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37565 discussions

Re: [SR-Users] Fail to add user account
by JinKevin 11 Jun '10

11 Jun '10

Sorry, pls ingore my previous email. I just forgot to remove some echo command added by me before the if statement. From: kevin.jzh(a)hotmail.com To: klaus.mailinglists(a)pernau.at; sr-users(a)lists.sip-router.org Subject: RE: [SR-Users] Fail to add user account Date: Fri, 11 Jun 2010 07:57:47 +0800 Have changed the kamctl from if [ $? -eq 0 ] ; then minfo "user '$1' already exists" exit 1 fi to if [ $? -ne 0 ] ; then minfo "user '$1' already exists" exit 1 fi Now I can add user successfully. Seems it's a bug in the script in kamailio 3.0.2 Thanks, Kevin > Date: Thu, 10 Jun 2010 16:48:13 +0200 > From: klaus.mailinglists(a)pernau.at > To: kevin.jzh(a)hotmail.com > Subject: Re: [SR-Users] Fail to add user account > > I have no idea. > > Try supplying a domain too: > > kamctl add user(a)yourdomain.com passwd > > regards > klaus > > Am 10.06.2010 16:32, schrieb JinKevin: > > Thanks Klaus, > > > > After change the LAST_LINE in the kamctlrc, the tail error is going, but > > it still says user "xxxx" already exists. > > > > # tail tool > > LAST_LINE="tail -1" > > > > I find the script below returns "false" which makes the kamctl exit. > > > > # params: user > > # output: false if exists, true otherwise > > is_user() { > > set_user $1 > > QUERY="select count(*) from $SUB_TABLE where \ > > $SUBSCRIBER_COLUMN='$OSERUSER' and $REALM_COLUMN='$OSERDOMAIN';" > > CNT=`$DBROCMD "$QUERY" "$DBRAWPARAMS" | $EGREP -v ERROR | $LAST_LINE` > > mdbg "is_user: user counter=$CNT" > > if [ "$CNT" = "0" ] ; then > > false > > else > > true > > fi > > } > > > > > > What could be the cause? > > > > Thanks, > > Kevin > > > > > > > Date: Thu, 10 Jun 2010 10:11:18 +0200 > > > From: klaus.mailinglists(a)pernau.at > > > To: kevin.jzh(a)hotmail.com > > > CC: sr-users(a)lists.sip-router.org > > > Subject: Re: [SR-Users] Fail to add user account > > > > > > > > > > > > Am 10.06.2010 03:12, schrieb JinKevin: > > > > AppSer01:root@/usr/local/kamailio-3.0/sbin$ kamctl add 2000 2000 > > > > usage: tail [+/-[n][lbc][f]] [file] > > > > tail [+/-[n][l][r|f]] [file] > > > > INFO: user '2000' already exists > > > > > > > > > between your kamctl command and the error message (user '2000' already > > > exists) is an error produced by the tail command. > > > > > > Looks like kamctl uses tail in a way which is not supported on your OS. > > > > > > regards > > > klaus > > > > ------------------------------------------------------------------------ > > 更多热辣资讯尽在新版MSN首页！立刻访问！ <http://cn.msn.com/> 聊天+搜索+邮箱想要轻松出游,手机MSN帮你搞定! 立刻下载！ _________________________________________________________________ 约会说不清地方？来试试微软地图最新msn互动功能！ http://ditu.live.com/?form=TL&swm=1

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Re: [SR-Users] Fail to add user account
by JinKevin 10 Jun '10

10 Jun '10

Have changed the kamctl from if [ $? -eq 0 ] ; then minfo "user '$1' already exists" exit 1 fi to if [ $? -ne 0 ] ; then minfo "user '$1' already exists" exit 1 fi Now I can add user successfully. Seems it's a bug in the script in kamailio 3.0.2 Thanks, Kevin > Date: Thu, 10 Jun 2010 16:48:13 +0200 > From: klaus.mailinglists(a)pernau.at > To: kevin.jzh(a)hotmail.com > Subject: Re: [SR-Users] Fail to add user account > > I have no idea. > > Try supplying a domain too: > > kamctl add user(a)yourdomain.com passwd > > regards > klaus > > Am 10.06.2010 16:32, schrieb JinKevin: > > Thanks Klaus, > > > > After change the LAST_LINE in the kamctlrc, the tail error is going, but > > it still says user "xxxx" already exists. > > > > # tail tool > > LAST_LINE="tail -1" > > > > I find the script below returns "false" which makes the kamctl exit. > > > > # params: user > > # output: false if exists, true otherwise > > is_user() { > > set_user $1 > > QUERY="select count(*) from $SUB_TABLE where \ > > $SUBSCRIBER_COLUMN='$OSERUSER' and $REALM_COLUMN='$OSERDOMAIN';" > > CNT=`$DBROCMD "$QUERY" "$DBRAWPARAMS" | $EGREP -v ERROR | $LAST_LINE` > > mdbg "is_user: user counter=$CNT" > > if [ "$CNT" = "0" ] ; then > > false > > else > > true > > fi > > } > > > > > > What could be the cause? > > > > Thanks, > > Kevin > > > > > > > Date: Thu, 10 Jun 2010 10:11:18 +0200 > > > From: klaus.mailinglists(a)pernau.at > > > To: kevin.jzh(a)hotmail.com > > > CC: sr-users(a)lists.sip-router.org > > > Subject: Re: [SR-Users] Fail to add user account > > > > > > > > > > > > Am 10.06.2010 03:12, schrieb JinKevin: > > > > AppSer01:root@/usr/local/kamailio-3.0/sbin$ kamctl add 2000 2000 > > > > usage: tail [+/-[n][lbc][f]] [file] > > > > tail [+/-[n][l][r|f]] [file] > > > > INFO: user '2000' already exists > > > > > > > > > between your kamctl command and the error message (user '2000' already > > > exists) is an error produced by the tail command. > > > > > > Looks like kamctl uses tail in a way which is not supported on your OS. > > > > > > regards > > > klaus > > > > ------------------------------------------------------------------------ > > 更多热辣资讯尽在新版MSN首页！立刻访问！ <http://cn.msn.com/> _________________________________________________________________ SkyDrive电子画册，带你领略精彩照片，分享“美”时“美”刻! http://www.windowslive.cn/campaigns/e-magazine/ngmchina/?a=c

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Match ip source of response into failure_route
by inge 10 Jun '10

10 Jun '10

Hi all, I try to do something like this on my failure_route[1] : if (t_check_status("503")) { if (ip_src==A.B.C.D) { ... } } The goal, is to prevent a 503 from a PSTN gateway and try another one for terminating the call. But, I realize that "src_ip" is the IP source of SIP Request. What could be a possible way to match IP source of the 503 responses ? Regards, Adrien L.

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Fail to add user account
by JinKevin 10 Jun '10

10 Jun '10

Hi All, Finally, the kamalio was installed on the Solaris 10 sparc and I can start it with the 'kamctl start', and it is running as showed below. However I cann't add user by using the kamctl, it always says the user already exists. And actually the susbciber table is still empty in mysql. Mysql is accessable with command "mysql -u openser -p ". What could be the issue? What are the log files I can check except the syslog? Didn't find an error in the syslog. ======= AppSer01:root@/usr/local/kamailio-3.0/sbin$ kamctl add 2000 2000 usage: tail [+/-[n][lbc][f]] [file] tail [+/-[n][l][r|f]] [file] INFO: user '2000' already exists AppSer01:root@/usr/local/kamailio-3.0/sbin$ kamctl ps Process:: ID=0 PID=8888 Type=attendant Process:: ID=1 PID=8890 Type=udp receiver child=0 sock=210.x.x.x:5060 Process:: ID=2 PID=8891 Type=udp receiver child=1 sock=210.x.x.x:5060 Process:: ID=3 PID=8892 Type=udp receiver child=2 sock=210.x.x.x:5060 Process:: ID=4 PID=8893 Type=udp receiver child=3 sock=210.x.x.x:5060 Process:: ID=5 PID=8894 Type=slow timer Process:: ID=6 PID=8895 Type=timer Process:: ID=7 PID=8896 Type=MI FIFO Process:: ID=8 PID=8897 Type=ctl handler Process:: ID=9 PID=8898 Type=tcp receiver child=0 Process:: ID=10 PID=8899 Type=tcp receiver child=1 Process:: ID=11 PID=8905 Type=tcp receiver child=2 Process:: ID=12 PID=8906 Type=tcp receiver child=3 Process:: ID=13 PID=8907 Type=tcp main process AppSer01:root@/usr/local/kamailio-3.0/sbin$ ======= Thanks, Kevin _________________________________________________________________ SkyDrive电子画册，带你领略精彩照片，分享“美”时“美”刻! http://www.windowslive.cn/campaigns/e-magazine/ngmchina/?a=c

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domain name
by Anders 10 Jun '10

10 Jun '10

Hi, I have a carrier who insists and have configured their system so they can receive INVITEs to their domain name only, and not the IP. I cannot configure the gw for anything but IPs, but thought that I might be able to replace the IP address in the cfg. So I tried with setting $dd = "domain.com"; but that's not working, - I guess $dd is read-only or something. Any ideas? Thanks!! //Anders

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Re: [SR-Users] 2 Asterisk servers re-inviting around kamailio even with t_relay being used?
by JR Richardson 09 Jun '10

09 Jun '10

> If that's really your complete Kamailio config, it's missing a number > of critical checks and handlers, such as loose_route(), which handles > sequential (in-dialog) requests differently from initial requests due > to the presence of a Route: header and a To header tag. > > You can't just stick cr_route() in the main route { ... } block > without those canonical checks and expect everything to work as desired. > > The reason that the proxy is not seeing the sequential requests > (reinvites, BYE, etc.) is because you don't have a Record-Route > header, which tells the endpoints to relay sequential requests through > the proxy on the network and transport level. Add this to your > initial request route prior to (cr_route())[1]: > > if(!is_method("REGISTER|OPTIONS|MESSAGE")) > record_route(); > > That should take care of it. > > -- Alex > > [1] And ideally, add quite a few other things as well. See the > stock configuration file for more insight. Thanks Alex. I am testing this in the lab so I don't have a production config loaded, I am really focused on the carrierroute module and trying to figure that out so I'm running bare bones using the module example. I looked into the RR functions and read in the CR module that it invokes the TM when necessary so I was thinking it would take care of keeping track of dialogs. The thing that threw me off was the handling of the scenario 1 session with sipp was ok but with asterisk scenario 2 it was not. So I loaded in the record_route function in the main routing block and it did take care of the complete session with the asterisk server, so that problem is solved. What I am running into now is the failure route. If the initial session can not compete to the fist carrierroute, the call switches over to the failure-route[1] which invokes carrierroute again with another domain. Then the receiving asterisk server retransmits the initial OK 6 times, so the transaction is lost or not held in kamailio state, then the whole session drops after a few more seconds. I think there is something going on with the append_branch function that is throwing me off. It is needed so the failure route completes but it's not handling the complete dialog or I could be totally off. Thanks. JR -- JR Richardson Engineering for the Masses

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Kamailio 3.0.1 and NAT
by Information 09 Jun '10

09 Jun '10

hi, I have configured kamailio 3.0.1 on fedora 12 and its working fine for ips on the network. But I am unable to get a working configuration for a natted client. If one client is natted and other is not, both can call each other through the kamailio server but only the natted client sends voice to the un-natted client while on tcpdump i found that the un-natted client is actually seeing the private ip of the natted client and trying to deliver packets there. So no sound reaches the natted client from the un-natted client. I have been searching for some days for some help in this regard but have not found anything i can use. help, regards. ***********************Confidentiality Notice*************************** The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Divinet or info(a)divinetaccess.com immediately and destroy all copies of this message and any attachments. Please print this message only if extremely necessary. Plant a tree today. Save the EARTH. ************************************************************************

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OT: phones which support out-of-dialog REFER
by Klaus Darilion 09 Jun '10

09 Jun '10

Hi! Does somebody know any SIP clients (hard/soft) which support out-of-dialog REFERs? thanks klaus

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2 Asterisk servers re-inviting around kamailio even with t_relay being used?
by JR Richardson 08 Jun '10

08 Jun '10

Hi All, I'm using kamailio with carrierroute to load balance calls to other servers. I have 2 testing scenarios set up: sipp><kamailio><asterisk server In the above scenario all SIP transactions, dialogs go through kamailio, INVITE, Trying, ACK, OK, BYE, OK Here is another one: phone><asterisk><kamailio><asterisk server In this scenario only the initial INVITE, ACK and OK go through kamailio, then the 2 asterisk servers finish the session directly to each other with ACK, BYE and OK In both asterisk servers, canreinvite=no is set in peers and general section in sip.conf. The kamailio cfg is using t_relay. No errors are coming up anywhere. Here is a pastebin of the 2 asterisk traces and the kamailio config. http://pastebin.com/Uk9qVhX2 Any ideas on why the call is not maintaining SIP session through the proxy? Thanks. JR -- JR Richardson Engineering for the Masses

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Strange behavior when rewriting "To" header from failure_route (kamailio - 3.0.2)
by Kosilov Fedor 08 Jun '10

08 Jun '10

piece of code: ........ failure_route[2] { if (!t_is_canceled()) { revert_uri(); sql_query("main", "select username from subscriber where alias='$tU' and alias_seq='2'", "alias_name"); sql_query("main", "select domain from subscriber where alias='$tU' and alias_seq='2'", "alias_domain"); subst_uri('/^sip:.*@.*$/sip:$dbr(alias_name=>[0,0])@$dbr(alias_domain=>[0,0])/'); subst('/^To:.*<sip:.*>/To: <sip:$dbr(alias_name=>[0,0])@$dbr(alias_domain=>[0,0])>\r/'); #<-----THIS ONE DOESN'T WORK PROPERLY sql_result_free("alias_name"); sql_result_free("alias_domain"); setflag(10); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_reply("404", "Not Found"); exit; case -2: exit; } } append_branch(); route(5); } } route[5] { if (!t_relay()) { sl_reply_error(); } exit; } And with ngrep i can see: # U 10.10.10.10.:5060 -> 10.10.10.15:53286 INVITE sip:ncm0000a@10.20.5.72:5060 SIP/2.0. Record-Route: <sip:10.10.10.10;lr=on;ftag=D1D33F74-16CA;nat=yes>. Via: SIP/2.0/UDP 10.10.10.10;branch=z9hG4bKc1ce.a89602e5.1. Via: SIP/2.0/UDP 10.20.5.10:5060;rport=57078;branch=z9hG4bK6F2A7E0. Remote-Party-ID: <sip:85559955500@10.20.5.10 <sip%3A85559955500(a)10.20.5.10> >;party=calling;screen=yes;privacy=off. From: <sip:9955500@10.20.5.10 <sip%3A9955500(a)10.20.5.10>>;tag=D1D33F74-16CA. To: <sip:ncm0000b@domain.com <sip%3Ancm0000b(a)domain.com>>.To: < sip:ncm0000a@domain.com <sip%3Ancm0000a(a)domain.com>>. <---------------DUPLICATE HEADERS Date: Tue, 08 Jun 2010 06:39:41 GMT. Call-ID: 73CF85D3-71FF11DF-B5C1D8D5-9276F617(a)10.20.5.10. Supported: 100rel,timer,resource-priority,replaces. Min-SE: 1800. Cisco-Guid: 1942740043-1912541663-2892365836-2242214896. User-Agent: Cisco-SIPGateway/IOS-12.x. Accept-Language: ru. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 69. Timestamp: 1275979181. Contact: <sip:85559955500@10.20.5.10:57078>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 401. ............... I'm trying to rewrite this one: To: <sip:ncm0000b@domain.com<sip%3Ancm0000b(a)domain.com> >. with this one: To: <sip:ncm0000a@domain.com<sip%3Ancm0000a(a)domain.com>>. I'm a newbie, so, please, show me where is my mistake. Sincerely yours, Kosilov Fedor.

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