- sr-users - kamailio.org

Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways
by Sarat C. Vemuri 07 Sep '11

07 Sep '11

Ovidiu, Thanks for your time. The "fixes" I pulled in is the latest rr module only. I didn't see anything in that diff to make this work. Could you elaborate a little? However, I was able to get "Outbound" working properly by adding ";r2=on" to the record_route_preset IPs. In going through the source code, I noticed that this is what regular record_route uses to enable double rr and it would automatically take both headers out if it sees that. That seems to have worked. Any pitfals with this? I am still working on "Inbound". For some reason my carrier GW keeps resending invites even after receiving ACK. I need to see if it is an issue with the carrier. Thanks SV. -----Original Message----- From: Ovidiu Sas [mailto:osas@voipembedded.com] Sent: Tuesday, September 06, 2011 9:50 AM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways It seems that something is miss configured on your server. The fixes that I made in the trunk (and you pulled in your local 3.1 repo) were designed to handle the scenario that you are trying to implement. The ACK should be handled properly and routed to the upstream carrier (following the same path as the initial INVITE). Regards, Ovidiu Sas On Mon, Sep 5, 2011 at 5:07 PM, Sarat C. Vemuri <sarat.vemuri(a)fthco.com> wrote: > Again, I apologize for this clumsy way of replying. > > Ovidiu, > > Thanks for the pointer on set_advertised_address. I had to patch rtpproxy module (and rr module for the two parameters to request_route_preset) since I am running 3.1. > > However, I still have a problem with ACKs after following what you suggested. > > INVITE from Internal to Carrier routes properly (two Request-Route headers, one internal IP and other public IP). On 200 OK, the carrier GW properly copies the route set in to Route header. Now the route contains two entries, the public IP and the private IP of Kamailio. The Internal UAC then sends the ACK back to Kamailio. Everything is fine till this point. Now, Kamailio removes the top entry, which is the private IP and then promptly sends the ACK to the public IP of itself!. Of course, that doesn't go anywhere. > > How do I remove the "public IP" entry from the route set before forwarding the reply to Internal UAC? Is there another way to deal with this? I've tried to set an alias= core parameter with the public IP, but doesn't seem to have any effect. The public IP is not reachable from internal network. > > Thanks for your help > > SV. > ------------------------------ > > Message: 3 > Date: Sat, 3 Sep 2011 16:44:22 -0700 > From: Ovidiu Sas <osas(a)voipembedded.com> > Subject: Re: [SR-Users] Kamailio + rtpproxy talking to multiple > carrier gateways - some via Firewall/NAT > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > > <CAND0Lkt_dpcTm2WKMywMhX6rdsX1ia0r=LYrzB1WfX3oN32Wrg(a)mail.gmail.com> > Content-Type: text/plain; charset=windows-1252 > > It is feasable to what you want: kamailio behind NAT proxying traffic > from/to public internet to/from private network. > You will need to properly craft the INVITE and use proper record route headers. > Use set_advertised_address when needed: > http://kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#set_advertis > ed_address Also, use record_route_preset (note that there are two > parameters): > http://kamailio.org/docs/modules/devel/modules_k/rr.html#id2547566 > > That should do it. You don't need any patches for rtpproxy. > Just use force_rtpp_proxy (and force the IP address): > http://kamailio.org/docs/modules/devel/modules/rtpproxy.html#id2546034 > > > Note: Make sure that you understand how in-dialog requests are routed > by a proxy in order to know how to properly handle the initial INVITE. > > > Regards, > Ovidiu Sas > > On Sat, Sep 3, 2011 at 2:53 PM, Sarat C. Vemuri <sarat.vemuri(a)fthco.com> wrote: >> We are trying to configure Kamailio ?(3.1.x) as a ?boarder proxy? >> where it acts as the front for various carrier gateways so that >> internal UACs and UASs are unaware of the carrier gateways. >> >> >> >> Let me try to present a clear picture of our setup. >> >> 1.?????? Kamailio has several NICs (physical or vlan).? Each on a >> different subnet. One subnet is internal/has routes for internal.? >> Other subnets are private connections to carriers or a ?route to public Internet. >> >> 2.?????? All of these subnets are non-routable from Internet. In >> addition , the carrier private connections are not routable internally. >> >> 3.?????? Connection to public internet is via a FW/NAT (one-to-one >> NAT), which maps to one of the interfaces. >> >> 4.?????? All internal? UAC/UAS connect on the internal subnet. >> >> 5.?????? We are using RTPProxy? (at least one instance per carrier) >> to relay media between internal and carrier subnets >> >> >> >> We are able to make this setup up work great except for one >> scenario.? One of the carriers is only reachable via public >> Internet.? Due to security requirements, our Kamailio cannot have a >> public IP address and must use FW/NAT. I guess this scenario is >> ??Proxy behind NAT? and not really encouraged. But I would like to >> see if there is a way to make this work.? We cannot use the >> ?advertised_address? since it changes the IP for every ?route?. >> >> >> >> We were able to get this mostly ?working by doing the following >> >> 1.?????? mhomed=1 >> >> 2.?????? Small patch in the rtpproxy module so that ?force_rtp_proxy >> actually uses the IP address passed (public IP). >> >> 3.?????? Using request_route_preset(?publicIP?) >> >> >> >> The above ?mostly? works.? By that I mean, the INVITE transaction is >> properly passed between internal UAS and carrier SBC and the call is setup. >> However, further transactions (BYE/re-INVITE) etc do not work >> properly. So, far-end hangups are not ?working etc. >> >> >> >> I?ve searched various archives of this and other SER lists looking to >> see if anyone was able to get this scenario working, but couldn?t >> find a definitive answer.? Most of them point to using ?advertised_address?. >> >> >> >> So, and ideas on how to make ?Proxy behind NAT? work without using >> advertised_address?? Am I wasting my time? >> >> >> >> Thanks in advance for any help you can offer. >> >> >> >> SV. >> >> >> >> _______________________________________________ >> 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 >> >> > > > > _______________________________________________ > 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 1

permanent usrloc entries in Kamailio
by Klaus Darilion 07 Sep '11

07 Sep '11

Hi! Recently I had a problem with permanent usrloc entries when upgrading a server from SER 0.9 to Kamailio 3.1. Obviously the problem was that the permanent entries had "flag=128" and now expires=0 is used, see old thread: http://old.nabble.com/usrloc-permanent-records-and-strptime-td17668417.html The strange thing is, that even setting the expiry to 2099-0...... did not worked and Kamailio deleted the "expired" entries. Only setting it to "1970-01-01 01:00:00" worked. Any ideas what could been wrong? Further I wonder why Kamailio uses 01:00:00 for the permanent entry? The server is currently running in timezon CEST (UTC+2), thus IMO 02:00:00 would be more logical. Above thread also mentions problems with day-light-saving changes. Are there really problems? How do you solve those problems? I think the best workaround is to set system time always to UTC. Thanks Klaus

2 2

Siremis User Role permissions
by Camila Troncoso 07 Sep '11

07 Sep '11

Hi, I have a question about the assignment of Role permissions to limit some Users privileges in Siremis. I defined a user , and then a Role denying some actions, but I couldn’t deny the action from de SER menu . The Role permissions only show the modules on Application Menu from Administration Tab . What if I want to deny access to some items inside SER Menu , like modifying the ACL services or /and Rounting services? Regards, *Camila Troncoso **|* Ingeniero de Desarrollo VoissNet S.A. *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

2 1

Check disconnected users
by Anto 06 Sep '11

06 Sep '11

Hello I want to store in a file access and disconnections to the system. To access (registrar), I thought scheduling a perl script and executed with the perl module, the problem I find with disconnections. Since the user can be disconnected for various reasons, had planned to do a script that runs an SQL table location and according to the expires field, see if disconnection has occurred or not. Would have to run this script in cron every X minutes. Is there a simpler way to check this or have Kamailio some function for it?. Thank you. Best regards

2 2

Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways
by Sarat C. Vemuri 06 Sep '11

06 Sep '11

Again, I apologize for this clumsy way of replying. Ovidiu, Thanks for the pointer on set_advertised_address. I had to patch rtpproxy module (and rr module for the two parameters to request_route_preset) since I am running 3.1. However, I still have a problem with ACKs after following what you suggested. INVITE from Internal to Carrier routes properly (two Request-Route headers, one internal IP and other public IP). On 200 OK, the carrier GW properly copies the route set in to Route header. Now the route contains two entries, the public IP and the private IP of Kamailio. The Internal UAC then sends the ACK back to Kamailio. Everything is fine till this point. Now, Kamailio removes the top entry, which is the private IP and then promptly sends the ACK to the public IP of itself!. Of course, that doesn't go anywhere. How do I remove the "public IP" entry from the route set before forwarding the reply to Internal UAC? Is there another way to deal with this? I've tried to set an alias= core parameter with the public IP, but doesn't seem to have any effect. The public IP is not reachable from internal network. Thanks for your help SV. ------------------------------ Message: 3 Date: Sat, 3 Sep 2011 16:44:22 -0700 From: Ovidiu Sas <osas(a)voipembedded.com> Subject: Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways - some via Firewall/NAT To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <CAND0Lkt_dpcTm2WKMywMhX6rdsX1ia0r=LYrzB1WfX3oN32Wrg(a)mail.gmail.com> Content-Type: text/plain; charset=windows-1252 It is feasable to what you want: kamailio behind NAT proxying traffic from/to public internet to/from private network. You will need to properly craft the INVITE and use proper record route headers. Use set_advertised_address when needed: http://kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#set_advertised_ad… Also, use record_route_preset (note that there are two parameters): http://kamailio.org/docs/modules/devel/modules_k/rr.html#id2547566 That should do it. You don't need any patches for rtpproxy. Just use force_rtpp_proxy (and force the IP address): http://kamailio.org/docs/modules/devel/modules/rtpproxy.html#id2546034 Note: Make sure that you understand how in-dialog requests are routed by a proxy in order to know how to properly handle the initial INVITE. Regards, Ovidiu Sas On Sat, Sep 3, 2011 at 2:53 PM, Sarat C. Vemuri <sarat.vemuri(a)fthco.com> wrote: > We are trying to configure Kamailio ?(3.1.x) as a ?boarder proxy? where it > acts as the front for various carrier gateways so that internal UACs and > UASs are unaware of the carrier gateways. > > > > Let me try to present a clear picture of our setup. > > 1.?????? Kamailio has several NICs (physical or vlan).? Each on a different > subnet. One subnet is internal/has routes for internal.? Other subnets are > private connections to carriers or a ?route to public Internet. > > 2.?????? All of these subnets are non-routable from Internet. In addition , > the carrier private connections are not routable internally. > > 3.?????? Connection to public internet is via a FW/NAT (one-to-one NAT), > which maps to one of the interfaces. > > 4.?????? All internal? UAC/UAS connect on the internal subnet. > > 5.?????? We are using RTPProxy? (at least one instance per carrier) to relay > media between internal and carrier subnets > > > > We are able to make this setup up work great except for one scenario.? One > of the carriers is only reachable via public Internet.? Due to security > requirements, our Kamailio cannot have a public IP address and must use > FW/NAT. I guess this scenario is ??Proxy behind NAT? and not really > encouraged. But I would like to see if there is a way to make this work.? We > cannot use the ?advertised_address? since it changes the IP for every > ?route?. > > > > We were able to get this mostly ?working by doing the following > > 1.?????? mhomed=1 > > 2.?????? Small patch in the rtpproxy module so that ?force_rtp_proxy > actually uses the IP address passed (public IP). > > 3.?????? Using request_route_preset(?publicIP?) > > > > The above ?mostly? works.? By that I mean, the INVITE transaction is > properly passed between internal UAS and carrier SBC and the call is setup. > However, further transactions (BYE/re-INVITE) etc do not work properly. So, > far-end hangups are not ?working etc. > > > > I?ve searched various archives of this and other SER lists looking to see if > anyone was able to get this scenario working, but couldn?t find a definitive > answer.? Most of them point to using ?advertised_address?. > > > > So, and ideas on how to make ?Proxy behind NAT? work without using > advertised_address?? Am I wasting my time? > > > > Thanks in advance for any help you can offer. > > > > SV. > > > > _______________________________________________ > 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 > >

3 2

KAMAILIO RTP PROXY
by Phillman25 Kyriacou 06 Sep '11

06 Sep '11

Hi I want to use Kamailio to handle both signalling and media always. My scenario is the following PSTN==>KAMAILIO==>ASTERISK. I have a dialplan that points to ASTERISK when a call comes into Kamailio. Signalling is fine, however i cant seem to get the Media to work, i have installed the RTPPROXY as suggested by RTPPROXY module and have implemented it into the routing logic, however i have one way audio. The PSTN side cannot hear anything. I dont use the Kamailio for sip registrations or for users behind NAT. I just use a simple dialplan for forwarding SIP messages to our certain VOIP carriers, however some of our carriers request that we use both signalling and media from the same IP thats why i need to implement this scenario. Is this possible? Thanking you in advance for your help! Phillip

2 1

db_mysql "Commands out of sync; you can't run this command now "
by MÉSZÁROS Mihály 06 Sep '11

06 Sep '11

Hi, I am using the git version of kamailio, and i am experiencing problem This message is repeated in log with all kind of mysql query (insert/delete/select) Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: db_mysql [km_dbase.c:124]: driver error on query: Commands out of sync; you can't run this command now Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: <core> [db_query.c:103]: error while submitting query Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: usrloc [udomain.c:570]: db_query failed Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: db_mysql [km_dbase.c:124]: driver error on query: Commands out of sync; you can't run this command now Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: <core> [db_query.c:186]: error while submitting query Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: usrloc [ucontact.c:513]: inserting contact in db failed Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: usrloc [urecord.c:474]: failed to insert in database Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: registrar [save.c:430]: failed to insert contact Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: db_mysql [km_dbase.c:124]: driver error on query: Commands out of sync; you can't run this command now Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: <core> [db_query.c:242]: error while submitting query Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: usrloc [urecord.c:426]: failed to delete from database Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: usrloc [udomain.c:896]: DB delete failed I am using stable debian so db_mysql.so linked with libmysqlclient.so.16 I read mysql doc and it is saying about this error the following: "Commands out of sync; you can't run this command now" http://dev.mysql.com/doc/refman/5.0/en/commands-out-of-sync.html So this problem cause it that not freed the last result. Only one assumption: Could it related to this change: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=df… Any help appreciated. Regards, Misi

3 18

Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways
by Sarat C. Vemuri 05 Sep '11

05 Sep '11

Alex, Sorry about this clumsy way of replying, but I am stuck with Outlook 2010 and Exchange. You stated the problem correctly. Carrier GWs (UAS) and internal clients (UAC) cannot reach each other directly (TCP/UDP). They have to go through Kamailio and rtpproxy. I am goint to reply to Ovidiu as well with some more details. Thanks SV. ------------------------------ Message: 2 Date: Sat, 03 Sep 2011 19:12:38 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways - some via Firewall/NAT To: sr-users(a)lists.sip-router.org Message-ID: <4E62B466.20808(a)evaristesys.com> Content-Type: text/plain; charset=windows-1252; format=flowed On 09/03/2011 05:53 PM, Sarat C. Vemuri wrote: > 3.Using request_route_preset(?publicIP?) > > The above ?mostly? works. By that I mean, the INVITE transaction is > properly passed between internal UAS and carrier SBC and the call is > setup. However, further transactions (BYE/re-INVITE) etc do not work > properly. So, far-end hangups are not working etc. So in other words, there is no interface to which Kamailio could be bound on a network that both the UAC and the UAS have direct network and transport-layer reachability to? If I understand correctly, it sounds like the fundamental problem here is that the Record-Route URI introduced by the proxy in handling the initial INVITE request has to be conserved in all subsequent requests and replies. It is a requirement to use the same route set in sequential (in-dialog) requests such as reinvites or BYEs. It is also mandatory for the UAS to copy the received Record-Route into the final reply (i.e. 200 OK). The problem is that you then have to choose the network address in the Record-Route. If either the UAC or the UAS cannot directly reach it, your problem arises. Is this an accurate statement of the problem? I wanted to verify before continuing. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

1 0

SIP Analysis & Testing tools
by Vladimir Broz 05 Sep '11

05 Sep '11

Hi, for sake of Vaclav's courage, let me introduce the "SIP Analysis & Testing tools". This is a set of very powerful utilities that may help you to do SIP call flows analysis. It can do SIP dialogs and transactions matching and analysis in large pcaps, create pretty call flow diagrams, check and verify SIP message content (headers, parameters,..) and many more.... It's available on: https://github.com/vkubart/sipat For more details, please ask Vaclav Kubart (vaclav.kubart(a)iptel.org). SIP Analysis & Testing tools ============================ This project offers set of tools for testing SIP. Dependencies (ubuntu packages) ============ libosip2-dev libpcap-dev libxml2-dev xsltproc cfanal ====== Call flow analyzer able to draw call flows from PCAP and verify callflows against given definition. Usage: cfanal -r captured-traffic.pcap -a 10.38.14.73=uac -a 10.38.14.74:5062=uas -print-cf drawing.cf cf2svg drawing.cf drawing.svg svg2png drawing.svg drawing.png callflow_diagrams ================= XSL template for painting call flow diagrams. ..... Thank you Vaclav a lot! -Vlada

4 4

question
by Allen D. Ford 05 Sep '11

05 Sep '11

Where are the instructions for this gui Im trying to setup a sip load balancer and rtp proxy for nating "Success is not measured by the positions you reach in life but by the obstacles you overcome -B.T.W" [Description: Description: Description: Description: Description: cyford]<http://www.cyfordtechnologies.com/>Director Of Technology Allen D. Ford 4290-106 Bells Ferry Road, Suite 20 , Kennesaw , GA 30144 Tel: 404-424-8270: Fax: 404-424-8271 Email: allen(a)cyfordtechnologies.com<mailto:allen@cyfordtechnologies.com> Web Address: http://www.cyfordtechnologies.com <http://www.cyfordtechnologies.com/> LinkedIn: http://www.linkedin.com/in/cyford FaceBook: http://www.facebook.com/profile.php?id=100002260975450 NOTICE: This email message is for the sole use of the intended recipient(s) and may contain confidential and privileged information. Any unauthorized review, use, disclosure, or distribution is prohibited. If you are not the intended recipient, please destroy all copies of the original message and notify the sender so that our address record can be corrected. Thank you

2 1

10 Years SER - Awards
by Daniel-Constantin Mierla 05 Sep '11

05 Sep '11

Hello, a surprise part of 10 Years SER Conference was granting *special* awards to the people attending the event that had a major contribution in the management and development of SER/Kamailio to date. Most of them are well known in the community, but not all of them. Hope you like them, it proved to be a very entertaining moment. Here is the pdf with the awards: * http://asipto.com/u/48 The plan is to extend it with the other people in our community as ideas with award titles pop up. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

10 Years SER - Follow UP
by Daniel-Constantin Mierla 05 Sep '11

05 Sep '11

Hello, first I want to thank to all participants at 10 Years SER event in Berlin last Friday. We spent great time, at least I did! We accepted 50 visiting participants to match the maximum capacity of the conference room, being full booked about 2 weeks before. Just few people didn't show up and didn't inform us at all in advance, which was a pity over all since several persons that registered later were not accepted, thus some of them could have attended. Folks at Fokus working with SER/Kamailio joined us and we had probably one of the very few occasions so far to get so dense knowledge of SIP/VoIP in the same place, making a day of interesting presentations and open discussions. Accompanied by nice weather in Berlin, the evening at Playa Paradiso on river Spree offered a great place for continuing the technical and mufti-cultural debates (I could count attendees from about 15 countries (including Brazil!)). Many thanks again to our sponsors for free drinks and food at this event: Sipgate (http://www.sipgate.de), Amooma (http://www.amooma.de), NG-Voice (http://www.ng-voice.com), Asipto (http://www.asipto.com), Frafos (http://www.frafos.com) and Fokus Institute (http://www.fokus.fraunhofer.de). I will upload the presentations soon and send an update here -- there were 15 talks, covering from technical concepts to business matters and open source philosophy. Therefore if you were a speaker and see this message, please send the pdf slides to me. Do not forget, you can join the next party for this celebration this week in Vienna, details at: * http://sip-router.org/10-years-ser-vienna/ Meanwhile, be sure you start/continue testing the upcoming release v3.2.0, we have to be sure it is stable to release it and start a new development cycle in the near future, to be able to present new features at the 11th celebration :-) . Here is a step by step guide to pull the code from GIT and install: * http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-devel-from-git Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 1

tls + tcp problems
by MÉSZÁROS Mihály 05 Sep '11

05 Sep '11

Hi, I am experiencing two problem: First: I am experiencing that tls read buffer overflow, if i use default value of tcp_rd_buf_size If i doubled the size of read buffer and i use tcp_rd_buf_size=8192 then the problem seems to be solved. Second: I am experiencing that TLS connection is dropped by sip proxy very frequently! And i think the tcp connection shouldn't dropped! So my guess is that TLS communication is not restarting the timeout counters! So in every 2 minutes the sip-router is restarting the TCP connection with tcp RST. like this: Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core> [tcp_main.c:4304]: tcp_main: entering timer for 0xac537c74 (ticks=446299809, ti meout=446299809 (0 s), wr_timeout=0 (240541717 s)), write queue: 0 bytes Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core> [tcp_main.c:4346]: tcp_main: timeout for 0xac537c74 Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x824cf80, 48, -1, 0x10) fd_no=41 called Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: tls [tls_server.c:597]: Closing SSL connection 0xac4a0374 Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core> [ip_addr.c:247]: tcpconn_new: new tcp connection: 10.10.10.10 Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core> [tcp_main.c:1081]: tcpconn_new: on port 36142, type 3 Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core> [tcp_main.c:1382]: tcpconn_add: hashes: 2473:3781:3367, 5 This second problem is not solved and very urgent for me. Any help suggestion is highly appreciated! Thanks, Misi

2 1

Carrier module for kamailio
by Mino Haluz 05 Sep '11

05 Sep '11

Hi, I need a module which could allow me to send traffic to various carriers and it has to support some important features. So some basic ones: - possibility to re-route the call in case the original route fails - peak/offpeak conditions (time-based) - route traffic according to prefix I found LCR and carrierroute module, but it does not have peak/offpeak feature. Correct me if I am wrong..

2 1

new developer: Sven Knoblich
by Henning Westerholt 05 Sep '11

05 Sep '11

Hi all, I'm happy to announce a new developer for the sip-router project: Sven Knoblich. Sven is an experienced C/C++ developer in our offices in Karlsruhe and since about a year part of the team that develops our VoIP backend services. Past contributions of him were the CDR based accounting extensions in the acc module and several other bugfixes and extensions for the dialog module. He'll maintain in the future this new acc and dialog functionality and also other code contributed from us. Best regards, Henning -- Henning Westerholt - Head of IT Operations Internet Access & Communications 1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany

3 2

Pass all rtp to RTPProxy
by Stas Bakulin 05 Sep '11

05 Sep '11

Hello! What is it right way to pass all rtp traffic through RTPProxy? How this can be configured in Kamailio? Stas Bakulin ---------------------------------------------------------------------- ogion(a)kvant-telecom.ru | www.kvant-telecom.ru Tel: +7 473 233 0330 (128)

2 1

Kamailio + rtpproxy talking to multiple carrier gateways - some via Firewall/NAT
by Sarat C. Vemuri 03 Sep '11

03 Sep '11

We are trying to configure Kamailio (3.1.x) as a "boarder proxy" where it acts as the front for various carrier gateways so that internal UACs and UASs are unaware of the carrier gateways. Let me try to present a clear picture of our setup. 1. Kamailio has several NICs (physical or vlan). Each on a different subnet. One subnet is internal/has routes for internal. Other subnets are private connections to carriers or a route to public Internet. 2. All of these subnets are non-routable from Internet. In addition , the carrier private connections are not routable internally. 3. Connection to public internet is via a FW/NAT (one-to-one NAT), which maps to one of the interfaces. 4. All internal UAC/UAS connect on the internal subnet. 5. We are using RTPProxy (at least one instance per carrier) to relay media between internal and carrier subnets We are able to make this setup up work great except for one scenario. One of the carriers is only reachable via public Internet. Due to security requirements, our Kamailio cannot have a public IP address and must use FW/NAT. I guess this scenario is "Proxy behind NAT" and not really encouraged. But I would like to see if there is a way to make this work. We cannot use the "advertised_address" since it changes the IP for every "route". We were able to get this mostly working by doing the following 1. mhomed=1 2. Small patch in the rtpproxy module so that force_rtp_proxy actually uses the IP address passed (public IP). 3. Using request_route_preset("publicIP") The above "mostly" works. By that I mean, the INVITE transaction is properly passed between internal UAS and carrier SBC and the call is setup. However, further transactions (BYE/re-INVITE) etc do not work properly. So, far-end hangups are not working etc. I've searched various archives of this and other SER lists looking to see if anyone was able to get this scenario working, but couldn't find a definitive answer. Most of them point to using "advertised_address". So, and ideas on how to make "Proxy behind NAT" work without using advertised_address? Am I wasting my time? Thanks in advance for any help you can offer. SV.

3 2

10 years of SER - statistics
by Daniel-Constantin Mierla 31 Aug '11

31 Aug '11

Hello, it is now one month till the 10 years SER event in Berlin. I am thinking to collect few statistics about usage of SER-based code at this moment (SER, OpenSER, Kamailio, OpenIMSCore ...) and present them during the conference and publish on the web. Everything is going to be like a combined report, not individual listing. Therefore, if you want to participate, your name or company won't be mentioned. Here are the stats I thought of: - type of usage: production, evaluation (testing), research - number of subscribers (phone lines) - percentage of phones behind NAT - number of calls per month - number of call minutes per month I am not looking for exact numbers, but rough estimation (e.g., about 10 000 phone lines, ...). Of course, some of the metrics don't apply always (e.g., if you do termination routed through SIP server, you don't have subscribers, but just calls traffic). Those doing deployments, can make one summary of statistics for all instances. You may send other statistics you think they worth publishing. Again, this is voluntary, naming is not required. I hope only those giving real number will write back -- I will try anyhow to figure out if someone is just dumping fantasy numbers. If privacy is really a big concern, from case to case, everything can be done under NDA. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 3

Resend INVITE in case of failure
by Mino Haluz 31 Aug '11

31 Aug '11

Hi, I would like to resend the INVITE to another gateway if the original fails (503, 500, etc.). Is there any module or mechanism which could do this ? Mino

4 3

Celebrating 10 years SER SIP Router in Vienna: 8th September 2011
by Klaus Darilion 31 Aug '11

31 Aug '11

Hi! If you haven't noticed yet, SER (the mother of the SIP proxy projects Openser, Kamailio, sip-router, opensips, ....) is celebrating their 10th year. There will be a main event happening in Berlin (http://sip-router.org/10-years-ser/). For those who can not travel to this event, there will be some smaller events around the world. In Vienna, this will happen on the 8th September at 7pm in the evening, organized by Andreas Granig and myself. We do not plan to have a big event with speaker, it is just an informal come together to meet with other SER-interested[1] guys, having some techie talks, maybe getting some business contacts and drinking some beers. Drinks will be sponsored by http://www.ipcom.at/ and http://www.sipwise.com/. The event will happen in the city center of Vienna. The location is not fixed yet, it depends on how many people are going to attend. Thus, please let us know if you going to attend (send us a short e-mail) and spread this e-mail to interested people you know. Cheers Andreas and Klaus [1] User's of all SER-flavors are welcome

1 1

Kamailio setup help - needed in Hamburg / Germany
by Rene Bormann 30 Aug '11

30 Aug '11

Hello SR List, our background is classic telephony with E1 interfaces, active telephony-cards etc. We now got our telephone-plattform transformed for SIP, but all behind a firewall. Actually we receive calls from some carriers throu the firewall on our Asterisk, which has E1 links to the IVR -Plattform. Now we want to eliminate Asterisk as kinda internal PSTN gateway and direcly route the calls throu Kamailio to our IVR services. So we thought, hey Kamailio can't be that complicated to setup, but now we have to admit - that we are kinda "LOST". We are located in Hamburg / Germany and need HELP. Is on this list anyboy - who can help us, maybe from around Hamburg? Get us on the kamailio train or maybe help us on a continous bases? Thanks from Hamburg, René

2 1

Kamailio 3.1.1 snmpstat help!!
by Stephen Dodge (Bistech) 30 Aug '11

30 Aug '11

Hello, I am having difficulty running the snmpstats module. We are running CentOS 5.6 and Kamailio version 3.1.1 (i386/linux). I've tried a fresh install of Kamailio, installing net-snmp from source but I can't seem to get it running - I've seen a couple of posts regarding the issue but nothing seems to work for us!! The issue I get when I start Kamailio is: 0(27523) ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/snmpstats.so>: /usr/lib/libnetsnmpagent.so.10: undefined symbol: hosts_ctl Any help would be much appreciated. Also, if anyone has any suggestions on how best to get either Solarwinds or PRTG to monitor our Kamailio servers that would be good. Thanks. Stephen Dodge Visit www.bistech.co.uk<http://www.bistech.co.uk/> for news and information Regards Stephen Dodge Visit www.bistech.co.uk<http://www.bistech.co.uk/> for news and information ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

2 4

registered("location") not working
by Skyler 30 Aug '11

30 Aug '11

Hi all, I am attempting to only reply to NOTIFY when the request is originating from a user/device that is authenticated on kamailio already. There are 2 ways I've tried (below) but neither will work. Any idea on what I am doing wrong or suggestions on how to do it right? TIA, Skyler if(is_method("NOTIFY") && is_registered("location")) { route(ROUTE_NOTIFY); } OR .... if(is_method("NOTIFY")) { # We only reply OK if coming from authenticated users or trusted peers if(!registered("location") || !from_any_gw("$avp(s:ip)")) { sl_send_reply("503", "Service Unavailable"); exit; } xlog("L_INFO", "Friendly Peer Poke - M=$rm F=$fu R=$ru\n"); sl_send_reply("200", "OK"); exit; }

2 3

SEMS 1.4.2 released
by Stefan Sayer 29 Aug '11

29 Aug '11

(apologies if you receive multiple copies of this mail) Hello, the SIP Express Media Server version 1.4.2 is available. Users of SEMS 1.4 versions and anyone else is recommended to upgrade to that version. Please find below the relevant changelog, the source at ftp://ftp.iptel.org/pub/sems/sems-1.4.2.tar.gz and Debian/Ubuntu packages as always at OBS https://build.opensuse.org/project/show?project=home%3Ateam-sems Many thanks to everyone who contributed with testing, bug fixes, documentation and feedback. * SEMS 1.4.2 release * auth'ed BYE (wait_for_bye_transaction) * fixes SIP auth for qop header format * xmlrpc: fix busyloop with keep-alive * a few minor SST issues * builds on Ubuntu 11.4 (build-deps) * ivr: release GIL on blocking file I/O * SBC: fix codec filter for unnamed payloads<96 * fixed DSM variables to outgoing call * some examples and documentation added This release introduces a new configuration option wait_for_bye_transaction=yes which enables authentication of BYE requests sent by SEMS, which is disabled by default. If this option is not enabled, SEMS behaves like in 1.4.1. With Best Regards Stefan Sayer for the SEMS community -- frafos.com

2 2

Re: [SR-Users] Kamailio as simple SIP proxy between client and SIP server.
by Edson - Lists 29 Aug '11

29 Aug '11

Hi, Henrik... Please, never, ever, forget to copy the list in a thread initiated there... other may be "curious" on it... ;) About your configuration, it seems that you just need a 'simple forward' server. It's very simple to accomplish... Make a simple forward configuration script (you can start with the provided one and cut all registration and whistles). Just be aware of all security issues binded to this choice. Ok... you still can load any of the control modules (pike, p.ex.) and have some prevention, but even so it's not your best choice. Edson. Em 29/08/2011 12:13, Henrik Aagaard Sørensen escreveu: > Hi Edson. > > I do know the different definitions as SIP-Proxy, SIP-Server etc. (I hope). > > That's why I want Kamailio to act as a SIP-Proxy, not a SIP-Server, as > it does currently. > > As I've understood Kamailio and SIP-Router is pretty much the same > thing. Going to sip-router.org <http://sip-router.org> for downloads > also refers to Kamailio. > > So shouldn't it be possible to use Kamailio as SIP-Proxy? Or have I > misunderstood that. > > I do not want my Kamailio to take registrations etc. It should just > forward SIP messages between client and SIP-Server, as I'm not allowed > to register towards my current SIP-Server with my current IP. But my > server, where the Kamailio is running, is allowed with it's IP. > > 2011/8/29 Edson - Lists <4lists(a)gmail.com <mailto:4lists@gmail.com>> > > Hi, Henrik... > > Sip-Router is "just" a SIP Proxy... but a dam good one... > > You have to take a step back... You are mixing concepts when > affirming that SR is playing SIP-Server roles... Please take a > better look on the definitions (SIP-Proxy, SIP-Server, Registrar, > etc) and find wich system play each role. > > In your actual scenario they are mixed concepts. > > Draw your target scenario with all the alternatives. You will find > many of them. Only them you will be able to point the roles each > component can and will have to play in the chosen scenario. Each of > this "play alternatives" (roles played by each component) will give > You Pros and Cons (security, scalability, velocity, etc, etc, etc) ... > > Make Your choice, try to implement it and this list will be able to > help you when problems on SR (SER or KAMAILIO flavors) configuration > come to play. > > Edson. > > Em 29/08/2011 11:46, Henrik Aagaard Sørensen escreveu: > > This is my current setup, which works: > SIP client <-> SIP server. > > As I've currently moved to another country I would like to be > able to do > the following setup: > SIP client <-> Kamailio SIP proxy <-> SIP server. > > As the SIP server only accept packets from specific IP's. > > Is it possible? > > I've managed to setup a simple Kamailio installation on my > server, which > does have a valid IP for the SIP server. > With the Kamailio installation I've also enabled NAT with RTPProxy. > > My problem is now, that the Kamailio installation doesn't work as a > simple proxy, but actually takes registrations etc. and acts as a > "full-blown" SIP server. > > It should only take SIP messages and transfer them between > clients and > server. > > Any ideas? > > > _________________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > >

1 0

Kamailio as simple SIP proxy between client and SIP server.
by Henrik Aagaard Sørensen 29 Aug '11

29 Aug '11

This is my current setup, which works: SIP client <-> SIP server. As I've currently moved to another country I would like to be able to do the following setup: SIP client <-> Kamailio SIP proxy <-> SIP server. As the SIP server only accept packets from specific IP's. Is it possible? I've managed to setup a simple Kamailio installation on my server, which does have a valid IP for the SIP server. With the Kamailio installation I've also enabled NAT with RTPProxy. My problem is now, that the Kamailio installation doesn't work as a simple proxy, but actually takes registrations etc. and acts as a "full-blown" SIP server. It should only take SIP messages and transfer them between clients and server. Any ideas?

2 1

Hunting Groups In SIP-Router
by afshin afzali 29 Aug '11

29 Aug '11

Hi Guys, I'm newbie in SER. I need to implement a hunting group service in SER. After some digging in the modules I could only find AVPOPS which seems to be able to use. I'll appreciate every advise on this. BEST, -- afshin

2 1

enum_pv_query
by Andrew Melton 28 Aug '11

28 Aug '11

I would like to construct an enum query using more than one pseudo variable. The result will be a non-e.164 string which is ok for this application. Ideally, it would be something like: enum_pv_query("$rU + $rp"); This would result in the username from the R-URI appended to the port of the R-URI. However, this doen't seem to work and I wind up with 1.2.1.2.5.5.5.2.1.2.1.$rp.e164.arpa. Instead of using the $rp pseudo variable, the string "$rp" is used to form the query. It should be: 1.2.1.2.5.5.5.2.1.2.1.*0.6.0.5*.e164.arpa. where the port 5060 is inserted between the R-URI username and the domain suffix. Is there a different or better way to use the variables within enum_pv_query to construct a query? Thanks.

4 3

Persistent dialog
by Jon Bonilla 27 Aug '11

27 Aug '11

Hi all I'm playing with the dialog module to count the calls and limit the number of calls per subscriber. After configured the proxy I've tested a call and I realize that the dialog list is not empty after the call ends. Here's a sample call log and the dialog list while the call was being stablished and 3 minutes after that: root@proxy1a:~# grep "fbrzjkplmvlauty@quenya" /var/log/kamailio.log New request - M=INVITE Dialog set mark TOTAL Proxy authentication failed New request - M=INVITE Dialog set mark TOTAL Dialog set mark user and userout to 'd5a78dafcb93' Dialog set mark type to outgoing Call to PSTN Dialog set mark peer and peerout to '4' Relaying request Request leaving server NAT-Reply - S=100 - Connecting NAT-Reply - S=488 - Not Acceptable Here Failure route for PSTN call Unset dialog mark peer and peerout '4' Filter reply code No failover routing needed for this response code New request - M=ACK While the call was being stablished or just after it was dropped (quickly) sercmd> dlg.list hash:3837:638039689 state:1 timestart:0 timeout:0 callid:fbrzjkplmvlauty@quenya from_tag:inobo to_tag: from_uri:sip:123456789@labsip.com to_uri:sip:0043987654321@labsip.com caller_contact:sip:123456789-sbhn1naovm6bc@111.111.111.164:5060;transport=udp caller_cseq:149 caller_route_set:<sip:111.111.111.135;r2=on;lr=on;ftag=inobo;nat=yes>,<sip:111.111.111.100;r2=on;lr=on;ftag=inobo;nat=yes> callee_contact: callee_cseq: callee_route_set: caller_bind_addr:udp:111.111.111.138:5062 callee_bind_addr: hash:3837:638039690 state:5 timestart:0 timeout:0 callid:fbrzjkplmvlauty@quenya from_tag:inobo to_tag: from_uri:sip:123456789@labsip.com to_uri:sip:0043987654321@labsip.com caller_contact:sip:123456789-sbhn1naovm6bc@111.111.111.164:5060;transport=udp caller_cseq:150 caller_route_set: <sip:111.111.111.135;r2=on;lr=on;ftag=inobo;nat=yes>,<sip:111.111.111.100;r2=on;lr=on;ftag=inobo;nat=yes> callee_contact: callee_cseq: callee_route_set: caller_bind_addr:udp:111.111.111.138:5062 callee_bind_addr: After a couple of minutes: hash:3837:638039689 state:1 timestart:0 timeout:0 callid:fbrzjkplmvlauty@quenya from_tag:inobo to_tag: from_uri:sip:123456789@labsip.com to_uri:sip:0043987654321@labsip.com caller_contact:sip:123456789-sbhn1naovm6bc@111.111.111.164:5060;transport=udp caller_cseq:149 caller_route_set: <sip:111.111.111.135;r2=on;lr=on;ftag=inobo;nat=yes>,<sip:111.111.111.100;r2=on;lr=on;ftag=inobo;nat=yes> callee_contact: callee_cseq: callee_route_set: caller_bind_addr:udp:111.111.111.138:5062 callee_bind_addr: sercmd> dlg.profile_get_size total Profile:total => profile:total value: count:1 sercmd> dlg.profile_get_size user Profile:user => profile:user value: count:0 sercmd> dlg.profile_get_size userout Profile:userout => profile:userout value: count:0 sercmd> dlg.profile_get_size peer Profile:peer => profile:peer value: count:0 sercmd> dlg.profile_get_size peerout Profile:peerout => profile:peerout value: count:0 Obviously, there's no such a call. But my dialog profile TOTOAL say's there's one, which is listed by sercmd. The dialog is persistent, I guess it will be destroyed after 12 hours, but I'd like to know why it's still there if I received a negative response and the ACK for it. Thanks, Jon

3 13

insert_hf in branch_route
by Gustavo Garcia Bernardo 26 Aug '11

26 Aug '11

Hi, I'm trying to add some headers in a branch_route making use of insert_hf method: insert_hf("Route: <sip:10.10.10.10:5080;lr=on>\r\n"); But the forwarded message doesn't include that new header. According to the doc, insert_hf can be used in a branch_route, Am I missing something? I'm using Kamailio 3.1. G. ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx

2 2

Re: [SR-Users] How to get mem_summary (SIGUSR1) on a worker process?
by Andrade Ricardo (CI/AFU1) 26 Aug '11

26 Aug '11

Hi, I'd just like to say that I found the answer to my own question in the documentation: With 3.x versions, you have to load ctl and cfg_rpc modules and execute: sercmd cfg.set_now_int core mem_dump_pkg <pid> http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory I did also write a patch to suppress the full memory dump, because I am only interested in the first lines of the status (total usage). This way I the server doesn't get blocked - full memory dump usually takes a VERY long time. Patch is attached and I'd be glad if something similar would be integrated in the trunk. I've introduced a mem_summary option "16" to control that behavior. Best regards, Ricardo.

2 2

issue with tm callbacks / premature cancel
by Andrew Pogrebennyk 26 Aug '11

26 Aug '11

I've got stuck today trying to implement some sequential forking scenario. When the call comes destined to the local user, I'm trying to reach it sequentially through two alternative extensions. For each extension, I'm calling t_set_fr with a timeout of 15sec. They reply only 180 and do not pick up the phone. After that the call should go to the "last resort" PSTN number. I'm calling then t_set_fr with a timeout of 30sec. But immediately after sending INVITE to PSTN gw Kamailio generates cancel and sends 408 Request Timeout to the initial INVITE. In the debug log I'm seeing this: Aug 1 17:34:43 localhost /usr/local/sbin/kamailio[28488]: DEBUG: tm [t_hooks.c:288]: DBG: trans=0xb60f543c, callback type 1048576, id 0 entered Aug 1 17:34:43 localhost /usr/local/sbin/kamailio[28488]: DEBUG: tm [t_reply.c:1634]: DEBUG: relay_reply: branch=1, save=1, relay=-1 Aug 1 17:34:43 localhost /usr/local/sbin/kamailio[28488]: DEBUG: tm [t_cancel.c:328]: DEBUG: cancel_branch: sending cancel... Does it look like something familiar to you? Kamailio version: 3.1.4 Config of tm module: modparam("tm", "fr_timer", 20000) modparam("tm", "fr_inv_timer", 120000) modparam("tm", "restart_fr_on_each_reply", 1) The rest of config is really simple, I can upload it somewhere if it's necessary. -- Sincerely, Andrew Pogrebennyk

2 5

Problems with pseudo-variables
by Rocío Ruz 26 Aug '11

26 Aug '11

Hello, I am using kamailio ims extensions and I am observing some problems with pseudo-variable handling. On the s-cscf the pseudo-variables are not 'translated' into the real values. When using xlog like this: xlog("L_ERR", " found pai: $ai time [$Tf] \n"); I am finding : tail -f /var/log/syslog | grep found invlcscs /usr/sbin/kamailio[21689]: ERROR: <script>: found pai: $ai time [$Tf] However on the p-cscf the same case is correctly treated: xlog("L_ERR", " found pai: $ai time [$Tf] \n"); invlcpcs /usr/sbin/kamailio[20540]: ERROR: <script>: found pai: sip:0188888881@open-ims.test;transport=UDP time [Fri Aug 19 14:05:44 2011] I am using 3.1 kamalio version and I have added pv and xlog modules on the s-cscf. Why is the S-CSCF unable to read pseudo-variables?Could anybody please help me? Thank you. RR.

2 1

Status of t_relay()
by Fabian Borot 25 Aug '11

25 Aug '11

Hi, If t_relay() fails, is it possible to retrieve the status/reason on a variable or something? I am using t_grep_status("code") and trying to catch each case that I am interested but I was wondering if there was a better way. One could do something like this: If (result > 400) { do action 1 } else { do action 2 } txs a lot fborot

3 5

report
by zmailer＠nic.funet.fi 25 Aug '11

25 Aug '11

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Dear user serusers(a)lists.iptel.org, mail server administrator of lists.iptel.org would like to let you know the following, We have detected that your e-mail account has been used to send a huge amount of junk email messages during this week. Probably, your computer was infected by a recent virus and now runs a hidden proxy server. [...] Content analysis details: (14.0 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.5 FSL_XM_419 Old OE version in X-Mailer only seen in 419 spam 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [87.235.117.21 listed in zen.spamhaus.org] 0.0 DEAR_EMAIL BODY: Message contains Dear email address 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.5000] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [87.235.117.21 listed in bb.barracudacentral.org] 0.3 FSL_UA FSL_UA 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS 2.6 MSOE_MID_WRONG_CASE MSOE_MID_WRONG_CASE 1.2 TO_NO_BRKTS_MSFT To: misformatted and supposed Microsoft tool 1.9 FORGED_MUA_OUTLOOK Forged mail pretending to be from MS Outlook 2.5 DOS_OE_TO_MX Delivered direct to MX with OE headers 1.5 TO_NO_BRKTS_DIRECT To: misformatted and direct-to-MX The original message was not completely plain text, and may be unsafe to open with some email clients; in particular, it may contain a virus, or confirm that your address can receive spam. If you wish to view it, it may be safer to save it to a file and open it with an editor.

1 0

Dumping Online Users stops at 2000 records
by Sebastian Damm 25 Aug '11

25 Aug '11

Hi, we have a script running on our Kamailio servers which dumps all online users with kamctl ul show, parses it and then sends it to database. (For explanation, we started doing it this way because we noticed at least with an older (open)ser release that SER would completely stop working for a couple of seconds while writing registered users to database every 5 minutes. When dumping it, we don't see any disruption.) With Kamailio 1.5 this works, with Kamailio 3.1 it works in the beginning, but when you pass the 2000 online user mark, it just stops working. I've started the Kamailio completely empty, then let users register against it. I did a "kamctl ul show" every 5 seconds, this is what the results were: Domain:: location table=512 records=82 max_slot=2 Domain:: location table=512 records=412 max_slot=4 Domain:: location table=512 records=640 max_slot=5 Domain:: location table=512 records=973 max_slot=7 Domain:: location table=512 records=1209 max_slot=8 Domain:: location table=512 records=1454 max_slot=11 Domain:: location table=512 records=1667 max_slot=12 Domain:: location table=512 records=1907 max_slot=12 500 command 'ul_dump' failed 500 command 'ul_dump' failed 500 command 'ul_dump' failed 500 command 'ul_dump' failed 500 command 'ul_dump' failed 500 command 'ul_dump' failed Is there a hard limit for dumping users through the fifo? Why? Or should we do it differently? Best Regards, Sebastian -- Dipl.-Inf. Sebastian Damm - VoIP-Engineer - damm(a)sipgate.de 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

4 5

Segmentation Fault in ACC module
by Bruno Bresciani 25 Aug '11

25 Aug '11

Hi all, ACC module is generating segmentation fault at acc_db_request function in kamailio 3.1.2 with database postgres. When module try to access the struct db_func_t acc_dbf.use_table (line 389) the child proccess that execute this operation is exited by a signal 11. Someone know why this segmentation fault is gererating? How I fix this problem? Best Regards

2 1

Kamailio 3.1.1 snmpstat help!!
by Stephen Dodge (Bistech) 25 Aug '11

25 Aug '11

Hello, I am having difficulty running the snmpstats module. We are running CentOS 5.6 and Kamailio version 3.1.1 (i386/linux). I've tried a fresh install of Kamailio, installing net-snmp from source but I can't seem to get it running - I've seen a couple of posts regarding the issue but nothing seems to work for us!! The issue I get when I start Kamailio is: 0(27523) ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/snmpstats.so>: /usr/lib/libnetsnmpagent.so.10: undefined symbol: hosts_ctl Any help would be much appreciated. Also, if anyone has any suggestions on how best to get either Solarwinds or PRTG to monitor our Kamailio servers that would be good. Thanks. Stephen Dodge Visit www.bistech.co.uk<http://www.bistech.co.uk/> for news and information ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

1 0

ACC with MySqsl problem after installing Siremis
by אורי שקד - חטיבת הנדסה ורשת - Uri Shacked 25 Aug '11

25 Aug '11

Hi, I managed to setup the ACC module with MySql. Now, after installing Siremis, the ACC table and the Missed_calls table are empty and do not populae the data from the call I make. Any ideas why? BR, Uri

4 8

Re: [SR-Users] ACC with MySqsl problem after installing Siremis
by Daniel-Constantin Mierla 25 Aug '11

25 Aug '11

On 8/25/11 1:29 PM, אורי שקד - חטיבת הנדסה ורשת - Uri Shacked wrote: > > I managed… the softphones clients were configured with “multicast” so > they where “talking behind my back” J…. I should have noticed it earlier… > > Thanks, and sorry…. > Don't worry, it happens many time to all of us to overlook the simplest things. Whenever the solution is found it is good to keep the mailing list cc-ed, so people that will read in the future the archive for similar issues, can get the resolution to this discussion. Cheers, Daniel > ------------------------------------------------------------------------ > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* Thursday, August 25, 2011 2:23 PM > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Cc:* אורי שקד- חטיבת הנדסה ורשת- Uri Shacked > *Subject:* Re: [SR-Users] ACC with MySqsl problem after installing Siremis > > Hello, > > do you get any error message in syslog from acc module. Depending on > your Linux OS, the syslog file should be /var/log/syslog or > /var/log/messages. > > Since you have acc to syslog enabled as well, do you see the ACC > records for syslog written? > > Cheers, > Daniel > > On 8/22/11 2:50 PM, אורי שקד- חטיבת הנדסה ורשת- Uri Shacked wrote: > > Hi, > > I checked and redo the changes… > > It doesn’t work. > > My config file is attached. > > I checked the DB tables and columns to see it is OK and it is….. > > Any Idea what else? > > Thanks, > > Uri > > ------------------------------------------------------------------------ > > *From:*Elena-Ramona Modroiu [mailto:ramona@asipto.com] > *Sent:* Monday, August 22, 2011 11:57 AM > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Cc:* אורי שקד- חטיבת הנדסה ורשת- Uri Shacked > *Subject:* Re: [SR-Users] ACC with MySqsl problem after installing Siremis > > Hi, > > if you installed Siremis 2.0 via the wizard, the structures of acc and > missed_calls tables were updated, so you have to change the parameters > of acc module in kamailio.cfg, like: > > http://kb.asipto.com/siremis:install20:accounting#config_file > > If you want to see the sql used to create the new acc and missed_calls > tables, see: > > http://siremis.git.sourceforge.net/git/gitweb.cgi?p=siremis/siremis;a=blob;… > > Regards, > Ramona > > > On 8/21/11 10:58 AM, אורי שקד- חטיבת הנדסה ורשת- Uri Shacked wrote: > > Hi, > > I managed to setup the ACC module with MySql. > > Now, after installing Siremis, the ACC table and the Missed_calls > table are empty and do not populae the data from the call I make. > > Any ideas why? > > BR, > > Uri > > > <http://robustus.impactia.com/AnalyticServer/redirect?cid=50f7cf7d3be2418b&m…><http://robustus.impactia.com/AnalyticServer/redirect?cid=50f7cf7d3be2418b&m…><http://robustus.impactia.com/AnalyticServer/redirect?cid=50f7cf7d3be2418b&m…><http://robustus.impactia.com/AnalyticServer/redirect?cid=50f7cf7d3be2418b&m…> > > ------------------------------------------------------------------------ > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > -- > Daniel-Constantin Mierla --http://www.asipto.com > Kamailio Advanced Training, Oct 10-13,Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Float Comparison
by Brandon Armstead 25 Aug '11

25 Aug '11

Hello, Is there anything special that needs to be done for float comparison? For example: if([5.5 >= 4.3]) .... or if(5.5 > 4.3) .... The conditional does not seem to be coming back as true like it should? This is simply an example... the precision in the actual case is different (longer decimal places). Thank you for all and any help in advance!

4 9

[regression] Message flags set in branch_route not saved into transaction
by Alex Hermann 25 Aug '11

25 Aug '11

Hello, It seems kamailio 3 does not save message flags set in branch route into the transaction. In reply_route and failure_route the flag set in branch_route is unset. In 1.4 this used to work. I would like to get that behaviour back, is this possible? Testscenario: route { setflag(0); t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); xlog("L_NOTICE", "[$rm] Request before relay: $mF"); t_relay(); } branch_route[1] { xlog("L_NOTICE", "[$rm] Branch begin: $mF"); setflag(8); xlog("L_NOTICE", "[$rm] Branch end: $mF"); } onreply_route[1] { xlog("L_NOTICE", "[$rm] Reply ($rs) begin: $mF"); setflag(4); xlog("L_NOTICE", "[$rm] Reply end: $mF"); } failure_route[1] { xlog("L_NOTICE", "[$rm] Failure ($T_reply_code): $mF"); } Log output on 3.x: [INVITE] Request begin: 00000000 [INVITE] Request before relay: 00000001 [INVITE] Branch begin: 00000001 [INVITE] Branch end: 00000101 [INVITE] Reply (100) begin: 00000001 [INVITE] Reply end: 00000011 [INVITE] Failure (408): 00000011 Log output on 1.4: [INVITE] Request begin: 00000000 [INVITE] Request before relay: 00000001 [INVITE] Branch begin: 00000001 [INVITE] Branch end: 00000101 [INVITE] Reply (100) begin: 00000101 [INVITE] Reply end: 00000111 [INVITE] Failure (408) begin: 00000111 -- Greetings, Alex Hermann

2 5

Kamailio Freezes and no calls are processed
by Omar 25 Aug '11

25 Aug '11

The only difference is we have added some AVPs variables to process, and kamailio stops processing new calls, is not regular, but seems is related to the number of calls received. No additional calls can be processed after certain time, or maybe some amount of calls. we were unable to isolate the problem, but kamailio.fifo is removed, which never happened before. No core dump created. did anybody have a similar issue. Omar

4 5

add carrierroute module
by אורי שקד - חטיבת הנדסה ורשת - Uri Shacked 25 Aug '11

25 Aug '11

Hi, I am trying to add the carrierroute module. (I installed Kamailio 3.1.4 with mysql). I add to the cfg file the ‘loadmodule “carrierroute.so”’ and all the modparams needed. When I started Kamailio I encountered errors. One of them was that the carrierroute.so could not be found…. So, can I add this module without the need to reinstall everything again? Please advice, Thanks, Uri

3 2

SIP to SMS/HTTP
by Mark Anthony C. Delfin 24 Aug '11

24 Aug '11

Hello list, I am not sure if this has been ask before but is there a way that Kamailio can sip received SIP method MESSAGES and then call a URL (Kannel URL). Thank you. Regards, Mark Anthony C. Delfin

3 2

WARNING: timer: add_timeout: 0 expire timer added
by Alex Hermann 23 Aug '11

23 Aug '11

Hello, I get this warning message when a request is relayed over tcp: WARNING: <core> [timer_funcs.h:119]: WARNING: timer: add_timeout: 0 expire timer added The warning is displayed after send_route() has run. (relevant) settings: modparam("tm", "fr_timer", 500) modparam("tm", "fr_inv_timer", 80000) modparam("tm", "wt_timer", 20000) Every other timer related setting is at its default. Just before t_relay() is called, fr timer is set with: t_set_fr($avp(ringtime) * 1000); where $avp(ringtime) = 80. An identical setup with udp does not show the warning. Where does this warning come from and how do i suppress it? -- Greetings, Alex Hermann

2 4

uac_req: pseudovariables, avps
by Jon Bonilla 23 Aug '11

23 Aug '11

Hi all I'm trying to generate a request using uac_req in a failure route when the call has been canceled. (Kamailio 3.1.3) Here's the code: failure_route[FAILURE_ROUTE_LALA] { if t_is_canceled() { $uac_req(method) = "INVITE"; $uac_req(ouri) = "sip:127.0.0.1:5090"; $uac_req(ruri) = $ru; $uac_req(furi) = $fu; $uac_req(turi) = $tu; $uac_req(hdrs) = "X-PUSH-Type: mytype\r\nX-PUSH-CLI: $avp(s:first_caller_cli)\r\nX-PUSH-DST: $rU\r\n"; uac_req_send(); route(ROUTE_STOP_RTPPROXY); exit; } ... } The problem here is that the headers X-PUSH-CLI and X-PUSH-DST have literal '$avp(s:first_caller_cli)' and '$rU' values. Here's the INVITE I send: U 2011/08/23 17:07:09.467666 127.0.0.1:5062 -> 127.0.0.1:5090 INVITE sip:ngcptest1@domain.com SIP/2.0' Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK925.85ac80a2000000000000000000000000.0' To: sip:ngcptest1@domain.com' From: sip:sipwise1@domain.com;tag=cf3b49e60a035342e3af7df009437068-9f9a' CSeq: 10 INVITE' Call-ID: 25f2b3a34ed8488c-3417(a)127.0.0.1' Content-Length: 0' User-Agent: Sipwise NGCP Proxy 2.X' X-PUSH-Type: mytype' X-PUSH-CLI: $avp(s:first_caller_cli)' X-PUSH-DST: $rU' I also tried to put the headers in a $var and in a $avp and set the value like: $var(cancel_push_request)="X-PUSH-Type: missed_call\r\nX-PUSH-CLI: $avp(s:first_caller_cli)\r\nX-PUSH-DST: $rU\r\n"; $uac_req(hdrs) = $var(cancel_push_request); If I xlog the values like this: xlog("L_INFO", "-------- Comprobando valores '$avp(s:first_caller_cli)' ------ '$rU'); The kamailio log show the correct value: INFO: <script>: -------- Comprobando valores '4312345' ------ 'ngcptest1' How could I use my stored values in the INVITE generation? cheers, Jon

2 2

max branches
by Jon Bonilla 23 Aug '11

23 Aug '11

Hi all Is there any configuration option in kamailio 3.1 to set the max number of branches? I have a system that needs to run over 30 times to failure_route and call append_brach() from there. At 11-12 iteractions I get the following errors: ERROR: tm [t_fwd.c:651]: ERROR: add_uac: maximum number of branches exceeded ERROR: tm [t_fwd.c:1528]: ERROR: t_forward_nonack: failure to add branches ERROR: tm [tm.c:1368]: ERROR: w_t_relay_to: t_relay_to failed ERROR: tm [t_reply.c:965]: ERROR: run_failure_handlers: Error in run_top_route cheers, Jon

4 4

Testing upcoming version 3.2.0
by Daniel-Constantin Mierla 23 Aug '11

23 Aug '11

Hello, for those looking to test v3.2.0 and need some instructions, for Kamailio flavour is a wiki page showing step-by-step installation: * http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-devel-from-git Practically 3.2.0 is the current devel branch. Testing SER flavour the difference is just using FLAVOUR=ser when compiling the application. Developers as well as users are encouraged to add details about the new additions in v3.2.0 to wiki page: * http://sip-router.org/wiki/features/new-in-devel Also, please add instructions for migrating from 3.1.x to 3.2.0 to the wiki page: * http://sip-router.org/wiki/install/3.1.x-to-3.2.x Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Development branch frozen for v3.2.0
by Daniel-Constantin Mierla 23 Aug '11

23 Aug '11

Hello, since several hours ago, the development of new features in master branch has been frozen, to allow proper testing for release of version 3.2.0. That means no new features should be committed to GIT master branch. Just before the release (expected in 1.5months) we will create a dedicate branch for 3.2.x series and master branch will be again open for new stuff. Meanwhile, developers can use personal branches to push code in the public space. GIT master branch will be used in this period for: - bug fixing - hammering of the new features in 3.2.0 - documentation improvements - merging of duplicated modules There are two good news for 3.2.0: - the number of the new features is very impressive. Besides the old squad, we gained many new developers lately, coming with interesting contributions, therefore I am sure you'll have lot of fun testing them - the core components (transport layers, asynchronous processing, timers, locking & memory managers, a.s.o.) were barely touched since 3.0, so this is the 3rd release in a row using the new scalable core framework. That shows the maturity and there will be no much to focus on them If you need assistance while trying some new features, feel free to email to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

RFC-5766 / RFC-6062 compliant TURN clients?
by Rafal Boni 23 Aug '11

23 Aug '11

I know this might not be totally on-topic for the list, but I figured the list participants might have experience in this domain, so I'm lobbing it out there... I've got some situations where I'd like to connect users who are oftentimes sitting on very locked-down networks (ie, corporate networks) to some SIP-based VoIP infrastructure, but the idea of opening random UDP ports on the networks in question is a no-go. As a result, I'm experimenting with TURN over TCP or TLS as a solution to enable connectivity... however, I can't find any clients that implement the latest specs (X-Lite 4 appears to speak an older version of the TURN protocol; QJsip doesn't work for me; etc. etc.). Do folks here have any suggestions for TURN-compatible clients? I care mostly about PC for the time being (ie, just a proof of concept) but a PC/Mac/Linux solution would be great too. Thanks for any insights, --rafal

2 1

E2E ACK is not logged with siptrace modul kamailio
by MÉSZÁROS Mihály 22 Aug '11

22 Aug '11

Hi, I want to see all incoming and outgoing sip messages in sip trace table, but i am missing messages. I am using TLS so i can't see really trace the conversation only this way. I am wondering if it is know limitation, or an issue: I am experiencing that E2E ACK is not logged in siptrace table. I opened a ticket with the details: http://sip-router.org/tracker/index.php?do=details&task_id=144&project=1&pa… Many Thanks, Misi

3 11

New developer - Matthew Williams
by Daniel-Constantin Mierla 22 Aug '11

22 Aug '11

Hello, a new person has joined our development squad - Matthew Williams, currently working for Flowroute, USA. He is already a developer of SEMS (SIP Express Media Server) and he going to add shortly, just in time for 3.2, some Json related extensions, including a JSON-RPC client module. His GIT commit user id is: mgw. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] decode_mime_type error
by Jijo 22 Aug '11

22 Aug '11

Hi Marius, The existing implementation is done considering the limitation of decode_mime_type but somehow missed the change in parse_content_type_hdr. I agree with you that its better to do the change in decode_mime_type as it is common for lot of other callers.. Regarding the example you pointed, According to my understanding from the code, I believe that existing implementation for parse_accept_body shall work as the mime types are predefined. So in the example you pointed for second invocation 'some value' is not a defined mime type as it not matching with predefined types. The decode_mime_type returns end pointer and parse_accept_body shall process only with the valid mime type text/plain. another example with multiple mime types.., Accept: text/html, multipart/mixed In this case parse_accept_body returns 2 mime types as both are predefined in the list. Thanks Jijo On Thu, Aug 18, 2011 at 5:58 PM, Bucur Marius <bucur_marius_ovidiu(a)yahoo.com > wrote: > Hi Jijo, > > In my opinion, decode_mime_type is broken, and parse_accept_body does not > work as expected. > For example, this is a valid accept header: > > accept: text/plain;param=",some value". > > but the parse_accept_body will return -1 because the first return value of > decode_mime_type is ",some value\"". The second invocation will think "some > value\"" is a mime type, which obviously isn't. > > I must say I never saw such an accept header, but the standard permits it. > Indeed, a quick fix would be to ignore the return code when comma is found. > This must be done for other functions that call decode_mime_type > like has_body(mime) in textops. > > This is just my opinion, but I think this solution is a bit hackish since > the convention, the "promise" that decode_mime_type should respect is to > return a non-NULL, non-end pointer ONLY when there are more mime types in > the body. And it doesn't. > A lot of functions that call it rely on this. I think it is easier/more > elegant to fix the bug in the decode_mime_type then to change the function > convention in all callers. > Also, this might work for most callers, but parse_accept_body will still be > buggy, as I explained in the example above. > > Cheers, > Marius > > ________________________________ > From: Jijo <realjijo(a)gmail.com> > To: Bucur Marius <bucur_marius_ovidiu(a)yahoo.com> > Sent: Thursday, August 18, 2011 10:48 AM > Subject: Re: [SR-Users] decode_mime_type error > > > Hi Marius, > > Right, the same function is used for Accept and Content-Type. > > I don't think there is any change required for parsing Accept. In case of > Accept the function decode_mime_type is called in a loop until it find all > the media types. On the returning from decode_mime_type(), the main function > parse_accept_xxx() checks the pointer is 'comma' or not. if its comma, then > the function decode_mime_type called again and find the remaining media > types. So i think the existing implementation for Parsing Accept is fine. > > The only change required here is not return error for comma while parsing > the Content-Type. > > Thanks > Jijo > > > On Thu, Aug 18, 2011 at 12:32 PM, Bucur Marius < > bucur_marius_ovidiu(a)yahoo.com> wrote: > > Hello Jijo, > > > >You are right, multiple mime-types are not allowed. But it seems like > decode_mime_type had the purpose of also parsing the Accept header: > > > >Accept: text/html, image/jpeg, multipart > > > >hence the search for commas :). > >The only function that uses this feature and calls decode_mime_type in a > loop is parse_accept_body. The other functions that call decode_mime_type > just fail when returning ret != end, and since multiple mime types in > content-type are not allowed the behavior is fine. > >The bad thing is the Accept header can have mime parameters too (which can > contain comma). > > > >Accept = "Accept" HCOLON [ accept-range *(COMMA accept-range) > ] > >accept-range = media-range *(SEMI accept-param) > >media-range = ( "*/*" / ( m-type SLASH "*" ) / ( m-type SLASH > m-subtype ) ) *( SEMI m-parameter ) > > > >So I think the best thing would be to change decode_mime_type so that it > does a more thorough parsing (e.g. check for parameter) hence a parameter > value can contain commas (only if quoted). > > > > > >Cheers, > > > >Marius > >________________________________ > >From: Jijo <realjijo(a)gmail.com> > >To: Bucur Marius <bucur_marius_ovidiu(a)yahoo.com>; SIP Router - Kamailio > (OpenSER) and SIP Express Router (SER) - Users Mailing List < > sr-users(a)lists.sip-router.org> > >Sent: Thursday, August 18, 2011 8:04 AM > >Subject: Re: [SR-Users] decode_mime_type error > > > > > > > >Hi Marius, > > > >Thanks for the response.. I checked the other functions, but they don't > have the check for ret !=end, but they check the pointer and if it is comma > then loop through again until it find all the media types. > > > >As per the RFC3261 multiple media-types are not supoorted in the > Content-Type. So I'm not sure why the author used comma to determine > multiple media types . Probably just followed like Route or Record-Route > headers..?? > > > >Content-Type = ( "Content-Type" / "c" ) HCOLON media-type > >media-type = m-type SLASH m-subtype *(SEMI m-parameter) > >m-type = discrete-type / composite-type > >discrete-type = "text" / "image" / "audio" / "video" > >/ "application" / extension-token > >composite-type = "message" / "multipart" / extension-token > > > >Record-Route = "Record-Route" HCOLON rec-route *(COMMA rec-route) > > > >Thanks > >Jijo > > > > > > > >On Thu, Aug 18, 2011 at 2:23 AM, Bucur Marius < > bucur_marius_ovidiu(a)yahoo.com> wrote: > > > >Hello Jijo, > >> > >>It seems like the decode_mime_type is a somehow broken. The comma is very > well allowed in boundary, as you said. The BNF specified in RFC2046 permits > it. > >>But, the decode_mime_type function ignores everything coming after comma. > More than that, it notifies the function caller that this content type has > multiple mime types. > >>I think the author of the function thought of something like: > >> > >>Content-Type: text/html, image/jpeg // very weird, though > >> > >>This is wrong, hence the comma can be in a parameter (like in your case). > >>I think the best fix, as you said is to remove that check, hence a > non-NULL return value doesn't mean anything (that the content type has > multiple mime types). > >>Another fix is to write a "good" decode_mime_type, that checks if the > comma is inside a parameter. I don't know if effort is worth it. > >> > >>One thing we need to do is to check if there are any functions in > Kamailio that call decode_mime_type and also perform this check (ret != > end). > >> > >>Cheers, > >>Marius > >> > >> > >>________________________________ > >>From: Jijo <realjijo(a)gmail.com> > >>To: sr-users(a)lists.sip-router.org; sr-dev(a)lists.sip-router.org > >>Sent: Wednesday, August 17, 2011 10:54 AM > >>Subject: [SR-Users] decode_mime_type error > >> > >> > >> > >>Hi All, > >> > >>The function parse_content_type_hdr() is failing in decode_mime_type() > when Content-Type parameter "boundary" has value comma as below. The error > message is "ERROR:parse_content_type_hdr: CONTENT_TYPE hdr contains " > >> "more then one mime type :-(! > >> > >>Content Type Header is as below, It works fine if the boundary value is > without comma. > >> > >>Content-Type: multipart/mixed;boundary=",AW" > >> > >>The parse_content_type_hdr() is failing in the following code, > >> > >> ret = decode_mime_type(msg->content_type->body.s, end , &mime); > >> if (ret==0) > >> goto error; > >> if (ret!=end) { > >> > >> LOG(L_INFO,"ERROR:parse_content_type_hdr: CONTENT_TYPE hdr > contains " > >> "more then one mime type :-(!\n"); > >> goto error ; > >> } > >> > >>I thought of fixing this issue by commenting the code for condition ret > !=end. I'm not sure why we we need that check, as mime variable has the > information to process the content. > >> > >>Please let me know if you see any impact. > >> > >>Thanks > >>Jijo > >> > >> > >> > >> > >> > >>_______________________________________________ > >>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 > >> > >>_______________________________________________ > >>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 2

min expires question
by Fabian Borot 22 Aug '11

22 Aug '11

Hello This is how my registrar module is configured, hoping that min-expires set to 30 wont allow registration attempts more often than 30 seconds, but if I configure a UAC to register every 15 seconds I see that kamailio allows it. Shouldn't it reply with "422 Session Interval Too Small" or "423 Interval Too Brief"? # ----- registrar params ----- modparam("registrar", "default_expires", 120) modparam("registrar", "min_expires", 30) 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) 131.380336 192.168.1.6 -> 192.168.1.194 SIP Request: REGISTER sip:192.168.1.194 131.381174 192.168.1.194 -> 192.168.1.6 SIP Status: 200 OK (1 bindings) 144.386014 192.168.1.6 -> 192.168.1.194 SIP Request: REGISTER sip:192.168.1.194 144.387110 192.168.1.194 -> 192.168.1.6 SIP Status: 200 OK (1 bindings) 157.395358 192.168.1.6 -> 192.168.1.194 SIP Request: REGISTER sip:192.168.1.194 157.396340 192.168.1.194 -> 192.168.1.6 SIP Status: 200 OK (1 bindings) 170.404905 192.168.1.6 -> 192.168.1.194 SIP Request: REGISTER sip:192.168.1.194 170.405771 192.168.1.194 -> 192.168.1.6 SIP Status: 200 OK (1 bindings) 183.414375 192.168.1.6 -> 192.168.1.194 SIP Request: REGISTER sip:192.168.1.194 183.415217 192.168.1.194 -> 192.168.1.6 SIP Status: 200 OK (1 bindings) txs fborot

3 4

Sip Router Issues (Topoh's module)
by Richard DEMONGEOT 22 Aug '11

22 Aug '11

Dear all, I've encountered two issues with Sip Router. Build : Kamailio 3.1.4 for Squeeze (Kamailio's debian Packages). When i'm using Topoh with Call-ID encoding, instruction dlg_bye("ALL") generate me an error on logfiles : Aug 4 17:46:26 kamtest /usr/sbin/kamailio[25267]: ERROR: topoh [th_mask.c:166]: invalid input string"dd04db6ccc7074deb9f8a1160946d331@0:0:0:0:0:0:0:0" Aug 4 17:46:26 kamtest /usr/sbin/kamailio[25267]: ERROR: topoh [th_msg.c:480]: cannot decode callid Aug 4 17:46:26 kamtest /usr/sbin/kamailio[25268]: ERROR: topoh [th_mask.c:166]: invalid input string"dd04db6ccc7074deb9f8a1160946d331@0:0:0:0:0:0:0:0" Aug 4 17:46:26 kamtest /usr/sbin/kamailio[25268]: ERROR: topoh [th_msg.c:480]: cannot decode callid Only "INCOMING LEG" is ended, and "OUTGOING LEG", with call-ID encoded continue. Have you any idea about this problem? Can we use dlg_bye("ALL") with call-id encoding? Best regards,

2 5

10 Years SER Conference, Berlin - full booked
by Daniel-Constantin Mierla 22 Aug '11

22 Aug '11

Hello, the 10 years SER conference in Berlin is full booked, meaning that new registrations will be accepted only if someone that registered previously will cancel his/her participation. Since we just discovered such a situation, in the case you did register and you haven't received a confirmation email (not the one automatically generated at registration time, but a second one), please reply to me or write to registration(a)lists.sip-router.org telling the date when you did the registration and we will try to solve it. Looking forward to meeting many of you soon in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Freezing development for v3.2.0 today
by Daniel-Constantin Mierla 22 Aug '11

22 Aug '11

Hello, just a short reminder that we will close the development of new features for version 3.2.0 at the end of today, GMT. Then the focus should be on testing (e.g., bug fixing) and enhancements to documentation. If you have something to commit, do it today, or if it is impossible for you to make it in the GIT repo today, let us know and we will see what can be done. But don't forget, 3.3.0 will come as well in the no-so-far future. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

masking carriers
by Anders 20 Aug '11

20 Aug '11

Hi, Background: Kamailio 1.5.x + rtpproxy I am not too crazy about telling our customers where we send our calls. But if they take a look at the sip dialog as it is right now, they can see our suppliers/carriers IP addresses. What can we do to mask this, so that the last piece of information stops at our IPs? We are running rtpproxy for the media - if that makes any difference. Someone told me I cannot mask anything without moving to Kamailio 3.x - is that the case? Thanks!! //Anders

2 2

Question about encryption with Kamailio
by margot.basa 19 Aug '11

19 Aug '11

Hello all, I would like to use Kamailio to encrypt contents of SIP messages (using SIP TLS) between 2 endpoints, i.e.: - To use 5061 port instead of 5060 port, - To use sips uri instead of sip uri... For example, T1 and T2 communicates with "Server A" like that: 1) T1 and T2 send REGISTER to "Server A" 2) T1 and T2 received 200 OK from "Server A" ... 3) "Server A" sends an INVITE message to T1 and T2 ... 4) RTP flow between T1 and T2 (this should not be encrypted) ... 5) "Server A" sends a BYE request to T1 and T2 ... All those exchanges are made on Transport layer TCP or UDP on port 5060. T1 and T2 are not able to support TLS but "Server A" needs to receive/send messages in SIP TLS. I would like to insert Kamailio between T1 and "Server A", T2 and "Server B" in order to encrypt contents of SIP messages. I have some questions about that: - I think Kamailio can do that but I am not sure, can you confirm that to me please? - Can I use Kamailio as it is to do that? - Do I have to add a "Route" header in requests in order that requests between T1 and "Server A" go through Kamailio or - Does Kamailio is able to intercept SIP packets automatically (with a certain configuration)? - Do you know difference between Freeswitch and Kamailio? (because I have seen that Freeswitch can do what I need: see Figure4: http://wiki.freeswitch.org/wiki/SIP_TLS) Thank you very much for your input. Regards Une messagerie gratuite, garantie à vie et des services en plus, ça vous tente ? Je crée ma boîte mail www.laposte.net

2 2

New developer: Alexandr Dubovikov
by Daniel-Constantin Mierla 19 Aug '11

19 Aug '11

Hello, a new person has now developer access to GIT repository - Alexandr Dubovikov, contributor in the past with patches, is going to work now on extensions related to Homer project - turning the SIP server in a SIP capturing server. His developer id is adubovikov. You can read more about Homer project at: * http://code.google.com/p/homer/ * http://www.kamailio.org/events/2011-Cluecon/AD-Homerv2.pdf Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 2

new functionality: CDR-based accounting
by Timo Reimann 19 Aug '11

19 Aug '11

Hi all, as announced quite a while ago, I finally checked in code that allows to produce CDRs (Call Data Records) directly from SIP-Router and generate logs accordingly. The main code portion resides in modules_k/acc and provides a switch to enable basic CDR generation including start time, end time, and duration. Analogous to the existing logging approach, you may define an extra parameter covering to-be-included dialog pseudo-variables that must be assigned in the configuration script. The new code will take care of transforming the basic and custom CDR fields into CDR logs at the end of a dialog. Speaking of dialogs: The implementation relies heavily on the dialog module. It takes advantage of dialog variables introduces by Carsten Bock and adds a few more features. Most notably, we had to change the dialog callback signature to provide both request and response messages. Having only one of them proved to be insufficient in certain cases; for instance, a locally generated 408 returned a FAKED_REPLY, thus rendering it impossible to access dialog variables through the PV framework. Other modules using dialog callbacks have been updated along the commit, third-party modules outside the repository will need to do so too (and think about whether using the request or response is the Right Thing to do). Due to the changes brought to the dialog module, I pushed the new acc and dialog code into a separate branch called treimann/acc-cdr. Feel free to give it a try by consulting the updated documentation and suggesting (or, if it's good enough, implementing :) ) improvements. A Kamailio 1.5 backport of the code has been in usage for quite some time with us, so generally there shouldn't be any major logical flaws. SIP-Router certainly needs more testing, however, so I'd be glad for any feedback. My plan is to merge the code into master branch prior to the 3.2 feature freeze, unless significant objections arise. Finally, big-time credits go to my co-worker Sven Knoblich who is the main contributor of the code. He's been working on this stuff for the past few months and dreams in CDRs by now. Cheers, --Timo

5 7

LCR Modules Comparison
by Spencer Thomason 19 Aug '11

19 Aug '11

Hello all, Can anyone shed some light into the differences between the available LCR modules? We have migrated our config from OpenSIPS where we used the drouting module to Kamailio where we are using the lcr module. Our ruleset is not crazy huge, about 100k entries. Are there compelling reasons to choose one module over the other? Is there one that has the most active development behind it and is a "standard" for the platform or is there one that will be deprecated in the foreseeable future? Thanks, Spencer

4 3

Re: [SR-Users] decode_mime_type error
by Bucur Marius 19 Aug '11

19 Aug '11

Hi Jijo, In my opinion, decode_mime_type is broken, and parse_accept_body does not work as expected. For example, this is a valid accept header: accept: text/plain;param=",some value". but the parse_accept_body will return -1 because the first return value of decode_mime_type is ",some value\"". The second invocation will think "some value\"" is a mime type, which obviously isn't. I must say I never saw such an accept header, but the standard permits it. Indeed, a quick fix would be to ignore the return code when comma is found. This must be done for other functions that call decode_mime_type like has_body(mime) in textops. This is just my opinion, but I think this solution is a bit hackish since the convention, the "promise" that decode_mime_type should respect is to return a non-NULL, non-end pointer ONLY when there are more mime types in the body. And it doesn't. A lot of functions that call it rely on this. I think it is easier/more elegant to fix the bug in the decode_mime_type then to change the function convention in all callers. Also, this might work for most callers, but parse_accept_body will still be buggy, as I explained in the example above. Cheers, Marius ________________________________ From: Jijo <realjijo(a)gmail.com> To: Bucur Marius <bucur_marius_ovidiu(a)yahoo.com> Sent: Thursday, August 18, 2011 10:48 AM Subject: Re: [SR-Users] decode_mime_type error Hi Marius, Right, the same function is used for Accept and Content-Type. I don't think there is any change required for parsing Accept. In case of Accept the function decode_mime_type is called in a loop until it find all the media types. On the returning from decode_mime_type(), the main function parse_accept_xxx() checks the pointer is 'comma' or not. if its comma, then the function decode_mime_type called again and find the remaining media types. So i think the existing implementation for Parsing Accept is fine. The only change required here is not return error for comma while parsing the Content-Type. Thanks Jijo On Thu, Aug 18, 2011 at 12:32 PM, Bucur Marius <bucur_marius_ovidiu(a)yahoo.com> wrote: Hello Jijo, > >You are right, multiple mime-types are not allowed. But it seems like decode_mime_type had the purpose of also parsing the Accept header: > >Accept: text/html, image/jpeg, multipart > >hence the search for commas :). >The only function that uses this feature and calls decode_mime_type in a loop is parse_accept_body. The other functions that call decode_mime_type just fail when returning ret != end, and since multiple mime types in content-type are not allowed the behavior is fine. >The bad thing is the Accept header can have mime parameters too (which can contain comma). > >Accept = "Accept" HCOLON [ accept-range *(COMMA accept-range) ] >accept-range = media-range *(SEMI accept-param) >media-range = ( "*/*" / ( m-type SLASH "*" ) / ( m-type SLASH m-subtype ) ) *( SEMI m-parameter ) > >So I think the best thing would be to change decode_mime_type so that it does a more thorough parsing (e.g. check for parameter) hence a parameter value can contain commas (only if quoted). > > >Cheers, > >Marius >________________________________ >From: Jijo <realjijo(a)gmail.com> >To: Bucur Marius <bucur_marius_ovidiu(a)yahoo.com>; SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> >Sent: Thursday, August 18, 2011 8:04 AM >Subject: Re: [SR-Users] decode_mime_type error > > > >Hi Marius, > >Thanks for the response.. I checked the other functions, but they don't have the check for ret !=end, but they check the pointer and if it is comma then loop through again until it find all the media types. > >As per the RFC3261 multiple media-types are not supoorted in the Content-Type. So I'm not sure why the author used comma to determine multiple media types . Probably just followed like Route or Record-Route headers..?? > >Content-Type = ( "Content-Type" / "c" ) HCOLON media-type >media-type = m-type SLASH m-subtype *(SEMI m-parameter) >m-type = discrete-type / composite-type >discrete-type = "text" / "image" / "audio" / "video" >/ "application" / extension-token >composite-type = "message" / "multipart" / extension-token > >Record-Route = "Record-Route" HCOLON rec-route *(COMMA rec-route) > >Thanks >Jijo > > > >On Thu, Aug 18, 2011 at 2:23 AM, Bucur Marius <bucur_marius_ovidiu(a)yahoo.com> wrote: > >Hello Jijo, >> >>It seems like the decode_mime_type is a somehow broken. The comma is very well allowed in boundary, as you said. The BNF specified in RFC2046 permits it. >>But, the decode_mime_type function ignores everything coming after comma. More than that, it notifies the function caller that this content type has multiple mime types. >>I think the author of the function thought of something like: >> >>Content-Type: text/html, image/jpeg // very weird, though >> >>This is wrong, hence the comma can be in a parameter (like in your case). >>I think the best fix, as you said is to remove that check, hence a non-NULL return value doesn't mean anything (that the content type has multiple mime types). >>Another fix is to write a "good" decode_mime_type, that checks if the comma is inside a parameter. I don't know if effort is worth it. >> >>One thing we need to do is to check if there are any functions in Kamailio that call decode_mime_type and also perform this check (ret != end). >> >>Cheers, >>Marius >> >> >>________________________________ >>From: Jijo <realjijo(a)gmail.com> >>To: sr-users(a)lists.sip-router.org; sr-dev(a)lists.sip-router.org >>Sent: Wednesday, August 17, 2011 10:54 AM >>Subject: [SR-Users] decode_mime_type error >> >> >> >>Hi All, >> >>The function parse_content_type_hdr() is failing in decode_mime_type() when Content-Type parameter "boundary" has value comma as below. The error message is "ERROR:parse_content_type_hdr: CONTENT_TYPE hdr contains " >> "more then one mime type :-(! >> >>Content Type Header is as below, It works fine if the boundary value is without comma. >> >>Content-Type: multipart/mixed;boundary=",AW" >> >>The parse_content_type_hdr() is failing in the following code, >> >> ret = decode_mime_type(msg->content_type->body.s, end , &mime); >> if (ret==0) >> goto error; >> if (ret!=end) { >> >> LOG(L_INFO,"ERROR:parse_content_type_hdr: CONTENT_TYPE hdr contains " >> "more then one mime type :-(!\n"); >> goto error ; >> } >> >>I thought of fixing this issue by commenting the code for condition ret !=end. I'm not sure why we we need that check, as mime variable has the information to process the content. >> >>Please let me know if you see any impact. >> >>Thanks >>Jijo >> >> >> >> >> >>_______________________________________________ >>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 >> >>_______________________________________________ >>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 1

The not so simple Kamailio Load Balancing of Asterisk
by Nick Khamis 19 Aug '11

19 Aug '11

Hello Everyone, I saw the simple load balancing of asterisk using x-SER documentation here "http://www.kamailio.org/dokuwiki/doku.php/asterisk:load-balancing-and-ha". Are there any up-tp-date not so simple documentation for setting up HA and LB asterisk instances using x-SER. Thanks in Advance, Nick Khamis Toronto Hydro Telecom

5 7

decode_mime_type error
by Jijo 18 Aug '11

18 Aug '11

Hi All, The function parse_content_type_hdr() is failing in decode_mime_type() when Content-Type parameter "boundary" has value comma as below. The error message is "ERROR:parse_content_type_hdr: CONTENT_TYPE hdr contains " "more then one mime type :-(! Content Type Header is as below, It works fine if the boundary value is without comma. Content-Type: multipart/mixed;boundary="*,*AW" The parse_content_type_hdr() is failing in the following code, ret = decode_mime_type(msg->content_type->body.s, end , &mime); if (ret==0) goto error; if (ret!=end) { LOG(L_INFO,"ERROR:parse_content_type_hdr: CONTENT_TYPE hdr contains " "more then one mime type :-(!\n"); goto error ; } I thought of fixing this issue by commenting the code for condition ret !=end. I'm not sure why we we need that check, as mime variable has the information to process the content. Please let me know if you see any impact. Thanks Jijo

2 2

Re: [SR-Users] mtree question
by Javier Gallart 18 Aug '11

18 Aug '11

Hi Juha based on my experience with the mtree module: -mt_match() will match the longest prefix in the tree, in your case 00358 -mt_ignore_duplicates makes sense when loading the data in the DB to the memory. There could be duplicate entries in the db and with that directive you tell kamailio how to handle that situation -I don't know about mt_tree_type...(never needed to test it) Hope it helps Regards Javi > ------------------------------ > > Message: 2 > Date: Fri, 12 Aug 2011 10:10:29 +0300 > From: Juha Heinanen <jh(a)tutpro.com> > Subject: [SR-Users] mtree question > To: sr-users(a)lists.sip-router.org > Message-ID: <20036.53733.20652.518606(a)sip.test.fi> > Content-Type: text/plain; charset=us-ascii > > after reading mtree readme, it is not clear to me if mt_match() matches > to longest or any matching prefix in the three. for example, if i have > tree with prefixes 00 and 00358 and string to be matched is > 0035892345670, will mt_match() set pv to value associated with 00 or > 00358? also, if mt_ignore_duplicates is set, will 00358 be considered > as duplicate of 00 or just prefixes that are exactly same. finally, > what are possible mt_tree_type values and what types they stand for? > > -- juha > > > > ------------------------------ > > _______________________________________________ >

3 5

lookup for specified user
by Efelin Novak 17 Aug '11

17 Aug '11

Hi folks, I discussed this issue in previous email ( [SR-Users Modifying $rU after lookup() ) but the question is a bit different now. I would like to change uri like lookup() function does it, but for the specific user. Problem is I want to have one registration (X) for several numbers. When this number is called I have to perform lookup() for this one common registration (X) and send this call there. To do this I changed $rU just before lookup() for X, and after lookup() I changed it back for the original number (both changes were done using variables $rU). However when there are two registrations (X1, X2), only X1 can be changed for the original number and the second one (X2) is sent to user without modification. I also found function reg_fetch_contacts(domain, uri, profile) but in documentation there is $ulc variable mentioned. This variable is not documented. Is there any way how to perform lookup for specific user? Thanks for the answer. Regards, Efelin

3 5

siptrace loosing an ACK
by Timo Klecker 17 Aug '11

17 Aug '11

Hello We are using the siptrace module to save the sip-messages to our database. All the messages are logged except one message. After INVITE, 100 trying and 200 OK we loose the outbound ACK which is forwarded by Kamailio statelessly in t_funcs.c. Is there any way to get a siptrace of this message, too? I've made a picture of the sip-messages and the lost ACK. Kind regards Timo Klecker

3 3

Error message
by Jon Bonilla 16 Aug '11

16 Aug '11

I have an error message in kamailio's log repeated many times. I tried to google it but no results (I think it's forst time I see no results in a google search) /usr/sbin/kamailio[3769]: ERROR: ctl [binrpc_run.c:942]: ERROR: binrpc: rpc_printf: binrpc_addstr failed: buffer too small (overflow) (-2) What does it mean? Any ideas? thanks, Jon

2 1

openSIPS vs Kamailio vs SIP-Router
by Nick Khamis 15 Aug '11

15 Aug '11

Hello Everyone, I can only imagine how many times this question has come up since post 2008. Please forgive my reoccurring of the issue. We are looking to provide carrier grade sip services to our clients world wide. What we need is a lightweight, robust and scalable solution that will allow us to terminate sip calls to our different carriers. Performance, and high throughput are factors very important to my employer. Features such as caller authentication, database back-end, load balancing, and interoperability with asterisk are things we are interested in, as was offered using OpenSER. With three+ open source proxy servers available on the net puts us in a situation where we have more solutions to choose from, at the same time wish the features from one were available in the other, and vice versa. With this in mind, we will have to fall back to other factors such as the most reliable, proven and active projects. As mentioned, we would choose functional stability over endless features that we will never use and that add to the projects fingerprint... I understand that all three projects are forks from OpenSER, people would naturally like to know what differentiates one from the other. Thanks in Advance, Nick Khamis Toronto Hydro Telecom

4 5

Kamailio presentations at Cluecon 2011
by Daniel-Constantin Mierla 15 Aug '11

15 Aug '11

Hello, I uploaded my slides presented at Cluecon 2011, last week in Chicago. In the first part the focus was on what has been happening in the last year around the project, on the second part showing some example of using it to secure VoIP networks. The PDF is available at: * http://asipto.com/u/42 Alexandr Dubovikov presented the recently announced Homer project which turns Kamailio in a scalable capturing system for SIP traffic, the slides for this talk are available at: * http://www.kamailio.org/events/2011-Cluecon/AD-Homerv2.pdf Stefan Wintermeyer presented Gemeinschaft 4 project, which is targeting to offer a very secure out-of-the-box IP PBX system, using Kamailio as one of the major components. The slides will be available in the near future (I will update the lists) -- he will present the same topic at 10 years SER conference in Berlin: * http://sip-router.org/10-years-ser/ Cheers, Daniel

3 2

10 Years SER Conference - update
by Daniel-Constantin Mierla 15 Aug '11

15 Aug '11

Hello, just wanted to let you know that only few seats are still available for the 10 Years SER Conference, on Sep 2, 2011, in Berlin, Germany -- more details at: * http://sip-router.org/10-years-ser/ If you plan to attend, do not wait any longer and register as soon as possible (form at: http://sip-router.org/registration-form/). The host FhG Fokus Institute has a medium size conference room, with no option to expand -- I expect the registration to be closed very soon. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

freezing for major release 3.2.0
by Daniel-Constantin Mierla 15 Aug '11

15 Aug '11

Hello, based on the last devel IRC meeting, the next major release should happen sometime by end of September, therefore we should freeze the development and go into testing phase about one month before. I propose Monday, August 22, the day to freeze development for version 3.2.0. By then, I hope every developer will be able to push to master branch what he/she wants to have in 3.2.0. Write here to mailing lists if you thing about alternatives. The very good side of 3.2.0 is that we didn't need to touch much the core and main modules, meaning we have a solid tested foundation, further testing should be focused on the new modules and the other enhancements. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 1

Wrong formatting of Request URI
by Mino Haluz 15 Aug '11

15 Aug '11

Hi, my kamailio is sometimes receiving ruri from remote clients which I assume are not right formatted. For ex.: sip:user@kamailio=3Buser=3Dphone Is there any mechanism which could transform this into something readable for kamailio or do I have to simply cut off everything after the host name of the RURI ? Mino

2 1

pua_xmpp questions
by laura testi 13 Aug '11

13 Aug '11

Hi all, we are trying to setup a kamailio presence server in SIP/SIMPLE domain to interwork with XMPP domains including the GTalk (see the config below). The architecture is like this: - SIP/SIMPLE server: kamailio3.14 with full configuration with XCAP integration - XMPP Gateway: another kamailio server + ejabberd server - component mode - xmpp moduleL: outbound proxy to the Kamailio SIP/SIMPLE server - pua_xmpp: no outbound proxy? - XMPP server: ejabberd The users from XMPP domain can add the SIP account and see the presence of the SIP users, but not vice versa. The chat from both directions works fine. in the Presence Server, we have configured the kamailio with XCAP integration. We have the following doubts: - does pua_xmpp/pua modules need the xcap integration for the presence integration with xmpp domains? because to use xcap for presence authorization rules, it needs the sip clients support the xcap - does pua_xmpp/pua support xcap? otherwise how to works? - in case of multiple SIP/SIMPLE presence server, how we can configure the server_address of pua_xmpp and presence parameter in xmpp gw? Can you help us to clarify the doubts please? Many thanks in advanced! Best Regards, Laura PS: following are the main configuration of the xmpp GW: --------------------------------------------------------------------------------------------- ... modparam("xmpp", "domain_separator", "*") modparam("xmpp", "backend", "component") modparam("xmpp", "gateway_domain", "<mygwdomain>") modparam("xmpp", "xmpp_domain", "<mygwdomain>") modparam("xmpp", "xmpp_host", "127.0.0.1") modparam("xmpp", "xmpp_password", "secret") modparam("xmpp", "outbound_proxy", "<my oubound proxy uri>") modparam("pua", "outbound_proxy", "<my outbound proxy uri>") modparam("pua", "update_period", 60) modparam("pua", "default_expires", 1200) modparam("presence", "server_address", "<my presence server uri>") modparam("pua_xmpp", "server_address", "<my presence server uri>") route{ route(REQINIT); t_check_trans(); if (uri =~ "sip:.+@.*<myxmppdomain>") { route(PRESENCE); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } route(CHAT); } xlog("L_INFO", "*** xmpp: unhandled message type\n"); t_reply("503", "Service unavailable"); return; } ... route[CHAT] { if(!is_method("MESSAGE")) return; if (!t_newtran()) { sl_reply_error(); exit; } xlog("L_INFO", "*** xmpp-handled MESSAGE message."); if ($cT=~"^text/plain") { if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } } else { xlog("L_INFO", "*** xmpp-handled MESSAGE, ignoring not text messages"); t_reply("200", "Accepted"); } t_release(); exit; } route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE|NOTIFY")) return; # create new transaction to catch retransmissions if (!t_newtran()) { sl_reply_error(); exit; } if( is_method("NOTIFY")) { xlog("L_INFO", "*** xmpp-handled NOTIFY message."); if(pua_xmpp_notify()) t_reply("200", "OK"); t_release(); exit; } else if( is_method("SUBSCRIBE")) { xlog("L_INFO", "*** xmpp-handled SUBSCRIBE message.\n"); handle_subscribe(); if($(hdr(Event))== "presence") { pua_xmpp_req_winfo("$ruri", "$hdr(Expires)"); } t_release(); exit; } else if(is_method("PUBLISH")) { handle_publish(); t_release(); exit; } } ...

2 8

subscriber table reload
by Fabian Borot 12 Aug '11

12 Aug '11

Hello When one creates a user using the cli command like this: "kamctl add 200 abc", it is my understanding that the information goes to the subscriber table and into memory. Is that right? But if one populates the subscriber table manually [adding a record on the table w/o the /kamctlcli command], do we need to "load" those accounts into memory before a client tries to register or does kamailio looks it up when the registration request comes in at that time? What we are trying to do is to bulk load subscribers on the subscribers table instead of using the "kamctl add" command. For example, with LCR parameters we modify the tables manually and then do a "lcr_reload" and all the changes will be in memory after the reload. thank you in advance fborot

4 5

mtree question
by Juha Heinanen 12 Aug '11

12 Aug '11

after reading mtree readme, it is not clear to me if mt_match() matches to longest or any matching prefix in the three. for example, if i have tree with prefixes 00 and 00358 and string to be matched is 0035892345670, will mt_match() set pv to value associated with 00 or 00358? also, if mt_ignore_duplicates is set, will 00358 be considered as duplicate of 00 or just prefixes that are exactly same. finally, what are possible mt_tree_type values and what types they stand for? -- juha

1 0

kamailio - proxy/relay
by Sunny 11 Aug '11

11 Aug '11

Hi list, I'm trying to configure kamailio to work as a OBP (Outbound Proxy). Basically KAMAILIO receive requests from UAs, after that it acts like a client and send those requests to the IP-PBX. On the same way the responses are forwarded from IP-PBX to the UAs. [UA] --> [KAMAILIO] --> [IP-PBX ] [UA] <-- [KAMAILIO] <-- [IP-PBX ] Following possible call flow: [UA ] -> [KAMAILIO] SIP/SDP Request: INVITE sip:12345678909@[KAMAILIO], with session description [KAMAILIO] -> [UA ] SIP Status: 100 Trying [KAMAILIO] -> [IP-PBX ] SIP/SDP Request: INVITE sip:12345678909@[IP-PBX ], with session description [IP-PBX ] -> [KAMAILIO] SIP Status: 100 Trying [IP-PBX ] -> [KAMAILIO] SIP Status: 407 Proxy Authentication Required [KAMAILIO] -> [IP-PBX ] SIP Request: ACK sip:12345678909@[IP-PBX ] [KAMAILIO] -> [UA ] SIP Status: 407 Proxy Authentication Required [UA ] -> [KAMAILIO] SIP Request: ACK sip:12345678909@[KAMAILIO] [UA ] -> [KAMAILIO] SIP/SDP Request: INVITE sip:12345678909@[KAMAILIO], with session description [KAMAILIO] -> [UA ] SIP Status: 100 Trying [KAMAILIO] -> [IP-PBX ] SIP/SDP Request: INVITE sip:12345678909@[IP-PBX ], with session description [IP-PBX ] -> [KAMAILIO] SIP Status: 100 Trying [IP-PBX ] -> [KAMAILIO] SIP/SDP Status: 183 Session Progress, with session description [KAMAILIO] -> [UA ] SIP/SDP Status: 183 Session Progress, with session description [IP-PBX ] -> [KAMAILIO] SIP/SDP Status: 180 Ringing, with session description [KAMAILIO] -> [UA ] SIP/SDP Status: 180 Ringing, with session description (...) [UA ] -> [KAMAILIO] SIP Request: CANCEL sip:12345678909@[KAMAILIO] [KAMAILIO] -> [UA ] SIP Status: 200 OK [KAMAILIO] -> [IP-PBX ] SIP Request: CANCEL sip:12345678909@[IP-PBX ] [IP-PBX ] -> [KAMAILIO] SIP Status: 200 OK [IP-PBX ] -> [KAMAILIO] SIP Status: 487 Request Terminated [KAMAILIO] -> [IP-PBX ] SIP Request: ACK sip:12345678909@[IP-PBX ] [KAMAILIO] -> [UA ] SIP Status: 487 Request Terminated [UA ] -> [KAMAILIO] SIP Request: ACK sip:12345678909@[KAMAILIO] Wondering if anyone could help to make this configuration work? Thanks, Sunny

3 4

time for acc records
by Daniel-Constantin Mierla 11 Aug '11

11 Aug '11

Hello, it seems that is some parts of the world it is required to have call duration with two decimals precision. By default acc module does the timestamp in seconds. Thinking of adding an option to save the milliseconds, I see two option: - a new column to store the milliseconds part as integer - a new column to store the seconds.milliseconds as double For those using various billing engines, what would be easier to digest when rating the CDRs? There is already the option of doing db extra column to store $TV(Sn), but this value is stored in a string (varchar) format, giving a little more troubles to convert to numeric format before being able to compute duration, so it might just worth doing the C coding to store in right data type. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 5

Re: [SR-Users] Default kamailio installation, syntax errors with Defined Values
by Gertjan Wolzak 11 Aug '11

11 Aug '11

Sorry, my bad.... Missing dependencies Regards, Gertjan Wolzak

1 0

Default kamailio installation, syntax errors with Defined Values
by Gertjan Wolzak 11 Aug '11

11 Aug '11

Hello, Just been installing Kamailio, version 3.1.x via Git, but when I use the default kamailio.cfg, Kamailio will not start and the debug gives the following errors. 0(25876) : <core> [cfg.y:3330]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 285, column 29-35: syntax error 0(25876) : <core> [cfg.y:3330]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 285, column 29-35: Invalid arguments 0(25876) : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 285, column 36: ERROR: bad config file (4 errors) When I check the line, column it is the following line of the acc params: modparam("acc", "log_flag", FLT_ACC) I assume the variable FLT_ACC is not set or read correctly. But as it is the default kamailio.cfg that seems strange to me. The defined values are: ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - 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 Does anyone has an idea why te values are not loaded into the variables.? With regards, Gertjan Wolzak

1 0

Using Kamailio to convert TCP to TLS
by aur.hoffmann＠free.fr 10 Aug '11

10 Aug '11

Hello, I would like to use Kamailio as a TCP <=> TLS convertor ? Do you think it's possible ? I have developed a software (which simulate a SIP phone) which can use UDP or TCP (not TLS). I need a SIP proxy to convert TCP to TLS. For example: UA => TCP OPTION (5060) => Kamailio => TLS OPTION (5061) => OXE UA <= TCP 200 OK (5060) <= Kamailio <= TLS 200 OK (5061) <= OXE Do you think Kamailio can do that ? Thanks in advance, Regards

3 2

$hu is not documented
by Juha Heinanen 10 Aug '11

10 Aug '11

xcap_server examples refer to pseudo variable $hu, which is not documented on pseudo variables wiki page: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel -- juha

2 2

Cancel issue
by Carl Wagner 09 Aug '11

09 Aug '11

Hi, I am having an issue getting Kamailio to properly cancel a call. I initiate the call with an INVITE from Asterisk and then hang up. According to rfc3665 the call flow should be: 3.8. Unsuccessful No Answer Alice Proxy 1 Proxy 2 Bob | | | | | INVITE F1 | | | |--------------->| INVITE F2 | | | 100 F3 |--------------->| INVITE F4 | |<---------------| 100 F5 |--------------->| | |<---------------| | | | | 180 F6 | | | 180 F7 |<---------------| | 180 F8 |<---------------| | |<---------------| | | | CANCEL F9 | | | |--------------->| | | | 200 F10 | | | |<---------------| CANCEL F11 | | | |--------------->| | | | 200 F12 | | | |<---------------| | | | | CANCEL F13 | | | |--------------->| | | | 200 F14 | | | |<---------------| | | | 487 F15 | | | |<---------------| | | | ACK F16 | | | 487 F17 |--------------->| | |<---------------| | | | ACK F18 | | | 487 F19 |--------------->| | |<---------------| | | | ACK F20 | | | |--------------->| | | In my case I only have one proxy - the Kamailio box, and the call is coming from Asterisk instead of Alice. The is the call flow that I am seeing: Asterisk Kamailio SIP_phone | | | | INVITE | | |--------------->| INVITE | | 100 |--------------->| |<---------------| 100 | | |<---------------| | | | | | 180 | | 180 |<---------------| |<---------------| | | CANCEL | | |--------------->| | | | CANCEL | | |--------------->| | 200 Canceling | | |<---------------| | | | 200 | | |<---------------| | | | | | | | | 487 | | |<---------------| | | ACK | | 487 |--------------->| |<---------------| | | ACK | | |--------------->| | | CANCEL | | |--------------->| | | 200 Canceling | | |<---------------| | | CANCEL | | |--------------->| | | 200 Canceling | | |<---------------| | It looks like the problem is that that "200 Canceling" has a To: tag and it shouldn't. ====================================== # cancel from Asterisk: U 2011/05/24 15:18:56.123714 3.3.3.3:5060 -> 2.2.2.2:5060 CANCEL sip:SIP_13034445555@2.2.2.2 SIP/2.0. Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK51c9602a;rport. From: "+13032223333" <sip:+13032223333@3.3.3.3>;tag=as4c415d87. To: <sip:SIP_13034445555@2.2.2.2>. Call-ID: 605b8de40a05924177085d71154bda77(a)3.3.3.3. CSeq: 102 CANCEL. User-Agent: VerbalWorld. Max-Forwards: 70. Content-Length: 0. # 200 canceling from Kamailio to asterisk: U 2011/05/24 15:18:56.127349 2.2.2.2:5060 -> 3.3.3.3:5060 SIP/2.0 200 canceling. Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK51c9602a;rport=5060. From: "+13032223333" <sip:+13032223333@3.3.3.3>;tag=as4c415d87. To: <sip:SIP_13034445555@2.2.2.2>;tag=4ed8ab9f3ed29a77613bb14b3431ff43-2f34. Call-ID: 605b8de40a05924177085d71154bda77(a)3.3.3.3. CSeq: 102 CANCEL. Server: kamailio (3.1.3 (i386/linux)). Content-Length: 0. ====================================== Why is kamialio adding a To: tag? Portion of kamailio.cfg ===================== ... route { ... route(WITHINDLG); ### only initial requests (no To: tag) from here on if (is_method("CANCEL")) { if (t_check_trans()) { xlog("L_INFO", "MAIN: Recieved Cancel, t_check_trans is true\n"); ... t_relay(); exit; } ... Thanks! Carl

3 9

Modifying $rU after lookup()
by Efelin Novak 09 Aug '11

09 Aug '11

Hi Folks, I would like to ask how can I change the $rU (user part of uri) in multiple appended branches? In my scenario I do lookup("location") and after this I try to modify an username using the $rU variable. This works fine until there are several records in the location table for the given user (registered from different locations). In this case only one branch is changed and unchanged uri is sent to an another location. Is there a way to change it in all branches together? Thanks for the answer. Regards, Efelin

5 7

Re: [SR-Users] Using SR for Number Translation & Number Portabilty Applications
by Henning Westerholt 09 Aug '11

09 Aug '11

On Monday 08 August 2011, Yaron Nachum wrote: > I have seen the match_mode but I haven't seen how is it possible to set it > up per domain. > > I would appreciate if you could explain how. Hi Yaron, please always CC to the mailling list. Its not possible to set the match_mode per domain, its a global parameter. With regards to the domains, this was another option you could use, e.g. use domain 0 for calls without the 'D' prefix, and then domain D for calls with 'D' prefix. Best regards, Henning

1 0

Re: [SR-Users] Using SR for Number Translation & Number Portabilty Applications
by Henning Westerholt 08 Aug '11

08 Aug '11

> Yaron wrote: > Thanks for the quick response. I have already played with the sqlops - > this is an obvious way to do it. I will further investigate cr and sqlops > and decide which is best for my purposes. Hello Yaron, > I liked the idea of having several instances of SIP-ROUTERs which are > independent of each other- I understand that cr answers it. Indeed, this is one advantage of using an in-memory cache. But you probably need some more DB access on the servers for things like authentication. > I have another question - I was trying to add alpha-numeric character > into the cr prefix and it seems it is not supported. We are using 'D' prefixes > in our network in order to route between different elements in the network. > Is there a way to support it? Maybe another routing module? The cr module support also a extended matching mode, but it will be probably not really feasible with millions of routes: http://kamailio.org/docs/modules/stable/modules/carrierroute.html#id2651893 Another thing you could do is use another "domain" for the D prefix, e.g. one for the default routing, and then another one for the D case. The module support matching for the domain as well. You could maybe also use flags for matching. Other possible modules for PSTN routing are lcr and mtree. Regards, Henning

1 0

Using SR for Number Translation & Number Portabilty Applications
by Yaron Nachum 08 Aug '11

08 Aug '11

Hi Everyone, I have downloaded SR r 3.1.4 and have been playing around with it. I am considering implementing Number Translation services (1800...) and Number Portability Services on it. We have around 20K entries on the Number Translation Service and around 3M entries on the Number portability Service. The volume is around 100 caps. I have started working with the carrierroute module - it seems perfect for the Number translation Service, but I am not sure about the Number portability. I have several questions: 1. Is it possible to use this module with several millions of entries? 2. What is the impact of loading the table? Is it possible to do it during traffic hours? I have read about PDB module, but I haven't understood how to setup the PDB server. I would appreciate for any guidance. Would you recommend using the PDB module for the Number Portability or the carrierroute? Thanks.

2 1

suggested number of TCP / UDP workers for version 3.1.x
by Francesco Castellano 08 Aug '11

08 Aug '11

Dear sirs, searching on the web and in old ML threads about the suggested number of TCP and UDP workers, I just found: - for TCP children (i.e., TCP receivers, isn't it?) "As a rule of thumb, (maximum simultaneous connections)/2000 should be OK" (from doc/tcp_tunning.txt) - In general for workers (from a past thread by Henning Westerholt): "In my experience the number of children is not that important for performance, you may just choose the default size of 8." In the last post there was, moreover, a suggestion that the number of children should be chosen with respect to RAM and CPU cores (that makes sense to me). The only thing I can think about the rule of thumb above comes from the possibility that a number of processes may be unavailable in blocking operations. So my question is: - with asynchronous TCP/TLS is it safe to assume that more children than CPU cores are more or less useless? - are there other blocking operations for which the number of UDP/TCP children should be incremented above the CPU cores number? Thank you! Best regards, Francesco Castellano

2 2

Kamailio Redundant Systems
by HM Kias 08 Aug '11

08 Aug '11

Hi, Anyone tried pacemaker, heartbeat or OpenAIS for building redundant kamailio systems? Please share your experiences. Also auggest a better solution. Regards, -- HM Kias 91-9443467600

2 1

Homer Project
by Alexandr Dubovikov 05 Aug '11

05 Aug '11

Hello sr-users, I would like to present a new project called "Homer". What is Homer ? Homer is a capture solution for SIP logging. How does it work? Homer has 4 components: 1. Capture Agent 2. Capture Node 4. Capture DB 3. Web GUI. The capture agent takes your local SIP traffic and sends it to the defined capture node in HEP protocol. If you are user of FreeSWITCH or Kamailio, use build-in capture agent. In other case, you can use an external generic daemon, but only for SIP/UDP. The capture node is based on kamailio, can receive SIP traffic encapsulated in IP proto 4 (IPIP) or in HEP procotol and store it into MySQL DB. To gain access to the capture DB, download a simple webinterface (PHP based). After it you are able to do search, generate PCAP file, callflow, look at the details of SIP messages etc. Homer works in our company and easy collect ~5 M packets per hour. The patches for kamailio and some other information can be found here: http://homer.googlecode.com Beta testers are welcome :-) I'll help everybody to setup/configure Homer :-) With best regards, Alexandr

3 3

$du gets set automatically before branch_route
by Alex Hermann 05 Aug '11

05 Aug '11

Hello, while trying to upgrade from 1.5 to 3.1 i noticed that between t_relay() and branch_route, $du gets automatically set to $ru if it has not been set before. This is a change from 1.5 i could not find documentation for, so it might have been unintened. Can someone enlighten me on the purpose of this change and if there is a way to determine if $du is (was) unset other than comparing $ru to $du, wich seems a bit inefficient? Is it even guaranteed that $du gets set to $ru in every possible scenario? -- Alex Hermann

3 8

udp vs. tcp receiver processes
by Juha Heinanen 05 Aug '11

05 Aug '11

i have 8 children specified in default file and in cfg file i have: listen=192.98.101.10:5060; listen=udp:127.0.0.1:5070; listen=udp:127.0.0.1:5070; listen=tcp:127.0.0.1:5070; listen=udp:127.0.0.1:5080; listen=tcp:127.0.0.1:5080; listen=tcp:127.0.0.1:6060; that results in 8 tcp receiver processes but 3 x 8 udp receiver processes: # sip-proxy_ctl ps 30720 attendant 30721 udp receiver child=0 sock=192.98.101.10:5060 30722 udp receiver child=1 sock=192.98.101.10:5060 30723 udp receiver child=2 sock=192.98.101.10:5060 30724 udp receiver child=3 sock=192.98.101.10:5060 30725 udp receiver child=4 sock=192.98.101.10:5060 30726 udp receiver child=5 sock=192.98.101.10:5060 30727 udp receiver child=6 sock=192.98.101.10:5060 30728 udp receiver child=7 sock=192.98.101.10:5060 30729 udp receiver child=0 sock=127.0.0.1:5070 30730 udp receiver child=1 sock=127.0.0.1:5070 30731 udp receiver child=2 sock=127.0.0.1:5070 30732 udp receiver child=3 sock=127.0.0.1:5070 30733 udp receiver child=4 sock=127.0.0.1:5070 30734 udp receiver child=5 sock=127.0.0.1:5070 30735 udp receiver child=6 sock=127.0.0.1:5070 30736 udp receiver child=7 sock=127.0.0.1:5070 30738 udp receiver child=0 sock=127.0.0.1:5080 30740 udp receiver child=1 sock=127.0.0.1:5080 30752 udp receiver child=2 sock=127.0.0.1:5080 30754 udp receiver child=3 sock=127.0.0.1:5080 30755 udp receiver child=4 sock=127.0.0.1:5080 30756 udp receiver child=5 sock=127.0.0.1:5080 30758 udp receiver child=6 sock=127.0.0.1:5080 30760 udp receiver child=7 sock=127.0.0.1:5080 30761 slow timer 30763 timer 30765 ctl handler 30803 TIMER NH 30805 TIMER RT 30806 tcp receiver child=0 30808 tcp receiver child=1 30810 tcp receiver child=2 30812 tcp receiver child=3 30813 tcp receiver child=4 30844 tcp receiver child=5 30846 tcp receiver child=6 30847 tcp receiver child=7 30849 tcp main process why cannot one udp receiver process handle all listening addresses like tcp processes seem to be able to do? the above seems like waste of resources. -- juha

3 4

Kamailio on a server with different network cards
by Rocío Ruz 05 Aug '11

05 Aug '11

Hello, My server has two different network cards with different IPs. I want kamailio to use both; the network card to be used depends on the destination. My problem is that I have to add an IP in the configuration file kamailio.cfg: listen=109.239.50.67 Will Kamailio listen only to packets sent to this IP? Can I add the two IPs corresponding to my two network cards here? Thank you RR

3 5

480 Temporarily Unavailable. in kamailio 3.1.4
by hala alramli 04 Aug '11

04 Aug '11

i installed kamailio and freeswitch according to http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc i installed also rtpproxy . i managed to create 101,102,103 and all of them registered , i am not able to make calls between them, 480 temporarily unavailable always the error . i cant find in the configuration file what cause this error. this is the track. i will be thankful for any help . Hala interface: eth0 (192.168.1.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) match: 101 U 2011/08/04 13:16:57.351872 192.168.1.175:5060 -> 192.168.1.233:5060 INVITE sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-6eabf9b3. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 101 INVITE. Max-Forwards: 70. Contact: <sip:101@192.168.1.175:5060>. Expires: 240. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 444. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. Content-Type: application/sdp. . v=0. o=- 986176 986176 IN IP4 192.168.1.175. s=-. c=IN IP4 192.168.1.175. t=0 0. m=audio 16450 RTP/AVP 8 0 2 4 18 96 97 98 100 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=fmtp:100 192-193. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 2011/08/04 13:16:57.352037 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-6eabf9b3. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7d11. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 101 INVITE. Proxy-Authenticate: Digest realm="192.168.1.233", nonce="TjqA1U46f6liNM8a6LX5HBJ2wXEBtOIX". Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. . U 2011/08/04 13:16:57.370381 192.168.1.175:5060 -> 192.168.1.233:5060 ACK sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-6eabf9b3. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7d11. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 101 ACK. Max-Forwards: 70. Contact: <sip:101@192.168.1.175:5060>. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 0. . U 2011/08/04 13:16:57.380065 192.168.1.175:5060 -> 192.168.1.233:5060 INVITE sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. Max-Forwards: 70. Proxy-Authorization: Digest username="101",realm="192.168.1.233",nonce="TjqA1U46f6liNM8a6LX5HBJ2wXEBtOIX",uri="sip:102@192.168.1.233",algorithm=MD5,response="a019828b7a7334539ffa6b668dfcfada". Contact: <sip:101@192.168.1.175:5060>. Expires: 240. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 444. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. Content-Type: application/sdp. . v=0. o=- 986176 986176 IN IP4 192.168.1.175. s=-. c=IN IP4 192.168.1.175. t=0 0. m=audio 16450 RTP/AVP 8 0 2 4 18 96 97 98 100 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=fmtp:100 192-193. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 2011/08/04 13:16:57.380329 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. . U 2011/08/04 13:16:57.488511 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=NDrNpDZ1g0jZe. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-5d37bc9 2011-08-03 09-05-01 -0500. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. . U 2011/08/04 13:16:57.488511 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=NDrNpDZ1g0jZe. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-5d37bc9 2011-08-03 09-05-01 -0500. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. Remote-Party-ID: "kb-102" <sip:kb-102@192.168.1.233>;party=calling;privacy=off;screen=no. . U 2011/08/04 13:16:57.502085 192.168.1.175:5060 -> 192.168.1.233:5060 ACK sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=NDrNpDZ1g0jZe. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 ACK. Max-Forwards: 70. Proxy-Authorization: Digest username="101",realm="192.168.1.233",nonce="TjqA1U46f6liNM8a6LX5HBJ2wXEBtOIX",uri="sip:102@192.168.1.233",algorithm=MD5,response="a019828b7a7334539ffa6b668dfcfada". Contact: <sip:101@192.168.1.175:5060>. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 0. .

2 1

How to reset t_on_failure/branch?
by Iñaki Baz Castillo 04 Aug '11

04 Aug '11

Hi, I remember that in Kamailio 1.5.X t_on_branch and t_on_failure still remain after rouning failure_route[XX], and I "think" there was a way to reset them. In 3.X it seems that t_on_branch and t_on_failure must be set again in failure_route to take effect in next t_relay() call, am I right? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

AVPs missing in onreply_route
by Sebastian Damm 04 Aug '11

04 Aug '11

Hi, I'm still fighting problems in our Kamailio 1.5 to 3.1 migration. Right now I'm stuck with a strange behavior of AVPs in my dialplan. I set a lot of AVPs for accounting in the route directive, and I just inserted an avp_print() statement right before t_relay() is called, all values are set. Then I call avp_print() as the first statement of onreply_route(), and it doesn't print out anything. I call it again at the end of onreply_route, and the only AVP printed out is the one I set directly there. In 1.5 there was an option in the tm module to control AVP behavior in replies, but that doesn't exist anymore, I even found a mailing list post from Daniel where he explained that all AVPs I set in route are also available in onreply_route. Am I doing something wrong? Is there still a secret option that I accidentally turned on? What else could make kamailio forget all AVPs after t_relay? I tried both version 3.1.3 and 3.1.4. Any help appreciated. Best regards, Sebastian

3 9

ERROR : when create database in kamailio 3.1.0
by hala alramli 04 Aug '11

04 Aug '11

hello i a trying to instal kamailio and free switch according to the tutorial in http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc but when i try to create the database and the extra tables this error apear could you help me? thanks Hala root@noga1:~# nano /usr/local/etc/kamailio/kamctlrc root@noga1:~# /usr/local/sbin/kamdbctl create MySQL password for root: database engine 'mysql' loaded INFO: test server charset INFO: creating database kamailio ... Creating core table: standard Creating core table: acc Creating core table: lcr Creating core table: domain Creating core table: group Creating core table: permissions Creating core table: registrar Creating core table: usrloc Creating core table: msilo Creating core table: alias_db Creating core table: uri_db Creating core table: speeddial Creating core table: avpops Creating core table: auth_db Creating core table: pdt Creating core table: dialog Creating core table: dispatcher Creating core table: dialplan INFO: Core Kamailio tables succesfully created. Install presence related tables? (y/n): y INFO: creating presence tables into kamailio ... INFO: Presence tables succesfully created. Install tables for imc cpl siptrace domainpolicy carrierroute userblacklist htable purple uac pipelimit? (y/n): y INFO: creating extra tables into kamailio ... Creating extra table: imc Creating extra table: cpl Creating extra table: siptrace Creating extra table: domainpolicy Creating extra table: carrierroute Creating extra table: userblacklist Creating extra table: htable Creating extra table: purple Creating extra table: uac Creating extra table: pipelimit /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql: line 225: /usr/local/share/kamailio//mysql/pipelimit-create.sql: No such file or directory ERROR: Creating extra tables failed!

2 2

Re: [SR-Users] Chicago, ClueCon next week
by Daniel-Constantin Mierla 03 Aug '11

03 Aug '11

Hello, for questions about kamailio (openser), the best place to address them is still the mailing list for users. There are many people there that might be able to answer quickly. Messages sent personally to me for technical questions about kamailio (openser) are discarded, unless they were a result of a discussion on the mailing list where I asked for sensitive data. Cheers, Daniel On 8/3/11 1:47 AM, ifeanyi okoye wrote: > hi Daniel..i would like to discuss some of the issues am having with Openser/Kamailio > -----Original Message----- > Date: Tuesday, August 02, 2011 5:15:39 pm > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> > Cc: ifeanyi okoye<ifeanyi_okoye(a)yahoo.com> > From: "Daniel-Constantin Mierla"<miconda(a)gmail.com> > Subject: Re: [SR-Users] Chicago, ClueCon next week > > Hello Ify, > > On 8/2/11 11:15 PM, ifeanyi okoye wrote: >> Hello Daniel, >> >> My name is Ify and I'm a research voip engineer.I would be attending >> the conference. Please can I reach you via telephone. >> > the phone might not be the most convenient at that time, so the best is > to ask the organizers to point you to me, unless you can spot me based > on the pictures you can find on the web. I will have the presentation on > the 2nd day, but I will be around on 1st day as well. > > Looking forward to meeting you, > Daniel > >> Regards, >> Ify >> >> --- On *Tue, 8/2/11, Daniel-Constantin Mierla /<miconda(a)gmail.com>/* >> wrote: >> >> >> From: Daniel-Constantin Mierla<miconda(a)gmail.com> >> Subject: [SR-Users] Chicago, ClueCon next week >> To: "kamailio users"<sr-users(a)lists.sip-router.org>, >> business(a)lists.kamailio.org >> Date: Tuesday, August 2, 2011, 1:59 AM >> >> Hello, >> >> I'll be in Chicago few days next week for ClueCon conference. If >> you are in the area, but no -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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