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37464 discussions

[Users] benefits of ser and openser
by Rao Ramaratnamma 27 Nov '06

27 Nov '06

This was excellent comment, I am surprised people did not answer too much. I thought too that if we take ser codebase, openser documentation, all contributed modules, and ser new configuration, and openser release cycle, we will obtain best of best to our benefit. Are people confused by subject line and is this why there is so little comments for this important discussion? I changed the subject line. Hopefuly people who can assess how difficult taking best-of-best would be will answer for me? rr ----- Original Message ---- From: Andrei Pelinescu-Onciul <andrei(a)iptel.org> To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Cc: Jiri Kuthan <jiri(a)iptel.org>; serusers(a)iptel.org; Kim Il <kim_il_s(a)yahoo.com>; users(a)openser.org Sent: Wednesday, November 22, 2006 6:59:04 PM Subject: Re: [Serusers] Re: Fw: [Users] TM : retransmission timers On Nov 22, 2006 at 12:39, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: [...] > > I'm being told that some other personal affiliated with > >the same company more or less copied-n-pasted TCP code from the allegibly > >discontinued SER to OpenSER. > > That's how open source works. I also copied lots of TLS extensions from > ser to openser, and even extended it. You can also copy my extensions > back to ser ... I would love to see it there :-) I'm sure Jiri has no problem with copying code from ser to openser. We will do the same with some openser modules and we already did with some fixes. The problem here was that it was claimed that ser was dead but in the same time new code commited in ser was added in openser (so the openser history writer was not very well intended). Speaking of copying/porting ser between ser and openser I have a few requests: 1. If someone sends a fix for code taken from ser, which was not modified in openser, please could you also bounce the message to the ser maintainer of that piece of code? If you don't to whom, just send it to me and I will make sure it reaches the right person. I'm asking this because my favourite pass-time is not to scan openser commitlogs for possible fixes to my code (e.g.:http://openser.cvs.sourceforge.net/openser/sip-server/fastlock.h?r1=1.…) and this will help me at a minimum effort for the openser developer part. Of course I think this is true not only for me, but also for other ser and openser developers and I (and I'm sure anybody else on the ser team) will return the favour. 2. Please give proper credits for the code you port from ser. I've saw several times things like: Foo sees something was fixed in ser or a new small feature was added and sends an email to the openser list (specifying that the change/patch comes form ser). The openser developers take the change, but the commit message says something like: fix for XyZ, credits go to Foo (ser is not mentioned at all). It would not only be polite, but it will also help us to filter more easily through the openser fixes. Comming back to the openser - ser performance / features discussion, I think the important part here is how we can improve both sers (and you should take the test results as a bug report and maybe ask yourself how did this go through testing, even old sers were faster then 4000 cps). If you think only of the users, then a combination of ser & openser will be the best version. While we disagree (probably) in many respects, I think there are at least a few clear advantages each version has. For example openser has very impressive documentation (at this time ser is very far away from this point of view) and lots of new modules. ser has also its share of new modules (though I think not so many) and a better/improved "infrastructure" (core, tm and lots of the "base" modules - we are actively improving them). ser code tends to also be more tested and stable (our release policies differ wildly). I've tried looking at openser core & tm commits and I haven't seen any significant changes (and this is not a flamewar attempt). Now the question is why doesn't openser take the core, tm and a big part of the modules from ser completely and concentrate then in adding new modules (which you seem to do anyway)? Does it make sense to re-invent the wheel ? Everybody (me too) suffers at least a little from the not-invented-here syndrome, but if you think of the users and at how much of your time it will save, it will make a lot of sense, even if some compromises would be necessary. I think the advantages will far outweigh the problems. Just think about it, we can concentrate each other on our favourite stuff and we can benefit much more easily from each other's work. Andrei _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com

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[Users] siptrace and traced_user
by Mark Price 27 Nov '06

27 Nov '06

The sip_trace table has a field called traced_user. This is undocumented, and I am finding that this field is empty in the table. Is this currently unused, or is there something I need to turn on? Mark

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[Users] Radius accounting: what happens if radius server is down?
by kjcsb 27 Nov '06

27 Nov '06

I currently have (open)SER configured to write accounting records to a radius server. However there is a possibility that the Radius server may become unavailable. What is the recommended way to deal with this scenario? Is there a way to cache accounting records? Or is it best to write records to both flatfile and the radius server? Something like this: modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 3) Any suggestions would be appreciated. Regards Cameron

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[Users] loose_route() function
by Gregoire 27 Nov '06

27 Nov '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am wondering where I can find more information about the loose_route() function. On Wiki, I can see that it is RFC 3261 compliant and understand what it's the difference between loose routing and strict routing, but in my conf. file I have a test like: if(loose_route()){...} and I 'd like to know what kind of value this function can return or when can it return something else than true? Does someone have some information about that? Thanks Greg -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFZv/iI8gmGeMTr0sRAgg8AKCA9CGb6hI9FEhdE8BMm5nlnXDKkACgjZBA NFrRWixi2BtEE6e2YrAfg9g= =ucXk -----END PGP SIGNATURE-----

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[Serusers] Contact in 200 OK response to REGISTER
by Stefan Sayer 27 Nov '06

27 Nov '06

Hi, registrar at iptel.org (0.10.99-iptelorg/tekelec-SOPv2) and ser 0.9.6 as well do not return a contact header in 200 response to REGISTER (see trace below), but in rfc (10.3) I read it's a MUST 8. The registrar returns a 200 (OK) response. The response MUST contain Contact header field values enumerating all current bindings. Each Contact value MUST feature an "expires" parameter indicating its expiration interval chosen by the registrar. The response SHOULD include a Date header field. Am I missing something (apart from the contact header) or is my register message not correct? Stefan U 192.168.5.101:5070 -> 213.192.59.75:5060 REGISTER sip:iptel.org SIP/2.0. Via: SIP/2.0/UDP 192.168.5.101:5070;branch=z9hG4bK15f6.4d8faeb.0. To: Stefan <sip:sayer@iptel.org>. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 10 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. Content-Length: 0. User-Agent: Sip EXpress router(0.9.6-sems (i386/linux)). Contact: <sip:sayer@192.168.5.101:5070>. Expires: 0. . # U 213.192.59.75:5060 -> 192.168.5.101:5070 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.10.144:5070;rport=5070;received=91.64.139.240;branch=z9hG4bK15f6.4d8faeb.0. To: Stefan <sip:sayer@iptel.org>;tag=17da4c1f77cf571798ae2c71e6d4b5c0.4003. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 10 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. P-Nat: Yes. WWW-Authenticate: Digest realm="iptel.org", nonce="456af887c3eac5c30c87bab4268389ee5e796372". Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)). Content-Length: 0. Warning: 392 213.192.59.77:5070 "Noisy feedback tells: pid=4141 req_src_ip=213.192.59.75 req_src_port=5060 in_uri=sip:iptel.org out_uri=sip:iptel.org via_cnt==2". . # U 192.168.5.101:5070 -> 213.192.59.75:5060 REGISTER sip:iptel.org SIP/2.0. Via: SIP/2.0/UDP 192.168.5.101:5070;branch=z9hG4bK25f6.dae56b14.0. To: Stefan <sip:sayer@iptel.org>. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 11 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. Content-Length: 0. User-Agent: Sip EXpress router(0.9.6-sems (i386/linux)). Contact: <sip:sayer@192.168.5.101:5070>. Expires: 0. Authorization: Digest username="sayer", realm="iptel.org", nonce="456af887c3eac5c30c87bab4268389ee5e796372", uri="sip:iptel.org", response="880ff39073b5eeb3cda68352b78de368", algorithm="MD5". . # U 213.192.59.75:5060 -> 192.168.5.101:5070 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.5.101:5070;rport=5070;received=91.64.139.240;branch=z9hG4bK25f6.dae56b14.0. To: Stefan <sip:sayer@iptel.org>;tag=17da4c1f77cf571798ae2c71e6d4b5c0.4003. From: Stefan <sip:sayer@iptel.org>;tag=0E520893-456AF6AD0009055B-B7963B90. CSeq: 11 REGISTER. Call-ID: 0D1543AB-456AF6AD00090738-B7C66B90(a)192.168.5.101. P-Nat: Yes. Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)). Content-Length: 0. Warning: 392 213.192.59.77:5070 "Noisy feedback tells: pid=4140 req_src_ip=213.192.59.75 req_src_port=5060 in_uri=sip:iptel.org out_uri=sip:iptel.org via_cnt==2". . -- Stefan Sayer Media Services Development iptego GmbH Am Borsigturm 40 13507 Berlin Germany stefan.sayer(a)iptego.de www.iptego.de

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[Users] avp_subst re problem
by Andreas Heise 27 Nov '06

27 Nov '06

Hello, I tried to use avp_subst to extract the part <......> of a Remote-Party-ID $avp(s:merker)= Remote-Party-ID: "test"<sip:12345@localhost>;party=calling;id-type=subscriber;screen=yes;privacy=full avp_subst("$avp(s:merker)/$avp(s:merker1)", "/(.*)>(.*)/\1>/"); the result for merker1 is Remote-Party-ID: "test"<sip:12345@localhost> what must I change in the re to get this result? <sip:12345@localhost> thanks in advance, Andreas

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[Serusers] Call Transfers in SER + Asterisk architecture
by Walter Rodrigues Filho 26 Nov '06

26 Nov '06

Ricardo - this is an Asterisk issue. Configure your SIP.conf in * creating a sip peer with insecure=very fotr the SER peer, like this: [ser-peer] type = friend ; should you want to receive and make calls to SER context=from-ser ; the context in dialplan with extensions allowed to be accessed by SER. Here you must have PSTN extensions capability. host=200.XXX.XXX.XXX ; the IP address for SER fromdomain=domain.com ;the Domain part of uri to be verified by asterisk on the INVITE received by SER. qualify= yes ; just to check the latency between SER and Asterisk (like this, if over 2000ms Ast will report as unavailable peer). disallow=all allow=alaw allow=g729 insecure= very ; this line garantees that any username part of Request URI sent by SER in INVITE to Asterisk will be accepted by Ast and routed to the dialplan. So, if SER send an INVITE to HYPERLINK "mailto:5531332818847@domain.com.br"5531332818847(a)domain.com.br , Asterisk will look for a section of type =user in SIP conf to match the user part first, it won't find, then it will look for a type=peer. It will find and try to match the IP address as in host= ...line and will accept any username part as per insecure=very line. Then, if context=from-ser in the Ast dialplan allows this dialing string (553132818847), it will proceed from there. Hope it helps. At. Walter -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.16/552 - Release Date: 26/11/2006 11:30

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[Users] Improper relaying of Invite requests
by Kaushik Kumar 26 Nov '06

26 Nov '06

Hi, I am working on a project which involves testing SER with some dummy clients. I have downloaded and installed OpenSER. I am using the very basic Hello.cfg configuration file available online. I find that when SER gets a INVITE request from the caller, it does not deliver it properly to the client (callee). Some debugging revealed that it relays the request to port 5060 at the client rather than the one which was registered by the callee earlier. I tried to look at the source code of SER and found that the callee's contact information never seem to be looked up at any stage. Is this due to some missing information in my configuration file? Any feedback would be appreciated. Thank you -Kaushik PS: I have attached (inline) the configuration file ser.cfg ------------------------------------ debug=9 fork=yes log_stderror=yes listen=128.42.3.3 port=5060 children=4 check_via=no # added dns=no rev_dns=no fifo="/tmp/ser_fifo" loadmodule "/scratch/ser-0.9.7-pre1/modules/sl/sl.so" # Stateless replies loadmodule "/scratch/ser-0.9.7-pre1/modules/tm/tm.so" # Transaction management loadmodule "/scratch/ser-0.9.7-pre1/modules/rr/rr.so" # Routing and record-routing loadmodule "/scratch/ser-0.9.7-pre1/modules/maxfwd/maxfwd.so" # Max fowards check loadmodule "/scratch/ser-0.9.7-pre1/modules/usrloc/usrloc.so" # User location support loadmodule "/scratch/ser-0.9.7-pre1/modules/registrar/registrar.so" # Registrar (requires userloc and sl) loadmodule "/scratch/ser-0.9.7-pre1/modules/textops/textops.so" # Message textual operations modparam("usrloc", "db_mode", 0) # 0 forces registration data to be stored in memory modparam("rr", "enable_full_lr", 1) route { # ------------------------------------------------------------------------ # Sanity Check Section # ------------------------------------------------------------------------ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ if (method!="REGISTER") { record_route(); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; append_hf("P-hint: userloc applied\r\n"); route(1); } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ if (!t_relay()) { log(2, "Failed t_relay"); sl_reply_error(); }; } route[2] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ if (!save("location")) { sl_reply_error(); }; }

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[Serusers] Analyse calling ID
by inge 26 Nov '06

26 Nov '06

Hi all, What sort of if structure can I use to test the Calling ID ? (ie : my calling id are member of subscriber table). For the moment I do something like this : "if(uri=~"^sip:119@") {" for the destination number, but I want to do the same for the calling ID. Thanks for your help. Best regards

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[Serusers]How to remove sip header field?
by Ren-Huang Liou 26 Nov '06

26 Nov '06

Dear all, I already know that there is a “remove_hf_f” function in the textops module. I can use this function to remove sip header fields which are supported by SER such as “From”, “To”, and “Expires” header fields, but it can’t remove sip header fields which are not support by SER such as “SIP-If-Match” and “SIP-ETag”. When I use “sl_send_reply” function to build and send sip response, this response contains some sip headers field which I don’t like. How can I remove unnecessary header fields? Thanks a lot, Ren-Huang Liou

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