sr-users February 2008

sr-users@lists.kamailio.org

146 participants
261 discussions

[Serusers] Problem with nat + forward on busy
by Martina 08 Feb '08

08 Feb '08

Hi, I have the following problem when using t_relay in a failure route: - next hop is correctly set but SIP message is forwaded to a wrong IP address (specifically to the address in request uri) failure_route[failure] { if (t_check_status("486|600")) { xlog("L_ERR", "LXLOG FAILURE_ROUTE: forward busy\n"); attr2uri("$tr.forward_busy"); $tu.uid=$tu.uid_busy; route(INBOUND); } } In route[INBOUND] several actions are performed including lookup_contacts and NAT_MANGLE Finally in the route[FORWARD] if (isflagset(FLAG_FAILUREROUTE)) { xlog("L_ERR","LXLOG: FORWARD: failure flag is set"); append_branch(); } # if this is an INVITE we might try another # (forwarding or voicemail) target after receiving an error if (method=="INVITE" && !$to_gw){ xlog("L_ERR","LXLOG: FORWARD: set_failure_route"); t_on_failure("FAILURE_ROUTE"); } # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP xlog("L_ERR","L_XLOG <%ru> request uri <%nh> <%ds>"); if (!t_relay()) { sl_reply_error(); xlog("L_ERR","LXLOG: FORWARD: !t_relay"); }; drop; The LOG IS: Feb 7 18:32:40 pc005 ser[11574]: LXLOG: FORWARD: set_failure_route Feb 7 18:32:40 pc005 ser[11574]: LXLOG: %ru = <sip:103@192.168.23.237> %nh = <sip:100.0.0.2:1037;dstip=100.0.0.1;dstport=5060> %ds = <Contact: <sip:103@192.168.23.237>;q=0, sip:103@192.168.23.237> but when I look at the sent packets using ethereal INVITE message is sent to 192.168.23.237 instead that 100.0.0.2. Do you have any suggestion about possible cause of this behaviour? Thanks in advance Martina

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[OpenSER-Users] Compilation problem on Debian Stable/SPARC architexture
by Schumann Sebastian 08 Feb '08

08 Feb '08

Dear all I am facing problems installing OpenSER 1.3 from branch on a Debian 4.0R2 running on a SUN UltraSPARC IIi server (sparc64). After retrieving all required files and executing # make mode=debug, the second file's compilation fails already: Compiling blacklist.c .... Error: Architecture mismatch on "membar". (Requires v9|v9a|v9b; requested architexture is sparclite.) Does anyone know what could be the reason for that. On i386 I never faced problems, and as OpenSER supports Solaris, it should also support Debian on sparc64 architecture or not? Thanks a lot in advance. Best regards Sebastian

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[OpenSER-Users] Tech prefix and CarrierRoute
by TCB 08 Feb '08

08 Feb '08

I need to append the callerid as tech prefix to all outgoing calls for complience with my billing Server. I'm using carrierroute model for routing. Can i do this or do i have to resort to do it from within the openser script -- TC

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[OpenSER-Users] How to set a fixed string as acc "extra" field?
by Iñaki Baz Castillo 08 Feb '08

08 Feb '08

Hi, imagine I need to log two extra fields in accounting: - Source-IP - Proxy-IP "Source-IP" will be dynamic containing the value of $si. "Proxy-IP" is a fixed string value = "88.88.88.88", but... how to log this value?: * modparam("acc", "radius_extra", "Source-IP=$si; Proxy-IP=88.88.88.88") Of course the above line fails since the IP string should be a text so should be between " ": * modparam("acc", "radius_extra", "Source-IP=$si; Proxy-IP="88.88.88.88"") This fails obviously. * modparam("acc", "radius_extra", "Source-IP=$si; Proxy-IP=\"88.88.88.88\"") This fails too. * modparam("acc", "radius_extra", "Source-IP=$si; Proxy-IP='88.88.88.88' ") This fails too. Is it impossible? any suggestion? PD: Yes, I know I can do a bit "dirty" trick: modparam("acc", "radius_extra", "Source-IP=$si; Proxy-IP=$var(proxy-ip)") ... $var(proxy-ip) = "88.88.88.88"; but I'd prefer not to do it. -- Iñaki Baz Castillo

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[OpenSER-Users] Local Call
by VoIP Forums www.Go4Calls.com 07 Feb '08

07 Feb '08

Hi friends, Could you please help me how i can make dialplan to call local each subscriber using some special prefix? I tried the following but it did not work. if (uri=~"sip:00[1-9][0-9]+@.*") { strip(2); rewritehostport("pstncarrier:5060"); route(1); exit; }else if (uri=~"sip:[1-9][0-9]+@.*") rewritehostport("pstncarrier:5060"); route(1); exit; }else if (uri=~"sip:01[1-9][0-9]+@.*") strip(2); if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; route(1); exit; }; The first two above condition is working fine but when we trying to call using 01 prefix + subscriber ID the call not going and the message comming 404 Not Found. Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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Re: [OpenSER-Users] How to handle messages received in WeSIP?
by Rajat Dudeja 07 Feb '08

07 Feb '08

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[Serusers] How to investigate/filter the sdp contents if Content type is text/plain
by Tomasz Zieleniewski 07 Feb '08

07 Feb '08

Hi, is it possible to check message contents if they contain some sting in case of content type text/plain?? will it work if I use search('some string'); from textops module? Regards Tomasz

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Re: [OpenSER-Users] newbie questions
by Doug McLetchie 07 Feb '08

07 Feb '08

Thanks for the reply Alex! I really appreciate it. I've been reading manuals & sip specs all day, and I think that I found my answer, and it really is an easy one. here's my own answer to my question, can you let me know if this sounds right? When a message is send from a user agent (on the public internet) to my gatweway & then to the proxy (on a private network), the gateway will add a via on to the message. that's how open SER (or any other proxy) can send a response message back to user agent on the public internet via the gateway. It's really the same thing as setting up a registrar, or any other kind of proxy/user agent that's going to communicate via a SBC of any kind. So, it that right? is that how the SIP via header is used? I understand the part about registrations. We're not currently doing that in our network, we're using all static IP addressing. I plan on using registrations in the future. the gateway that I'm using can act as a session border controller, but it can also act as a class 4/5 switch (TDM or IP). so it's some serious overkill for what I want to do here. :o) Thanks again for the reply, --Doug ----- Original Message ---- From: Alex Balashov <abalashov(a)evaristesys.com> To: Doug McLetchie <dougmcletchieatwork(a)yahoo.com> Cc: users(a)lists.openser.org Sent: Wednesday, February 6, 2008 1:23:51 AM Subject: Re: [OpenSER-Users] newbie questions Doug, Doug McLetchie wrote: > For inbound calls (calls coming from another carrier to my big expensive > gateway & destined for a specific PBX), I'd like my gateway to send the > call to the proxy, who will determine which PBX to send the call to, and > then send the call to to correct PBX via the Gateway. I don't > understand how to set up OpenSER to send the call via the Gateway. If I > provision the static ip address of the PBX in the proxy, won't it try to > send directly to that IP address instead of sending it to the Gateway? > I think that the feature that I'm looking for is something like an > outbound proxy, for the proxy. (does that make sense?) OpenSER can certainly do what you are trying to accomplish. SIP routing is done by URI, which consists of a "user" part and a "domain" part. The "user" part is the number (or alphanumeric identification, in the case of pure-VoIP peering) and the "domain" part is the IP "place" at which the "user" part is reachable. When you route a call to some URI, what you are really saying is, "Here, domain, you must know what to do with this 'user' part - i.e. have reachability information for it (a SIP contact bound from a registration, for example)." A proxy by itself isn't enough. If you need to reach these PBXs, you clearly need to know how to reach them. This requires a SIP registrar somewhere, so that the PBXs can register against it and say, "Here, you can reach me at such and such IP and port." Or, I suppose, you can define these contacts statically with a database interface from the proxy, in which case you don't need to use a registrar. I don't know what this Big Expensive Gateway is, but if it's something like a Session Border Controller, it should be able to forward SIP REGISTER requests to your proxy/registrar. Or do they register against the gateway? Either way, you can perform this resolution with OpenSER. The problem you *might* run into is sending the same logical call leg back to the Gateway. Depending on what it is, it may not like that and perceive a call routing loop, because the call that went through it to the proxy is the "same" call (in terms of SIP Call-ID, and other things that make up a logical call "leg") that is now being sent back around to it. This problem is often solved with the introduction of a back-to-back user agent which can re-originate a different call leg. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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[OpenSER-Users] Accounting through Mediaproxy
by Abdul 07 Feb '08

07 Feb '08

Good Day All, I am trying to get accrue AcctSessionTime of calls. I am using Openser+Asterisk but some time asterisk CDR are not accoutered, record missing + if some one call 15 min it will come 2 hrs in cdr. Is Mediaproxy will be best choice to get trust able AcctSessionTime? If Mediaproxy AcctSessionTime is trustable, i have to use ACC module of Openser or only dbaccounting configuration will be enough to store records direct from Mediaproxy into MySQL backend? Thank You Abdul Latef --------------------------------- Never miss a thing. Make Yahoo your homepage.

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[OpenSER-Users] How to handle messages received in WeSIP?
by Rajat Dudeja 06 Feb '08

06 Feb '08

Hi, I'm using SEAS module to interact with WeSIP Application Server. I'm using the version wesip-02-04-07-beta-0.1.4.tar.gz of WeSIP. I've managed to configure it and I'm also seeing that the messages are received by WeSIP. I'ld like to know: 1. Are there any special files to which WeSIP writes the received messages? 2. In case, some one wants to modify the received message then is there a possibilty to send the modified message back to OpenSER? 3. I have used the above tar file, to me it appears that this file only contains the binary form of WeSIP. So where can I get the complete source code of WeSIP? Many thanks, Rajat =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

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sr-users February 2008