sr-users October 2008

sr-users@lists.kamailio.org

93 participants
169 discussions

[Kamailio-Users] OpenSER ActiveWatcher Table

by George Lee

Hi Expert, When a SUBSCRIBE message is received, OpenSER creates an entries in the ActiveWatcher table. This is used to generate NOTIFY message to watchers. The problem I have seen is that OpenSER keeps on creating new entries to the ActiveWatchers if the SIP UA deregister, register, deregister, register, etc. for several times. This triggers multiple NOTIFY messages to be sent to watchers. How does OpenSER remove entries in the ActiveWatcher table? It deosn't look like a SUBSCRIBE with expires=0 would delete an entry in the table. Thanks, George

15 years, 6 months

[Kamailio-Users] URI length

by VoIP Forums www.Go4Calls.com

Hi All, I was trying to make dialplan if dialed number length >= 8 it should go for PSTN call otherwise local call. I did the following but PSTN is working well if length is greater than 8 but when number comes for local call i am getting the error "Message to Big" if ($(rU{s.len}) >= 8) { rewritehostport("89.21.XXX.XX:5060"); route(1); exit; }else{ lookup("location"); route(1); exit; }; Thank psl suggest us how i can do the local call if it is < to 8 thank you _________________________________________________________________ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

15 years, 6 months

[Kamailio-Users] Record Route and proxy behind static NAT

by Zappasodi Daniele

Hello, I have my proxy behind a NAT (static NAT) with SIP clients that are both on my LAN and on public IP. UAC1 Private IP <--> <--> UAC3 PublicIP Proxy <--> NAT <--> WAN UAC2 Private IP <--> <--> UAC4 PublicIP I'm looking for the correct configuration of record route for this scenario. I looked for examples with proxy behind the NAT and I have always found a config section for the record route like this # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("<MY OPENSER PBULIC IP>:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; In this way the proxy uses "record_route_preset" when the caller is private (client_nat_test("3")) and "record_route" when the caller is public, without consider the callee. Consequently the proxy puts in the record route the public IP when the caller is private and the private IP address when the caller is public. I can't understand how it can work: when a client on an public ip calls a client on an public ip, they find in record route the private ip and they will try to send the SIP method (ACK,BYE, reINVITE ecc) to a private IP, so this messages will be lost. This is the beahaviour that I'm observing in my test. Maybe this configuration isn't adapted for my scenario or I miss something else (I have already added advertised_address/port) or I haven't correctly understand the instructions or the Record Route's usage. ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

15 years, 6 months

[Kamailio-Users] UUID column in usr_preferences - How unique does it need to be?

by Graham Wooden

Hi, By looking at the indexes set for the table usr_preferences, it appears that UUID is going to be unique against the attribute column. So if user A has a attribute of custom_parameter_Z and user B has the same attribute, then both UUIDs need to be different, right? UUID can really be anything right? Its set to VARCHAR(64), so heck - a MD5 checksum against say a epoch timestamp would be fine. I just want to confirm how unique each row (in respects to the UUID) it needs to be. Thanks, -graham

15 years, 6 months

[Kamailio-Users] Registrar performance

by Pascal Maugeri

Hi I would like to have some rough estimate of the registrar performance (with DB). If there is no figure available, does it make sense to use call performance benchmark like this one http://www.transnexus.com/White%20Papers/OpenSER_RTPproxy_test.htm and extrapolate to registrar performances ? eg. can I say that the SIP REGISTRAR is able to handle >600 registrations per second in the case of TransNexus testbed ? Regards, Pascal

15 years, 6 months

[Kamailio-Users] Pike module is not working!

by Arif-Uz-Zaman

Hi, I've configured OpenSER with Pike module but it's not working as expected (doesn't block IP according to the configured time). J Here is a simple configuration by considering the following: ---------------------------------------------------------------------------- --------------- modparam("pike", "sampling_time_unit", 60) #1 minute modparam("pike", "reqs_density_per_unit", 200) #200 requests modparam("pike", "remove_latency", 7200) #Block IP for 2 hours if (!pike_check_req()) xlog("L_CRIT","#########TOO MANY HITS FROM $si##########\n"); exit; } ---------------------------------------------------------------------------- ----------------- If the number of SIP messages from a single IP address to my SIP Proxy exceeds 200 per minute. Action: Block IP for 2 hours. I'm not sure whether I'm doing wrong or is there any problem with Pike module? Please help me with valuable feedback. Thanks in advance! ARIF

15 years, 6 months

[Kamailio-Users] Pike module is not working!

by Arif-Uz-Zaman

Hi, I've configured OpenSER with Pike module but it's not working as expected (doesn't block IP according to the configured time). :-) Here is a simple configuration by considering the following: ---------------------------------------------------------------------------- --------------- modparam("pike", "sampling_time_unit", 60) #1 minute modparam("pike", "reqs_density_per_unit", 200) #200 requests modparam("pike", "remove_latency", 7200) #Block IP for 2 hours if (!pike_check_req()) xlog("L_CRIT","#########TOO MANY HITS FROM $si##########\n"); exit; } ---------------------------------------------------------------------------- ----------------- If the number of SIP messages from a single IP address to my SIP Proxy exceeds 200 per minute. Action: Block IP for 2 hours. I'm not sure whether I'm doing wrong or is there any problem with Pike module? Please help me with valuable feedback. Thanks in advance! ARIF

15 years, 6 months

[Kamailio-Users] Re write any SIP Header Value!

by arif.zaman

Hi, Is it possible to update any SIP Header (i.e. User-Agent) value before it saving to “location” database? If yes, please help me with sample config. For example: User-Agent value always save as "arif" in location database. Thanks in advance, ARIF -- View this message in context: http://www.nabble.com/Rewrite-any-SIP-Header-Value%21-tp20067253p20067253.h… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

15 years, 6 months

[Kamailio-Users] (moved to kamailio)Why PUA_BLA generates a SUBSCRIBE "Event: dialog" that is never supported by OpenSIPS

by Klaus Darilion

Iñaki Baz Castillo wrote: > El Domingo, 19 de Octubre de 2008, Iñaki Baz Castillo escribió: >>> Hi, I using PUA_BLA and PUA_DIALOGINFO modules. >>> >>> I can't understand why when the proxy receives an INVITE it generates a >>> SUBSCRIBE to itselfs with: >>> Event: dialog >>> >>> Shouldn't it be: >>> Event: dialog;sla >>> ? > > Well, I've realized that it's because PUA_DIALOG info (Event: dialog). > > And more sorry! since it's just in Kamailio. The pua_dialoginfo modules sends SUBSCRIBE requests to itself? It can not send SUBSCRIBE - only PUBLISH. I guess the SUBSCRIBE is coming from pua_bla regards klaus

15 years, 6 months

[Kamailio-Users] Remove function

by michel freiha

Dear All, I have the following function: if($rU =~ "^[+].*") { route(8); exit; } I need to remove the + sign from the dilaed number before sending the call to route(8) Regards

15 years, 6 months

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sr-users October 2008