i call user 1002 from user 1000, through the kamailio redirect server, which redirects to corresponding servers. Mike Claudi Pedersen

2 1

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 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. #############################################################################

1 0

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) = $rU; if (avp_db_load("$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

3 4

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 <----------------------------> Switch<----------------------------> Router | | clientA <--------------------- RTP(audio/video) ? --------------------> client B Now when the SIP session established (SIP Packets have to pass through all the routers and switch to reach Kamailio server), RTP session will start between Client A and B. But What is meant by client to client here ? what is the RTP packets path between the clients to reach their destinations ? Is the RTP packets have to pass through Routers and switch to reach its destination client ? (i.e Client A--->Router 1---->switch -----> Router 2-----> client B) ? What if i connect both client A and client B to the same router ? PS: I have not configured any media-relays (RTPproxy/Media-proxy) Please help me in clarifying all this doubts. Anybody's help will greatly appreciate. Regards, Nandini

1 0

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 11:23 AM, "Uri Shacked" <ushacked at gmail.com <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> wrote: >>* 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*

2 1

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 but the results are same as like in Kamailio/RTPproxy case. What could be the problem with this set-up ? How can i solve this issue ? Please anybody help me in resolving this problem. Any help will greatly appreciate. Regards. Ravi.

4 7

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

2 1

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

3 3

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 only have one branch, but it doesn't enter failure route for 30x if you have another active branch (e.g. in ringing state). Would it be possible to create a new branch directly from the reply route, or force to enter failure route for a specific reply/branch? Andreas

3 5

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 : Invalid arguments Line 311 is kamailio.cfg is "modparam ("usrloc", "db_url", DBURL)" And 'DBURL' is defined as #! define DBURL "mysql://openser:5mil5My5ql@192.168.1.20/openser" Can some one please help me out in resolving this error. Secondly I am deploying my database on seperate server for the first time. I have deployed kamailio-3.1 already but with database placed on same machine. Can someone please guide what changings do I need to make in kamailio.cfg and kamctlrc file to make it work. I have made some as per my understanding. Maybe they are a cause of this error. Thanks, Regards, Owais

2 1

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

1 0

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users