- sr-users - kamailio.org

[Kamailio-Users] Access List
by michel freiha 24 Sep '08

24 Sep '08

Hi All, Hi All,I need please to configure kamailio in order to force the user to register to my it from a specific IP...It's like creation of an access list...If the customer tries to register from another IP frm the one specified in kamailio he'll be denied Regards

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[Kamailio-Users] digest or not digest ...
by BERGANZ François 24 Sep '08

24 Sep '08

Hello all, I need to choose if my users do digest auth or not! If I have : modparam("auth_db", "calculate_ha1", yes)àauth or modparam("auth_db", "calculate_ha1", 0)àauth digest! How can I choose if this user is digest after having look at a param in the database For example, user 1 have a column in the database which say ok he can do digest auth How can I ask to the client an auth digest ? Thank you Cordialement BERGANZ François http://www.acropolistelecom.net <http://www.acropolistelecom.net/>

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[Kamailio-Users] openser.org and project documentation
by Jeremy McNamara 23 Sep '08

23 Sep '08

Whoever has taken over openser.org should do the community as a favor and at the very least HTTP 301 redirect the documentation links over to kamailio.org. There are soo many sites that index URLs like: http://openser.org/docs/modules/ ...however openser.org gives back a 404 not found.

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[Kamailio-Users] About Changing URI's
by Ali Jawad 23 Sep '08

23 Sep '08

Hi I have the following setup Cell Phone "Any Possible Number" ----Calls a DID number-----> DID Provider --> Forwards Call to my Openser --> OpenSer Server--> Forwards Call to Asterisk with A2B --> Forwards it to the cell phone of the user In short, a person can call a DID and it gets forwarded to Openser, if the user is online using his sip phone, the sip phone rings and all is OK. However, if the user is not online I do forward that call to Asterisk and I want to forward it to the cell phone of the user. The problem is that the FROM field and the SOURCE URI keep changing,they are in the from phonenumberOfCaller(a)DID_Service_Provider.com and I do want to charge the user for the call. To do that I need to have the FROM field and the source URI to be the USER ID so that I can charge him for the call. So, I might have to change the source URI and FROM field of the packet and send it to asterisk..it would send the packet back to Openser and I have to change those back to what they were and then send it to the DID provider. Might sound confusing, and it might sound stupid. Please help me set this straight, can it be done ? Maybe not ? Maybe Yes... Any suggestions are welcome. With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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[Kamailio-Users] un-subscribing a pending subscription is also pending
by jh＠tutpro.com 23 Sep '08

23 Sep '08

a user complained that un-subscribing a pending subscription is also pending, i.e., kamailio responds with 202 instead of 200. i verified the behavior. now the question is, what do the relevant RFCs say about it? it appears logical to me that if a user gives up waiting to get his/her subscription confirmed, he/she should be able to forget the whole thing by sending an un-subscribe. any comments on this? -- juha

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[Kamailio-Users] CDRTool 6.6.3/6.6.4 with freeradius-cdrtool-1.1
by Diego Zuaneti Arruda 23 Sep '08

23 Sep '08

Hi all, it is possible to use CDRTool 6.6.3 or 6.6.4 with freeradius-cdrtool-1.1 ? The problem is in the code of freeradius-cdrtool.py, where in the code have "DebitBalance From=%s To=%s Duration=%s\n" , "MaxSessionTime From=%s To=%s Duration=%s Lock=1\n" and others, but the newer CDRTool need have a gateway parameter and Call Id parameter. Anybody have a sucess with this versions ? Thanks in advanced, Diego.

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[Kamailio-Users] how to change source number
by BERGANZ François 23 Sep '08

23 Sep '08

Hello, I need to change the source number, to have an international format (e164) but when I use : $avp(s:245)=$fu; avp_subst("$avp(s:245)/$avp(s:246)/d", "/sip:/sip:33/"); uac_replace_from("", "$avp(s:246)"); in fact, my phone have in the screen sip:33..., before, it have my name I think that I have to use another function to change the source number, but which function? Thank you Cordialement BERGANZ François http://www.acropolistelecom.net <http://www.acropolistelecom.net/>

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Re: [Kamailio-Users] nested redirect
by Daniel-Constantin Mierla 23 Sep '08

23 Sep '08

Hello, what you sent is the debug from kamailio/openser, which is useful, but I would need the sip messages, you can use ngrep for that, do ngrep -d any -qt -W byline port 5060 on your openser/kamailio server and send the output. Thanks, Daniel On 09/17/08 02:22, Kent Pirlo wrote: > ... is my core switch, > ... is my openser/Kamailio redirect server > ... is a redirect server from a carrier of ours that returns a billing TN in contact header.. > > > Here is the debug of the incoming ACK from the core switch after being sent the 300... > > > Sep 16 15:49:50 [6505] DBG:core:parse_msg: SIP Request: > Sep 16 15:49:50 [6505] DBG:core:parse_msg: method: <ACK> > Sep 16 15:49:50 [6505] DBG:core:parse_msg: uri: <sip:...@...> > Sep 16 15:49:50 [6505] DBG:core:parse_msg: version: <SIP/2.0> > Sep 16 15:49:50 [6505] DBG:core:parse_headers: flags=2 > Sep 16 15:49:50 [6505] DBG:core:parse_to_param: tag=391854ebdd4d0c63cf50836cbb12191e-6708 > Sep 16 15:49:50 [6505] DBG:core:parse_to: end of header reached, state=29 > Sep 16 15:49:50 [6505] DBG:core:parse_to: display={}, > Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: <To> [75]; > Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: to body > Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: cseq <CSeq>: <1> <ACK> > Sep 16 15:49:50 [6505] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKdce3ac34730f843a5a9b9c7002690f22>; state=16 > Sep 16 15:49:50 [6505] DBG:core:parse_via: end of header reached, state=5 > Sep 16 15:49:50 [6505] DBG:core:parse_headers: via found, flags=2 > Sep 16 15:49:50 [6505] DBG:core:parse_headers: this is the first via > Sep 16 15:49:50 [6505] DBG:core:receive_msg: After parse_msg... > Sep 16 15:49:50 [6505] DBG:core:receive_msg: preparing to run routing scripts... > Sep 16 15:49:50 [6505] DBG:sl:sl_filter_ACK: to late to be a local ACK! > Sep 16 15:49:50 [6505] DBG:core:destroy_avp_list: destroying list (nil) > Sep 16 15:49:50 [6505] DBG:core:receive_msg: cleaning up > > > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: Tuesday, September 16, 2008 5:10 AM > To: kentp(a)globalpops.com > Cc: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] nested redirect > > Hello, > > can you post here the SIP messages? Will help to spot if there is > anything wrong with the ACK or something else in the signaling. > > Cheers, > Daniel > > > On 09/13/08 04:11, kentp(a)globalpops.com wrote: > >> i have a simple redirect server working: >> >> Call comes from switch (A) to kamailio redirect server (B).. >> (B) runs perl script to determine route >> (B) sends Stateless 300 redirect reply to (A). >> (A) sends ACK to (B). >> >> Now, I need to add a nested redirect... >> >> call comes from sitch (A) to kamailio redirect server (B).. >> (B) then needs to send the call to third party proxy (C).. >> (C) will return 302 redirect.. >> (B) sends ACK to (C) >> (B) needs to read the contact header, then run perl script to determine route (using the contact value >> returned from (c)) >> (B) then sends 300 redirect back to (A). >> (A) sends ACK back to (B) >> >> >> so far i have been trying something like this, but the final (A) ACK gets ignored and (B) keeps sending >> multiple 300's to (A): >> >> route { >> route(1); >> } >> >> route[1] { >> t_on_reply("1"); >> t_on_failure("2"); >> if (!t_relay("proxy.server.c:5060","0x02")) { >> sl_reply_error(); >> } >> onreply_route[1] { >> if(t_check_status("302")){ >> $var(lrn_from_bv) = $ct; >> } >> } >> failure_route[2] { >> >> if (t_check_status("3[0-9][0-9]")) { >> route(2); >> exit; >> } >> route[2] { >> if(perl_exec("callcheck")){ >> avp_pushto("$ru", "$avp(s:routes)/g"); >> t_reply("300","Go Here"); >> exit; >> } >> } >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >> > > -- > Daniel-Constantin Mierla > http://www.asipto.com > -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Question about redirect if user is not online
by Ali Jawad 23 Sep '08

23 Sep '08

Hi All I do forward calls to a DID number to my openser box, if my softphone is online it rings directly because I am using the DID number as an alias. However I want the call to be forwarded to my asterisk box if the user is not online. What statement should I use ? THanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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[Kamailio-Users] preparing release 1.4.1
by Daniel-Constantin Mierla 22 Sep '08

22 Sep '08

Hello, I propose to release kamailio 1.4.1 next Thursday, Sep 25, 2008. Please send to devel(a)lists.kamailio.org any critical issues you are aware of, to try to fix them. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] Handling server with broken record-route handling
by Marcus Better 22 Sep '08

22 Sep '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have problems with a remote server, outside of my control, that seems to have very broken request URI handling. It is possibly suffering from an Asterisk bug [1] or something similar. On my side I have a UAC (IP address 172.16.1.102) and a SER 2.0 proxy. The UAC sends an INVITE, and the remote UAS ends the dialog with a BYE request. However this BYE returns to my proxy with missing route headers, so SER cannot route it. I need to work around the problem with SER on my side and wonder what the best solution is. Here is the relevant part of the conversation as seen on my proxy 80.81.82.3 (slightly massaged, irrelevant parts cut): INVITE sip:00463112341234@proxy.remote.com SIP/2.0. Record-Route: <sip:80.81.82.3;avp=Z3oDBwBhY2NvdW50AwB5ZXM;lr>. Call-ID: f730a3a6a5c450508274739a8133ae7f(a)172.16.1.102. CSeq: 1 INVITE. To: <sip:00463112341234@proxy.remote.com>. From: <sip:17473605948@proxy.remote.com>;tag=2045565953650740630. Via: SIP/2.0/UDP 80.81.82.3;branch=z9hG4bK761b.b1548ac4.0. Via: SIP/2.0/UDP 172.16.1.102:5060;rport=5060;appname=myapp;branch=z9hG4bK74592b710287e6cfa94b4cf9c10e15bf. Contact: "17473605948" <sip:172.16.1.102:5060;transport=udp>. SIP/2.0 200 OK. Via: SIP/2.0/UDP 80.81.82.3;branch=z9hG4bK761b.b1548ac4.0. Via: SIP/2.0/UDP 172.16.1.102:5060;rport=5060;appname=myapp;branch=z9hG4bK74592b710287e6cfa94b4cf9c10e15bf. Record-Route: <sip:123.12.13.14;lr;ftag=2045565953650740630>. Record-Route: <sip:80.81.82.3;avp=Z3oDBwBhY2NvdW50AwB5ZXM;lr>. From: <sip:17473605948@proxy.remote.com>;tag=2045565953650740630. To: <sip:00463112341234@proxy.remote.com>;tag=as438ee174. Call-ID: f730a3a6a5c450508274739a8133ae7f(a)172.16.1.102. CSeq: 1 INVITE. Contact: <sip:011463112341234@130.94.88.93:5060>. ACK sip:01146739835569@130.94.88.90:5060 SIP/2.0. Record-Route: <sip:80.81.82.3;avp=Z3oDBwBhY2NvdW50AwB5ZXM;lr>. Call-ID: f730a3a6a5c450508274739a8133ae7f(a)172.16.1.102. CSeq: 1 ACK. Via: SIP/2.0/UDP 80.81.82.3;branch=0. Via: SIP/2.0/UDP 172.16.1.102:5060;rport=5060;appname=myapp;branch=z9hG4bK96a96b64dfd5aba5c896c37e0de18850. From: <sip:17473605944@proxy.remote.com>;tag=7737826714334905221. To: <sip:0046739835569@proxy.remote.com>;tag=as543c9f3f. Route: <sip:123.12.13.14;lr;ftag=7737826714334905221. Content-Type: application/sdp. BYE sip:80.81.82.3:5060 SIP/2.0. Record-Route: <sip:123.12.13.14;lr;ftag=as438ee174>. Via: SIP/2.0/UDP 123.12.13.14;branch=z9hG4bK100d.d617f387.0. Via: SIP/2.0/UDP 130.94.88.93:5060;branch=z9hG4bK0256ad7c;rport=5060. Route: <sip:80.81.82.3;avp=Z3oDBwBhY2NvdW50AwB5ZXM;lr>. From: <sip:00463112341234@proxy.remote.com>;tag=as438ee174. To: <sip:17473605948@proxy.remote.com>;tag=2045565953650740630. Call-ID: f730a3a6a5c450508274739a8133ae7f(a)172.16.1.102. CSeq: 102 BYE. I think this BYE is incorrect. It should have the originator's Contact (sip:172.16.1.102:5060;transport=udp) in the request URI and the Record-Routes of the original INVITE in the Route header. Or, if it is a strict router, it could have the current request URI but add the Contact as the last Route header. Now it has destroyed the originator's route information. What is the best way to get SER to remember/reconstruct this last hop to the UAC? Cheers, Marcus [1] http://bugs.digium.com/view.php?id=3609 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkjXkbQACgkQXjXn6TzcAQnhrQCeP9T0pR+AlPJjTr8OCS5nNOAA GBIAnR5pw68rqe1PR00jDLvcYh+eQXPP =sdca -----END PGP SIGNATURE-----

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[Kamailio-Users] Kamailio+Asterisk Need Help
by VIP Carrier 22 Sep '08

22 Sep '08

Hi, I'm trying to setup an HA cluster based of 2 Proxy 4 Asterisk 2 DB's Can some one please help me with this.

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[Kamailio-Users] Crate prepaid account on CDRTool
by Adrian Georgescu 20 Sep '08

20 Sep '08

Click on the Rating link then select Prepaid table. There you can add new accounts Adrian

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[Kamailio-Users] New MediaProxy release 2.1.0
by Adrian Georgescu 20 Sep '08

20 Sep '08

MediaProxy2 uses the Linux kernel for routing packets.So it uses all the cores and CPUs recognized by the kernel. Adrian

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[Kamailio-Users] OT: openser/Asterisk management GUI
by Klaus Darilion 19 Sep '08

19 Sep '08

Hi! FYI: I just found this one: http://carrierclass.net/ looks like it is a central management webinterface for a openser-asterisk setup. regards klaus

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[Kamailio-Users] Problem Dynamically Setting Forward
by Jon Farmer 18 Sep '08

18 Sep '08

Hi I am trying to call a script via exec_avp and then use the output of the script to use as an argument to forward(). Such as route[2] { # forward to AGCF exec_avp("/usr/utils/select_incoming_agcf.php"); record_route(); xlog("forward to: $avp(i:1)\n"); #forward("$avp(s:1)"); forward("$avp(s:1); exit; } However syslog complains as follows Sep 18 11:42:12 vla openser[19470]: ERROR:core:parse_phostport: bad protocol in $avp(s:1) Sep 18 11:42:12 vla openser[19470]: ERROR:core:fix_actions: ERROR:fix_actions: FORWARD/SEND bad argument Sep 18 11:42:12 vla openser[19470]: ERROR:core:main: failed to fix configuration with err code -6 The output of the script is 10.0.0.1:5060 Any ideas? Regards Jon

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[Kamailio-Users] Using asterisk to pstn integration
by Tiago Rés 18 Sep '08

18 Sep '08

Hello all, I'm running openser 1.2.2, and i want to make/receive pstn calls. I have installed asterisk 1.4. in the same machine. Do I need to change the port in asterisk? and how to do it? What are the basic lines in the configuration files of asterisk to accomplish that? I don't think I have problem with openser.cfg. Sorry about the questions but I don't seem to find the exact documentation I need. Thanks in advance! Tiago Res

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[Kamailio-Users] Crate prepaid account on CDRTool
by Diego Zuaneti Arruda 18 Sep '08

18 Sep '08

Hello, Somebody have a example to create a prepaid account on CDRTool database ? Thanks. Diego.

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[Kamailio-Users] double resquests
by BERGANZ François 18 Sep '08

18 Sep '08

Hello, I need help because, when myopenser look at a frame, it forward it in double always. You can look at it on the join picture. Thank you for advices. Cordialement BERGANZ François http://www.acropolistelecom.net <http://www.acropolistelecom.net/>

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[Kamailio-Users] Help Me on OpenXCAP...
by mahesh.peddi＠info-spectrum.com 18 Sep '08

18 Sep '08

Hi ALL, I installed openxcap 1.0 version and its dependences. When i run "openxcap --no-fork" and openser 1.3.2v. I am getting these errors. ************************************************************************************************************* [root@info infopresence-1.2.0-tls]# openxcap --no-fork Starting Open XCAP 1.0.0 Traceback (most recent call last): File "/usr/local/bin/openxcap", line 53, in <module> from xcap.server import XCAPServer File "/usr/local/lib/python2.5/site-packages/xcap/server.py", line 21, in <module> from xcap import authentication File "/usr/local/lib/python2.5/site-packages/xcap/authentication.py", line 21, in <module> from xcap.appusage import getApplicationForURI, namespaces File "/usr/local/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 42, in <module> class ServerConfig(ConfigSection): File "/usr/local/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 45, in ServerConfig backend = Backend('Database') File "/usr/local/lib/python2.5/site-packages/xcap/appusage/__init__.py", line 40, in __new__ raise ValueError("Couldn't find the '%s' backend module: %s" % (value.lower(), str(e))) ValueError: Couldn't find the 'database' backend module: No module named _mysql_exceptions [root@info infopresence-1.2.0-tls]# ******************************************************************************************************************** Here with i am enclosing the config.ini file. Please guide me on this. Thanks & Regards, Mahesh Peddi

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[Kamailio-Users] Rtpproxy limit
by alasupcom 18 Sep '08

18 Sep '08

hi all :working: When i use force_rtp_proxy() in my openser.cfg i have 20 calls/s, if i try to increase the number i have this error: force_rtp_proxy2:incorrect port 0 in reply from rtp proxy and i can't use rtpproxy in this calls. But in this linkc http://www.b2bua.org/wiki/RTPproxy/FAQ i have the capacity rtpproxy 1500-2000. Please, if one have the solution send me a reply.:confused: any suggest is appreciate -- View this message in context: http://www.nabble.com/Rtpproxy-limit-tp19532035p19532035.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] RTPproxy
by Ricky Gutierrez 18 Sep '08

18 Sep '08

hi list rtpproxy version 1.1 runs well with kamailio 1.4 or openser 1.3.2 ?, some person in the list that this making tests with her Sep 17 21:55:26 xserver /sbin/kamailio[3522]: WARNING:nathelper:rtpp_test: support for RTP proxy <udp:127.0.0.1:7890> has been disabled temporarily Sep 17 21:55:26 xserver /sbin/kamailio[3531]: WARNING:nathelper:rtpp_test: support for RTP proxy <udp:127.0.0.1:7890> has been disabled temporarily Sep 17 21:55:26 xserver /sbin/kamailio[3535]: WARNING:nathelper:rtpp_test: support for RTP proxy <udp:127.0.0.1:7890> has been disabled temporarily Sep 17 21:55:26 xserver /sbin/kamailio[3518]: WARNING:nathelper:rtpp_test: can't get version of the RTP proxy Sep 17 21:55:26 xserver /sbin/kamailio[3518]: WARNING:nathelper:rtpp_test: support for RTP proxy <udp:127.0.0.1:7890> has been disabled temporarily Sep 17 21:55:25 xserver /sbin/kamailio[3537]: ERROR:nathelper:send_rtpp_command: proxy <udp:127.0.0.1:7890> does not respond, disable it Sep 17 21:55:26 xserver /sbin/kamailio[3537]: WARNING:nathelper:rtpp_test: can't get version of the RTP proxy Sep 17 21:55:26 xserver /sbin/kamailio[3537]: WARNING:nathelper:rtpp_test: support for RTP proxy <udp:127.0.0.1:7890> has been disabled temporarily I have some problems with the one regards rickygm

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[Kamailio-Users] New MediaProxy release 2.1.0
by Adrian Georgescu 18 Sep '08

18 Sep '08

Hello, There is a new release of MediaProxy software available. For people running Debian testing or unstable on an i386 architecture there is now an official public repository provided by AG Projects. To use it, add these lines in /etc/apt/sources.list # AG Projects software deb http://ag-projects.com/debian unstable main deb-src http://ag-projects.com/debian unstable main Install the AG Projects debian software signing key: wget http://download.ag-projects.com/agp-debian-gpg.key apt-key add agp-debian-gpg.key After that, run: apt-get update apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web- sessions to install all the packages, or you can install only the packages you actually need on that specific system. The change log follows: mediaproxy (2.1.0) unstable; urgency=low . * Mention libnetfilter-conntrack in the installation requirements section * Added note about using distribution provided kernel images * Renamed OpenSER to OpenSIPs * Do accounting only after a 200 OK response was received, which is indicated by the presence of the to_tag that is not sent by OpenSIPS for provisional replies, but only with the 200 OK * Have synthetic test scripts look for the mediaproxy module in current and parent directories automatically * In the synthetic tests have OpenSIPS send the to tag only with 200 OK * Added synthetic test script to test provisional responses from two different PSTN gateways with varying to tags * Enhanced kernel version parsing to support non-standard named kernels (i.e. major.minor.revision.extra) * Updated debian standards version * Rephrased description in manual pages and debian control file * Restructured and improved documentation to better describe how to install * Improved dictionary parsing in RADIUS accounting module * Update support section in readme * Added management interface for OpenSIPS * Fix for relay cleanup timer when dispatcher is restarted * Modified dispatcher to store dialog_id for sessions and use the OpenSIPS management interface to send BYEs when a session did timeout * Added dialog_id value to test scripts * Simplified parsing of headers received from OpenSIPs in dispatcher * Fixed ExpireWatcher port comparison to be done as a 16-bit unsigned int * Remove all matching iptables entries on Inhibitor dealloc in order to remove stray entries that are leftover after crashes * Added per-stream post-dial delay to the session statistics * Record the post dial delay from the first stream into radius accounting * Added note to INSTALL about official Debian repositories from AG Projects * Added note to INSTALL about where to find the changelog * Updated the feature list in README with the latest additions * Fixed regression from 1.x that prevented chaining of mediaproxies * Ignore invalid accounting module names in the configuration and log an error about them instead of raising an exception * Gracefully report the errors that occur while reading the radius configuration file and then exit, instead of throwing an exception * The radius configuration file may be relative to the config directory * Do not cache database inserted rows to avoid the need to clean the cache or otherwise have an ever increasing application memory footprint * Capture and report database errors when inserting statistics records * Use MyISAM for the media_sessions table * Improved the database accounting speed about 3 times using raw queries * Create the table for database accounting on startup if it doesn't exist * Removed pool_size database option and use a single database connection * Reworded the tls README a bit for clarity * Added OpenSIPS section to the config.ini.sample * Added documentation sections for accounting and closing expired calls * Added minimum version dependency for libnetfilter-conntrack in INSTALL * Encode applications and codecs from all streams, not only the first one, in the radius accounting backend * Log the post-dial delay in radius accounting as Acct-Delay-Time * Capture and log radius exceptions For more information, download of source code and the documentation, visit the home page of the product: http://ag-projects.com/MediaProxy.html Regards, Adrian

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[Kamailio-Users] Presence server (acive and stand-bye setup)
by Yoo Chan Jeon 18 Sep '08

18 Sep '08

Hi, I just start to use the Openser as the presence server. I tested with the eyebeam. The server saved the presentity information to the MySQL server. It worked well. I have questions. I would like to setup two presence servers. One is active, and the other is stand-bye. If the active is down, the client sends a message to the stand-bye. How does the presence server know that it is active or stand-bye? How do I set it up? How do I replicate the database between servers? If the active is down, how do I make it up running again? Thanks. Jeon

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[Kamailio-Users] CDRTool 6.6.3 not rating calls
by Adrian Georgescu 18 Sep '08

18 Sep '08

The data in your tables seems to be consistent. Why you still do not get the price is not clear from your email. Adrian

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[Kamailio-Users] Issues with routing
by Jon Farmer 17 Sep '08

17 Sep '08

Hi all I have to following setup for a incoming call Asterisk -> OpenSER -> 3rd Party SBC So the call comes into Asterisk which sends it to OpenSER which forwards it to the 3rd party SBC. I need to use a forward() to send to the SBC as the URI needs to be preserved as it came from Asterisk. This is all working fine, however when the called party answers and the OK is sent for the INVITE it is not sent back to Asterisk. How do I get this to work? Regards Jon

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[Kamailio-Users] CDRTool 6.6.3 not rating calls
by Diego Zuaneti Arruda 17 Sep '08

17 Sep '08

Hi, The CDRTool is not generate rating and prices. My configuration: -- billing_rates | id | gateway | domain | subscriber | name | destination | durationRate | trafficRate | application | connectCost | +----+---------+--------+------------+------+-------------+--------------+-------------+-------------+-------------+ | 1 | | | | 551 | 55 | 1500 | | audio | 0 | | 2 | | | | 552 | 55 | 1000 | | audio | 0 | -- billing_profiles | id | gateway | domain | subscriber | name | rate_name1 | hour1 | rate_name2 | hour2 | rate_name3 | hour3 | rate_name4 | hour4 | +----+---------+--------+------------+------+------------+-------+------------+-------+------------+-------+------------+-------+ | 1 | | | | 551 | 551 | 18 | 552 | 0 | | 0 | | 0 | | 2 | | | | 552 | 552 | 18 | 552 | 0 | | 0 | | 0 | -- billing_customers | id | gateway | domain | subscriber | profile_name1 | profile_name1_alt | profile_name2 | profile_name2_alt | timezone | increment | min_duration | country_code | +----+---------+-------------------+------------+---------------+-------------------+---------------+-------------------+-------------------+-----------+--------------+--------------+ | 1 | | mydomain.com | | 551 | | 552 | | America/Sao_Paulo | 0 | 0 | | -- destinations | id | gateway | domain | subscriber | dest_id | dest_name | +----+---------+--------+------------+---------+-----------+ | 1 | | | | 55 | Brazil | -- And the radacct: | 51 | 1vpzOG09ZNwBth5S(a)192.168.2.29 | 695fa85e025f57eb | 87002(a)mydomain.com | mydomain.com | | 5060 | | 2008-09-17 18:05:58 | 2008-09-17 18:06:03 | 5 | NULL | NULL | | 0 | 0 | 00551533334444(a)mydomain.com | 87002(a)mydomain.com | 200 | Sip-Session | | | 0 | 0 | NULL | Invite | 200 | as7e8c15b3 | gluy7ejee3XZ1ckO | =2B551533334444(a)200.200.200.200 | | audio | | | | 201.201.201.201 | 5060 | 00551533334444(a)mydomain.com | | 0 | | | 0.0000 | 1 | | NULL | | | | | The configuration is correct? Thanks. Diego.

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[Kamailio-Users] digest authentication
by BERGANZ François 17 Sep '08

17 Sep '08

Hello, I need help to do digest authentication ! I have: Route{ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "db_url", "mysql://root:xxx@localhost/openser") if (!proxy_authorize("realm.org", "subscriber")) { proxy_challenge("realm.org", "1"); exit; }; } And, in my database : username domain password realm Ha1 gaia Realm.org xxxxx Where can be my problem ? Cordialement BERGANZ François http://www.acropolistelecom.net <http://www.acropolistelecom.net/>

1 1

[Kamailio-Users] new modules for dialog-info
by Klaus Darilion 17 Sep '08

17 Sep '08

Hi! I have just commited 2 new modules into SVN trunk: - pua_dialoginfo: gets dialog states from dialog module and sends PUBLISH with the dialog state in RFC 4235 conform XML documents - presence_dialoginfo: extends presence module to handle PUBLISH with dialog-info. Allows aggregation of multiple XML documents into one document. With this module you can have BLF without the need for an PBX (Asterisk). It is know to work with SNOM and Linksys phones. Linksys requires some tweaking with module parameters. The work was sponsored by Silver Server (www.sil.at) For a full description please refer to the very extensive documentation in the associated READMEs. regards Klaus

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Re: [Kamailio-Users] t_relay
by Daniel-Constantin Mierla 17 Sep '08

17 Sep '08

sabino frisardi wrote: > OK, so if i call tmb->t_relay(msg,0,0,0,0,0,0) from a module it's the > same if i call t_relay() from script? yes. Daniel > > 2008/9/3 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > On 09/02/08 13:20, Klaus Darilion wrote: > > Hi Sabino! > > Do you really need to call it from within your module? The > usual way is that modules prepare the request (set RURI, DURI, > branches ...) and then in script you call t_relay(). This way > is more flexible as it also allows you to perform some task in > script before sending out the request (if needed). > > klaus > > sabino frisardi schrieb: > > > Hi all, > > I'm using openser 1.4. Can anyone tell me how can i call > t_relay() from another module. > Can anyone help me? any suggestion? > > > if you need it, then tm module exports an API that you can use. > You can have a look at msilo module to see how the TM api is > imported and used later. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > > > -- > Sabino Frisardi -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Iam getting ERROR:db_mysql:db_mysql_new_connection: driver error
by mahesh.peddi＠info-spectrum.com 16 Sep '08

16 Sep '08

Hi, I am working on opensips 1.4.0 and while running "openser -C" command I am getting the following errors. Please any one can help me. Sep 16 21:07:05 localhost opensips[3200]: ERROR:db_mysql:db_mysql_new_connection: driver error: Access denied for user 'opensips'@'localhost' (using password: YES) Sep 16 21:07:05 localhost opensips[3200]: ERROR:core:db_do_init: could not add connection to the pool Sep 16 21:07:05 localhost opensips[3200]: ERROR:rls:mod_init: while connecting database Sep 16 21:07:05 localhost opensips[3200]: ERROR:core:init_mod: failed to initialize module rls Sep 16 21:07:05 localhost opensips[3200]: ERROR:core:main: error while initializing modules Sep 16 21:07:05 localhost opensips[3200]: NOTICE:presence:destroy: destroy module ... Sep 16 21:57:10 localhost yum-updatesd: error getting update info: Cannot find a valid baseurl for repo: extras Waiting for your replies. Thanks & Regards, Mahesh

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[Kamailio-Users] nested redirect
by kentp＠globalpops.com 16 Sep '08

16 Sep '08

i have a simple redirect server working: Call comes from switch (A) to kamailio redirect server (B).. (B) runs perl script to determine route (B) sends Stateless 300 redirect reply to (A). (A) sends ACK to (B). Now, I need to add a nested redirect... call comes from sitch (A) to kamailio redirect server (B).. (B) then needs to send the call to third party proxy (C).. (C) will return 302 redirect.. (B) sends ACK to (C) (B) needs to read the contact header, then run perl script to determine route (using the contact value returned from (c)) (B) then sends 300 redirect back to (A). (A) sends ACK back to (B) so far i have been trying something like this, but the final (A) ACK gets ignored and (B) keeps sending multiple 300's to (A): route { route(1); } route[1] { t_on_reply("1"); t_on_failure("2"); if (!t_relay("proxy.server.c:5060","0x02")) { sl_reply_error(); } onreply_route[1] { if(t_check_status("302")){ $var(lrn_from_bv) = $ct; } } failure_route[2] { if (t_check_status("3[0-9][0-9]")) { route(2); exit; } route[2] { if(perl_exec("callcheck")){ avp_pushto("$ru", "$avp(s:routes)/g"); t_reply("300","Go Here"); exit; } }

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[Kamailio-Users] OpenSER 1.3 crash
by Robin Vleij 16 Sep '08

16 Sep '08

Hi guys! Posted this to the dev list, but I'm not sure if that's the right forum. Looking at the posts there, it's more about patches and dev stuff (hence the name). Hoped to get some feedback on a crash I had a week ago, now I'm trying it here. :) We've been running OpenSER (1.3.0 ntls, compiled for x66_64) for quite a while now and only had a crash once. We suspected then that this was caused by hitting the max_while_loops for a while (never happened again after fixing the max-while-loops). Now I had a crash and the backtrace looks like the following. Do you guys have any pointers on what is causing this? #0 free_hostent (dst=0x6e3fe0) at proxy.c:203 203 for (r=0; dst->h_addr_list[r];r++) { (gdb) backtrace #0 free_hostent (dst=0x6e3fe0) at proxy.c:203 #1 0x0000000000436530 in free_proxy (p=0x6e3fc0) at proxy.c:312 #2 0x00002adec6dc01ea in t_relay_to (p_msg=0x6de4a0, proxy=0x6e3fc0, flags=<value optimized out>) at t_funcs.c:230 #3 0x00002adec6dd1fcd in w_t_relay (p_msg=0x6de4a0, proxy=0x0, flags=0x0) at tm.c:929 #4 0x000000000040ba9f in do_action (a=0x646e80, msg=0x6de4a0) at action.c:816 #5 0x000000000040dfac in run_action_list (a=<value optimized out>, msg=0x6de4a0) at action.c:132 #6 0x000000000040ca06 in do_action (a=0x646fb0, msg=0x6de4a0) at action.c:689 #7 0x000000000040dfac in run_action_list (a=<value optimized out>, msg=0x6de4a0) at action.c:132 #8 0x000000000040d780 in do_action (a=0x647460, msg=0x6de4a0) at action.c:695 #9 0x000000000040dfac in run_action_list (a=<value optimized out>, msg=0x6de4a0) at action.c:132 #10 0x000000000040ca06 in do_action (a=0x6475a0, msg=0x6de4a0) at action.c:689 #11 0x000000000040dfac in run_action_list (a=<value optimized out>, msg=0x6de4a0) at action.c:132 #12 0x000000000040e309 in run_top_route (a=0x643398, msg=0x6de4a0) at action.c:112 #13 0x000000000044b8bf in receive_msg ( buf=0x625ca0 "ACK sip:013104104@sip-corporate.tele2.se SIP/2.0\r\nVia: SIP/2.0/UDP 130.244.190.46;branch=z9hG4bK7b7c7b36\r\nVia: SIP/2.0/UDP 130.244.190.46;branch=z9hG4bK7b7c7b36\r\nVia: SIP/2.0/UDP 130.244.190.46;branch"..., len=3703, rcv_info=0x7fffe4d8d840) at receive.c:156 #14 0x0000000000488154 in udp_rcv_loop () at udp_server.c:438 #15 0x0000000000425081 in main (argc=9, argv=0x7fffe4d8da48) at main.c:834

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[Kamailio-Users] help to query a table with new function in core
by Marcio 15 Sep '08

15 Sep '08

Hi all, I need a help to solve some problems in the clearance of a function to perform a query Mysql....I created a function based on documentation of asipto / kamailio devel which is very good... But as I did not want to create a new module I'm trying to use the module domain... whose goal a proxy SIP receive msg that will should validate the realm comparing with a table set up in MySQL(trusdomain)... it should be used into config file to to declare of the table created in the Mysql: modparam ( "domain", "trusdomain_table," trusdomain) the version table of the openser has updated with the creation of this new table (trusdomain) and created a function into file: domain.c / h and a part was need be exported to config file in domain_mod.c / h....but on the tests a call..this proxy kills the pid openser ... Sep 5 06:32:21 notehome openser[7962]: INFO:core:handle_sigs: child process 7964 exited by a signal 11 Sep 5 06:32:21 notehome openser[7962]: INFO:core:handle_sigs: core was generated Sep 5 06:32:21 notehome openser[7962]: INFO:core:handle_sigs: terminating due to SIGCHLD Sep 5 06:32:21 notehome openser[7967]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7963]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7968]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7966]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7965]: INFO:core:sig_usr: signal 15 received The function created in domain.c: int is_from_trusted(struct sip_msg* _msg, char* _s1, char* _s2) { struct sip_uri *puri; /* struct str* _host; */ /* str table_trus; str col_id; str col_dom; str col_lastmod; */ str* _host = (str*)(&(puri->host)); char* db_url = DEFAULT_RODB_URL; db_key_t keys[1]; /* Tipo da coluna .. Nome das colunas */ db_val_t vals[1]; db_key_t cols[1]; db_res_t* res = NULL; if ((puri=parse_from_uri(_msg))==NULL){ LM_ERR("Error while parsing From header\n"); return -2; } /* Bind the database module */ if (bind_dbmod(db_url, &domain_dbf )) { LM_ERR("Cannot bind to database module!\n"); return -1; } /* Check for SELECT capability */ if (!DB_CAPABILITY(domain_dbf, DB_CAP_QUERY)){ LM_ERR("Database modules does not provide all functions needed\n"); return -1; } /* Connect to DB */ db_handle = domain_dbf.init(db_url); if (!db_handle){ LM_ERR("failed to connect database\n"); return -1; } keys[0]=trusdomain_col.s; cols[0]=trusdomain_col.s; LM_INFO(">>domain.c>>-2-IF RES_ROW_N(res)=>%d\n",RES_ROW_N(res)); /* Estrutura dos vls das colunas - arq. db/db_val.h */ VAL_TYPE(vals) = DB_STR; VAL_NULL(vals) = 0; VAL_STR(vals).s = _host->s; VAL_STR(vals).len = _host->len; /* LM_INFO(">>domain.c>>-1-connector=%d and tab=%s\n",db_handle,trusdomain_table.s); */ if (domain_dbf.use_table(db_handle, trusdomain_table.s) < 0) { LM_ERR("Error while trying to use trusdomain table\n"); return -1; } if (domain_dbf.query(db_handle, keys, NULL, vals, cols, 1, 1, 0, &res) < 0) { LM_ERR("Error while querying database\n"); domain_dbf.close(db_handle); return -1; } /* RES_ROW_N(res) Num. de linhas no fetch */ if (RES_ROW_N(res) <= 0 || RES_ROWS(res)[0].values[0].nul != 0){ LM_DBG("no value found\n"); if (res != NULL && domain_dbf.free_result(db_handle, res) < 0) LM_DBG("failed to free the result\n"); domain_dbf.close(db_handle); } else if (RES_ROW_N(res) >= 1) { /* Teste do que estah na tab. trusdomain com o que vem na URI */ LM_DBG("Realm '%.*s' is trusted\n",_host->len, ZSW(_host->s)); } /* Free the result */ domain_dbf.free_result(db_handle, res); domain_dbf.close(db_handle); return 0; } And declaration in domain_mod.c : ... /* by mrg */ #define TRUSDOMAIN_TABLE "trusdomain" #define TRUSDOMAIN_TABLE_LEN (sizeof(TRUSDOMAIN_TABLE) - 1) #define TRUSDOMAIN_COL "trusdomain" #define TRUSDOMAIN_COL_LEN (sizeof(TRUSDOMAIN_COL) - 1) ... ... str trusdomain_table = {TRUSDOMAIN_TABLE, TRUSDOMAIN_TABLE_LEN}; /* Name of trusted table */ str trusdomain_col = {TRUSDOMAIN_COL, TRUSDOMAIN_COL_LEN}; /* Name of trusted column */ ... ... /* * Exported functions */ static cmd_export_t cmds[] = { ............. {"is_domain_local", w_is_domain_local, 1, parameter_fixup, 0, REQUEST_ROUTE|FAILURE_ROUTE|BRANCH_ROUTE}, {"is_from_trusted", is_from_trusted, 0, 0, 0, REQUEST_ROUTE|FAILURE_ROUTE|BRANCH_ROUTE}, {0, 0, 0, 0, 0, 0} }; .../* * Exported parameters */ static param_export_t params[] = { ............ {"domain_table", STR_PARAM, &domain_table.s}, {"domain_col", STR_PARAM, &domain_col.s }, {"trusdomain_table", STR_PARAM, &trusdomain_table.s}, {"trusdomain_col", STR_PARAM, &trusdomain_col.s}, {0, 0, 0} }; ... ... static int mod_init(void) { int vertr....; ..... trusdomain_table.len = strlen(trusdomain_table.s); trusdomain_col.len = strlen(trusdomain_col.s); ..... if (db_mode != 0) { if (domain_db_init(db_url.s)<0) return -1; /* Check table version */ ver = domain_db_ver(&domain_table); vertr = domain_db_ver(&trusdomain_table); if ((ver < 0) && (vertr < 0)) { LM_ERR("Error while querying table version\n"); goto error; } else if ((ver < TABLE_VERSION) && (vertr < TABLE_VERSION)) { LM_ERR("Invalid table version <%d> trustable version <%d> (should be %d)\n", ver, vertr, TABLE_VERSION); goto error; } ....... ....... /* Initializing hash tables and hash table variable */ ........ domain_db_close(); } return 0; error: domain_db_close(); return -1; } So I think that I must be something doing wrong ..... as would be the right and as I could learn fast to resolve error debugging of pID of Openser that has been deads...??? thx Marcio

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[Kamailio-Users] project updates
by Daniel-Constantin Mierla 15 Sep '08

15 Sep '08

Hello everybody, just want to give some updates concerning two main topics. First, regarding the project name, many expressed the concern that a rename in short time will create more confusion. As we are not in front of a new major release, I think it is better to postpone this process for a while and focus on the second topic, the organization for the project. For this one, I have to thank personally to many people from the list that wrote us, giving suggestions and helping finding solutions. As a result, we are working to create the draft of statute for organization, but we are also investigating the option to join an existing organization, dedicated to hosting of open source projects. We will keep you updated. Meanwhile, the svn repository got couple of new features as well as refactoring of some modules. 1.4.1 will be released soon as we got a bunch of important fixes in the 1.4 branch. Most probably another release in 1.3 branch will follow. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] record_route to control my cancel.
by BERGANZ François 15 Sep '08

15 Sep '08

Hello, I dont really understand, but : I just need to do in the INVITE : record_route_preset('$dd:$dp;codec=alaw'); forward(); and for the CANCEL: if(check_route_param(";codec=alaw")){ route(2); } Is it possible? Thank you Cordialement BERGANZ François http://www.acropolistelecom.net

5 6

[Kamailio-Users] Question about Authentication
by mark morreny 15 Sep '08

15 Sep '08

Hi, I am able to get authentication setup with Mysql and Freeradius. Just one simple question, for pure authentication purpose, is it better just to use the Mysql database for authentication instead of setting up Freeradius? Is there any advantage of using Freeradius on top of Mysql? If I use Mysql to serve authentication for Openser, is it possible to encrypt the password in Mysql? Is it possible to specify the table and fields to be used for authentication instead of using the default mysql tables? Thanks for your help. Regards, Mark

3 3

[Kamailio-Users] DNS_name to IP coredumps on calee bye
by Stefan Broddesson 15 Sep '08

15 Sep '08

Greetings, Rev. is 1.3.3 on debian_Etch_64 Have a scenarion that always triggers a core dump. On Calee initiated <bye> and where destination address is a hostname not IP address. Executing forward() or t_relay() in script triggers the coredump. dns & rev_dns On/Off with the same result. --- DBG:core:comp_scriptvar: str 20 : siphost.domain.com DBG:tm:t_newtran: transaction on entrance=0xffffffffffffffff DBG:core:parse_headers: flags=ffffffffffffffff DBG:core:parse_headers: flags=78 DBG:tm:t_lookup_request: start searching: hash=7994, isACK=0 DBG:tm:matching_3261: RFC3261 transaction matching failed DBG:tm:t_lookup_request: no transaction found DBG:tm:run_reqin_callbacks: trans=0x2b4e6b2e0f48, callback type 1, id 0 entered DBG:siptrace:trace_onreq_in: trace off... DBG:core:mk_proxy: doing DNS lookup... DBG:core:sip_resolvehost: no port, no proto -> do NAPTR lookup! DBG:core:filter_and_sort_naptr: found valid SIP+D2U -> _sip._udp.siphost.domain.com DBG:core:filter_and_sort_naptr: skipping SIPS+D2T -> _sips._tcp.siphost.domain.com DBG:core:filter_and_sort_naptr: skipping SIP+D2T -> _sip._tcp.siphost.domain.com Have a workaround that rewrites destination in case of hostName trig. impractical though. -hints? //s __________________________________________________________ Ta semester! - sök efter resor hos Kelkoo. Jämför pris på flygbiljetter och hotellrum här: http://www.kelkoo.se/c-169901-resor-biljetter.html?partnerId=96914052

3 2

[Kamailio-Users] MediaProxy 2.0.3 on OpenSUSE 11
by Darren Sessions 15 Sep '08

15 Sep '08

Just finished some documentation on how to install MediaProxy 2.0.3 on OpenSUSE 11. Due to the length of the post and the init scripts provided, I won't be re-posting this one in the lists like usual. Thanks do go out to Ruud Klaver @ AG Projects for a clever Linux version detection hack for the relay module (hopefully will make it into the next release of MediaProxy). Everything can be found on my site at http://www.darrensessions.com. Cheers! _____________________________ Darren Sessions dmsessions(a)gmail.com http://www.darrensessions.com _____________________________

1 0

[Kamailio-Users] Help to create in core
by Marcio 13 Sep '08

13 Sep '08

Hi all, I appreciate very much to help solve some problems in the clearance of a function to perform a query Mysql .... created the function based on documentation of asipto / kamailio devel which is great ... but I did not want to create a new module used the module in domain ... whose goal is a proxy to receive SIP msg should validate the realm with what is in a table (Trustee) was created through a query ... it should be used for the config file to declare of the table created in the Mysql: modparam ( "domain", "trusdomain_table," trusdomain) the table version updated with the creation of this table (trusdomain) and create the own function in the file: domain.c / h part to be exported in domain_mod.c / h....but on the tests a call SIP to proxy kills a openser's pid ... Sep 5 06:32:21 notehome openser[7962]: INFO:core:handle_sigs: child process 7964 exited by a signal 11 Sep 5 06:32:21 notehome openser[7962]: INFO:core:handle_sigs: core was generated Sep 5 06:32:21 notehome openser[7962]: INFO:core:handle_sigs: terminating due to SIGCHLD Sep 5 06:32:21 notehome openser[7967]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7963]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7968]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7966]: INFO:core:sig_usr: signal 15 received Sep 5 06:32:21 notehome openser[7965]: INFO:core:sig_usr: signal 15 received *The function created in domain.c:* int is_from_trusted(struct sip_msg* _msg, char* _s1, char* _s2) { struct sip_uri *puri; /* struct str* _host; */ /* str table_trus; str col_id; str col_dom; str col_lastmod; */ str* _host = (str*)(&(puri->host)); char* db_url = DEFAULT_RODB_URL; db_key_t keys[1]; /* Tipo da coluna .. Nome das colunas */ db_val_t vals[1]; db_key_t cols[1]; db_res_t* res = NULL; if ((puri=parse_from_uri(_msg))==NULL){ LM_ERR("Error while parsing From header\n"); return -2; } /* Bind the database module */ if (bind_dbmod(db_url, &domain_dbf )) { LM_ERR("Cannot bind to database module!\n"); return -1; } /* Check for SELECT capability */ if (!DB_CAPABILITY(domain_dbf, DB_CAP_QUERY)){ LM_ERR("Database modules does not provide all functions needed\n"); return -1; } /* Connect to DB */ db_handle = domain_dbf.init(db_url); if (!db_handle){ LM_ERR("failed to connect database\n"); return -1; } keys[0]=trusdomain_col.s; cols[0]=trusdomain_col.s; LM_INFO(">>domain.c>>-2-IF RES_ROW_N(res)=>%d\n",RES_ROW_N(res)); /* Estrutura dos vls das colunas - arq. db/db_val.h */ VAL_TYPE(vals) = DB_STR; VAL_NULL(vals) = 0; VAL_STR(vals).s = _host->s; VAL_STR(vals).len = _host->len; /* LM_INFO(">>domain.c>>-1-connector=%d and tab=%s\n",db_handle,trusdomain_table.s); */ if (domain_dbf.use_table(db_handle, trusdomain_table.s) < 0) { LM_ERR("Error while trying to use trusdomain table\n"); return -1; } if (domain_dbf.query(db_handle, keys, NULL, vals, cols, 1, 1, 0, &res) < 0) { LM_ERR("Error while querying database\n"); domain_dbf.close(db_handle); return -1; } /* RES_ROW_N(res) Num. de linhas no fetch */ if (RES_ROW_N(res) <= 0 || RES_ROWS(res)[0].values[0].nul != 0){ LM_DBG("no value found\n"); if (res != NULL && domain_dbf.free_result(db_handle, res) < 0) LM_DBG("failed to free the result\n"); domain_dbf.close(db_handle); } else if (RES_ROW_N(res) >= 1) { /* Teste do que estah na tab. trusdomain com o que vem na URI */ LM_DBG("Realm '%.*s' is trusted\n",_host->len, ZSW(_host->s)); } /* Free the result */ domain_dbf.free_result(db_handle, res); domain_dbf.close(db_handle); return 0; } *And declaration in domain_mod.c :* ... /* by mrg */ #define TRUSDOMAIN_TABLE "trusdomain" #define TRUSDOMAIN_TABLE_LEN (sizeof(TRUSDOMAIN_TABLE) - 1) #define TRUSDOMAIN_COL "trusdomain" #define TRUSDOMAIN_COL_LEN (sizeof(TRUSDOMAIN_COL) - 1) ... ... str trusdomain_table = {TRUSDOMAIN_TABLE, TRUSDOMAIN_TABLE_LEN}; /* Name of trusted table */ str trusdomain_col = {TRUSDOMAIN_COL, TRUSDOMAIN_COL_LEN}; /* Name of trusted column */ ... ... /* * * Exported functions */ static cmd_export_t cmds[] = { {"is_from_local", is_from_local, 0, 0, 0, REQUEST_ROUTE|BRANCH_ROUTE|FAILURE_ROUTE}, {"is_uri_host_local", is_uri_host_local, 0, 0, 0, REQUEST_ROUTE|BRANCH_ROUTE|FAILURE_ROUTE}, {"is_domain_local", w_is_domain_local, 1, parameter_fixup, 0, REQUEST_ROUTE|FAILURE_ROUTE|BRANCH_ROUTE}, {"is_from_trusted", is_from_trusted, 0, 0, 0, REQUEST_ROUTE|FAILURE_ROUTE|BRANCH_ROUTE}, {0, 0, 0, 0, 0, 0} }; .../* * Exported parameters */ static param_export_t params[] = { {"db_url", STR_PARAM, &db_url.s }, {"db_mode", INT_PARAM, &db_mode }, {"domain_table", STR_PARAM, &domain_table.s}, {"domain_col", STR_PARAM, &domain_col.s }, {"trusdomain_table", STR_PARAM, &trusdomain_table.s}, {"trusdomain_col", STR_PARAM, &trusdomain_col.s}, {0, 0, 0} }; ... ... static int mod_init(void) { int vertr....; ..... trusdomain_table.len = strlen(trusdomain_table.s); trusdomain_col.len = strlen(trusdomain_col.s); ..... if (db_mode != 0) { if (domain_db_init(db_url.s)<0) return -1; /* Check table version */ ver = domain_db_ver(&domain_table); vertr = domain_db_ver(&trusdomain_table); if ((ver < 0) && (vertr < 0)) { LM_ERR("Error while querying table version\n"); goto error; } else if ((ver < TABLE_VERSION) && (vertr < TABLE_VERSION)) { LM_ERR("Invalid table version <%d> trustable version <%d> (should be %d)\n", ver, vertr, TABLE_VERSION); goto error; } ....... ....... /* Initializing hash tables and hash table variable */ ........ domain_db_close(); } return 0; error: domain_db_close(); return -1; } So I think I must be doing everything wrong ..... as would be the right and as I could learn so fast as this error debugging of pID of Openser is dead...??? Any help would be greatly appreciated... thx Marcio

1 0

Re: [Kamailio-Users] mediaproxy on centos 5.x
by Ricky Gutierrez 13 Sep '08

13 Sep '08

Hi list, this was the problem my version of libnetfilter-conntrack-dev thank Dan and list rickygm ----- Original Message ---- From: Dan Pascu <dan(a)ag-projects.com> To: Ricky Gutierrez <xserverlinux(a)yahoo.com> Sent: Friday, September 12, 2008 4:57:14 AM Subject: Re: mediaproxy on centos 5.x On Friday 12 September 2008, Ricky Gutierrez wrote: > Hi Dan , I am trying to install mediaproxy on a centos 5.2, but when I > am Settling it shows this error and it doesn't finish installing > > mediaproxy/interfaces/system/_conntrack.c: In function > âForwardingRule_initâ: mediaproxy/interfaces/system/_conntrack.c:223: > error: âNFCT_SOPT_SETUP_REPLYâ undeclared (first use in this function) > mediaproxy/interfaces/system/_conntrack.c:223: error: (Each undeclared > identifier is reported only once > mediaproxy/interfaces/system/_conntrack.c:223: error: for each function > it appears in.) error: command 'gcc' failed with exit status 1 > > I have followed this mail, I have installed these dependences but it > even fails me Make sure you have libnetfilter-conntrack-dev at least version 0.0.89 The symbol it doesn't find was introduced on 4 Jun 2007, so I guess you have a pretty old version (it first appeared in 0.0.80, but we recommend you use at least 0.0.89 since that is what we used while developing it and it's guaranteed to work with). Better yet use the latest version (0.0.96) > > http://lists.opensips.org/pipermail/users/2008-August/000131.html > > could you help me? > > God blesses you > > rickygm -- Dan

1 0

[Serusers] ser authentication user
by Amos Shahar 12 Sep '08

12 Sep '08

Hi, how can I add user to ser where the auth user is diffrent the user (e.g my DID)? for example I have: user 12127773456 (my number in the header, from, contact....) pass 1234 but the authentication user is 10000 (for the autherization proxy) using serctl does not have this parameter: serctl add <username> <password> <email> .. add a new subscriber for example in X-lite I want to have diffrent user name and autherization user name and to be able to register to SER. Thanks! Amos

1 0

[Kamailio-Users] SIP NOTIFY for Message-Waiting
by George Lee 12 Sep '08

12 Sep '08

Hi, I know Kamailio SIP server can handle SUBSCRIBE for message-waiting indication. Is there a way to ask the SIP server to generate a NOTIFY with message summary event and its related headers like Message-waiting, Message-Account, Voice-Message, etc.? Or I must use Asterisk as my mail server and forward all mwi Subscribes to Asterisk? Thanks, George

2 3

[Kamailio-Users] after an append_hf
by BERGANZ François 12 Sep '08

12 Sep '08

hello all I need help for 'append_hf' : I know how to add sip header with append_hf but how can look at this after (in another server) ? I need to do append_hf("user: XXX"\r\n"); and after (with another OpenSer) look at the string after user: ? How to do? Cordialement BERGANZ François http://www.acropolistelecom.net <http://www.acropolistelecom.net/>

3 10

[Kamailio-Users] Changing Source IP per Route
by Ali Jawad 12 Sep '08

12 Sep '08

Hi All Can the following be achieved.: I do have a subscription at a carrier with two routes, on normal route and one premium route. The problem is that I do have to use a different source IP with each of these two routes. This would be easy to achieve with two different openser boxes. However I do have only one at current time. So, is it possible to use a different source IP per route ? Thanks With Regards Ali Jawad

4 3

[Kamailio-Users] branch_route and onreply_route for CANCEL and ACK
by Alex Hermann 12 Sep '08

12 Sep '08

Hello, I use t_onbranch and t_on_reply for all request passing the proxy. Why doesn't the branch_route get called for CANCEL's and ACK's (loose routed as well as statefull)? The loose routed BYE does pass through the branch_route. And why isn't the reply on a CANCEL passed through the transactional onreply_route? It is passed through the default onreply_route. The documentation doesn't mention any selectivity on request method. -- Greetings, Alex Hermann

2 1

[Serusers] DNS server and registered users
by Magnus Sörman 12 Sep '08

12 Sep '08

Hi, How do I do the following in SER 2.0: 1. How do I specify which DNS server that should be used? E.g. That a reverse look-up should go to the DNS server running on the same machine as SER, instead of the once specified in /etc/resolv.conf. 2. If I'm not using the mysql database, how can I see how many users that are registered in usrloc/RAM? Btw, 'ser_ctl user show <uri>' didn't even find the successfully registered users, so maybe that's something I somehow need to address first. Thanks, /Magnus

1 0

[Serusers] How to use record route to change path of BYE messages
by 邱磊 12 Sep '08

12 Sep '08

Hi,everyone For voice communication I use a SIP+RTP proxy together with SER. For PSTN calls SER routes the INVITE messages to the RTP proxy. Everything works fine there, but when the call is ended by one client (which is connected to SER) or the PSTN user, the BYE message arrive to SER but dont forward it. If our RTP proxy does not see the message coming from SER, call doesn't end. That's why I need all of the BYE messages to go through SER. How can I do that ? In the documents they said that the bye message loose route if there are ";lr" and If the message should be loose routed then SER should simply relay the message to the next destination as specified in the top-most Record-Route header field .but i have a try and nothing happened! that is my config block: # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" # ------------------------- request routing logic ------------------- # main routing logic route { record_route(); if(message=="BYE") loose_route(); }; my ser is 1.2.3.4:5060 this is my sip message: BYE sip:0000123@19.68.150.20:31562 SIP/2.0^M Via: SIP/2.0/UDP 23.43.53.4:5060;branch=z9hG4bK5a69ae91;rport^M Record-Route: <sip:1.2.3.4:5060;ftag=28118cbe-30554463;lr>^M Record-Route: <sip:12.45.32.61:5060;r2=on;lr;ftag=28118cbe-30554463>^M From: <sip:123456@1.2.3.4:5060;user=phone>;tag=as7232fbd6^M To: "beurself212" <sip:0000123@19.68.150.20:31562>;tag=28118cbe-30554463^M Call-ID: 76ad6e55-86f19c97-MzA1NTQ0NjM(a)192.168.50.50^M Thanks, Jimmy 2008-09-12 邱磊

1 0

[Serusers] failure route problem - duplicate invites
by Matthew McGuire 11 Sep '08

11 Sep '08

I'm having a problem with the route sections below. When SER receives an Invite to proxy to a PSTN gateway, SER is sending 2 invites very fast to the gateway in the first route block which ends up creating a mess with the gateway. Either I have something wrong in the routing code or I have an SER/OS/hardware timing problem. Any suggestions ? route[1] { xlog("L_INFO", "Sending to route 1\n%mb\n"); rewritehostport ("1.1.1.1:5060"); append_branch(); t_on_failure("1"); t_relay(); } failure_route[1] { if(t_check_status("487")) { break; } xlog( "L_INFO", "failure on route 1\n%mb\n"); append_branch(); route(2); break; } route[2] { xlog("L_INFO", "Sending to route 2\n%mb\n"); rewritehostport ("2.2.2.2:5060"); append_branch(); t_on_failure("2"); t_relay(); } failure_route[2] { if(t_check_status("487")) { break; } xlog( "L_INFO", "failure on route 2\n%mb\n"); append_branch(); route(3); break; } route[3] { xlog("L_INFO", "Sending to rout 3\n%mb\n"); rewritehostport ("3.3.3.3:5060"); t_relay(); } Thanks, Matt

2 3

Re: [Kamailio-Users] help installing mediaproxy in centos 5
by Ricky Gutierrez 11 Sep '08

11 Sep '08

I install the dependences in my centos "kernerl-devel" and the netfilter kernel module . "libnetfilter-conntrack-dev" but it still shows me the installation error I read this mail, thinking that that it could be my problem http://lists.opensips.org/pipermail/users/2008-August/000131.html thank you for all help that I can be offered rickygm ----- Original Message ---- From: Peter Lemenkov <lemenkov(a)gmail.com> Sent: Wednesday, September 10, 2008 11:13:57 PM Subject: Re: [Kamailio-Users] help installing mediaproxy in centos 5 > I've got different advise. After quick googling I found that these >constants are related to netfilter kernel module, so you probably need > to install kernel-headers and pass properly additional includedirs (if > required) to mediaproxy buildsystem.

1 0

[Serusers] Difficulties "making" SER v2
by Steven C. Blair 11 Sep '08

11 Sep '08

I'm trying build make SERv2 on a RedHat 2.6.18-92.1.10.el5xen SMP machine. I'm having difficulty with include definition conflicts. Are there specific instructions for making SER on 64bit architecture machines? The first error occurs with futex.h Thanks

1 0

[Kamailio-Users] custom module depends on usrloc, the load order
by toly 11 Sep '08

11 Sep '08

Greetings, I have 2 questions. 1. I've created custom module where I'd like to get number of users from usrloc module. Here is the code: LOG(L_INFO,"PRESOURCE: Initializing\n"); bind_usrloc = (bind_usrloc_t)find_export("ul_bind_usrloc", 1, 0); if ( !bind_usrloc ) { LOG(L_CRIT, "Can't bind usrloc (find_export) check if loaded"); goto error; } if ( !bind_usrloc(&ul) ) { LOG(L_CRIT, "Can't bind usrloc"); goto error; } When this code is enabled if fails on second if. Looks like usrloc module is not loaded. Here's the log 09/10 18:50:57 4431 info <statistics.c: 135> INFO: statistics manager successfully initialized 09/10 18:50:57 4431 info <uac.c: 139> UAC - initializing 09/10 18:50:57 4431 info <hslot.c: 44> INFO:ul_init_locks: locks array size 512 09/10 18:50:57 4431 info <pike.c: 108> PIKE - initializing 09/10 18:50:57 4431 info <ip_tree.c: 86> INFO:pike:init_lock_set: probing 256 set size 09/10 18:50:57 4431 info <avpops.c: 164> AVPops - initializing 09/10 18:50:57 4431 info <sl.c: 130> StateLess module - initializing 09/10 18:50:57 4431 info <tm.c: 586> TM - initializing... 09/10 18:50:57 4431 notice <maxfwd.c: 97> Maxfwd module- initializing 09/10 18:50:57 4431 info <presource.c: 146> PRESOURCE: Initializing 09/10 18:50:57 4431 crit <presource.c: 155> Can't bind usrloc 09/10 18:50:57 4431 err <sr_module.c: 465> init_mod(): Error while initializing module presource In config script loadmodule("usrloc.so") is prior to load my custom module. When I disable above code, then load order follows config file order. 2. Here's how I wanted to get number of users: unsigned long get_number_of_users() { return ul.get_number_of_users(); } But usrloc_api_t does not contain this cmd function despite it's exported in usrloc module typedef struct usrloc_api { int use_domain; int db_mode; unsigned int nat_flag; register_udomain_t register_udomain; get_all_ucontacts_t get_all_ucontacts; insert_urecord_t insert_urecord; delete_urecord_t delete_urecord; get_urecord_t get_urecord; lock_udomain_t lock_udomain; unlock_udomain_t unlock_udomain; release_urecord_t release_urecord; insert_ucontact_t insert_ucontact; delete_ucontact_t delete_ucontact; get_ucontact_t get_ucontact; update_ucontact_t update_ucontact; register_watcher_t register_watcher; unregister_watcher_t unregister_watcher; register_ulcb_t register_ulcb; } usrloc_api_t; Thanks, Toly -- View this message in context: http://www.nabble.com/custom-module-depends-on-usrloc%2C-the-load-order-tp1… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 3

[Serusers] Sarfaraz has invited you to join iDeezire - Keeps Connected!
by Sarfaraz 11 Sep '08

11 Sep '08

1 0

[Serusers] Sarfaraz has invited you to join iDeezire - Keeps Connected!
by Sarfaraz 11 Sep '08

11 Sep '08

1 0

[Kamailio-Users] preparing 1.4.1 release
by Daniel-Constantin Mierla 11 Sep '08

11 Sep '08

Hello, there are already several important fixes since 1.4.0, so we should look to build the first maintenance release for branch 1.4. I am thinking of 2 weeks time frame from now. Make sure you point outstanding issues you are aware of as soon as possible on the tracker. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 2

[Kamailio-Users] help installing mediaproxy in centos 5
by Ricky Gutierrez 11 Sep '08

11 Sep '08

Hi list is trying to install mediaproxy 2.0.3 in centos 5 and when I make python ./setup.py install it generates me this error: mediaproxy/interfaces/system/_conntrack.c:490: warning: implicit declaration of function ânfct_catchâ mediaproxy/interfaces/system/_conntrack.c:491: warning: implicit declaration of function ânfct_callback_unregisterâ mediaproxy/interfaces/system/_conntrack.c:502: warning: implicit declaration of function ânfct_get_attr_u8â mediaproxy/interfaces/system/_conntrack.c:502: error: âATTR_ORIG_L3PROTOâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:503: error: âATTR_REPL_L3PROTOâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:504: error: âATTR_ORIG_L4PROTOâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:505: error: âATTR_REPL_L4PROTOâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:513: error: âATTR_ORIG_IPV4_SRCâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:514: error: âATTR_ORIG_PORT_SRCâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:515: error: âATTR_DNAT_IPV4â undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:515: error: âATTR_REPL_IPV4_SRCâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:516: error: âATTR_DNAT_PORTâ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:516: error: âATTR_REPL_PORT_SRCâ undeclared (first use in this function) error: command 'gcc' failed with exit status 1 I already have installed the bookstore python any suggestion is appreciated? best regards rickygm

4 6

[OpenSER-Users] Problems with SerMyAdmin Installation
by Mark Koopmann 10 Sep '08

10 Sep '08

Hi everybody, does anybody has experience with installing the java tool SerMyAdmin? The Tool is running but every action ends with an runtime exception. Trying to register a new user ends with the following message: Grails Runtime Exception Error Details Message: Caused by: java.lang.NullPointerException Class: RegisterUserController At Line: [98] Code Snippet: (empty) Stack Trace org.codehaus.groovy.runtime.InvokerInvocationException: java.lang.NullPointerException 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.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.WebInterceptorFilter.doFilterInternal(WebInterceptorFilter.java:106) at org.jsecurity.web.servlet.SecurityContextFilter.doFilterInternal(SecurityContextFilter.java:93) at org.jsecurity.web.servlet.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:106) 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:595)Caused by: java.lang.NullPointerException at RegisterUser.toString(RegisterUser.groovy) at RegisterUserController$_closure8.doCall(RegisterUserController.groovy:98) at RegisterUserController$_closure8.doCall(RegisterUserController.groovy) ... 29 more Unfortunately, i dont have much experience with installing 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 Mark Koopmann

5 7

[Kamailio-Users] controle the dispatcher
by BERGANZ François 10 Sep '08

10 Sep '08

Hello, I need to check where is send the frame to the next server, I need to do something like: ds_select_dst("1","0"); xlog("L_ERR", " methode <$rm> ruri:$ruri ru:$ru\n"); forward(); exit; I want to log to next server! How to do? Thank you Cordialement BERGANZ François http://www.acropolistelecom.net TEL/FAX : 33 (0) 1 70 72 50 15

3 2

[Kamailio-Users] Prefix with variable?
by Robin Vleij 10 Sep '08

10 Sep '08

Hi! I'm running into a little problem using the prefix() function. According to the docs I can only use strings as prefixes and this works fine everywhere in our config. Now I'm trying to do something like this: prefix($avp(s:t_prefix)); Customer should be able to choose their own prefixes for certain types of traffic. We get a custom prefix in from the PSTN gateways, where we want to then replace that "internal" prefix (strip(4)) with a configurable prefix. Why can't we use variables in the prefix function? I could try to use $ru = $avp(s:t_prefix) + $original_ru or something, but on a short try-run this didn't seem as easy as that. Any ideas or anyone doing something like this? /robin -- Robin Vleij robin(a)swip.net

2 2

[Kamailio-Users] CANCEL can't be forward!
by BERGANZ François 10 Sep '08

10 Sep '08

hello a problem with my CANCEL frames! user-------openser------asterisk invite>>>>>>>|>>>>>>invite ringing<<<<<<|<<<<<<ringing cancel>>>>>>>| in fact, I don't know where forward the cancel in stateless! I have no information in the frame to forward it to the right asterisk (one for g711, one for g729...) have you an idea? thank you all Cordialement BERGANZ François <http://www.acropolistelecom.net> http://www.acropolistelecom.net TEL/FAX : 33 (0) 1 70 72 50 15

3 4

[Kamailio-Users] failure route problem - duplicate invites
by Matthew McGuire 10 Sep '08

10 Sep '08

I'm having a problem with the route sections below. When my proxy receives an Invite to proxy to a PSTN gateway, the proxy is sending 2 invites very fast to the gateway in the first route block which ends up creating a mess with the gateway. Either I have something wrong in the routing code or I have an SER/OS/hardware timing problem. Any suggestions ? route[1] { xlog("L_INFO", "Sending to route 1\n%mb\n"); rewritehostport ("1.1.1.1:5060"); append_branch(); t_on_failure("1"); t_relay(); } failure_route[1] { if(t_check_status("487")) { break; } xlog( "L_INFO", "failure on route 1\n%mb\n"); append_branch(); route(2); break; } route[2] { xlog("L_INFO", "Sending to route 2\n%mb\n"); rewritehostport ("2.2.2.2:5060"); append_branch(); t_on_failure("2"); t_relay(); } failure_route[2] { if(t_check_status("487")) { break; } xlog( "L_INFO", "failure on route 2\n%mb\n"); append_branch(); route(3); break; } route[3] { xlog("L_INFO", "Sending to rout 3\n%mb\n"); rewritehostport ("3.3.3.3:5060"); t_relay(); } Thanks, Matt

5 11

[Serusers] help required
by Engr. Muhammad Noman Saeed 10 Sep '08

10 Sep '08

Dear All, I need your help for establishing SER as a SIP server for my organization. I have downloaded it at my fedora 8.0 machine, but as i am new with linux and SER it is hard to find me how to install/configure. kindly help me as a new SER user. regards -- ========================= Noman =========================

2 1

[Serusers] how can i config ser.cfg
by 邱磊 10 Sep '08

10 Sep '08

hi friends: i need some help! frist,my asterisk receive a sip message from sip proxy via SER.when my asterisk send 200 to SER and then receive ACK from proxy directly! when asterisk send BYE to the proxy and can hangup the call. but now i redirect the BYE to SER, i found the SER can not send the message to proxy! what can i do in the ser.cfg? thanks in advance 2008-09-10 邱磊

1 0

[Kamailio-Users] add number to an avp ?
by BERGANZ François 09 Sep '08

09 Sep '08

I need to add 1 to an avp like : $avp(i:681)=$avp(i:681)+1; but don't work! how to? thank you Cordialement BERGANZ François http://www.acropolistelecom.net TEL/FAX : 33 (0) 1 70 72 50 15

2 2

[Serusers] Dubt on RFC 4484 and SIP Proxy Server behavior
by Francesco la Torre 09 Sep '08

09 Sep '08

Hi all, I've a doubt regarding a statement at the end of page 5 "... Some important design decisions are associated with carrying assertions in a SIP request. If an assertion is carried by value, or uses a MIME-based content indirection system, then proxy servers will be unable to inspect the assertion themselves. If the assertion were referenced in a header, however, it might be possible for the proxy to acquire and inspect the assertion itself. There ..." Why a proxy server will be unable to inspect the assertion themself ? SER has functions by which it could be read message body, does this behavior not respect any rfc ? I'm missing something :-S Thanks in advance, Francesco la Torre

2 4

[Kamailio-Users] [Fwd: Re: Unconditional call forward]
by Stefano Palleschi 09 Sep '08

09 Sep '08

Hi Gustavo, thanks for your reply. I've changed the *avp type* from "2" to "0" but I get always the same error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> Now I'm going to reboot the server... If I'll have news I'll let you know. Regards, Stefano -------- Messaggio Originale -------- Oggetto: Re: [Kamailio-Users] Unconditional call forward Data: Wed, 3 Sep 2008 10:38:42 -0300 Da: Gustavo Mistrinelli <gmistrinelli(a)gmail.com> A: Stefano Palleschi <stefano.palleschi(a)okcom.it> CC: users(a)lists.kamailio.org Referenze: <48BE6299.5000805(a)okcom.it> Hi Stefano, Please check *avp type* that you are using, use type 0 instead of 2 See below the different types for avps: 0 - *AVP* with string name and string value 1 - *AVP* with string name and integer value 2 - *AVP* with integer name and string value 3 - *AVP* with integer name and integer value Regards, Gustavo On Wed, Sep 3, 2008 at 7:10 AM, Stefano Palleschi <stefano.palleschi(a)okcom.it <mailto:stefano.palleschi@okcom.it>> wrote: Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 <http://81.29.176.57> | callfwd | sip:0666653218@81.29.176.57 <mailto:sip%3A0666653218@81.29.176.57> | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57 <http://81.29.176.57>> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi _______________________________________________ Users mailing list Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users -- Gustavo Mistrinelli

2 2

[Kamailio-Users] 302 redirect inside redirect
by Kent Pirlo 09 Sep '08

09 Sep '08

Ok, I am trying to get a redirect server working correctly, only cache is I need to hit another redirect server first, and grab info from the contact in the that 302 reply before processing and sending back a 300 to the original switch. call flow would be something like this: call comes into kamailio, 1) kamailio sends the invite to another proxy(1), 2) that proxy returns a 302,(if it times out it moves on to step 5) 3) kamailio sends an ACK back to that proxy(1), 4) kamailio looks at the contact header of the 302 reply, 5) depending on that info, I decide where to tell the originating switch where to go, 5) kamailio sends originating switch a 300 redirect with new contact header. 6) originating switch sends back ACK. I have tried doing this with a first step going to route(1) , which does a "t_on_reply" and "t_relay" to the second proxy. The onreply route, graps the contact saves it into variable and then sends to route(2), Route(2) looks at the variable, does some other lookups and sends t_reply("300","redirect"). The switch then sends the ACK, but kamailio says its not a local ACK and ignores it.. Then it keeps sending 300's back to the switch... Please help.. Thanks in Advance. kent

2 1

[Kamailio-Users] Documentation Site Down
by Jon Farmer 09 Sep '08

09 Sep '08

Hi When I go to http://kamailio.org/docs/ i get a 404 error. Any ideas anyone? Regards Jon

5 5

[Kamailio-Users] Forward call
by Thiago Rondon 09 Sep '08

09 Sep '08

Folks, My problem for forward call, that I need to change username/password and realm to do my forward. So, I try this: modparam("uac","from_restore_mode", "auto") modparam("uac","auth_realm_avp","$avp(s:uac_realm)") modparam("uac","auth_username_avp","$avp(s:uac_username)") modparam("uac","auth_password_avp","$avp(s:uac_password)" route { ... if(uri=~"^sip:0[0-9]{10}@") { xlog("L_INFO","uac_atual: $avp(s:uac_username) !!"); $avp(s:uac_username)="username"; $avp(s:uac_realm)="1.1.1.1"; $avp(s:uac_password)="password"; xlog("L_INFO","uac_trocado $avp(s:uac_username) !!"); route(3); exit; } .. } ... route[3] { t_on_failure("3"); # reset flag to mark no authentication yet performed resetflag(7); # forward to PSTN uac_replace_from("username","sip:username@1.1.1.1"); rewritehostport("1.1.1.1:5060"); xlog("L_INFO","rewritehost !!!!!!!!"); t_relay(); } .. So, when I do xlog, in the first entry, uac_username don't display, why ? And its strange, because in one supplier its ok, and in another its not run. But I don't undestand why uac_username is empty in the first xlog! Thanks, -Thiago Rondon

2 1

[Kamailio-Users] populate_leg_info
by David Villasmil 08 Sep '08

08 Sep '08

Anyone lnow what this means? dialog:populate_leg_info: bad sip message or missing Contact hdr ERROR:dialog:dlg_onreq: could not add further info to the dialog openser 1.3.2 thnks

2 1

[Kamailio-Users] New OpenXCAP release 1.0.0
by Adrian Georgescu 08 Sep '08

08 Sep '08

Hello, There is a new release of OpenXCAP server available. It contains many bug fixes and additions for important features like XML partial manipulation, xcap-caps, xcap-diff and RLS services that were not available at the initial release last year. With this features OpenXCAP fulfills now most if not all relevant requirements for an XCAP server implementation. A generic xcap client library and command line client is available in the clients directory, you can use it to manipulate documents on OpenXCAP or other XCAP server implementation. The server software is also operational at http://sip2sip.info so you can test against it by registering a SIP account. Changes in version 1.0.0 * Added RLS services (RFC 4662 and RFC 4826) * Added support for xcap-diff based on draft-ietf-simple-xcap-diff-09 * Added partial get/put/delete of elements in the XCAP documents * Added test suite for rls-services, resource-lists and partial updates * Many bug fixes from field experiences * Development status from beta to production * Switched to Python 2.5 * Improved documentation and testing suite * Changed license to GPL * Fixed MySQL operational error (2006) The software can be downloaded from: http://openxcap.org Many thanks to Denis Bilenko who did most of the works for this new release. Kind regards, Adrian Georgescu

1 0

[Kamailio-Users] CDRTool and Prepaid
by Adrian Georgescu 08 Sep '08

08 Sep '08

Hello Thanks for the feedback from everyone. You can try cdrtool version 6.6.0, I have added parallel prepaid call support. The algorithm is described in PREPAID.txt cdrtool (6.6.0) unstable; urgency=low * Allow simultaneous calls for prepaid accounts, see the Simultaneous prepaid calls section in doc/PREPAID.txt * Added pseudo code examples for call control engine * Apply the changes from setup/mysql/alter_tables.mysql * Added interface to kill ongoing sessions in Prepaid page Testing and feedback is useful and welcome. Regards, Adrian

1 0

[Kamailio-Users] call forward problem
by muhammad akl 06 Sep '08

06 Sep '08

hey all after making call forward feature using avpops module and its exported functions avp_db_load and avp_pushto i have a problem : assuming all calls to extension 12001 will be forwarded to extension 12008, in opensers's database i have added the following : username=12001 domain =mydomain attribute = callfwd value = sip:12008@mydomain type = 0 untill that every thing was fine and all calls comes to 12001 forward to 12008 without any problem , the problem comes from when i am using 12008 to call 12001 it calls itself ( mean 12008) so anyone can tell me what i have missed ? i have configured my openser.cfg like this in loaded modules section added loadmodules "avpops.so" > in module parameter section added : modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") in Invite message handler added : if(avp_db_load("$ru/username","$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); route(1); exit; } any suggestions ?

4 7

[Kamailio-Users] two proxies sharing the same database
by toly 05 Sep '08

05 Sep '08

Greetings, I have 2 proxies having the same database on backend. One is the primary, second is standby - OpenSER 1.2.1. That's how I implement DNS based failover. What I see, when all UA's registered on primary and I start standby, I's starting to cleanup location table according to it's own timer. So record is gone, when primary trying to updated the record on re-register it of course fails, as a result location table is wrong. What will be a solution? Thanks, Toly -- View this message in context: http://www.nabble.com/two-proxies-sharing-the-same-database-tp19335185p1933… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

[Kamailio-Users] How to solve early media during serial forking
by Iñaki Baz Castillo 05 Sep '08

05 Sep '08

Hi, AFAIK there is an issue in typical SIP/PSTN enviroments in the case of failover using different gateways serially. The main issue is that SIP doesn't define how a UAC should handle the case in which it receives various "183 Session Progress" from different early-dialogs, each of them containing early-media. It occurs to me that a SIP/PSTN gateway sometimes replies 183 with early media for 1 or 2 seconds, and inmediatelly it replies a 502. Well, this 502 could be caused by other node in the PSTN network, it's not necessarialy a problem fo the gateway [1]. In this case I do serial forking forwarding the request to other gateway. But the UAC already received early media from the first gateway, and most of the SIP phones just render the first early media they receive in case of multiple 183, and drop others. So basically the problem is that the UAC user doesn't listen the ringing coming from PSTN after the request has been forwarded to the second gateway. Is there something I could do at proxy level to avoid this issue? Sincerely I don't think so, since it makes no sense to disallow the first 183 (I cannot know that the gateway will later reply a 502). Thanks. [1] Anyway, *who* is creating the early media? Why the PSTN generates it if the request doesn't arrive to the called? "imagine there is no PSTN..." -- Iñaki Baz Castillo

1 0

[Kamailio-Users] custom routing
by toly 05 Sep '08

05 Sep '08

Greetings, Suppose I need to do custom routing for outbound calls from UA - softphone. Each user account may have the following info attached to it: 1:provider_a:2:provider_b:3:provider_c which means: forward call to provider_a, if failed forward call to provider_b if failed forward call to provider_c if failed send response back to UA I do database lookup based on from, and get above string. Suppose I wrote my own dispatcher module which initially gets set to above string: 1:provider_a:2:provider_b:3:provider_c and forwards to first provider, then on failure_route I call next_provider and so on. Will it work across multiple messages? Will it work correctly for each child - I mean joe@domain and paul@domain should be handled according their own providers? Any other suggestions? Thanks, Toly -- View this message in context: http://www.nabble.com/custom-routing-tp19322478p19322478.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

4 4

[Kamailio-Users] connection to remote mysql database
by Asim Riaz 05 Sep '08

05 Sep '08

I am using Kamailio 1.2 , in t openser.cfg I defined the the url for db, which is not local ("auth_db|permissions|uri_db|usr_loc", "db_url", "mysql://openser:db_password@db_ip_address/openser") but still getting the error message below when start openser; /sbin/openser[13064]: permissions - initializing /sbin/openser[13064]: WARNING: File not found: //etc/openser/permissions.allow /sbin/openser[13064]: Default allow file (//etc/openser/permissions.allow) not found => empty rule set /sbin/openser[13064]: WARNING: File not found: //etc/openser/permissions.deny /sbin/openser[13064]: Default deny file (//etc/openser/permissions.deny) not found => empty rule set /sbin/openser[13064]: new_connection: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) /sbin/openser[13064]: ERROR: group_db_bind: unable to connect to the database /sbin/openser[13064]: ERROR:group:mod_init: unable to open database connection /sbin/openser[13064]: init_mod(): Error while initializing module group /sbin/openser[13064]: INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill any idea ? thanks in advance Asim

2 2

[Kamailio-Users] Openser & Centile
by BERGANZ François 05 Sep '08

05 Sep '08

Hello, I have some problem with, openser & centile! When do a call from centile to openser, the call don't do "ringing", just "trying" and after, centile say "BYE" Have you an idea? Thank you

2 3

Re: [Kamailio-Users] Openser & Centile
by BERGANZ François 05 Sep '08

05 Sep '08

I could have a look, but Asterisk-Openseràno problems Centile-openseràdont work ! Cordialement BERGANZ François <http://www.acropolistelecom.net> http://www.acropolistelecom.net TEL/FAX : 33 (0) 1 70 72 50 15 De : Darren Sessions [mailto:dmsessions@gmail.com] Envoyé : vendredi 5 septembre 2008 15:26 À : BERGANZ François Objet : Re: [Kamailio-Users] Openser & Centile Would be helpful to see some of the SIP signaling. _____________________________ Darren Sessions dmsessions(a)gmail.com http://www.darrensessions.com _____________________________ On Sep 5, 2008, at 7:24 AM, BERGANZ François wrote: Hello, I have some problem with, openser & centile! When do a call from centile to openser, the call don't do "ringing", just "trying" and after, centile say "BYE" Have you an idea? Thank you _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

1 0

[Kamailio-Users] CDRTool and Prepaid
by Adrian Georgescu 05 Sep '08

05 Sep '08

The problem with concurrent prepaid calls and single balance is that you have to correlate between the call control and rating angine somehow so that all calls terminate when balnce becomes zero. The problem is a bit complex: Example: Balance = 10. A call starts to destination XXX, for the sake of example max session time = 2 minutes After one minute, you start second call to destination YYY which has a different price and your balance is not anymore 10 but depends on the duration of the first call which is in progress. What is the maximum session time for it given that the first call is already in progress? What should happen with the first call? I am looking for suggestions on implementing a proper algorithm to deal with this situation in the rating engine. If you have any I would be glad to hear it. Adrian

7 8

[Serusers] /tmp/ser_fifo disaapears after rebooting
by Joseph Myalla 05 Sep '08

05 Sep '08

I have revmoved the ser and reinstalled again, just after the installation ser works, i can see the PID, calls are made and text messages between computers, lokking in the tmp/ser_fifo is availabe, but it desappears after rebooting, what could be the reason On Wed, Sep 3, 2008 at 11:00 AM, <serusers-request(a)lists.iptel.org> wrote: > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. Error opening ser's fifo /tmp/ser_fifo (Joseph Myalla) > 2. Fwd: Error opening ser's fifo /tmp/ser_fifo sorry its ubuntu > 8.04LTs (Joseph Myalla) > 3. Re: Error opening ser's fifo /tmp/ser_fifo (Atle Samuelsen) > 4. SER 0.9.7-pre3 and timers (Steven C. Blair) > 5. how can i get the ip ? ( ?? ) > 6. how can i get the ip? ( ?? ) > 7. how can i get the IP? ( ?? ) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 2 Sep 2008 12:12:39 +0100 > From: "Joseph Myalla" <chanangae(a)gmail.com> > Subject: [Serusers] Error opening ser's fifo /tmp/ser_fifo > To: serusers(a)lists.iptel.org > Message-ID: > <531a7e5d0809020412n27b87d6g8f7346a569d98286(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I have downloades ser's debian packages and installed them in Ubuntu > Linux 6.0Ls Desktop edition > ser_0.9.7_i386.deb and the correspoding ser_mysql modules, > > trying commands serctrl ps or ul show i get an error "Error opening > ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I > get error SER/FIFO not accessible: 2 > > the are some instructions in the mailing list saying I should add > fifo_mode=0777 in the configuration file, I have done it but still the > errors exist. > > can any one help. > > > -- > Myalla J.C > Kibaha Education Centre > P.O Box 30054 Kibaha > Pwani > Tanzania > +255787680744 > > > ------------------------------ > > Message: 2 > Date: Tue, 2 Sep 2008 12:14:24 +0100 > From: "Joseph Myalla" <chanangae(a)gmail.com> > Subject: [Serusers] Fwd: Error opening ser's fifo /tmp/ser_fifo sorry > its ubuntu 8.04LTs > To: serusers(a)lists.iptel.org > Message-ID: > <531a7e5d0809020414q25b69c15ya029023b51ce367e(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I have downloades ser's debian packages and installed them in Ubuntu > Linux 8.0Ls Desktop edition > ser_0.9.7_i386.deb and the correspoding ser_mysql modules, > > trying commands serctrl ps or ul show i get an error "Error opening > ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I > get error SER/FIFO not accessible: 2 > > the are some instructions in the mailing list saying I should add > fifo_mode=0777 in the configuration file, I have done it but still the > errors exist. > > can any one help. > > > -- > Myalla J.C > Kibaha Education Centre > P.O Box 30054 Kibaha > Pwani > Tanzania > +255787680744 > > > > -- > Myalla J.C > Kibaha Education Centre > P.O Box 30054 Kibaha > Pwani > Tanzania > +255787680744 > > > ------------------------------ > > Message: 3 > Date: Tue, 2 Sep 2008 13:18:02 +0200 > From: Atle Samuelsen <clona(a)cyberhouse.no> > Subject: Re: [Serusers] Error opening ser's fifo /tmp/ser_fifo > To: Joseph Myalla <chanangae(a)gmail.com> > Cc: serusers(a)lists.iptel.org > Message-ID: <20080902111802.GD63335(a)mystic.cyberhouse.no> > Content-Type: text/plain; charset=us-ascii > > Hi Joseph, > > have you started ser? because, if ser is not running you will not be > able to get access to /tmp/ser_fifo. > > You can also check if it's there by doing a ls /tmp/ser_fifo :) > > > - Atle > > > * Joseph Myalla <chanangae(a)gmail.com> [080902 13:13]: >> I have downloades ser's debian packages and installed them in Ubuntu >> Linux 6.0Ls Desktop edition >> ser_0.9.7_i386.deb and the correspoding ser_mysql modules, >> >> trying commands serctrl ps or ul show i get an error "Error opening >> ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I >> get error SER/FIFO not accessible: 2 >> >> the are some instructions in the mailing list saying I should add >> fifo_mode=0777 in the configuration file, I have done it but still the >> errors exist. >> >> can any one help. >> >> >> -- >> Myalla J.C >> Kibaha Education Centre >> P.O Box 30054 Kibaha >> Pwani >> Tanzania >> +255787680744 >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > > > ------------------------------ > > Message: 4 > Date: Tue, 2 Sep 2008 11:14:46 -0400 > From: "Steven C. Blair" <blairs(a)isc.upenn.edu> > Subject: [Serusers] SER 0.9.7-pre3 and timers > To: "serusers(a)iptel.org" <serusers(a)iptel.org> > Message-ID: > <20C52563FC353043A5A9B80D847590E102D6824900(a)MAILBOX.exchange.upenn.edu> > > Content-Type: text/plain; charset="us-ascii" > > > I'd like to ask if anyone if running a service using SER 0.9.7-pre3 software with greater than 2000 registrations? If so are any of you experiencing issues which seem to be related to timers not firing when expected. We set the inv_timeout to a number which approximates a certain number of rings. This worked well when our subscriber base was small however as the number of subscribers, and associated calling activity, grew the timers seem to take longer to fire. We are planning on upgrading to a newer version of SER but I'd like to hear about other real world experiences. > > Thanks, Steve > > > > >

2 1

[Kamailio-Users] Mediaproxy vs. RTPproxy.
by Alex Balashov 05 Sep '08

05 Sep '08

This must be an incredibly dumb question, but why would anyone ever use mediaproxy over rtpproxy? If one is written in Python and one in C, it seems to me that the performance and scalability makes the choice of rtpproxy a no-brainer. What benefit is there to using mediaproxy? More features and capabilities, ease of configuration perhaps? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

2 6

[Kamailio-Users] FW: Call Hangup from Caller-End not working with TLS
by Ali Jawad 05 Sep '08

05 Sep '08

Hi All I am using using openser 1.3..if I make a call between two softphones on the same lan or a a pstn call to my mobile phone..and the called/receiver party does hang-up the call. It works fine in UDP mode and the call get's hang-up. However in TLS mode this does not work. Anything I might have missed here? Since both udp and tls use the same routes, and voice is fine and no one way audio ..etc. I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in the logs of openser I can't see any errors. However on the wire shark I can see icmp destination unreachable...port unreachable. I would have said it is a NAT issue. However it works for simple UDP. However I did notice the following in the logs Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000 Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794 (45900000) Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler: retransmission_handler : done Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request: Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE> Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri: <sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200 8F980980E97E42F8EC;nat=yes> Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0> Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2 Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type 232, <branch> = Shouldn't the transport=TLS ? Regards

4 15

[Serusers] how can i get the ip?
by 邱磊 05 Sep '08

05 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regards 2008-09-05 邱磊

1 0

[Kamailio-Users] Help with SerMyAdmin
by Tiago Rés 04 Sep '08

04 Sep '08

Hello all, I'm a new user in this list, but I have already seen some posts about serMyAdmin. I have openser 1.2.2 installed, running with mysql Version 14.12. I'm trying to use a web interface for openser, such as sermyadmin or openser administrator, but I'm unable to accomplish that. I have followed the instructions from sermyadmin website, but I had some errors: I couldn't add a user in mysql with that command, so I used something like: openserctl add admin openserrw admin@localhost The remaining steps went ok, but everytime I try to open http://(myserver):8080/serMyAdmin I get an error page from Tomcat. Any suggestion? Thanks in advance. Tiago

2 3

[Kamailio-Users] Fax Detection
by Abdul Hakeem 04 Sep '08

04 Sep '08

Hello, Is there a way to identify in incoming call as a fax (perhaps by detecting CNG tone ). Regards, Abdul Hakeem

2 1

[Kamailio-Users] ERROR:mysql:db_mysql_new_connection: driver error: Too many connections
by Samael - 04 Sep '08

04 Sep '08

Hi, We are deploying OpenSER in a large infrastructure together with Asterisk to support thousands of current transactions. We are using udp listener and mysql module in openser. We had to increase the children parameter of the openser.cfg to 200 because Openser couldn't handle the high amount of concurrent SIP REQUESTs from ASTERISK (Not only INVITES, but also a lot of SIP OPTIONs), that means that Openser couldn't handle more than 'children' REQUEST loosing the rest of the concurrent REQUEST. children=200 But then, the following error appeared in the openser: ERROR:mysql:db_mysql_new_connection: driver error: Too many connections This is because the Standard parameter of mysql "max_connections" is set to 100. In mysql CLI, we can see that every child process of openser has a mysql connection. (see in Mysql> SHOW PROCESSLIST) The recommended max_connections parameter is 500-1000 => we have a limit of 500-1000 concurrent SIP REQUEST to be processed by the openser. Why there is a process for every child? Why is neccesary a mysql connection for every child? Can the openser have a pool of mysql connection to be used by thousands of children? Regards

4 6

[Kamailio-Users] OpenSER reply 513 when register use FQDN
by Steven Wu 04 Sep '08

04 Sep '08

Hi, I just want use FQDN in my networking environment. My SJPhone use FQDN register to OpenSER 1.2.2 but was rejected by 513 Message to big. However it can be registed use the same profile just replace FQDN to IP address. The DNS query process worked fine and got the right IP address. My environment elments looks like: OpenSER 1.2.2 at host 192.168.1.120 / ipcsip.org DNS server at host 10.57.39.53 SJPhone at host 192.168.1.105 REGISTER package looks like: 192.168.1.105 -> 192.168.1.120 REGISTER sip:ipcsip.org:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.105;branch=z9hG4bKc0a801690000001f48bf8e4700006b9400000002;rport From: "unknown" <sip:1000@ipcsip.org:5062>;tag=328715c58a2 To: <sip:1000@ipcsip.org:5062> Contact: <sip:1000@192.168.1.105> Call-ID: 444CD1265B43407EBBF2B26BD4C01C6B0xc0a80169 CSeq: 1 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 192.168.1.120 -> 192.168.1.105 SIP/2.0 513 Message too big Via: SIP/2.0/UDP 192.168.1.105;branch=z9hG4bKc0a801690000001f48bf8e4700006b9400000002;rport=5060 From: "unknown" <sip:1000@ipcsip.org:5062>;tag=328715c58a2 To: <sip:1000@ipcsip.org:5062>;tag=2560d490c3265ff35995c6bbde62a7c3.b7d8 Call-ID: 444CD1265B43407EBBF2B26BD4C01C6B0xc0a80169 CSeq: 1 REGISTER Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 I got same result evenif enabled dns & rev_dns in openser.cfg (dns=yes rev_dns=yes). So, What wrong with me? Coudle some give me some instructions? Best regards, Steven Wu

4 7

[Kamailio-Users] Unconditional call forward
by Stefano Palleschi 04 Sep '08

04 Sep '08

Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 | callfwd | sip:0666653218@81.29.176.57 | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi

2 1

[Serusers] Which kind of SIP message can I use to exchange Data out of session?
by Peter Cai 04 Sep '08

04 Sep '08

Hi All, If a SIP client has to exchange some data out of a Session with the proxy server, such as get his "buddy list", what SIP Message can I use? I didn't find any thing suit for this purpose. Should I use something beside SIP such as HTTP.

3 3

[Serusers] cannot load module acc.so
by 静于 04 Sep '08

04 Sep '08

hello, I cannot load the acc.so module.The config file is below: # ------------- version 0.8.11-0 # ------------- Initial global variables debug=3 fork=yes log_stderror=no listen=10.1.2.3 listen=127.0.0.1 # hostname matching an alias will satisfy the condition uri==myself". alias=bigu.edu alias=10.1.2.3 # dns - Uses dns to check if it is necessary to add a "received=" field # to a via. Default is no. # rev_dns - Same as dns but use reverse DNS. dns=no rev_dns=no port=5060 children=4 # check_via - Turn on or off Via host checking when forwarding replies. # Default is no. arcane. looks for discrepancy between name and # ip address when forwarding replies. check_via=yes # syn_branch - Shall the server use stateful synonym branches? It is # faster but not reboot-safe. Default is yes. syn_branch=yes # memlog - Debugging level for final memory statistics report. Default # is L_DBG -- memory statistics are dumped only if debug is set high. memlog=3 # sip_warning - Should replies include extensive warnings? By default # yes, it is good for trouble-shooting. sip_warning=yes # fifo - FIFO special file pathname fifo="/tmp/ser_fifo" # server_signature - Should locally-generated messages include server's # signature? By default yes, it is good for trouble-shooting. server_signature=yes # reply_to_via - A hint to reply modules whether they should send reply # to IP advertised in Via. Turned off by default, which means that # replies are sent to IP address from which requests came. reply_to_via=no # user | uid - uid to be used by the server. 99 = nobody. uid="nobody" # group | gid - gid to be used by the server. 99 = nobody. gid="nobody" # mhomed -- enable calculation of outbound interface; useful on # multihomed servers. mhomed=0 # ------------- external module loading loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # ------------- tm parameters modparam("tm", "fr_timer", 12) modparam("tm", "fr_inv_timer", 24) # ------------- rr parameters # set ";lr" tag to ?�?;lr=true?�? modparam("rr", "enable_full_lr", 1) # ------------- accounting parameters modparam("acc", "log_missed_flag", 3) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) # ------------- usrloc parameters # 2 enables write-back to persistent mysql storage for speed # disable=0, write-through=1 modparam("usrloc", "db_mode", 2) # minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) # database location modparam("usrloc", "db_url", "sql://ser:>password>@localhost/ser") # ------------- auth parameters # database location modparam("auth_db", "db_url", "sql://ser:>password>@localhost/ser") # allows clear text passwords in the mysql database modparam("auth_db", "calculate_ha1", yes) # name of password column in mysql database modparam("auth_db", "password_column", "password") # ------------- routing logic route { # ------------- routine checks # stop forwarding at 10 hops to prevent infinite loops if (!mf_process_maxfwd_header("10")) { log(1, "LOG: Too many hops\n"); sl_send_reply("483", "Too many hops"); break; }; # prevents private ip space from being used if (search("^(Contact|m): .*(a)(192\.168\.|10\.|172\.16)")) { if (method=="REGISTER") { log(1, "LOG: Someone trying to register from private IP\n"); sl_send_reply("479", "Please don't use private IP addresses" ); break; }; }; # separate the destination r-uri from the set of proxies that must be traversed loose_route(); # if the host portion of the request uri is not local, send it directly # to route processing. if (!(uri==myself)) { route(2); break; }; # All REGISTER attempts are processed and must always be authenticated if (method=="REGISTER") { # make sure that users don't register infinite loops if (search("^(Contact|m): .*(a)(10\.1\.2\.3|(proxy\.)?bigu\.edu)")) { log(1, "LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; # challenge/response if (!www_authorize("bigu.edu", "subscriber")) { www_challenge("bigu.edu", "0"); break; }; # only registered users are allowed if (!is_user("replicator") & !check_to()) { log(1, "LOG: unregistered user registration attempt\n"); sl_send_reply("403", "Only registered users are allowed"); break; }; # it is an authenticated request, update Contact database now if (!save("location")) { sl_reply_error(); }; break; }; # process traffic local to BigU and the PSTN # Find the canonical username lookup("aliases"); # check domain again, if it is not still local after the alias # table lookup, just send it on its way. We do not authenticate # traffic we forward if (!(uri=~"^sip:(.+@)?(10\.1\.2\.3|(proxy\.)?bigu\.edu)([:;\?].*)?$")) { route(5); break; }; # now check for destinations through the gateway. 911 and 9911 # are always sent to the gateway. The assumption is that other all # numeric usernames between 5 and 20 digits are really pstn numbers # and so they are routed to the gateway if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") | (uri=~"sip:[0-9]{5,20}@.*") ) { route(3); break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail (ser->grp) group?) if (is_user_in("Request-URI", "voicemail")) { t_on_failure("4"); setflag(4); }; # handle local SIP destinations not found in usrloc db # mostly offline or non-existent users if (!lookup("location")) { route(4); break; }; # check whether some inventive user has uploaded gateway # contacts to usrloc to bypass authorization logic if (uri=~"@10\.1\.2\.5([;:].*)*" ) { log(1, "LOG: Gateway address in UsrLoc\n"); route(3); break; }; # this flag is used with the acc module to report missed calls # to syslog. setflag(3); # do it (words to live by) append_hf("P-hint: USRLOC\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } /* end of initial routing logic */ # ------------- process traffic leaving BigU for Internet route[2] { # outbound requests are allowed only for registered BigU users if (!(src_ip==10.1.2.3) & !(proxy_authorize("bigu.edu", "subscriber"))) { # ACK and CANCEL have no security mechanisms so they are just # noted if (method=="ACK" | method=="BYE") { log(1, "LOG: failed outbound authentication for ACK granted\n"); } else if (method=="CANCEL") { log(1, "LOG: failed outbound authentication for CANCEL granted\n"); } else { proxy_challenge("bigu.edu", "0"); break; }; }; # to maintain credibility of our proxy, we check >From in INVITEs if (!src_ip==10.1.2.3 & method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; append_hf("P-hint: OUTBOUND ON INTERNET\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process traffic leaving Internet for PSTN route[3] { # all calls through the gateway must be record routed to assure # acl acceptance on the gateway record_route(); # send out emergency calls to pstn gateway immediately if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") ) { rewritehostport("10.1.2.5:5060"); forward(uri:host, uri:port); break; }; # five digit numeric addresses are internal freebies sent to the pbx # without authentication if (uri=~"^sip:[0-9]{5}@(10.1.2.3|(proxy\,)?\.bigu\.edu)") { rewritehostport("10.1.2.5:5060"); forward(uri:host, uri:port); break; }; # all numeric addresses beginning with 9 go to the pbx on the way # to the PSTN # first the caller needs to be authenticated if (uri=~"^sip:9[0-9]*@(10.1.2.3|(proxy\.)?bigu\.edu)") { if (!(src_ip==10.1.2.3 | method==ACK | method=="CANCEL" | method=="BYE")) { if (!proxy_authorize("bigu.edu", "subscriber")) { proxy_challenge( "bigu.edu","0"); break; } else if (method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; }; if (method=="INVITE") { # if the r-uri begins 91, does the authenticated user have # permission for long distance if (uri=~"sip:91[0-9]*@.*") { if (!is_user_in("credentials", "ld")) { sl_send_reply("403", "Local calls only"); break; }; }; }; # authenticated and authorized, now accounting is set setflag(1); }; rewritehostport("10.1.2.5:5060"); append_hf("P-hint: GATEWAY\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process calls for users offline route[4] { if (!t_newtran()) { sl_reply_error(); }; if (!t_reply("404", "Not Found")) { sl_reply_error(); }; break; } # ------------- process aliased outbound traffic # inbound requests that have been aliased to a non-BigU domain # are not authenticated by BigU route[5] { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- CC-Diversion to voicemail failure_route[4] { append_branch("sip:80000@10.1.2.5"); append_urihf("CC-Diversion: ", "\r\n"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); t_relay(); } Can anyone help BR Ivy ___________________________________________________________ �Ż��䣬��䣡 http://cn.mail.yahoo.com/

1 0

Re: [Kamailio-Users] svn diff
by Vadim Berezniker 04 Sep '08

04 Sep '08

Oops hit Ctrl+Enter svn diff http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.2/ http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.3/ I've been trying to run this against the public SVN but it never completes.... I hate sourceforge. From: Vadim Berezniker Sent: Wednesday, September 03, 2008 1:57 PM To: 'users(a)lists.kamailio.org' Subject: svn diff Can someone with access to the dev svn run this for me?

3 3

[Kamailio-Users] Could you please help me how to integrate RLS AS and PS AS with SERIMS?
by mahesh.peddi＠info-spectrum.com 04 Sep '08

04 Sep '08

Hi, I am working on openser 1.3.2 as RLS application server and presence application server. I want to integrate this two application servers with SERIMS. Could you please help me how to integrate and make run this. To do this I have run SERIMS components on two servers? Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

1 0

[Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Kristian Kielhofner 03 Sep '08

03 Sep '08

Hello everyone, Feel free to set $OSSPROXY to whatever you like = Kamailio/OpenSIPS/OpenSER (for those of you who can't let go). $OSSPROXY is my favorite SIP toolkit. In addition to being an excellent proxy and registrar it serves as a platform to do SIP mangling and wrangling. With that in mind let me propose a crazy idea... How hard would it be to get $OSSPROXY to be able to parse what is commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has no business getting too involved in the SDP but I feel it makes sense in some instances (like this one). What if I want to make a routing decision based on an encapsulated ISUP parameter of some sort? Who knows what it could be!?!? Maybe there are some other uses but that makes the most sense and seems to be the most practical. It should be possible to develop an $OSSPROXY module to: - Use the SDP parser to find the ISUP SDP part: http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing - Pass that part into a "real" ISUP parser (if necessary). I couldn't seem to find the perfect library but perhaps something like this could work: http://www.openss7.org/isup.html - Make the ISUP parameters available via AVPs, pseudo variables, etc for use in the $OSSPROXY script. I don't really have a need for this but it seems like it could be cool. Am I nuts? -- Kristian Kielhofner http://blog.krisk.org

3 3

[Serusers] Is setdsturi supported in SER
by Steven C. Blair 03 Sep '08

03 Sep '08

I am wondering if setdsturi is supported in SER 0.9.7-pre3 Thanks

1 0

[Kamailio-Users] svn diff
by Vadim Berezniker 03 Sep '08

03 Sep '08

Can someone with access to the dev svn run this for me?

1 0

[Kamailio-Users] Unconditional call forward
by Stefano Palleschi 03 Sep '08

03 Sep '08

Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 | callfwd | sip:0666653218@81.29.176.57 | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi

2 1

[Kamailio-Users] VQ_MALLOC memory manager
by Henning Westerholt 03 Sep '08

03 Sep '08

Hi all, does somebody uses the VQ_MALLOC memory manager? I thought about removing this one, as it don't work correctly at the moment (it lacks a realloc implementation). This would be a good cleanup in the core. If somebody still depends on this memory manager, perhaps has also some fixes created for this, please reply to this thread. If nobody speaks up i'll remove the VQ_MALLOC implementation at the end of this month. Cheers, Henning

1 0

[Kamailio-Users] NAT transversal without mediaproxy or rtpproxy
by BERGANZ François 03 Sep '08

03 Sep '08

Hello, I am looking for to do nat transversal without mediaproxy or rtpproxy! I cant find any example without mediaproxy or rtpproxy. Have you an idea? I need to register my users, and contact them after Thank you

4 6

[Kamailio-Users] t_relay to two diffrent proxy
by Asim Riaz 03 Sep '08

03 Sep '08

Hi , I am trying to relay invite from Kamailio to one of two proxies, as long as it receive the reply from one it should not forward to second but incase if the proxy1 does not reply then relay to proxy2. here is my config route(1); if((src_ip != proxy1_ip_address) || (src_ip !=proxy2_ip_address) ) { if(!t_relay_to_udp("proxy1_ip_address", "5060")) || (!t_relay_to_udp("proxy2_ip_address","5060")) { end_media_session(); sl_reply_error(); } } I am also allowing only my proxies can see the location tables; f ((src_ip== Proxy1_ip_address) || (src_ip= Proxy2_ip_address)) { if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; for some reason incoming calls from my proxies are looping means Kamailio sending back to one of my proxyies, i dont know if there is somthing missing in my configuration or somthing else is causing this problem. Regards Asim Riaz

2 2

[Kamailio-Users] Kamailio dispatcher Module redirection
by Asim Riaz 03 Sep '08

03 Sep '08

Hi All I am trying to configure Openser as loadblancer for two more Kamailio. I wan loardbalancer not stay in the middle once routed to other kamilio servers. my first question is, dispathcer list always take IPs or I can configure proxy names as well. at the moment I am checking the to_uri and then redirecting the traffic to my one of Kamilio on my network, it seems to work fine with Linksys but not other phones e.g polycom xlite or even sjphone. if(to_uri=~"sip:.+@sip.mydomain.ie <sip%3A.%2B(a)sip.mydomain.ie>") { ds_select_domain("1", "4"); sl_send_reply("300","Redirect"); exit; } any suggestion. Thanks in advance Regards Asim Riaz

4 8

[Kamailio-Users] uac_redirect question
by Zahid Mehmood 03 Sep '08

03 Sep '08

Hi, Need a little help with get_redirects(), using openser 1.2. I am using the following in failure_route to capture contacts from a "multiple choices" reply from cisco media gateway. Following code sets the ruri to the last contact, however, the $ds show more information about other contacts. Is there a way to reset this so that only ruri information is used? Cisco response and xlog information is also given below. if(!get_redirects("*:*")) { xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci \n"); acc_db_request("480", "acc"); t_reply("480", "Temporarily Unavailable"); exit; } # get last URI from destination-set and set it as R-URI xlog("L_INFO", "Redirect from UAC intercepted 1 - M= $rm RURI=$ru D=$ds B=$bR \nF=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:tmp)/g"); $avp(s:tmp) = $ds; avp_subst("$avp(s:tmp)", "/.*(sip:.+@[^:;>]+).*$/\1/"); avp_pushto("$ru", "$avp(s:tmp)"); setflag(29); xlog("L_INFO", "Redirect from UAC intercepted 2 - M= $rm RURI=$ru D=$ds B=$bR \ntmp=$avp(s:tmp) \nF=$fu T=$tu IP=$si ID=$ci \n"); append_branch(); route(17); # process exit; } Thanks in advance for your advice. -- Zahid The response from gateway: U 2008/08/26 08:19:04.440411 10.10.0.32:5060 -> 10.10.0.98:5060 SIP/2.0 300 Multiple Choices. Via: SIP/2.0/UDP 10.10.0.98;branch=z9hG4bK901b.335cb695.0,SIP/2.0/UDP 10.10.12.140;branch=z9hG4bK612a4f20B9293665. From: "10521" <sip:10521@devproxy.myip.org>;tag=B395F9F2-4618AC3F. To: <sip:40001@devproxy.myip.org;user=phone>;tag=195535C4-11AC. Date: Tue, 26 Aug 2008 12:19:04 GMT. Call-ID: f7932e23-d89cd014-e3ea20a9(a)10.10.12.140. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 INVITE. Allow-Events: telephone-event. Diversion: <sip:40001@10.10.0.32>;reason=unconditional;counter=1. Contact: <sip:10512@10.10.0.98>,<sip:10512@10.10.0.32>. Content-Length: 0. xlog entries: Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 1 - M=INVITE RURI=sip:40001@10.10.0.32:5060;transport=udp D=Contact: sip:40001@10.10.0.32:5060;transport=udp, <sip:10512@10.10.0.32 >;q=0.01, <sip:10512@10.10.0.98>;q=0.01 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140 Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 2 - M=INVITE RURI=sip:10512@10.10.0.98 D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98 >;q=0.01 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01 tmp=sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140 Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted - M=INVITE RURI=sip:10512@10.10.0.98 D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98 >;q=0.01, sip:10512@10.10.0.98 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01, sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140

2 1

[Kamailio-Users] t_relay
by sabino frisardi 03 Sep '08

03 Sep '08

Hi all, I'm using openser 1.4. Can anyone tell me how can i call t_relay() from another module. Can anyone help me? any suggestion? Thanks for your help. Regards -- Sabino Frisardi

3 2

[Serusers] how can i get the IP?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? 2008-09-03 邱磊

1 0

[Serusers] how can i get the ip?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regard~ 2008-09-03 邱磊

1 0

[Serusers] how can i get the ip ?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regard~ 2008-09-03 邱磊

1 0

Re: [Kamailio-Users] CDRTool and Prepaid
by Matteo D'Amato 03 Sep '08

03 Sep '08

Thanks David for your answer, but can you explain how radius can do that since sending BYE messages is more at the signaling level. I dont quite understand how radius can even communicate at the SIP signaling level. From: David Villasmil Sent: Sunday, August 31, 2008 4:33 PM To: Matteo D'Amato Subject: Re: [Kamailio-Users] CDRTool and Prepaid You could use radius. On Sun, Aug 31, 2008 at 10:09 PM, Matteo D'Amato > wrote: Hello, In "/doc/PREPAID.txt" it talks about "an external call control module (not provided by CDRTool) that maintains call status and terminates calls by sending BYEs to both SIP end-points" What is this 3-party module and where can I get it? --Matteo _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

9 19

[Serusers] SER 0.9.7-pre3 and timers
by Steven C. Blair 02 Sep '08

02 Sep '08

I'd like to ask if anyone if running a service using SER 0.9.7-pre3 software with greater than 2000 registrations? If so are any of you experiencing issues which seem to be related to timers not firing when expected. We set the inv_timeout to a number which approximates a certain number of rings. This worked well when our subscriber base was small however as the number of subscribers, and associated calling activity, grew the timers seem to take longer to fire. We are planning on upgrading to a newer version of SER but I'd like to hear about other real world experiences. Thanks, Steve

1 0

[Serusers] Error opening ser's fifo /tmp/ser_fifo
by Joseph Myalla 02 Sep '08

02 Sep '08

I have downloades ser's debian packages and installed them in Ubuntu Linux 6.0Ls Desktop edition ser_0.9.7_i386.deb and the correspoding ser_mysql modules, trying commands serctrl ps or ul show i get an error "Error opening ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I get error SER/FIFO not accessible: 2 the are some instructions in the mailing list saying I should add fifo_mode=0777 in the configuration file, I have done it but still the errors exist. can any one help. -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744

2 1

[Serusers] Fwd: Error opening ser's fifo /tmp/ser_fifo sorry its ubuntu 8.04LTs
by Joseph Myalla 02 Sep '08

02 Sep '08

I have downloades ser's debian packages and installed them in Ubuntu Linux 8.0Ls Desktop edition ser_0.9.7_i386.deb and the correspoding ser_mysql modules, trying commands serctrl ps or ul show i get an error "Error opening ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I get error SER/FIFO not accessible: 2 the are some instructions in the mailing list saying I should add fifo_mode=0777 in the configuration file, I have done it but still the errors exist. can any one help. -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744 -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744

1 0

[Serusers] Local number portability
by inge 02 Sep '08

02 Sep '08

Hi all, Is someone know if it's possible to use SER for prefix calls for LNP ? Then the call will be sent to AS5350 and the prefix must define the carrier for out ported number. I search information on this topic but I find nothing about SIP and LNP. Thanks for our informations. Regards

1 0

[Kamailio-Users] route number
by BERGANZ François 02 Sep '08

02 Sep '08

Hello, I need to have a route number like route(1000) But after some tests, I cant have more than route(39) How can I have more? Thank you

2 1

[Kamailio-Users] How to config OpenSER as IMS server
by Steven Wu 01 Sep '08

01 Sep '08

Hi everybody, Currently we are working on an IMS project. This project need a IMS server. Finally we choiced OpenSER as the IMS server. But have no experience how to use OpenSER as a IMS server. Could someone tell me how to config it? Best regards, Steven Wu

3 2

[Kamailio-Users] Mediaproxy 2.0 - Dispatcher vs Relay
by Nuno Marques 01 Sep '08

01 Sep '08

Hi, I'm trying to understand better how mediaproxy works. Just read the mediaproxy-ng.org page... i understood the necessity of using mediaproxy for NAT traversal, but i have one doubt... - What's the difference between the dispatcher and the relay? Can someone explain me the concept or point me to some literature so i can clear my mind :) Thanks in advance, Nuno

3 2

[Kamailio-Users] kamailio 1.5 and beyond
by Henning Westerholt 01 Sep '08

01 Sep '08

Hi all, i want to share some thoughts where i think the priorities for the next major release(s) of Kamailio should be. This is the continuation of the work i (and many others) have done for the previous releases, but i thought it makes sense to summarize it a bit on a higher (abstraction) level. 1. Refactoring and cleanup We still have quite a lot of old code in our repository, which is no longer working or obsolete. Some implementations uses way to much gotos, or macros that are several pages long. This makes important parts of the core and modules harder to understand and maintain then necessary. Especially in the area of utility functions there is still to much duplicated code. proposed actions: - identification and removal of no longer working code in the core and modules - identification and unification of redundant (duplicated) code in core and modules to one implementation - identification and removal of obselete code parts in core and modules. - refactoring of code to use less gotos and macros 2. User and developer documentation With regards to user documentation quite a lot of work has been done in the past releases. But with the developer documentation, especially in certain modules, there is still plenty of content missing. As the involvement of individuals and companies can change, documentation is necessary to ensure a continued maintenance of the code. proposed actions: - conversion of already existing documentation to doxygen format - add a short doxygen module definition to every module, explaining the purpose - add a short doxygen file definition to every file, explaining the purpose - extend doxygen docs in important areas like the core and modules like TM - autogeneration of database related module documentation from the XML scheme definitions 3. Consolidation in core and modules At the moment we've 85 modules in our repository. Some modules implement only one or two functions, which makes no real sense with regards to the maintainance overhead that comes with it. 24 of the modules are using database access and 7 implement database interfaces. There is no clear boundary in some areas of the core and modules, this hinders the further development of the server. proposed actions: - integrate several modules that share responsibilities and not implement that much functionality on their own into one - autogeneration of module interface and database infrastructure code from the XML database scheme definitions - identify obselete module functions that can be done easily with pseudo-variables now. Remove these functions. - move certain core parts that do not need to be there into modules, making the core smaller and more flexible - identify core APIs which are used from modules, evaluate eventual generalisations to make the modules less dependent from the actual core implementation 4. Quality assurance The complexity of several modules, and the special dependencies that some of them carry increase the work to test the server and reproduce bugs. Without proper test coverage maintenance work is slowed down and flexibility is lost. All "easy" bugs that get caught automatically will not reach the user, increasing the quality. proposed actions: - extend available test suite to cover more of core and modules - continue maintenance of available infrastructure for build tests (build bot, debian package autobuilder), evaluate extensions to increase coverage All this work will be done as in the past in an open and accountable way on the developer list. For bigger changes a request for comment will be posted in advance on this list. If existing functionality is deprecated, then this RFC will be also posted to the user list. If some functionality is moved or changed than this will be documented in the wiki (porting docs) as usual. I understand some changes will be probably add work on your side, in porting, configuration adjustments and the like. But i think this is necessary to ensure the further development of the Kamailio project, and to be able to deal with new challenges that will come up in the future. Any input and help with this regards to this individual tasks is of course much appreciated. In the end the success of the project depends in the future (like it did in the past) on your support. Without your involvement in contributions, discussions and bug reports i would be only to a much lesser extend involved in this project. Thank you, Henning Westerholt -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

2 2

[Kamailio-Users] About Changing 401 Message
by Ali Jawad 31 Aug '08

31 Aug '08

Hi All When I do logon with a wrong username/pwd I get the following error 401 UNAUTHORIZED. If I do want to change that error "for whatever stupid reason :-)" I know I could change that message using the header files and recompile. However I only know where to change the UNAUTHORIZED keyword, what about the 401 number...apart from that could I change that message using ss_reply ? Thanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

4 6

Re: [Kamailio-Users] CDRTool and Prepaid
by Matteo D'Amato 31 Aug '08

31 Aug '08

Hello, In "/doc/PREPAID.txt" it talks about "an external call control module (not provided by CDRTool) that maintains call status and terminates calls by sending BYEs to both SIP end-points" What is this 3-party module and where can I get it? --Matteo

1 0

Re: [Kamailio-Users] [Kamailio-Business] project foundation
by Daniel-Constantin Mierla 31 Aug '08

31 Aug '08

Hello, On 08/13/08 08:37, Max van Biezen wrote: > Hi, > > >> a quick operational expenses estimation for the first year brought up >> the figure of ~3000Euro, that includes the expenses for creating the >> foundation, two hosted servers (one main, one backup), domain names and >> trademark registrations. The costs regarding TM registration expenses >> (US&EU) are not yet very clear to me, maybe someone can shed some light >> if he/she did it recently. >> > > I can only give an indication on EU trademark. That is 850,- one time > fee. This fee is due regardless the fact some company will complain or > not. As happened to me. > > The process can be done by yourself no lawyers needed. However there are > lawyers outhere that simply complain to everything in the hope some > company which they claim to represent willl pay for it. Anyway worst > case scenario would then be to drop the registration whihc means the 850 > is gone > > Bottom line just try it yourself. It is a long process before > registration is final (more than 6 months) > thank you for the details. Cheers, Daniel > Max van Biezen > > > -- Daniel-Constantin Mierla http://www.asipto.com

1 1

[Kamailio-Users] issue when adding subscriber to mysql
by Alex R.S.M 30 Aug '08

30 Aug '08

Hi, Ijust added mysql module to my openser 1.2 and tried to test it by adding a couple of SIP users to the database. I can verified there is only one default sub in the database: mysql> select count(*) from subscriber; +----------+ | count(*) | +----------+ | 1 | +----------+ 1 row in set (0.00 sec) mysql> show tables; I can add the first subscriber: # openserctl add 100 100 100@localhost -e database engine 'MYSQL' loaded -e Control engine 'FIFO' loaded -e is_user: user counter=0 -e check_db_alias: alias counter=0 -e mysql_query -e insert -e 0 -e new user '100' added Howver, When I try to add the second subscriber I am getting the following error: # openserctl add 101 101 101@localhost -e database engine 'MYSQL' loaded -e Control engine 'FIFO' loaded -e is_user: user counter=0 -e check_db_alias: alias counter=0 ERROR 1062 (23000) at line 1: Duplicate entry '' for key 3 -e mysql_query -e insert -e 0 -e \E[37;31mERROR: introducing the new user '101' to the database failed root@bling1:/etc/openser# What might be the problem? Thanks, Alex _________________________________________________________________ See what people are saying about Windows Live. Check out featured posts. http://www.windowslive.com/connect?ocid=TXT_TAGLM_WL_connect2_082008

2 1

[Serusers] SER Version
by Arun Kumar 30 Aug '08

30 Aug '08

Hi All, I'm planning to roll out a new production SER for my company needs suggestion on SER version. At present I run 0.9.6 and having some issue like waiting transaction, extra connection to mysql, so which version should I roll out. thanks arun

4 9

[Kamailio-Users] openser children parameter
by toly 29 Aug '08

29 Aug '08

Greetings, What would be value of children parameter for few thousands user? Thanks, Toly -- View this message in context: http://www.nabble.com/openser-children-parameter-tp19192868p19192868.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 4

[Serusers] AVP and MySQL question
by Steven C. Blair 29 Aug '08

29 Aug '08

I have a SER 0.9.7-pre3 system. I use avopops to load flags from mysql to implement call andling features. I'd like to read a list of values for a single attribute from the database (and I can today) however the order in which the results are return is not consistent. I read that MySQL4 has an issue where results of a query are returned in a nondeterministic fashion. I think I can get my desired result by issuing my own query but I do not see an avpops function which allows for a user specified query. Does such a function exist? If so what is it's name? Thanks

2 1

[Kamailio-Users] Question regarding outbound configuration.
by Siddhardha Garige 29 Aug '08

29 Aug '08

Hello all, I am working on an outbound proxy configuration and here is my setup. UA ------> Outbound ------->firewall(NAT) -------->Proxy. Both UA and outbound are behind the NAT. I did some RFC study on outbound configuration for routing calls both inbound and outbound through outbound proxy. According to RFC 3327 Extension Header field for Registering Non-Adjacent contacts, PATH variable has to be introduced into REGISTRATIONS. Here are few questions and any help with answers is greatly appreciated. 1. Default openser.cfg will route request to destination if (URI != myself) and appends p-hint:outbound. What exactly is the significance of P-hint:outbound? http://tools.ietf.org/html/draft-ietf-sip-outbound-15 says ob parameter has to be added to PATH filed or Contact Field to retain path. P-hint got to do any thing regarding this? 2. I enabled path.so module. But openser does not at PATH filed to outgoing requests. Can someone direct me to a config example? Thanks in advance, Sid

1 1

[Kamailio-Users] event notifications and user aliases
by Alexander 'Leo' Bergolth 28 Aug '08

28 Aug '08

Hi! I'd like to implement P2P presence and extension monitoring for my SIP network. Our users register by using a numeric extension (e.g. sip:226050@wu-wien.ac.at), but most of the time, they are contacted using a user alias that looks like their email-address (e.g. sip:bergolth@wu-wien.ac.at). So the addresses stored in their buddy-lists won't be the "real"-SIP-URI of the users. Will there be a problem if I translate those aliases on subscription, so that a user may also subscribe for the presence of a user alias? If I rewrite SUBSCRIBE requests to use the "real" SIP-URI on subscription, the following NOTIFY messages will contain the wrong URI in the From-header and in the pidf body. But maybe this won't be a problem, since the watcher can assign it to the correct username because the NOTIFY belongs to the same dialog as the SUBSCRIBE? Will the user agents be able to correctly assign those notifications to their contacts? Any experiences with this? Cheers, --leo -- e-mail ::: Leo.Bergolth (at) wu-wien.ac.at fax ::: +43-1-31336-906050 location ::: Computer Center | Vienna University of Economics | Austria

4 4

[Serusers] Call forwarding: Diversion header
by caio 28 Aug '08

28 Aug '08

Hi, Does anyone know how to implement draft-levy diversion header on for example bilnd call forwarding? I have this scenario: xlite ------ SER-0.9.7 ------ PSTN-GW ------ PSTN Case: Call from xlite-579950 to 579917 (both registered at SER), 579917 has fwdblind enabled to pstn-501866. where SER modifies INVITE when detect that user 579917 has fwdblind feature enabled. The call must be forwarded to the pstn number. but the PSTN-GW returns to SER a "302: Moved Temporarily" with a confused diversion header.. Here a few sip messages: = SER -----> PSTN-GW U +0.000824 200.2.127.187:5060 -> 200.2.127.153:5060 INVITE sip:501866@200.2.127.153:5060 SIP/2.0. Record-Route: <sip:200.2.127.187;ftag=2d0ad32d;lr=on>. Via: SIP/2.0/UDP 200.2.127.187;branch=z9hG4bK396d.c60b3015.0. Via: SIP/2.0/UDP 192.168.132.22:35318;branch=z9hG4bK-d8754z-c16e4506aa72e334-1---d8754z-;rport=35318. Max-Forwards: 16. Contact: <sip:579950@192.168.132.22:35318>. To: "579917"<sip:579917@wsa.lab>. From: "579950"<sip:579950@wsa.lab>;tag=2d0ad32d. Call-ID: MzI2MjQ4ZTMyMGQzOWRlYzNkMWFkZTVmNGY3MzIzNDE.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1100l stamp 47546. Content-Length: 385. = PSTN-GW ----> SER U +0.051198 200.2.127.153:5060 -> 200.2.127.187:5060 SIP/2.0 302 Moved Temporarily. Via: SIP/2.0/UDP 200.2.127.187;branch=z9hG4bK396d.c60b3015.0. Via: SIP/2.0/UDP 192.168.132.22:35318;branch=z9hG4bK-d8754z-c16e4506aa72e334-1---d8754z-;rport=35318. To: "579917" <sip:579917@wsa.lab>;tag=121881197764d164f2-9431-47bb-be95-8af6684ebc04. From: "579950" <sip:579950@wsa.lab>;tag=2d0ad32d. CSeq: 2 INVITE. Call-ID: MzI2MjQ4ZTMyMGQzOWRlYzNkMWFkZTVmNGY3MzIzNDE.. Contact: <sip:579917@200.2.127.187:5060>. Diversion: <sip:501866@200.2.127.153:5060>;reason=unknown. Server: Epygi Quadro SIP User Agent/v4.1.23 (SOHOGW). Content-Length: 0. What is wrong? How the flow must be to satisfies diversion of levy draft? Any comments are welcome...i'm confused if SER must modify INVITE or "To" header or if a diversion must be set (add_diversion of diversion.so module) when SER send the first invite to PSTN-GW... Regards, -- caio

1 0

[Kamailio-Users] Why dialplan and drouting modules were removed from openser-1.3.3?
by Peter Lemenkov 28 Aug '08

28 Aug '08

Hello All! Subj. -- With best regards!

3 4

[Kamailio-Users] How to test RLS and trace messages?
by mahesh.peddi＠info-spectrum.com 28 Aug '08

28 Aug '08

Hi, I am working on RLS Application server and integrated with SER IMS, i made changes in the configuration file with the help of Readme file. Its working fine. But how to confirm and test RLS is working. And how to trace the messages? Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

1 0

[Kamailio-Users] Using pua_bla for BLF
by catalina oancea 28 Aug '08

28 Aug '08

Hello I am trying to make some phones work with BLF using Kamailio and I found an older messge ( http://lists.kamailio.net/pipermail/users/2007-October/013700.html) saying that I could do this using BLA. I have a Thomson ST2030 as the message says, but it doesn't work with presence and pua_bla. I get the error: ERROR:presence:handle_subscribe: Missing or unsupported event header field value ERROR:presence:handle_subscribe: event= dialog I undestood from that message that the event is supposed to be 'dialog', so what's the problem then? Thanks a lot Catalina

3 6

[Kamailio-Users] Fw: regarding 3rd party register request.
by srinivasa.reddy＠info-spectrum.com 28 Aug '08

28 Aug '08

Hi all, This is Srinivas, i am working on OpenIMScore and Openser( as Application Server) . from few days i stuck at third party regaester request(that is interface between SCSCF and Presence ). You said that, the registration-event package is not enabled at the IMS Core side that mean in the scscf.cfg is it? If it is could you please provide where and what i have to change and i am attaching the scscf.cfg . And regarding 2nd point Also the Openser presence server, as much I know, will be for the "presence" package. Not for the registration-event "reg" package That means openser does not support reg package or ? WITH BEST REGARDS, SRINIVAS. -------------------------------------------------------------------------------------------------------------------------------- Hi, >From your mail below, I could infer only that, the registration-event package is not enabled at he IMS Core side. Make sure, that you have done the necessary settings for that. Also the Openser presence server, as much I know, will be for the "presence" package. Not for the registration-event "reg" package. Thanks, Noble Antony T. On Wed, Aug 27, 2008 at 3:54 PM, <mahesh.peddi(a)info-spectrum.com> wrote: Hi, I have one issue regarding openser could you kindly reply if you know . I am implimented openser1.2.0 and 1.3.2 as Presence server. And i am connected it with IMS core .And for testing i was used UCTIMS client during the register with uctims client actually i need to get 4 messages. 1.Third party register request is going to presence server from scscf. 2.Notify messages coming from presence server to uctclient. 3.subscribe message must take route through the presence server. 4.publish message must take route through the presence server. But at present i am getting 3 and 4 messges corectly i am not getting 1 and 2. So please tell me what are the modifications or changes i need to do in any xml files or cfg files. Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764 Regards, Srinivasa Reddy Kadiri Software Developer Infospectrum India Pvt. Ltd. Cell: +91 9921710062 Email: srinivasa.reddy(a)info-spectrum.com IP-Phone Ext. - 759

1 0

Re: [Kamailio-Users] How to test and trace the RLS messages?
by Daniel-Constantin Mierla 28 Aug '08

28 Aug '08

Hello, please keep cc-ing to mailing lists, there are people that can answer you quicker and better. As a matter of fact, although I have studied a bit IMS docs, that happened long time ago and don't know how those messages should be routed in IMS and where the mistake can be in your routing logic. Cheers, Daniel On 08/27/08 13:24, mahesh.peddi(a)info-spectrum.com wrote: > Hi, > > I have one issue regarding openser could you kindly reply if you know . > > I am implimented openser1.2.0 and 1.3.2 as Presence server. And i am > connected it with IMS core .And for testing i was used UCTIMS client > during the register with uctims client > actually i need to get 4 messages. > 1.Third party register request is going to presence server from scscf. > 2.Notify messages coming from presence server to uctclient. > 3.subscribe message must take route through the presence server. > 4.publish message must take route through the presence server. > But at present i am getting 3 and 4 messges corectly i am not getting > 1 and 2. > So please tell me what are the modifications or changes i need to do > in any > xml files or cfg files. > > Regards, > Mahesh Peddi > Infospectrum India Pvt. Ltd. > Cell: +91 9765775176 > IP-Phone Ext. - 764 > > ----- Original Message ----- From: "Daniel-Constantin Mierla" > <miconda(a)gmail.com> > To: <mahesh.peddi(a)info-spectrum.com> > Cc: "userlist" <users(a)lists.kamailio.org> > Sent: Wednesday, August 20, 2008 10:44 PM > Subject: Re: [Kamailio-Users] How to test and trace the RLS messages? > > >> Hello, >> >> On 08/20/08 09:44, mahesh.peddi(a)info-spectrum.com wrote: >>> Hi All, >>> I am working on openser 1.3.2 version with RLS as >>> "integrated_xcap_server". Its running without erros. >>> But my query is how to test, And please send me call flow between >>> IMS components and RLS server. >> perhaps you can find some hints in IMS specifications. You need in >> first place a sip phone with rls support, afaik, there are not that >> many. >> >> Cheers, >> Daniel >> >>> Regards, >>> Mahesh Peddi >>> Infospectrum India Pvt. Ltd. >>> Cell: +91 9765775176 >>> IP-Phone Ext. - 764 >>> ------------------------------------------------------------------------ >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] openser v1.3.3. released
by Daniel-Constantin Mierla 27 Aug '08

27 Aug '08

Hi everybody, OPENSER v1.3.3 is out - it is a minor release of the branch 1.3, that includes the fixes since v1.3.2 committed so far- configuration file and database compatibility is preserved. OPENSER 1.3.3 is based on the latest version of branch 1.3, therefore those running 1.3.2 or earlier version in this branch are advised to upgrade. Source tarballs are available at: http://www.kamailio.org/pub/openser/1.3.3/src/ Detailed changelog: http://www.kamailio.org/pub/openser/1.3.3/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.3 openser Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.3.3/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Serusers] Radius Auth : 407 proxy auth and Get-User-Attr - crash
by Samuel Muller 27 Aug '08

27 Aug '08

Hello all, i'm using SER 2.0 with Radius accounting and authentication - based on x64 ubuntu machine. i've a problem with the Radius auth, and only for the proxy auth mechanism (method INVITE) : depending of the topology of the UAC, it crashes the SER or a there's a 407 proxy auh required. when the UAC is behind a simple NAT, i get the 407, with "preauth: credentials with realm 'sip.720.fr' not found" - i tried many configs, no way it doesn't work. but the REGISTER is ok - it's the same Radius request ? when the UAC is behind NAT but in a VPN (our customers), proxy auth is successfull even if the preauth don't find the credentials with the realm (why? it does not differ from another uac without VPN ???), but when SER requests the Radius to get the user attributes, it crashes ! in each case, there's no pbm with REGISTER requests. Radius user sample : 1010(a)sip.720.fr User-Password = "pwd" User-Name = "1010", SER-UID = "1010(a)sip.720.fr", SER-DID = "1", SER-Digest-Username = "1010", SER-Digest-Realm = "sip.720.fr", SER-Uri-User = "1010(a)sip.720.fr" Does anyone have any idea, or met already this problem ? What I have to verify ? Thanks for help, Samuel MULLER

1 0

[Kamailio-Users] multiple extensions/domains
by voipmo 27 Aug '08

27 Aug '08

I have been running asterisk servers for a few years now to host multiple organizations. I have been working on setting up OpenSER so that we can have failover and load balancing. My question is how do I keep my different organizations separate? I currently have the following registrations: 1001(a)test.example.com 1002(a)test.example.com 1001(a)another.example.com The problem is that if 1002(a)test.example.com dials 1001 it rings both domains 1001 phones. Am I going about this the right way? Is there a better method? How might I prevent calls between domains? -- View this message in context: http://www.nabble.com/multiple-extensions-domains-tp19173400p19173400.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 2

[Serusers] Help with Ser and LCR
by Jai Rangi 26 Aug '08

26 Aug '08

All, I need little help with LCR module in ser, Say these are the entries on my LCR tables, Prefix, | From_URI | Grp_id | Priority "13612499080" ;"%"; "46"; "1" "1" ;"%"; "1"; "1" "1" ;"%" "2" "2" Now any Say if someone calls 19495556666, Then ser sends the call to Server in Grp_id 1, if the grp_id=1 server does not respond, then SER sends the call to Server with grp_Id 2. Perfect, that exactly what I need. Now if some calls 13612499080", then SER will send the call to server with grp_id 46. If the call failes it should not send the call to server grp_id 1 or two. Right? In my case it is sending the call to server grp_id 1 after the call fails on server grp_id 46. How can I stop that. I want the calls for 13612499080 forwarded to only server grp_id 46 and is that fails then just terminate the call. I will appreciate if someone can provide any help or hint. Thank you, -JP

1 1

[Kamailio-Users] AVP to store user contact information
by Asim Riaz 26 Aug '08

26 Aug '08

Hi All, I am wondering if there is a way to store the user contact info into avp, relay it to existing Regitrar and store the original contact info into location through a function(which i dont know) once i received successful reply from my sip registrar. Thanks in advance Regards Asim Riaz

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[Kamailio-Users] FW: | SMS FALAS | <--- did anyone else on this list get this message?
by Vadim Berezniker 26 Aug '08

26 Aug '08

Sorry this is offtopic but.. Did anyone else get this email? I'm just wondering if they scraped emails from this mailing list. -----Original Message----- From: KLIENT I NDERUAR [mailto:info@hi5.com] Sent: Tuesday, August 26, 2008 11:38 AM To: Vadim Berezniker Subject: | SMS FALAS | Klient i/e Nderuar Nese Deshironi te Ndegjoni Muzika ma te rejat ather vizitone Kosova-SMS dhe do te bindeni me muziken qe ofrojm ne. Gjithashtu per Vizitoret e Kosova-SMS ka Startu me ofert te RE. SMS FALAS MUNDSUAR NGA Kosova-SMS 60 SMS NE DIT NGA NR I YT KLIKO PER ME SHUM. --> http://www.kosova-sms.tk

1 0

[Serusers] Notice call ending
by inge 26 Aug '08

26 Aug '08

Hi all, I look for a workarround to notice the caller that the call will be hang up, by generating a beep. Is someone can help me because I don't find anything about this subject ? Thanks. Regards, Adrien.

1 0

[Kamailio-Users] Having Problem routing an ACK
by Stagg Shelton 26 Aug '08

26 Aug '08

Hello, I am having an issue routing an ACK back from a new upstream carrier. The basic situation is my openser 1.3.2 receives the INVITE. I do an alias_db_lookup, set some accounting flags, and send the call to t_relay The call makes it to the PBX, as expected, and the PBX sends a 100 trying, 180 ringing. The 180 ringing is sent to the carrier. When the incoming call is answered the PBX sends a 200OK to openser, and that 200OK is sent to the carrier. When the carrier sends the ACK back it makes it to OpenSER, but is not sent to the PBX. The PBX will send several more 200 OK attempts, but each ACK from the carrier stops at openser. From what I can tell from the wireshark packet captures, the Request- Line header in the ACK from the carrier is sip:TELNUMBER@MY_OPENSER_IPADDR I have compared these packets to similar traces from other carriers and the Request-Line header from the other carriers is sip:TELNUMBER@MY_PBX_IPADDR This new carrier is telling me that they are not doing anything wrong with the ACK. RFC3261 seems specific about the construction of the ACK, but at the same time i am getting lost going to all of the other sections trying to identify details. Basically I just need to get the ACK to the PBX. What other information can I provide with this problem. Thanks Stagg Shelton

4 20

[Serusers] ser 0.10 doucmentation
by Steven C. Blair 26 Aug '08

26 Aug '08

I was wondering if anyone has documentation on the new syntax used in SER 0.10.x. I see the following in the ser-advanced.cfg example file but I do not see these parameters documented anywhere. Thanks,Steve if ($t.fr_inv_timer) { if ($t.fr_timer) { t_set_fr("$t.fr_inv_timer", "$t.fr_timer"); } else { t_set_fr("$t.fr_inv_timer"); } } Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

6 32

[Kamailio-Users] CDRTool not applying rates
by Brian Del Shasta 25 Aug '08

25 Aug '08

I believe my rating tables are configured correctly now thanks to David Villasmil who sent me examples from his working setup. However, I am still unable to see any rates. The rating engine seems to be started and I have no problem identifying and matching calls based on specific criteria. But I still see no rates. I'm wondering if anyone knows the best place to troubleshoot the rating engine would be or if anyone has any other ideas? Thank you in advance, and thanks again David for your help!

3 7

Re: [Kamailio-Users] presence fail
by Ricky Gutierrez 25 Aug '08

25 Aug '08

in this guide, he doesn't have the module presence_xml, but he has the parameter modparam (" presence ", " force_active ", 1) http://www.kamailio.net/dokuwiki/doku.php/presence:configuration-file it is strange, I believe to be seen in the wiki a same example for openser 1.3.2 greetingss rickygm

2 1

Re: [Serusers] call forward no answer (solved)
by caio 25 Aug '08

25 Aug '08

append_branch() is missed at ser examples... On Fri, Aug 15, 2008 at 4:22 PM, caio <elcaio(a)gmail.com> wrote: > Hi all, > I can't make call forwarding on no answer works on my ser-0.9.7 installation. > Here you can see a snippet of log, where the call falls on > failure_route and then to route_1 where t_relay is called. > > Log at: http://rafb.net/p/xwTmn195.html > > The call is from sip:579951 to sip:579915(who has enabled the feature > fwdnoanswer to sip:579916. > But I do not know why the INVITE after the append_branch() is still to > R-URI: sip:579915@.... > > After the relay then the call terminates with: > > Aug 15 16:13:25 [/usr/sbin/ser] SIP Reply (status):_ > Aug 15 16:13:25 [/usr/sbin/ser] version: <SIP/2.0>_ > Aug 15 16:13:25 [/usr/sbin/ser] status: <487>_ > Aug 15 16:13:25 [/usr/sbin/ser] reason: <Request Terminated>_ > > I follow the example of ser guide.., but do not work.. > I would appreciate any comment.. > > Regards, > > -- > caiogf > -- caio

1 0

[Serusers] ser + media server
by caio 25 Aug '08

25 Aug '08

Hi all, What do you recommend for media server application to integrate wih ser-0.9.x? SEM or Asterisk? Specifically i'm talking about running services such as IVR, conference and voicemail.. Thanks your suggestions.. -- caio

1 0

[Kamailio-Users] crashing problem
by Jonathan K. Creasy 25 Aug '08

25 Aug '08

We're using Carrier route with about 53,000 routes stored in a mysql databse. We're getting this crash in production that we never had in the lab. The bt is below. Does anyone have any suggestions? nline static void free_to_params(struct to_body* tb) { struct to_param *tp=tb->param_lst; struct to_param *foo; while (tp){ foo = tp->next; <-- **** Line 75 ***** pkg_free(tp); tp=foo; } } -Jonathan #0 free_to (tb=0x817cf10) at parser/parse_to.c:75 #1 0x080daf70 in clean_hdr_field (hf=0x8a8ee18) at parser/hf.c:182 #2 0x080db05b in free_hdr_field_lst (hf=0x8a905b8) at parser/hf.c:209 #3 0x080db12d in free_sip_msg (msg=0x8193030) at parser/msg_parser.c:660 #4 0x08094695 in receive_msg ( buf=0x8168220 "SIP/2.0 183 Session Progress\r\nf: \"16512552900\"<sip:16512552900@206.80.65.90>;tag=as781b438d\r\nt: <sip:+16513401989@lefty.netlogic.net>;tag=c7ad5f5020080822192510\r\ni: 396fb801653abd6f605a12153e0e3e72@20"..., len=1271, rcv_info=0xbf82f844) at receive.c:206 #5 0x080d3e0d in udp_rcv_loop () at udp_server.c:438 #6 0x0806c8f5 in main (argc=3, argv=0xbf82fa34) at main.c:834 (gdb) backtrace full #0 free_to (tb=0x817cf10) at parser/parse_to.c:75 No locals. #1 0x080daf70 in clean_hdr_field (hf=0x8a8ee18) at parser/hf.c:182 __FUNCTION__ = "clean_hdr_field" #2 0x080db05b in free_hdr_field_lst (hf=0x8a905b8) at parser/hf.c:209 No locals. #3 0x080db12d in free_sip_msg (msg=0x8193030) at parser/msg_parser.c:660 No locals. #4 0x08094695 in receive_msg ( buf=0x8168220 "SIP/2.0 183 Session Progress\r\nf: \"16512552900\"<sip:16512552900@206.80.65.90>;tag=as781b438d\r\nt: <sip:+16513401989@lefty.netlogic.net>;tag=c7ad5f5020080822192510\r\ni: 396fb801653abd6f605a12153e0e3e72@20"..., len=1271, rcv_info=0xbf82f844) at receive.c:206 msg = (struct sip_msg *) 0x8193030 __FUNCTION__ = "receive_msg" #5 0x080d3e0d in udp_rcv_loop () at udp_server.c:438 len = 1271 buf = "SIP/2.0 183 Session Progress\r\nf: \"16512552900\"<sip:16512552900@206.80.65.90>;tag=as781b438d\r\nt: <sip:+16513401989@lefty.netlogic.net>;tag=c7ad5f5020080822192510\r\ni: 396fb801653abd6f605a12153e0e3e72@20"... tmp = <value optimized out> from = (union sockaddr_union *) 0x8a91318 fromlen = 16 ri = {src_ip = {af = 2, len = 4, u = {addrl = {2850040017, 135426432, 3213031528, 3085483479}, addr32 = {2850040017, 135426432, 3213031528, 3085483479}, addr16 = {10449, 43488, 29056, 2066, 63592, 49026, 48599, 47080}, addr = "Ñ(à©\200q\022\bhø\202¿×½è·"}}, dst_ip = {af = 2, len = 4, u = {addrl = {509300942, 0, 0, 0}, addr32 = {509300942, 0, 0, 0}, addr16 = {20686, 7771, 0, 0, 0, 0, 0, 0}, addr = "ÎP[\036", '\0' <repeats 11 times>}}, src_port = 5060, dst_port = 5060, proto = 1, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 2, sa_data = "\023ÄÑ(à©\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 2850040017}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 2850040017, sin6_addr = {in6_u = {u6_addr8 = '\0' <repeats 15 times>, u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x818dd90} __FUNCTION__ = "udp_rcv_loop" #6 0x0806c8f5 in main (argc=3, argv=0xbf82fa34) at main.c:834 cfg_log_stderr = 0 cfg_stream = (FILE *) 0xa179008 c = <value optimized out> r = <value optimized out> tmp = 0x8148b9c "ÿÿÿÿ" tmp_len = <value optimized out> port = <value optimized out> proto = <value optimized out> ret = <value optimized out> seed = 639236262 rfd = 4 __FUNCTION__ = "main"

4 9

Re: [Kamailio-Users] Google reference to openser users and devel Archives
by H Quintana 25 Aug '08

25 Aug '08

Hi, My temporary solution was to edit the etc/hosts file in my windows xp machine. The path is: C:\WINDOWS\system32\drivers\etc\hosts And I added the lines: 193.22.119.66 www.openser.org 193.22.119.66 openser.org to point to the Kamailio servers. Hope this helps, Humberto > Subject: Re: [Kamailio-Users] Google reference to openser > users and > devel Archives > To: users(a)lists.kamailio.org > Message-ID: > <200808251326.44077.henning.westerholt(a)1und1.de> > Content-Type: text/plain; charset="utf-8" > > On Friday 15 August 2008, Rafael Risco Gonzales-Vigil > wrote: > > This is for openser/kamailio and opensips list > administrators: > > google still has references to > http://www.openser.org/pipermail/users/ and > > http://www.openser.org/pipermail/devel/ for openser > related searchings and > > those urls does not contains old files anymore so its > dificult search > > information for learning and troubleshooting about > openser/kamailio or > > opensips... I hope someone can solve this issue for > the benefit of all > > current and future community users. > > Hi Rafael, > > we don't have access to the openser.org domain, so > there is nothing what we > can do here, sorry. All access of web and mail services to > openser.org will > go to opensips infrastructure. > > If you encounter such old links with openser.org, you can > simply > change "openser.org" to "kamailio.org", > then you'll be able to access the > content, apart from using the google archives, as Alex > pointed out. > > Cheers, > > Henning >

2 1

[Kamailio-Users] failure route configuration
by Jonathan K. Creasy 25 Aug '08

25 Aug '08

I have this block in my configuration: <excerpt from failure_route[1]> if(!cr_rewrite_uri("1", "call_id")){ t_reply("403", "$fU@$fd Not allowed"); } else { # In case of failure, re-route the request prefix("+"); t_on_failure("2"); if (is_method("INVITE")) t_on_reply("1"); # Relay the request to the gateway t_relay(); } The output in my log file: Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5622]: INFO: carrierroute:carrier_rewrite_msg: uri 8909 was rewritten to sip: 8909@gateway_one Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: INFO: Failure Route Started Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: INFO: carrierroute:carrier_rewrite_msg: uri 8909 was rewritten to sip: 8909@gateway_two Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: ERROR: tm:t_forward_nonack: no branch for forwarding Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: ERROR: tm:w_t_relay: t_forward_nonack failed What am I missing that I need to do prior to calling t_relay()? -Jonathan

6 11

[Kamailio-Users] Google reference to openser users and devel Archives
by Rafael Risco Gonzales-Vigil 25 Aug '08

25 Aug '08

This is for openser/kamailio and opensips list administrators: google still has references to http://www.openser.org/pipermail/users/ and http://www.openser.org/pipermail/devel/ for openser related searchings and those urls does not contains old files anymore so its dificult search information for learning and troubleshooting about openser/kamailio or opensips... I hope someone can solve this issue for the benefit of all current and future community users. rafael Lima - Peru

3 2

[Kamailio-Users] need help on starting opensips v1.4.1
by Do Nguyen Ha 25 Aug '08

25 Aug '08

hi opensips debian:/etc/opensips# /etc/init.d/opensips start Starting opensips: opensipsListening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.1.254 [192.168.1.254]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.1.254 [192.168.1.254]:5060 Aliases: tcp: debian:5060 tcp: localhost:5060 udp: debian:5060 udp: localhost:5060 . debian:/etc/opensips# /etc/init.d/opensips status Status of opensips: opensips is not running. could anyone tell me the tips for this Thanks Ha`

2 1

[Kamailio-Users] Why does Kamailio reply 408 when 'fr_inv_timer' expires?
by Iñaki Baz Castillo 25 Aug '08

25 Aug '08

Hi, when 'fr_inv_timer' expires Kamailio CANCELs the pending branches and reply "408 Request Timeout" to the UAC. Well, I wonder if 408 is appropiate. I think "480 User Unavailable" or "487 Cancelled" would be better: I received those replies in other maillist: "408 indicates transaction timeout. You have sent request to next hop and it has not responded back. If the gateway has not responded at all (Including 100 trying), 408 is appropriate. 480 is more of application usage meaning that the user is not available and would be appropriate if the gateway has at least send one provisional response i.e. 100, 180, 183..." "408 is used consistently to mean IP-level timeout in sending a request from one element to another. It is also defined to be returned by a UA when the time specified in "Expires" has been reached. (In theory, two different error codes should be defined for these two situations, but that has not been done.) However, in a ring-no-answer situation, it is more common for an upstream proxy to send a CANCEL to the UA/gateway, in which case the UA/gateway provides a 487 response to the INVITE." This last point suggest that Kamailio could do the following: - If the proxy has received provisional responses from callee's then not send a 408 upstream since when the proxy sends CANCEL to UAS's it will receive some 487 and would return a 487 upstream. - If the proxy hasn't received provisional responses then this is a clear "408 Request timeout". So a possibility is that Kamailio just reply a 408 in real timeouts and reply 487 when 'fr_inv_timer' expires (or just reply nothing since UAS's will generate the 487). Opinions? -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 2

[Kamailio-Users] Where have the OLD Module Docs gone?
by Douglas Garstang 25 Aug '08

25 Aug '08

The module documentation for versions 1.2, 1.3 etc used to be available on the web site. Where has it gone? When you clicked on the documentation link, you used to be able to choose the version of OpenSER(Kam...OpenSIPS whatever) that you wanted. Now, your taken immediately to what appears to be the documentation for the LATEST version. Doug.

7 7

[Kamailio-Users] presence fail
by Ricky Gutierrez 25 Aug '08

25 Aug '08

Hi, lists I have some problems with the simple presence in kamailio 1.4, I add to my kamailio.cfg the following line and when restarting the it shows me errors in this it lines Aug 24 19:31:41 [21946] ERROR:core:set_mod_param_regex: parameter <force_active> not found in module <presence> Aug 24 19:31:41 [21946] CRITICAL:core:yyerror: parse error in config file, line 50, column 19-20: Can't set module parameter Aug 24 19:31:41 [21946] ERROR:core:main: bad config file (1 errors) Aug 24 19:31:41 [21946] NOTICE:presence:destroy: destroy module ... I remove of the Makefile the modul presence kamailio.cfg loadmodule "presence.so" modparam("presence|auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence", "force_active", 1) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:192.168.10.1:5060") where the error can be? regards rickygm

2 1

[Kamailio-Users] preparing openser 1.3.3
by Daniel-Constantin Mierla 24 Aug '08

24 Aug '08

Hello, I propose to build a new minor release for openser 1.3 branch on the 27th of August (Wednesday). There are already a lot of fixes since 1.3.2. Probably couple of backports from 1.4 will get in until Wednesday, the rest, if it is the case, shall get in 1.3.4, probably in about another month, as we are going to maintain 2 stable branches. Perhaps 1.4.1 will be out by same time or so. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] FreeRadius patch for OpenSer1.2.2
by Nuno Marques 24 Aug '08

24 Aug '08

Hi, I'm following the examples on the book "Building Telephony Systems with OpenSer". I came to Accounting and Billing chapter using FreeRadius. Flavio (the author) says we need to apply a patch to the freeradius to "fix a problem with accounting type 15". My question is - is it still necessary to apply that patch or the new version of freeradius allready covers it? I'm having big problems applying that patch because of the key signature problems. Appreciate any help in this subject. Thanks in advance, Nuno

5 5

[Kamailio-Users] AUTO: Miodrag Radulovic is out of the office (returning 07-09-2008)
by Miodrag Radulovic 23 Aug '08

23 Aug '08

I am out of the office until 07-09-2008. Privremeno nedostupan - Out-of-Office Message. Poruka koju ste poslali je prispela i bice sacuvana, ali napominjem da sam na odmoru van zemlje do 07.09.2008. i moj pristup elektronskoj posti bice veoma ogranicen. Ukoliko se radi o potrebi za hitnom intervencijom, dostupan sam preko mobilnog telefona 065/3016576. The message that you have sent will be saved in my inbox, but please note that due to summer holliday I will be out of the country until 07.09.2008. In that period my access to my email will be very limited. In case of urgency, I am available on my mobile phone: +381 65 3016576. Miodrag Radulovic General Manager COMUTEL Note: This is an automated response to your message "Users Digest, Vol 39, Issue 83" sent on 23.8.2008 0:49:18. This is the only notification you will receive while this person is away.

1 0

Re: [Serusers] SER - Waiting Transaction
by Arun Kumar 22 Aug '08

22 Aug '08

Hi, thanks for your reply. I'll ignore this but I see so many open connection to MySQL from SER and they don't get close automatically also resulting in denial of other service using my MySQL. Thanks arun On Mon, Aug 18, 2008 at 10:12 PM, Jiri Kuthan <jiri(a)iptel.org> wrote: > ignore it > > Arun Kumar wrote: > >> Hi, >> >> >> I'm running SER 0.96 in my production for a long but there is strange >> problem , when I do serctl moni and looks for transaction it shows waiting >> transaction in () almost 400+ can some one tell me why there are so many >> transaction in waiting while I see processing only 10-20 transaction. >> >> Thanks >> arun >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >

4 5

[Kamailio-Users] Problem to run MediaProxy 2.0.3
by Leonardo Santana 22 Aug '08

22 Aug '08

Hi all, I´m using mediaproxy 2.0.3 on OpenSuSE 11.0. When I run the "media-dispatcher" I see this message bellow: PS.: The OpenSuSE 11.0 doesn´t have the module "libgnutls-extra.so.13" only the libgnutls-extra.so.26". ######## Aug 21 17:18:32 nana media-dispatcher[5502]: [-] Log opened. Aug 21 17:18:32 nana media-dispatcher[5502]: [-] Starting MediaProxy Dispatcher 2.0.3 Aug 21 17:18:32 nana media-dispatcher[5502]: [-] Traceback (most recent call last): Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/local/mediaproxy-2.0.3/media-dispatcher", line 46, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from mediaproxy.dispatcher import Dispatcher Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/local/mediaproxy-2.0.3/mediaproxy/dispatcher.py", line 31, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from mediaproxy.tls import X509Credentials, X509NameValidator Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/local/mediaproxy-2.0.3/mediaproxy/tls.py", line 12, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls import crypto Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/crypto.py", line 11, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.validators import method_args, none, ignore, list_of, one_of Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/validators.py", line 9, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.constants import * Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/constants.py", line 38, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.library import constants Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/library/__init__.py", line 7, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.library import errors Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/library/errors.py", line 14, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.library.functions import gnutls_strerror, gnutls_alert_get Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/library/functions.py", line 6, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] _libraries['libgnutls-extra.so.13'] = CDLL('libgnutls-extra.so.13') Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/ctypes/__init__.py", line 348, in __init__ Aug 21 17:18:32 nana media-dispatcher[5502]: [-] self._handle = _dlopen(self._name, mode) Aug 21 17:18:32 nana media-dispatcher[5502]: [-] OSError: libgnutls-extra.so.13: cannot open shared object file: No such file or directory ##### Thanks a lot!!!

2 1

[Kamailio-Users] RFC: branches/location records control
by Daniel-Constantin Mierla 22 Aug '08

22 Aug '08

Hello, one of the most hidden parts of openser to the config is the branch processing. Behind a branch is following structure: struct branch { char uri[MAX_URI_SIZE]; unsigned int len; /* Real destination of the request */ char dst_uri[MAX_URI_SIZE]; unsigned int dst_uri_len; /* Path vector of the request */ char path[MAX_PATH_SIZE]; unsigned int path_len; int q; /* Preference of the contact among contact within the array */ struct socket_info* force_send_socket; unsigned int flags; }; I am thinking in offering a fine access to this structure in configuration file, so the admin can control much closer the branch processing. By using while and and index, checks of the attributes in branches could be done. By doing it in main route, one can reduce the overload of stateful processing, by dropping a branch before going to transaction, for example. I am looking at something similar for user location records. Such a record has many attributes, and from config is no way to access them without doing a lookup and changing r-uri/branches. There could be many benefits, for example: - being able to detect if a user calls from a registered phone or not - detecting how many contacts a user has at a moment - ability to check and, eventually, drop processing before doing changes to branches and so on... The access will be provided via pseudo-variables, perhaps the location records will be fetched in variables upon calling a function that gets an AoR as parameter. I have seen on mailing lists discussions about needs that will be solved by this new feature. Any comments? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

4 7

[Kamailio-Users] kamailio-1.4.0 and SerMyAdmin
by luzango mfupe 22 Aug '08

22 Aug '08

Hi, Im bit cnfsd, does SerMyAdmin support Kamailio-1.4.0?. I did my configuration using SerMyAdmin 0.7-Beta.war and then 0.7-beta.war files, but everytime i pointed my browser to the IP address of my installation my Tomcat6 server return the message that SerMyAdmin does not exist!!! Can someone pls'e clarify this?? .. wbr, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

2 4

[Kamailio-Users] openser 1.2.1 errors in the log
by toly 22 Aug '08

22 Aug '08

Greetings, Proxy compiled with TLS support, tls is not turned on Routing script inclulded below. Underlying database is Sybase. The following is happening: There are 25 users having as a contact each other. Softphone is Bria. Without any calls, just subscribe/notify proxy is running out of the socket in one hour. The users are or outside coming via corp firewall, or if inside, the route is setup the way the going via corp firewall, so they are natted. Proxy is in DMS on the same network as firewall. There are no problems with nat traversal. Daniel, could you please take look at the script? Thanks a lot, Toly. this is from log: <receive.c: 206> receive_msg: cleaning up 08/15 20:12:58 29156 debug <ip_addr.c: 109> tcpconn_new: new tcp connection to: xxx.xxx.xxx.xx 08/15 20:12:58 29156 debug <tcp_main.c: 402> tcpconn_new: on port 58544, type 2 08/15 20:12:58 29156 debug <tcp_main.c: 497> tcpconn_add: hashes: 96, 45 08/15 20:12:58 29156 debug <tcp_main.c: 993> handle_new_connect: new connection: 0xb6485590 98 flags: 0002 08/15 20:12:58 29156 debug <tcp_main.c: 935> WARNING: send2child: no free tcp receiver, connection passed to the least busy one (1) this is routing script: log_facility=LOG_LOCAL2 debug=7 fork=yes log_stderror=no listen=PROXY_ENV # This is where OpenSER installed port=5070 children=20 tcp_connect_timeout=2 tcp_max_connections=8192 reply_to_via = no sip_warning = yes check_via = no dns=no rev_dns=no disable_dns_blacklist=true disable_tls = 1 listen = tls:PROXY_ENV:5061 tls_verify_server = 1 tls_verify_client = 0 tls_require_client_certificate = 0 tls_method = SSLv23 tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" ###########################################################33 ###########################################################33 ###########################################################33 mpath = "/usr/local/lib/openser/modules" loadmodule "uac.so" loadmodule "psybase.so" loadmodule "pike.so" modparam("pike", "remove_latency", 120) modparam("pike", "reqs_density_per_unit", 10000) modparam("pike", "sampling_time_unit", 60) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0666 ) modparam("mi_fifo", "reply_dir", "/tmp/" ) modparam("mi_fifo", "reply_indent", "\t" ) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0) loadmodule "avpops.so" modparam("avpops", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "use_domain", 1) loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) 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) loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "uri.so" loadmodule "uri_db.so" modparam("uri_db", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") loadmodule "alias_db.so" modparam("alias_db", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") 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 "dispatcher.so" modparam("dispatcher", "list_file", "/usr/local/etc/openser/dispatcher.list") #modparam("dispatcher", "dst_avp", "$avp(i:271)") #modparam("dispatcher", "grp_avp", "$avp(i:272)") #modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "flags", 2 ) # failover mode set flag=2 loadmodule "usrloc.so" modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("usrloc", "nat_bflag", 6) loadmodule "textops.so" loadmodule "acc.so" # set the reporting log level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "log_flag", 1 ) #modparam("acc", "log_flag", 0 ) modparam("acc", "db_flag", 1 ) modparam("acc", "log_missed_flag", 1 ) modparam("acc", "db_missed_flag", 1 ) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("acc", "log_extra", "src_ip=$si;src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "db_extra", "src_ip=$si;src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "report_cancels", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") 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", 0) modparam("auth_db", "load_credentials", "rpid") loadmodule "domain.so" modparam("domain", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "mediaproxy.so" modparam("mediaproxy","natping_interval", 10) modparam("mediaproxy","mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "received_avp", "$avp(i:801)") modparam("nathelper", "sipping_from", "sip:pinger@parusivoip.net") modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", 7) loadmodule "registrar.so" modparam("registrar", "received_avp", "$avp(i:801)") modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 50) 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", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) loadmodule "rr.so" modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "permissions.so" modparam("permissions", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("permissions", "db_mode", 1) ######################################################################## ######################################################################## ######################################################################## route { if ( is_method("REGISTER") ) { if ( !pike_check_req() ) { xlog("L_NOTICE", "TRACKING [0] ALARM - TOO MANY HITS ON REGISTER on IP=$si !!"); exit; } }; if ( is_method("INVITE") ) { if ( !pike_check_req() ) { xlog("L_NOTICE", "TRACKING [0] ALARM - TOO MANY HITS ON INVITE on IP=$si !!"); exit; } }; # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; xlog("L_NOTICE", "TRACKING [0] NEW REQUEST - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); force_rport(); # ----------------------------------------------------------------- # Record Route Section and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && nat_uac_test("19")) { record_route_preset("PROXY_ENV:5070;nat=yes"); # This is where OpenSER installed xlog("L_NOTICE", "TRACKING [0] RECORD ROUTE PRESET- M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); } else if (method!="REGISTER") { xlog("L_NOTICE", "TRACKING [0] RECORD ROUTE - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); record_route(); }; append_hf("P-hint: rr-enforced\r\n"); if ( search("User-Agent: Parus.*") && is_method("BYE|REFER")) { # check the transport of callee $avp(i:44) = $(ru{uri.user}); if ( $avp(i:44) != $tU ) { $avp(i:45) = $avp(i:44); } else { $avp(i:45) = $tU; } avp_db_query("exec oserIsTcpTransport @to_user='$avp(i:45)', @td='$td'", "$avp(i:11)"); xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si AVP11=$avp(i:11) AVP44=$avp(i:44) TD=$td"); if ( $avp(i:11) == "tcp" ) { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS TCP - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); add_uri_param("transport=tcp"); } else if ( $avp(i:11) == "tls" ) { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS TLS - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); add_uri_param("transport=tls"); } else if ( $avp(i:11) == "udp" ) { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS UDP - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); add_uri_param("transport=udp"); } else { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS NOT TCP - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); } } # ----------------------------------------------------------------- # ACC Section # ----------------------------------------------------------------- if ( is_method("CANCEL|BYE|INVITE|MESSAGE") ) { setflag(1); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if ( is_method("BYE|CANCEL") ) { xlog("L_NOTICE", "TRACKING [0] CALL TEAR DOWN: END_MEDIA_SESSION - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { xlog("L_NOTICE", "TRACKING LOOSE_ROUTE - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); if (!has_totag()) { xlog("L_NOTICE", "TRACKING LOOSE_ROUTE Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if ( is_method("INVITE") ) { if ( !allow_trusted() ) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); xlog("L_NOTICE", "TRACKING LOOSE_ROUTE proxy authorization failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } if (!check_from()) { sl_send_reply("403", "Spoofed from-URI Detected"); xlog("L_NOTICE", "TRACKING LOOSE_ROUTE spoofed from-URI etectet - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } consume_credentials(); xlog("L_NOTICE", "TRACKING LOOSE_ROUTE proxy authorized - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); }; }; if ( (nat_uac_test("19") || search("^Route:.*;nat=yes")) && is_method("INVITE|REFER|ACK") ) { setbflag(6); # NONONO fix_nated_contact() xlog("L_NOTICE", "TRACKING LOOSE_ROUTE using media proxy - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); use_media_proxy(); }; route(1); exit; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if ( !is_uri_host_local() ) { if ( is_from_local() || allow_trusted() ) { xlog("L_NOTICE", "TRACKING[0] IS_FROM_LOCAL OR TRUSTED - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); } else { xlog("L_NOTICE", "TRACKING[0] NOT IS_FROM_LOCAL OR NOT TRUSTED - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); sl_send_reply("403", "Forbidden"); exit; } }; if ( is_method("CANCEL|BYE") ) { route(4); exit; } else if ( is_method("ACK") ) { route(1); exit; } else if ( is_method("INVITE") ) { route(3); exit; } else if ( is_method("REGISTER") ) { route(2); exit; } else if ( is_method("OPTIONS") ) { route(11); exit; } else if ( is_method("MESSAGE") ) { xlog("L_NOTICE", "TRACKING [14] ROUTE14 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); route(14); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; xlog("L_NOTICE", "TRACKING[0] calling default route(1) M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); route(1); } # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- route[1] { xlog("L_NOTICE", "TRACKING [1] - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); if(isbflagset(6)) { xlog("L_NOTICE", "TRACKING [1] USING NATed reply - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); t_on_reply("1"); } else { xlog("L_NOTICE", "TRACKING [1] USING Standard reply - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); t_on_reply("2"); } if (!t_relay()) { xlog("L_NOTICE", "TRACKING [1] t_relay() Failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; exit; } # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- route[2] { sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!search("^Contact:\ +\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); setbflag(7); xlog("L_NOTICE", "TRACKING [2] Fixed nated register - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); } if (!save("location")) { xlog("L_NOTICE", "TRACKING [2] Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); sl_reply_error(); }; xlog("L_NOTICE", "TRACKING [2] Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); } # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- route[3] { xlog("L_NOTICE", "TRACKING [3] - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if ( !allow_trusted() ) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); xlog("L_NOTICE", "TRACKING [3] proxy authorization failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } if (!check_from()) { sl_send_reply("403", "Spoofed from-URI Detected"); xlog("L_NOTICE", "TRACKING [3] spoofed from-URI etectet - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } consume_credentials(); }; if (client_nat_test("3")) { fix_nated_contact(); setbflag(6); xlog("L_NOTICE", "TRACKING [3] FIX_NATED_CONTACT - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); }; if ( alias_db_lookup("dbaliases") ) { xlog("L_NOTICE", "TRACKING [3] callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); } if ( !is_domain_local("$rd") ) { xlog("L_NOTICE", "TRACKING [3] call for foreign domain M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); route(5); route(1); exit; } if ( does_uri_exist() ) { xlog("L_NOTICE", "TRACKING [3] callee was local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (lookup("location")) { xlog("L_NOTICE", "TRACKING [3] LOOKUP LOCATION SUCC - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); append_hf("P-hint: usrloc applied\r\n"); remove_hf("Alert-Info"); if ( proto==tls ) { append_hf("Alert-Info: 1\r\n"); } route(5); t_on_branch("1"); route(1); exit; } }; xlog("L_NOTICE", "TRACKING [3] SOFTPHONE LOOKUP LOCATION NOT SUCC M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); # # call pstn handler # if ( route(7) ) { xlog("L_NOTICE", "TRACKING [3] call to PSTN handler successful M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; # will never hit it } else { xlog("L_NOTICE", "TRACKING [3] call to PSTN handler failed M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; # exit for now } # # never hit this for now, possibly in the future if # forward to different domain or so # xlog("L_NOTICE", "TRACKING[3] calling default route(1) M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); route(1); exit; } # ----------------------------------------------------------------- # PSTN call section # ----------------------------------------------------------------- route[7] { xlog("L_NOTICE", "TRACKING [37] PSTN SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (uri =~ "^sip:011.*@.*") { xlog("L_NOTICE", "TRACKING [37] PSTN SECTION ATTEMPT TO MAKE INTERNATIONAL CALL M=$rm RURI=$ru F=$fu T=$tu TU=$tu"); sl_send_reply("403", "Forbidden - no international calls"); return (-1); }; if ($(tU{s.len}) > 11 ) { strip(1); xlog("L_NOTICE", "TRACKING [37] PSTN SECTION MORE THEN 11 DIGITS M=$rm RURI=$ru F=$fu T=$tu TU=$tu"); sl_send_reply("403", "Forbidden - no international calls"); return (-1); } if ($(tU{s.len}) == 11 ) { strip(1); xlog("L_NOTICE", "TRACKING [37] PSTN SECTION STRIP ONE DIGIT M=$rm RURI=$ru F=$fu T=$tu TU=$tu"); } if ( ($(rU{s.len}) != 10) && ($(rU{s.len}) != 2) && ($(rU{s.len}) != 3) && ($(rU{s.len}) != 4) && ($(rU{s.len}) != 1) ) { xlog("L_NOTICE", "TRACKING [37] S.LEN != 10 or 1 or 2 or 3 or 4 M=$rm RURI=$ru F=$fu T=$tu rU=$rU IP=$si ID=$ci"); sl_send_reply("403", "Forbidden - invalid number"); xlog("L_NOTICE", "TRACKING [7] LEAVING PSTN SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); return (-1); }; if (uri =~ "^sip:[2-9]{1}[0-9]{2}[2-9]{1}[0-9]{2}[0-9]{4}@.*") { xlog("L_NOTICE", "TRACKING [37] PSTN OUTBOUND $fU -> $ruri "); avp_db_query("exec oserGetCallIdForBvox @fU='$fU', @from_domain='$fd'", "$avp(i:12)"); uac_replace_from("$avp(i:12)"); xlog("L_NOTICE", "TRACKING [37] REPLACED FROM PSTN avp = $avp(i:12[1]) $fU -> $ruri "); route(5); # select next gateway simulates NEXTONE # 1 - DEV, 2 - PROD asterisk, 3 - john's nextone, 4 - prod nextone, 5 - qa ds_select_domain("GWGROUPID", "4"); t_on_failure("2"); xlog("L_NOTICE", "TRACKING [37] DISPATCHER FORWARD PSTN $fu -> $ruri "); route(1); return (1); # will never hit this } else { xlog("L_NOTICE", "TRACKING [37] Attempt to dial invalid number $fu -> $tu"); sl_send_reply("403", "Forbidden - invalid number"); xlog("L_NOTICE", "TRACKING [37] LEAVING PSTN SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); return (-1); }; } # ----------------------------------------------------------------- # DEV / TEST section # ----------------------------------------------------------------- route[6] { xlog("L_NOTICE", "TRACKING [6] DEV / TEST SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); } # ----------------------------------------------------------------- # CANCEL or BYE handler # ----------------------------------------------------------------- route[4] { xlog("L_NOTICE", "TRACKING [4] - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (client_nat_test("3")) { fix_nated_contact(); xlog("L_NOTICE", "TRACKING [4] FIX_NATED_CONTACT - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); }; route(1); } # ----------------------------------------------------------------- # NAT traversal section # ----------------------------------------------------------------- route[5] { xlog("L_NOTICE", "TRACKING [5] NAT TRAVERSAL SECTION - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (isbflagset(6)) { if (!isflagset(8)) { setflag(8); xlog("L_NOTICE", "TRACKING [5] BFLAG 6 IS SET, USING MEDIA PROXY - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); avp_printf("$avp(i:450)", "$dd"); if (avp_check("$avp(i:450)", "eq/$src_ip/g")) { xlog("L_NOTICE", "TRACKING [5] LOOKUP SUCC, FLAGS 6 AND 7 BEHIND SAME NAT $fu -> $tu $dd $src_ip"); } use_media_proxy(); }; }; } # ------------------------------------------------------------------------ # OPTIONS Message Handling # ------------------------------------------------------------------------ route[11] { xlog("L_NOTICE", "TRACKING [11] OPTIONS - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } xlog("L_NOTICE", "TRACKING [11] - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); sl_send_reply("200", "Got it"); exit; } # ----------------------------------------------------------------- # NAT reply route # ----------------------------------------------------------------- onreply_route[1] { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[1] NAT REPLY M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci"); if (nat_uac_test("1")) { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[1] CLIENT_NAT_TEST FIXED NATED CONTACT M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci"); fix_nated_contact(); }; # we are checking here for a progressing return... ie a 180 Ringing or # 183 session progress -- if this occurs we don't care from here on # about failures as a gateway is handling the call... if( status =~ "18[0-9]" ) { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[1] [$Tf] ORR: $ci -- SIP-$rs Reset t_on_failure()\n"); t_on_failure("0"); } else { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[1] [$Tf] ORR: $ci -- $rs $rr\n" ); } if ( isbflagset(6) && (status=~"(180)|(183)|2[0-9][0-9]")) { #if (!search("^Content-Length:\ +0")) { if ( $cl > 0 ) { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[1] USE MEDIA PROXY M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci"); use_media_proxy(); }; }; exit; } # ----------------------------------------------------------------- # standard reply route # ----------------------------------------------------------------- onreply_route[2] { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[2] STANDARD REPLY- M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); # we are checking here for a progressing return... ie a 180 Ringing or # 183 session progress -- if this occurs we don't care from here on # about failures as a gateway is handling the call... if( status =~ "18[0-9]" ) { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[2] [$Tf] ORR: $ci -- SIP-$rs Reset t_on_failure()\n"); t_on_failure("0"); } else { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[2] [$Tf] ORR: $ci -- $rs $rr\n" ); } exit; } # ----------------------------------------------------------------- # FAILURE route # ----------------------------------------------------------------- failure_route[1] { xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [1] - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if(t_check_status("408|500|503")) { if ( t_check_status("408") ) { if ( ds_next_domain() ) { xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [1] TRYING NEXT GW - M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); t_on_failure("1"); route(1); return; } } else { xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [1] NOT 408 STATUS - M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); t_reply("503", "Service not available"); return; } } } # ----------------------------------------------------------------- # FAILURE route # ----------------------------------------------------------------- failure_route[2] { # If fr_timer expires t_check_status("408") is true, although $rs is <null> if( t_check_status("408") ){ xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- TIMEOUT for Gateway $rd\n" ); } else { xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- $rs reason $rr\n" ); } # basically what we do is # 1. not worry about reasonable failures (ie busy, wrong number etc...) # 2. go to next destination gateway # 3. if no destination gateway then 503 (service unavailable) # 403 -- typically ISDN network says 'not a valid number' etc.. if( t_check_status("403") ){ xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Forbidden -> ISDN Cause Code 1\n" ); end_media_session(); return; } # 408 -- timeout -- typically the end party has not answered # Since we cancel t_on_failure() on a provisional response we should not be # getting a 408 timeout from a gateway at this stage.. it will just "fall through" #if( t_check_status("408") ){ # xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Timeout\n" ); # return; #} # 486 -- User Busy if( t_check_status("486") ){ xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Destination Busy\n" ); end_media_session(); return; } # 487 -- Request Cancelled (usually in response to a CANCEL transaction) if( t_check_status("487") ){ xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Request Cancelled\n" ); end_media_session(); return; } # ok... so at this stage we try the next gateway (unless we don't have one) # if no next gateway we bail... if( ds_next_domain() ){ #t_on_reply("1"); t_on_failure("2"); xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci Next gateway $fU -> $tU via $rd\n" ); route(1); return; } else { xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci No more gateways -> 503.\n" ); end_media_session(); t_reply("503", "Service unavailable -- no more gateways" ); return; } } # ----------------------------------------------------------------- # BRANCH route # ----------------------------------------------------------------- branch_route[1] { if (isbflagset(6)) { xlog("L_NOTICE", "TRACKING BRANCH_ROUTE[1] Branch marked as NATed- M=$rm branch=$T_branch_idx, branch flags=$bf F=$fu T=$tu IP=$si ID=$ci"); } else { xlog("L_NOTICE", "TRACKING BRANCH_ROUTE[1] Branch marked as not NATed- M=$rm branch=$T_branch_idx, branch flags=$bf F=$fu T=$tu IP=$si ID=$ci"); } } # ------------------------------------------------------------------------ # MESSAGE Message Handling # ------------------------------------------------------------------------ route[14] { if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } -- View this message in context: http://www.nabble.com/openser-1.2.1-errors-in-the-log-tp19007938p19007938.h… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 4

[Kamailio-Users] openser 1.2.1 errors in the log - tcp blocking
by toly 22 Aug '08

22 Aug '08

Greetings, This is openser 1.2.1 compiled with TLS, TLS is not turned on. This is part of the config file: listen=xxx.xxx.xxx.xxx # This is where OpenSER installed port=5070 children=25 tcp_connect_timeout=2 tcp_max_connections=8192 reply_to_via = no sip_warning = yes check_via = no dns=no rev_dns=no disable_dns_blacklist=true here's the problem: I have 25 users having each other as a contact. Softphone is Bria, transport is tcp here comes subscribe: 08/21 15:07:46 8029 debug <loose.c: 115> find_first_route: No Route headers found 08/21 15:07:46 8029 debug <loose.c: 883> loose_route: There is no Route HF 08/21 15:07:46 8029 notice <xlog.c: 148> TRACKING [18] - M=SUBSCRIBE RURI=sip:dmarkiew@tamtam.com F=sip:sgarratt@tamtam.com T=sip:dmarkiew@tamtam.com IP=333.64.100 .15 ID=YTBmOTJhNGI5M2RmZjYyNmM2NGFhMmI1NmRjMmY4ZWE. 08/21 15:07:46 8029 debug <dset.c: 445> rewrite_uri: Rewriting Request-URI with 'sip:dmarkiew@76.222.53.112:53363;transport=TCP;rinstance=575ebcf03ae72b26' 08/21 15:07:46 8029 debug <dset.c: 285> ******* setting for branch 0 flags C0 08/21 15:07:46 8029 debug <dset.c: 285> ******* setting for branch 1 flags C0 08/21 15:07:46 8029 notice <xlog.c: 148> TRACKING [1] - M=SUBSCRIBE RURI=sip:dmarkiew@76.222.53.112:53363;transport=TCP;rinstance=575ebcf03ae72b26 D-URI=sip:76.222.53.11 2:53363;transport=TCP F=sip:sgarratt@tamtam.com T=sip:dmarkiew@tamtam.com IP=333.64.100.15 ID=333.64.100.15 08/21 15:07:46 8029 notice <xlog.c: 148> TRACKING [1] USING NATed reply - M=SUBSCRIBE RURI=sip:dmarkiew@76.222.53.112:53363;transport=TCP;rinstance=575ebcf03ae72b26 D-UR I=sip:76.222.53.112:53363;transport=TCP F=sip:sgarratt@tamtam.com T=sip:dmarkiew@tamtam.com IP=333.64.100.15 ID=333.64.100.15 08/21 15:07:46 8029 debug <t_lookup.c: 1037> DEBUG: t_newtran: T on entrance=0xffffffff 08/21 15:07:46 8029 debug <parser/msg_parser.c: 310> parse_headers: flags=ffffffffffffffff 08/21 15:07:46 8029 debug <parser/msg_parser.c: 310> parse_headers: flags=78 08/21 15:07:46 8029 debug <t_lookup.c: 407> t_lookup_request: start searching: hash=41567, isACK=0 and later I get this: 08/21 15:07:48 8029 err <tcp_main.c: 291> ERROR: tcp_blocking_connect: timeout 2 s elapsed from 2 s 08/21 15:07:48 8029 err <tcp_main.c: 459> ERROR: tcpconn_connect: tcp_blocking_connect failed 08/21 15:07:48 8029 err <tcp_main.c: 721> ERROR: tcp_send: connect failed 08/21 15:07:48 8029 err <../../forward.h: 105> msg_send: ERROR: tcp_send failed 08/21 15:07:48 8029 err <t_fwd.c: 767> ERROR:tm:t_forward_nonack: sending request failed 08/21 15:07:48 8029 debug <tcp_main.c: 718> tcp_send: no open tcp connection found, opening new one And I get a lot of it. Eventually I may get SO_ERROR(111) and then I have to restart proxy. I can post entire config script if anybody had/resolved/willing to take a look at the problem. Thanks a lot, Toly -- View this message in context: http://www.nabble.com/openser-1.2.1-errors-in-the-log---tcp-blocking-tp1909… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 1

[Kamailio-Users] Res: Res: Help with SerMyAdmin
by Flavio Goncalves 21 Aug '08

21 Aug '08

Hi Edson, Please check the tomcat logs. They can be found at /usr/local/tomcat6/logs. When tomcat6 don't start, usually is a typo in the configuration files. Best regards, Flavio E. Goncalves ----- Mensagem original ---- De: Edson - Listas <4lists(a)gmail.com> Para: Flavio Goncalves <flavio(a)asteriskguide.com> Cc: Asim Riaz <ariaz78(a)gmail.com>; users <users(a)lists.kamailio.org> Enviadas: Quinta-feira, 21 de Agosto de 2008 14:05:27 Assunto: Re: [Kamailio-Users] Res: Help with SerMyAdmin Yesterday, we faced a similar/related problem, but because Tomcat process was killed. When we try to run it again, SerMyAdmin refuse to start, altought all configurations and DB definitions where there... Any ideas on how to restore it? Edson. Flavio Goncalves escreveu: > Hi Assim, > > I'm glad it is working. SerMyAdmin creates the database fields during > the first run, that's why it worked after restarting the tomcat. The sql > file creates the required users. I'm working to have a debian package > for SerMyAdmin soon. > > Regards, > > Flavio E. Goncalves > > ----- Mensagem original ---- > De: Asim Riaz <ariaz78(a)gmail.com> > Para: users <users(a)lists.kamailio.org> > Enviadas: Quinta-feira, 21 de Agosto de 2008 12:02:50 > Assunto: Re: [Kamailio-Users] Help with SerMyAdmin > > Its working thanks for the help. > I removed my existing serMyAdmin directory from tomcat, restarted the > tomcat and run the script mysql -u root -p openser <openser.sql > > regards > Asim Riaz > > On Thu, Aug 21, 2008 at 3:46 PM, Nuno Marques <yangsengopenser(a)gmail.com > <mailto:yangsengopenser@gmail.com>> wrote: > > How did you add the user? With openserctl or with sermyadmin? > Note that you have to use the same login/pass that are on the sql > script - admin@setup for login and secret for pass. > > > 2008/8/21 Asim Riaz <ariaz78(a)gmail.com <mailto:ariaz78@gmail.com>> > > When I am adding a user i am getting > java.lang.NullPointerException; > > also I cant login using admin username. > > On Thu, Aug 21, 2008 at 3:06 PM, Nuno Marques > <yangsengopenser(a)gmail.com <mailto:yangsengopenser@gmail.com>> > wrote: > > Hi again, > > I was looking at the database schema of openser and i > noticed that the subscriber table only has 12 columns till rpid. > Looking at the openser.sql script, it has more than 12 > columns (including the 'version' column...). > What is wrong? Should SerMyAdmin add this columns to the > subscriber table? > Is it lacking me something in the database subscribet table? > Any help with this? > > Thanks in advance, > > Nuno > > 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com > <mailto:muhammad.akl@gmail.com>> > > > > Hi Nuno > > i faced some problems your already facing and succeeded > to solve them so i'd like to as you > > did you install SERweb tables ? > > if though the problems come from there especially the > phplib_id field a duplication happened between it and > the "id" filed so you have to remove SERweb tables and > all thing will go smoothly without any errors > > p.s i recommend you to remove the whole database e.g > openser and install it again using openser_mysql.sh script > > e.g /path/to/openser_mysql.sh create > > press yes for presence and that stuff but for SERweb no > > try and tell me what you get > > BR > > Muhammad > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] Res: Help with SerMyAdmin
by Flavio Goncalves 21 Aug '08

21 Aug '08

Hi Assim, I'm glad it is working. SerMyAdmin creates the database fields during the first run, that's why it worked after restarting the tomcat. The sql file creates the required users. I'm working to have a debian package for SerMyAdmin soon. Regards, Flavio E. Goncalves ----- Mensagem original ---- De: Asim Riaz <ariaz78(a)gmail.com> Para: users <users(a)lists.kamailio.org> Enviadas: Quinta-feira, 21 de Agosto de 2008 12:02:50 Assunto: Re: [Kamailio-Users] Help with SerMyAdmin Its working thanks for the help. I removed my existing serMyAdmin directory from tomcat, restarted the tomcat and run the script mysql -u root -p openser <openser.sql regards Asim Riaz On Thu, Aug 21, 2008 at 3:46 PM, Nuno Marques <yangsengopenser(a)gmail.com> wrote: How did you add the user? With openserctl or with sermyadmin? Note that you have to use the same login/pass that are on the sql script - admin@setup for login and secret for pass. 2008/8/21 Asim Riaz <ariaz78(a)gmail.com> When I am adding a user i am getting java.lang.NullPointerException; also I cant login using admin username. On Thu, Aug 21, 2008 at 3:06 PM, Nuno Marques <yangsengopenser(a)gmail.com> wrote: Hi again, I was looking at the database schema of openser and i noticed that the subscriber table only has 12 columns till rpid. Looking at the openser.sql script, it has more than 12 columns (including the 'version' column...). What is wrong? Should SerMyAdmin add this columns to the subscriber table? Is it lacking me something in the database subscribet table? Any help with this? Thanks in advance, Nuno 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com> Hi Nuno i faced some problems your already facing and succeeded to solve them so i'd like to as you did you install SERweb tables ? if though the problems come from there especially the phplib_id field a duplication happened between it and the "id" filed so you have to remove SERweb tables and all thing will go smoothly without any errors p.s i recommend you to remove the whole database e.g openser and install it again using openser_mysql.sh script e.g /path/to/openser_mysql.sh create press yes for presence and that stuff but for SERweb no try and tell me what you get BR Muhammad _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

2 1

Re: [Kamailio-Users] Help with SerMyAdmin
by Asim Riaz 21 Aug '08

21 Aug '08

Its working thanks for the help. I removed my existing serMyAdmin directory from tomcat, restarted the tomcat and run the script mysql -u root -p openser <openser.sql regards Asim Riaz On Thu, Aug 21, 2008 at 3:46 PM, Nuno Marques <yangsengopenser(a)gmail.com>wrote: > How did you add the user? With openserctl or with sermyadmin? > Note that you have to use the same login/pass that are on the sql script - > admin@setup for login and secret for pass. > > > 2008/8/21 Asim Riaz <ariaz78(a)gmail.com> > >> When I am adding a user i am getting java.lang.NullPointerException; >> >> also I cant login using admin username. >> >> On Thu, Aug 21, 2008 at 3:06 PM, Nuno Marques <yangsengopenser(a)gmail.com>wrote: >> >>> Hi again, >>> >>> I was looking at the database schema of openser and i noticed that the >>> subscriber table only has 12 columns till rpid. >>> Looking at the openser.sql script, it has more than 12 columns >>> (including the 'version' column...). >>> What is wrong? Should SerMyAdmin add this columns to the subscriber >>> table? >>> Is it lacking me something in the database subscribet table? >>> Any help with this? >>> >>> Thanks in advance, >>> >>> Nuno >>> >>> 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com> >>> >>> >>>> >>>> Hi Nuno >>>> >>>> i faced some problems your already facing and succeeded to solve them so >>>> i'd like to as you >>>> >>>> did you install SERweb tables ? >>>> >>>> if though the problems come from there especially the phplib_id field a >>>> duplication happened between it and the "id" filed so you have to remove >>>> SERweb tables and all thing will go smoothly without any errors >>>> >>>> p.s i recommend you to remove the whole database e.g openser and install >>>> it again using openser_mysql.sh script >>>> >>>> e.g /path/to/openser_mysql.sh create >>>> >>>> press yes for presence and that stuff but for SERweb no >>>> >>>> try and tell me what you get >>>> >>>> BR >>>> >>>> Muhammad >>>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

1 0

[Kamailio-Users] Help with SerMyAdmin
by Nuno Marques 21 Aug '08

21 Aug '08

Hi, I´m trying to install SerMyAdmin in my Openser 1.2.2 machine. I follow the steps that were on www.sermyadmin.org. I accomplished the installation task, but now when i try to do something on the DB (for example: create a new user) it gives me a lot of errors. My experience with Grails is null so i don't know were to start. Below i'll show the error messages. If anyone could give me an hand i'll appreciate. Thanks in advance, Nuno *Message:* could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] *Caused by:* org.springframework.dao.DataIntegrityViolationException: could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] *Class:* UserController *At Line:* [147] *Code Snippet:* Stack org.codehaus.groovy.runtime.InvokerInvocationException: org.springframework.dao.DataIntegrityViolationException: could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] 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.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.WebInterceptorFilter.doFilterInternal(WebInterceptorFilter.java:106) at org.jsecurity.web.servlet.SecurityContextFilter.doFilterInternal(SecurityContextFilter.java:93) at org.jsecurity.web.servlet.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:106) 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:595) Caused by: org.springframework.dao.DataIntegrityViolationException: could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] at UserController$_closure8.doCall(UserController.groovy:147) at UserController$_closure8.doCall(UserController.groovy) ... 29 more Caused by: org.hibernate.exception.ConstraintViolationException: could not insert: [User] at $Proxy3.saveOrUpdate(Unknown Source) ... 31 more Caused by: com.mysql.jdbc.exceptions.MySQLIntegrityConstraintViolationException: Duplicate entry '4000-192.168.137.1' for key 2 at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:1011) at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:956) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3491) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3423) at com.mysql.jdbc.MysqlIO.sendCommand(MysqlIO.java:1936) at com.mysql.jdbc.MysqlIO.sqlQueryDirect(MysqlIO.java:2060) at com.mysql.jdbc.ConnectionImpl.execSQL(ConnectionImpl.java:2542) at com.mysql.jdbc.PreparedStatement.executeInternal(PreparedStatement.java:1734) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:2019) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1937) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1922) at org.apache.tomcat.dbcp.dbcp.DelegatingPreparedStatement.executeUpdate(DelegatingPreparedStatement.java:102) ... 32 more

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[Kamailio-Users] Fwd: Help with SerMyAdmin
by Asim Riaz 21 Aug '08

21 Aug '08

---------- Forwarded message ---------- From: Asim Riaz <ariaz78(a)gmail.com> Date: Thu, Aug 21, 2008 at 2:49 PM Subject: Re: [Kamailio-Users] Help with SerMyAdmin To: Nuno Marques <yangsengopenser(a)gmail.com> my issue is same as Nano tried this command mysql -u root -p openser <openser.sql but still got ERROR 1054 (42S22) at line 1: Unknown column 'version' in 'field list' Regards Asim On Thu, Aug 21, 2008 at 12:26 PM, Nuno Marques <yangsengopenser(a)gmail.com>wrote: > Hi Muhammad, > > I droped the database and then created a new database as you told me... > Now i have to create all users again and an admin for sermyadmin... > But when i tried to run the mysql commands that are on > www.sermyadmin.org/openser/openser.sql they gave me an error: unknown > column 'version' in 'field list'. > Have any idea of what is going on? > By the way... do you know what happened to www.sermyadmin.org - it's seem > like it has been hacked or something... > > Many tahnks, > > Nuno > > > 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com> > > >> >> Hi Nuno >> >> i faced some problems your already facing and succeeded to solve them so >> i'd like to as you >> >> did you install SERweb tables ? >> >> if though the problems come from there especially the phplib_id field a >> duplication happened between it and the "id" filed so you have to remove >> SERweb tables and all thing will go smoothly without any errors >> >> p.s i recommend you to remove the whole database e.g openser and install >> it again using openser_mysql.sh script >> >> e.g /path/to/openser_mysql.sh create >> >> press yes for presence and that stuff but for SERweb no >> >> try and tell me what you get >> >> BR >> >> Muhammad >> > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >

2 1

[Kamailio-Users] Change project name in Surceforge?
by Iñaki Baz Castillo 21 Aug '08

21 Aug '08

Hi, is not time to change the project name in SourceForge? XD It appears still as "OpenSER SIP Server". -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 1

[Kamailio-Users] SIP Session timeouts
by Jason Penton 21 Aug '08

21 Aug '08

Hi All, I have a question about the best solution to ensure that SIP calls are terminated when the 2 parties drop off the network (wireless). Is there a std way of doing this (maybe using sst module) or should this be implemented using some specialised code? If not supported in a std manner in Kamailio, i thought about creating a child process within the DLG module that continuously checks all dialogs and the connected parties and pings them for connectivity. Should connectivity of any participant drop for more than a specific unit of time then the call is focefully torn down by openser. Any ideas? Cheers Jason

2 2

[Kamailio-Users] RATELIMIT link on modules docs page?
by Alex Balashov 21 Aug '08

21 Aug '08

Why does the RATELIMIT link on http://www.kamailio.org/docs/modules/1.4.x/ go to the documentation for the Perl module? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] Permissions module error !
by Nuno Marques 21 Aug '08

21 Aug '08

Hi, I tried to install on my Openser 1.2.2 the permissions module so i can use an asterisk gateway. I populated the trusted table with information about the gateway. When i tried to run OpenSer it gave me some errors. Can anyone tell me what could be the problem? Below are the erros and the data on trusted table. Thanks in advance, Nuno 0(3824) init_tcp: using epoll_lt as the io watch method (auto detected) 0(0) INFO: statistics manager successfully initialized 0(0) StateLess module - initializing 0(0) TM - initializing... 0(0) Maxfwd module- initializing 0(0) INFO:ul_init_locks: locks array size 512 0(0) TextOPS - initializing labvoipua:/etc/openser# 0(0) permissions - initializing 0(0) Default allow file (//etc/openser/permissions.allow) not found => empty rule set 0(0) Default deny file (//etc/openser/permissions.deny) not found => empty rule set 0(0) hash_table_insert(): Unknown protocol 0(0) ERROR: permissions: trusted_reload(): Hash table problem 0(0) init_trusted(): Reload of trusted table failed 0(0) Error while initializing allow_trusted function 0(0) init_mod(): Error while initializing module permissions ERROR: error while initializing modules 0(0) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy TRUSTED TABLE +----+----------------+-------+--------------+------+---------+ | id | src_ip | proto | from_pattern | tag | version | +----+----------------+-------+--------------+------+---------+ | 1 | 192.168.137.20 | Any | ^sip:.*$ | NULL | 3 | +----+----------------+-------+--------------+------+---------+

3 2

[Kamailio-Users] Speed dial
by salman farci 21 Aug '08

21 Aug '08

Hi, I have openser 1.2.2 installed with mysql as backend. I was able to run a simple call service after registering users in location table. Now that I'm trying implement a speed_dial, it returns "404 call not found" error at the Xlite panel after dialing a short dial number as per speed_dial table entry. But if I short dial my own short dial number it dials back to me(working) ! What might b the problem ? script file : ........... if (uri=~"sip:[0-9]{2}@.*") sd_lookup("speed_dial"); ............ speed_dial table entry : +----+------------+----------+----------------+--------------+----------------+-----------+----------+-----------------+ | id | username | domain | sd_username | sd_domain | new_uri | fname | lname | description | +----+------------+----------+----------------+--------------+----------------+-----------+----------+-----------------+ | 5 | 103 | | 33 | 10.1.2.214 | 103(a)10.1.2.214 | | | 4 | 102 | | 22 | 10.1.2.214 | 102(a)10.1.2.214 | | | 6 | 101 | | 11 | 10.1.2.214 | 101(a)10.1.2.214 | | | 9 | 105 | | 55 | 10.1.2.214 | 105(a)10.1.2.214 | | +----+------------+----------+----------------+--------------+----------------+----------+----------+-----------------+ can anyone clarify ? Thanx regards Farci

1 0

[Kamailio-Users] Second Logo
by Darren Sessions 20 Aug '08

20 Aug '08

Good work. I tinkered a bit and came up with something very similar. - Darren _____________________________ dmsessions(a)gmail.com http://www.darrensessions.com _____________________________ On Aug 20, 2008, at 4:52 PM, Ricky Gutierrez wrote: > logo with basic colors > > Regards > > rickygm > > ----- Original Message ---- > From: Johansson Olle E <oej(a)edvina.net> > To: Ricky Gutierrez <xserverlinux(a)yahoo.com> > Cc: devel(a)lists.kamailio.org > Sent: Wednesday, August 20, 2008 4:55:21 AM > Subject: Re: [Kamailio-Devel] logo > > Ricky, > I really like your logo. But I also see issue with using it in pure > black/white situations - no greyshades. > > Can you try to create such a version of it as well? > > This is often needed when producing t-shirts and other stuff, where an > outline is used. > > Thanks. > /O > > <kamailio-new.png>_______________________________________________ > Devel mailing list > Devel(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/devel

1 0

[Kamailio-Users] Commercial support?
by Robert McGilvray 20 Aug '08

20 Aug '08

With the departure of voice system from the project are there any commercial entities associated with the development team that offer support for Kamailio? Regards, 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. (W1)

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[Kamailio-Users] How to test and trace the RLS messages?
by mahesh.peddi＠info-spectrum.com 20 Aug '08

20 Aug '08

Hi All, I am working on openser 1.3.2 version with RLS as "integrated_xcap_server". Its running without erros. But my query is how to test, And please send me call flow between IMS components and RLS server. Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

2 1

Re: [Serusers] Content-Type: application/isup read encoded message
by Jiri Kuthan 19 Aug '08

19 Aug '08

I think the best learning experience is OOB http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/ser-oob.cfg?re… it is not really simple, on the contrary -- it is quite complete, but for that it gives you all kinds of examples that can be probably useful to you. -jiri michael(a)teldelnort.com wrote: > Hello jiri, > > Ok, > > Do yo have an idea of a simple example that will help me familiarize the > usage in ser.cfg? > > -michael > > -----Original Message----- > *From:* Jiri Kuthan [mailto:jiri@iptel.org] > *Sent:* Tuesday, August 19, 2008 03:44 PM > *To:* michael(a)teldelnort.com > *Subject:* Re: [Serusers] Content-Type: application/isup read > encoded message > > I don't think so for this particular one -- just try to use it as > other select operators. -jirimichael(a)teldelnort.com wrote:> Hello > Jiri,> > Thanks for the response.> > Is there any documentation on > how to use select operator with @context?> > > I am trying to filter > a search on the context for "II Digits:"> > -michael> > > > -----Original Message-----> *From:* Jiri Kuthan > [mailto:jiri@iptel.org]> *Sent:* Tuesday, August 19, 2008 02:09 AM> > *To:* michael(a)teldelnort.com> *Cc:* serusers(a)iptel.org> *Subject:* > Re: [Serusers] Content-Type: application/isup read> encoded message> > > I think the select operator, @content, should get you access to > the> body. There is however noparsing of SDP to my knowledge > (unless> someone corrects me, I remember it has been> > frequentlymentioned.)-jirimichael(a)teldelnort.com wrote:> Hello,> >> > How can I evaluate the Content-Type: multipart/mixed;boundary= with> > SER?> > Content-Type: application/isup; base=ansi92; version=ansi> > >> Is there a module that allows SER read the contents of the > encoded> message?> I am try to get the > > "Originating Line > Information> ----> II Digits:"> > from the message.> > -michael> > > >> > ------------------------------------------------------------------------>> > > _______________________________________________> Serusers > mailing> list> Serusers(a)lists.iptel.org>> > http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Kamailio-Users] Ratelimit outgoing SIP INVITE messages after dispatcher
by sergejf 19 Aug '08

19 Aug '08

Hello, We are connected to several VoIP carriers that impose a calls per second (CPS) rate limit on their SIP trunks. For example, on TRUNK1 we can only make call attempts at 10 CPS. Our calls are originally made by a dialer that does not support rate limiting, hence it always blasts calls as fast as it can. Upon receiving these SIP INVITE messages we determine what carrier to use and perform a mapping to a dispatcher group. Call attempts are then load-balanced between all SIP trunks from the chosen carrier. My question is whether it is possible to use the ratelimit module in conjunction with the dispatcher (or carrierroute) to ensure we don't overload the carrier SIP trunks? If this is possible, then I wonder how to configure ratelimit to support a 10 CPS call attempts rate. Thanks in advance for any advice on the topic! Serge -- View this message in context: http://www.nabble.com/Ratelimit-outgoing-SIP-INVITE-messages-after-dispatch… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 3

[Serusers] Content-Type: application/isup read encoded message
by michael＠teldelnort.com 19 Aug '08

19 Aug '08

Hello, How can I evaluate the Content-Type: multipart/mixed;boundary= with SER? Content-Type: application/isup; base=ansi92; version=ansi Is there a module that allows SER read the contents of the encoded message? I am try to get the "Originating Line Information ---- II Digits:" from the message. -michael

3 2

[Kamailio-Users] About Asterisk2Billing
by Ali Jawad 19 Aug '08

19 Aug '08

Hi All Was anyone able to get the accounting part of asterisk2billing to work with Openser ? This looks like a pretty mature solution to me. Thanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

5 4

[Kamailio-Users] Accounting for OPenser
by Ali Jawad 19 Aug '08

19 Aug '08

Hi I am looking for a accounting solution, that I can use with Openser without involving Asterisk. Has anyone tested something similar ? I need the solution to work for my prepaid customers, for example he has got 10$ once the credit is used. Deny any more call.s With Regards Ali Jawad System Administrator

3 2

[Kamailio-Users] SIPp dispatcher load testing failure with too many hops - why?
by sergejf 18 Aug '08

18 Aug '08

Hello, I'm attempting to load test a simple dispatcher script in OpenSER 1.3.x using SIPp (built-in UAC and UAS scenarios). My dispatcher.list only has one address 8.XX.XX.12 (a SIPp instance, UAS). SIPp on 8.XX.XX.10 sends the INVITE messages to 63.XXX.XXX.110. Everything goes well until the UAC sends ACK and BYE in quick sequence. OpenSER tries to t_relay() it to itself, which results in a "too many hops" error: 2008-08-18 11:16:56:708 1219072616.708649: Aborting call on unexpected message for Call-Id '1-3548(a)8.XX.XX.10': while expecting '200' (index 8), received 'SIP/2.0 483 To Many Hops Via: SIP/2.0/UDP 8.XX.XX.10:5061;branch=z9hG4bK-3548-1-7 From: sipp <sip:sipp@8.XX.XX.10:5061>;tag=3548SIPpTag001 To: sut <sip:service@63.XXX.XXX.110:5060>;tag=2148SIPpTag011 Call-ID: 1-3548(a)8.XX.XX.10 CSeq: 2 BYE Server: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 I have tried modifying the SIPp UAC script to re-use the Contact: header from the SIP 200 OK message in the R-URI of the SIP ACK and SIP BYE, to no avail. I have loaded the TM module but I suppose that since the SIP 200 OK is the final message of the transaction, SIP ACK and SIP BYE won't automatically be relayed to 8.XX.XX.12. Is this an OpenSER issue or a SIPp scripting issue? Any advice would be appreciated. Please find below some relevant lines from my openser.cfg: route{ xlog("TRACE:ROUTE: src($si:$sp) dst($Ri:$Rp) msg($mb)\n"); # initial checkings if ( !mf_process_maxfwd_header("10") ) { xlog("SCRIPT:ERROR: $rm (from $si:$sp) too many hops\n"); sl_send_reply("483","To Many Hops"); exit; }; if (method==CANCEL) { if (t_check_trans()) t_relay(); exit; } # routing if (has_totag()) { xlog("SCRIPT0:INFO: $rm RURI=[$ru] - routing to dst-uri [$du] cnt [$avp(i:273)] dst set [$avp(i:271)]\n"); loose_route(); if (method=="INVITE") record_route(); route(1); exit; } The SIP ACK and SIP BYE follow the route defined in the above if {} statement. The SIP INVITE follows the dispatcher route shown below: record_route(); # perform load balancing # set algorithm (4 = round robin) $avp(alg) = 4; # set the group $avp(grp) = 2; if (method=="INVITE") xlog("CALL_START: RURI=[$ru] group=[$avp(grp)] callid=$ci\n"); # do balancing if(!ds_select_dst("$avp(grp)", "$avp(alg)")) { xlog("CALL_DROP:INTERNAL: no destinations for [$ru] group=[$avp(grp)], callid=$ci\n"); sl_send_reply("404", "no dst"); exit; } if(avp_check("$avp(i:273)", "eq/i:1")) { if(!avp_pushto("$ruri","$avp(i:271)")) { xlog("CALL_DROP:INTERNAL: cannot push to ruri [$avp(i:271)], group=[$avp(grp)], callid=$ci\n"); sl_send_reply("500", "cannot get dst"); exit; } } else { # redundancy t_on_failure("1"); } xlog("SCRIPT:INFO: $rm RURI=[$ru] - routing to dst-uri [$du] cnt [$avp(i:273)] dst set [$avp(i:271)]\n"); # do forward route(1); exit; -- View this message in context: http://www.nabble.com/SIPp-dispatcher-load-testing-failure-with-too-many-ho… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 2

[Kamailio-Users] poll: project name
by Daniel-Constantin Mierla 18 Aug '08

18 Aug '08

Hello everybody, some people expressed inconveniences regarding the new name, so we would like to get your opinion about, therefore this is a poll that will get us to a decision. Here is the question: Do you like the name "Kamailio"? 1) Yes, I love it. Please keep it. 2) It is OK. 3) I do not mind the name at all 4) I do not like it 5) I hate the name and we should change it There is no need of explanations, as it is up to personal taste and decision. Just vote by option. Cheers, Daniel PS. for those interested... As short history about renaming story. On the 28th of March the board was informed about issues with SER part of project name. You can easily find who has the trademarks. On the 7th of April the board was informed about an agreement for the deadline as of September 30, 2008 to use old name. From there on, all old releases, documents, etc. can be referred by the old name, but nothing new. Other few discussions about what should/could be kept/changed followed and on 12th of May started the discussion about the new name with two main direction, something similar, to remind about old project or just something completely new, to be used as brand name. Couple of proposals and discussions regarding possible names followed up to 16th of July when was the deadline and the start of vote. Majority was reached on 17th but it was required to have opinions from all members of the board, last came from Bogdan, on the 28th of July. All discussions about migration were on the board mailing list. No change were done to openser.org configuration, just the new domain with kamailio in it was pointed to same server. Every step done was reported to board, following a schedule discussed there. Another discussion was about making the renaming and 1.4.0 release in same day. Being practically impossible to do all the work in one day, it was decided to do first the announcement about renaming. That happened on 28th when all agreed with the new name. As you can see, there was no 5 months of renaming duration. There was no change in openser.org DNS configuration undertaken with/without the knowledge of all board members. The new name domains were reserved when majority was achieved, in any case; dns entries for new name were pointed to old server shortly after booking the domains. -- Daniel-Constantin Mierla http://www.asipto.com

34 40

[Serusers] SIP and IP multicast
by Steven C. Blair 18 Aug '08

18 Aug '08

Has anyone had any experience using SIP via SER to setup an IP multicast session with a group of phones?

2 1

[Kamailio-Users] Looking for a VOIP Engineer
by Jonathan K. Creasy 18 Aug '08

18 Aug '08

Voxitas Inc. is a national VOIP provider based in St. Louis, MO. and we are currently looking for qualified applicants to fill Tier Three support positions and a VOIP Engineer position. The reason I am posting to this list is because we are looking to fill the VOIP Engineer position with someone who has OpenSER experience and is comfortable configuring and maintaining these systems. For details please contact our VP of Operations Zack Sargent. Zack Sargent VP Operations and Engineering Voxitas, Inc. 210 N. Tucker, St. Louis, MO. 63101 314-266-4000 x108 zsargent(a)voxitas.com Thank you! -Jonathan

1 0

[Kamailio-Users] PortaBilling
by Ali Jawad 18 Aug '08

18 Aug '08

Hi All Has anyone tried portabilling with Openser ? If so can you please provide a sample of your config file ? THanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

1 0

[Serusers] SER - Waiting Transaction
by Arun Kumar 18 Aug '08

18 Aug '08

Hi, I'm running SER 0.96 in my production for a long but there is strange problem , when I do serctl moni and looks for transaction it shows waiting transaction in () almost 400+ can some one tell me why there are so many transaction in waiting while I see processing only 10-20 transaction. Thanks arun

1 0

[Kamailio-Users] pick_branch failed (lowest==-1) for code 302--HELP!!
by Krunal Patel 18 Aug '08

18 Aug '08

Hi, Here is the scenario: -----------------------Invite--------------> <----------100 Trying---------------------- <----------302 Moved temporarily-------- -----------------------ACK--------------> Now I would like to route call to contact returned in 302 responce. But I am getting problem: Aug 17 18:10:41 [16667] DBG:core:parse_headers: flags=80 Aug 17 18:10:41 [16667] DBG:tm:t_should_relay_response: T_code=100, new_code=302 Aug 17 18:10:41 [16667] DBG:tm:t_pick_branch: picked branch -1, code 999 Aug 17 18:10:41 [16667] CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 302 Aug 17 18:10:41 [16667] DBG:tm:relay_reply: branch=0, save=0, relay=-1 Please help me to fix this. Thanking you Krunal Patel

1 1

[Serusers] Gentilini, Paul is out of the office.
by PGentilini＠intellicare.com 17 Aug '08

17 Aug '08

I will be out of the office starting 08/15/2008 and will not return until 08/18/2008. Please contact the CSC at 888-899-4227 if you need immediate assistance.

1 0

[Serusers] Acc module extra patches
by caio 16 Aug '08

16 Aug '08

Hello, Is there patches for acc extra params for ser version 0.9.7? I've found this link http://www.voice-system.ro/docs/acc-headers/ but do not know if is it for v0.9.x Thanks... -- caio

1 0

[Serusers] on Ubuntu Installation & Config
by Ameed Jamous 16 Aug '08

16 Aug '08

Hi all, Am trying to install & Configure SER on ubuntu 8.04. After i intsalled the SER file i can see this in my terminal: Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.1.104 [192.168.1.104]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.1.104 [192.168.1.104]:5060 Aliases: tcp: Ubuntu-AM.local:5060 tcp: localhost:5060 udp: Ubuntu-AM.local:5060 udp: localhost:5060 ok now how can i add a user to register to SER ? How can i monitor the logs and the prosses happening? i tried using serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config Even though i have line fifo=/tmp/ser_fifo in my config. Can anyone help with this? Regards, Ameed -- Best Regards, Ameed Jamous Project Manager/Carriers Relations. Direct Line: +962 777 835819. www.joodtelecomgroup.com

1 0

[Serusers] call forward no answer
by caio 15 Aug '08

15 Aug '08

Hi all, I can't make call forwarding on no answer works on my ser-0.9.7 installation. Here you can see a snippet of log, where the call falls on failure_route and then to route_1 where t_relay is called. Log at: http://rafb.net/p/xwTmn195.html The call is from sip:579951 to sip:579915(who has enabled the feature fwdnoanswer to sip:579916. But I do not know why the INVITE after the append_branch() is still to R-URI: sip:579915@.... After the relay then the call terminates with: Aug 15 16:13:25 [/usr/sbin/ser] SIP Reply (status):_ Aug 15 16:13:25 [/usr/sbin/ser] version: <SIP/2.0>_ Aug 15 16:13:25 [/usr/sbin/ser] status: <487>_ Aug 15 16:13:25 [/usr/sbin/ser] reason: <Request Terminated>_ I follow the example of ser guide.., but do not work.. I would appreciate any comment.. Regards, -- caiogf

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[Kamailio-Users] Question
by Edgar Guadamuz 15 Aug '08

15 Aug '08

Hello, I'm a newbie and I don't know the difference between opensips (where openser.org redirects) and kamailio, If someone can give me a light on this it would be great, thanks,

3 2

[Kamailio-Users] project foundation
by Daniel-Constantin Mierla 15 Aug '08

15 Aug '08

Hello everybody, I think it is the time to move forward with project management structure. During last days there were talks and discussions on the lists, the solution to make the project independent and long term trustable is to create a foundation. We made a draft of purpose and organization structure and we want to get your comments. http://www.kamailio.net/dokuwiki/doku.php/admin:foundation-draft For the statute we are looking at similar cases of foundations around open source projects. Considering the developer population and contributors, I find as good and reliable place to register the foundation to be Germany. There is still long way to go and a lot to discuss: * membership fees * how one can become a full member (associate member can be anyone that is friend of the project and pays the membership fee) * how a member loses the board or full member rights * how the board is supervised o when and how the reports of the board are presented o when and how the general assembly meeting takes place o how the reports are approved by full members * how to change the statute over time ... and perhaps more, but if we start and put some effort we can have it ready in reasonable time, avoiding trust and reliability issues in long term. Let's have debates here and contribute the results to the wiki. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] project status and my decisions
by Daniel-Constantin Mierla 15 Aug '08

15 Aug '08

Hello all, following the last events everybody perhaps wonders what is going on. I was surprised by the way everything happens and I am not sure yet how this resulted. Probably a bit of time is needed to clarify relations between developers themselves and community as well. There is not any official statement that a developer quited the support for this project, but I guess the trust between some is at lowest level possible. It is clear that we cannot say anything about a board right now that manages the project. From board discussions summary you may see that discussions were conducted to create a legal entity around the project, a non profit foundation. This might be hard to achieve in this status, but as a personal decision, I am not going in any other management group unless there is a legal framework backing up. Human relations prove to be pretty unpredictable and not conforming with initial statements. What is sure now - openser 1.3.x and 1.4.x will be maintained properly and I am volunteering to take care but hope others will join. It is really a good achievement of a sip server. Decision about future development will be based on community and developers feedback for this project. I am not going to promise anything by myself as I admit one person (to be more specific: myself) cannot take care of the entire code base -- I am just realistic, not willing to sell dreams. I think the people that still believe in the project can come with suggestions and alternatives of future development. What I can tell for sure, I am not going to join a fork of the original openser project. If something happened to make developers not working together, solution to solve nicely could have been found, in the worst, maybe just by splitting teams, but keeping old domain together just to show the new alternatives. "Why not joining a fork in this manner?"> Because I was one of the founders of this one and involved in development and management at top level. If I failed to build something I started, I had all the tools in my hands, why just creating something similar saying the old is crap but this new one will be the best? What is the difference? Why I didn't do it first time? Would I be able to fool someone? Does leaving my ship in secrecy on back door will bring more trust on me? I doubt and perhaps you, too. I am certainly assuming share of the failure in managing this project. I was there, in management, period. There is no single person responsible for mis-management, but it comes just easy for some to blame the others. But I am really confident we built outstanding sip server, all of us contributed with code, documentation, testing to an application that routes million and millions of minutes per day. There were accuses of bad things about project, I will argument in a separate message, this will get too long. So, I am asking: - how do you see a proper management structure for the future - feedback about future involvement, suggestions and alternative for going on Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

5 4

[Serusers] Register to SER with UDP
by Steven Wu 15 Aug '08

15 Aug '08

Hi All, Does SIP server at iptel.org support UDP register? When my UA rigister to iptel.org whith UDP, the 401 response always replied by SER at port 100xx(example 10028). But in my contact of rigister port assigned to 5060. So, my UA can't recive the 401 response, register failure. But use TCP everyting is OK? Can anyone give me help? Best regards, Steven Wu

2 1

[Kamailio-Users] KAMAILIO v1.4.0 is released
by Daniel-Constantin Mierla 15 Aug '08

15 Aug '08

KAMAILIO v1.4.0 is released August 7, 2008 Kamailio is the new name of the SIP server known so far as OpenSER. Due to trademark issues, the OpenSER project had to change the name. Officially, the new name was announced on July 28, 2008. Version 1.4.0 brings many new features since OpenSER v1.3.0 as well as improvements to existing components. An important share of effort was directed to code cleanup and stability. Moreover, new features in core and old modules were added while other 6 new modules were introduced. Download: You can download the tarball of the released sources at http://www.kamailio.net/pub/kamailio/latest/src/ Binary packages for several distributions you can find at http://www.kamailio.net/pub/kamailio/latest/bin/ http://www.kamailio.net/pub/kamailio/latest/packages/ Packages will be uploaded as soon as they are submitted. Documentation: * modules' documentation: http://www.kamailio.net/docs/modules/1.4.x/ * migration guide : http://www.kamailio.net/dokuwiki/doku.php/install:1.3.x-to-1.4.0 Summary of the new stuff in core: * overhauled DB API - better performance and safety, more common functionality integrated in the core * extensive cleanups in database drivers - integrate common functionality into the core for more stability and maintainability * SDP parser - provides an internal API for parsing SDP * fixup functions - provides an internal API for fixing module parameters * new free fixup functions - making safer to use some functions exported by modules in embedded languages such as Perl * extension to module interface - its now possible to use up to 6 parameters in module functions New modules: * db_oracle - ORACLE DB driver - connect to Oracle SQL Server * dialplan - Dialplan management - dialplan regular expression based translations * nat_traversal - NAT traversal helper * peering - RADIUS based peering allowing SIP providers to verify via a broker if source or destination request is from a trusted peer * ratelimit - SIP ttraffic shaping control and server load control based on your rules * userblacklist - User-specific blacklists New in existing modules: * acc module * it is possible to log accounting related output to a different log facility * cfgutils module * new FIFO functions get_config_hash and check_config_hash for config file checks * shared PV exported to configuration script * debugging helpers usable from config script for abort, shm_status and pkg_status * mysql module * enable timeouts on connect, read and write to prevent blocking on errors * database modules * renaming of all database modules, addition of the prefix 'db_', e.g. mysql is now db_mysql * carrierroute module * extension: much more flexibility in routing and database supported failure routing, improved internal structure * dialog module * internal API reworked for better flexibility * direction of the message provided via the internal API * new dialog callback types * new mi command: dlg_list_ctx * dispatcher module * ability to load destination URI flags from database at startup or reload * new algorithm to hash the content of a pseudo-variable * enum module * enum_fquery([...]) replaced by enum_pv_query([...]) * msilo module * notification system refurbished - message body, content-type, from address and contact header can be dynamically specified with pseudo-variables * pdt module * accepts now same destination domain bound to different prefixes * internal structures and logic optimized for memory and performance * lcr module * refactor module functions * pseudo-variable support * mediaproxy module * update to version 2.0 * better performance and scalability as packets are forward in kernel space New testing suite * a testing suite to ensure quality checking and regression tests reports has been developed, included in the repository and used during this development cycle Other important changes * modules' documentation has been migrated to DOCBOOK XML format * doxygen documentation extended and developer guide published * serweb tables have been removed * command line tool to manage dbtext in a SQL fashion These are not all, there is a big change log that gives more details: * http://www.kamailio.net/pub/kamailio/latest/ChangeLog More on http://www.kamailio.net Best regards, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] As a board member
by Adrian Georgescu 15 Aug '08

15 Aug '08

Hello, Because of sheer incompetence and abuse conduct of one of the recently adopted board members I decided to put my activities as a management board member of OpenSER/Kamailio project on hold. I will be able to resume my activities after the voluntary leave of Henning Westerholt. For those who do not know who I am, I was involved with OpenSER from Day 0 of its inception. Kind regards, Adrian

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[Serusers] SIP conference.
by Peter Cai 15 Aug '08

15 Aug '08

Hi all! I know H323 support conference but SIP says nothing about it. I want to make some study on this subject but there seems to be little information on Internet. Could some one give me some clues? Is there any RFC draft on this concept?

2 1

[Kamailio-Users] Voicemail implementation...
by Gerard A. Matthew 14 Aug '08

14 Aug '08

Ok, I'm trying to get the voicemail system working, however for some reason my Asterisk box is receiving a duplicated IP Address on the c line from OpenSER....below is the sample sip header. Is there anything in my openSER config file that would cause such a duplication? Rgds, Gerard INVITE sip:u5073@<DOMAIN>:5061 SIP/2.0 Record-Route: <sip:<DOMAIN_IP>;ftag=1d498377;lr=on> Via: SIP/2.0/UDP <DOMAIN_IP>;branch=z9hG4bKaed4.96adae34.1 Via: SIP/2.0/UDP 192.168.1.112:51006;received=<CLIENT_XIP>;branch=z9hG4bK-d8754z-0127430e81514e54-1---d8754z-;rport=51006 Max-Forwards: 69 Contact: <sip:0078@<CLIENT_XIP>:51006;nat=yes;nat=yes> To: "5073"<sip:5073@<DOMAIN>> From: "G Laptop"<sip:0078@<DOMAIN>>;tag=1d498377 Call-ID: NDZiOTUyNWFmYTQ3M2JlMjNiYzdmZjcyMGYyZDljZWE. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 481 P-hint: usrloc applied v=0 o=- 6 2 IN IP4 192.168.1.112 s=CounterPath X-Lite 3.0 *c=IN IP4 <XXX.XX.XXX.XXX><XXX.XX.XXX.XXX>* t=0 0 m=audio 3649036490 RTP/AVP 107 119 100 106 0 105 98 8 3 101 a=alt:1 1 : 4IoIRnGz iYQAZz1E 192.168.1.112 16720 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=nortpproxy:yes a=nortpproxy:yes

2 1

[Kamailio-Users] Translate name in RURI to number
by Ali Jawad 14 Aug '08

14 Aug '08

Hi all..my users are all registered with names...I want to forward their calls to my pstn gw..however those only take numerical ANI numbers. So I need to change the sip packet on the Openser server to a number (diff number per user) before forwarding that to the PSTN Gateway ..any ideas ? Would ENUM work ? THanks

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[Kamailio-Users] [OT] Any domain registrar offering SRV, NAPTR and AAAA?
by Iñaki Baz Castillo 14 Aug '08

14 Aug '08

Hi, first of all I'm sorry for the OffTopic. I'm looking for a domain registrar and nameserver that offers me SRV, NAPT and AAAA registers ni their domain control panel. Unfortunatelly in Spain *NO ONE* offers it. Is there any one "outside"? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 0

[Kamailio-Users] Kamailio & CDRTool 6.5.XX & MediaProxy 2.0
by luzango mfupe 14 Aug '08

14 Aug '08

Hi guys,I read the release note of MediaProxy 2.0 that to be able to have it fully functioning one needs CDRTool version 6.5.X and above i have been searching all over with no avail can someone point where it if found?? by the way i just found the old version 6.4.1 in the author's site wbr, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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