sr-users July 2007

sr-users@lists.kamailio.org

163 participants
262 discussions

[OpenSER-Users] a=nortpproxy:yes in SDP and fix_nated_*
by liran tal 14 Aug '07

14 Aug '07

Hey everyone, When I'm doing sequential forking I'm seeing that for each gateway that doesn't answer and I move forward in the list of available gateways to forward the INVITE to, openser adds an "a=nortpproxy:yes" into the SDP so what happens is that the final gateway which is alive receives the sip header with multiple lines of "a=nortpproxy:yes" repeating themselves in the SDP and throws away the packet. >From my browsing in nathelper's code I see the only place that writes a "a=nortpproxy:yes" string to the SDP is the fix_nated_sdp function, and I'm actually not using it in my cfg but rather using fix_nated_register and fix_nated_contact which from what the README says are not handling the SDP part. And so I'm asking, where could it be coming from? And also, it would be easier if I can insert a piece of code that can do some regular expression miracles with the SIP header and remove any duplicate lines of the "nortpproxy" string. I'm running on openser 1.1.1 Thanks, Lir.

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[Serusers] problem with sercmd module
by dizel247＠aol.com 06 Aug '07

06 Aug '07

Hello, I have downloaded and intalled ser v2 from http://www.iptel.org/ser_2_0_0_release_candidate_1 Followd all the instructions. In Section Creating your first config. Run /usr/local/sbin/sercmd and type ctl.listen to see if SER is running and listening. Type help to see other commands available. I get an error. [root@localhost serweb-0.9.6]# /usr/local/sbin/ser -f /usr/local/etc/ser/ser.cfg Listening on ???????????? udp: 127.0.0.1 [127.0.0.1]:5060 ???????????? udp: 192.168.1.117 [192.168.1.117]:5060 ???????????? tcp: 127.0.0.1 [127.0.0.1]:5060 ???????????? tcp: 192.168.1.117 [192.168.1.117]:5060 Aliases: ???????????? tcp: localhost:5060 ???????????? tcp: localhost.localdomain:5060 ???????????? udp: localhost:5060 ???????????? udp: localhost.localdomain:5060 [root@localhost serweb-0.9.6]# /usr/local/sbin/sercmd ERROR: connect_unix_sock: connect(/tmp/ser_ctl): No such file or directory [2] can someone tell me why am i getting that error. Roman ________________________________________________________________________ AOL now offers free email to everyone. Find out more about what's free from AOL at AOL.com.

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[OpenSER-Users] loadmodule "perl.so"
by Richard Timsit 06 Aug '07

06 Aug '07

Hey Everyone, when i try to load module perl.so, i get this message : ERROR: load_module: could not open module </usr/local/lib/openser/modules/perl.so>: /usr/local/lib/openser/modules/perl.so: undefined symbol: boot_OpenSER In /usr/local/lib/openser/modules/ i have not only perl.so but also the lib/perl dir. openser-1.2.0-tls or openser-1.2.1-tls Any idea ? Thanks, -- Richard Timsit <Richard.Timsit(a)epfl.ch> EPFL

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[OpenSER-Users] OpenSER looping routes back to itself.
by Taylor Carpenter 02 Aug '07

02 Aug '07

Running OpenSER v1.1.1 with internal ip for sip client connections and ext ip to connect to sip providers. I am getting loops in my calls sometimes... They stop after reaching the max hops trying to send the BYE message to it self. I am wondering if it is the record- route that I see that includes the internal IP. The 483 error is U 2007/07/30 00:49:56.133157 192.168.18.22:5060 -> 192.168.18.22:5060 SIP/2.0 483 Too Many Hops. From: <sip:15126461516@192.168.18.1:5060>;tag=nu-481d-6c62. To: <sip:18065436468@192.168.18.22:5060>;tag=6738. Call-ID: 421ac0c6-1dd2-11b2-8093-df998124da20(a)192.168.18.91. CSeq: 11269143 BYE. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.7fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.6fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.5fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.4fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.3fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.2fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.1fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.0fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.fee832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.eee832a5.0. Via: SIP/2.0/UDP 192.168.18.91:5060;rport=5060. Server: OpenSer (1.1.1-notls (i386/linux)). Content-Length: 0. Any ideas?

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[OpenSER-Users] openser+radius
by ram 02 Aug '07

02 Aug '07

Hi iam installing openser SVN and Freeradius 1.16 followed the link as mentioned in document when i run radiusd -X iam getting below problem 1. rlm_sql (sql): Connected new DB handle, #4 2. Module: Instantiated sql (sql) 3. Module: Loaded Acct-Unique-Session-Id 4. acct_unique: key = "User-Name, Acct-Session-Id, NAS-IP-Address, Client-IP-Address, NAS-Port" 5. rlm_acct_unique: Cannot find attribute 'NAS-IP-Address' in dictionary 6. radiusd.conf[1159]: acct_unique: Module instantiation failed. 7. radiusd.conf[1753] Unknown module "acct_unique". 8. radiusd.conf[1747] Failed to parse preacct section. any suggestiong ram

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[OpenSER-Users] PRACK handling
by Zahid Mehmood 02 Aug '07

02 Aug '07

Hello everyone, What is the preferred way of handling PRACKS? simply relay them if they are in loose_route? should they be authenticated? What, if any, are the potential risks of not authenticating PRACKS? Thanks in advance. -- Zahid

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[OpenSER-Users] disable media/nathelper
by Aymeric Moizard 02 Aug '07

02 Aug '07

Hi, I'm facing an issue with openser. I checked the latest SVN code and it seems my issue is not solved even in the svn version. My UA is sending an SDP with 2 m lines: one for audio and one for video. In a re-INVITE, I want to disable video and I'm sending m=video 0 RTP/AVP.. My openser is currently modifying the port number for the video line even if it's 0 which means it's disabled... This behavior would of course happen even for audio lines that ones wants to disable. I was not able to find a clean fix in nathelper.c and hope to get some help for proposing one... I guess it would be something close to: if (oldport==0) skip altering ip/port for media and continue; but I don't want to insert any mem leak or lose a session... tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/

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[OpenSER-Users] how does OpenSER run on IPV6?
by null_neo＠sina.com 02 Aug '07

02 Aug '07

Dear sir: First,I am a Chinese,and my English is poor.^_^ We are testing kinds of applications on IPV6 including VOIP. And we use Linphone as our UA. If  I don't use domain name,how can I add a account? On IPV4,I add account " # openserctl add test test test@ipv4_address" .And in the file "openser.cfg" ,I directly write my SIP server ipv4_address . Can I directly replace "ipv4_address" by "ipv6_address"?  I has tried,and the resault is bad. I am very sorry that I am amateurish for VOIP ,because Our work is only to test. Thank you!                                                                                                 Li  Zhonglei ------------------------------------------------------------------- 吉贝克 ——商业智能的领军者( http://d1.sina.com.cn/sina/limeng3/mail_zhuiyu/2007/mail_zhuiyu_20070723.ht… ) =================================================================== 注册新浪2G免费邮箱（ http://mail.sina.com.cn/chooseMode.html ）

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[OpenSER-Users] about openSER and SIPp
by bozhang＠nec-labs.com 02 Aug '07

02 Aug '07

Hi there, I'm doing sipp performance measurements with openser 1.2. I run OpenSER on machine A, then run one SIPp on machine B, one SIPp on machine C. For the SIPp on machine C, I start it using UAS mode, ie., ./sipp -sn uas For the SIPp on machine B, if I start is using the following command: ./sipp -sn uac C -rsa A -m 1 -r 1 Everything is correct. The OpenSER can relay messages correctly. But if I start SIPp on machine B using the following command: ./sipp -sn uac -rsa A -m 1 -r 1 -d 5000 Then the SIPp client on machine B will crash because of receiving unexpected "ringing 180" message. Note: the only difference is I add a pause 5 seconds using "-d 5000" option. What I found is: the OpenSER server will keep sending "INVITE" to C during the '5 second pause', so C will keep reponding with "180 Ringing" to B. Does anybody know why OpenSER keeps sending "INVITE" to C while B only issues one "INVITE" request? BTW, the configuration file for OpenSER is very simple: # # $Id: openser.cfg 1676 2007-02-21 13:16:34Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=32 log_facility=LOG_LOCAL0 disable_tcp=yes disable_dns_blacklist=yes disable_dns_failover=yes # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # listen=udp:192.168.2.102:5060 # ------------------ module loading ---------------------------------- loadmodule "modules/tm/tm.so" modparam("tm", "wt_timer", 2) # ------------------------- request routing logic ------------------- # main routing logic route{ t_relay(); } Thanks a lot! -Bo Zhang

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[OpenSER-Users] Can't use variables as function parameter?
by Iñaki Baz Castillo 01 Aug '07

01 Aug '07

Hi, in URI as "ping(a)domain.org" I need to change URI to just "domain.org" (without the username@). But I have multidomain so can't rewrite uri directly and must use the requested URI domain ($rd). So I try: if (uri=~"sip:ping[@]+.*") { rewriteuri("sip:$rd"); xlog("L_INFO", "-- New URI = $ru\r\n"); ... but I get: -- New URI = sip:$rd so it seems that I cannot use variables for making function parameters, is it true? so then, how should I do it? maybe I need using AVP's for that? Thanks for all. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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sr-users July 2007