sr-users February 2006

sr-users@lists.kamailio.org

189 participants
374 discussions

[Serusers] Serweb installation
by Ismael 02 Feb '06

02 Feb '06

Hello, where i specify the hostname server ? On the 8th step of the serweb installation, it is wrote : "Open the file http://<your-host><your-install-dir>/admin/index.php" What do you mean by <your-install-dir> ? For the moment, i tried with : http://192.168.1.71/var/libserweb/html/admin/index.php, but it doesn't function ?? Could you help me ? Thanks, Ismaël

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[Users] DNS CNAME query in openser?
by Greg Fausak 02 Feb '06

02 Feb '06

Can I do a cname or txt query in openser? I'm trying to create a cname for a enum soa. That is, I want to do: soa: e164.test.com cname: other.test.com -> e164.test.com Is there any way to accomplish this sort of thing through DNS? -g -- Greg Fausak greg(a)thursday.com

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[Serusers] SER & SEMS on separate machines
by Amos M. Nungu 02 Feb '06

02 Feb '06

Hello members, I have SER and SEMS on the same machine and it works fine. For performance reasons, I would like to them to be on separate machines. My query is the fact that SEMS uses the email address from the subscriber table in SER db. Do I need to have SER db on the machine where SEMS is running (means 2 identical dbs) or how can I achieve this? Regards, Amos.

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[Users] INVITE transaction timer vs. SIP Proceeding timer
by Arek Bekiersz 02 Feb '06

02 Feb '06

Dear all, I think it will be useful for everybody. I have discussion with SIP device manufacturer about expiration time of call attempt. When call is made from UA it sends INVITE with "Expires: 65". It is controlled by device's timer setting "ring_time_limit". This timer can be set to anything between 10 and 600 seconds, we set it to 65 seconds. Fine. However at the same time it starts a special timer, say "max_response_time_invite". It can be set to anything between 8 and 20 seconds. We set it to 20 sec. Fine so far. CASE 1: Imagine we call from UA to PSTN number, using one of VoIP->PSTN providers. We have more than one provider, so we setup SER with "fr_timer" value of, let's say 25 seconds and "fr_inv_timer" of 60. We prepare failure routes in case when those timers hits in PSTN call. When "fr_timer" hits, we simply reroute to another gateway. Great. SER forwards INVITE from UA to remote PSTN gateway and at the same time sends back "100 Trying" to UA. Fine. Imagine now that it is busy-hour and it takes PSTN provider 21 seconds to send back "180 Ringing" (or "183 Session Progress") and after 8 seconds more remote callee picks up ("200 OK"). Unfortunately device wil send CANCEL to SER, because it hasn't received "180 Ringing" or "183 Session Progress" within the "max_response_time_invite" setting of 20 seconds. It only received "100 Trying" at the beginning. This ruined any failure route attempts from SER, as call failed. CASE 2: Imagine now a second call. This time PSTN provider send "183 Session Progress" after 14 seconds and remote callee picked up after further 15 seconds. This time everything went good. UA did not hit "max_response_time_invite" timer - it waited patiently for whole 29 seconds for remote callee to pick up. I think that when UA receives provisional response (like "100 Trying") from SER it should stop timer "max_response_time_invite" and use 'ring_time_limit'. What device does now, it uses: * "max_response_time_invite" when "100 Trying" received * "ring_time_limit" when "180 Ringing" or "183 Session Progress" received I haven't really found explanation of this in RFC3261. When you look at SIP state machine on page 128, it is not explicitly said which timer controls behaviour of UA when in "Proceeding" state ("Proceeding" is entered oin receipt of ANY provisional response). QUESTION 1: what timer controls behaviour of UA when in "Proceeding" state? Especially regarding timeout. Is this still the 'timer B' started at the beginning of INVITE transaction? If it is 'timer B', why UA is CANCELing transaction after 20 seconds, if it informed by "Expires: 65" field in INVITE that it set 'timer B' to 65 seconds. QUESTION 2: What should I propose to manufacturer for maximum value for "max_response_time_invite"? -- Thanks, Arek Bekiersz

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Re: [Users] forward_tcp
by Klaus Darilion 02 Feb '06

02 Feb '06

Hi Ramin! Please always cc to the list Ramin Dousti wrote: > Thanks, Klaus. > > Here is the full picture: > > I receive a SIP URI "sip:user@dest" on a UDP socket. The only thing I need to do > is to proxy this session on a TCP socket to "dest". > > I fugured I can do: > forward_tcp(dest); > > But when I extract dest into an AVP, forwasrd_tcp does not work with an AVP as > its argument. > > avp_printf("s:dst", "$td"); > forward_tcp("s:dst"); > > it tries to resolve "s:dst" literally. forward_tcp does not support pseudo variables. try forward_tcp() without parameters. regards klaus >>If the destination is stored in an AVP, use avp_pushto >>(http://openser.org/docs/modules/1.0.x/avpops.html#AEN291) to push the >>new URI into the request URI. >> >>Then use t_relay (make sure the URI contains a transport=tcp parameter). >> >>If the destination is static, use t_relay_to_tcp(ip, port) >>http://openser.org/docs/modules/1.0.x/tm.html#AEN320 >> >>regards >>klaus >> >> >> >>Ramin Dousti wrote: >> >>>Hello, >>> >>>I'm trying to forward a SIP message that's been received by UDP, to >>>$td through TCP. but the following configuration doesn't work: >>> >>> send_tcp("$td", 5060 ); >>>or >>> send_tcp($td, 5060 ); >>>or >>> avp_printf("s:dst", "$td"); >>> send_tcp("s:dst", 5060 ); >>> >>>Could you please help? >>> >>>-- >>>Ramin >>> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> > > > -- > Ramin > >

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[Users] How to check Displayname in From Tag
by martin＠campus-merseburg.de 02 Feb '06

02 Feb '06

Hi, i want to change the "From:" tag with uac_replace_from when the displayname is "anonymous". with from_uri i can only check the uri. is there a way to check the displayname in the "From: " tag? for example: From: "Anonymous <sip:1234@blah.net>" i want to change it to: From: "Anonymous <sip:Anonymous@Anonymous>" Ciao

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[Serusers] Serweb
by Ismael 02 Feb '06

02 Feb '06

hello, i have followed serveb installation instructions, and know i am on the last step : "Open the file http://<your-host>/<your-install-dir>/admin/index.php" But i don't understand what i have to put on "your-host" ? Thanks, Ismaël

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[Serusers] Install and configure SER
by Ismael 02 Feb '06

02 Feb '06

Hello, I want to install SER on my debian. Could you explain me step by step how to install and configure ser server on my computer ?? How attach mysql to SER server ? Thanks to advance, Ismaël

1 0

[Serusers] Need help to install SER and Serweb
by Ismael 02 Feb '06

02 Feb '06

Hello, I installed Debian on my computer, and now i want to install ser proxy ans serweb. To install SER, i did : // installation of apache, mysql, and phpmyadmin apt-get install apache mysql-server phpmyadmin // installation of SER dpkg -i ser_0.9.6-0.1_i386.deb // restart of SER service /etc/init.d/ser restart To install serweb, i just did : tar -xvzf serweb-0.9.4.tar.gz What do i have to do now ?? Thanks for your help, Ismaël

2 1

[Serusers] TCP sockets and NATs
by Teemu Harju 02 Feb '06

02 Feb '06

Hi, This is a question that has been bugging me for some time now. If I use SER and the clients connected to it use only TCP, will there be a problem with the number of sockets that can be open at the same time, if these clients are behind NAT? I mean that for NAT bindings to stay open, each client has to keep open TCP connection to SER all the time after registration. What is the maximum number of open sockets that Linux can handle at the same time? I've changed the maximum number of open files to 65535 (ulimit -n 65535). Doesn't this also mean the maximum number of sockets? Is this the absolute maximum? Also, does this mean the maximum number of clients that can connect to my proxy? Regards, Teemu -- Teemu Harju http://www.teemuharju.net

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sr-users February 2006