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42 participants
37519 discussions

[Serusers] off-topic: Sofphones with Redirect Suport
by Felipe Louback 13 Nov '06

13 Nov '06

Hi all, I need a softphone that supports redirect. Let me explain: In all free softphones I download, I have an "outbound proxy" field. So, if my server returns and 302 Moved Temporarily, the softphone will get the Contact header field and try to call the user again using the outbound proxy, and the outbound will reply with an 302 and I will get an loop. I need a softphone who will use a location server to query the location of a user and after that the softphone will make the call directly, without using a proxy. Could anyone point me a softphone that will do this? Best Regards, Felipe -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "Come to me, all you who are weary and burdened, and I will give you rest." Matthews 11:28

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[Serusers] SER pre-release ser_ctl
by SER LIST 13 Nov '06

13 Nov '06

Hello, When I try to run the ser_ctl command to do something like ser_ctl version or any other command, I get the error ser_ctl: (111, 'Connection refused'): error SER is running while I type the command. Any hints will be appreciated. _________________________________________________________________ Buy, Load, Play. The new Sympatico / MSN Music Store works seamlessly with Windows Media Player. Just Click PLAY. http://musicstore.sympatico.msn.ca/content/viewer.aspx?cid=SMS_Sept192006

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Re: [Serusers] SER pre-release - PRESENCE ISSUES
by SER LIST 12 Nov '06

12 Nov '06

Thank you. settig the xcap_root for the xcap module fixed the two errors relating to xcap_root. BUT I am still getting the two errors relating to the lookup_domain command, namely: ------------------------------------- ser: parse error (192,24-25): unknown command, missing loadmodule? ser: parse error (261,26-27): unknown command, missing loadmodule? ser: ERROR: bad config file (2 errors) ser startup failed ---------------------------------- I am using the presence example file. The excerpt from my ser.cfg file is (I have marked the LINES where the error occurs): ---- start --- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/dialog.so" loadmodule "/usr/local/lib/ser/modules/rls.so" loadmodule "/usr/local/lib/ser/modules/pa.so" loadmodule "/usr/local/lib/ser/modules/xcap.so" loadmodule "/usr/local/lib/ser/modules/presence_b2b.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/fifo.so" loadmodule "/usr/local/lib/ser/modules/xmlrpc.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" { LINES TRUNCATED} if (uri==myself) { if (!lookup_domain("To")) { <-------------------------- LINE 192 xlog("L_ERR", "Unknown domain to: %tu from: %fu\n"); route(1); break; } if (method=="SUBSCRIBE") { if (!t_newtran()) { sl_reply_error(); break; }; (lines truncated) if (lookup_domain("From")) { <---------------------- LINE 261 if (lookup_user("From")) { if (is_simple_rls_target("$uid-list")) { # log(1, "it is simple subscription!\n"); # takes From UID and makes XCAP query for user's # list named "default" if (!query_resource_list("default")) { t_reply("404", "No such user list"); break; } } } } ----------- END --------------------- Any suggestions?? THanks in advance >From: samuel <samu60(a)gmail.com> >To: "SER LIST" <sergrp(a)hotmail.com> >CC: serusers(a)iptel.org, serdev(a)iptel.org >Subject: Re: [Serusers] SER pre-release - PRESENCE ISSUES >Date: Fri, 10 Nov 2006 10:14:17 +0100 > >XCAP parameters where removed from rls and pa and a new module, called >xcap contains the xcap root so you have to delete pa and rls xcap >parameter, load the new module xcap.so and set the xcap_root parameter >of xcap module to the directory having the doc structure. > >About the lookup_domain, there can be a typo. Can you post the config >file syntax?? > >Hope it helps! > >Samuel > >2006/11/10, SER LIST <sergrp(a)hotmail.com>: >>I installed the SER pre-release along w/ all modules. The problem is that >>presence now does not run (it was working w/ the presence-release-10.99). >>The errors I get are: >>---------------------------------------------------------------------------------------------------------- >>set_mod_param_regex: parameter <xcap_root> not found in module <rls> >>ser: parse error (82,20-21): Can't set module parameter >>set_mod_param_regex: parameter <auth_xcap_root> not found in module <pa> >>parse error (97,20-21): Can't set module parameter >>ser: parse error (186,24-25): unknown command, missing loadmodule? >>parse error (255,26-27): unknown command, missing loadmodule? >>ERROR: bad config file (4 errors) >>------------------------------------------------------------------------------------------------------------- >>I checked the pa module code and also the rls code and the parameters >>auth_xcap_root and xcap_root are NOT defined !!!! >> >>The error on line 185 and 255 refer to the "Lookup_domain" function. I >>verified that the domain module has been compiled and included in the >>ser.cfrg file _________________________________________________________________ Ready for the world's first international mobile film festival celebrating the creative potential of today's youth? Check out Mobile Jam Fest for your a chance to WIN $10,000! www.mobilejamfest.com

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[Users] Forwarding requests
by Harry Harcrow 12 Nov '06

12 Nov '06

I have my SER server running on a Linux machine and I am attempting to forward some requests to a Cisco SIP server. I have been running X-lite clients on windows and the same Linux box that the SER server is running on. I have no problem completing calls between the X-lite clients OR with either X-lite client calling the Cisco SIP server directly. However, when I try to make a call from either X-lite client through the SER server I eventually get timeout methods. I am trapping those calls and wrote the following route block: route [5] { record_route(); forward(uri:host, uri:port); } I have tried replacing the forward with send() and t_relay() method calls, but nothing seems to work. The header from the X-lite client directly to the Cisco SIP server, which works is: INVITE sip:2158200@65.221.7.xxx SIP/2.0 Via: SIP/2.0/UDP 166.34.149.xxx:5061;rport;branch=z9hG4bK5963AD09C90F6F3A16F55A271C98F573 From: Harry Harcrow <sip:1000@166.34.149.xxx:5061>;tag=1279481120 To: <sip:2158200@65.221.7.xxx> Contact: <sip:1000@166.34.149.xxx:5061> Call-ID: 28E5B723-9B02-BF38-1A81-1FECC0A85D5E(a)166.34.149.xxx CSeq: 6254 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 313 The header from SER to the Cisco server, which does not work is: INVITE sip:2158200@65.221.7.xxx SIP/2.0 Record-Route: <sip:166.34.149.xxx;ftag=2061592395;lr=on> Via: SIP/2.0/UDP 166.34.149.xxx;branch=0 Via: SIP/2.0/UDP 166.34.149.xxx:5061;rport=5061;branch=z9hG4bK4EA9A90B8CE7CE52BFB69DA431560E5 7 From: Harry Harcrow <sip:1000@166.34.149.xxx:5061>;tag=2061592395 To: <sip:2158200@65.221.7.xxx> Contact: <sip:1000@166.34.149.xxx:5061> Call-ID: 38CA166F-59A6-E44B-F34E-9E1D6307698B(a)166.34.149.xxx CSeq: 3009 INVITE Max-Forwards: 16 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 313 The only thing that I can see different is the record_route record and an additional Via record. Does anyone have any idea what I am doing wrong? Thanks Harry Harcrow

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[Serusers] Suggestion
by Eddie Johnson Jr 12 Nov '06

12 Nov '06

I am new to setting up SER and this is the test setup. 1. DHCP running from router 2. SJ Phone on laptop 192.168.X.X 3. Grandstream Budgetone 101 192.168.X.X 4. SIP Express Router 192.168.X.X 5. Asterisk Server 192.168.X.X attached to tdm400p card is an analog phone Will this work internally? There is no connection to a telco. The SJ phone is showing the following: nat/firewall: Port Restricted Cone NAT SIP calls use outbound proxy: 192.168.X.X (SER) Will I have to register the SJ, Grandstream and Asterisk PBX? Ed

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[Serusers] SER not sending 486 Busy Here
by Ravi ***** 11 Nov '06

11 Nov '06

Hi Users , I am Using SER-0.96 , When I make a call from SER to Pstn it works fine I mean is all "INV"BYE"CAN". But when I recieved a call from Pstn I cannot cancel the call because X-lite is sending to pstn gateway 480 temp. service not available instead of 486 Busy Here. I am using default config. but I dont understand where it went wrong . Any Idea or help greatly appreciated Thank You. Regards, Ravi.

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Re: [Serusers] Re: How to change Caller ID in FROM field ?
by lawrence k.y. lin 10 Nov '06

10 Nov '06

Hi Kristian. Thanks for your email. I am using ser 0.9.6 and did not find uac_replace_from function. Then I did a search and found that it is in openser. I did install openser and it worked. Do you or anyone know if this function will be added into ser release ? Please advise, thanks, Larry >------------------------------ >Message: 20 >Date: Wed, 25 Oct 2006 13:11:06 +0200 >From: Kristian Larsson <kristian(a)netatonce.se> >Subject: Re: [Serusers] Re: How to change Caller ID in FROM field ? >To: "lawrence k.y. lin" <lawlin888(a)hotmail.com> >Cc: serusers(a)lists.iptel.org >Message-ID: <20061025111106.GJ30882(a)juniks.net> >Content-Type: text/plain; charset=iso-8859-15 > >On Tue, Oct 24, 2006 at 03:45:01PM -0700, lawrence k.y. lin wrote: >>Hi all, >> >> Appreciate if someone can give me a suggestion or even an answer. >> Since the call from SIP device to PSTN phone did not have Caller ID >>and our Telco carriers request for the Caller ID, we have to set a number >>before the calls are routed to PSTN gateway. >> I did it in Asterisk before with the function calls SetCallerID(CLID) >>or Set(CALLERID(number)=CLID) (depends on the Asterisk version) and these >>function calls worked. But because of the scalabliity issue, I changed to >>SER and would like to do the similar thing as in Asterisk. >> I read the SER document and awared SIP_HF_FROM has the data for the >>entired FROM field. Is there any other variables for Caller ID only ? >> I searched the mailing list for SER and did not find any related topic >>(only found one with title: caller-id with raius using sip-rpid, but it is >>irrelevant). >There is a function called uac_replace_from which >can replace values in the From header, however I >would need some help with it myself. >For example, towards one of my SIP "peers", I need >to prefix a "0" on all From headers which is no >problem, just do a uac_replace_from("0$fU","sip:0$fU@$fd"); >$fU is the from user part and $fd is the from >domain... >But how do I do if I first want to strip a few >characters of from $fU and then add a few others? >Regards, > Kristian. >-- >Kristian Larsson _________________________________________________________________ Find a local pizza place, music store, museum and morethen map the best route! http://local.live.com?FORM=MGA001

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[Serusers] fr_inv_timer + avp not working under high load
by Marcus Hunger 10 Nov '06

10 Nov '06

Hi list, I am trying to realise dynamic invite-timeout-timers loading timer-settings per call from my database using avp. It seems to work perfectly on my development-system but fails nearly completly in my high loaded life-environment. According to my logs, timer values are set correctly (eg.: reply_received: FR_INV_TIMER = 15, see modules/tm/t_reply.c:1371), but most of the times the timers have no effect. My setup is similar to http://siprouter.onsip.org/doc/gettingstarted/ch09.html, but avps are not written by hand, but acquired with: modparam("tm", "fr_inv_timer_avp", "mytimer"); avp_db_load("$ruri/username","s:mytimer"); 'mytimer' is an attribute in an usr_preferences-table. Ser's version is 0.9.6. I already searched the list-archives without finding anything regarding this issue. I Also failed accessing the bug-tracker (http://bugs.sip-router.org/), it gave me an internal server error. Has anybody experienced this effect? Is this a bug? Is there a workaround? Best regards, Marcus Hunger

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[Users] Problem with TCP INVITE being bridged by OpenSER to UDP
by Russ Daigle 10 Nov '06

10 Nov '06

I have a scenario where user Alice registered with OpenSER with UDP port 5070. Then, I have user Bob sending an invite to Alice via the OpenSER using TCP. Hence, OpenSER is bridging my TCP SIP requests to UDP (and vice versa for the responses). Here is the sequence: Alice ----- TCP Invite -----> OpenSER ---------- UDP Invite ------ > Alice Alice responds back with 180 ringing, then 200 Ok. Upon getting the 200 OK, Bob sends an ACK (to complete the INVITE), followed by an Immediate BYE. THE TROUBLE IS...... the ACK and the BYE that Bob sends are dropped by OpenSER, and never forwarded onto Alice!!!! I see nothing wrong with the messages. WHat is happening? sip:~ russdaigle$ sudo tcpdump -i en0 -A -q -s 2000 port 5060 Password: tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on en0, link-type EN10MB (Ethernet), capture size 2000 bytes The following shows an sip-invite-bye from mutator using TCP to mjsip (via OPENSER). Everything after the ACK is dropped and not sent on. Then, is a similar but UDP exchange. In that case, we get a 404 not found in response to the BYE!!! 01:17:59.553909 IP 10.10.2.6.50135 > 10.10.1.234.sip: tcp 588 INVITE sip:alice@siptest2.com:5060 SIP/2.0 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Contact: <sip:bob@10.10.2.6>;transport=TCP Max-Forwards: 70 Content-Type: application/sdp Content-Length: 195 v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 10.10.2.6 t=0 0 m=audio 5000 RTP/AVP 0 4 97 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:97 telephone-event/8000 01:17:59.554929 IP 10.10.1.234.sip > 10.10.2.6.50135: tcp 536 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Server: OpenSer (1.1.0-notls (i386/darwin)) Content-Length: 0 Warning: 392 10.10.1.234:5060 "Noisy feedback tells: pid=11962 req_src_ip=10.10.2.6 req_src_port=50135 in_uri=sip:alice@siptest2.com: 5060 out_uri=sip:alice@10.10.2.6:5070 via_cnt==1" 01:17:59.555035 IP 10.10.1.234.sip > 10.10.2.6.5070: UDP, length 814 INVITE sip:alice@10.10.2.6:5070 SIP/2.0 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Contact: <sip:bob@10.10.2.6>;transport=TCP Max-Forwards: 69 Content-Type: application/sdp Content-Length: 195 P-hint: usrloc applied v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 10.10.2.6 t=0 0 m=audio 5000 RTP/AVP 0 4 97 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:97 telephone-event/8000 01:17:59.667962 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 364 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 To: Alice <sip:alice@siptest2.com:5060> From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Server: mjsip stack 1.6 Content-Length: 0 01:17:59.699680 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 505 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060> From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Server: mjsip stack 1.6 Content-Length: 0 01:17:59.699683 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 728 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 01:17:59.700273 IP 10.10.1.234.sip > 10.10.2.6.50135: tcp 440 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060> From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Server: mjsip stack 1.6 Content-Length: 0 01:17:59.700299 IP 10.10.1.234.sip > 10.10.2.6.50135: tcp 663 SIP/2.0 200 OK Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 01:17:59.804566 IP 10.10.2.6.50135 > 10.10.1.234.sip: tcp 435 ACK sip:alice@siptest2.com:5060 SIP/2.0 Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Route: <sip:10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK59972 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 CSeq: 1 ACK Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Max-Forwards: 70 01:17:59.839493 IP 10.10.2.6.50135 > 10.10.1.234.sip: tcp 435 BYE sip:alice@siptest2.com:5060 SIP/2.0 Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Route: <sip:10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK68444 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 CSeq: 2 BYE Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Max-Forwards: 70 01:18:00.199306 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 728 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000

2 3

[Serusers] IM and silo table in MySQL
by davor jovanovic 10 Nov '06

10 Nov '06

Hi all, I have a problem with silo table in MySQL. messages are stored in the table when user is offline, and when comes back online messages are delivered to the client. Problem is that the messages aren't erased from table after delivery . Am I missing something in my ser.cfg? I'm using ser-0.9.6 and XLite 3.0. I will appreciate any help. thanks /davor loadmodule "/usr/lib/ser/modules/msilo.so" modparam("auth_db|uri_db|usrloc|permissions|group|acc_db|msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo", "check_time", 10) modparam("msilo", "clean_period", 3) modparam("msilo", "use_contact", 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; }; ... } else if (method=="MESSAGE") { route(8); break; }; ... route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!www_authorize("","subscriber")) { www_challenge("","0"); break; ... m_dump(); } route[8] { ########################## # Message Handler # ########################### if (!lookup("location")) { if (method == "MESSAGE") { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("1")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; }; }; route(1); t_on_failure("1"); } ......... failure_route[1] { if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); };

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