sr-users February 2007

sr-users@lists.kamailio.org

158 participants
285 discussions

[Users] Error Handling with Parallel Forking

by Chris Heiser

Hello gurus, I'm attempting to do some rather crazy parallel forking and have found a few issues I'm hoping to solve. I thought I'd solicit some advice from the experts. Some background: Each device registers with its own unique information. No two devices share the same credentials. Since a random user might want their number sent to multiple phones, and maybe even a cell phone, I've done the following: if(method=="INVITE" && !isflagset(5)) { if(avp_db_load("$ru/username","fork")) { t_on_branch("1"); avp_pushto("$ru/username","$avp(s:fork)/g"); }; }; Then the branch route looks like: branch_route[1] { setflag(5); route(3); } The idea here is the fork avp contains one or more possible destinations, but we don't store a complete URI here. The branch route sends us back through the normal routing logic to lookup local users on the system, and maybe route something to an external PSTN gateway, or another proxy. The problem: I haven't found a way to gracefully handle the event when one of the possible destinations is a device that hasn't registered. Say we allow the user to have a soft client, so we add the username to the list of fork avps. If the device isn't registered, I have 2 options: 1) Use a default route 2) Detect that I should know who this is, and error out Either way, an error is produced, but I don't see a good way of detecting if we should push the 404, or generate a 404 back to the client. The polite method would be: Are there other branches still going? If yes, just exit this branch If no, relay, or generate an error back to the client So my question is how to do this last part. Am I crazy for even thinking about doing this in OpenSER? BTW, I'm using v1.1.1 --Chris

17 years, 9 months

[Users] uri module question

by Klaus Darilion

Hi! I have problems understanding the uri module. As far as I understand I can map aliases to subscribers - e.g. my main identity is klaus@mydomain, but I also want to use nickname@cooldomain in From: header with a single SIP account (single subscriber) Thus I can use uri_db module and uri table to check if klaus@mydomain is allowed to use nickname@cooldomain. Thus, what do I enter in the uri table? The definition is: id | username | domain | uri_user | last_modified ----+----------+--------+----------+--------------- In which column do I have to put which values? thanks klaus -- Klaus Darilion nic.at

17 years, 9 months

[Serusers] Call from Asterisk to SER to a foreign subscriber

by karim basraoui

Hi, I've set up a small architecture in which an ASterisk proxy is connected to a SER proxy in two different domains. I have sip phones connected to both of them. when an asterisk subscriber "travels" to SER domain it tries to register to SER. I've configured SER to accept foreign subscribers and to REGISTER them to their initial domain by the command "t_replicate". Till now everything works good, the UA is registered and Asterisk is aware of its new location, it displays in peers list the UA and its new IP address and port. The problem is when i'm trying to call the foreign UA from an Asterisk subscriber, it doesn't work. i have a TRYING response but no call establishement. how can i overcome this problem please? -- BASRAOUI Karim www.basraouik.afrikart.net

17 years, 9 months

[Users] No logging. OpenSER on OpenBSD

by Ben Ferguson - backcountry.com

I can't get any logging out of openSER for some reason. I've tried using the log_facility setup via /etc/syslog.conf and i still cannot get any output. What am I missing? Here's the top of my openser.cfg file: debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 sip_warning=yes #log_facility=LOG_LOCAL7 I've tried un-commenting that last line to correspond to my /etc/syslog.conf: # don't log messages with LOG_LOCAL7 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL7 in /var/log/ser.log local7.* -/var/log/ser.log What am i doing wrong? Any ideas? Thanks, Ben

17 years, 9 months

[Serusers] authenticate INVITEs

by Octavarium

I have enabled mysql authentication of REGISTER messagess, but i need that only INVITE mesagges from devices properly registered in SER can go through, and that INVITE mesagges from devices NOT registered are dropped. How can i do this? do you have a sample script? Thanx a lot!

17 years, 9 months

[Users] How to reduce load of mediaproxy

by Alex

Hello, I'm sorry if this question was asked before, I googled and did not found references to it. I'm interested in reducing load of mediaproxy (using mediaproxy 1.8.0, openser-devel). Is it possible to connect callers not behind NAT directly and how configuration should look like? I want to direct traffic by this scheme: Caller Proxy Called NAT <- mediaproxy -> NAT NAT <- mediaproxy -> not NAT not NAT <- mediaproxy -> NAT not NAT <- direct connect -> not NAT I tried configuration something like this: route { if (client_nat_test(3)) { use_media_proxy(); t_on_reply("1"); } } onreply_route[1] { if (client_nat_test("2")) { use_media_proxy(); } } And it works fine then caller is behind NAT and called is not (NAT <-> not NAT) - traffic was directed via mediaproxy. But, then caller is not behind NAT and called is (not NAT <-> NAT), this configuration does not work. Any suggestions? Thanks.

17 years, 9 months

[Serusers] Polycom IP 601 help needed

by Steve Blair

I have been given the task of getting a Polycom IP 601 running SIP v2.0.1.0291 to register with our SER proxy and be able to interact with our Asterisk server for voice mail. The Asterisk server currently sends unsolicited NOTIFY messages to turn on/off the message waiting light. Most of the configuration is working however I cannot get the phone to register with SER if I set the voIpProt.server.1.address to the SRV name of our SIP domain. The only way the phone will register is if I set this parameter to the IP address of our SER server which is something we do not want to do. Is there any way to make this phone perform a SRV lookup for the server address? Thanks,Steve

17 years, 9 months

[Users] tr_inv_timer

by Ibrahim Hamouda

Is there a way to change the value of the timer from inside the script Ibrahim Hamouda Sphinx Information Technologies Inc. IT Canada International Tel: 403-668-6880 Fax: 403-229-0407 N.A.: 1-877-500-7664 Mobile: 403-714-3336

17 years, 9 months

[Users] CISCO & OpenSER & more ...

by Daniel-Constantin Mierla

Hello, OpenSER was chosen to be the SIP proxy of Cisco Service Node for Linksys One. That shows the very good level of stability and flexibility as well as wide set of capabilities in OpenSER: http://www.openser.org/index.php?option=com_content&task=view&id=90&Itemid=9 I would like to express my thanks to all supporting the project by any means (coding, testing, community support ...). Release 1.2.0 is knocking at the door -- it is a huge step forward since 1.1.0 adding lot of new stuff, plus a handful of improvements to existing features, just to remember some of news in 1.2.0: - SIMPLE Presence support - SNMP - PERL API - JAVA SIP Servlet Application Server (http://wesip.eu) - script variables - DNS failover - IP blacklists - XMPP IM gateway - XMLRPC - SIP session timers - secure federation peering - retransmission timer accuracy - automatic error handling So, here I wanted to get: help with testing 1.2.0 is very appreciated these days :-) For those looking forward to meet other OpenSER folks, the next two major events are: - Fosdem, Brussels, Belgium, this weekend - Von Spring, San Jose, USA, in March Cheers, Daniel

17 years, 9 months

[Serusers] Rewriting To: field

by Stéphane Alnet

Hello, I am doing interop testing with Level3, and need to be able to rewrite the To: field in requests to match their requirements. I can fake the rewrites inside route[], but although the packets sent out look like they use the new "To:", the header is never parsed again, so the internal structures of SER still hold the former (incorrect) value. When a request hit failure_route[] (for example when a CANCEL is sent when an INVITE times out) the original, incorrect To: header is then sent (and there's no way to modify the content of the CANCEL from within failure_route[]). Is there a standard way to alter the internally-stored value for To: (and From:, since I'm asking)? If not, could someone point me to the right place in the code to start adding this kind of feature? Thanks, Stéphane Alnet -- http://carrierclass.net/ http://www.linkedin.com/in/stephalnet

17 years, 9 months

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sr-users February 2007