- sr-users - kamailio.org

add_contact_alias
by Klaus Darilion 17 Jan '10

17 Jan '10

Hi Juha! Does the alias functions also support IPv6 addresses? regards klaus

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[Kamailio-Users] [kamailio-users] Implement kamailio sever failover
by RAJNIKANT VANZA 16 Jan '10

16 Jan '10

Hi, I want to implement kamailio server failover. so, anybody know about how to configure kamailio server failover? e.g. 2 kamailio severs run and when any one server fail then other server take over thats resources. -- Best Regards, Rajnikant Vanza

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[Kamailio-Users] How to enable all sip_trace messages?
by Dinesh gautam 16 Jan '10

16 Jan '10

Dear all, I have enabled the sip_trace module and able to log the messages ACK, INVITE, REGISTER , SUBSCRIBE. I want to log other messages like RING, OK, BYE etc. Please help me out how can I enable theses messages. -- With Regards, Dinesh Kumar Gautam

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Re: [Serusers] configure SER to work as a gateway
by "Martínez García, Rebeca" 15 Jan '10

15 Jan '10

Authentication in my upstream server is a MUST in my case and I cannot disable it, sorry. In fact I would be going against the standards for what I am configuring if I disabled authentication. I only wanted SER for the special case of internal calls inside a private LAN, but I am realising it is not as simple as I though it would be. In any case, thanks for the support. Regards, Rebeca Martínez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: vie 15/01/2010 12:06 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Not that I know of... Anyway, we had the same problem....but solved it by disabling authentication on the upstream server. After all...you are now in a "trusted zone"....you have successfully "auth challenged" on the SER...so no need to do it twice....SER "IS" the Registrar, thats what it is designed to be....an architecture with 2 Registrars is not good. If you can turn off auth on the upstream server then this is best. When you t_replicate Registers SER just ignores the 200ok back from the upstream server....after all it is just sending the register as a courtesy gesture. Rupert ________________________________ From: "Martínez García, Rebeca" [mailto:remartinez@indra.es] Sent: 15 January 2010 10:54 To: Organ,RJE,Rupert,DRK R Cc: serusers(a)lists.iptel.org; "Huertas García, Victor" Subject: RE: [Serusers] configure SER to work as a gateway Hi all, Rupert you were right, SER does not permit to save the location based on the reg of a different upstream server; which means that I have to make "save_contacts("location") when SER receives the REGISTER. However, as it then sends directly to my client a 200 OK, when the client receives the "401 Unauthorized" error from my server, it does not authenticate with it because it has received a previous 200 OK (from SER...). So, here is my question: Is it possible to make SER not send a 200 OK when saving the location of the user? (I am only asking because I do not know). Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: lun 11/01/2010 12:37 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Rebeca, i am not an expert, but if I had to guess...then i would guess it would be because there is no contacts in the 200 to a Register. In our ser.cfg we save("location"); after testing if (method=="REGISTER") { but I can see that you are making registration and saving of info into location dependent on successful reg on a different upstream server. I do not think that is possible because you need to make the decision to save into location based on the reg that SER has received...and not based on the response of an upstream element....but if I am wrong I would be interested in the answer... Rupert ________________________________ From: "Martínez García, Rebeca" [mailto:remartinez@indra.es] Sent: 11 January 2010 11:21 To: Organ,RJE,Rupert,DRK R Cc: serusers(a)lists.iptel.org; "Huertas García, Victor" Subject: RE: [Serusers] configure SER to work as a gateway Hi all, Thanks Rupert for your answer, I realise that my explanation was not very clear but the scenario I am working into is a little bit complex. Finally, I have decided to change my configuration and make the client sends its REGISTER message to SER and then it will forward the same message to my Server. However, the Server carries out the authentication process (I can not change it), so SER has to save the location contact after the client is authenticated in the Server. My function for the REGISTER message in SER is the following one: route[REGISTER] { t_newtran(); # send the register to the proxy rewritehostport("open-ims.test:4060"); t_on_reply("REGISTER_reply"); #t_on_failure("REGISTER_failure"); if (!t_relay()) { sl_send_reply("500","Error forwarding to P-CSCF"); break; } } onreply_route[REGISTER_reply] { if (t_check_status("401")){ log(-1, "A response from the P-CSCF!"); break; } if (t_check_status("200")){ log(-1, "User registered in the IMS Core!"); save_contacts("location"); break; } } But SER does not start. It shows this error: cng@cng:~$ sudo /etc/init.d/ser start -P /var/run/ser/ser.pid Starting ser: ser 0(2545) parse error (147,19-20): Command cannot be used in the block I have checked where is the error and it complains about the "save_contacts("location"))", but I do not know why I can not use it inside the onreply_route. Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: vie 08/01/2010 15:03 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Hi Rebeca, I think I understand what you are trying to do, albeit you describe it in a strange way. SER is designed to be a Registrar / SIP Proxy. It is not designed to proxy REGISTER messages. However I had to interface it to a SIP Application Server that wanted to see the REGISTER messages. However SER was my REGISTRAR and performed my security / authentication. Therefore I filter on the REGISTER messages using if (method=="REGISTER") { after doing the auth challenge in the ser.cfg, and after doing an eNum lookup and some user agent field checks I simply t_replicate the REGISTER. i.e. t_replicate("rebeca.node.es.com", "5060"); or if you do not use DNS t_replicate("10.1.x.y", "5060"); Note this replication does not include the auth challenge....but it satisfys my SIP app server functionality nicely, Also you can overwrite the useragent field if you wish... e.g. if (subst_uri("/;userAgent=rupert")) { subst('/^User-Agent:.*$/User-Agent: Rebeca/ig'); Good luck Rupert ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of "Martínez García, Rebeca" Sent: 08 January 2010 11:52 To: serusers(a)lists.iptel.org Cc: "Huertas García, Victor" Subject: [Serusers] configure SER to work as a gateway Hello all, Solved the installation problem I had, now I am trying to configure SER in a certain way. My configuration is the following one: Client ------------------------- SER ---------------------------- Server/Proxy 192.168.x.x 192.168.x.y 10.1.x.x 10.1.x.y Unlike the normal case where SER is used as the Registrar, my client (in a private LAN) must register in a Server/Proxy and SER has to act as a SIP gateway. However, at the same time, SER has to register locally this client (for internal calls). So, put in other words, SER should forward the REGISTER message (directed from the client to the Server/Proxy) and save locally the location of that client. Once registered the user, SER should act as a transparent proxy. But in the practice, it does nothing if I put that the client registers in the Server/Proxy. The messages reach SER but it does not forward them, nor it registers locally the clients. So I tried configuring the client to register in SER instead of the Server/Proxy and then SER started to work. Is a listening problem? What I have to include to the ser.cfg so SER handles also messages which are not directed to it but to the Server/Proxy? I hope somebody could help me. Thanks in advance, Rebeca Martinez

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Re: [Serusers] configure SER to work as a gateway
by "Martínez García, Rebeca" 15 Jan '10

15 Jan '10

Hi all, Rupert you were right, SER does not permit to save the location based on the reg of a different upstream server; which means that I have to make "save_contacts("location") when SER receives the REGISTER. However, as it then sends directly to my client a 200 OK, when the client receives the "401 Unauthorized" error from my server, it does not authenticate with it because it has received a previous 200 OK (from SER...). So, here is my question: Is it possible to make SER not send a 200 OK when saving the location of the user? (I am only asking because I do not know). Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: lun 11/01/2010 12:37 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Rebeca, i am not an expert, but if I had to guess...then i would guess it would be because there is no contacts in the 200 to a Register. In our ser.cfg we save("location"); after testing if (method=="REGISTER") { but I can see that you are making registration and saving of info into location dependent on successful reg on a different upstream server. I do not think that is possible because you need to make the decision to save into location based on the reg that SER has received...and not based on the response of an upstream element....but if I am wrong I would be interested in the answer... Rupert ________________________________ From: "Martínez García, Rebeca" [mailto:remartinez@indra.es] Sent: 11 January 2010 11:21 To: Organ,RJE,Rupert,DRK R Cc: serusers(a)lists.iptel.org; "Huertas García, Victor" Subject: RE: [Serusers] configure SER to work as a gateway Hi all, Thanks Rupert for your answer, I realise that my explanation was not very clear but the scenario I am working into is a little bit complex. Finally, I have decided to change my configuration and make the client sends its REGISTER message to SER and then it will forward the same message to my Server. However, the Server carries out the authentication process (I can not change it), so SER has to save the location contact after the client is authenticated in the Server. My function for the REGISTER message in SER is the following one: route[REGISTER] { t_newtran(); # send the register to the proxy rewritehostport("open-ims.test:4060"); t_on_reply("REGISTER_reply"); #t_on_failure("REGISTER_failure"); if (!t_relay()) { sl_send_reply("500","Error forwarding to P-CSCF"); break; } } onreply_route[REGISTER_reply] { if (t_check_status("401")){ log(-1, "A response from the P-CSCF!"); break; } if (t_check_status("200")){ log(-1, "User registered in the IMS Core!"); save_contacts("location"); break; } } But SER does not start. It shows this error: cng@cng:~$ sudo /etc/init.d/ser start -P /var/run/ser/ser.pid Starting ser: ser 0(2545) parse error (147,19-20): Command cannot be used in the block I have checked where is the error and it complains about the "save_contacts("location"))", but I do not know why I can not use it inside the onreply_route. Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: vie 08/01/2010 15:03 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Hi Rebeca, I think I understand what you are trying to do, albeit you describe it in a strange way. SER is designed to be a Registrar / SIP Proxy. It is not designed to proxy REGISTER messages. However I had to interface it to a SIP Application Server that wanted to see the REGISTER messages. However SER was my REGISTRAR and performed my security / authentication. Therefore I filter on the REGISTER messages using if (method=="REGISTER") { after doing the auth challenge in the ser.cfg, and after doing an eNum lookup and some user agent field checks I simply t_replicate the REGISTER. i.e. t_replicate("rebeca.node.es.com", "5060"); or if you do not use DNS t_replicate("10.1.x.y", "5060"); Note this replication does not include the auth challenge....but it satisfys my SIP app server functionality nicely, Also you can overwrite the useragent field if you wish... e.g. if (subst_uri("/;userAgent=rupert")) { subst('/^User-Agent:.*$/User-Agent: Rebeca/ig'); Good luck Rupert ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of "Martínez García, Rebeca" Sent: 08 January 2010 11:52 To: serusers(a)lists.iptel.org Cc: "Huertas García, Victor" Subject: [Serusers] configure SER to work as a gateway Hello all, Solved the installation problem I had, now I am trying to configure SER in a certain way. My configuration is the following one: Client ------------------------- SER ---------------------------- Server/Proxy 192.168.x.x 192.168.x.y 10.1.x.x 10.1.x.y Unlike the normal case where SER is used as the Registrar, my client (in a private LAN) must register in a Server/Proxy and SER has to act as a SIP gateway. However, at the same time, SER has to register locally this client (for internal calls). So, put in other words, SER should forward the REGISTER message (directed from the client to the Server/Proxy) and save locally the location of that client. Once registered the user, SER should act as a transparent proxy. But in the practice, it does nothing if I put that the client registers in the Server/Proxy. The messages reach SER but it does not forward them, nor it registers locally the clients. So I tried configuring the client to register in SER instead of the Server/Proxy and then SER started to work. Is a listening problem? What I have to include to the ser.cfg so SER handles also messages which are not directed to it but to the Server/Proxy? I hope somebody could help me. Thanks in advance, Rebeca Martinez

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[Serusers] Max-Forwards missing in SER CANCEL during forking
by Andres 15 Jan '10

15 Jan '10

Hi, We encountered an issue in 0.9.6 when SER forks requests because there are several endpoints registered with the same account (ie office and home phones). When one of the endpoints answers the call, SER sends a CANCEL to the other endpoints so they can stop further call processing. It turns our that this SER generated CANCEL does not contain the "Max-Forwards" header which apparently violates RFC3261. Here is an example: U SER IP:5060 -> 10.10.1.50:36679 CANCEL sip:3055551234@10.10.1.50:36679;rinstance=C926AD943D258534F5F62913216C50ADC9D895D2; transport=udp SIP/2.0..Via: SIP/2.0/UDP 10.10.1.21:5065;branch=z9hG4bK1c1e.32eeb702.1.. From: "9545551234" <sip:9545551234@10.10.1.20:5066>;tag=as7d1243c9.. Call-ID: 4385c66230a310e714f020926aa2028c(a)10.10.1.20.. To: <sip:3055551234@X.X.X.X:5065>..CSeq: 102 CANCEL.. User-Agent: Sip EXpress router(0.9.6 (i386/linux)).. Content-Length: 0.... Is there any fix or workaround for this? How can we get that "Max-Forwards" header in the Message if it is internally generated by SER and no something that flows through the config script. Thanks, Andres http://www.telesip.net

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[Kamailio-Users] mk_proxy: could not resolve hostname
by kévin l'heveder 14 Jan '10

14 Jan '10

Hi, I'm using kamailio V3.0 In the kamailio.cfg when I use the "send" function with an IP inside (as send("127.0.01:51001"), I see the following error : mk_proxy: could not resolve hostname: "127.0.01:51001" At first, kamailio was sending request to the DNS server which define in resolv.conf to resolve "127.0.01:51001" The DNS could not resolve this name then the error occured So I removed this defined DNS and now no request are sending but still have the same error. I tried to set dns and use_dns_search_list parameters to no, but that's change nothing Anybody have any ideas? Regards, Koon

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[Serusers] ser scalability
by Sabatella, Michael 14 Jan '10

14 Jan '10

I just wanted to run this by the group. I did a complete overhaul on one of my ser environments and upgraded the physical server and all software around my sip proxies. I am also maintaining a complete front end for provisioning and billing that I also moved(PHP/mysql) I am running this on a Virtual machine (vmware esxi). If I need more proxies I can clone the VM and build in new service/capacity pretty fast. I am running 0.9.6 on Redhat ES 5. I have 4 cores dedicated to this server running Xeon 2.93GHz processers and 4 GIG memory. I am running the 32 bit OS. I hardly see the processor tick up to 1%. The machine I am testing has about 300 registered endpoints in the proxy db. My questions are: 1. What is the scalability of ser processing calls? 2. Are there any limits on the number of users that can be registered at one time. Is the registration memory based and am I severely limited with 4GB of memory? 3. Is there anyone that has 1000s of registration to the proxy and if so I would like to hear how you are scaling it. What type of server are you using? 4. What are some good metrics to look at on the sip proxy to gauge its performance? 5. Is there a way to capture calls per minute? Is anyone doing this? I have call accounting set up. Thanks, Michael ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

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[Kamailio-Users] 64kbit, quad/dual core support
by mbsip 14 Jan '10

14 Jan '10

Hi all, I am wondering if kamailio do support 64-bit OS systems and dual/quad core cpu platforms? The question is raised because i am about to prepare hardware specification for kamailio server. Thanks in advance, Maciej.

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Re: [Kamailio-Users] about kamailio with sctp
by Daniel-Constantin Mierla 14 Jan '10

14 Jan '10

Hi Bruce, On 1/14/10 12:54 PM, Bruce McAlister wrote: > Hi Daniel, > > >> you may ask this one on sr-dev(a)lists.sip-router.org since the devel of >> SCTP use to be more active there. >> >> > Thanks, I will ask there too. > > >> When I implemented in in old kamailio, it was no option for me than >> using kamailio-to-kamailio SCTP channel. >> >> > When you did the testing between the two kamailio servers, how did you > enforce SCTP as the transport? > > Does t_relay offer a way to specify UDP/TCP/SCTP as the transport to relay > the messages between servers? > > if you set dst uri with transport parameter then t_relay() will use it, e.g.: $du = "sip:10.0.0.1;transport=sctp"; t_relay(); Or you can use t_relay_to_sctp() or t_relay_to() with appropriate parameters: http://kamailio.org/docs/modules/3.0.x/modules/tm.html Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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Re: [Kamailio-Users] about kamailio with sctp
by Daniel-Constantin Mierla 14 Jan '10

14 Jan '10

Hello, On 1/14/10 2:25 AM, Bruce McAlister wrote: > Hi Klaus, > > >> I do not know. For kamailio 1.5 I would just grep for SCTP in the >> sources to get the answer. >> For kamailio 3.0 (based on sip-router with a new (I guess much better) >> SCTP implementation there is probably another method needed. A cite from >> another email from Andrei: >> >> >>> Yes, make cfg SCTP=1 and then make all will compile with SCTP support. >>> I'm currently pondering whether or not I should make the SCTP support >>> automatic (if the needed *.h files are installed compile it >>> > automatically). > > I am compiling Kamailio 3.0.0 on Solaris 10 which does have SCTP support > inherent in the OS. I did a 'grep -i sctp Makefile*' and saw that there were > checks for SCTP in the makefiles. > > I ended up building Kamailio 3.0.0 with the following command line: > > make prefix=/opt/kamailio \ > SCTP=1 \ > CC_EXTRA_OPTS=-I/usr/gnu/include \ > group_include="standard postgres presence" \ > include_modules="snmpstats perl tls" \ > all > > The build went through successfully and I can see that sctp_server.o has > been linked into the kamailio binary when the build is running. > > Does anyone know of any tool that I can use to test Kamailio using the SCTP > protocol? > you may ask this one on sr-dev(a)lists.sip-router.org since the devel of SCTP use to be more active there. When I implemented in in old kamailio, it was no option for me than using kamailio-to-kamailio SCTP channel. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] running Qjsimple with kamailio 1.5 + tls
by mustafa samara 14 Jan '10

14 Jan '10

Hello i am trying Qjsimple 0.6.3 with Kamailio 1.5 with tls support, i make the user register successfully but when i am trying to invite another user i got the following error: " transport_srtp Failed generating random key: unspecified failure pjsua_call.c Error initializing media channel: Unknown error 259800 [status=259800] Dialog destroyed QjSimple: Error calling buddy: Unknown error 259800 [status=259800] " the sip setting as the following: protocol: tcp srtp : mandatory srtp requirements: tls also i got the same error if the sip setting as the following: protocol: tls srtp : mandatory srtp requirements: tls or: protocol: tls srtp : mandatory srtp requirements: sips any body help me please? -- Eng.Mustafa Al-Samara

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Re: [Kamailio-Users] about kamailio with sctp
by Klaus Darilion 14 Jan '10

14 Jan '10

Bruce McAlister schrieb: > Hi All, > > I am wanting to play with Kamailio 3.0.0 and SCTP as well. I just want to > verify that I am building it in to Kamailio properly. When I build Kamailio > do I need to have USE_SCTP=1 as an option on the make command line to enable > SCTP support? > > I can see there are updates in the ChangeLog for SCTP but I cannot find how > to enable support for it in the INSTALL/README-MODULES documentation. > > Can someone point me in the right direction please. I do not know. For kamailio 1.5 I would just grep for SCTP in the sources to get the answer. For kamailio 3.0 (based on sip-router with a new (I guess much better) SCTP implementation there is probably another method needed. A cite from another email from Andrei: > Yes, make cfg SCTP=1 and then make all will compile with SCTP support. > I'm currently pondering whether or not I should make the SCTP support > automatic (if the needed *.h files are installed compile it automatically). regards klaus > > Thanks > Bruce > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org > [mailto:users-bounces@lists.kamailio.org] On Behalf Of Daniel-Constantin > Mierla > Sent: 11 January 2010 09:57 > To: mustafa samara > Cc: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] about kamailio with sctp > > Hello, > > On 1/11/10 1:37 AM, mustafa samara wrote: >> Hello >> i do not know how to try sctp, imake kmailio 1.5 with sctp=1 >> could you help me? > if you want to play with sctp then I recommend to use 3.0.0 (to be > released today). It has far more improvements and new features than > older versions to SCTP. You can take it from git now: > http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git > > Some highlights about sctp and 3.0.0: > http://by-miconda.blogspot.com/2010/01/best-of-new-in-kamailio-300-15-sctp.h > tml > > Cheers, > Daniel >

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[Kamailio-Users] Rtpproxy not recording media stream
by Vikram Ragukumar 13 Jan '10

13 Jan '10

Hello, I have setup Kamailio(v1.5) and Rtpproxy(v1.2.1) on a Centos5.4 linux box. Kamailio's config file has been edited to include Nat and mysql support. With Kamailio running, i started rtpproxy using rtpproxy -l ip_addrs -s udp:127.0.0.1:7722 -a -r /var/log -F However rtpproxy does not seem to be recording any session. I read that the -a switch can be used to record all sessions. Am i missing a switch/doing something wrong while invoking rtpproxy ? Regards, Vikram.

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[Kamailio-Users] Logical operators
by Antonio Reale 13 Jan '10

13 Jan '10

Hi, I'm trying this: if(is_avp_set("$avp(s:avp)") && !isflagset(1) || is_avp_set("$avp(s:avp)") && !isflagset(2)) { ...; } The first condition before OR operator is never matched. Could you please suggest the correct syntax? Thank you very much. Regards. -- Antonio

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[Kamailio-Users] SIP proxy is expecting "Route:" header to be filled information from the last "Contact:" header ?
by Julien Chavanton 13 Jan '10

13 Jan '10

Hi all, first thank you for maintaining this list and free/Open source SIP proxy software. I am working on one interco problem with a remote SIP proxy running on Kamailio. Cisco(U1) --> Opensips(P1) --> Internet --> Kamailio(P2) --> Asterisk(U2) The remote proxy(P2) is expecting us to fill the "Route:" header taking information from the "Contact:" header, according to my research the "Route:" header as to be filled using the information found in the last "Record-route:" header. Note that the "Contact:" header IP address is the one of U2, as it is the destination UA. I think our partner using Kalamailio only interconnected UA not Proxy, since they never faced this problem. Of course I do not suspect Kalamailio to be doing something wrong, I just want to validate what we should expect, and investigate further from there. I will have to contact the technicien that as configured the Kalamailio server to validate why face this behavior. -------------------------------------------------------------------------- " 16.12.1.3 Rewriting Record-Route Header Field Values " http://www.ietf.org/rfc/rfc3261.txt -------------------------------------------------------------------------- We found a temporary solution to make it worko, where we hardcoded the replacement of the IP address in the "Route:" header with the one of the U2 subst('/^Route: <sip:P2;(.*)$/Route: <sip:U2;\1/');

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[Kamailio-Users] Does uac_auth work for you?
by Andrey Utkin 13 Jan '10

13 Jan '10

Hi all. Please share your experience. If you ever tried authenticating outgoing calls with Kamailio, did it work for you? Which devices or servers you know do/don't accept such authentication? Do you use it in some tricky way to get it working?

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[Kamailio-Users] about kamailio with sctp
by mustafa samara 13 Jan '10

13 Jan '10

Hello i do not know how to try sctp, imake kmailio 1.5 with sctp=1 could you help me? thnaks -- Eng.Mustafa Al-Samara

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Kamailio (OpenSER) v3.0.0 Released
by Daniel-Constantin Mierla 12 Jan '10

12 Jan '10

Hi everybody, this ML was forgotten yesterday to be cc-ed ...for the sake of archiving, here it is the announcement. After ten months since previous major release 1.5.0, I am glad to announce that Kamailio (OpenSER) v3.0.0 is out. It is a special release, being the first based on SIP Router project. Along with new things developed during the past months, you can blend features and modules provided by previous versions of Kamailio (OpenSER) and SIP Express Router (SER) in same configuration file. Given these, 3.0.0 offers an impressive number of new features and improvements, read release notes at: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v3.0.0-Release-Notes Wiki page with more details about what is new: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x The tarball with sources: http://www.kamailio.org/pub/kamailio/latest/src/ Binary packages for several distributions: http://www.kamailio.org/pub/kamailio/latest/bin/ http://www.kamailio.org/pub/kamailio/latest/packages/ Modules' documentation: - http://www.kamailio.org/docs/modules/3.0.x/ Cookbooks and more wiki docs: - http://www.kamailio.org/dokuwiki/ Migration guidelines: - http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 From Kamailio project point of view, this is the first major milestone accomplished within SIP Router project, proving its viability. The work done so far, to offer the features brought out by this release, together with the development teams working together, make the future of the SIP Router project very solid. I would like to thank to everybody that contributed to get SIP Router project at this stage and I am eager to get to next major release from it source tree -- the work will continue in order to offer a more tight integration of Kamailio (OpenSER) and SIP Express Router (SER) projects, of course, along with development of new features. Cheers, Daniel

1 0

[Kamailio-Users] Checking the destination IP after dispatcher
by Geoffrey Mina 12 Jan '10

12 Jan '10

What pseudo variable would i check after running ds_next_dst() to check the IP we are about to forward the INVITE to? Basically I want to have a check for $si == [something] to ensure I am not about to send the INVITE to the same UI which just requested it. While I have your attention, I may as well see if anyone has a better solution for what I am trying to accomplish: I am doing this because asterisk can't handle a hairpin scenario and I need to generate two distinct calls in my network, so I figure i'll just forward the request to another server in the network to deal with this. INVITE from PSTN --> Kamailio --> Asterisk#1 [Asterisk#1 needs to create a new session, but stay on-net] --> INVITE from Asterisk#1 to Kamailio --> [Dispatcher] --> INVITE from Kamailio to Asterisk#2 This would basically create 3 legs of the call, which is what I want PSTN --> Asterisk#1 --> Asterisk#2 I would prefer if Asterisk#1 could just INVITE to LocalHost, but that doesn't seem to be allowed. Not sure if that's a limitation of SIP or Asterisk... Thanks!

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[Kamailio-Users] job offer: VoIP developer, team Bucharest
by Henning Westerholt 12 Jan '10

12 Jan '10

Job offer: Interested in working with one of the biggest Voice over IP networks in europe? We're hiring and have one additional VoIP developer position open in the team in Bucharest, Romania at the moment. For more informations (requirements, how to apply..) please refer to the opening on the job web page. Some facts about our VoIP system can be found at e.g. [1], please feel free to contact me per mail as well. http://cariere.1and1.ro/xml/JobsDetail?job_id=2142 Best regards, Henning Westerholt [1] http://www.kamailio.org/events/2009-linuxtag/presentation.pdf -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 0

[Kamailio-Users] This has probably been asked a thousand times before...
by Robert B 12 Jan '10

12 Jan '10

Regardless, despite my Googlage, I can't seem to find what I am looking for. I've looked at the Wiki and the documentation. While I do have a programming background, I still am having a hard time grasping some of these tutorials. Many of them seem very specific and not at all relevant to what I am trying to do. I feel like I'm left without a good starting point. My goal is to have Kamailio setup as an ITSP would. That is, I have three sipxecs PBX systems here, and rather than have each one configured with umpteen SIP trunks that I'll need to manage, I want to have (for the time being) just one -- that one would be my personal Kamailio server. When a user makes a call, it first goes to my Kamailio server which then distribute the calls to my ITSPs (like Bandwidth.com, voip.ms, etc). I want the ACLs to be setup based on IP addresses and not user registrations. I want nothing to do with the media or RTP streams. First step would be termination (outbound). The obvious next step would be setting it up for origination (inbound). I've downloaded and installed the SIREMIS web UI for OpenSER 1.4.x, but as I said my lack of understanding of the terminology's context prohibits me from really exploring anything. Some areas look familiar, others seem redundant, etc. The reason I started with the web UI was because it's obviously easier and it also produces output that is syntactically correct for my own examination and learning purposes. What would the appropriate terminology for this kind of setup be? Can anyone point out a Wiki article which I may have missed? To me, this seems like the most basic of basic configurations and a seemingly obvious launching point for anyone getting into the software... Thanks! -- Robert

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[Kamailio-Users] Kamailio (OpenSER) v3.0.0 Released
by Daniel-Constantin Mierla 11 Jan '10

11 Jan '10

Hi everybody, after ten months since previous major release 1.5.0, I am glad to announce that Kamailio (OpenSER) v3.0.0 is out. It is a special release, being the first based on SIP Router project. Along with new things developed during the past months, you can blend features and modules provided by previous versions of Kamailio (OpenSER) and SIP Express Router (SER) in same configuration file. Given these, 3.0.0 offers an impressive number of new features and improvements, read release notes at: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v3.0.0-Release-Notes Wiki page with more details about what is new: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x The tarball with sources: http://www.kamailio.org/pub/kamailio/latest/src/ Binary packages for several distributions: http://www.kamailio.org/pub/kamailio/latest/bin/ http://www.kamailio.org/pub/kamailio/latest/packages/ Modules' documentation: - http://www.kamailio.org/docs/modules/3.0.x/ Cookbooks and more wiki docs: - http://www.kamailio.org/dokuwiki/ Migration guidelines: - http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 From Kamailio project point of view, this is the first major milestone accomplished within SIP Router project, proving its viability. The work done so far, to offer the features brought out by this release, together with the development teams working together, make the future of the SIP Router project very solid. I would like to thank to everybody that contributed to get SIP Router project at this stage and I am eager to get to next major release from it source tree -- the work will continue in order to offer a more tight integration of Kamailio (OpenSER) and SIP Express Router (SER) projects, of course, along with development of new features. Cheers, Daniel

3 2

[Kamailio-Users] preparing Kamailio 3.0.0 release
by Daniel-Constantin Mierla 11 Jan '10

11 Jan '10

Hello, as announced previously, today will package the kamailio 3.0.0 release. Please commit any change to kamailio_3.0 branch before 14:00UTC to avoid overlappings. If you are aware of new bugs, just drop an email to devel mailing list. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 1

Re: [Kamailio-Users] Users Digest, Vol 56, Issue 19
by mustafa samara 11 Jan '10

11 Jan '10

thank you very much i will see kamailio 3.0 but is there any voip soft phone which support sctp, please? thanks

2 1

Re: [Serusers] configure SER to work as a gateway
by "Martínez García, Rebeca" 11 Jan '10

11 Jan '10

Hi all, Thanks Rupert for your answer, I realise that my explanation was not very clear but the scenario I am working into is a little bit complex. Finally, I have decided to change my configuration and make the client sends its REGISTER message to SER and then it will forward the same message to my Server. However, the Server carries out the authentication process (I can not change it), so SER has to save the location contact after the client is authenticated in the Server. My function for the REGISTER message in SER is the following one: route[REGISTER] { t_newtran(); # send the register to the proxy rewritehostport("open-ims.test:4060"); t_on_reply("REGISTER_reply"); #t_on_failure("REGISTER_failure"); if (!t_relay()) { sl_send_reply("500","Error forwarding to P-CSCF"); break; } } onreply_route[REGISTER_reply] { if (t_check_status("401")){ log(-1, "A response from the P-CSCF!"); break; } if (t_check_status("200")){ log(-1, "User registered in the IMS Core!"); save_contacts("location"); break; } } But SER does not start. It shows this error: cng@cng:~$ sudo /etc/init.d/ser start -P /var/run/ser/ser.pid Starting ser: ser 0(2545) parse error (147,19-20): Command cannot be used in the block I have checked where is the error and it complains about the "save_contacts("location"))", but I do not know why I can not use it inside the onreply_route. Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: vie 08/01/2010 15:03 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Hi Rebeca, I think I understand what you are trying to do, albeit you describe it in a strange way. SER is designed to be a Registrar / SIP Proxy. It is not designed to proxy REGISTER messages. However I had to interface it to a SIP Application Server that wanted to see the REGISTER messages. However SER was my REGISTRAR and performed my security / authentication. Therefore I filter on the REGISTER messages using if (method=="REGISTER") { after doing the auth challenge in the ser.cfg, and after doing an eNum lookup and some user agent field checks I simply t_replicate the REGISTER. i.e. t_replicate("rebeca.node.es.com", "5060"); or if you do not use DNS t_replicate("10.1.x.y", "5060"); Note this replication does not include the auth challenge....but it satisfys my SIP app server functionality nicely, Also you can overwrite the useragent field if you wish... e.g. if (subst_uri("/;userAgent=rupert")) { subst('/^User-Agent:.*$/User-Agent: Rebeca/ig'); Good luck Rupert ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of "Martínez García, Rebeca" Sent: 08 January 2010 11:52 To: serusers(a)lists.iptel.org Cc: "Huertas García, Victor" Subject: [Serusers] configure SER to work as a gateway Hello all, Solved the installation problem I had, now I am trying to configure SER in a certain way. My configuration is the following one: Client ------------------------- SER ---------------------------- Server/Proxy 192.168.x.x 192.168.x.y 10.1.x.x 10.1.x.y Unlike the normal case where SER is used as the Registrar, my client (in a private LAN) must register in a Server/Proxy and SER has to act as a SIP gateway. However, at the same time, SER has to register locally this client (for internal calls). So, put in other words, SER should forward the REGISTER message (directed from the client to the Server/Proxy) and save locally the location of that client. Once registered the user, SER should act as a transparent proxy. But in the practice, it does nothing if I put that the client registers in the Server/Proxy. The messages reach SER but it does not forward them, nor it registers locally the clients. So I tried configuring the client to register in SER instead of the Server/Proxy and then SER started to work. Is a listening problem? What I have to include to the ser.cfg so SER handles also messages which are not directed to it but to the Server/Proxy? I hope somebody could help me. Thanks in advance, Rebeca Martinez

1 0

[Kamailio-Users] oracle sqlops URL
by alex pappas 11 Jan '10

11 Jan '10

Hi all, happy 2010 for all of us! Dear friends I'm trying to setup a sqlops connection to an Oracle db. I loaded the db_oracle module with (loadmodule "db_oracle.so") without parameters. The connection string that I gave in sqlops module is : modparam("sqlops","sqlcon","ca2=>user/pass@ip_addres:1521:db_name") When i try to start kamailio I get the following errors and kamailio does NOT start: ERROR:core:db_check_api: module db_user/pass@ip does not export db_use_table function ERROR:core:init_mod_child: failed to initializing module sqlops, rank 2 ERROR:core:main_loop: init_child failed for UDP listener Thank in advance and merry Christmas Alex

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[Kamailio-Users] about kamailio and tls
by mustafa samara 11 Jan '10

11 Jan '10

i am mustafa samara master degree student. i try to test qjsimple with kamailio openser is it possible (to test the tls support) ? also i want to ask about ( in sip preferences) what is the deference when we you use tls as a protocol or when we use (tls or sips) as a SRTP requirements. clould you help me please? i wat to include the result in my thesis -- Eng.Mustafa Al-Samara

3 2

[Kamailio-Users] PDD
by Anders 09 Jan '10

09 Jan '10

Hi, Anyone with a smooth solution for tracking PDD of every call? Thanks!

5 6

[Serusers] configure SER to work as a gateway
by "Martínez García, Rebeca" 08 Jan '10

08 Jan '10

Hello all, Solved the installation problem I had, now I am trying to configure SER in a certain way. My configuration is the following one: Client ------------------------- SER ---------------------------- Server/Proxy 192.168.x.x 192.168.x.y 10.1.x.x 10.1.x.y Unlike the normal case where SER is used as the Registrar, my client (in a private LAN) must register in a Server/Proxy and SER has to act as a SIP gateway. However, at the same time, SER has to register locally this client (for internal calls). So, put in other words, SER should forward the REGISTER message (directed from the client to the Server/Proxy) and save locally the location of that client. Once registered the user, SER should act as a transparent proxy. But in the practice, it does nothing if I put that the client registers in the Server/Proxy. The messages reach SER but it does not forward them, nor it registers locally the clients. So I tried configuring the client to register in SER instead of the Server/Proxy and then SER started to work. Is a listening problem? What I have to include to the ser.cfg so SER handles also messages which are not directed to it but to the Server/Proxy? I hope somebody could help me. Thanks in advance, Rebeca Martinez

3 2

[Kamailio-Users] Unreadable script
by Bruno Machado 08 Jan '10

08 Jan '10

Hi all I'm using the version 1.5.2 and I'd like to know from you if it is possible to 'hide' the config file. Using other words, becoming the file unreadable for us, humans. There are people that have root access to the machine and I don't want to let them read the script. Does Kamailio have a function to do this kind of job? tks, Machado ____________________________________________________________________________________ Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

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[Kamailio-Users] Problems with avp_printf in Kamailio-3.0.0
by Santiago Gimeno 07 Jan '10

07 Jan '10

Hi, I'm getting errors when trying to use avp_printf in K-3.0.0. The script line, for example, is: avp_printf("$avp(i:905)", "$rU"); The error is: kamailio: DEBUG: <core> [sr_module.c:516]: find_export_record: <avp_printf> not found I've tried pv_printf with no luck either. This was working correctly in K-1.5 Could this be a bug? Best regards, Santi

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[Kamailio-Users] openser/kamailio subscribe / MWI
by Joel Webster 07 Jan '10

07 Jan '10

I am looking to MWI messages openser/kamailio and send the MWI message back based on a database lookup or other means within a routing block, when the users register though, i get errors like: Jan 3 12:07:20 ser1 /usr/local/sbin/kamailio[20317]: ERROR:presence:get_database_info: No matching subscription dialog found in database Jan 3 12:07:20 ser1 /usr/local/sbin/kamailio[20317]: ERROR:presence:handle_subscribe: getting stored info and when I disable the DB setup I had, I get a similar error: Jan 4 16:59:48 ser1 /usr/local/sbin/kamailio[5211]: ERROR:presence:get_stored_info: record not found in hash_table Jan 4 16:59:48 ser1 /usr/local/sbin/kamailio[5211]: ERROR:presence:handle_subscribe: getting stored info

2 1

Re: [Kamailio-Users] [kamailio-users] Kamailio - Polycom Sound Point
by Ramu 07 Jan '10

07 Jan '10

Hi, Your configuration looks fine. Can you try with 2 X-Lite phones on different machines just to narrow down the problem? On Wed, Jan 6, 2010 at 4:29 PM, Chandrakant Solanki < solanki.chandrakant(a)gmail.com> wrote: > Hi > > Thanks for reply > > Here is kamailio.cfg file > > ================================================================================================ > listen = 192.168.94.30 > port = 5060 > mpath = "/lib/kamailio/modules" > debug = 9 > fork = yes > disable_tcp = no > log_facility = LOG_LOCAL0 > log_stderror = no > dns = no > rev_dns = no > > loadmodule "usrloc.so" > modparam("usrloc", "user_column", "username") > modparam("usrloc", "domain_column", "domain") > modparam("usrloc", "contact_column", "contact") > modparam("usrloc", "expires_column", "expires") > modparam("usrloc", "q_column", "q") > modparam("usrloc", "callid_column", "callid") > modparam("usrloc", "cseq_column", "cseq") > modparam("usrloc", "methods_column", "methods") > modparam("usrloc", "flags_column", "flags") > modparam("usrloc", "user_agent_column", "user_agent") > modparam("usrloc", "received_column", "received") > modparam("usrloc", "socket_column", "socket") > modparam("usrloc", "use_domain", 0) > modparam("usrloc", "desc_time_order", 0) > modparam("usrloc", "timer_interval", 60) > modparam("usrloc", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("usrloc", "db_mode", 1) > modparam("usrloc", "matching_mode", 0) > modparam("usrloc", "cseq_delay", 20) > modparam("usrloc", "nat_bflag", 6) > > loadmodule "pv.so" > > loadmodule "registrar.so" > modparam("registrar", "default_expires", 3600) > modparam("registrar", "min_expires", 60) > modparam("registrar", "max_expires", 0) > modparam("registrar", "default_q", 0) > modparam("registrar", "append_branches", 1) > modparam("registrar", "case_sensitive", 0) > modparam("registrar", "received_param", "received") > modparam("registrar", "max_contacts", 100) > modparam("registrar", "retry_after", 0) > modparam("registrar", "method_filtering", 0) > modparam("registrar", "path_mode", 2) > modparam("registrar", "path_use_received", 0) > modparam("registrar", "received_avp", "$avp(i:801)") > > loadmodule "rr.so" > modparam("rr", "enable_full_lr", 0) > modparam("rr", "append_fromtag", 1) > modparam("rr", "enable_double_rr", 1) > modparam("rr", "add_username", 0) > > loadmodule "tm.so" > modparam("tm", "fr_timer", 3) > modparam("tm", "fr_inv_timer", 35) > modparam("tm", "wt_timer", 5) > modparam("tm", "delete_timer", 2) > modparam("tm", "ruri_matching", 1) > modparam("tm", "via1_matching", 1) > modparam("tm", "unix_tx_timeout", 2) > modparam("tm", "restart_fr_on_each_reply", 1) > modparam("tm", "pass_provisional_replies", 0) > modparam("tm", "fr_inv_timer_avp", "$avp(s:callee_fr_inv_timer)") > > loadmodule "xlog.so" > modparam("xlog", "buf_size", 4096) > modparam("xlog", "force_color", 0) > > loadmodule "mi_fifo.so" > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > > loadmodule "domain.so" > modparam("domain", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("domain", "db_mode", 1) > modparam("domain", "domain_table", "domain") > modparam("domain", "domain_col", "domain") > > loadmodule "nathelper.so" > modparam("nathelper", "natping_interval", 60) > modparam("nathelper", "ping_nated_only", 1) > modparam("nathelper", "sipping_method", "OPTIONS") > modparam("nathelper", "received_avp", "$avp(i:801)") > > loadmodule "textops.so" > > loadmodule "sl.so" > modparam("sl", "enable_stats", 1) > > loadmodule "maxfwd.so" > modparam("maxfwd", "max_limit", 256) > > loadmodule "db_mysql.so" > modparam("db_mysql", "ping_interval", 300) > modparam("db_mysql", "auto_reconnect", 1) > > loadmodule "auth.so" > modparam("auth", "nonce_expire", 300) > modparam("auth", "rpid_suffix", > ";party=calling;id-type=subscriber;screen=yes") > modparam("auth", "rpid_avp", "$avp(s:rpid)") > > loadmodule "auth_db.so" > modparam("auth_db", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("auth_db", "user_column", "username") > modparam("auth_db", "domain_column", "domain") > modparam("auth_db", "password_column", "password") > modparam("auth_db", "password_column_2", "ha1b") > modparam("auth_db", "calculate_ha1", 1) > modparam("auth_db", "use_domain", 1) > modparam("auth_db", "load_credentials", "$avp(s:caller_uuid)=uuid") > > loadmodule "uri_db.so" > modparam("uri_db", "db_url", "mysql:// > database:database@192.168.90.12/openser") > > loadmodule "avpops.so" > modparam("avpops", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("avpops", "avp_table", "usr_preferences") > modparam("avpops", "use_domain", 1) > modparam("avpops", "uuid_column", "uuid") > modparam("avpops", "username_column", "username") > modparam("avpops", "domain_column", "domain") > modparam("avpops", "attribute_column", "attribute") > modparam("avpops", "value_column", "value") > modparam("avpops", "type_column", "type") > > loadmodule "lcr.so" > modparam("lcr", "db_url", "mysql://database:database@192.168.90.12/openser > ") > modparam("lcr", "gw_table", "gw") > modparam("lcr", "gw_name_column", "gw_name") > modparam("lcr", "ip_addr_column", "ip_addr") > modparam("lcr", "port_column", "port") > modparam("lcr", "uri_scheme_column", "uri_scheme") > modparam("lcr", "transport_column", "transport") > modparam("lcr", "grp_id_column", "grp_id") > modparam("lcr", "lcr_table", "lcr") > modparam("lcr", "strip_column", "strip") > modparam("lcr", "prefix_column", "prefix") > modparam("lcr", "from_uri_column", "from_uri") > modparam("lcr", "priority_column", "priority") > modparam("lcr", "flags_avp", "$avp(i:712)") > modparam("lcr", "gw_uri_avp", "$avp(i:1400)") > modparam("lcr", "ruri_user_avp", "$avp(i:1402)") > modparam("lcr", "fr_inv_timer_avp", "$avp(s:fr_inv_timer_avp)") > modparam("lcr", "fr_inv_timer", 90) > modparam("lcr", "fr_inv_timer_next", 30) > modparam("lcr", "rpid_avp", "$avp(s:rpid)") > > loadmodule "alias_db.so" > modparam("alias_db", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("alias_db", "user_column", "username") > modparam("alias_db", "domain_column", "domain") > modparam("alias_db", "alias_user_column", "alias_username") > modparam("alias_db", "alias_domain_column", "alias_domain") > modparam("alias_db", "use_domain", 0) > > loadmodule "uac.so" > modparam("uac", "rr_store_param", "vsf") > modparam("uac", "from_restore_mode", "auto") > modparam("uac", "from_passwd", "s1p:Wiz4rd!") > > loadmodule "acc.so" > modparam("acc", "db_url", "mysql://database:database@192.168.90.12/openser > ") > modparam("acc", "db_flag", 1) > modparam("acc", "db_missed_flag", 2) > modparam("acc", "log_flag", 1) > modparam("acc", "log_missed_flag", 2) > modparam("acc", "log_extra", > "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") > modparam("acc", "db_extra", > "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") > > loadmodule "mi_datagram.so" > modparam("mi_datagram", "socket_name", "udp:127.0.0.1:8080" ) > modparam("mi_datagram", "children_count", 3 ) > modparam("mi_datagram", "unix_socket_mode", 0600 ) > modparam("mi_datagram", "unix_socket_group", "root" ) > modparam("mi_datagram", "unix_socket_user", "root" ) > modparam("mi_datagram", "socket_timeout", 2000 ) > > loadmodule "mediaproxy.so" > modparam("mediaproxy", "mediaproxy_socket", > "/var/run/mediaproxy/dispatcher.sock") > modparam("mediaproxy", "mediaproxy_timeout", 500) > modparam("mediaproxy", "signaling_ip_avp", "$avp(s:nat_ip)") > modparam("mediaproxy", "media_relay_avp", "$avp(s:media_relay)") > > loadmodule "nat_traversal.so" > modparam("nat_traversal", "keepalive_interval", 90) > modparam("nat_traversal", "keepalive_method", "OPTIONS") > modparam("nat_traversal", "keepalive_from", "sip:keepalive@mydomain.com<sip%3Akeepalive(a)mydomain.com> > ") > > route[0] > { > xlog("L_DBG", "New Request - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > force_rport(); > > if(msg:len > max_len) > { > xlog("L_DBG", "Message too big - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > sl_send_reply("513", "Message Too Big"); > exit; > } > if (!mf_process_maxfwd_header("10")) > { > xlog("L_DBG", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > sl_send_reply("483", "Too Many Hops"); > exit; > } > if(!is_method("REGISTER")) > { > if(nat_uac_test("19")) > { > record_route(";nat=yes"); > } > else > { > record_route(); > } > } > if (is_method("SUBSCRIBE")) { > # challenge/response > if (!www_authorize("", "subscriber")) { > www_challenge("", "0"); > exit; > } > exit; > } > > if(is_method("CANCEL") || is_method("BYE")) > { > end_media_session(); > } > if(loose_route()) > { > if(nat_uac_test("19") || search("^Route:.*;nat=yes")) > { > fix_nated_contact(); > setbflag(6); > } > route(8); > } > if(is_method("REGISTER")) > { > route(7); > } > if(is_method("INVITE")) > { > use_media_proxy(); > route(9); > } > if(is_method("CANCEL") || is_method("ACK")) > { > route(13); > } > route(14); > } > > ######################################################################## > # Request route 'clear-usr-preferences-caller' > ######################################################################## > route[1] > { > xlog("L_DBG", "Clear caller preferences - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > avp_delete("$avp(s:caller_cli)/g"); > avp_delete("$avp(s:clir)/g"); > } > > ######################################################################## > # Request route 'clear-usr-preferences-callee' > ######################################################################## > route[2] > { > xlog("L_DBG", "Clear callee preferences - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > avp_delete("$avp(s:callee_fr_inv_timer)/g"); > avp_delete("$avp(s:cfu)/g"); > avp_delete("$avp(s:cfc)/g"); > } > > ######################################################################## > # Request route 'usr-preferences-caller' > ######################################################################## > route[3] > { > route(1); > xlog("L_DBG", "Load caller preferences for uuid > '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > # load caller avps > avp_db_load("$avp(s:caller_uuid)", "*"); > avp_copy("$avp(s:cli)", "$avp(s:caller_cli)/d"); > if(is_avp_set("$avp(s:clir)/n") && avp_check("$avp(s:clir)", > "eq/i:1")) > { > # mark for anonymization > setflag(28); > } > } > > ######################################################################## > # Request route 'usr-preferences-callee' > ######################################################################## > route[4] > { > xlog("L_DBG", "Load callee preferences for uuid > '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > # load callee avps > avp_db_load("$avp(s:callee_uuid)", "*"); > if(is_avp_set("$avp(s:cfu)/s")) > { > xlog("L_DBG", "Call-forward-unconditional to '$avp(s:cfu)' > found - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > avp_delete("$avp(s:caller_uuid)/g"); > avp_copy("$avp(s:callee_uuid)", "$avp(s:caller_uuid)/d"); > avp_pushto("$ru", "$avp(s:cfu)"); > > route(3); > route(10); > exit; > } > if(is_avp_set("$avp(s:ringtimeout)/n")) > { > xlog("L_DBG", "Setting ring timeout to $avp(s:ringtimeout) > secs - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > avp_copy("$avp(s:ringtimeout)", > "$avp(s:callee_fr_inv_timer)/d"); > } > } > > ######################################################################## > # Request route 'clir' > ######################################################################## > route[5] > { > if(isflagset(28) && !isflagset(27)) > { > setflag(27); > > xlog("L_DBG", "Anonymize caller - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > > uac_replace_from("Anonymous","sip:anonymous@anonymous.invalid"); > if(is_present_hf("Privacy")) > { > remove_hf("Privacy"); > } > append_hf("Privacy: id\r\n"); > } > } > > ######################################################################## > # Request route 'stop-rtp-proxy' > ######################################################################## > route[6] > { > if(isflagset(22)) > { > end_media_session(); > } > } > > ######################################################################## > # Request route 'base-route-register' > ######################################################################## > route[7] > { > sl_send_reply("100", "Trying"); > if(!www_authorize("", "subscriber")) > { > xlog("L_DBG", "Register authentication failed - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > www_challenge("", "0"); > exit; > } > if(!check_to()) > { > > xlog("L_DBG", "Spoofed To-URI detected - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > sl_send_reply("403", "Spoofed To-URI Detected"); > exit; > } > consume_credentials(); > if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) > { > fix_nated_register(); > setbflag(6); > } > > if(!save("location")) > { > xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > sl_reply_error(); > exit; > } > xlog("L_DBG", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > exit; > } > > ######################################################################## > # Request route 'base-outbound' > ######################################################################## > route[8] > { > if(is_present_hf("P-Asserted-Identity")) > { > remove_hf("P-Asserted-Identity"); > } > if(is_present_hf("Remote-Party-ID")) > { > remove_hf("Remote-Party-ID"); > } > if(is_avp_set("$avp(s:caller_cli)/s")) > { > if(!isflagset(28)) > { > xlog("L_DBG", "Set caller CLI '$avp(s:caller_cli)' > - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > append_hf("P-Asserted-Identity: > <$avp(s:caller_cli)>\r\n"); > } > } > > route(5); > if(isbflagset(6)) > { > if(!isflagset(22) && !search("^Content-Length:[ ]*0")) > { > setflag(22); > # use_media_proxy(); > } > t_on_reply("2"); > } > else > { > t_on_reply("1"); > } > if(!isflagset(21)) > { > t_on_failure("2"); > } > if(isflagset(29)) > { > append_branch(); > } > if(is_present_hf("Proxy-Authorization")) > { > # consume_credentials(); > } > > xlog("L_DBG", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > # no 100 (we already sent it) and no DNS blacklisting > if(!t_relay("0x05")) > { > sl_reply_error(); > if(is_method("INVITE") && isbflagset(6)) > { > end_media_session(); > } > } > exit; > } > > ######################################################################## > # Request route 'base-route-invite' > ######################################################################## > route[9] > { > sl_send_reply("100", "Trying"); > if(from_gw()) > { > $avp(s:caller_uuid)="0"; > xlog("L_DBG", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > setflag(23); > $avp(s:ringtimeout)=1000; > > route(3); > if (nat_uac_test("3")) > { > xlog("L_DBG", "Inside NATed INVITE Processing\n"); > fix_nated_contact(); > > fix_nated_sdp("3"); > > t_on_reply("1"); > } > else > { > t_on_reply("1"); > } > > if(nat_uac_test("19")) > { > setbflag(6); > } > > route(51); > } > else > { > if(!proxy_authorize("", "subscriber")) > { > xlog("L_DBG", "Proxy authentication failed - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > proxy_challenge("", "0"); > exit; > } > if(!check_from()) > { > xlog("L_DBG", "Spoofed From-URI detected - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > sl_send_reply("403", "Spoofed From-URI Detected"); > exit; > } > > route(3); > if(nat_uac_test("19")) > { > fix_nated_contact(); > setbflag(6); > } > route(10); > } > } > > ######################################################################## > # Request route 'invite-find-callee' > ######################################################################## > route[10] > { > if(alias_db_lookup("dbaliases")) > { > xlog("L_DBG", "Callee was aliased - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > } > > route(2); > if(!is_domain_local("$rd")) > { > setflag(20); > $avp(s:callee_uuid) = "0"; > route(12); > } > avp_delete("$avp(s:callee_uuid)"); > avp_db_query("select uuid from subscriber where username = '$rU'", > "$avp(s:callee_uuid)"); > if(is_avp_set("$avp(s:callee_uuid)/s")) > { > xlog("L_DBG", "Callee is local, uuid='$avp(s:callee_uuid)' > - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > route(11); > } > else > { > $avp(s:callee_uuid) = "0"; > xlog("L_DBG", "Callee is not local - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > # > # *86 - sending user direct to access voicemail from sip > client > # > if($rU=="*86") > { > route(12); > } > else > { > if($(rU{s.len})<=7) > { > route(11); > } > else > { > route(12); > } > } > } > exit; > } > > ######################################################################## > # Request route 'invite-to-internal' > ######################################################################## > > route[11] > { > route(4); > > > revert_uri(); > seturi("sip:*89@openser.org"); > > rewritehostport("voicemailserver.databasepbx.com:5060"); > route(8); > exit; > } > > ######################################################################## > # TEST ROUTE 51 > ######################################################################## > route[51] > { > if(alias_db_lookup("dbaliases")) > { > xlog("L_DBG", "Callee was aliased - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > } > > route(2); > if(!is_domain_local("$rd")) > { > setflag(20); > $avp(s:callee_uuid) = "0"; > route(12); > } > avp_delete("$avp(s:callee_uuid)"); > avp_db_query("select uuid from subscriber where username = '$rU'", > "$avp(s:callee_uuid)"); > if(is_avp_set("$avp(s:callee_uuid)/s")) > { > xlog("L_DBG", "Callee is local, uuid='$avp(s:callee_uuid)' > - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > route(52); > } > else > { > $avp(s:callee_uuid) = "0"; > xlog("L_DBG", "Callee is not local - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > route(12); > } > exit; > } > > ######################################################################## > #TEST 52 > ######################################################################## > route[52] > { > route(4); > if(!lookup("location")) > { > xlog("L_DBG", "Local user offline - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > sl_send_reply("404", "User Offline"); > } > else > { > xlog("L_DBG", "Local user online - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > avp_delete("$avp(s:callee_srcip)"); > avp_db_query("select received from location where username > = '$rU'", "$avp(s:callee_srcip)"); > avp_printf("$avp(i:13)", "$(du{s.select,1,:})" ); > avp_printf("$avp(i:14)", > "$(avp(s:callee_srcip){s.select,1,:})" ); > xlog("L_DBG", "source IP [$avp(i:13)] destination IP > [$avp(i:14)]\n"); > if (avp_check("$avp(i:13)","eq/$avp(i:14)/i")) > { > xlog("L_INFO", "Detected Two Clients Behind the > Same NAT - Disabling Mediaproxy"); > # Do not use mediaproxy as the clients seem to be > behind the same NAT > xlog("L_DBG", "source IP is the same as > destination IP \n"); > resetflag(2); > resetflag(3); > } > avp_delete("$avp(i:13)/g"); > avp_delete("$avp(i:14)/g"); > route(8); > } > exit; > } > > ######################################################################## > # Request route 'invite-to-external' > ######################################################################## > route[12] > { > if(isflagset(20)) > { > xlog("L_DBG", "Call to foreign domain - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > route(8); > exit; > } > if(!isflagset(23)) > { > if(uri =~ "^sip:\*86@.*") > { > xlog("L_DBG", "User accessing Voicemail - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > rewritehostport(" > voicemailserver.databasepbx.com:5060"); > route(8); > } > # don't allow calls relaying from PSTN to PSTN, if not > explicitely forwarded > else if(uri =~ "^sip:[0-9]+@") > { > # only route numeric users to PSTN > if(!load_gws()) > { > xlog("L_ERR", "Error loading PSTN gateways > - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > sl_send_reply("503", "PSTN Termination > Currently Unavailable"); > exit; > } > if(!next_gw()) > { > xlog("L_ERR", "No PSTN gateways available - > M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > sl_send_reply("503", "PSTN Termination > Currently Unavailable"); > exit; > } > setflag(21); > t_on_failure("1"); > route(8); > } > } > > xlog("L_DBG", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > sl_send_reply("404", "User Not Found"); > exit; > } > > ######################################################################## > # Request route 'base-route-local' > ######################################################################## > route[13] > { > t_on_reply("1"); > if(t_check_trans()) > { > xlog("L_DBG", "Request leaving server - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > } > else > { > xlog("L_DBG", "Dropping mis-routed request - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > exit; > } > if(!t_relay()) > { > sl_reply_error(); > } > exit; > } > > > ######################################################################## > # Request route 'base-route-generic' > ######################################################################## > route[14] > { > xlog("L_DBG", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > sl_send_reply("501", "Method Not Supported Here"); > exit; > } > > ######################################################################## > # Request route 'base-filter-failover' > ######################################################################## > route[15] > { > if(!t_check_status("408|500|503")) > { > xlog("L_DBG", "No failover routing needed for this response > code - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > route(6); > exit; > } > } > > > ######################################################################## > #Send the call out > ######################################################################## > > route[16] > { > #send the call outward > if(method=="INVITE" && !isflagset(10)) > { > t_on_failure("2"); > } > > if (!t_relay()) > { > xlog("L_WARN", "ERROR: t_relay failed"); > sl_reply_error(); > } > } > > ######################################################################## > # Request route 'Voicemail and Asterisk for further processing' > ######################################################################## > failure_route[3] > { > if(!t_was_cancelled()) > { > revert_uri(); > rewritehostport("voicemailserver.databasepbx.com:5060"); > append_branch(); > #PREVENT SOME CRAZY VOICEMAIL LOOP > xlog("L_DBG", "No Answer Sending the call to Voicemail - > M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > setflag(10); > route(8); > } > } > > ######################################################################## > # Reply route 'base-standard-reply' > ######################################################################## > onreply_route[1] > { > xlog("L_DBG", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); > if (nat_uac_test("1")) > { > fix_nated_contact(); > } > exit; > } > > ######################################################################## > # Reply route 'base-nat-reply' > ######################################################################## > onreply_route[2] > { > xlog("L_DBG", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si > ID=$ci\n"); > if(nat_uac_test("1")) > { > fix_nated_contact(); > xlog("L_DBG", "NAT-Reply==[Fix_nated_contact] - S=$rs D=$rr > F=$fu T=$tu IP=$si ID=$ci\n"); > } > > if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") > { > if(!search("^Content-Length:[ ]*0")) > { > use_media_proxy(); > xlog("L_DBG", "NAT-Reply==[Force Rtp proxy] - S=$rs > D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); > } > } > exit; > } > > ######################################################################## > # Failure route 'pstn-failover' > ######################################################################## > failure_route[1] > { > xlog("L_DBG", "Failure route for PSTN entered - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > route(15); > if(!next_gw()) > { > xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > route(6); > exit; > } > > t_on_failure("3"); > route(8); > } > > ######################################################################## > # Failure route 'base-standard-failure' > ######################################################################## > failure_route[2] > { > if(!t_was_cancelled()) > { > revert_uri(); > route(10); > } > if(t_check_status("422|481|487")) > { > xlog("L_DBG", "Final reply - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > route(6); > exit; > } > > t_on_branch("1"); > xlog("L_DBG", "Redirect from UAC intercepted - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > route(10); > if($avp(s:cfc) != NULL) > { > avp_pushto("$ru", "$avp(s:cfc)"); > setflag(29); > append_branch(); > > t_on_branch("1"); > xlog("L_DBG", "CFC detected - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > route(10); > } > > route(15); > route(6); > } > > ######################################################################## > # Branch route 'cfc-drop-local' > ######################################################################## > branch_route[1] > { > if(is_domain_local("$rd")) > { > xlog("L_DBG", "Dropping local branch - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > drop(); > } > } > > =============================================================================================== > > MediaProxy's config.ini > > ; > ; Configuration file for MediaProxy > ; > > [Relay] > relay_ip = 192.168.94.30 > dispatchers = 192.168.94.30 > port_range = 8000:20000 > log_level = DEBUG > on_hold_timeout = 1800 > stream_timeout = 90 > > [Dispatcher] > socket = /var/run/mediaproxy/dispatcher.sock > listen = 192.168.94.30 > listen_management = 192.168.94.30 > log_level = DEBUG > management_use_tls = no > > [TLS] > certs_path = tls > verify_interval = 300 > > [OpenSIPS] > max_connections = 100 > > Is there any problem in config file then please suggest me. This > configuration was running with other phone but not with * > PolycomSoundPointIP-SPIP_600-UA/3.1.3.0439*. > > > Thanks in advance > -- > Regards, > > Chandrakant Solanki > > On Wed, Jan 6, 2010 at 3:38 PM, Ramu <contactramu(a)gmail.com> wrote: > >> Hi, >> >> You have to do NATing if the UAs are behind NAT when REGISTER and INVITE >> packet comes to Kamailio. >> >> You have to call use_media_proxy() function for INVITE and 200 OK >> messages. >> >> Please send your kamailio.cfg file to resolve the issues. >> >> >> On Wed, Jan 6, 2010 at 11:12 AM, Chandrakant Solanki < >> solanki.chandrakant(a)gmail.com> wrote: >> >>> Hello >>> >>> I have installed kamailio - 1.5.2 with MediaProxy. >>> >>> I registered 101(a)domain.com and 102(a)domain.com from * >>> PolycomSoundPointIP-SPIP_600-UA/3.1.3.0439*. >>> >>> But when I look for location table into database, I show received field >>> is NULL, while from other phone like X-Lite it shows some value in same >>> field like sip:X.X.X.X:5061. >>> >>> When I call from 101 to 102 using *PolycomSoundPointIP *phone, The phone >>> rings on the other extension but when I pick up no voice is heard, the >>> caller phone keeps hearing the wait-for-other party. >>> >>> Is there any changes required into kamailio.cfg or not. ?? >>> >>> Thanks in advance >>> >>> -- >>> Regards, >>> >>> Chandrakant Solanki >>> >>> _______________________________________________ >>> Kamailio (OpenSER) - Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >> >> >> Thanks, >> Ramu >> > > -- Ramu

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[Kamailio-Users] [kamailio-users] Kamailio - Polycom Sound Point
by Chandrakant Solanki 06 Jan '10

06 Jan '10

Hello I have installed kamailio - 1.5.2 with MediaProxy. I registered 101(a)domain.com and 102(a)domain.com from * PolycomSoundPointIP-SPIP_600-UA/3.1.3.0439*. But when I look for location table into database, I show received field is NULL, while from other phone like X-Lite it shows some value in same field like sip:X.X.X.X:5061. When I call from 101 to 102 using *PolycomSoundPointIP *phone, The phone rings on the other extension but when I pick up no voice is heard, the caller phone keeps hearing the wait-for-other party. Is there any changes required into kamailio.cfg or not. ?? Thanks in advance -- Regards, Chandrakant Solanki

2 1

[Kamailio-Users] memlog and mem_dbg : kamailio 3.0.0
by Ricardo Martinez 05 Jan '10

05 Jan '10

Hello. I have a doubt about the memlog and memdbg log levels. I have in my configuration file the next lines : debug=2 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL0 children=4 port=5060 memdbg=0 memlog=0 I was expecting not to see any memory debugs, instead I have this lines in the log file : Jan 5 16:20:14 - /usr/local/sbin/kamailio[16672]: WARNING: <core> [mem/q_malloc.c:406]: qm_malloc(0x826db60, 8) returns address 0x82f9408 frag. 0x82f93f0 (size=8) on 1 -th hit Jan 5 16:20:14 - /usr/local/sbin/kamailio[16672]: WARNING: <core> [mem/q_malloc.c:428]: qm_free(0x826db60, 0x82f9408), called from nathelper: nathelper.c: nh_timer(3252) Jan 5 16:20:14 - /usr/local/sbin/kamailio[16672]: WARNING: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x82f93f0 alloc\'ed from nathelper: nathelper.c: nh_timer(3143) Is this ok? How can I remove such debug? Regards, Ricardo Martinez

2 1

[Kamailio-Users] t_replicate in Kamailio-3.0.0 ?
by Santiago Gimeno 05 Jan '10

05 Jan '10

Hi, I'm trying to use t_replicate in K-3.0.0 with no luck. t_replicate("sip:192.168.1.1:5060"); The logs report "unknown command" and by looking at the documentation of the TM module I've seen that t_replicate is no longer in the TM module. Is this correct? Any workaround? Thanks in advance. Best regards, Santi

2 3

[Kamailio-Users] odd query
by Anders 05 Jan '10

05 Jan '10

Hi, Using SIREMIS, there is a column called dst_domain in both the acc and cdrs tables. This is supposedly the ID of the gw used for the call. However, this seems to be messed up when there is more than one gw that has the same IP. I have gws with the same IP but different prefixes, but then it just seems to put into dst_domain whatever the ID is of the first gw with that IP. Not so good, since I need the info of which gw used to be able to find out how much the call cost me. Anyone knows about how to handle this? Is there a fix, hatch or work-around? Thanks!

1 0

[Kamailio-Users] multiple kamctl configurations
by Klaus Darilion 04 Jan '10

04 Jan '10

Hi! I have several Kamailio servers on a single server running. Is there a way to tell kamctl to use a different configuration (e.g. for fifo)? regards klaus

2 1

[Kamailio-Users] re-invite mid call?
by Antonio Goméz Soto 04 Jan '10

04 Jan '10

Hi, maybe I should go to the -dev list for this, but thought I'd ask here first. Is it possible to let openSER send a SIP reinvite in the middle of a call, to redirect the sound stream to some rtpproxy? Maybe using the MI? Or should OpenSER be modified to do this? Thanks, Antonio

5 25

[Kamailio-Users] Kamailio 3.0.0 - Jan 11, 2010
by Daniel-Constantin Mierla 04 Jan '10

04 Jan '10

Hello, it was a bit longer way than usual with the testing, but the achievements pay the bill - you can run kamailio and ser modules at the same time, as well as most of the core features from both applications. I propose to release Kamailio (OpenSER) 3.0.0 on 11th of January, by that time we should fix the small issues discussed on the devel list lately. If you are aware of other issues, report them to: sr-dev [at] lists.sip-router.org I encourage everyone to contribute to wiki pages for what is new and migration from 1.5.x: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 To get it from GIT and test, see: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git I started to detail some of the new features in blog posts, see ToC at: http://bit.ly/4qy9cn A great 2010 for kamailians and sip router users! Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Serusers] problems when starting ser
by "Martínez García, Rebeca" 04 Jan '10

04 Jan '10

Hello all, I have installed ser in Ubuntu Karmic by means of Synaptic without any problems or errors during the installation. But before starting to configure the ser.cfg, I decided firstly to try the configuration by default and I got this error (I only uncommented the "log_stderror=yes"): cng@cng:/etc/ser$ ser D E Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.1.3.207 [10.1.3.207]:5060 udp: 192.168.50.10 [192.168.50.10]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.1.3.207 [10.1.3.207]:5060 tcp: 192.168.50.10 [192.168.50.10]:5060 Aliases: tcp: cng.local:5060 tcp: cng:5060 tcp: localhost:5060 udp: cng.local:5060 udp: cng:5060 udp: localhost:5060 0(2956) init_tcp: using epoll_lt as the io watch method (auto detected) cng@cng:/etc/ser$ 0(2958) Maxfwd module- initializing 0(2958) ERROR: fifo_server.c:857: Can't create FIFO: No such file or directory (mode=384) 0(2958) ERROR: ctl: mod_init: init ctrl. sockets failed 0(2958) init_mod(): Error while initializing module ctl ERROR: error while initializing modules It seems that it starts ok but suddenly finds an error and stops. However if I run: cng@cng:/etc/ser$ sudo /etc/init.d/ser start Starting ser: serListening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.1.3.207 [10.1.3.207]:5060 udp: 192.168.50.10 [192.168.50.10]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.1.3.207 [10.1.3.207]:5060 tcp: 192.168.50.10 [192.168.50.10]:5060 Aliases: tcp: cng.local:5060 tcp: cng:5060 tcp: localhost:5060 udp: cng.local:5060 udp: cng:5060 udp: localhost:5060 0(3016) init_tcp: using epoll_lt as the io watch method (auto detected) . 0(3018) unable to create pid file /var/run/ser/ser.pid: No such file or directory It only complains about the ser.pid file. I am a little bit confused. Where is the problem? Thanks in advance, Rebeca Martinez

3 4

[Kamailio-Users] New install woes - 1.5.3 w/ mysql
by Graham Wooden 02 Jan '10

02 Jan '10

Hi all, Apparently I am missing something fundamentally here; trying to get mysql functions working with the kamdbctl command, but it seems that while it locates the kamdbctl.mysql file, it can¹t locate/use the the locate_tool() function from the kamctl.base¹ (which is called at the top of kamdbctl.mysql). I have been going back through trunk/INSTALL file and not seeing anything that is standing out. Neither when I ³strace² the kamdbctl create¹ command. But seems that my environment is missing something. I don¹t recall such issues with my production 1.4.2 machine. It¹s a fresh install CentOS 5.4 with most items (e.g. mysql server/client) installed via the standard yum repo¹s. I tried 1.5.3 from both building it from source as as well as the linux/i386 binary set. Still the same results. Thoughts? [root@misfits kamailio]# kamctl start database engine 'MYSQL' loaded Control engine 'FIFO' loaded INFO: Starting Kamailio : INFO: started (pid: 9006) [root@misfits kamailio]# kamdbctl create /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 25: locate_tool: command not found error: 'mysql' tool not found: set MYSQL variable to correct tool path

2 5

[Kamailio-Users] A great 2010 for everyone!
by Daniel-Constantin Mierla 31 Dec '09

31 Dec '09

Hello, before starting the party, I want to wish everyone all the best for 2010. Many thanks for those helping during 2009 with Kamailio and SIP Router projects, with code, documentation, advocating and support on mailing list. It was a lot of work done and I am confident you will enjoy Kamailio 3.0.0 and its great improvements and new features. I started a series of posts trying to highlight the best of it from my point of view, link to first is here: http://by-miconda.blogspot.com/2009/12/best-of-new-in-kamailio-300-1-includ… Not finished, expect more before Kamailio 3.0.0 is out, planned to happen before mid of January 2010. The draft of new features is available on wiki: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x A fantastic night for all! Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] avpops OR sqlops
by alex pappas 31 Dec '09

31 Dec '09

Hi all, I'm trying to make an sql query through my kamailio script in order to check for number portability. i have a table with 2 col (phonenumber,prefix). I'm trying both modules avpops and sqlops but I'm not able to retreive any data. modparam("avpops", "db_url","mysql://openser:openserrw@localhost/openser") avp_db_query("SELECT prefix FROM ported WHERE phonenumber='$rU'","$avp(s:lnp_prefix)") ;The query run perfectly in command line. The return code is always 1 but when I print $avp(s:lnp_prefix) I get no result. I have 2 questions : 1. Which is the best way to check portability in my case 2. What I'm doing wrong and I'm not getting any result from the query. Thank you ALL Alex

5 46

[Kamailio-Users] 1.5.3 dialog module: dialogs stay in database when in "shutdown only" mode
by Nathan Angelacos 30 Dec '09

30 Dec '09

Hi, It appears that in "shutdown only" mode (modparm db_mode = 3 ) dialogs are not cleared from the database on startup: 1. kamctl start initiate call 2. kamctl stop 3. kamctl start terminate call (dialog is terminated correctly) 4. kamctl db show dialog shows the dialog record 5. kamctl stop 6. kamctl db show dialog shows the (now very dead) dialog record Perhaps the dialog table should be cleared right after step 3 or at the beginning of step 5 when the dialog module is in "shutdown only" mode.

2 1

[Kamailio-Users] Problems with Sip Redirect Client
by Daniel Cardoso Alves 30 Dec '09

30 Dec '09

Hello... I have a SIP Redirect Serve with a kamailio and it works fine with others system(NexTone, Asterisk, etc), but I have a problem with a SIP redirect client with Kamailio. I am using http://kamailio.org/docs/modules/1.5.x/uac_redirect.html to try manipulate the 3xx message. I would like to send one destiny at a time, but the get_redirects("10:2","redirect") send only 2 contacts and at teh same time. I get from my SIP Redirect Server this message: SIP/2.0 301 Moved Permanently..Via: SIP/2.0/UDP 2XX.XX.XXX.XXX:5080;branch=z9hG4bK633e.a3e32ff2.0;rport=5080;received=2XX.XX.XXX.XXX..Via : SIP/2.0/UDP 192.168.0.133:7191;received=201.XX.X.XXX;branch=z9hG4bK-d87543-254f7f3a1058e232-1--d87543-;rport=7191..To: <sip:05433330000 @2XX.XX.XXX.XXX:5080>;tag=609b19d62bd6886602a6c902875525b0.c8b2..From: "1938"<sip:1938@2XX.XX.XXX.XXX:5080>;tag=187fec3c..Call-ID: 4959bc 623e556571@ZGVzZW4tcGMtMDguZ292b2lwLmNvbS5icg....CSeq: 1 INVITE..Contact: sip:55341#5433330000@192.168.0.2, q=1, sip:123#5433330000@1 92.168.0.3, q=2, sip:55341#5433330000@192.168.0.4, q=3, sip:55341#5433330000@192.168.0.5, sip:55341#5433330000@192.168.0.6, sip:553 41#5433330000(a)192.168.0.7, sip:55341#5433330000@192.168.0.8, sip:55341#5433330000@192.168.0.9, sip:55341#5433330000@192.168.0.10. .Server: Test Redirect..Content-Length: 0.... How I can configure to send fist to sip:55341#5433330000@192.168.0.10 and after if I not sucess it sends to sip:55341#5433330000@192.168.0.9, etc...? Thanks... ____________________________________________________________________________________ Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

3 6

[Kamailio-Users] [kamailio-users] Kamailio 1.5.0 + MediaProxy 2.3.8
by Chandrakant Solanki 30 Dec '09

30 Dec '09

Hi Kamailio - 1.5.0 MediaProxy - 2.3.8 CentOS 5.2 - 2.6.18-92.el5PAE My kamailio is running on *192.168.X.Y* ** *Here is my /etc/mediaproxy/config.ini* ------------------------------------------------- ; Configuration file for MediaProxy ; [Relay] dispatchers = 192.168.X.Y relay_ip = <public ip> port_range = 8000:10000 log_level = DEBUG stream_timeout = 90 [Dispatcher] socket = /var/run/mediaproxy/dispatcher.sock listen = 192.168.X.Y listen_management = 192.168.X.Y log_level = DEBUG [TLS] certs_path = tls verify_interval = 300 [OpenSIPS] max_connections = 100 =========================================================== Start media-dispatcher / media-relay # media-dispatcher --no-fork # media-relay --no-fork When I see log it show dispatcher to relay and relay to dispatcher connected.. But when I call 101 to 102 it gives me error, as show below *Media Dispatcher error* error: No suitable relay found *Media Relay error* warning: Cannot use port pair 8684/8685 warning: Cannot use port pair 8686/8687 -- Regards, Chandrakant Solanki

3 2

Re: [Kamailio-Users] Caller and Callee behind the same NAT
by juergen＠glowka.de 29 Dec '09

29 Dec '09

On Tuesday 29 December 2009 01:55:27 juergen(a)glowka.de wrote: > Hello, > > I'm new here. So I say "hello" to everybody. > > Maybe, somebody can help me or push me in the right direction. > > If there are Caller and Callee behind the same NAT, so it should > be better, they connect directly (e.g. 192.168.x.1 <-->192.168.x.2), > instead of using the rtp-proxy (this causes a jitter noise) . > > > I'm using Kamailio 1.5.3 with RTPPROXY 1.2.1. > > Has anybody a code snip or a sample kamailio.cfg for me? > I would be thankful for it. Better than giving you the solution, try to guest it ... Think a little ... what have in common a REQUEST from UAC A to UAC B if they are behing THE SAME NAT ROUTER ? -- Raúl Alexis Betancor Santana Dimensión Virtual ____________________________________________ Dear Mr. Santana Thanks for trying to help. oh, I guess, they've got the same IP-Address ;-) I found this code-snip in : dokuwiki/doku.php/examples:caller-callee-behind-same-nat ______________________________________ else if ( isflagset(2) and isflagset(3) ) { log(1, "Both Clients are behind NAT"); # Store the destination domain into an AVP avp_printf("$avp(i:450)", "$dd"); if ( avp_check("i:450", "eq/$src_ip/g") ) { xlog("L_INFO", "2 Clients Behind the Same NAT - Disabling Mediaproxy"); # Do not use mediaproxy as the clients seem to be behind the same NAT resetflag(2); resetflag(3); } } _______________________________________ But this Code-snip contains many mistakes so I changed it to : _______________________________________ if ( isflagset(5) and isbflagset(6) ) { lookup("location"); log(1, "Both Clients are behind NAT"); # Store the destination domain into an AVP avp_printf("$avp(i:450)", "$dd"); if ( avp_check("$avp(i:450)", "eq/$src_ip/g") ) { xlog("L_INFO", "2 Behind the Same NAT - Disabling Mediaproxy"); # Do not use mediaproxy as the clients seem to be behind the same NAT resetflag(5); resetbflag(6); } } _______________________________________ This seems to work, but I still have got this looping, jittering feeping noise witch is getting louder and louder ... Two or three words I can talk, then it sounds, as if a group of Navajo-Indians gone attack me .... :-) Could it be the client? I use 2 PDAs, Windows Mobile, Portsip Ver. 2.6 Best Wishes, also for me Juergen Glowka

3 2

[Serusers] using ser to involve nat
by WangZhenzhong 29 Dec '09

29 Dec '09

Hi every body I am using ser to solve the nat problem, In my environment: invite A-->SER-->B 1. SER receive the invite message REQA from A(invite B) 2. SER judge the use B, if the B is in public network, will fix nat in the REQA message if judge the B is in private network , will not fix anything and I want to know, how to judge the B's network, when the SER receive the INVITE from A? Is there any examle code used in ser.cfg thank you very much 2009-12-29 姓名：王振中公司：艾诺通信系统(苏州)有限公司地址：苏州工业园区苏虹西路81号苏虹工业坊A8单元二楼

1 0

[Kamailio-Users] Caller and Callee behind the same NAT
by juergen＠glowka.de 29 Dec '09

29 Dec '09

Hello, I'm new here. So I say "hello" to everybody. Maybe, somebody can help me or push me in the right direction. If there are Caller and Callee behind the same NAT, so it should be better, they connect directly (e.g. 192.168.x.1 <-->192.168.x.2), instead of using the rtp-proxy (this causes a jitter noise) . I'm using Kamailio 1.5.3 with RTPPROXY 1.2.1. Has anybody a code snip or a sample kamailio.cfg for me? I would be thankful for it. Best wishes Juergen Glowka _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

2 1

Re: [Kamailio-Users] adding inbound and outbound routes seperately
by Klaus Darilion 28 Dec '09

28 Dec '09

please always Cc: the list.... On 23.12.2009 16:24, Rodney wrote: > Well could you explain how to actually get peering functioning > between Kamailio on inbound and outbound? > There is no "peering functioning" in kamailio. You self have to define how you want to peer. For example if you want to peer with somebody sending you inbound requests, you have to define at least: - number format - authentication (IP-based, TLS, digest) One you have defined those parameters, you can implement it, e.g. digest authentication or IP based authentication. The same for outbound route - you have to configure Kamailio so that outbound requests are sent in the proper format. And Kamailio can not register to any other SIP service. If you need such a feature, you better use Asterisk. regards klaus > > -----Original Message----- From: Klaus Darilion > [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 23, > 2009 8:49 AM To: Rodney Cc: users(a)lists.kamailio.org Subject: Re: > [Kamailio-Users] adding inbound and outbound routes seperately > > > On 22.12.2009 22:35, Rodney wrote: >> Ok, I have kamailio and siremis, and I need to know how to set up >> a separate in bound and outbound route. >> >> Peering with inbound and registering with outbound >> > > Kamailio is a sip proxy, thus it can not register to other SIP > proxy. > > regards klaus > >> Any ideas guys? >> >> R >> >> >> >> _______________________________________________ Kamailio (OpenSER) >> - Users mailing list Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

3 6

[Kamailio-Users] Newbie questions
by Antonio Goméz Soto 28 Dec '09

28 Dec '09

Hi, I have used asterisk before, but now am looking at kamailio for scaling up. I've read 'Building Telephony Systems with OpenSER', but I still have lots of questions.... so, here it goes: - I could not find a way to reboot a phone remotely, can it send SIP NOTIFY to all phones? - does it have something like the Asterisk Manager Interface, so I can connect/create an operator panel for it? Does that already exist somewhere? Something like FOP where operators can transfer calls, and check availability? - according to the docs it is not a BTBUA. What does this mean? I know that asterisk is a BTBUA, but what can a BTBUA do that a SIP proxy cannot? Is it only the moh and conferences? - lastly, is there a good tutorial on load-balancing somewhere? Thanks, Antonio.

5 12

[Serusers] About iptel 400 missing cookie
by Dennis.Cheng 28 Dec '09

28 Dec '09

Hi folks, I used a Draytek Vigor 2820PBX to register iptel, and I found 400 missing cookie issue. When my pbx send the reinvite to hold another side, iptel always responded pbx 400 missing cookie. But I checked invite Via field, its branch included magic cookie "z9hG4bK". I attached wireshark packets. BR Dennis

1 0

[Kamailio-Users] Merry Christmas and a Happy New Year!
by Elena-Ramona Modroiu 25 Dec '09

25 Dec '09

Hi everybody, before you leave the computers and go on holidays, I want to thank you all for the good work and support of the Kamailio and SIP Router projects. I wish you a Merry Christmas and a Happy New Year! As the project name is a Hawaiian word - the "Kamailian" greeting is: "Mele Kalikimaka me ka Hau'oli Makahiki Hou!" How would it be in your language? May the new year bring you joy, peace, good health and prosperity! Enjoy the winter holidays! Ramona

9 9

[Kamailio-Users] Transparent bridge mode
by Vikram Ragukumar 24 Dec '09

24 Dec '09

Hello All, I am trying to setup a test scenario, where i have Kamailio and rtpproxy running on one CentOS box (Server1) and i have Asterisk running on another CentOS box (Server2). Server1 has 2 NIC's eth0 and eth1 that are both assigned Public IP's. There is a transparent bridge br0 connecting eth0 and eth1 which also has its own Public IP. Finally eth0 on Server2 also has a Public IP. Server2 must be assigned a Public IP. My goal is to modify rtpproxy so that i can intercept packets traveling to Server2, process them and let them resume along their original path. I would like to know if there is another way of setting this up so that i dont use as many Public IP's ? Do any of you see a problem with this setup, things that may not work eventually, or any other concerns ? Thanks, Vikram.

5 12

[Kamailio-Users] Server Internal Error with sample config file : Kamailio 3.0 rc3
by Ricardo Martinez 24 Dec '09

24 Dec '09

Hello list. I'm using the sample configuration that comes with Kamailio RC3. (version: kamailio 3.0.0-rc3 (i386/linux)). I have two IP phones registered in the proxy. If I call from one to another and the call is cancelled a "500 Server Internal Error" message is send back to the caller (please see the debug attached) by the proxy. This is not happening with Kamailio 1.4 or Opensips 1.6 (using the sample configurations). Could be some problem? 10.0.0.220 - Caller 10.0.0.211 - Proxy 10.0.0.152 - Callee U 10.0.0.220:5060 -> 10.0.0.211:5060 INVITE sip:5501122334@10.0.0.211:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 70. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.220:5060;transport=udp>;+sip.instance="<urn:uuid: 00000000-0000-1000-8000-00085D1B6C7A>". Supported: gruu, path, timer, 100rel, replaces. User-Agent: Aastra 57i/2.5.2.1010. Content-Type: application/sdp. Content-Length: 285. . v=0. o=MxSIP 0 0 IN IP4 10.0.0.220. s=SIP Call. c=IN IP4 10.0.0.220. t=0 0. m=audio 3000 RTP/AVP 18 0 8 101. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=silenceSupp:off - - - -. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Server: kamailio (3.0.0-rc3 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.211:5060 "Noisy feedback tells: pid=9830 req_src_ip=10.0.0.220 req_src_port=5060 in_uri=sip:5501122334@10.0.0.211:5060 out_uri=sip:5501122334@10.0.0.152:5062 via_cnt==1". . U 10.0.0.211:5060 -> 10.0.0.152:5062 INVITE sip:5501122334@10.0.0.152:5062 SIP/2.0. Record-Route: <sip:10.0.0.211;lr=on>. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 69. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.220:5060;transport=udp>;+sip.instance="<urn:uuid: 00000000-0000-1000-8000-00085D1B6C7A>". Supported: gruu, path, timer, 100rel, replaces. User-Agent: Aastra 57i/2.5.2.1010. Content-Type: application/sdp. Content-Length: 285. . v=0. o=MxSIP 0 0 IN IP4 10.0.0.220. s=SIP Call. c=IN IP4 10.0.0.220. t=0 0. m=audio 3000 RTP/AVP 18 0 8 101. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=silenceSupp:off - - - -. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. User-Agent: T20 9.41.0.70. Content-Length: 0. . U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Record-Route: <sip:10.0.0.211;lr=on>. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Contact: <sip:5501122334@10.0.0.152:5062>. User-Agent: T20 9.41.0.70. Allow-Events: talk,hold,conference,refer,check-sync. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Record-Route: <sip:10.0.0.211;lr=on>. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Contact: <sip:5501122334@10.0.0.152:5062>. User-Agent: T20 9.41.0.70. Allow-Events: talk,hold,conference,refer,check-sync. Content-Length: 0. . U 10.0.0.220:5060 -> 10.0.0.211:5060 CANCEL sip:5501122334@10.0.0.211:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 70. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. Supported: gruu, path. User-Agent: Aastra 57i/2.5.2.1010. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.152:5062 CANCEL sip:5501122334@10.0.0.152:5062 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Max-Forwards: 69. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 200 canceling. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cd 47. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. Server: kamailio (3.0.0-rc3 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.211:5060 "Noisy feedback tells: pid=9831 req_src_ip=10.0.0.220 req_src_port=5060 in_uri=sip:5501122334@10.0.0.211:5060 out_uri=sip:5501122334@10.0.0.211:5060 via_cnt==1". . U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. User-Agent: T20 9.41.0.70. Content-Length: 0. . U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. User-Agent: T20 9.41.0.70. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.152:5062 ACK sip:5501122334@10.0.0.152:5062 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Max-Forwards: 69. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 ACK. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM). Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cd 47. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Server: kamailio (3.0.0-rc3 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.211:5060 "Noisy feedback tells: pid=9830 req_src_ip=10.0.0.220 req_src_port=5060 in_uri=sip:5501122334@10.0.0.211:5060 out_uri=sip:5501122334@10.0.0.152:5062 via_cnt==2". . U 10.0.0.220:5060 -> 10.0.0.211:5060 ACK sip:5501122334@10.0.0.211:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 70. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cd 47. Call-ID: c8e55b952ba85701. CSeq: 13486 ACK. User-Agent: Aastra 57i/2.5.2.1010. Content-Length: 0. . Ricardo.-

3 9

[Kamailio-Users] kamailio/ sip-router at 26c3
by Henning Westerholt 24 Dec '09

24 Dec '09

Hi, I'll (like in the last years) participate in the annual congress of the CCC in Berlin next week - the 26c3. More informations can be found on their site: http://events.ccc.de/congress/2009/wiki/Welcome So if you are in Berlin as well in this week (not necessarly on this event) and are interested in meeting for some discussions, please drop me a mail. Best regards, Henning

1 0

[Serusers] How to remove npdi & rn parameters from INVITE messages?
by Frank Durda IV 24 Dec '09

24 Dec '09

I have another large carrier that I am trying to send calls to, and they have a requirement that you not send them the npdi and rn=nnnnnnnn parameters for all calls. If you do, the call handling will be botched and the call will fail, even if the LRN information is exactly correct. Apparently this carrier (who once owned the Death Star) is incapable of filtering or ignoring this information on their side, and the sending switch is incapable of not sending this information because they thought RFC 4694 said they MUST ALWAYS send npdi/rn on every SIP call. Of course, the RFC does not say this. Don't get me started on which party is dumber. Anyway, I was hoping that SER could act as diplomat or interpreter and with some rewrite-something() calls, SER could remove the npdi/rn, or alter them to the point that the receiving switch no longer recognized the parameters as rn/npdi and so happily does it's own thing, and the calls don't die. However, based on what documentation exists, none of the various rewrite functions appear to operate on this part of the INVITE message and replace doesn't appear to work on the first line of a message. So there seems to be a spot whose contents can't be reached or manipulated by the existing functions. Am I correct in thinking this? So, is there a way to do something with rn=NNNNNNNNNN and npdi/npdi=yes parameters without having to write a custom function to do it? Thanks in advance!

2 1

[Serusers] How to drop a redundant/broken SIP reply message?
by Frank Durda IV 24 Dec '09

24 Dec '09

I have a large carrier that I am trying to send calls to, and their major brand SIP/ISUP switch (whose name begins with a "S") has a quirk where it emits a 183 Session Progress message with no SDP payload immediately after 100 Trying, and before ACM has actually been received. When ACM is received, it emits either a 180 Ringing with a SDP payload, or a 183 Session Progress with a SDP payload. So INVITE 100 Trying 183 Session Progress (no SDP payload) 180 Ringing (or 183 Session Progress) (with SDP payload) 200 OK (with SDP payload) is what you normally get, with the two 18x messages coming one or two seconds apart. However, if the called number is busy, you get 100 Trying 183 Session Progress (no SDP payload) 486 Busy Here What the calling party experiences for that is a second or two of ring-back commencing with the 183 Session progress, then a busy signal. That's very non-compliant. So far, the carrier has been unable to get this switch to stop emitting the first 183 Session Progress message. What I would like to do is have SER detect the redundant 183 Session Progress message (which I can already detect), and then have SER discard it, eg not send that one message back to whomever the INVITE came from. So far, I have been unable to get SER to do this. While "drop" is allowed in onreply_route sections, it is silently ignored, and you get a reply anyway. I tried putting a route(666) in the onreply_route section and putting the drop in that route["666"] section, but that had no obvious effect either, although some other functions can now be used in route["666"] that so far aren't solving the problem either. It occurred to me that I could change the 183 Session Progress of the offending message to something else in the "so-what" department, such as a "181 Your Call Is Being Forwarded", but "replace()" doesn't seem to work on the first line of replies, but can certainly alter other things in the reply message. There also doesn't seem to be any way (that actually works) to change the numeric SIP Response Code value to something else, even though clearly SER has it in the variable "status". I even thought of changing the port number or IP address via a rewrite command and send the message off to nowhere, but those don't work on replies or don't appear to. So, any suggestions on how to simply drop a chosen reply message, or misdirect it or malform it to the point that the call originator doesn't recognize it anymore? Yeah, kludgy but I only need it to work for two months. Thanks in advance!

2 2

[Serusers] Ser2.1.0.
by Aleš Podskalský 23 Dec '09

23 Dec '09

Hi all, I tried to install the newest snapshot of the Ser (version 2.1.0+cvs20091126). Server has been instaled correctly, but I can't add new user to the database. I noticed that serctl was changed with new tool called sercmd. With this tool I should be able to create new user using appropriate command, but I can't find some help for it. So my question is, if this version is unable for normal use by then, or if I can fill user data in some another way and how. Thank a lot. Ales P.

2 1

[Kamailio-Users] adding inbound and outbound routes seperately
by Rodney 23 Dec '09

23 Dec '09

Ok, I have kamailio and siremis, and I need to know how to set up a separate in bound and outbound route. Peering with inbound and registering with outbound Any ideas guys? R

3 2

Re: [Kamailio-Users] PCRE Regex / Transformations Bug? (Daniel-Constantin Mierla)
by Brandon Armstead 23 Dec '09

23 Dec '09

Inaki: That is what I was looking for and it does appear you are correct! Daniel, I could see that being as a useful addition, do you imagine there being much involved to implement such a method? Thank you guys for all of your help! Sincerely, Brandon Armstead

1 0

[Kamailio-Users] Question about Contact in NAT
by Ricardo Martinez 22 Dec '09

22 Dec '09

Hello list. I know that probably this questions is more related to the SIP protocol, but is causing me problems and this is a very supportive list. I'm using a HT-286 from Grandstream behind NAT. This is the REGISTER sequence : U 200.120.46.99:29786 -> 200.100.154.35:5060 REGISTER sip:myproxy.mydomain.net SIP/2.0. Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKafcb21cb60db91bb. From: <sip:5501234567@myproxy.mydomain.net;user=phone>;tag=478555f337ca8f10. To: <sip:5501234567@myproxy.mydomain.net;user=phone>. Contact: *. Supported: replaces, timer. Call-ID: a51e541e3b0f66fd(a)192.168.1.100. CSeq: 100 REGISTER. Expires: 0. User-Agent: Grandstream HT287 1.1.0.42 DevId 000b822197fe. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE. Content-Length: 0. . U 200.100.154.35:5060 -> 200.120.46.99:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKafcb21cb60db91bb;received=200.120.46.99. From: <sip:5501234567@myproxy.mydomain.net;user=phone>;tag=478555f337ca8f10. To: <sip:5501234567@myproxy.mydomain.net;user=phone>. Call-ID: a51e541e3b0f66fd(a)192.168.1.100. CSeq: 100 REGISTER. Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 200.100.154.35:5060 "Noisy feedback tells: pid=29777 req_src_ip=200.120.46.99 req_src_port=29786 in_uri=sip:myproxy.mydomain.net out_uri=sip:myproxy.mydomain.net via_cnt==1". . U 200.100.154.35:5060 -> 200.120.46.99:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKafcb21cb60db91bb;received=200.120.46.99. From: <sip:5501234567@myproxy.mydomain.net;user=phone>;tag=478555f337ca8f10. To: <sip:5501234567@myproxy.mydomain.net;user=phone>;tag=6eb14299de2cc731d17 3c135064daec5.1fe9. Call-ID: a51e541e3b0f66fd(a)192.168.1.100. CSeq: 100 REGISTER. WWW-Authenticate: Digest realm="myproxy.mydomain.net", nonce="4b30e9281c52f640a2713c3011bb835d06005946", qop="auth". Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 200.100.154.35:5060 "Noisy feedback tells: pid=29777 req_src_ip=200.120.46.99 req_src_port=29786 in_uri=sip:myproxy.mydomain.net out_uri=sip:myproxy.mydomain.net via_cnt==1". As you can see the initial REGISTER has a Contact header like this : Contact: *. Is this ok?. Is this ok by RFC? What does this means? In my cfg file I have : route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; So I think is not matching the regexp "(!search("^Contact:\ +\*")" therefore is not fixing the contact. Any ideas? Ricardo.-

2 3

[Kamailio-Users] append_hf
by Corey Edwards 22 Dec '09

22 Dec '09

I'm upgrading from OpenSER 1.2.2 to Kamailio 1.5.3 and I've run into a problem with append_hf. It seems that my header only gets appended once in a while. Here's the shortened version of what I'm doing: failure_route[1]{ if (t_check_status("(486)|(600)")){ # busy handler route(15); } } route[15]{ # look up forwarding number .... if (does_uri_exist()){ append_hf("Diversion: <sip:$var(orig_user)@$Ri>;user-id=\"$var(orig_user)@$Ri\";reason=busy\r\n"); t_relay(); } } Unfortunately, the Diversion header doesn't get set every time. I can't see any difference in the logs from when it does get set and when it doesn't. I added an xlog right above append_hf and confirmed that append_hf runs each time. On a whim I added a branch_route and called it using t_on_branch. I then added duplicate code to call append_hf. I now seem to be getting more instances of Diversion being set (twice!), but still not every time. Is there something obvious I'm missing here? Any way I can ensure those headers get set every time? Corey

1 1

[Kamailio-Users] listing users that were added
by Rafi Gordon 22 Dec '09

22 Dec '09

Hello, I had added successfully several users by : kamctl add xxx yyy How can I list all users that exist in the DB? Rgs, Rafi

3 5

[Kamailio-Users] Question about AVP in route()
by Ricardo Martinez 21 Dec '09

21 Dec '09

Hello. I have the next situation. In my config file I have something like this.. route { ... If INVITE route(3); } ... route[1] { Route(39); t_relay() } route[3] { ... different check for the INVITE radius_load_caller_avps("$fU@$fd") (I load the next AVPS from the radius server : Attributes: SIP-AVP = "tra:sip:0217001234567@10.0.0.211:5060" SIP-AVP = "channels:1") is_avp_set("$avp(s:channels)/n") <-- this AVP seem to be load and is set OK. route(10); } route[10] { rewritehost("myproxy.mydomain.com"); route(1); } route[39] { is_avp_set("$avp(s:channels)/n") } The last check in the route(39) is not working!. Why? Ricardo Martinez Ingeniero de Desarrollo VOISSNET S.A.

2 2

[Kamailio-Users] Presence_Dialoginfo versioning
by kamailio.org＠spam.lublink.net 21 Dec '09

21 Dec '09

Hey, I have noticed that the XML body of the dialog messages contains a version attribute. The server is counting the versions using the latest subscription as a reference point, and the phone is counting the versions from the first subscription ( at reboot ). Which is the correct way to count these versions? Consider : 10:00 Subscription 10:05 Notification version 1 10:35 Notification version 2 11:01 Subscription 11:05 Notification version X Is X = 3 because it is the third notification or 1 because it is the first after the last subscription? If it's version 1, it could confuse the phone cause a notification that is sent at the same time as the notification would have a confusing version. On the other hand, each subscription, has it's own versioning, so would it not logically follow that the different subscriptions for the same device have seperate versioning? From my phone : BLF Notify received for line: 3 has older version: 3 last version:135 To confirm my theory, I redialed the number 133 times, and once the version number had run up to 136, the lights started flashing again. Is this a bug in Grandstream, Kamailio, or perhaps an RFC which was unclear? Thanks, David

7 23

[Kamailio-Users] Clients with late offer-answer
by Klaus Darilion 21 Dec '09

21 Dec '09

Hi! I want to test rtpproxy with late offer-answer (SDP in 200-ok and ACK). Thus, I'm looking for a SIP client which support that. Do you know any? thanks klaus

3 2

[Kamailio-Users] add Allow header to sl_send_reply
by Andreas Heise 19 Dec '09

19 Dec '09

Hello, if I create a response with sl_send_reply then the message contains not an Allow header. How could I add an Allow header to this responses? thanks in advance, Andreas

4 10

[Kamailio-Users] log not showing when using uac_replace_from( ) function
by Robert R 19 Dec '09

19 Dec '09

Hi, I simply trying to modify URI and FROM headers as follow. tcpdump packet capture shows the values are changed for both URI and FROM headers. However, the syslog only shows the new value for URI and not for FROM header. How can I verify new value in FROM header in the log file? # main request routing logic route { $rU = avp(i:1); $rd = avp(i:2); .... uac_replace_from("$avp(i:3)"); xlog ("fu after perl: $fu"); xlog ("ru after perl: $ru"); ... } Thanks, Rob

2 1

[Kamailio-Users] Transparent bridge mode
by Sungjoo Kim 17 Dec '09

17 Dec '09

Hello All, I'm trying to setup a test scenario, where i have Kamailio and rtpproxy running on one CentOS box(server1) and asterisk running on another CentOS box(server2). Server1 has 2 NIC's installed, eth0 connected to the WAN and eth1 connected to Server2 in a bridge mode. I will be modifying rtpproxy to intercept data flowing to Server2 and process them before i let them continue to flow along their original path. Server2 has to have a Public IP. For this i am having to use 4 Public IP's- eth0,eth1 and bridge br0 on Server1 and eth0 on Server2. Is there an alternate way of doing this ? Do you suspect there will be a problem with setting things up this way while trying to use rtpproxy to intercept data. Thanks and regards, Vikram.

1 0

[Kamailio-Users] Kamailio + CSQL
by Ramu 17 Dec '09

17 Dec '09

Hi All, Did anybody tried to setup Kamailio with CSQL database. Please post your experiences using CSQL and what are parameters we need to change in modules to connect CSQL DB Server. Thanks, Ramu

2 2

[Kamailio-Users] Kamailio and RTPProxy
by Bruce McAlister 16 Dec '09

16 Dec '09

Hi All, I am trying to setup kamailio 1.5.3-notls for use with rtpproxy 1.2.1. I have a partially working setup, however, when I have a softphone configured behind nat it appears that rtpproxy is using the internal ip (prior to nat) to setup the sdp ports. I have rtpproxy configured in bridge mode as follows: rtpproxy -f -F -l public_ip/internal_ip -s udp:*:11211 I have kamailio's nathelper module setup as follows: modparam("nathelper", "rtpproxy_sock", "udp:internal_ip:11211") I then use a softphone (x-lite) behind a nat firewall (ip = 192.168.11.2) to establish a call, and the log of rtpproxy is as follows: INFO:handle_command: new session ZmRkNjRlNjM0NzExZjQ5MWUxYjllOTFhZWFmZGZkMjY., tag 8330585e;1 requested, type strong INFO:handle_command: new session on a port 52572 created, tag 8330585e;1 INFO:handle_command: pre-filling caller's address with 192.168.11.2:35498 INFO:handle_command: lookup on ports 52572/40380, session timer restarted INFO:handle_command: pre-filling callee's address with internal_ip:11250 INFO:handle_delete: forcefully deleting session 1 on ports 52572/40380 INFO:remove_session: RTP stats: 906 in from callee, 0 in from caller, 906 relayed, 0 dropped INFO:remove_session: RTCP stats: 3 in from callee, 0 in from caller, 3 relayed, 0 dropped INFO:remove_session: session on ports 52572/40380 is cleaned up In the log above it looks like rtpproxy is using 192.168.11.2:35498 as the remote ip/port combination to contact for the sdp session of the x-lite softphone. I would have thought that it would use the IP address that initial request was received from to establish this connection (via the fix_nated_contact() function). My route logic in kamailio is a slightly modified default configuration that is installed with kamailio : route{ . . . # NAT detection route(4); . . . route(1); } route[1]{ if (check_route_param("nat=yes")) { setbflag(6); } # if (isflagset(5) || isbflagset(6)) { route(5); # } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } Would the "fix_nated_contact" function in the "nat detection" route not fix the contact before passing over the request to rtpproxy to establish the sdp session? route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ # force_rtp_proxy(); if(src_ip==internal_ip_of_media_server) force_rtp_proxy("ocfaei"); else force_rtp_proxy("ocfaie"); } if (!has_totag()) add_rr_param(";nat=yes"); return; } onreply_route[1] { xdbg("incoming reply\n"); # if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { if (status=~"(183)|(2[0-9][0-9])") { # force_rtp_proxy(); if(src_ip == internal_ip_of_media_server) force_rtp_proxy("ocfaei"); else force_rtp_proxy("ocfaie"); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } } Any pointers would be greatly appreciated. Thanks Bruce

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[Kamailio-Users] Problem
by Ricardo Martinez 15 Dec '09

15 Dec '09

Hello. What this error means? Dec 14 15:20:19 kamailio: ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/nat_traversal.so>: /usr/local/lib/kamailio/modules_k/nat_traversal.so: undefined symbol: register_module_stats Dec 14 15:20:19 kamailio: : <core> [cfg.y:3210]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 48, column 12-29: failed to load module I have : # ------------------ module loading ---------------------------------- #set module path mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/module s/" # Uncomment this if you want to use SQL database loadmodule "db_mysql.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "siputils.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "domain.so" loadmodule "permissions.so" loadmodule "group.so" loadmodule "mi_fifo.so" loadmodule "avpops.so" loadmodule "xlog.so" loadmodule "dialog.so" loadmodule "nathelper.so" loadmodule "nat_traversal.so" loadmodule "mediaproxy.so" I'm using Kamailio version: kamailio 3.0.0-rc2 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. @(#) $Id$ main.c compiled on 18:09:39 Dec 9 2009 with gcc 4.3.2 In the directory /usr/local/lib/kamailio/modules_K/ the file nat_traversal.so exists : [root@ nat_traversal]# ls /usr/local/lib/kamailio/modules_k/ acc.so cfgutils.so drouting.so mi_datagram.so permissions.so registrar.so speeddial.so uac.so alias_db.so db_text.so exec.so mi_fifo.so pike.so rr.so sqlops.so uri_db.so auth_db.so dialog.so group.so msilo.so presence_mwi.so rtimer.so sst.so userblacklist.so auth_diameter.so dispatcher.so htable.so nathelper.so pua_mi.so siptrace.so statistics.so usrloc.so auth.so diversion.so imc.so nat_traversal.so pv.so siputils.so textops.so xlog.so benchmark.so domainpolicy.so kex.so path.so qos.so sl.so tmx.so call_control.so domain.so maxfwd.so pdt.so ratelimit.so sms.so uac_redirect.so Can someone help me? Regards, Ricardo

2 3

Re: [Kamailio-Users] kamailio load problem
by Daniel-Constantin Mierla 15 Dec '09

15 Dec '09

Hello, cc-ed back the list as others may look for same issues in the future ... On 12/15/09 11:30 AM, alex pappas wrote: > Daniel hi, > > In my config I don't have any authentication. I also configured in > mysql the log-slow-query with long_query_time = 2 > In my kamailio.cfg is a lot of ACC loging: > > Log acc to a file (extra fields) > Log to mysql with axtra fields(I disabled this for the moment because > was causing high load) > Log to a Radius (the radius is in the same machine and as far as I > know the radius after i swriting his data to a mysql also) I diden't > know that we have radius installed localy > > From the log-slow-query now I see a message about sip_trace query wich > obiesly took time more than 2 sec. Now I know that I'm doing also > sip_trace inserts in mysql sip_trace is writing big amount of data, this can make mysql do a lot of work. Try without it and see the difference. Also, doing acc to syslog brings some penalty if syslog is not configured in asyncronous mode (see my previous email). Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

Re: [Kamailio-Users] kamailio load problem
by Daniel-Constantin Mierla 15 Dec '09

15 Dec '09

Hello, cc-ed back the list as others may look for same issues in the future ... On 12/15/09 11:30 AM, alex pappas wrote: > Daniel hi, > > In my config I don't have any authentication. I also configured in > mysql the log-slow-query with long_query_time = 2 > In my kamailio.cfg is a lot of ACC loging: > > Log acc to a file (extra fields) > Log to mysql with axtra fields(I disabled this for the moment because > was causing high load) > Log to a Radius (the radius is in the same machine and as far as I > know the radius after i swriting his data to a mysql also) I diden't > know that we have radius installed localy > > From the log-slow-query now I see a message about sip_trace query wich > obiesly took time more than 2 sec. Now I know that I'm doing also > sip_trace inserts in mysql sip_trace is writing big amount of data, this can make mysql do a lot of work. Try without it and see the difference. Also, doing acc to syslog brings some penalty if syslog is not configured in asyncronous mode (see my previous email). Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] debian packages for kamailio 3.0.0-rc3
by Daniel-Constantin Mierla 15 Dec '09

15 Dec '09

Hello, I have uploaded debian packages for stable version: http://www.kamailio.org/pub/kamailio/3.0.0-rc3/packages/debian-lenny/ If you want to create debs by yourself, get Kamailio 3.0 branch from GIT, see here: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git And then run: make deb Testing and feedback is very much appreciated. Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] kamailio load problem
by alex pappas 15 Dec '09

15 Dec '09

Dear users, I'm using kamailio in the following scenario: 1. A call comes in and is checked against PDT module. From this module depend from the prefix I'm getting back the domain which is used as the dialplan group. 2. After that the call is checked against Dialplan module using the previous value from PDT as Dialplan dpid. From the Dialplan module i get back the the attributes values. 3. these attributes values are user in the dispathcher module where I continue with failover scenarios. Also the script is working with a radius for accounting purposes. This scenario works perfecttly and it is still in a testing mode. An the problem: I'm running SIPP against this scenario and until the CPS is 5 everything looks good. The scenario for SIPP is the simplest. uac and uas default. Uac is sending a call and uas is answering. When the CPS goes above 9 then I have call failures. All the statistics are going down. I'm cheking for CPU load or other system problems but I don't see anything strange. Does anyone understand why this is happening? I'm attachng the 2 routes that i'm using: test values in order to understand the script *PDT value = 1 (used for the dialplan group) Dialplan value = 100.101.102.# Dispatcher values: 100 = 10.0.0.5 101 = 10.0.0.6 102 = 10.0.0.7* Thank you in advance Alex route[7] { if(prefix2domain("2", "0")) { $var(dial_grp) = $(rd{s.int}); if(dp_translate("$var(dial_ grp)", "$rU/$rU")) { $var(i) = 0; while($(avp(s:dest){s.select,$var(i),.})!="#") { $avp(s:dstgrp) = $(avp(s:dest){s.select,$var(i),.}{s.int}); $var(i) = $var(i) + 1; } $avp(s:user) = $rU; if(ds_select_domain("$avp(s:dstgrp)", "4")) { if($(ru{uri.param,prefix})!=null) { $ru = "sip:" + $(ru{uri.param,prefix}) + $rU + "@" + $rd; } else { $ru = "sip:" + $rU + "@" + $rd; } $avp(s:dstgrp) = null; t_on_failure("2"); t_relay(); exit; } } sl_send_reply("404", "No route found"); exit; } } failure_route[2] { if(t_was_cancelled()) { exit; } if(t_check_status("4[0-9][0-9]|5[0-9][0-9]")) { if(ds_next_domain()) { if($(ru{uri.param,prefix})!=null) { $ru = "sip:" + $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; } else { $ru = "sip:" + $avp(s:user) + "@" + $rd; } t_on_failure("2"); append_branch(); t_relay(); exit; } else { if($avp(s:dstgrp)!=null) { if(ds_select_domain("$avp(s:dstgrp)", "4")) { if($(ru{uri.param,prefix})!=null) { $ru = "sip:" + $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; } else { $ru = "sip:" + $avp(s:user) + "@" + $rd; } $avp(s:dstgrp) = null; t_on_failure("2"); append_branch(); t_relay(); exit; } } else { t_reply("444", "No more tries for you!"); } } }

2 6

[Kamailio-Users] Reboot phones
by David 14 Dec '09

14 Dec '09

Hey, I have noticed that when I try to use kamctl to reboot phones, it comes back with simply "400". I tried using the fifo interface and the phone asked the server to identify itself using the phone's username and password. How do people generally use the kamctl to reboot phones? Thanks, David

3 2

[Kamailio-Users] Kamailio 3.0.0-RC3 Released
by Daniel-Constantin Mierla 14 Dec '09

14 Dec '09

Kamailio v3.0.0-RC3 is out - the tarball with sources is available at: http://www.kamailio.org/pub/kamailio/3.0.0-rc3/src/kamailio-3.0.0-rc3_src.t… Lot of fixes to packaging, code and documentation have been committed since RC1, Kamailio 3.0.0-RC3 becoming ready for pre-production phase. If you like to work with sources, then use the tutorial guiding the installation of Kamailio 3.0 branch from GIT repository: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git Two wiki pages were created to collect what is new in Kamailio 3.0 and how to migrate to this version: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 Feel free to contribute to wiki pages, helping to build migration tutorials. This is the last RC before full 3.0.0 relase. Please report any issue you find to sr-dev(a)lists.sip-router.org Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] preparing RC3 for kamailio 3.0
by Daniel-Constantin Mierla 14 Dec '09

14 Dec '09

Hello, I plan to release Kamailio 3.0, hopefully today, if no outstanding issue out there. Ending of last week, I managed to get debian packaging working (tested on Ubuntu) using Andrei's new Makefile system, with module groups. If you are aware of issues you want fixed before 3.0RC3, drop an email. This is supposed to be latest RC before full 3.0.0 release. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] Bflags
by David 14 Dec '09

14 Dec '09

Hey, I am having some issues with NAT and I think the issue is with Bflags. So I have some clarifications : 1. What does the flags column in location contain? 2. What does the cflags column in location contain ? I am guessing that cflags represents the bflags? So if I have a value of 192, does that mean flags 7 and 8 are set ? This confuses me cause I set flags 6 and 7. Do the bflags appear magically in the routing scripts or is there a command that I have to execute for the script to have the bflags? Thanks! David

5 8

What to download?
by Antonio Goméz Soto 13 Dec '09

13 Dec '09

Hi, I'm a longtime asterisk user, needing to scale up. I figured it would be smart to use and develop for sip-router, because it'll come out soon. But I've never used one of ser, openser, or siblings, and now I'm confused by the download links on you site. I see kamailio_3.0, master, sr_3.0, stable and unstable top level directories. What should I download? What's the diference between them? Thanks, Antonio

2 2

[Serusers] Warning 392 : Noisy Feedback Tells
by Abdul Khadar 11 Dec '09

11 Dec '09

Hi, I need help for registering with iptel.org I have created one user called hitechabdul with iptel.org When I am trying to register with this domain, It is returing error *401 Unauthorized *with *Warning 392 : Noisy Feedback Tells* please find attached ethereal packet and help me to resolve this problem. The details filling with register message : username : hitechabdul password : xxxxxx Domain Name : iptel.org Other than this Is I have to given any more data ? -- thanks & regards, Abdul Khadar Shaik,

2 1

[Kamailio-Users] Rewrite $tU
by David 11 Dec '09

11 Dec '09

Hey, I need to rewrite the $tU variable on a packet before it is relayed. It is ok if the change is only on the relayed packet and the $tU variable change. The important thing is that I modify the To header, it seems that rewritting $tU would be the easiest. I know that user agents do not like this changing, but I am changing it on an intermediary server and when the reply comes back, I change it back to the original value. Thanks! David

5 6

Case sensitivity in usrloc "location" table
by Bruce McAlister 10 Dec '09

10 Dec '09

Hi All, I am trying to integrate Kamailio (1.5.2-notls) into our existing database structure using a view for the subscriber table and using the location table to store the registration information. I am successfully able to register a phone using this method. One of our requirements is to set an "outboundproxy" and "outboundproxyport" parameter in the location table for a particular sip account (username). I am trying to do this with the sqlops module. What I have come across though is that when the registration information is saved to the location table the username is stored in lower case. So when my sqlops query is run against the username it does not match. Is there a way to tell the usrloc module to store the username in the same case as it is received? For example, in the subscriber table I have view defined that for my user account returns the following information: ------------- select * from subscriber where username = 'BruceNW9440'; username | domain | password | ha1 | ha1b -------------+---------------+------------+----------------------------------+---------------------------------- BruceNW9440 | 10.10.10.10 | 00minger00 | 1f2d688f65be04a61cab36a1fb3f426a | 6529d45f5c418daf35be7b3dd4a858cb (1 row) ------------- Now, if I check the above against the location table: -------------- select * from location where username = 'BruceNW9440'; 0 rows returned -------------- If I run the same query using "brucenw9440" as the username (lowercase) then the correct information is returned. I have the following log line in my kamailio configuration script: xlog("L_INFO","Username: $au, OutboundProxy: $Ri, OutboundProxyPort: $Rp"); The details of this log appear correct when it is logged: Dec 9 14:21:03 ibis /opt/kamailio/sbin/kamailio[16972]: [ID 197553 local0.info] Username: BruceNW9440, OutboundProxy: 10.10.10.10, OutboundProxyPort: 5060 However, when my sqlopt query is run it does not find a match because of the incorrect case: sql_query("bfl", "UPDATE location SET outboundproxy = '$Ri', outboundproxyport = '$Rp' WHERE username = '$au'", "res"); I've had a read through the usrloc module documentation and I cannot seem to find a reference to any options that will preserve the case. I also cannot find if there is a parameter to save('location') that will preserve the case. I cant seem to find a reference to the save() function on the wiki. I must be missing something small. We have some sip usernames that are case sensitive so any help in being able to resolve this would be greatly appreciated. Thanks Bruce

4 9

[Kamailio-Users] Question regarding TM module DNS failover with SRV records
by Nathan Angelacos 10 Dec '09

10 Dec '09

According to the 1.5.x TM module documentation, DNS failover is triggered by a 503 reply or a timeout. When using SRV records, it appears a 503 triggers the failure_route instead. Given: host -t srv _sip._udp.voip.example.net _sip._udp.voip.example.net has SRV record 1 0 5060 fast-server.example.net. _sip._udp.voip.example.net has SRV record 2 0 5060 slow-server.example.net. On an invite to voip.example.net, if fast-server does not respond within "fr_timer" seconds, t_relay fires off a serial branch (t_on_branch is triggered) and the invite goes to slow-server. If fast-server responds with a 503, the t_on_failure route is triggered instead, and the invite is not sent on to slow-server. If its not a bug in the module, does anyone have clues of how to coax the next DNS SRV record out of the t_on_failure route? Or what can be done to get t_relay to consume the 503? Thanks!

3 7

[Kamailio-Users] ldap_search get result URL
by Andrey Utkin 10 Dec '09

10 Dec '09

How can i access LDAP URI of ldap_search() results in routing script? I need to do recursive search in company directory within persons that belong to groups, i need to get group id without storing groupid as person attribute, but as an element of resulting URL. Query base name: dc=example,dc=com. Query result is cn=user1,ou=group1,dc=example,dc=com. I can easily get user1 attributes, but how to get ou?

3 3

[Kamailio-Users] kam 1.5 dispatcher priority usage
by Robert McGilvray 10 Dec '09

10 Dec '09

What is the function of the priority field in dispatcher module? The docs are pretty light on the subject and it's not clear to me what it's meant to do. I'm looking for a way to assign a priority to preferred gateways within the same group. Thanks Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

4 4

[Kamailio-Users] Siremis login page once and again.
by Vicente 10 Dec '09

10 Dec '09

Hello, I've installed siremis 0.9.4 with kamailio 3.0.0rc2 on a test computer and it was working without a problem. But now I'm trying to install it on a server and I can't go beyond the login screen. If I try a wrong password I get the error message. But if I use the right one I'm redirected to the login screen again and again. Both systems are ubuntu Linux with the following differences: Test computer Server Ubuntu 9.10 Ubuntu 8.10 apache 2.2.12 apache2.2.9 PHP 5.2.10 PHP 5.2.6 mysqld 5.1.37 mysqld 5.0.67 The bigger difference that I can see is mysqld version.But I've installed PHPOpenBiz framwork and the demo app creates and populates the demo data base. So it doesn't look like related to this. I've used the apache configuration on the siremis install page to set up apache for both systems. Although on the server there are some previous settings, I've created a new virtual server for siremis. apache logs only show: File does not exist: /var/www/siremis-0.9.4/siremis-web/js/sound.js But this error also appears on the working computer. And doesn't seem to be related with the problem. The log_DEBUG.html on siremis only shows the query to the database. ¿ Is there any log file on siremis that can point to the problem ? ¿ Has anyone else had this problem ? Sorry if I ask something silly but I've been trying to find what I'm doing wrong for two days. Best regards. Vicente.

2 1

Perl Module Logging Query
by Bruce McAlister 09 Dec '09

09 Dec '09

Hi All, I have been trying to use the perl module with Kamailio 1.5.3-notls. I have managed to load the module successfully. I have a simple script with a single function that strips the "<" or ">" from a contact header and logs the info, then sets an AVP. Kamailio seems to load the perl module and the script properly, but when it runs I get an error on the logging side of it. I cannot for the life of me see where the error lies. Could one of you guru's out there please cast an eye over my config and see if you can spot something I am missing. My kamailio configuration for perl is as such: ---- loadmodule "perl.so" modparam("perl", "modpath", "/opt/kamailio/lib64/perl") modparam("perl", "filename", "/opt/kamailio/perl-scripts/kamailio-perl.pl") ---- My perl script looks as follows: ---- use OpenSER ; use OpenSER::Message ; sub clean_contact { my $contact_header = shift; log("contact header passed: $contact_header\n"); $contact_header =~ s/^<|>$//g; log("contact header fixed: $contact_header\n"); OpenSER::AVP::add("clnct", "$contact_header"); return 1; } ---- I have the following section in the route logic that calls this function as follows: ---- perl_exec_simple("clean_contact","$ct"); ---- When the perl script executes, it gets to the first log line and dies with the following message: ---- ERROR:core:XS_OpenSER__Message_log: perl error: Can't take log of 0 at /opt/kamailio/perl-scripts/kamailio-perl.pl line 7. ---- I have had a look at the logging.pl script in the sample directory for the perl module, and have tried all the methods shown in the example, but I just cannot seem to get past this particular logging issue. Any tips would be greatly appreciated. Thanks Bruce

1 1

[Kamailio-Users] Problem with REGISTER
by Uriel Rozenbaum 09 Dec '09

09 Dec '09

Hello guys, I'm having a strange problem with my Kamailio 1.5.3 when trying to register an Audiocodes Gateway. I'm using authentication for REGISTER messages. The GW sends a REGISTER without credentials, so I do a proxy_challenge. Then the GW sends a REGISTER with credentials but the proxy_authorize is returning -4. I did an NGREP to check the messages and they don't seem malformed, can you tell me if there is something on the functions that might be loosing the Proxy-Authorization header? This happens only with this GW, not if i register a softphone (XLITE). Within route block: ... if (is_method("REGISTER")) { route(11); save("location"); exit(); } ... Route(11): route[11] { # ----- AUTENTICACION DE UN MENSAJE ----- if (!proxy_authorize("", "subscriber")) { switch($retcode) { case -1: xlog("L_INFO", "AUTH: Invalid username\n"); sl_send_reply("403", "Forbidden"); exit; case -2: xlog("L_INFO", "AUTH: Invalid password\n"); sl_send_reply("403", "Forbidden"); exit; case -3: xlog("L_INFO", "AUTH: Nonce ambiguo\n"); break; case -4: xlog("L_INFO", "AUTH: No credentials\n"); break; #default: #xlog("L_INFO", "AUTH: Unknown error\n"); #break; } #xlog("L_DBG", "AUTH: -Data- Auth Username = $au\n"); #xlog("L_DBG", "AUTH: -Data- Domain = $fd\n"); #xlog("L_DBG", "AUTH: -Data- From Username = $fU\n"); proxy_challenge("192.168.200.130","0"); exit; }; xlog("L_INFO", "AUTH: Valid username and credentials\n"); consume_credentials(); } NGREP: # U 10.0.1.200:5060 -> 192.168.200.130:5060 REGISTER sip:192.168.200.130 SIP/2.0. Via: SIP/2.0/UDP 10.0.1.200;branch=z9hG4bKac26395445. Max-Forwards: 70. From: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200> >;tag=1c26389068. To: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200>>. Call-ID: 154992670511200004613(a)10.0.1.200. CSeq: 68 REGISTER. Contact: <sip:testdevice@10.0.1.200:5060>;expires=180. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Expires: 180. User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006. Content-Length: 0. . # U 192.168.200.130:5060 -> 10.0.1.200:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 10.0.1.200;branch=z9hG4bKac26395445. From: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200> >;tag=1c26389068. To: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200> >;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.ee1a. Call-ID: 154992670511200004613(a)10.0.1.200. CSeq: 68 REGISTER. Proxy-Authenticate: Digest realm="192.168.200.130", nonce="4b142e8900000000188436d32ee472791b706f36952f4162". Server: Kamailio (1.5.3-notls (i386/linux)). Content-Length: 0. . # U 10.0.1.200:5060 -> 192.168.200.130:5060 REGISTER sip:192.168.200.130 SIP/2.0. Via: SIP/2.0/UDP 10.0.1.200;branch=z9hG4bKac26766380. Max-Forwards: 70. From: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200> >;tag=1c26389068. To: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200>>. Call-ID: 154992670511200004613(a)10.0.1.200. CSeq: 69 REGISTER. Proxy-Authorization: Digest username="testdevice",realm="192.168.200.130",nonce="4b142e8900000000188436d32ee472791b706f36952f4162",uri="sip:192.168.200.130",algorithm=MD5,response="9fa0904b94f94152619a16c46ffa6485". Contact: <sip:testdevice@10.0.1.200:5060>;expires=180. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Expires: 180. User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.023.006. Content-Length: 0. . # U 192.168.200.130:5060 -> 10.0.1.200:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 10.0.1.200;branch=z9hG4bKac26766380. From: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200> >;tag=1c26389068. To: <sip:testdevice@10.0.1.200 <sip%3Atestdevice(a)10.0.1.200> >;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.b7c8. Call-ID: 154992670511200004613(a)10.0.1.200. CSeq: 69 REGISTER. Proxy-Authenticate: Digest realm="192.168.200.130", nonce="4b142e8900000001515ab7891a3d534cc3f15e5627322da3". Server: Kamailio (1.5.3-notls (i386/linux)). Content-Length: 0. . Thanks in advance! Uriel

3 5

Upgrade Serweb from version 0.9.4 to version 2.0.1-dev1
by inge 09 Dec '09

09 Dec '09

Hi, I have to upgrade Serweb to the version 2.0.1-dev1 (that work with SER 2.0). Is it possible to use Serweb 0.9.4 (using PHP v.5.1.6) with SER 2.0 however the database "ser" has a new structure? Regards, David.

1 0

Build error: "no type or storage class" when building "presence" modules on solaris (pa module)
by Bruce McAlister 09 Dec '09

09 Dec '09

Hi All, I have come across another build issue when building the presence module bundle on Solaris 10. The specific error is as follows: config.mak included make[1]: Entering directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/modules_s/pa' make[1]: Leaving directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/modules_s/pa' config.mak included make[1]: Entering directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/modules_s/pa' /usr/sfw/bin/gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m32 -minline-all-stringops -falign-loops -mtune=athlon -Wall -I/usr/include/libxml2 -I/usr/local/include/libxml2 -I/usr/local/include -I../../lib -I../.. -DNAME='"ser"' -DVERSION='"2.99.99-pre3"' -DARCH='"i386"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/usr/sfw/bin/gcc 3.4.3"' -D__CPU_i386 -D__OS_solaris -DSER_VER=2099099 -DCFG_DIR='"/opt/kamailio/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -DSER -DSER_MOD_INTERFACE -DMOD_NAME='"pa"' -c async_auth.c -o async_auth.o In file included from async_auth.c:1: pa_mod.h:54: error: syntax error before "pa_dbf" pa_mod.h:54: warning: type defaults to `int' in declaration of `pa_dbf' pa_mod.h:54: warning: data definition has no type or storage class make[1]: *** [async_auth.o] Error 1 make[1]: Leaving directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/modules_s/pa' make: *** [modules_s] Error 1 I have read the INSTALL text file and it seems that the pa module requires libxml2. I have checked that this library is on the box. The version of libxml2 on Solaris is as follows: user@soldev:~ $ /usr/bin/xml2-config --version 2.6.23 user@soldev:~ $ /usr/bin/xml2-config --libs -L/usr/lib -R/usr/lib -lxml2 -lz -lpthread -lm -lsocket -lnsl user@soldev:~ $ /usr/bin/xml2-config --cflags -I/usr/include/libxml2 I am building the following version of SIPRouter: ser-2.99.99-pre3_src_2009-12-05_b8e67f The way I am building SIPRouter is as follows: make prefix=/opt/kamailio group_include="standard postgres presence" all If I manually change directory into the modules_s/pa directory and issue a "make" then the build still fails with the same error. If I remove the "presence" group then the build completes successfully. Any pointers for a resolution would be greatly appreciated. If you require any additional information please dont hesitate to ask. Thank you Bruce

3 4

PerlVDB Usage
by Bruce McAlister 09 Dec '09

09 Dec '09

Hi All, We are looking to integrate SIPRouter/Kamailio into our existing infrastructure. Currently we have a sip proxy and registrar that handles registration and routing. The main issue we have with integrating SIPRouter/Kamailio into our environment is that the table structure for kamailio's tables are vastly different to our existing database tables. We have several application built around our current database structure so I think it is fair to assume that we will be forced to use the existing structures. I had a look at perlvdb which seems to do what I had in plan, where the table structure is transparent to kamailio. so for example, when a registration request comes in and we then authenticate the user and want to save the location, could we use the perlvdb module to intercept the save and manipulate it so that it inserts/updates/reads data from a table structure that is different to what it is expecting? I've been trying to find an example of how to use perlvdb but have been unsucessfull in my google searches and mailing list searches. Does anyone have an example of how to use the perlvdb module? Thanks Bruce

4 6

[Kamailio-Users] check SIP agent health
by Robert R 08 Dec '09

08 Dec '09

Hi, I have two SIP agents for redundancy, node1 and node2. Kamailo is supposed to route SIP traffic to node1 unless if it is down. Then it will route the traffic to node2. Is there a way to script Kamailo to check if a SIP agent is alive or not. Thanks, R

2 1

[Kamailio-Users] Add a module's file descriptor to the main loop
by Olivier Détour 08 Dec '09

08 Dec '09

Hi, I would like to create a new module. This module has to read an external file descriptor and call a callback. I don't want to use extra process and shared memory or something like that. Is it possible to add FD in the main Select and call my callback ? Regards, -- Olivier Détour EPITA 2009

2 3

[Kamailio-Users] Problem with ACK
by Ricardo Martinez 07 Dec '09

07 Dec '09

Hello List. I'm testing the last kamailio versión 1.5.3. I'm using the default configuration file and a gateway sending calls to a registered device. Gw : 10.0.0.7 Proxy : 10.0.0.11 Registered device : 10.0.0.20 This is what I see : U 10.0.0.7:5060 -> 10.0.0.11:5060 INVITE sip:5501234567@10.0.0.11 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 INVITE. Supported: timer, replaces, early-session. Min-SE: 1800. Date: Mon, 07 Dec 2009 16:17:13 GMT. User-Agent: AddPac SIP Gateway. Contact: <sip:3341876@10.0.0.7>. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO. Content-Type: application/sdp. Content-Length: 238. Max-Forwards: 70. . v=0. o=3341876 1260202633 1260202633 IN IP4 10.0.0.7. s=AddPac Gateway SDP. c=IN IP4 10.0.0.7. t=1260202633 0. m=audio 23068 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. U 10.0.0.11:5060 -> 10.0.0.7:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 INVITE. Server: Kamailio (1.5.3-notls (i386/linux)). Content-Length: 0. . U 10.0.0.11:5060 -> 10.0.0.20:5060 INVITE sip:5501234567@10.0.0.20:5060;transport=udp SIP/2.0. Record-Route: <sip:10.0.0.11;lr=on>. Via: SIP/2.0/UDP 10.0.0.11;branch=z9hG4bKd506.b3a3dd83.0. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 INVITE. Supported: timer, replaces, early-session. Min-SE: 1800. Date: Mon, 07 Dec 2009 16:17:13 GMT. User-Agent: AddPac SIP Gateway. Contact: <sip:3341876@10.0.0.7>. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO. Content-Type: application/sdp. Content-Length: 238. Max-Forwards: 69. . v=0. o=3341876 1260202633 1260202633 IN IP4 10.0.0.7. s=AddPac Gateway SDP. c=IN IP4 10.0.0.7. t=1260202633 0. m=audio 23068 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. U 10.0.0.20:5060 -> 10.0.0.11:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.0.0.11;branch=z9hG4bKd506.b3a3dd83.0. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. Record-Route: <sip:10.0.0.11;lr=on>. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>;tag=3079903260. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.20:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1B6C7A>". Server: Aastra 57i/2.5.0.82. Supported: gruu, path. Content-Length: 0. . U 10.0.0.11:5060 -> 10.0.0.7:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. Record-Route: <sip:10.0.0.11;lr=on>. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>;tag=3079903260. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.20:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1B6C7A>". Server: Aastra 57i/2.5.0.82. Supported: gruu, path. Content-Length: 0. . U 10.0.0.20:5060 -> 10.0.0.11:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.11;branch=z9hG4bKd506.b3a3dd83.0. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. Record-Route: <sip:10.0.0.11;lr=on>. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>;tag=3079903260. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.20:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1B6C7A>". Server: Aastra 57i/2.5.0.82. Supported: gruu, path, timer, replaces. Content-Type: application/sdp. Content-Length: 210. . v=0. o=MxSIP 0 0 IN IP4 10.0.0.20. s=SIP Call. c=IN IP4 10.0.0.20. t=0 0. m=audio 3000 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U 10.0.0.11:5060 -> 10.0.0.7:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. Record-Route: <sip:10.0.0.11;lr=on>. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>;tag=3079903260. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.20:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1B6C7A>". Server: Aastra 57i/2.5.0.82. Supported: gruu, path, timer, replaces. Content-Type: application/sdp. Content-Length: 210. . v=0. o=MxSIP 0 0 IN IP4 10.0.0.20. s=SIP Call. c=IN IP4 10.0.0.20. t=0 0. m=audio 3000 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U 10.0.0.7:5060 -> 10.0.0.11:5060 ACK SIP/2.0. Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK894be985a441. From: <sip:3341876@10.0.0.7>;tag=894be985a4. To: <sip:5501234567@10.0.0.11>;tag=3079903260. Call-ID: 892a1d4b-35e7-e926-8085-0002a401f510(a)10.0.0.7. CSeq: 41 ACK. Route: <sip:10.0.0.11;lr=on>. Content-Length: 0. Max-Forwards: 70. . The call is answered and then the ACK is sent without R-URI.. making the Proxy fails with : Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:core:parse_uri: uri too short: <> (0) Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:core:parse_sip_msg_uri: bad uri <> Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:rr:loose_route: failed to parse Request URI Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:core:parse_uri: uri too short: <> (0) Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:core:parse_sip_msg_uri: bad uri <> Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:pv:pv_get_ruri_attr: failed to parse the R-URI Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:core:parse_uri: uri too short: <> (0) Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:core:parse_sip_msg_uri: bad uri <> Dec 7 17:22:17 proxy /usr/local/sbin/kamailio[14219]: ERROR:pv:pv_get_ruri_attr: failed to parse the R-URI What could be the problem? I'm using the default configuration file, and this start happening to me since the upgrade from 1.5.2 to 1.5.3.. Hope that someone could help me. Ricardo Martinez.

2 1

[Serusers] Via: branch tags from SER for CANCELs/BYEs frequently don't match value in the INVITE
by Frank Durda IV 07 Dec '09

07 Dec '09

As you may recall, this is a problem where frequently (well, virtually all of the time) when SER passed a BYE or CANCEL message through, the device receiving such messages from SER would reject them, because the Via: header line that SER added to the BYE or CANCEL message didn't match the one that SER added to the INVITE message sent earlier. I brought this up a couple of times in the list previously. As there was not any response on this point at all, I have done some digging and have a simplified example and analysis of what SER appears to be doing wrong. First, here is an example of the problem occurring. Here are key header lines from the INVITE and a BYE message of a sample call: INVITE INPUT TO SER From: <sip:9715551212@10.11.12.13;isup-oli=62;user=phone>;tag=000a0285+1+9c6d0174+ff665515 To: <sip:3719110000@10.20.30.40;user=phone> Call-ID: 16305237000a0285 CSeq: 252558264 INVITE SER EMITTED THIS INVITE (to called party) Via: SIP/2.0/UDP 66.77.88.99;branch=z9hG4bKe47c.7be033a8ca37f0e8a1c6bfe63b043efe.0 From: <sip:9715551212@10.11.12.13;isup-oli=62;user=phone>;tag=000a0285+1+9c6d0174+ff665515 To: <sip:3719110000@10.20.30.40;user=phone> Call-ID: 16305237000a0285 CSeq: 252558264 INVITE BYE INPUT TO SER (from calling party) Call-ID: 16305237000a0285 From: <sip:9715551212@10.11.12.13;isup-oli=62;user=phone>;tag=000a0285+1+9c6d0174+ff665515 To: <sip:3719110000@10.20.30.40;user=phone>;tag=15358744 CSeq: 252558265 BYE SER EMITTED THIS BYE (to called party) Via: SIP/2.0/UDP 66.77.88.99;branch=z9hG4bKf47c.c79c6922afd022e9b4cdf00110f17fd5.0 Call-ID: 16305237000a0285 From: <sip:9715551212@10.11.12.13;isup-oli=62;user=phone>;tag=000a0285+1+9c6d0174+ff665515 To: <sip:3719110000@10.20.30.40;user=phone>;tag=15358744 CSeq: 252558265 BYE The called switch rejects the BYE with a 481 because top-most Via: in the BYE does not match the one from the INVITE. Here are the two VIAs added by SER that do not match: INVITE Via: SIP/2.0/UDP 66.77.88.99;branch=z9hG4bKe47c.7be033a8ca37f0e8a1c6bfe63b043efe.0 BYE Via: SIP/2.0/UDP 66.77.88.99;branch=z9hG4bKf47c.c79c6922afd022e9b4cdf00110f17fd5.0 A review of how SER constructs the branch=nnnn value reveals where the differences are coming from: #1. e47c vs f47c difference caused by the Cseq: number changing between INVITE and BYE messages. The Cseq: value is supposed to be different in the INVITE and BYE per RFC 3261. The Cseq: should not be used in generating the branch=nnnn value. In this example, the BYE just happened to get the very next sequence number, but it will always be a value other than the sequence number seen in the INVITE. #2. 7be033a8ca37f0e8a1c6bfe63b043efe vs c79c6922afd022e9b4cdf00110f17fd5 is caused by ;tag=15358744 added to BYE To: header line. At minimum, any tags present on the To: and From: header should not be used in generating the branch=nnnn value because of the handling of tags in called-party-BYE situations. Also, there is no requirement in RFC 3261 that the To: nor From: header lines be used in generating the branch=nnnn value at all. Based on what I am seeing, it appears that neither the Cseq: nor the complete To: or From: header lines should have ever been used in constructing a branch=nnnn value, because of the adverse affects on CANCEL and BYE messages. Four things appear to be key to causing the branch=nnnn computation to generate non-compliant results in this example, as well as the case of a CANCEL message sent prior to the completion of the INVITE. (There could be other scenarios.) #1. The Cseq: value of the BYE will be different than the INVITE (per RFC 3261). The Cseq: header line should not ever be used in generation the hash for a branch=nnnn value as it will prevent the branch in BYE from matching the one in the INVITE. #2. If the called party generated the BYE message, the tags on the To: and From: header lines will be exchanged (also per RFC 3261), so the entire To: and From: header line's content cannot be used for generating a hash for a branch=nnnn value that is expected to match earlier messages that had a branch=nnnn value computed prior to the tag swap. #3. If a tag was previously absent on the To:/From:, a BYE from either party will add one. Again, the entire To: and From: header lines cannot be used for generating a hash for a branch=nnnn value that matches earlier messages prior to the addition of any tag. #4. Any deviation in the From: and To: header lines between INVITE and BYE/CANCEL will yield a different branch=nnnn value, even though the parameters of the From: and To: header lines may still be perfectly legal and technically identical. Any deviation may be the result of a sloppy SIP implementation in the calling equipment, but isn't valid grounds for not handling the message correctly. Examples include things like alternate number formats (as in To: sip:8885551212@66.77.88.99:5060 (seen in INVITE) versus To: <sip:8885551212@66.77.88.99:5060> (seen in CANCEL from same calling device), or differences in the amount of whitespace (including trailing whitespace), which is supposed to be ignored in SIP messages but the MD5 generator doesn't appear to know that it should ignore whitespace. Any one or a combination of these items (and possibly others) will cause the current branch=nnnn generation code in SER to generate a different branch=nnnn value and a receiving system who is following RFC 3261 will reject the BYE or CANCEL message that has passed through SER. This is non-compliant behavior. For BYE, you only get into this situation if you are using Record-Route to force the BYE message to pass through SER so that it can tear down the call (unforce_rtp_proxy() and similar actions), but for a CANCEL message, anybody can run into this problem, if syn_branch=0. So, while this defect in the branch=nnnn generation doesn't affect everybody, it still appears to be wrong. Is there by chance a compliant version of char_msg_val() exists that doesn't have these issues? That is, it doesn't use the Cseq: header line, and if it uses the From: or To: header line at all, it computes using only the called/calling digits, and not anything else that may appear on those header lines. I also don't understand why this is being MD5'ed. The RFC only says that the branch must be unique, and there are lots of cheaper has algorithms out there. Thanks for reading this far and thanks in advance for anyone who has fixed this or can come up with a fix.

4 7

[Kamailio-Users] Add Reason Header in CANCEL request (forked call)
by heinz huber 07 Dec '09

07 Dec '09

Hi everyone, I've the following question: Is it possible to add a reason header in a CANCEL request? This is helpful in the following scenario: A call gets forked by the proxy (since there are more contacts for one AOR). On the of the SIP-destination-UAs accepts the call and sends a 200 OK to the proxy. The proxy afterwards generates the CANCEL request for all open UAC transactions. In this case it would be helpful to add additional infos, like "Call completed" elsewhere, so that the call is not listed in the "missing calls" list in the destinations UAs. I've found a linke in the tracker on http://sourceforge.net/tracker/?func=detail&aid=2581449&group_id=139143&ati… Has this feature been included in the last official kamailio version, or is it still open. thx && br Andy -- Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3.5 - sicherer, schneller und einfacher! http://portal.gmx.net/de/go/atbrowser

2 1

[Kamailio-Users] log not showing when using uac_replace_from( ) function
by Robert R 07 Dec '09

07 Dec '09

Hi, I simply trying to modify URI and FROM headers as follow. tcpdump packet capture shows the values are changed for both URI and FROM headers. However, the syslog only shows the new value for URI and not for FROM header. How can I verify new value in FROM header in the log file? # main request routing logic route { $rU = avp(i:1); $rd = avp(i:2); .... uac_replace_from("$avp(i:3)"); xlog ("fu after perl: $fu"); xlog ("ru after perl: $ru"); ... } Thanks, Rob

3 2

[Kamailio-Users] finding out which gw id was used
by Anders 07 Dec '09

07 Dec '09

Hi, I would like to know which gw was used for a call. Right now, it is doing this: sql_query("cb", "select id from gw where ip_addr='$rd'", "rb"); ...in route[1], but that's not good enough, because I have more than one gw with the same IP, so it just takes the first one available. If I could do something like "select id from gw where grp_id='grp_id_from_lcr'" , where grp_id_from_lcr is the the grp_id in the lcr table, which lead to the gw used... then I'd have the right id of the gw, since I use unique numbers for the grp_id in the gw table. How do I do this? Thanks!!

2 1

[Kamailio-Users] Can't connect with rtpproxy-0.2_2007-07-28
by RAJNIKANT VANZA 07 Dec '09

07 Dec '09

Hi Friends, i have problem about rtpproxy connection. /var/log/messages/ --------------------------- ERROR:nathelper:send_rtpp_command: can't connect to RTP proxy ERROR:nathelper:send_rtpp_command: proxy <unix:/var/run/rtpproxy.sock> does not respond, disable it WARNING:nathelper:rtpp_test: can't get version of the RTP proxy WARNING:nathelper:rtpp_test: support for RTP proxy <unix:/var/run/rtpproxy.sock> has been disabled temporarily i have installed successful rtpproxy-0.2_2007-07-28. run from /usr/local/sbin/rtpproxy in /var/run/ to files exists as -rw-r--r-- 1 root root 6 Dec 4 19:48 rtpproxy.pid srwxr-xr-x 1 root root 0 Dec 4 19:48 rtpproxy.sock kamailio.cfg ------------------ loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") -- Thanks and Regards Rajnikant Vanza Call : +91-97374 56583

2 1

Symbol referencing errors (building ser-2.99.99-pre3_src_2009-12-03_ca351a on Solaris)
by Bruce McAlister 07 Dec '09

07 Dec '09

Hi All, When attempting to build SIPRouter version ser-2.99.99-pre3_src_2009-12-03_ca351a on Solaris 10 I am presented with the following error: ---- config.mak included make[1]: Entering directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/modules/ctl' Makefile.defs defs skipped make[2]: Entering directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/utils/sercmd' make[2]: Leaving directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/utils/sercmd' Makefile.defs defs skipped make[2]: Entering directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/utils/sercmd' /usr/sfw/bin/gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m32 -minline-all-stringops -falign-loops -mtune=athlon -Wall -DNAME='"sercmd"' -DVERSION='"0.2"' -DMOD_NAME="utils/sercmd" -c parse_listen_id.c -o parse_listen_id.o /usr/sfw/bin/gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m32 -minline-all-stringops -falign-loops -mtune=athlon -Wall -DNAME='"sercmd"' -DVERSION='"0.2"' -DMOD_NAME="utils/sercmd" -c sercmd.c -o sercmd.o /usr/sfw/bin/gcc -m32 -O2 parse_listen_id.o sercmd.o -o sercmd Undefined first referenced symbol in file bind sercmd.o getservbyname parse_listen_id.o gethostbyname sercmd.o socket sercmd.o connect sercmd.o ld: fatal: Symbol referencing errors. No output written to sercmd collect2: ld returned 1 exit status make[2]: *** [sercmd] Error 1 make[2]: Leaving directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/utils/sercmd' ERROR: make -C ../../utils/sercmd install-if-newer failed make[1]: *** [install-utils] Error 1 make[1]: Leaving directory `/export/home/user/packages/BUILD/ser-2.99.99-pre3/modules/ctl' make: *** [install-modules] Error 1 ---- Here are the steps that I have followed to get to this point: [1] make prefix=/opt/kamailio [2] make prefix=/opt/kamailio 'include_modules=db_unixodbc db_postgres cpl-c snmpstats' modules [3] make basedir=/var/tmp/pkgbuild-user/SIPRouter-2.99.99-pre3-build prefix=/opt/kamailio 'include_modules=db_unixodbc db_postgres cpl-c snmpstats' install The above error I receive is during the last step ([3]), the installation phase of SIPRouter. I'm not sure where to go from here, although it looks like it may be a resolver library that is not included along the line. Any help would be appreciated. If you need any further information, please dont hesitate to ask. Thanks Bruce

2 4

[Kamailio-Users] [kamailio-users] Error : SIP/2.0 401 Unauthorized
by RAJNIKANT VANZA 05 Dec '09

05 Dec '09

Hi Friends, need to help. *I have problem about sip : SIP/2.0 401 Unauthorized* Is it require to nathelper module in kamailio ? *what can i write kamailio.cfg file when kamailio and Asterisk on same network?* Scenario is like as : ----------------------------- 1) kamailio server on 172.18.100.74 kamailio.cfg ( nathelpler module ) ----------------- loadmodule "nathelper.so" modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") 2) Asterisk server on 172.18.100.65 sip.conf ----------- [rajnikant] nat=yes disallow=all allow=alaw allow=ulaw allow=gsm type=peer context=default host=172.18.100.74 fromdomain=rajnikant.net mailbox=user@context -- Thanks and Regards Rajnikant Vanza

2 1

[Kamailio-Users] Kamailio RTP Proxy issues
by General Lee 04 Dec '09

04 Dec '09

Hi, I am currently integrating an H323 to SIP gateway with Kamailio and trying to route all calls through RTP Proxy. I have a problem where RTP Proxy treats both the incoming H323 gateway call (Caller) and outgoing SIP call (Callee) as the 'caller' in the RTP Proxy syslog. RTP Proxy doesn't assign a 'callee' therefore not able to setup a call (See syslog below). I have configured Kamailio to accept the 'ACK' with SDP and this is working correctly. When I enable H323 Fast connect and the SDP is included in the INVITE, the call connects correctly and is routed through RTP Proxy. I feel the problem is related to RTP Proxy receiving an INVITE from the H323-SIP gateway without SDP. Can anyone explain why RTP Proxy treats both the incoming H323 Gateway call and outgoing SIP call as the 'caller' in the RTP Proxy syslog. How can I make RTP Proxy treat the incoming H323 call as the 'callee'? Thanks, More information below ******************************************************************************* My H323 endpoints use H323 Slow Connect, so when the H323-SIP Gateway delivers the 'INVITE' to Kamailio there is no SDP included in the INVITE. I added a 'onreply_route' to the Kamailio configuration file which handles the 'ACK' with SDP which is working correctly. All of my SIP calls (Signalling + Media) are forced though RTP Proxy and I would like to force all H323-SIP Gateway calls through RTP Proxy. When placing a call from my H323 endpoint to my SIP UA, the RTP Proxy syslog records the incoming and outgoing call, however RTP Proxy states that the ‘callee’ is actually the 'caller'. The RTP Proxy syslog also states that the caller is the caller so there is no 'callee' (see below). In the syslogs both the ‘callee’ and ‘caller’ are recognised as the ‘caller’ so RTP Proxy has no callee to send the traffic back to. When the INVITE is received the ‘callee’ is populated in the syslogs as the ‘caller’. The H323 Gateway call is not recorded until after the ‘ACK’ with SDP is received from the gateway. Oct 27 17:33:05 rtpproxy[24086]: INFO:handle_command: pre-filling caller's address with 72.19.211.106:49620 (Should be callee) Oct 27 17:33:05 rtpproxy[24086]: INFO:handle_command: pre-filling caller's address with 72.19.211.106:49622 (Should be callee) Then after the ‘ACK’ is received from the Gateway, the H323 call is mentioned in the syslog as well as the other caller who is supposed to be the callee. Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling caller's address with 69.72.11.51:10204 Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling caller's address with 69.72.11.51:10214 Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling caller's address with 72.19.211.106:49620 Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling caller's address with 72.19.211.106:49622 -- View this message in context: http://www.nabble.com/Kamailio-RTP-Proxy-issues-tp26112049p26112049.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 12

Build Error for v3.0.0-rc2 (2009-12-03_3ed668) on Solaris
by Bruce McAlister 03 Dec '09

03 Dec '09

Hi All, I am trying to build Kamailio v3.0.0 RC2 on Solaris 10 (note: *not* OpenSolaris), but I have been faced with a couple issues I hope you can help me with. The first issue has to do with GNU as (/usr/sfw/bin/gas), there appears to be a check somewhere for what I assume to be the version number of the GNU assembler. When the make is running "/usr/sfw/bin/gas -V" it just hangs at this point with the following output from make: ---- make prefix=/opt/kamailio Makefile:248: config.mak: No such file or directory normal Makefile.defs exec target architecture <i386>, host architecture <x86_64> ---- I did a process listing and noticed the following process waiting: 1719 make prefix=/opt/kamailio 1763 /bin/sh -c if [ -z "/usr/sfw/bin/gas" ] ; then echo "as" ; else if /usr/sfw/bin 1764 /usr/sfw/bin/gas -V I ended up making a wrapper script that just output the version information for "/usr/sfw/bin/gas -V" which let me get a little further. The wrapper script looks as follows: ---- #/bin/ksh if [ "$1" = "-V" ]; then echo "GNU assembler version 2.15 (i386-pc-solaris2.10) using BFD version 2.15" exit else /usr/sfw/bin/gas.bin $@ fi ---- However, when I now try a "make prefix=/opt/kamailio" I am faced with the following error: ---- gcc: installation problem, cannot exec `/usr/sfw/bin/gas': Exec format error ---- The complete output of the make is as follows: ---- make prefix=/opt/kamailio Makefile:248: config.mak: No such file or directory normal Makefile.defs exec target architecture <i386>, host architecture <x86_64> making config... yacc -d -b cfg cfg.y conflicts: 5 shift/reduce flex cfg.lex Makefile.defs defs skipped /usr/sfw/bin/gcc -g -O9 -funroll-loops -Wcast-align -m32 -minline-all-stringops -falign-loops -mtune=athlon -Wall -DNAME='"kamailio"' -DVERSION='"3.0.0-rc2"' -DARCH='"i386"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/usr/sfw/bin/gcc 3.4.3"' -D__CPU_i386 -D__OS_solaris -DSER_VER=3000000 -DCFG_DIR='"/opt/kamailio/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DSTATISTICS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -c action.c -o action.o gcc: installation problem, cannot exec `/usr/sfw/bin/gas': Exec format error make: *** [action.o] Error 1 ---- If I remove the wrapper script then the make just hangs on the command "/usr/sfw/bin/gas -V". Would anyone know how I could work around the issue? Just an FYI, I am able to build Kamailio 1.5.2-notls on the same machine without any errors/hangs, although we are considering moving to Kamailio 3/Sip-Router when it is officially released, this is why I am trying to build v3.0.0 RC2 at the moment. If you need any further information please dont hesitate to ask, any help in resolving the issue would be greatly appreciated. Thanks Bruce

3 4

[Kamailio-Users] log not showing when using uac_replace_from( ) function
by Robert R 03 Dec '09

03 Dec '09

Hi, I simply trying to modify URI and FROM headers as follow. tcpdump packet capture shows the values are changed for both URI and FROM headers. However, the syslog only shows the new value for URI and not for FROM header. How can I verify new value in FROM header in the log file? # main request routing logic route { $rU = avp(i:1); $rd = avp(i:2); .... uac_replace_from("$avp(i:3)"); xlog ("fu after perl: $fu"); xlog ("ru after perl: $ru"); ... } Thanks, Rob

1 0

[Kamailio-Users] log not showing when using uac_replace_from( ) function
by Robert R 03 Dec '09

03 Dec '09

Hi, I simply trying to modify URI and FROM headers as follow. tcpdump packet capture shows the values are changed for both URI and FROM headers. However, the syslog only shows the new value for URI and not for FROM header. How can I verify new value in FROM header in the log file? # main request routing logic route { $rU = avp(i:1); $rd = avp(i:2); .... uac_replace_from("$avp(i:3)"); xlog ("fu after perl: $fu"); xlog ("ru after perl: $ru"); ... } Thanks, Rob

1 0

[Kamailio-Users] Kamailio Perl OpenSER::AVP
by Örn Arnarson 03 Dec '09

03 Dec '09

Hello everyone, I've been successfully using the kamailio Perl module, but now I'm trying to set AVPs. Unsuccessfully. Whenever I do "use OpenSER::AVP;" and restart kamailio, the following error messages appear and kamailio doesn't start: Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:core:XS_OpenSER__Message_log: perl error: Can't locate OpenSER/AVP.pm in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.10.0 /usr/local/share/perl/5.10.0 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.10 /usr/share/perl/5.10 /usr/local/lib/site_perl .) at /etc/kamailio/checkip.pl line 4. Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:core:XS_OpenSER__Message_log: perl error: Can't locate OpenSER/AVP.pm in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.10.0 /usr/local/share/perl/5.10.0 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.10 /usr/share/perl/5.10 /usr/local/lib/site_perl .) at /etc/kamailio/checkip.pl line 4. BEGIN failed--compilation aborted at /etc/kamailio/checkip.pl line 4. Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:perl:parser_init: failed to load perl file "/etc/kamailio/checkip.pl". Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:core:init_mod: failed to initialize module perl Dec 3 12:42:51 proxy02 /usr/sbin/kamailio[5392]: ERROR:core:main: error while initializing modules Now, I cannot find this AVP module within the perl package anywhere. I downloaded the source of 1.5.0 and it's nowhere to be found. It is, however, in the .1.5 documentation. Does anyone have any info on this for me? Best regards, Örn

3 3

[Kamailio-Users] Using Sip Message in a third application.
by Olivier Détour 03 Dec '09

03 Dec '09

Hello, I'm using Kamailio 1.5.3. I configured it in a proxy mode. I would like to know what is the best way to forward the SIP message to a C application via a socket, block kamailio.cfg script, and reforward it to kamailio to continue his treatment. I found a ugly bay using PV as lock with a while loop, but this solution is not acceptable for me. I saw SEAS module, but it is only for Java Application server. Do you know a simple way to do it with an existing module ? Do I have to write my own module ? Thanks for your response, Regards, -- Olivier Détour EPITA 2009

3 5

topoh: remove Via headers
by Klaus Darilion 03 Dec '09

03 Dec '09

Hi! IIRC correctly. topoh modules rewrites existing Via headers. Would it be possible to remove existing headers completely so that the forwarded request has only one Via header? regards klaus

1 0

[Kamailio-Users] Integrate Skype into Kamailio (Must support video)
by General Lee 03 Dec '09

03 Dec '09

Hi, I would like to integrate Skype into my Kamailio video system? Has anyone implemented this successfully before? Or suggest how this could be done? Key features I would like are: 1. Must support Video 2. Ideally support H264 video codec 3. Share presence information 4. Ability to make calls in both directions I would appreciate any information you could provide. Thanks, -- View this message in context: http://old.nabble.com/Integrate-Skype-into-Kamailio-%28Must-support-video%2… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

4 6

[Kamailio-Users] ERROR:tm:add_uac: maximum number of branches exceeded
by alex pappas 02 Dec '09

02 Dec '09

Hello there, I'm using dispatcher to serial fork a call through many gateways. For a call to number 2XXXXXXXXX , first a try to connect through gateways A,B,C after that D,F , after that E,R , after that M,N and finally after those K,L. When the call is trying gateway N then I'm getting the ERROR:tm:add_uac: maximum number of branches exceeded. I found on internet that I'm exceeding the MAX_BRANCHES: branch=t->nr_of_outgoings; if (branch==MAX_BRANCHES) { LOG(L_ERR, "ERROR:tm:add_uac: maximum number of branches exceeded\n"); ret=E_CFG; goto error; } >From a trace I got , I see that kamailio is sending to each destination IP 8 INVITES and if there is no answer to that is continuing with next destination gateway sending also there 8 INVITES. When the number of INVITES is 52 then I get this error. Does anyone can suggest a way to change the MAX_BRANCHES MAX number ? Is is important to keep open the INVITEs from the first gateways because is an LCR scenario and if I get an answer from these gateways is important to complete through them. Any help is apreciated Thanks Alexandros

2 6

[Kamailio-Users] Random looped back audio issues with RTPPROXY
by tony＠everestbn.com 02 Dec '09

02 Dec '09

Hello, I'm having random audio issues (<10% of the time) with blind call transfers on a Polycom phone. According to RTPPROXY's logs, it seems it is substituting the same IP and port for the caller's and callee's address causing one end to hear their own voice while the other end hears nothing. INFO:rxmit_packets: caller's address filled in: a.b.c.199:17856 (RTP) INFO:rxmit_packets: guessing RTCP port for caller to be 17857 INFO:rxmit_packets: callee's address filled in: a.b.c.199:17856 (RTP) INFO:rxmit_packets: guessing RTCP port for callee to be 17857 The SDP in the SIP messages has all the correct IPs and ports and it appears from a SIP overview everything should work fine. Here is some background information: I am using Kamailio 1.5.3 with RTPPROXY 1.2.1. SIP PATH is as follows: PSTN Gateway---Kamailio--SIP Proxy Server--Polycom Phone RTP PATH: PSTN Gateway---Kamailio----Polycom Phone *There are nor firewalls or filters. Call Flow: I place a call from the gateway to the ip phone. Call is answered. When I do a blind transfer from the Polycom, the SIP Proxy sends an invite with no SDP, the gateway responds with a 200 with SDP and the SIP Proxy then sends an ACK with SDP. As I mentioned, this works most of the time and the SIP messages look identical for when the transfer works and when it doesn't. When it doesn't work the gateway end hears their own voice while the recipient Polycom transfer (usually another phone on the same proxy) hears nothing. Here is a portion of my configuration file: ..... if (is_method("INVITE")) { if((search("^Content-Type:[ ]*application/sdp")) || (search("^Content-Type:application/sdp"))){ rtpproxy_offer("fcr"); setflag(12); } } if (is_method("ACK")) { if((search("^Content-Type:[ ]*application/sdp")) || (search("^Content-Type:application/sdp"))) { rtpproxy_answer("fcr"); } } t_on_reply("1"); .... } onreply_route[1] { if (status=~"(180)|(183)|(2[0-9][0-9])"){ if((search("^Content-Type:[ ]*application/sdp")) || (search("^Content-Type:application/sdp"))) { if (isflagset(12)) { rtpproxy_answer("fcr"); } else { rtpproxy_offer("fcrl"); } } } } Any help would be greatly appreciated! Thanks, Tony

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Re: [Kamailio-Users] [sr-dev] sr and kamailio autogenerated tarballs
by Daniel-Constantin Mierla 02 Dec '09

02 Dec '09

On 12/2/09 1:50 PM, Andrei Pelinescu-Onciul wrote: > Daily snapshots of sr_3.0, kamailio_3.0 and sr master branch can be > downloaded from: http://sip-router.org/tarballs/sr/ . > Note that new tarballs are generated only if there are changes for the > corresponding branch (so if you don't see a tarball with today's date it > means there was no change from the previous one). > > The tarballs are generated via make tar, which makes sure the correct > repository version will be included in the compiled binary (make tar > generates first autover.h before creating the archive). > Thanks Andrei! I cc-ed Kamailio users mailing list, since there might be people interested as well. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Serusers] Memory leak issue
by Piyush.Bansal＠relianceada.com 02 Dec '09

02 Dec '09

Hi All, I'm using SER version 2.0.0-rc1 as a SIP server and in that facing lot of memory leaks issue. I'm not able to enable the memory logs in SER. Can anybody help me identifying memory leaks or tell me how to enable memlogs so that I myself will try removing all the leaks... Plz help me ASAP, its urgent... Thanx n regards, --Piyush The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

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[Kamailio-Users] Kamailio -> Asterisk Redirect/Reinvite and Remote-Party-ID
by Florian Meister 02 Dec '09

02 Dec '09

Hi, basically I'm using this structure at the moment: SIP Users <----> Kamailio <-----> Asterisk <-----> PSTN I have to add a diversion-functionality at kamailio-level, so to simply rewrite $ru with something else defined in the database. That's working without problems. For billing issues, I also have to add a Remote-Party-ID header, set to the SIP user, which initiated the redirect in the database. Now to the problem: When a call is coming from PSTN, it's passing the asterisk server, then at kamailio level $ru is rewritten and sent back to asterisk (I'm talking about a redirect to a number in PSTN here) What I've seen from the logs is, that asterisk is seeing that it gets an invite back with the same call-id, and therefore it cancels the original invite and handles the whole call internally via the Local Channel. The Problem is, that in the invite sent from kamailio back to asterisk, I've set a Remote-Party-ID header to tell asterisk to set the Callerid correctly for billing purposes. Now it seems that asterisk is _ignoring_ this header from the second invite. So is this an expected behavior ? If yes, how to do it correctly ? Below you can see the verbose output of asterisk. Since the call is handled at "Local" Channels the function to read sip headers does not work. The only message I get is "thanks to SIP/tpsiptestproxyu01-00d0a0b8". -- Called tpsiptestproxyu01/+435572949012 -- Now forwarding DAHDI/2-1 to 'Local/066480588134@from-internal' (thanks to SIP/tpsiptestproxyu01-00d0a0b8) -- Executing [066480588134@from-internal:1] NoOp("Local/066480588134@from-internal-d69e;2", "435572501134") in new stack [Dec 1 15:14:50] WARNING[20506]: chan_sip.c:15797 func_header_read: This function can only be used on SIP channels. -- Executing [066480588134@from-internal:2] NoOp("Local/066480588134@from-internal-d69e;2", "") in new stack -- Executing [066480588134@from-internal:3] Dial("Local/066480588134@from-internal-d69e;2", "DAHDI/G0/066480588134") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G0/066480588134 -- DAHDI/124-1 is proceeding passing it to Local/066480588134@from-internal-d69e;2 -- Local/066480588134@from-internal-d69e;1 is proceeding passing it to DAHDI/2-1 In the SIP debug you can see that asterisk is cancelling the dialog with kamailio and doing it itself: 13:49:30.589054 IP [--ASTERISK--].5060 > [--KAMAILIO--].5060: SIP, length: 938 E.......@..L..,L..,N........INVITE sip:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]> Contact: <sip:435572501134@[--ASTERISK--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Remote-Party-ID: "435572501134" <sip:435572501134@[--ASTERISK--]>;privacy=off;screen=yes Date: Tue, 01 Dec 2009 12:49:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 930830518 930830518 IN IP4 [--ASTERISK--] s=Asterisk PBX 1.6.1.5 c=IN IP4 [--ASTERISK--] t=0 0 m=audio 16924 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 13:49:30.591037 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length: 342 E..r..@.@.[0..,N..,L.....^.aSIP/2.0 100 Trying Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 INVITE Server: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 13:49:30.594345 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length: 1093 E..a..@.@.XA..,N..,L.....M+LINVITE sip:066480588134@[--ASTERISK--]:5060;transport=udp SIP/2.0 Record-Route: <sip:[--KAMAILIO--];lr;ftag=as27658014> Via: SIP/2.0/UDP [--KAMAILIO--];branch=z9hG4bK06.05390227.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 Max-Forwards: 69 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]> Contact: <sip:435572501134@[--ASTERISK--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 01 Dec 2009 12:49:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 Remote-Party-ID: "435572949012" <sip:435572949012@tpseru01.tele.net>;party=caller;privacy=none;screen=yes v=0 o=root 930830518 930830518 IN IP4 [--ASTERISK--] s=Asterisk PBX 1.6.1.5 c=IN IP4 [--ASTERISK--] t=0 0 m=audio 16924 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 13:49:30.594605 IP [--ASTERISK--].5060 > [--KAMAILIO--].5060: SIP, length: 460 E.......@..)..,L..,N....... CANCEL sip:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 CANCEL User-Agent: Asterisk PBX 1.6.1.5 Remote-Party-ID: "435572501134" <sip:435572501134@[--ASTERISK--]>;privacy=off;screen=yes Content-Length: 0 13:49:30.596307 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length: 387 E.....@.@.[...,N..,L.......KSIP/2.0 200 canceling Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]>;tag=45db18648893e7acabf725621374d382-4ddb Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 CANCEL Server: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 13:49:30.596801 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length: 396 E.....@.@.Z...,N..,L.......kSIP/2.0 487 Request Terminated Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]>;tag=45db18648893e7acabf725621374d382-4ddb Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 INVITE Server: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 13:49:30.596842 IP [--ASTERISK--].5060 > [--KAMAILIO--].5060: SIP, length: 539 E..7....@.....,L..,N.....#.oACK sip:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]>;tag=45db18648893e7acabf725621374d382-4ddb Contact: <sip:435572501134@[--ASTERISK--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.5 Remote-Party-ID: "435572501134" <sip:435572501134@[--ASTERISK--]>;privacy=off;screen=yes Content-Length: 0 Thanks, Florian

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[Kamailio-Users] client registers with private IP
by Joao Gomes Pereira 01 Dec '09

01 Dec '09

Hello I'm trying to register in Kamailio, an Asterisk working behind Nat. Here is the trunk: The problem is: Asterisk is registering with its internal IP (192.168.1.25), as you can see here: sip:s@192.168.1.25 Q= Expires:: 81 Callid:: 480b40aa13ddd8707787b21a69656535(a)127.0.0.1 Cseq:: 103 User-agent:: Asterisk PBX How can I force Asterisk to register with its public IP? Is there some configuration I can do in Kamailio? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira(a)startel.pt

3 2

[Kamailio-Users] CDRTool Mediaproxy etc.. AG stuff
by Brian Chamberlain 01 Dec '09

01 Dec '09

Hi List, Apologies if you get this twice, for some reason the last post didn't seem to appear on the list.. I am running Debian lenny. Can I just change the repos in sources.list from lenny to testing and unstable to get the correct python dependencies etc for the ag-project tools? Thanks, Brian

4 4

[Kamailio-Users] presence send system info
by alex pappas 01 Dec '09

01 Dec '09

Dear friends, I'm trying to send some system info through presence in my X-Lite client , following the example http://www.kamailio.org/dokuwiki/doku.php/presence:pua-modules#using_with_x… Does anyone knows what configuration should i have in the X-Lite side? Thanks Alex

2 7

[Kamailio-Users] t_on_failure() t_on_reply()
by BERGANZ François 01 Dec '09

01 Dec '09

Hello, Is it possible to do that : ? t_on_reply("1"); t_on_failure("1"); t_relay(); François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 3

[Kamailio-Users] Fwd: Binding to multiple IP addresses on the same interface.
by Geoffrey Mina 01 Dec '09

01 Dec '09

yes it does: [root@atl-sipgateway1 ~]# ip addr show dev eth0 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast qlen 1000 link/ether 00:1e:c9:b6:4c:fa brd ff:ff:ff:ff:ff:ff inet 67.220.XX.XX/27 brd 67.220.XX.XX scope global eth0 inet 67.220.XX.XX/27 brd 67.220.XX.XX scope global secondary eth0:1 inet6 fe80::21e:c9ff:feb6:4cfa/64 scope link valid_lft forever preferred_lft forever On Mon, Nov 30, 2009 at 12:15 PM, Alex Hermann <alex(a)speakup.nl> wrote: > On Sunday 29 November 2009 20:57:59 Geoffrey Mina wrote: >> I have a Kamailio machine that has multiple IPs assigned to a single >> interface. >> >> eth0 = 1.2.3.4 >> eth0:1 = 1.2.3.5 > > No idea if it has any influence on Kamailio behaviour, but those seem to be > two interfaces as far as applications are concerned. Assigning multiple IP's > to one interface means that for all IP's, the interface should be the same. > > Does 'ip addr show dev eth0' show 2 IP's? > > -- > Alex Hermann > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >

1 0

[Kamailio-Users] 487
by Anders 30 Nov '09

30 Nov '09

Hi, This question may belong in a general SIP forum - but I'll drop it here anyway, - hope I stir no big anger ;-) I'm running Kamailio 1.5, - use it for wholesale, and my customers typically send me calls from their Asterisk platform. I'm experiencing a very share of calls ending up in 487. This should be the client terminating the call - but is it my client (the provider with the Asterisk platform) or is it the final client - my client's client? Could Asterisk send a 487 for some reason? I've researched this (maybe not good enough!), but cannot find the answer: Does a 487 always mean that the client gave up and hung up? If this is the case, could high PDD be my problem - it's taking too long and the client gives up? Thanks!

2 1

[Kamailio-Users] Binding to multiple IP addresses on the same interface.
by Geoffrey Mina 30 Nov '09

30 Nov '09

Hello, I have a Kamailio machine that has multiple IPs assigned to a single interface. eth0 = 1.2.3.4 eth0:1 = 1.2.3.5 In my Kamailio config i have listen=eth0:5060 And it doesn't appear to be binding to my second IP address. What is the proper way to tell the server to bind to all IPs on a single interface? I don't want to specify the IP directly for binding as I use the same config across 4 Kamailio servers. listen=eth0:1:5060 (i think that's not correct) Thanks, Geoff p.s. I would do some testing, but this is a live machine and I don't want to bring it up and down.

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Re: [Kamailio-Users] Compiling nonce.o (Auth Module) with TLS=1 fails
by Martin Koenig 30 Nov '09

30 Nov '09

Hello Laszlo, Good hint. We are building against openssl-0.9.7f-7.10. For testing purposes I have commented out the INCLUDE of md5.h in nonce.c, now the module compiles with the following warning: gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"auth"' -DNAME='"kamailio"' -DVERSION='"1.5.3-tls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DUSE_LOCAL_ROUTE -DF_MALLOC -DUSE_TLS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/ssl/include -I/usr/local/include -I/usr/include/openssl -c nonce.c -o nonce.o nonce.c: In function âcalc_nonceâ: nonce.c:116: warning: implicit declaration of function âMD5Initâ nonce.c:126: warning: implicit declaration of function âMD5Updateâ nonce.c:129: warning: implicit declaration of function âU_MD5Finalâ I'll try to see if this is functional now. However, it feels like a "dirty hack". Best regards, Martin ________________________________ Von: voipfreak.net(a)gmail.com [mailto:voipfreak.net@gmail.com] Im Auftrag von Laszlo Gesendet: Montag, 30. November 2009 17:57 An: Martin Koenig Betreff: Re: [Kamailio-Users] Compiling nonce.o (Auth Module) with TLS=1 fails Hi Martin, [...] Which version of openssl you have? You might find this interesting: http://bugs.gentoo.org/146780 (botton of the page) -Laszlo

1 0

[Kamailio-Users] Compiling nonce.o (Auth Module) with TLS=1 fails
by Martin Koenig 30 Nov '09

30 Nov '09

Hello all, I am trying to compile kamailio 1.5.3 with TLS=1. The whole kamailio package compiles just fine, with the exception of nonce.o: gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"auth"' -DNAME='"kamailio"' -DVERSION='"1.5.3-tls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DUSE_LOCAL_ROUTE -DF_MALLOC -DUSE_TLS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/ssl/include -I/usr/local/include -I/usr/include/openssl -c nonce.c -o nonce.o In file included from /usr/include/openssl/evp.h:84, from /usr/include/openssl/x509.h:67, from /usr/include/openssl/ssl.h:177, from ../sl/../../tls/tls_config.h:34, from ../sl/../../version.h:29, from ../sl/../../sr_module.h:54, from ../sl/sl_api.h:33, from auth_mod.h:39, from nonce.c:40: /usr/include/openssl/md5.h:105: error: conflicting types for âMD5_CTXâ ../../md5.h:42: error: previous declaration of âMD5_CTXâ was here nonce.c: In function âcalc_nonceâ: nonce.c:116: warning: passing argument 1 of âMD5Initâ from incompatible pointer type nonce.c:126: warning: passing argument 1 of âMD5Updateâ from incompatible pointer type nonce.c:128: warning: passing argument 1 of âMD5Updateâ from incompatible pointer type nonce.c:129: warning: passing argument 2 of âU_MD5Finalâ from incompatible pointer type make: *** [nonce.o] Error 1 Do you guys have any hints on what is going wrong? Best regards, Martin Koenig

1 0

[Kamailio-Users] Can kamailio bypass rtp media to backend sip server?
by Lei Tang 30 Nov '09

30 Nov '09

Hi everyone, I'm new to kamailio. I plan to setup a sip server cluster, Does someone can give me some suggestions if I can use kamailio as front server, which handle sip message and bypass rtp media message to backendserver, this mean the kamailio handle only sip message, the rtp packets are sent directed to backend server. The sip sequences is 1.softswitch send invite to kamailio 2.kamailio send invite to backend server. 3.backend server return 200 ok to kamailio 4.kamailio return 200 ok to softswitch, the ip and port in sip is from the backend server's msg, this make the softswitch send rtp direct to backend server. 5.rtp packets are transfered between softswitch and backend server. .... 6.backend server send bye to kamailio 7.kamailio send bye to softswitch. -- Lei.Tang lei.tlfly(a)gmail.com

1 0

[Kamailio-Users] Adding Columns to Database
by JR Richardson 29 Nov '09

29 Nov '09

Hi All, Using openser 1.1.0-notls (i386/linux) from debian etch package. Working with mysql database integration, working fine, using permissions module with 'trusted' table and alias_db module with the 'dbalias' table. I created the database and tables with openser_mysql cmd and things seemed to work out of the box. I ran into trouble when I dropped the database and manually created just the 'trusted' and 'dbalias' tables and added a few more columns in the tables [accountcode,notes,id(auto increment)]. When I restarted openser, no go, was getting table version and structure errors. So my question: is the table structure strict as per the openser_mysql script and are all 19 tables required for a properly running instance? I have not tried yet, but would it break if I added the new columns to the openser_mysql script created tables? Thanks. JR -- JR Richardson Engineering for the Masses

2 2

[Kamailio-Users] compile error for perl module on Fedora 11
by Zen Kato 28 Nov '09

28 Nov '09

I have installed the following packages on Fedora 11: perl-5.10.0-82.fc11.i586 perl-devel-5.10.0-82.fc11.i586 perl-libs-5.10.0-82.fc11.i586 perl-LDAP-0.34-5.fc11.noarch perl-IPC-Shareable-0.60-9.fc11.noarch But, when I compile perl modules: # make include_modules="jabber db_mysql perl perlvdb" all ....(snip)....... openserxs.c:1968: error: ‘PL_scopestack_ix’ undeclared (first use in this function) openserxs.c:1969: error: ‘XSRETURN_YES’ undeclared (first use in this function) make[1]: ディレクトリ `/usr/local/src/kamailio/1.5.3/kamailio-1.5.3-notls/modules/perl' から出ます make: *** [modules] エラー 2 What modules should I install on Fedora 11? Regards, Zen

1 1

Problems with PRESENCE/DIALOG functionality in sip-router
by klaus.feichtinger＠gmx.net 27 Nov '09

27 Nov '09

Hi all, I've tested the current release candidate for Kamailio these days and found that the presence/dialog state functionality is not given in the current version. The first (and worst) difference to the latest stable release is that the handling of "setflag(X)" for tracing the dialogs resulted in a crash of kamailio with following error message: ********************** EXCERPT ****************** Nov 24 13:35:47 lennysrv /usr/local/sbin/kamailio[13814]: : tm [t_hooks.c:190]: BUG:tm:register_tmcb: callback type TMCB_REQUEST_IN can't be register along with types Nov 24 13:35:47 lennysrv /usr/local/sbin/kamailio[13814]: ERROR: dialog [dlg_handlers.c:615]: failed cache in T the shortcut to dlg Nov 24 13:35:53 lennysrv /usr/local/sbin/kamailio[13815]: NOTICE: acc [acc.c:275]: ACC: call missed: timestamp=1259134133;method=INVITE;from_tag=4051501197-3958320-1259134127622;to_tag=3762694574-3958280-1259134127622;call_id=140363298-3958320-1259134127622(a)10.16.48.60;code=603;reason=Decline;src_user=5003;src_domain=10.16.48.205;dst_ouser=5002;dst_user=5002;dst_domain=10.16.48.60 Nov 24 13:35:56 lennysrv /usr/local/sbin/kamailio[13815]: INFO: pua_usrloc [ul_publish.c:203]: should not send ul publish Nov 24 13:35:56 lennysrv /usr/local/sbin/kamailio[13815]: INFO: <script>: --- pua-set-publish() was executed --- Nov 24 13:35:58 lennysrv kernel: [176699.348059] kamailio[13821]: segfault at c0c0c0c0 ip b7de43b3 sp bfb03eb8 error 5 in libc-2.7.so[b7d6e000+155000] Nov 24 13:35:58 lennysrv /usr/local/sbin/kamailio[13837]: : <core> [pass_fd.c:283]: ERROR: receive_fd: EOF on 12 Nov 24 13:35:58 lennysrv /usr/local/sbin/kamailio[13812]: ALERT: <core> [main.c:721]: child process 13821 exited by a signal 11 Nov 24 13:35:58 lennysrv /usr/local/sbin/kamailio[13812]: ALERT: <core> [main.c:724]: core was generated Nov 24 13:35:58 lennysrv /usr/local/sbin/kamailio[13812]: INFO: <core> [main.c:736]: INFO: terminating due to SIGCHLD Nov 24 13:35:58 lennysrv /usr/local/sbin/kamailio[13814]: INFO: <core> [main.c:787]: INFO: signal 15 received *************************************************** However, the module documentation of the source code (READMEs) does not include major news. The configuration is "the same" (regarding handle_publish, handle_subscribe, pua_set_publish) as I've used in the kamailio-1.5.3 version. But - no NOTIFY messages with XML bodies are sent to the subscribers / watchers. Instead of using the setflag(X) method I´ve tried using the method "dlg_setflag(X)" for prohibiting crashes - that was accepted by kamailio. Were these modules tested (in combination with the dialog module) in the new version 3.0 RC1? - presence - presence_xml - presence_dialoginfo - pua - pua_dialoginfo - pua_usrloc It does not work in the SER variant of sip-router, too. Yesterday I've downloaded the current version with GIT, compiled the code, adapted the configuration. But the result is the same as with kamailio-3 RC1 (crash and no XML). For being sure that I have not made any mistake in the configuration, I've tested the functionality in the version 1.5.3 of Kamailio today. It works fine - the subscribers receive NOTIFY messages with XML body after every PRESENCE and DIALOG state change. Can anybody give me a hint? Thanks in advance! regards, Klaus P.S. the tested versions: PROBLEMS: ======== version: kamailio 3.0.0-rc1 (i386/linux) 78f069 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. @(#) $Id$ main.c compiled on 13:34:37 Nov 20 2009 with gcc 4.3.2 version: ser 2.99.99-pre3 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. @(#) $Id$ main.c compiled on 11:12:49 Nov 23 2009 with gcc 4.3.2 SUCCESS: ======= version: kamailio 1.5.3-notls (i386/linux) flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 13:31:21 Nov 24 2009 with gcc 4.3.2 -- Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3.5 - sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser

3 4

Looking for documentation with SER 2.0
by inge 27 Nov '09

27 Nov '09

Hello here, I'm testing SER 2.0 on a dev plateform. I see that there's a lot of modifications beside SER 0.9.9. But I have a lot of pain to find documentation like SER Getting Started for 0.9.x to be able to understand the new routing method (ie. ser.cfg) and the databases scheme and all other parameters at the top of the config file. I install SERWeb too, the version 2.1, and it looks like very different too (don't know how to create an admin user, separate greeting/sip password, etc.). What's your recommandation about SERWeb version with SER2.0 ? Regards, Adrien .L

3 4

[Kamailio-Users] Inbound NAT detection and fix_nated_register() very poorly documented
by Alex Balashov 27 Nov '09

27 Nov '09

Greetings, I was faced with the following problem in Kamailio 1.5.2: I was using nathelper + rtpproxy + registrar and needed to relay media for calls going inbound to a NAT'd registrant through rtpproxy - only if the registrant is NAT'd. Here is the scenario: Media gateway Kamailio (on public IP) ----> registrar -----> NAT'd registrant (w/nathelper) I was handling registrations from NAT'd endpoints like this: if(nat_uac_test("1")) { force_rport(); fix_nated_contact(); } I was aware of fix_nated_register() but was not clear on its relationship to the 'received_param' parameter stored in 'location' and set as a modparam to both the 'registrar' and 'nathelper' modules, so I was not using it. Very quickly I ran into an obvious problem: if the contact stored in the 'contact' column in 'location' is already fixed up with the received IP:port, there is no way to know that the endpoint to which the call is going is behind NAT -- when lookup() is called, the RURI is set to the public IP:port. There are no flags of any kind that one can set during save() that persist while the contact binding is present for that AOR and can be resurrected on lookup(). Anyway, I eventually figured out how to fix this problem by empirical means, with no clear help from the nathelper documentation. It turns out that if I set the 'received_param' as a modparam to 'nathelper' and 'registrar' and handle registrations from NAT'd endpoints with fix_nated_register() instead, it will magically work. I made the following discoveries to arrive at this conclusion, both of which are not documented. This is why it will work: 1) When the original (RFC1918) 'contact' is stored in the 'location' table (if using DB, which I am), the 'received' parameter is stored alongside it and contains the public IP:port. When lookup() is called, the RURI domain is set to private address, e.g. if(!lookup("location")) { # Error handling here exit; } xlog("L_INFO", "[R-2:$ci] -> Registration resolved to RURI: $ru\n"): $ru here will contain something like this: sip:s@10.1.0.2:5060, the original and unmodified contact supplied by the UAC. But somehow, magically, when t_relay() is called the request will be relayed to a different RURI - one with the 'received' IP:port substituted in the domain portion. So, the request goes end up going to the correct place. This is, of course, because of the way the 'received' parameter is supposed to work when appended to a Contact URI. But the point is that it is not documented; nowhere in the documentation for lookup() or t_relay() does it say that this will transpire, and I have no way of knowing it except by observation. 2) For some reason, nat_uac_test("1") returns a positive result after the lookup() and confirms that the destination is NAT'd. This allows me to set a flag and then mangle the SDP in the reply to use rtpproxy for NAT traversal of media as well, e.g. route[2] { ... if(!lookup("location")) { # Error handling here exit; } if(nat_uac_test("1")) setflag(9); ... t_on_reply("1"); if(!t_relay()) sl_reply_error(); ... } ... onreply_route[1] { if(t_check_status("(180|183|200)")) { if(nat_uac_test("5")) fix_nated_contact(); if(search("Content-Type: application/sdp")) { if(isflagset(9)) { set_rtp_proxy_set("1"); force_rtp_proxy(); } } } } But if you examine carefully the documentation for nat_uac_test(), it says the following for bit flag 1: * 1- Contact header field is searched for occurrence of RFC1918 addresses. To me this means that nat_uac_test() should not work after the lookup() above. The only Contact header value that is present in the inbound INVITE handler is the Contact URI of the media gateway, which is a public IP address! Yet for some reason it works, as if by magic; apparently it is somehow implicit that a hidden "received" attribute of the RURI is also part of this check. This is also not documented, and is completely counterintuitive. Is there any possibility of clearing up documentation as to this point? Perhaps I have done something wrong here unknowingly; I have no idea. I know that my solution works but I cannot justify why in terms of the documentation. Is it very much to ask that the documentation be explicit about hidden but critically important mysteries like this? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

2 6

[Kamailio-Users] tmx.so: undefined symbol: _tm_table
by Andrey Utkin 27 Nov '09

27 Nov '09

Hello all. I try to use config from http://sip-router.org/wiki/migration/kamailio-3.0-config I use a kamailio compiled from git://git.sip-router.org/sip-router, branch kamailio_3.0, commit 9b482424cb8aff4050426033908e7e243f03ae83 Kamailio fails to launch with such error: root@oba:~# kamailio 0(24231) DEBUG: <core> [cfg.y:1551]: loading modules under /usr/lib/kamailio loading modules under /usr/lib/kamailio 0(24231) DEBUG: <core> [cfg.y:1544]: loading module modules/db_mysql.so 0(24231) DEBUG: <core> [sr_module.c:349]: load_module: module file not found <modules/db_mysql.so> 0(24231) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </usr/lib/kamailio/modules/db_mysql.so> 0(24231) DEBUG: <core> [cfg.y:1544]: loading module modules_k/mi_fifo.so 0(24231) DEBUG: <core> [sr_module.c:349]: load_module: module file not found <modules_k/mi_fifo.so> 0(24231) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </usr/lib/kamailio/modules_k/mi_fifo.so> 0(24231) DEBUG: <core> [cfg.y:1544]: loading module modules_k/kex.so 0(24231) DEBUG: <core> [sr_module.c:349]: load_module: module file not found <modules_k/kex.so> 0(24231) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </usr/lib/kamailio/modules_k/kex.so> 0(24231) DEBUG: <core> [cfg.y:1544]: loading module modules_k/sl.so 0(24231) DEBUG: <core> [sr_module.c:349]: load_module: module file not found <modules_k/sl.so> 0(24231) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </usr/lib/kamailio/modules_k/sl.so> 0(24231) DEBUG: <core> [cfg.y:1544]: loading module modules/tm.so 0(24231) DEBUG: <core> [sr_module.c:349]: load_module: module file not found <modules/tm.so> 0(24231) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </usr/lib/kamailio/modules/tm.so> 0(24231) DEBUG: <core> [cfg.y:1544]: loading module modules_k/tmx.so 0(24231) DEBUG: <core> [sr_module.c:349]: load_module: module file not found <modules_k/tmx.so> 0(24231) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </usr/lib/kamailio/modules_k/tmx.so> 0(24231) ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/lib/kamailio/modules_k/tmx.so>: /usr/lib/kamailio/modules_k/tmx.so: undefined symbol: _tm_table 0(24231) : <core> [cfg.y:3210]: parse error in config file //etc/kamailio/kamailio.cfg, line 117, column 12-29: failed to load module .... Has anybody succeed to load this module from last git sources?

2 1

[Kamailio-Users] Presence
by Andrey Utkin 27 Nov '09

27 Nov '09

Hello, all. My objective is to make a kamailio installation where phones call to each other by number and are able to monitor others using SUBSCRIBE. I use Kamailio 3.0.0 rc1 pulled from git. My config file is http://pastebin.ca/1675069 I had bad results with establishing calls, possibly because i frequently restarted kamailio, only once i had INVITE passing from one phone to other. RTPProxy mustn't work on a box with grey IP, i know. But my greatest convern now is notifications. I just want to say that i never got NOTIFYs with xml containing bodies, what was usual in Asterisk. First of all NOTIFYs that go immediately after SUBSCRIBEs also have no xml body. Please direct me what is wrong in my configuration regarding presence setup. Tell me what system info to provide.

2 2

[Kamailio-Users] we don't follow the RFCs!
by BERGANZ François 27 Nov '09

27 Nov '09

Hello all, As I could see in the source, we dont follow the RFCs. If I send an INVITE, kamailio response a Giving a try It isnt in the RFCs! It is normaly Trying. I could change it in -src/modules/tm/t_funcs.c -line: static str relay_reason_100 = str_init("Giving a try"); I hope you understand what I mean François BERGANZ Ingénieur R&D VoIP Acropolis Telecom P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

8 18

Re: [Kamailio-Users] i want to get user defined header value
by Alex Balashov 27 Nov '09

27 Nov '09

Please don't forget to copy your responses back to the list so that others can benefit from this discussion. I must have misunderstood your intended meaning of "user-defined." $hdr(...) takes only string literals as arguments, not pseudovariables. I do not believe that what you are wanting is possible. RAJNIKANT VANZA wrote: > Hi Alex, > > i have tested like : > > 1. ${hdr(TEST)) > 2. $hdr($avp(s:TEST)) > > but it's not work. > if u know any other suggestion, send me. > > On Mon, Nov 23, 2009 at 3:56 PM, Alex Balashov > <abalashov(a)evaristesys.com <mailto:abalashov@evaristesys.com>> wrote: > > RAJNIKANT VANZA wrote: > > i want to get user-defined header value. > e.g. i have create header name : TEST and its value : vtec in > asterisk. > then how can i get this value from this header in kamailio. > > > Try: $hdr(TEST) > > -- > Alex Balashov - Principal > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > > > > > -- > Thanks and Regards > Rajnikant Vanza > Call : +91-97374 56583 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

5 11

[Kamailio-Users] prb while loading presence.so
by ioanna lampropoyloy 27 Nov '09

27 Nov '09

Openser cant start while i am trying to load presence.so I get the following error Nov 24 12:57:11 localhost openser: ERROR: load_module: could not open module <//lib/openser/modules/presence.so>: //lib/openser/modules/presence.so: undefined symbol: ctime_buf Nov 24 12:57:11 localhost openser: parse error (58,13-14): failed to load module and i also get errors which refer to the parameters of presence since the module cant be loaded. Any ideas?

2 1

[Kamailio-Users] delete the SDP?
by BERGANZ François 27 Nov '09

27 Nov '09

Hello, How can I delete all the SDP ? Thank you François BERGANZ P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

3 2

how i should process replies on nathelper pings?
by Andres Moya 27 Nov '09

27 Nov '09

Hi. nathelper sending empty OPTIONS messages to clients. It have only one VIA, server's IP address(195.167.152.55). Then it got reply i have: U 195.167.152.55:5060 -> 81.195.165.250:5060 OPTIONS sip:81.195.165.250:5060 SIP/2.0..Via: SIP/2.0/UDP 195.167.152.55:5060;branch=0..From: sip:pinger@riki.ru;tag=e64c73d5..To: sip:81.195.165.250:5060..Call-ID: 680bed34-d2e5c741-0eff517(a)195.167.152 .55..CSeq: 1 OPTIONS..Content-Length: 0.... # U 81.195.165.250:5060 -> 195.167.152.55:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 195.167.152.55:5060;branch=0..Contact: <sip:10.0.52.103:5060>..To: <sip:81.195.165.250:5060>;tag=021c6509..From: <sip:pinger@riki.ru>;tag=e64c73d5..Call-ID: 680bed34-d2e 5c741-0eff517@195.167.152.55..CSeq: 1 OPTIONS..Accept: application/sdp, application/sdp..Accept-Language: en..Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE..User-Agen t: Zoiper rev.5324..Allow-Events: presence..Content-Length: 0.... Nov 25 23:59:26 v /usr/local/sbin/kamailio[9697]: ERROR: <core> [forward.c:695]: ERROR: forward_reply: no 2nd via found in reply # U 195.167.152.55:5070 -> 86.154.150.209:5060 OPTIONS sip:86.154.150.209:5060 SIP/2.0..Via: SIP/2.0/UDP 195.167.152.55:5070;branch=0..From: sip:pinger@riki.ru;tag=f64c73d5..To: sip:86.154.150.209:5060..Call-ID: 680bed34-e2e5c741-7eff517(a)195.167.152 .55..CSeq: 1 OPTIONS..Content-Length: 0.... # U 86.154.150.209:5060 -> 195.167.152.55:5070 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 195.167.152.55:5070;branch=0..To: <sip:86.154.150.209>;tag=931uruqh0thc6crbneub..From: <sip:pinger@riki.ru>;tag=f64c73d5..Call-ID: 680bed34-e2e5c741-7eff517(a)195.1 67.152.55..CSeq: 1 OPTIONS..Content-Length: 0.... Nov 25 23:59:33 v /usr/local/sbin/kamailio[9698]: ERROR: <core> [forward.c:695]: ERROR: forward_reply: no 2nd via found in reply Reply should be consumed by nathelper? or i should eat it somewhere? i can't do so in request route. and i can't use reply route because i can't set t_on_reply, it is not my request. There should be easy way... tired get ERROR: forward_reply: no 2nd via found in reply thanks

3 7

[Kamailio-Users] Dialplan regular expressions
by alex pappas 26 Nov '09

26 Nov '09

Dear all, I'm having difficulties in the dialplan module. My problem is in the regular expressions added to the database. I tried some examples from previous users but they don't work for me. example: dpid 10 pr 100 match_op 0 match_exp ^123(.*)$ match_len 0 subst_exp ^123 repl_exp 0030 Previous example from Daniel that I found in user lists: s/subst_exp/repl_exp/ subst_exp - is regular expression that can be used to group parts of the input repl_exp - is used to build the output Example: you want to remove the first 0 and prefix +44 to a number: subst_exp=^0(.+)$ repl_exp=+44\1 Can someone provide an example that works with the latest kamailio please? Also where I can find more docs about dialplan? Thank you all Alex

3 7

[Serusers] newbie
by Carlos Felipe Solano Fernandez 26 Nov '09

26 Nov '09

Greetings. I had to read some thigs about SER. I like configure a sip server but I don't know that I need. I like to take calls (pc-phone)(ip phone-phone) from my customers, and resend them to a voip provider. and I need a billing system (preaid), to control my customers, I need to configure several rates for calls. Can somebody tell me that I need to make this? I'm sorry but my english isn't very well. (I speak spanish) Atte, Felipe Solano Explore the seven wonders of the world Learn more! _________________________________________________________________ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx

1 0

[Kamailio-Users] RTCP
by kamailio.org＠spam.lublink.net 26 Nov '09

26 Nov '09

Hey, I was wondering how I can log all the RTCP messages and automatically associate them with the CDRs? Thanks, David

5 6

[Kamailio-Users] tm retransmits and times out even though reponse is received
by Alex Hermann 26 Nov '09

26 Nov '09

Hello, I'm seeing an increasing amount of spurious timeouts on my proxies. Even though the proxy receives a "100 Trying" response immediately after relaying the request, tm will send retransmits and times out eventually. Below is a commented log snippet. The timeout occurs on the proxy named 'balancer1'. Both are on the same host. I verified that the 100 Trying response is received in a t_on_reply block, so it is recognized by tm as belonging to a transaction. When needed, I'll provide a debug log when it occurs again. Meanwhile, any idea what is causing this? This is on kamailio 1.5+svn5768 fr_timer = 10 fr_timer_avp = 2 # Initial request received on balancer May 15 12:32:01 balancer1[24170]: [2 INVITE] - New request. <sip:uac@domain> => <sip:uas@domain> May 15 12:32:01 balancer1[24170]: [2 INVITE] Relaying # Initial request received on proxy May 15 12:32:01 proxy1[23803]: [2 INVITE] - New request. <sip:uac@domain> => <sip:uas@domain> # Balancer receives 100 response May 15 12:32:01 balancer1[24166]: [2 INVITE] - Reply. 100 Trying # Proxy relays request May 15 12:32:01 proxy1[23803]: [2 INVITE] Relaying # Proxy receives retransmission from balancer ??? May 15 12:32:02 proxy1[23804]: [2 INVITE] - New request. <sip:uac@domain> => <sip:uas@domain> # Balancer times out ??? May 15 12:32:02 balancer1[24171]: [2 INVITE] Failure (Status: 408) May 15 12:32:02 balancer1[24171]: [2 INVITE] No more gateways May 15 12:32:02 proxy1[23804]: [2 CANCEL] - New request. <sip:uac@domain> => <sip:uas@domain> -- Greetings, Alex Hermann

2 4

[Kamailio-Users] DSYSLOG_ASYNC option in Makefile.defs
by Torben Friese 26 Nov '09

26 Nov '09

Hi all, I have a problem with the Kamailio 1.5, if the DSYSLOG_ASYNC option is set I receive no log messages on a lenny-64bit system(fresh installed). Rsyslog is the standard log daemon in lenny. If I start directly from the command line with -dd I receive log messages. Any hints or ideas what is going wrong here ? The same option on a etch system works fine. best regards and thanking you in anticipation Torben

2 1

[Kamailio-Users] Commercial 24x7x365 Support
by Martin Werne 26 Nov '09

26 Nov '09

Hi! I have a customer (service-provider) that want to implement redundant openSER type of solution. On two locations (Europe and USA). They need access to high level support 24 hours a day, 365 days a year (by phone and remote-assistance via VPN). Do you know of any companies that delivers this kind of service? Anyone that has a list of companies that delivers solutions based on openSER? As I'm only one man, I'm not able to do this by myself...... ;-) Best regards, Martin

2 1

[Kamailio-Users] Core dump on xl_get_pai in Openser 1.2.2
by Marcello Lupo 26 Nov '09

26 Nov '09

Hi to all, i implemented the P-Asserted-Identity and P-Preferred-Identity Header management. On the development server all is working properly but when i activate the new config on our production system i start to get crashes of the openser process: openser[31483]: segfault at 0000000000000000 rip 0000000000487853 rsp 00007fff94c9f1a0 error 4 openser[6511]: segfault at 0000000000000025 rip 00000000004214ce rsp 00007fff80f05480 error 4 openser[8967] general protection rip:4214ce rsp:7fff05fbb530 error:0 openser[9872] general protection rip:4214ce rsp:7fffd2b630d0 error:0 I checked the core dump with gdb and i have this info from the core: Core was generated by `/usr/sbin/openser'. Program terminated with signal 11, Segmentation fault. #0 xl_get_pai (msg=0x2b4ad9030d00, res=0x7fffd2b63190, param=<value optimized out>, flags=<value optimized out>) at items.c:1050 1050 res->rs.len = get_pai(msg)->uri.len; We are using the 1.2.2 version and i know it is quite old but i like to understand if it happen for some known bug or if i made something wrong. For the moment is not possible for us to manage the upgrade to a newer version of the proxy till we will not have it redundant on another location. Thank you for help. Bye, MArcello

2 3

[Kamailio-Users] Grails Runtime Exception while adding user with SerMyAdmin
by Fred 25 Nov '09

25 Nov '09

Hello, when i push create button, I get the following: Grails Runtime Exception Error Details Message: org.apache.commons.codec.digest.DigestUtils Caused by: java.lang.NoClassDefFoundError: org.apache.commons.codec.digest.DigestUtils Class: UserController At Line: [226] Code Snippet: Stack Trace org.codehaus.groovy.runtime.InvokerInvocationException: java.lang.NoClassDefFoundError: org.apache.commons.codec.digest.DigestUtils at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:290) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationDispatcher.invoke(ApplicationDispatcher.java:630) at org.apache.catalina.core.ApplicationDispatcher.processRequest(ApplicationDispatcher.java:436) at org.apache.catalina.core.ApplicationDispatcher.doForward(ApplicationDispatcher.java:374) at org.apache.catalina.core.ApplicationDispatcher.forward(ApplicationDispatcher.java:302) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.jsecurity.web.servlet.JSecurityFilter.doFilterInternal(JSecurityFilter.java:384) at org.jsecurity.web.servlet.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:183) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:233) at org.apache.catalina.core.StandardContextValve.invoke(StandardContextValve.java:175) at org.apache.catalina.core.StandardHostValve.invoke(StandardHostValve.java:128) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:102) at org.apache.catalina.core.StandardEngineValve.invoke(StandardEngineValve.java:109) at org.apache.catalina.connector.CoyoteAdapter.service(CoyoteAdapter.java:286) at org.apache.coyote.http11.Http11Processor.process(Http11Processor.java:844) at org.apache.coyote.http11.Http11Protocol$Http11ConnectionHandler.process(Http11Protocol.java:583) at org.apache.tomcat.util.net.JIoEndpoint$Worker.run(JIoEndpoint.java:447) at java.lang.Thread.run(Thread.java:619) Caused by: java.lang.NoClassDefFoundError: org.apache.commons.codec.digest.DigestUtils at UserController$_closure11.class$(UserController.groovy) at UserController$_closure11.doCall(UserController.groovy:226) at UserController$_closure11.doCall(UserController.groovy) ... 28 more Unfortunately, i am not familiar with such java stuff. It would be great if one of you can give me a hint or something. I dont have any idea how to solve this problem. Thanks for any comment! Best regards, Fred

3 2

[Kamailio-Users] /usr/local/sbin/kamdbctl create error
by vivi 25 Nov '09

25 Nov '09

Hi all I have specify the wanted db type (DBENGINE=MYSQL) in the /usr/local/etc/kamailio/kamctlrc then I using "/usr/local/sbin/kamdbctl create" to create MySQL database, but I got this error: ERROR: database engine not specified, please setup one in the config script root@acer:/usr/local/src/kamailio-1.5.0/sip-server# vim /usr/local/etc/kamailio/kamctlrc root@acer:/usr/local/src/kamailio-1.5.0/sip-server# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR: Creating core database and grant privileges failed! Cheers, vivi

3 2

[Kamailio-Users] Call duration with RADIUS support
by mbsip 25 Nov '09

25 Nov '09

Hi, I am struggling with kamailio and radius configuration to achieve proper Call duration notification. Here is what i have: - kamailio 1.6 - radiusclient-ng - freeradius on different server Kamailio configuration has following radius-extra: modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$ruri; \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") which produce following radius data START and STOP: Mon Nov 23 16:26:38 2009 Acct-Status-Type = Start Service-Type = IAPP-Register Sip-Response-Code = 200 Sip-Method = Invite Event-Timestamp = 1258990835 Sip-From-Tag = "c3716c01" Sip-To-Tag = "11544D08-C80" Acct-Session-Id = "Nzg5MmZlMDNkMGUwODgwN2ZkZWZkMmQ3NzYwYzNkNWY." User-Name = "Mac(a)xx.yy.zz.vv" Calling-Station-Id = "sip:Mac@xx.yy.zz.vv" Called-Station-Id = "sip:123456789@xx.yy.zz.vv" Sip-Translated-Request-URI = "sip:123456789@aa.bb.cc.dd" X-Ascend-Send-Secret = 0x3135372e32352e3139322e32 X-Ascend-Receive-Secret = 0x3a7c X-Ascend-Netware-timeout = 0x3c7369703a4d616369656a403139322e3136382e332e3132353a36323034323e NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = xx.yy.zz.vv Client-IP-Address = xx.yy.zz.vv Acct-Unique-Session-Id = "a62f94528d34acfe" Timestamp = 1258989998 Mon Nov 23 16:26:38 2009 Acct-Status-Type = Stop Service-Type = IAPP-Register Sip-Response-Code = 200 Sip-Method = 8 Event-Timestamp = 1258990836 Sip-From-Tag = "c3716c01" Sip-To-Tag = "11544D08-C80" Acct-Session-Id = "Nzg5MmZlMDNkMGUwODgwN2ZkZWZkMmQ3NzYwYzNkNWY." User-Name = "Mac(a)xx.yy.zz.vv" Calling-Station-Id = "sip:Mac@xx.yy.zz.vv" Called-Station-Id = "sip:123456789@xx.yy.zz.vv" Sip-Translated-Request-URI = "sip:123456789@aa.bb.cc.dd:5060" X-Ascend-Send-Secret = 0x3135372e32352e3139322e32 X-Ascend-Receive-Secret = 0x3a7c X-Ascend-Netware-timeout = 0x3c7369703a4d616369656a403139322e3136382e332e3132353a36323034323e NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = xx.yy.zz.vv Client-IP-Address = xx.yy.zz.vv Acct-Unique-Session-Id = "a62f94528d34acfe" Timestamp = 1258989998 The task is to upload all stored radius messages into MYSQL with proper call duration info. I've ran through many posts and found a few way-outs: 1) to use AVPOPS module: modparam("acc", "radius_extra", "Acct-Session-Time=$avp(call_length)") modparam( "avpops", "avp_aliases", "start_timestamp=i:100") modparam( "avpops", "avp_aliases", "end_timestamp=i:101") modparam( "avpops", "avp_aliases", "call_length=i:102") avp_op("$avp(end_timestamp)","sub/$avp(start_timestamp)"); avp_op("$avp(end_timestamp)/$avp(call_length)","sub/$avp(start_timestamp)"); 2) to use DIALOG module modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_extra", "Acct-Session-Time=$avp(i:921)") if (is_method("BYE")) { xlog("L_INFO", "dialog_status:$DLG_status duration=$DLG_lifetime\n"); $avp(i:921) = $DLG_lifetime{s.int}; } else { $avp(i:921) = 0; } setflag(1); 3) to use a script or db triggers(after STOP). Could you advice which one is better/work properly. Maybe there are any other configuration i could use. Thanks in advance, Maciej

2 2

[Kamailio-Users] Same "RADIUS Problem"
by Masoom Alam 25 Nov '09

25 Nov '09

I am having the same problem: I am configuring RADIUS with openser and CDRtool. when i dial a number it loges into in mysql of RADIUS server but it gives the following error. and when i check CDRs of CDRtool it shows all calls "*inprogress". * Tue Nov 24 19:47:37 2009 : Error: rlm_sql (sql) in sql_accounting: stop packet with zero session length. [user '1000@ip address'<%271000@ip%20address%27>, nas 'ipaddress'] Can any body help me, that whether the RADIUS server is not sending the INVITE and BYE traffic correctly or not singallng the CDRTool as well. Thanking you in advance.

2 1

[Kamailio-Users] RADIUS problem
by toqeer ali 24 Nov '09

24 Nov '09

I am configuring RADIUS with openser and CDRtool. when i dial a number it loges into in mysql of RADIUS server but it gives the following error. and when i check CDRs of CDRtool it shows all calls "*inprogress". * Tue Nov 24 19:47:37 2009 : Error: rlm_sql (sql) in sql_accounting: stop packet with zero session length. [user '1000@ip address'<'1000@ip%20address'>, nas 'ipaddress'] -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

1 0

[Kamailio-Users] please help
by Masoom Alam 24 Nov '09

24 Nov '09

Hi i am configuring RADIUS accounting with openser and i facing the problem with radius_extra module in acc all other parameters are ok but when i add $hdr something i got the following error. parameters with avp are fine but problem is with $hdr. modparam("acc", "radius_extra", ......; X-RTP-Stat=$hdr (X-RTP-Stat; ......) Nov 24 13:30:50 kamailio /sbin/openser[3760]: xl_fill_extra_spec: extra item [hdr] not found Nov 24 13:30:50 kamailio /sbin/openser[3760]: xl_lookup_spec_name: not found PV [hdr] Nov 24 13:30:50 kamailio /sbin/openser[3760]: ERROR:xl_parse_spec: error searching pvar "hdr" Regards, MM Alam SERG Pakistan http://serg.imsciences.edu.pk http://www.imsciences.edu.pk/profiles/masoom_alam.html

2 1

[Kamailio-Users] Strip core function with variable
by Antonio Reale 24 Nov '09

24 Nov '09

Hi all, strip core function accepts only an integer as parameter. What do you thing about implementing the strip function with a variable as parameter in a future release? strip($var(x)); or something like that... Is this doable? Best regards. -- Antonio

2 2

Re: [SR-Users] [sr-dev] recursive calls to failure_route strange behavior
by Miklos Tirpak 23 Nov '09

23 Nov '09

[please keep the list CC-d because others may also be interested in the solution or may know the answer better.] On 11/20/2009 07:06 PM, Andres Moya wrote: > Can i ask one more question here. It is complicated, i am using > uac_replace_from and uac_auth. > I am using failure route to process authentication if necessary and > redirect on next carrier. > > If i authenticated with one provider, then fot let say 415 ( i set only > speex in UAC to get it ;) ). Ok SER send request to second provider, i use > uac_replace_from once again in my LOAD_AUTH route > then uac_auth again. > > Ok. now i see from ngrep that uac_replace_from did nothing in from field > and use user@domain for first provider, authentication failed :( Which authentication fails? The first or the second one? The first should work, at least the from header should be rewritten by the function. The second authentication will not work this way (if it requires a different from header) because the proxy "remembers" for the header changes done before the first t_relay() function call and applies the same header modifications also for any other branch added from failure route. Hence, the outgoing SIP request to the second provider will contain the same from HF as the request to the first provider. The easiest way is to apply the header modifications in branch route if you do not need to reuse them later from failure route. Modifications done in branch routes are valid only within that branch. > > I moved uac_replace_from to my failure route to call once again before > uac_auth, but got config error as i can't use uac_replace_from in > failure_route. I think you already use this function from failure route because it is in a route block that is included from failure route. The only difference is that the syntax checker does not recognize the issue. I am not familiar with uac_replace_from() but after having a quick look at the function I think it is safe to use it from here. > Ok i will use textops to rewrite, but it is ugly? The main difference is that uac_replace_from() restores the original From HF when the response if forwarded. If you use textops module then you need to restore the header manually. > no. Maybe i should > call uac_replace_from in branch route? I would suggest this way. Both for the first and for the second provider. Miklos > > Thanks

3 6

Re: [Kamailio-Users] i want to get user defined header value
by RAJNIKANT VANZA 23 Nov '09

23 Nov '09

On Mon, Nov 23, 2009 at 5:54 PM, RAJNIKANT VANZA <rajnivanza(a)gmail.com>wrote: > On Mon, Nov 23, 2009 at 4:20 PM, Thiago Rondon <thiago(a)aware.com.br>wrote: > >> >> Ranikant, >> >> You can you use Perl module for this, please look at : >> >> http://kamailio.org/docs/modules/1.6.x/perl.html >> >> See the script example, headers.pl >> >> -Thiago Rondon >> >> >> Alex Balashov escreveu: >> >> Please don't forget to copy your responses back to the list so that >>> others can benefit from this discussion. >>> >>> I must have misunderstood your intended meaning of "user-defined." >>> $hdr(...) takes only string literals as arguments, not pseudovariables. >>> >>> I do not believe that what you are wanting is possible. >>> >>> RAJNIKANT VANZA wrote: >>> >>> Hi Alex, >>>> >>>> i have tested like : >>>> >>>> 1. ${hdr(TEST)) >>>> 2. $hdr($avp(s:TEST)) >>>> >>>> but it's not work. >>>> if u know any other suggestion, send me. >>>> >>>> On Mon, Nov 23, 2009 at 3:56 PM, Alex Balashov < >>>> abalashov(a)evaristesys.com <mailto:abalashov@evaristesys.com>> wrote: >>>> >>>> RAJNIKANT VANZA wrote: >>>> >>>> i want to get user-defined header value. >>>> e.g. i have create header name : TEST and its value : vtec in >>>> asterisk. >>>> then how can i get this value from this header in >>>> kamailio. >>>> >>>> >>>> Try: $hdr(TEST) >>>> >>>> -- Alex Balashov - Principal >>>> Evariste Systems >>>> Web : http://www.evaristesys.com/ >>>> Tel : (+1) (678) 954-0670 >>>> Direct : (+1) (678) 954-0671 >>>> >>>> >>>> >>>> >>>> -- >>>> Thanks and Regards >>>> Rajnikant Vanza >>>> Call : +91-97374 56583 >>>> >>> > > Hi friends, > > thank for reply. > > i have add header through asterisk api by below statement and its display > correct on asterisk CLI > but not able to get header value through kamailio: > > *asteisk statement is :* SipAddheader(TEST=vtec); > above statement in header name is "TEST" and its value is "vtec". > > i want get "header TEST" from sip message when kamailio receive sip message > (INVITE) from asterisk server. > > > -- > Thanks and Regards > Rajnikant Vanza > Call : +91-97374 56583 > -- Thanks and Regards Rajnikant Vanza Call : +91-97374 56583

1 0

[Kamailio-Users] i want to get user defined header value
by RAJNIKANT VANZA 23 Nov '09

23 Nov '09

Hi friends, i want to get user-defined header value. e.g. i have create header name : TEST and its value : vtec in asterisk. then how can i get this value from this header in kamailio. -- Thanks and Regards Rajnikant Vanza

2 1

[Kamailio-Users] how can i get sip header value
by RAJNIKANT VANZA 23 Nov '09

23 Nov '09

Hi Friends, i am a new user for kamailio. let me know, how can i get sip header value ? -- Thanks and Regards Rajnikant Vanza

2 1

[Kamailio-Users] [kamailio-users] Read SIP Header from call request
by Chandrakant Solanki 23 Nov '09

23 Nov '09

Hi I am using kamailio 1.5.0.2 How can i read SIP Header in kamailio for every sip request. Is there any method / function which will help to read SIP Header...?? -- Regards, Chandrakant Solanki

2 1

[Kamailio-Users] TCP problem/question
by Torben Friese 21 Nov '09

21 Nov '09

Hi all, I need a hint regarding TCP handling. I want to receive messages as TCP and relay or forward them as UDP. This is realized with dispatcher module. A explicit entry with sip:IP;transport=udp in the dispatcher list. Receiving via TCP works fine, but the proxy send the request also via TCP out. Any ideas where I can search or how to debug the problem ? best regards Torben

3 2

[Kamailio-Users] ERROR:mi_fifo:mi_fifo_server: fifo command must begin with :: early_dialogs (1.5 rev 5834)
by Iñaki Baz Castillo 21 Nov '09

21 Nov '09

Hi, I use the Munin plugin I did in various servers running Kamailio 1.5 (Debian 64 bits). The munin plugins performs the Mi command: kamctl fifo get_statistics inuse_transactions In one of them the installed revision is 5834 and the plugin works perfectly, no errors in 3 months with high traffic. However in the other server (revision 5925) there are no traffic yet but the plugin fails sometimes (maybe once each 2-3 days). Sometimes it gets an error: ERROR:mi_fifo:mi_fifo_server: fifo command must begin with :: early_dialogs and sometimes it gets wrong value: kamctl fifo get_statistics inuse_transactions => 2819624 (with no traffic !!!) In the second case the Munin grapsh gets corrupted due to so high value. Could it be a bug in dialog module? Should I upgrade to revision 5939 in which I see a change for modules/dialog/dlg_handlers.c: ------------------------------------------------------------------------------------------------------------------------- ~# svn diff -r HEAD modules/dialog/dlg_handlers.c Index: modules/dialog/dlg_handlers.c =================================================================== --- modules/dialog/dlg_handlers.c (revisión: 5939) +++ modules/dialog/dlg_handlers.c (copia de trabajo) @@ -824,14 +824,13 @@ if(dlg->toroute>0 && dlg->toroute<RT_NO) { + dlg_set_ctx_dialog(dlg); fmsg = faked_msg_next(); if (exec_pre_req_cb(fmsg)>0) { - dlg_set_ctx_dialog(dlg); LM_DBG("executing route %d on timeout\n", dlg->toroute); set_route_type(REQUEST_ROUTE); run_top_route(rlist[dlg->toroute], fmsg); - dlg_set_ctx_dialog(0); exec_post_req_cb(fmsg); } } ------------------------------------------------------------------------------------------------------------------------ I see some differences between the dialog module versions running in both servers, but I think the main difference is a bug fix for the case in which there is empty Record-Route in the response (Kamailio crashed before the fix): ----------------------------------------------------------------------------------------------------------------------- # svn diff -r 5834:5925 modules/dialog/dlg_handlers.c Index: modules/dialog/dlg_handlers.c =================================================================== --- modules/dialog/dlg_handlers.c (revisión: 5834) +++ modules/dialog/dlg_handlers.c (revisión: 5925) @@ -163,10 +163,10 @@ int populate_leg_info( struct dlg_cell *dlg, struct sip_msg *msg, struct cell* t, unsigned int leg, str *tag) { - unsigned int skip_recs; - str cseq; - str contact; - str rr_set; + unsigned int skip_recs = 0; + str cseq = {0, 0}; + str contact = {0, 0}; + str rr_set = {0, 0}; /* extract the cseq number as string */ if (leg==DLG_CALLER_LEG) { @@ -184,48 +184,41 @@ /* extract the contact address */ if (!msg->contact&&(parse_headers(msg,HDR_CONTACT_F,0)<0||!msg->contact)){ LM_ERR("bad sip message or missing Contact hdr\n"); - goto error0; + } else { + if ( parse_contact(msg->contact)<0 || + ((contact_body_t *)msg->contact->parsed)->contacts==NULL || + ((contact_body_t *)msg->contact->parsed)->contacts->next!=NULL ) { + LM_ERR("bad Contact HDR\n"); + } else { + contact = ((contact_body_t *)msg->contact->parsed)->contacts->uri; + } } - if ( parse_contact(msg->contact)<0 || - ((contact_body_t *)msg->contact->parsed)->contacts==NULL || - ((contact_body_t *)msg->contact->parsed)->contacts->next!=NULL ) { - LM_ERR("bad Contact HDR\n"); - goto error0; - } - contact = ((contact_body_t *)msg->contact->parsed)->contacts->uri; - /* extract the RR parts */ if(!msg->record_route && (parse_headers(msg,HDR_RECORDROUTE_F,0)<0) ){ LM_ERR("failed to parse record route header\n"); - goto error0; - } - - if (leg==DLG_CALLER_LEG) { - skip_recs = 0; } else { - /* was the 200 OK received or local generated */ - skip_recs = dlg->from_rr_nb + - ((t->relaied_reply_branch>=0)? - (t->uac[t->relaied_reply_branch].added_rr):0); - } + if (leg==DLG_CALLEE_LEG) { + /* was the 200 OK received or local generated */ + skip_recs = dlg->from_rr_nb + + ((t->relaied_reply_branch>=0)? + (t->uac[t->relaied_reply_branch].added_rr):0); + } - if(msg->record_route){ - if( print_rr_body(msg->record_route, &rr_set, leg, - &skip_recs) != 0 ){ - LM_ERR("failed to print route records \n"); - goto error0; + if(msg->record_route){ + if( print_rr_body(msg->record_route, &rr_set, leg, + &skip_recs) != 0 ){ + LM_ERR("failed to print route records \n"); + } } - } else { - rr_set.s = 0; - rr_set.len = 0; } if(leg==DLG_CALLER_LEG) dlg->from_rr_nb = skip_recs; LM_DBG("route_set %.*s, contact %.*s, cseq %.*s and bind_addr %.*s\n", - rr_set.len, rr_set.s, contact.len, contact.s, - cseq.len, cseq.s, + rr_set.len, (rr_set.len)?rr_set.s:"", + contact.len, (contact.len)?contact.s:"", + cseq.len, (cseq.len)?cseq.s:"", msg->rcv.bind_address->sock_str.len, msg->rcv.bind_address->sock_str.s); ----------------------------------------------------------------------------------------------------------------------- Perhaps a regression in rev 5925 so I should upgrade? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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change from header in branches works strange
by Andres Moya 21 Nov '09

21 Nov '09

Hi, it is me again. I gave up on uac_replace_from, and trying with subst('/^From:(.*)sip:[^@]*@[a-zA-Z0-9\.]+(.*)$/From:\1sip:$avp(i:21)@$avp(i:24)\2/i'); first time it substitute ok, on next branch i got result: From: <sip:andres.moya4@sipdiscount.com;transport=UDP>;tag=ee12f132.sip:andres.moya4@sipdiscount.com Something really strange, looks like there is no way i can authenticate to different SIP providers in one transaction :( Anyone know how i can hack From field actually?

1 0

crazy uac_replace_from
by Andres Moya 21 Nov '09

21 Nov '09

Hi, not sleeping second night. I am trying to authenticate to sequence of SIP provider, if one fail - another. I am calling uac_replace_from for each. uac_replace_from("sip:$avp(i:21)@$avp(i:24)"); first time it set ok: From: <sip:andres.moya4@sipdiscount.com>;tag=572ac509. second time i am getting From: <sip:andres.moya4@sipdiscount.comsip:21100036838@terrasip.net>;tag=572ac509. grrrr.... is it a joke :) Someone ... please help

1 0

[Kamailio-Users] No free TCP receiver
by luis.guaman＠interlancompu.com 19 Nov '09

19 Nov '09

3 2

[Kamailio-Users] Dialog timed calls
by Jason Penton 19 Nov '09

19 Nov '09

Hi All, Has anyone else noticed the significant inaccuracy of the timeout in the dialog module? We are using: modparam("dialog", "timeout_avp", "$avp(i:10)") However, the actual call is torn down a NUMBER of seconds past the timeout supplied. In some instances, for long calls, it is out by 30seconds or more....? For shorter calls it is accurate with a second or so. Anyone have any input, ideas? Cheers Jason

2 1

[Kamailio-Users] How to configure alias_db_lookup in config script
by JR Richardson 18 Nov '09

18 Nov '09

Hi All, I have a real simple load balancer proxy setup using the dispatcher module, works great. I want to do an alias lookup before I send the calls through to the dispatcher, but I can't seem to find a good example of how that should be setup in the config script. I've tried different things, the aliase_db_lookup is working but if a user is not in the database, the call loops and does not continue on to the dispatcher. Here is my current route script (this does not work): route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; alias_db_lookup("dbaliases"); t_on_failure("1"); t_relay(); ds_select_dst("1", "4"); t_on_failure("1"); t_relay(); } failure_route[1] { ds_next_dst(); forward(); } I think there should be an 'if' 'then' or 'if' 'else' separating the alias_db_lookup and the ds_select functions, but I'm not sure of the syntax and where the curly brackets need to be. Thanks. JR -- JR Richardson Engineering for the Masses

2 5

[Kamailio-Users] openser freeradius connectivity error
by toqeer ali 18 Nov '09

18 Nov '09

Hi all, I configured freeradius for accounting of openser. but when i make a call the call establishwa and gives this error for Freeradius. Nov 18 06:50:58 kamailio /sbin/openser[11590]: rc_ip_hostname: couldn't look up host by addr: 0A000005 Nov 18 06:50:58 kamailio /sbin/openser[11590]: rc_send_server: no reply from RADIUS server unknown:1813 Nov 18 06:50:58 kamailio /sbin/openser[11590]: ERROR:acc:acc_rad_request: radius-ing failed Nov 18 06:51:20 kamailio /sbin/openser[11593]: rc_ip_hostname: couldn't look up host by addr: 0A000005 Nov 18 06:51:20 kamailio /sbin/openser[11593]: rc_send_server: no reply from RADIUS server unknown:1813 Nov 18 06:51:20 kamailio /sbin/openser[11593]: ERROR:acc:acc_rad_request: radius-ing failed please help -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

2 3

[Kamailio-Users] Play Early Media
by alex pappas 18 Nov '09

18 Nov '09

Dear all, I'm trying to play Early Media instead of Ringing WHEN an 180 Ringing OR 182 Call Queue message is received from the destination UAC. In the onreply_route[1] I added the following lines in bold: I use kamailio 1.5 and I have NAT enabled. onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"*(180)|*(183)|(2[0-9][0-9])") { force_rtp_proxy(); } * if (status=~"(180)") { rtpproxy_stream2uac("/usr/local/etc/rtpproxy/ast-ivr.alaw",""); } else { rtpproxy_stop_stream2uac(); }* if (isbflagset(6)) { fix_nated_contact(); } } When I get from a UAC an 180 Ringing does not play the media and also in the syslog I see the following ERRORs: Nov 17 14:11:29 YGOR /usr/local/sbin/kamailio[4681]: ERROR:nathelper:extract_body: message body has length zero Nov 17 14:11:29 YGOR /usr/local/sbin/kamailio[4681]: ERROR:nathelper:force_rtp_proxy: can't extract body from the message Nov 17 14:11:29 YGOR /usr/local/sbin/kamailio[4681]: ERROR:nathelper:select_rtpp_node: script error -no valid set selected Nov 17 14:11:29 YGOR /usr/local/sbin/kamailio[4681]: ERROR:nathelper:rtpproxy_stream: no available proxies Thanks in advance for any help Alex

4 8

[Kamailio-Users] Dialplan Module Confusion
by JR Richardson 18 Nov '09

18 Nov '09

Hi All, I'm new to the list and fairly new to the vastness which is kamailio and the tons of modules and options. I'm trying to figure out the simplest method to accomplish the following scenario: SIP call comes in, match DID in database (MySQL), rewrite ruri with associated database entry matching DID, relay call to specified gateway, if no matching entry for DID is in database, forward call to dispatcher module and continue processing call (I already use the dispatcher module for load balancing, just need to implement DID call routing before I send calls to dispatcher) I've been reading about the following modules, dialplan, LCR, drouting, alias_db. I think dialplan is what I really need, but I'm confused about what entries to put in the database to match on 10 digit DID: match-op = 0 (equal) match_exp= 1112223333 match_len= 10 subst_exp= ?? is this where I would put the ruri sip:1112223333@10.10.10.10<sip%3A1112223333(a)10.10.10.10>(I'm not sure of the syntax here) repl_exp= ?? not sure what this is for can more than one database entry have the same id? and also I could use a practical example of how this would work int he config file: dp_translate("240", "$ruri.user/$ruri.user"); t_relay?? Any guidance on this would be much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

2 1

[Kamailio-Users] RADIUS for billing
by Rajesh 17 Nov '09

17 Nov '09

Dear All, I am trying to configure openSER for billing.After a lot og googling, i came to know that RADIUS is needed for anabling billing module. I would like to know if RADIUS is a MUST for billing or can i manage using mysql alone. I have configured accounting module in openser.can i use mysql and a CDR tool for billing?.. -- Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

3 4

[Kamailio-Users] dbtext, domain and caching
by Andreas Granig 17 Nov '09

17 Nov '09

Hi, I was trying to use caching in domain module with the dbtext backend, but on startup I get the error ERROR:domain:reload_domain_table: Database problem Looks like something is wrong with the column type when it's checked in line if ((ROW_N(row) == 1) && (VAL_TYPE(val) == DB_STRING)) { ... } The db file for domain module looks like: id(int) domain(str) last_modified(int) 1:test.local:0 Any hints on that? Probably a problem with dbtext module, since it works with mysql? Best regards, Andreas

2 5

[Kamailio-Users] Test a string concatenated with an AVP
by Antonio Reale 17 Nov '09

17 Nov '09

Hi all, actually I test if the R-URI contains the value *44* with: if (uri=~"^sip:\*44\*[0-9]+@") { ... } Now I want to read the value 44 or *44* from an AVP. Tried using directly the avp or assigning it to a variable, without success. ex.: $var(x) = "sip:" + $avp(s:code) + "[0-9]+@"; How can I solve this problem? Thank you very much. Regards. -- Antonio

2 6

[Serusers] (no subject)
by Chahn Kim 16 Nov '09

16 Nov '09

Hi all, Need some help from experts. I have beeb using $hdr(via[0]) to fetch the VIA header to get top via field, which will show entire top via filed as; <SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK-9aa0fded>. Now I am in need of getting IP:port# portion only to get assigend to a variable. What would be the most effecient way of doing it? I didn't find any other pseudo variable to provide it based on docu. Thanks in advance. John Chahn Kim email: johnk(a)koreanet.us

1 1

[Kamailio-Users] Question about force_send_socket() as it relates to record_route()
by Geoffrey Mina 16 Nov '09

16 Nov '09

Hello, I have a scenario where I need to send traffic to my carrier from two different source IP addresses. The carrier charges me one rate for dialer based traffic and another rate for standard outbound origination. I am planning on using the force_send_socket() method to manually set the source IP address. My question is how does this correlate to my usage of the record_route() function? Do I need to force_send_socket() BEFORE inserting the Record-Route header? Are there any other things I need to be concerned with while using the TM module? Thanks!

2 3

[Kamailio-Users] Freeradius Configuration
by toqeer ali 16 Nov '09

16 Nov '09

I am configuring Freeradius server with openser... By default there is openser dictionary file that the path of that file should be included freeradius master dictionary file... now i am confuse that command $INCLUDE {path of the openser dictioanry file} how i will include that over the network in dictionary file of Freeradius... if i copy the Freeradius server then it gives error... Basically my radius and openser are both on diffrent servers. -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

2 1

[Kamailio-Users] Possible setups question
by Varga Levente 16 Nov '09

16 Nov '09

Hello! I'm new to Kamailio so I want to ask a few questions before to be sure that I'm not trying to do something that cannot be done using Kamailio. The question would be if it is possible to do the following things using Kamailio: I want to setup the following configuration, two or more Asterisk servers with one redundant server in standby using Kamailio to route and partition the number. of users in a fixed manner. What I mean by that for instance if I have 200 users I want that the first 100 user to register and use server number one and the other 100 user to use the server number two. A user cannot migrate from one server to another they have to be always on the designated server from start. Kamailio would act as a unified front-end/gateway for the client (they will use the same entry point not being aware that there a number of Asterisk server beside it). Also there is a need for a N+1 redudancy scheme, that is if one of the servers fail or get very busy the new calls and registration should be redirected to the standby server on the fly (user account data replication would be taken care of on the spot). Can all this be done using Kamailio as a gateway and SIP router? Thank you! levi -- Lege, lege, relege, ora, labora et invenies.

3 2

[Kamailio-Users] please help me
by GEMINORIUM crux 15 Nov '09

15 Nov '09

hi i am transferring tls protocol to udp protocol. client <---tls----> kamilio <----udp---> asterisk how can i add in contact "transport=tls" when i am sending 200 OK message. please help me. any suggestion is appreciated.

2 1

[Kamailio-Users] CDRTool Mediaproxy etc.. AG stuff
by Brian Chamberlain 14 Nov '09

14 Nov '09

Hi List, I am running Debian lenny. Can I just change the repos in sources.list from lenny to testing and unstable to get the correct python dependencies etc for the ag-project tools? Thanks, Brian

1 0

[Serusers] CNAM in SER
by Brian Artigas 13 Nov '09

13 Nov '09

Hello, Our VoIP provider doesn't provide CNAM to us so we have to access an outside provider for this. Can anyone point me in the right direction for running an http request and passing the result back out so the pbx can pick grab the callerid. I know that OpenSER they has the Utils module but SER doesn't have that. Any ideas? Thanks in advance, Brian Artigas Allstate Computers, LLC brian(a)allstatecomputers.com www.allstatecomputers.com P: 561.743.1521 F: 561.575.9234

1 0

[Kamailio-Users] Remote management and security
by Brian 13 Nov '09

13 Nov '09

Hi, Does anyone know how I can do remote management of Kamailio in a secure way? Looking at the documentation for MI_DATAGRAM and MI_XMLRPC, there does not seem to be a way to authenticate the request, or a way to restrict requests to specific IP ranges. I want to be able to send management commands remotely, but I don't to make this wide open. Thanks, Brian.

3 3

Re: [Kamailio-Users] Register Request Forward
by toqeer ali 13 Nov '09

13 Nov '09

My rtpproxy works now basically i am using kamailio 1.2 with LCR, but when i dial one number via LCR it dial this number and it works but when i dial second number it says "*filtered destination*", i have to restart the service and then it dial second number... i think LCR not de-alocating the resources any idea ? 2009/11/12 Denis Putyato <denis7979(a)mail.ru> > 1) I would done not ” modparam("nathelper", "rtpproxy_sock", "udp: > 127.0.0.1:8899")” but “modparam("nathelper", "rtpproxy_sock", > "unix:/var/run/rtpproxy.sock")” > > 2) http://www.kamailio.org/docs/modules/1.5.x/nathelper.html#id2468157 > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Thursday, November 12, 2009 2:10 PM > > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Thanks Denis, > > > > It is indeed the problem of NAT. We have tried with live ip and its work > fine. However, behind the NAT it is not working. I have installed RTPproxy , > but still it is not working. Here are the configuration that we use in our > cfg file. > > > > modparam("nathelper","received_avp", "$avp(i:42)") > modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:8899") > modparam("nathelper", "natping_interval", 30) > modparam("nathelper", "ping_nated_only", 0) > modparam("nathelper", "sipping_bflag", 7) > modparam("nathelper", "sipping_from", "sip:pinger@127.0.0.1<sip%3Apinger(a)127.0.0.1> > ") > > route { > > ## NAT Detection > # > force_rport(); > if (nat_uac_test("19")) { > if (method=="REGISTER") { > fix_nated_register(); > } else { > fix_nated_contact(); > }; > setflag(5); > }; > > > ................. > > > > and rtpproxy is also configured on the same server. > > > > > > > 2009/11/12 Denis Putyato <denis7979(a)mail.ru> > > No two way communication is a problem of NAT in general. If session has > been established then it’s not codec problem. > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Thursday, November 12, 2009 9:25 AM > > > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > i have a problem with codec. when i call to canada from kamailio to PSTN, i > could not listen his/her voice as they listen( no two way communication) > > but when call to other countries it works... Basically call to canada > support g729 can i define any codec information in kamailio to G729. > > > > Basically the problem is only with local number when you dial with > extension 1 and when you dial with 08 it works... 08 is international > gateway of canada... > > > > X-lite i am using as a softphone. > > > > please help > > > > 2009/11/11 Denis Putyato <denis7979(a)mail.ru> > > “can i use regular expression in $fu=XXX” > > > > You can use regexp. $fu=~”expression” > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 3:08 PM > > > *To:* Denis Putyato > *Cc:* users@lists.kamailio..org <users(a)lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Hmmm > > > > it works now i just remove Record Route function and it works,,, now i can > dial from 602 to 601... thanks :). > > > > i will start work on how i will register multiple extension via kamailio to > asterisk ... can i use regular expression in $fu=XXX instead of actual > extension extension... > > 2009/11/11 Denis Putyato <denis7979(a)mail.ru> > > Registration is a process for telling SoftSwitch (in your variant - > Asterisk) where (on which username, IP address and port) the SoftSwitch must > send INVITE when received incoming call for 601. Try to understand where > your asterisk try to send INVITE when you try to call from 602 to 601. > > ngrep utility will help you (http://ngrep.sourceforge.net/download.html) > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 2:18 PM > > > *To:* Denis Putyato > *Cc:* users@lists.kamailio..org <users(a)lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Here is the update: > > > > what i have done so for is that I have "Register extension 601" to asterisk > via kamailio and it is registered!. > > > > When i call to 602 (configured on 192.168.0.1 -- asterisk box) it calls and > make connection and voice call also works (RTP Traffic is ok). > > > > However, when i call to 601 from 602 (which is directly connected to > asterisk without kamailio ), 601 can't receive call via kamailio. This is > the main problem. Please guide me that how i will define this in kamailio > (ip = 192.168.0.2) that xlite with extension 601 can receive call through > 602 which is another extension on Asterisk. > > > > In a nutshell: 601 (registered through Kamailio to Asterisk) ---can call > ---> 602 (registered via Asterisk) > > However: 602----cannot call ------> 601 > > > > please help. > > > > > > 2009/11/11 Denis Putyato <denis7979(a)mail.ru> > > Try to use dlg_bridge(); function in dialog module > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 12:13 PM > > > *To:* Denis Putyato > *Cc:* users@lists.kamailio..org <users(a)lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > > "that client send INVITE to asterisk via kamailio but no replay is in on > this INVITE" > > > > basically Asterisk receive the request and send back to kamailio but > kamailio can't send back to xlite.. > > > > 2009/11/11 Denis Putyato <denis7979(a)mail.ru> > > «but call does not route back to me» > > > > You mean, that incoming call from asterisk to client via kamailio doesn’t > work or that client send INVITE to asterisk via kamailio but no replay is in > on this INVITE? > > > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 10:16 AM > > > *To:* Denis Putyato > *Cc:* users@l > > ists.kamailio.org > > > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Thanks Again for the prompt reply... Basicaly i want to route Register > Requests for multiple asterisk via kamailio ... by using pattern matching > of extensions... > > > > is i am on the right path ?? or i can achieve this from dispatcher or load > balancer module .... Basically i want to register specific extensions > to specific asterisk boxes.... > > > > > > from your previous reply i can send call to asterisk and register to as > well but call does not route back to me... any clue ... > > 2009/11/11 Denis Putyato <denis7979(a)mail.r > > “The secret is that the other sip server was asterisk and i want route > call to asterisk via kamailio :).” > > I don’t understand why do you want to do any session with asterisk via > kamailio… not directly to asterisk… but any way the decision is same – > send() or rewritehost() > > > > route { > > if ($fU==”…”&&is_method(“INVITE”)) > > record_route(); > > rewritehost(“192.168.0..1”); > > t_relay(); > > } > > > > Do this before authentication checking is made in kamailio. > > *From:* toqeer ali [mailto:toqeer83@gmail.com] > *Sent:* Wednesday, November 11, 2009 6:49 AM > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org > *Subject:* Re: [Kamailio-Users] Register Request Forward > > > > Thanks it Registered but i can't call on this .. "authentication Required > from proxy". what i should do now. > > > > The secret is that the other sip server was asterisk and i want route call > to asterisk via kamailio :). > > 2009/11/10 Denis Putyato <denis7979(a)mail.ru> > > Hi > > If not secret, why do you want to use such scheme? > > > > Your decision > > send(); - core function. > > > > { > > If (is_method(“REGISTER”)&&$fU==”….”) send(“udp:192.168.0.1:5060”); > > } > > > > or > > > > { > > If (is_method(“REGISTER”)&&$fU==”….”) { > > rewritehost(“192.168..0.1”); > > t_relay(); > > } > > } > > > > *From:* users-bounces(a)lists.kamailio.org [mailto: > users-bounces(a)lists.kamailio.org] *On Behalf Of *toqeer ali > *Sent:* Tuesday, November 10, 2009 1:56 PM > *To:* users(a)lists.kamailio.org > *Subject:* [Kamailio-Users] Register Request Forward > > > > Hi all, > > > > My question is how to forward Register Request to another sip server. > > > > For example one user (X-lite) want to register with 192.168.0.1(sip > server1) via 192.168.0.2(sip server 2). > > it means that in the domain part of xlite is 192.168...0.2 and > it basically register with 192.168.0.1. > > 192.168.0.2 will forward Register request to 192.168...0.1. > > it will be very helpful if any one can answer. > > > > Thanks > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > > > > > -- > Toqeer Ali Syed > > Red Hat Certified Engineer > mob: +92 321 9059916 > -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

2 1

[Kamailio-Users] SKYPE or voipstunt
by Masoom Alam 13 Nov '09

13 Nov '09

whether skype or voipstunt are using SIP protocols. I have experienced that in many countries, where voice ports are blocked by the Telecom providers but still voipstunt and skype works for calling to PSTN. Please give me some links, for further reading on this.

2 1

[Kamailio-Users] Register Request Forward
by toqeer ali 12 Nov '09

12 Nov '09

Hi all, My question is how to forward Register Request to another sip server. For example one user (X-lite) want to register with 192.168.0.1(sip server1) via 192.168.0.2(sip server 2). it means that in the domain part of xlite is 192.168.0.2 and it basically register with 192.168.0.1. 192.168.0.2 will forward Register request to 192.168.0.1. it will be very helpful if any one can answer. Thanks -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

2 14

[Kamailio-Users] record-route
by toqeer ali 12 Nov '09

12 Nov '09

Let me describe my scenario: SipUser1 -------register request------------> kamailio (SIP Proxy) ------- forwarded register request-----------> Asterisk 1. In the above scenario, when I register to Asterisk through Kamailio, it works fine when I give the "record-route()" function in the cfg file. Further, using this, I can also call to another extension 602 which is also registered to Asterisk but 602 cannot call to 601 which is registered via kamailio. 2. However, If I remove the "record-route()" function without unregistering 601 and restarting the server, 602 can call to 601 . route { if ($fU=="601"&&is_method("INVITE")) record_route(); rewritehost("10.0.10.111"); //asterisk host route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } should i add anything in script further... -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

2 1

[Kamailio-Users] Listening on more than one port
by Sven Schulz 12 Nov '09

12 Nov '09

Is it possible to have kamailio (1.4.4) to listen on more than one port? Is it as simple as adding another ³port=xxxx² in the config? Sven Schulz Penn State University Telecommunications and Network Services 814.865.6116 sip:sven@psu.edu

2 1

[Kamailio-Users] sending calls to voip provider
by Andres Moya 11 Nov '09

11 Nov '09

Hi, I am with openser about my first 40 hours. So far i managed to install 3.0. Was able to make calls between registered phones. Enabled accounting, etc. Looks i have control ;) Now i have a very simple question. I want to send calls for land lines via voip provider, sipdiscount.com in my case. I want to enable so registered client agent can send invite to that server, but it should re authenticate somehow. I was looking for proxy_authorize www_authorize but it looks like it authenticate my phones only. I was trying rewrite rewritehostport, but of course was getting authentications errors. Is there any manual, previous discussion on this topic? Thank you.

1 0

[Kamailio-Users] change contact address
by GEMINORIUM crux 11 Nov '09

11 Nov '09

hi 1. i want to change contact address .i want to add transport protocol how can i do this ? 2. i am getting Error on $var(query) = "INSERT INTO sip_user(sipuser) VALUES(' "+$fU+"' )"; if(!avp_db_query("$var(query)","$avp(s:test)")){ xlog("ERROR IN Query = $var(query)"); sl_send_reply("400","Bad Request"); exit; } but if i run that query in mysql it give me expected output. why i am getting error ? pls help me ? any suggestion is appreciated. thanks

1 0

[Kamailio-Users] Problem in configuring LCR module
by Masoom Alam 11 Nov '09

11 Nov '09

Hi there, I am a first time user of Kamilio 1.5. We are trying to configure the LCR, we get an error in the syslog. The error in the syslog is: Nov 6 15:12:09 kam /sbin/kamailio[5396]: ERROR:lcr:mod_init: AVP fr_inv_timer_avp has not been defined Nov 6 15:12:09 kam /sbin/kamailio[5396]: ERROR:core:init_mod: failed to initialize module lcr Nov 6 15:12:09 kam /sbin/kamailio[5396]: ERROR:core:main: error while initializing modules We have already configured FR_inv_timer_avp parameter like this in the Kamilio.cfg: modparam("tm", "fr_inv_timer_avp", "$avp(i:70") The "tm" module is already loaded in the Kamilio.cfg. Please guide, where we are making problems?

7 13

[Kamailio-Users] Translate "486 Busy Here" to "600 Busy Everwhere"
by geoffreymina＠gmail.com 11 Nov '09

11 Nov '09

Hello, I have a need to translate a 486 from my asterisk server to a 600 when sending upstream to my carrier. The problem I am seeing is that when I return 486, the carrier re-presents the call to my next server in the SRV rotation... I really want the 486 to be a final disposition, so I am assuming sending a 600 would solve my issue. I am not sure how to go about forwarding the message, but modifying the SIP code... is that even allowed? This would be something like what I'm trying to do: failure_route[1]{ if(t_check_status("

3 3

Re: [Kamailio-Users] Translate "486 Busy Here" to "600 Busy Everwhere"
by Geoffrey Mina 10 Nov '09

10 Nov '09

Apologies... sent directly to Alex and not the list. On Tue, Nov 10, 2009 at 12:02 PM, Geoffrey Mina <geoffreymina(a)gmail.com> wrote: > Alex, > Thanks for the info. We are on the same page about what a proxy > should, and should not do... however, I am strugging with my ITSP > (Level3) a bit on this. Unfortunately, Asterisk is my media end-point > here and there is no mechanism in Asterisk to send back a 600. Level3 > is exhibiting very odd behaivor, and I have a ticket open with them, > but who knows if it will ever go anywhere. > > CallID: 1627795841_10332(a)4.55.4.227 > Level3 --> INVITE --> Kamailio --> Asterisk > Asterisk --> 183 Progress --> Kamailio --> Level3 > Asterisk --> 486 Busy --> Kamailio --> Level3 > Level3 --> ACK --> Kamailio --> Asterisk > > [at this point Level3 IMMEDIATELY sends me another presentation of the > call with a different CallID and the process repeats] > > CallID: 604627041_120490736(a)4.55.20.99 > Level3 --> INVITE --> Kamailio --> Asterisk > Asterisk --> 183 Progress --> Kamailio --> Level3 > Asterisk --> 486 Busy --> Kamailio --> Level3 > Level3 --> ACK --> Kamailio --> Asterisk > > > I am pretty sure I am in the right here, and Level3 shouldn't be > failing over in a 486 scenario... unfortunately my customer doesn't > care why our systems aren't talking properly :) > > Thanks! > > > > On Tue, Nov 10, 2009 at 10:38 AM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> geoffreymina(a)gmail.com wrote: >>> >>> Hello, >>> I have a need to translate a 486 from my asterisk server to a 600 when >>> sending upstream to my carrier. The problem I am seeing is that when I >>> return 486, the carrier re-presents the call to my next server in the SRV >>> rotation... I really want the 486 to be a final disposition, so I am >>> assuming sending a 600 would solve my issue. >>> >>> I am not sure how to go about forwarding the message, but modifying the >>> SIP code... is that even allowed? >>> >>> This would be something like what I'm trying to do: >>> >>> failure_route[1]{ >>> if(t_check_status(" >> >> Sure, you can do that. It is not a good idea to do so in principle because >> it violates how proxies should behave; proxies are very thin interoperation >> layers that are designed to statefully pass requests and replies back and >> forth more or less strictly as intended by the endpoints involved, and a >> B2BUA would more ideally fit your goal. Nevertheless, it is possible. >> >> failure_route[1] { >> >> if(t_was_cancelled()) >> exit; >> >> if(t_check_status("486")) { >> t_reply("603", "Declined"); >> exit; >> } >> } >> >> >> -- >> Alex Balashov - Principal >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> >

2 2

[Kamailio-Users] Double acc records in DB
by Alex Hermann 10 Nov '09

10 Nov '09

Hello, I noticed that sometimes I get 2 identical acc records in the database, although I get only 1 in the syslog (I use both db_flag and log_flag). Of course, the second record in the DB shouldn't be there. There are no retransmissions of neither request nor response and only a single branch exists. How is this possible, and more importantly, how can I prevent the double records? This is on Kamailio 1.4+svn. -- Greetings, Alex Hermann

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[Kamailio-Users] Question about translating 486 to 600
by Geoffrey Mina 10 Nov '09

10 Nov '09

Hello, I have a need to translate a 486 from my asterisk server into a 600 to my ITSP. I haven't figured out a way to force asterisk to send the 600, so I guess I need to rely on Kamailio to do that for me. The problem is that when the INVITE is sent to me, if I return a 486, my ITSP immediately sends a new INVITE to the next gateway in my SRV records. I want the 486 to be a final disposition in this scenario, so I am guessing I need to send 600 for the carrier to not try another gateway. How would I go about doing this? failure_route[1]{ if(t_check_status("486")){ t_relay("600","Busy Here"); } } I know the above example isn't valid... but that is basically what I need to accomplish. Thanks!

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[Kamailio-Users] avp
by Anders 10 Nov '09

10 Nov '09

Hi all, Here I am - back with more "Kamailio 101"-questions! ;-) I need to strip the first character from the uri and write this to the 'acc' table. What I'm doing so far in kamailio.cfg is: modparam("acc", "db_extra", "............., $avp(s:user_ip_tag)"); So I've added the avp-thing in that line. And independently I've created a new column in the 'acc' table called "user_ip_tag". If the above is correct, my next step is to write something to this variable. In the routing, I have this: if (uri=~"^sip:[1-9]00*") { do_something_here_to_take_first_character_of_the_uri_and_put_to_avp...but_how? strip(1); } What I want to do, is to take the 1-9 digit, if it starts with that and has to zeroes after, and put that to the avp - and the strip it off before I move on in the routing. Yes, I have browsed the documentation, but I'm not sharp enough to understand it, it seems. (or alternatively, the documentation is not made for beginners). Thanks!

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[Kamailio-Users] accounting
by Rajesh 09 Nov '09

09 Nov '09

Dear all, I would like to know if i can get the duration between two calls in Accounting module..I am in an attempt to do "Billing" module for OpenSER and hence would like to know the same. -- Regards, Rajesh

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SR or Kamailio.
by Vicente 09 Nov '09

09 Nov '09

Hi, This is my first message to the list. So, sorry if I say something stupid. I've installed sip-router from GIT repository. It is now compiled and installed. I can see that bins ser, sercmd and ser_mysql.sh has been copied to /usr/local/sbin. ser runs properly but sercmd tries to use /tmp/ser_ctl (ERROR: connect_unix_sock: connect(/tmp/ser_ctl): No such file or directory [2]) while /temp/sip-router_ctl is the socket generated by ser. ser_mysql.sh also fails with error: ./ser_mysql.sh: 326: Bad substitution The INSTALL file on the repository instructs to use serctl utility to watch server health but no binary has been generated with that name. Is it replaced by sercmd? What is the status of sip-router.org? Is it ready for production? Or is it a better option to use Kamailio at the moment? Best regards.

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[Kamailio-Users] How to recognize the caller ID
by Masoom Alam 09 Nov '09

09 Nov '09

Hi Every one, Is the following scenario possible and if yes, how? I want to identify the caller ID inside the Kamailio so that a specific caller can be routed to an Asterisk box. Actually, the caller -- a SIP client or an IP phone client is not registered at Kamailio, but Kamailio has to act as a Proxy to a group of Asterisk servers. The sketch in my mind is to use some if-then-else statements within the [route] where user can call a particular number say NXXXXXXX201 and then the Kamailio can strip off the first number and re-route it to an Asterisk server. Regards, MM Alam SERG Pakistan http://serg.imsciences.edu.pk http://www.imsciences.edu.pk/profiles/masoom_alam.html

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[Serusers] 500 internal error
by Sabatella, Michael 08 Nov '09

08 Nov '09

Hi, I am getting a 500 internal server error. I am not sure why as my config did not change. I do not want to restart the ser proxy. Could I get a list of causes for a 500 server error? Thanks, Michael Sabatella Network Services Engineer Office: 1-201-253-2280 Cell: 1-908-380-5321 ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

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[Kamailio-Users] A architectural question
by Masoom Alam 08 Nov '09

08 Nov '09

Hi there, Is it possible that Kamailio is used in front of a group of Asterisk servers. That is, Kamailio as a REGISTRAR Translation can proxy the SIP registration and identify which PBX to send on LAN by the ID before the EXT underlined from IP PHONE. Please advise with some links through which it is possible to handle such an architecture. Regards, MM Alam SERG Pakistan http://serg.imsciences.edu.pk http://www.imsciences.edu.pk/profiles/masoom_alam.html

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[Kamailio-Users] OCS?
by Jay R. Worthington 06 Nov '09

06 Nov '09

Hi, has someone managed to "federate" kamailio with an Office Communications Server? I'm mainly looking for IM&Presence (to gate into ejabberd), voice would be nice to have... Regards, Jay

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