sr-users August 2006

sr-users@lists.kamailio.org

184 participants
361 discussions

[Users] 'socket' field in location db table
by Christian Schlatter 18 Aug '06

18 Aug '06

hi all, I'm running two openser instances accessing one central MySQL database using the usrloc db-only mode. Now I'm getting warnings in the log like WARNING:usrloc:get_all_db_ucontacts: non-local socket <udp:xxx.xxx.xxx.xxx:5060>...ignoring Although this is only a 'warning' level message, I'm a little concerned what effects this could have. Does anybody know what the 'socket' field in the location table is actually good for? thanks, Christian

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[Users] haze about the REGISTER method
by Jeremie Le Hen 17 Aug '06

17 Aug '06

Hi, first of all, excuse me for posting these questions twice, but the previous mail was bloated with too verbose sentences. I'll be straightforward then. 1) Why do all the configuration files I was given to see on the web call record_route() when the method is *not* a REGISTER ? 2) What is the harm of calling fix_nated_contact() in addition to fix_nated_register() in a REGISTER message ? The configuration file [1] carefully avoids doing this. [1] http://voip-info.org/wiki/view/OpenSER+And+RTPProxy Thank you very much. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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[Serusers] How to send Calls from Ser to Asterisk Voicemail?
by Ricardo Carvalho 17 Aug '06

17 Aug '06

How should I configure ser.cfg to enable Ser to send calls to Asterisk Voicemail when after ringing for some while, the phone isn't answered? Thanks, Ricardo.

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[Serusers] iptel.org problem
by Klaus Darilion 17 Aug '06

17 Aug '06

Hi! The SIP service of iptel.org does not work correctly - calls to other SIP domains do not work (404). regards klaus

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[Users] heders in route & failure_route blocks
by Jose Silva 17 Aug '06

17 Aug '06

>Message: 1 >Date: Tue, 01 Aug 2006 15:27:20 +0300 >From: OpenSer Users <openser-users(a)list.coretech.ro> >Subject: [Users] heders in route & failure_route blocks >To: users(a)openser.org >Message-ID: <44CF48A8.4020503(a)list.coretech.ro> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed >hello, >I want to add Remote-Party-ID header in route block and change it in >failure_route block. >the problem is that after the failure_route block the message sent out >has two Remote-Party-ID headers. >in failure_route I am using remove_hf("Remote-Party-ID") >scenario: >original invite has no Remote-Party-ID header >in route block I use: remove_hf("Remote-Party-ID"); append_rpid_hf("<", >">;party=calling;privacy=full") >in failure_route block I use: remove_hf("Remote-Party-ID"); >append_rpid_hf("<", ">;party=calling;privacy=off") >the invite sent out after failure_route block has two Remote-Party-ID >headers, one with privacy=full and another with privacy=off. >is there a way to correct the problem ? >from what I understand the failure_route block works on a copy of the >original message, but the original message has no Remote-Party-ID >header, so why the header from the route block is still visible ? >on the other hand if the header added from the route block is visible >why is it no seen by the remove_hf function ? >thanks, >Razvan Radu Hi I all so have a similar problem! Is this behavior normal? Is it possible to have this type of manipulation of the message on the failure-route? Thanks in advance, Jose Silva

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[Serusers] SIP MESSAGE method revisited.
by sip 17 Aug '06

17 Aug '06

Okay, Since I've been unable to pass SIP MESSAGEs back and forth through our server, but knowing full well that I once was able, I decided to take our configuration and remove everything in it that we've added over time (slowly regressing back to the beginning of our SVN code repository for the SER config) and see if I could tell what had changed so that I can no longer pass a SIP MESSAGE back and forth between clients. It didn't work. SO... I decided to take the hello-world config and try with it. No luck. What am I missing here? Is there some special handling that is needed to pass a SIP MESSAGE in SER 0.9.6? Is there a chance that something basic would have changed between 0.9.0 (our original server) and 0.9.6 that would have broken our ability to pass the MESSAGE method without additional tinkering? Sure, if I put in a block like: if(method=="MESSAGE") { sl_send_reply("200", "OK"); }; ... it's clear the message gets there and I receive the 200 OK, but of course it doesn't go anywhere after that. Without that block in there, it ends up getting a 513 Message too big response (which seems to be somewhat of a default for messages that don't end up getting handled in any other way that SER understands). I'm now completely at a loss. Pointers. Tips. Snide remarks. Anything helpful would be greatly appreciated. N.

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[Serusers] Restrict user for one login
by ram 17 Aug '06

17 Aug '06

Hi I have installed SER and testing But in my both sipura config same Extension both are registering, how can i restrict the extensions. If the user already loging, reject other user to login using same extension. any sample rules. Ram

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[Users] Problem with a NATing proxy
by Jeremie Le Hen 17 Aug '06

17 Aug '06

Hi, first of all, I am new to the SIP world, so please forgive me if I ask anything dumb. I am trying to set up a NAT-traversal SIP proxy, with the help of the nathelper module and rtpproxy. I only have access to the proxy and one softphone behind it. I am facing two problems: when someone in the outer world wants to call my softphone (which is behind the proxy), the proxy forwards the INVITE message to itself and recurses until the mf_process_maxfwd_header() statement stops further processing. I am not sure how to handle this. My guess is that I should use the registrar module and when the proxy receives an INVITE message, it should use registered() method in order to know where to forward the request ; but I don't know how to do that. Help would be welcome. My second problem is that when the peer hangs off the phone, I should receive a BYE message but I can't see it. Given I have no access on the other end, I guessed that the BYE message had been sent elsewhere. That's why I fixed the Contact header field. However, I still can't see actions coming from the outer phone. Any idea ? (I didn't attached any packet dump for now because I think it can be more misleading than informative because there are two proxies on the same machine: one that is a tunnel broker to encapsulate VoIP traffic in HTTP, and the other one is OpenSER. Basically, every SIP session is being tunneled in HTTP, then the datagram in unwrapped by the first proxy and then forwarded to the second one, which is OpenSER. Nonetheless I would be glad to provide any packet dump upon request.) My configuration file is attached. If you find any other mistakes or inconsistencies in this configuration file, please tell me. Best regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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[Serusers] Ser with Asterisk for Voicemail
by Ricardo Carvalho 17 Aug '06

17 Aug '06

Hi, I'm deploying a SER + Asterisk architecture, where SER is used to manage acc, users database and sip routing, and Asterisk is used for voicemail and PSTN gateway. How should I edit ser.cfg to make Ser redirect calls to Asterisk when the user doesn't answer his phone or when the user exists but it hasn't any phone registered? If I'm using also Asterisk for Gateway purpose, I guess all calls to be sent to PSTN, Ser redirects them to Asterisk, but it will try to find a mailbox for every number dialled, when it should distinguish calls for PSTN not to be sent to Voicemail. How can Asterisk distinguish between a call sent for voicemail and a call sent by ser to be redirected to PSTN? It depends on the configuration made in ser.cfg? Thanks, Ricardo.

1 0

[Users] External Script execution?
by Dominik Bay 17 Aug '06

17 Aug '06

Hi, how can i execute an external script to lookup if a prefix exists in a mysqltable? I want to have prefix(""); filled from a database. -- Mit freundlichen Grüßen / Kind regards Dominik Bay Cablesurf Technik

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sr-users August 2006