sr-users September 2009

sr-users@lists.kamailio.org

69 participants
105 discussions

by jh＠tutpro.com

if i have understood correctly, if uac_replace_from() is called in "auto" mode, encoded from uri is added into an extra param of r-r header when the request is forwarded towards uas. does the module check that reply to the request from uas really includes the added parameter in its r-r header or is correct operation of in-dialog requests at the mercy of the uas? -- juha

14 years, 5 months

[Kamailio-Users] kamctl online

by Chandrakant Solanki

hello I would like to print user@domainname while i execute kamctl online command... Any Idea -- Regards, Chandrakant Solanki

14 years, 6 months

[Kamailio-Users] Presence issue - pua dialoginfo

by Francisco Javier Lizarán Vilches

Hi all: I have a problem using Kamailio 1.5.2 as proxy&presence server. User A is subscribed to B's event dialog, and Kamailio handles this subscription. When a call arrives to pstn number C, the proxy resolves it via ENUM and sends the INVITE to B. The issue is that the PUBLISH request uri is C, because pua_dialoginfo module looks into To uri instead of Request uri, so A is not informed about B state. Has anybody any idea to work around this? Thanks: Francisco Javier Lizarán Vilches ---------------------------------------------- BARIK (Grupo Ormazabal) Departamento Técnico Tel.: +34 917 479 900 www.barik.es <http://www.barik.es/> ----------------------------------------------

14 years, 6 months

[Kamailio-Users] How can I access dialog attributes from within timeout route?

by Nathan Angelacos

Hi, In kamailio 1.5.2-notls, I'm trying to get the CallID of a dialog that has timed out: if (is_method( "INVITE" ) && !has_totag()) { $dlg_ctx(timeout_route) = 21; } ... route[21] { xlog ("L_WARN", "Dead dialog: $dlg(callid)"); exit; } However, syslog says: daemon.warn /usr/sbin/kamailio[6578]: Dead Dialog: <null> daemon.warn /usr/sbin/kamailio[6578]: WARNING:dialog:dlg_ontimeout: timeout for dlg with CallID \ '8b2eb5e3-20493c20-14e681d(a)192.168.1.1' and tags '87F489DA-922DF157' '2XrwB5OcEJ2zQvyiTUIh-l2Zd1wjL51M' dlg_handlers.c leads me to believe that the context is loaded before calling the route(21), and the dialog timeout handler is nice enough to report it in syslog just after my script fails. Am I missing something fundamental, or is this just not possible? Thanks!

14 years, 6 months

[Serusers] Can SER 2.0.0-rc1 pstn.cfg file actually work with BYE from PSTN?

by Frank Durda IV

Does SER 2.0.0-rc1/examples/pstn.cfg or the much more complete one in Chapter 8 of "SER Getting Started" actually work when it comes to a BYE comming from the PSTN gateway? So far, it appears these two cfg files don't cope with the called party on the PSTN hangs up first, and it isn't clear how it could be made to work in anything more complex than a one-SIP-call-source one pstn-gateway configuration. Let's say I have four clients from which a call can originate and arrive at SER, with the caller source IP addresses of "A", "B", "C" and "D". "C" sends the INVITE of a call that SER determined is destined for the PSTN gateway, which is at IP address "P". Here, the INVITE, 100 Trying, 180/183, 200 OK and ACK messages all go to the right places and look more or less correct. If "C" sends a BYE to end the call, this also works correctly, with SER forwarding that on to "P" and the 200 OK being returned. So far, so good. However, if the called party on the PSTN hangs up first, "P" sends SER a BYE and SER simply sends the BYE right back to "P", whom responds with a "481" back. The originating party at "C" never finds out that the call has ended, because SER didn't pass the BYE on to the calling device. The SIP phone stays in the "Connected" state. In my case, the calling sites A, B, C and D don't include a Record-Route of their own, but sometimes have a Via: header. In either case, SER still doesn't send a BYE or re-INVITE coming from P to the correct destination of A, B, C or D. I don't know how I could force SER to send the BYE or INVITE on to the right place, given four possible call-initiator IP addresses, of which only "C" would know about this sample call. I don't believe I can stick in an extra Record-Route when handling the original INVITE that has the call source IP address in it (in addition to a Record-Route of my own), simply because I don't believe SER will let me dynamically select which senders IP address should go in the forged Record-Route, or if doing so would really help the situation later when that trail of crumbs might be helpful. Also, even though the pstn.cfg sample in the book has lots of steps to check for fraud before processing a re-INVITE coming from the PSTN (and the footnotes say that this is what these checks are for), I don't see how SER would know the right place to send the re-INVITE that came from the PSTN once it the authentication checks were done. It looks like SER will always send it to "P" regardless of which direction it should be sent, which doesn't help much, and puts any call using Session-Expires: to failure, when 50% of that time (plus 32.5 seconds) has passed. So, is there with a corrected pstn.cfg out there, or better still, has anybody actually made these two situations work, which are: (1) The called party on the PSTN hangs-up first which causes a BYE to be sent to SER. SER forwards the BYE to the right caller (or whatever the so-called proxy is supposed to do), AND (2) the SIP-PSTN gateway device sends a re-INVITE to SER and it goes somewhere that will generate an OK reply If someone has made it work, can they show an example, rather than just say "it should be possible" or "works for me"? Such responses are encouraging, but not that helpful. Thanks in advance!

14 years, 6 months

asking odd mediaproxy,,

by Rakhmadhany Primananda

I've just setting SIP Express Router and Mediaproxy in 1 PC. Then, I'm setting in NAT Router that only activated for MASQUERADE like this: iptables -A POSTROUTING -s 172.17.2.0/24 -o eth1 -j MASQUERADE NB:I assume in network 10.100.0.0/24 (eth1) as public IP and 172.17.2.0/24as private IP. When I calling with x-lite and activated mediaproxy service, the connection running well with 2 direction (full duplex from 10.x.x.x to 172.x.x.x conversely).But,when I stopping mediaproxy service (for proving that voice packet if pass NAT will connect half duplex or maybe broken),apparently this connection still running well with 2 direction my question,, why this happened? are having trouble with mediaproxy or SIP Express Router? i'm confusing,, thanks very much,, :)

14 years, 6 months

[Serusers] Quick question on pattern matching in the uri variable

by Frank Durda IV

The following piece of code is supposed to change a uri of 18885551212 to 8885551212 and change a uri of +18885551212 to 8885551212. The 18885551212 situation works fine, but in the +1 situation, the "if" test fails for some reason the code below falls to the "nothing to strip" log message. What am I missing? route[STRIP_LEADING_ONE] { if (method == "INVITE") { if (uri =~ "^sip:1.*" ) { #Leading 1? xlog("L_ERR", "Strip leading '1' - From URI:<%fu> From Tag:<%ft> Request's R-URI:<%ru> Before treatment To URI:<%tu>\n"); strip(1); } else if (uri =~ "^sip:+1.*" ) { #Leading +1? xlog("L_ERR", "Strip leading '+1' - From URI:<%fu> From Tag:<%ft> Request's R-URI:<%ru> Before treatment To URI:<%tu>\n"); strip(2); } else { xlog("L_ERR", "Nothing to Strip - From URI:<%fu> >From Tag:<%ft> Request's R-URI:<%ru> Before treatment To URI:<%tu>\n"); } } } Thanks in advance!

14 years, 6 months

Re: [Kamailio-Users] content about rtpproxy

by zhangchao00001

Hello everyone, i meet a problem when intergrating rtpproxy with kamailio. The two software start well, can communicate with each other, however, the rtpproxy always report "can't create listener" error. Dose anyone meet the same problem?

14 years, 6 months

Re: [Kamailio-Users] kamailio 1.4.3 slow

by Juan Perez

we finally found what was causing the prob, at least the prob is gone after doing this: In the syslog config file of the server where kamailio was working fine we found this entry: local0.none /var/log/messages we added it to the server with probs and problem solved. Since we were using the facility 0 to log the messages from kamailio apparently we need to add that entry listed above as well to make the logging work better. this is how we setup kamailio's logging, using the example from the web: # log messages with LOG_LOCAL0 in /var/log/ser.log local0.* -/var/log/proxy.log I do not know the reason why but it fixed our issue. I wanted to share with you. cheers and thank you a lot for your assistance jp ________________________________ From: Juan Perez <jperezsip2008(a)yahoo.com> To: Henning Westerholt <henning.westerholt(a)1und1.de>; users(a)lists.kamailio.org Cc: myoung(a)redmonsters.net Sent: Thursday, September 17, 2009 8:49:00 AM Subject: Re: [Kamailio-Users] kamailio 1.4.3 slow than you a lot Henning for your suggestions, I will try to do all of that but the lab stuff because we already have a lab setup but it works superb too, only on the new machine is when we have found the prob. in all our other scenarios kamailio runs super fast, processing a lot of calls per seconds and the cpu is 99-100 % idle. fortunately we can do this in the same machine because it has an amount of traffic controllable by us so at any given time we can take it out or put more or use sipp, I will post the results as soon as I have something new, thank you guys a lot again cheers jp ________________________________ From: Henning Westerholt <henning.westerholt(a)1und1.de> To: users(a)lists.kamailio.org Cc: Juan Perez <jperezsip2008(a)yahoo.com>; myoung(a)redmonsters.net Sent: Thursday, September 17, 2009 6:05:07 AM Subject: Re: [Kamailio-Users] kamailio 1.4.3 slow On Mittwoch, 16. September 2009, Juan Perez wrote: > In another scenario in our secondary site I am using another Dell Poweredge > 1950, with dual quad core almost at the same speed and 64 bits , same > version of CentOS and it is working like a charm. A lot of calls and CPU > idle almost 100% all the time. That is we decided to buy a more powerful > server to have a more robust platform in our main site but we haven't been > able to move forward with this because of the bad performance. Any other > ideas guys ? Hello Juan, as this looks more like a hardware issue to me i'd try to find issue that causes this particular server to be that slow. Somewhere it must spend this time. Some ideas: - Start kamailio with only one worker children, and then use strace to see what its doing when its hangs - enable debugging (only for a few seconds if you've production load on the system) and then look to the timestamps where its spend the time - use the benchmark module as Daniel already suggested - try to replicate the setup in a lab environment and then use a load tester (e.g. sipp) to give some traffic to the system. then change certain parts of the configuration and setup to find the culprit Cheers, Henning

14 years, 6 months

[Kamailio-Users] Siremis JSON error

by Luis Guaman

Hi, When we tried to run a Siremis Chart got the following error message: Open Flash Chart JSON Parse Error [Syntax Error] Error at character 0, line 1: 0: <body style='font-family:Arial; font-size:12px; background-color:#FCFCFC;'>Error m Some help?? Atentamente, Luis Guamán Desarrollo de Negocios República 396 y Diego de Almagro Ph:+593.2.2526585 Cel:+593.9.4264354 Quito - Ecuador logoReducido

14 years, 6 months

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sr-users September 2009