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44 participants
37550 discussions

[Serusers] tls module documentation not available
by tzieleniewski 09 May '07

09 May '07

Hi, The tls module documentation is unavailable page - http://www.iptel.org/ser/doc/modules/tls. is it temporary?? Cheers Tomasz

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[Users] How to use forward command with script variables?
by Noel R. Morais 09 May '07

09 May '07

Hi All, I am trying to do something like this: route[1] { $var(forwardTo) = "192.168.192.100"; forward($var(forwardTo)); exit; } But it do not work. you know a way to do it? Thanks Noel

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[Serusers] isup-oli
by Jai Rangi 09 May '07

09 May '07

Hello, Can some please tell me how can I capture isup-oli field from a incoming call. We need to capture that to recognize the incoming call from pay phone. Thank you, -Jai

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[Users] XCAP-Lite problems
by Schumann Sebastian 09 May '07

09 May '07

Dear all, starting to build up OpenSER as presence server, I had no mentionable problems. Now, I tried to implement an Apache2 based XCAP server to store presence info, buddy lists etc. there. I have a problem setting up the XCAP-Lite server v0.9. Searching the web and the mailing list did not lead to a successful answer why the following happens: After setting up Apache2 with all required modules, virtual host etc. I try to use Eyebeam 1.5 to write the presrules and resource-list.xml to the database. This failes. I have successfully installed Apache2 with pear (+ extensions), php, mysql support. I created a virtual host with the following settings: NameVirtualHost * <VirtualHost *> <Location "/"> Dav On #necessary for PUT method </Location> ServerName 172.16.167.128 DocumentRoot /var/www/ #/var/www/xcap-root didn't work for GET as well, changed RewriteEngine On RewriteLog /var/log/apache2/rewrite.log RewriteLogLevel 2 RewriteRule /xcap-root/(.*) /xcap-root/index.php ProxyVia On </VirtualHost> I putted all files from XCAP-Lite archive in /var/www/xcap-root. After using PUT method through Eyebeam (change in contact list) I receive the 204 No content message. BUT: Not the database content is changed, the putted XML file overwrites index.php instead, which means, further requests fail. Does anybody know why the access to database does not work? Password etc. in config.php is ok. Thanks in advance. Best regards Sebastian PS: As I didn't find a mail address of the author of the program, I hope he is also subscribed to that list. There is probably an error in xcap-lib.php on line 766. I think there is a space between function and XCAPclient necessary.

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Re: [Serusers] Which version of SER to use?
by lists＠infoway.net 09 May '07

09 May '07

Excellent news. Thanks for the feedback. Overall, and please don't take this negatively, I think documentation for SER is difficult for beginners, so if we have to go through the learning curve anyway, we might as well do it in the newer version. Thanks again On Tue, May 8, 2007 5:04 pm, SIP <sip(a)arcdiv.com> said: > For all intents and purposes, SER 2.0 (the new one) is ready for use. > The only thing it's really waiting on to be 'stable' is some better > documentation. However, it's best, if you're just getting started, to > deploy with the new version and maybe muddle your way through a bit > until documentation is complete. > > N. > > > lists(a)infoway.net wrote: >> We are planning on deploying a couple of SER servers and integrate them with >> either SEMS or Asterisk for voicemail. However, we are a bit confused as to which >> version of SER to go with? From what we can tell, the latest stable version is >> 0.9.6. However, we know of the new 0.10.X version. Reading up on the site, there >> is some documentation that explains a little bit of the differences and/or >> migration recommendations/steps. >> >> So, in an effort to keep life as simple as possible, which version should we work >> with? We are afraid that if we go through a learning cycle with 0.9.6 we're going >> to "suffer" when migrating to the new version, either because of the learning >> curve or because of a potential painful migration process. >> >> Any recommendations? >> >> Thanks >> >> >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > >

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Re: [Users] Proxy/Registrar and NAT
by Bernie Hoeneisen 09 May '07

09 May '07

Hi Klaus, Alex et al. I was thinking about the same problem. My setup has two redundant SIP Proxy/Registrar (OpenSER 1.1) Servers (P1 and P2), no additonal edge proxies. It uses either MySQL cluster or Master-Master Replication to exchange the state of the location table. For NAT Processing of REGISTER I use the Nathelper module, which writes the original (inside NAT) address to the 'contact' column and the information, where the REGISTER request was received from (outside NAT socket) goes to 'received' column of the location table. e.g.: contact column: sip:behoe@192.168.1.56:2051 received column: sip:130.56.88.20:5432 When an INVITE arrives at P1 (or P2), the 'contact' column will go to the R-URI and the 'received' value will be used as the next target of the INVITE request. INVITE requests that do not arrive at the proxy, where the UA has registered, have (depending on the NAT type) some likelihood to fail. Appart from that and as long as there is no path value saved in the location table, this work fine. If there is a path value, its content will go to the Route header field and the the request is sent to the topmost Route entry. Therefore it looses the information that was stored in the 'received' column....:-( The preserve this information and to make sure that LL requests are routed via the proxy, the UA has registered to, I have some idea, that uses path. (Unfortunately not all of my UAs support 'outbound'...:-( ): When the REGISTER arrives at the Proxy P1, I would insert a Path header field with the address of P1 (i.e. the IP address and port of the proxy processing the request) Furthermore I would add a new header field parameter to the contact header field (let's call it "extsock" for the time being) containing the external NAT socket (i.e. the same information that also goes to 'received' column) e.g. Path: <sip:136.59.10.85:5060;lr>, Contact: <sip:behoe@192.168.1.56:2051;extsock=sip:130.56.88.20:5432;lr> This will then be stored in the 'path' / 'contact' columns of the location table at save(). When the INVITE arrives e.g. at P2, it does the location lookup: The Route header field will be populated with the content of the 'path' column and the R-URI will be rewritten with the 'contact' column. This means the request will be normally forwarded to P1 (topmost Route). The SIP INVITE looks e.g. as follows: INVITE sip:behoe@192.168.1.56:2051;extsock=sip:130.56.88.20:5432 SIP/2.0 Route:<sip:136.59.10.85:5060;lr> [...] After the request arrives at P1, it checks, whether there is an extsock parameter in the R-URI, and if yes, it uses its values as the next target of the request i.e. sip:130.56.88.20:5432 (Maybe P1 could even remove the extsock parameter from the R-URI at this stage to make some Nokia E-Series' phones happier...) My questions: - Is this idea feasible? Does it work in any case? Any issues with forking? - Could it easily be configured to OpenSER, or is there a change in the source code necessairy? - How much does it break SIP standards? ;-) Looking forward to your feedback. Have a nice weekend! cheers, Bernie > On Wednesday 02 May 2007 08:33, Klaus Darilion wrote: >> Hi Alex! >> >> Without having done this: You can configure the SIP proxies as load >> balancers too which distribute the load over all the proxy/registrars >> (including itself). > > I thought of that, but then every REGISTER needs to be forwarded to another > proxy to get the path info right. This result in a great amount of traffic > between the proxies and extra processing power. I want the REGISTER request > to be handled on the first host it arrives on. Then only some INVITE's need > to create inter-proxy traffic.

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[Serusers] Which version of SER to use?
by lists＠infoway.net 08 May '07

08 May '07

We are planning on deploying a couple of SER servers and integrate them with either SEMS or Asterisk for voicemail. However, we are a bit confused as to which version of SER to go with? From what we can tell, the latest stable version is 0.9.6. However, we know of the new 0.10.X version. Reading up on the site, there is some documentation that explains a little bit of the differences and/or migration recommendations/steps. So, in an effort to keep life as simple as possible, which version should we work with? We are afraid that if we go through a learning cycle with 0.9.6 we're going to "suffer" when migrating to the new version, either because of the learning curve or because of a potential painful migration process. Any recommendations? Thanks

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[Serusers] Clueless newbie question - integrating SER and Asterisk
by John Breen 08 May '07

08 May '07

Help! I've followed the SER documentation and searched the web like a mad thing and found nothing useful. I have an asterisk box here at home and want to put SER on it so I can use SIP clients out in the real world (on the other side of my firewall) to get to it. Now, I have got SER set up and the phones register fine and everything, BUT I cannot see how to get it to pass calls through to Asterisk. I'm not using Asterisk realtime or anything like that - just plain old asterisk with plain old config files. I thought I had it right, but calls are just not getting to asterisk so far as I can see. I know it must be possible, I just can't see how to do it. Regards John Breen

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[Serusers] SER/PSTN Gateway conf
by flavio 08 May '07

08 May '07

Hi to all, I'm newbie for SER and I'm testing my own configuration to send/receive calls through my pstn gateway, according to follow simple scenario: Phone --> PSTN --> Gateway --> SER --> SIP Phone I send and receive calls successfully, but when Phone over PSTN hangs up SER reply to BYE message with 404 Not Found.Here is my ser.cfg: debug=3 fork=no log_stderror=yes listen=10.28.19.202 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://root@10.28.19.202/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://root@10.28.19.202/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_allow_file", "/usr/local/etc/ser/permissions.allow") modparam("permissions", "default_deny_file", "/usr/local/etc/ser/permissions.deny") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("16")) { 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()) { # if ((method=="INVITE" || method=="REFER") && !has_totag()) { # sl_send_reply("403", "Forbidden"); # break; # }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(3); route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); } lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:0[0-9]*@") { # calls to PSTN #strip(0); route(4); break; }; if(uri=~"^sip:3[0-9]*@") { calls to Mobile route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; route(1); } route[4]{ #----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("10.28.52.105"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(1); } Any suggestions about my issue? Thanks 4 all, Flavio

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[Serusers] re-invite problem
by olivier.taylor 08 May '07

08 May '07

Hello, using ser 0.9.6 The following dialog is a re-invite, the re-invite is initiated by the callee after putting the call on hold. Problem is that SER answer with a 407 on the wrong port, Invite is coming from port 62907 and SER answer to port 5060. Problem doesn't occurs for the initial invite. We have tried with other Uas without problems, just having problems with this Ua (Avaya). Any idea? Best regards, Olivier U xxx.xx.xx.xx:62907 -> yy.yyy.yyy.yy:5060 INVITE sip:990099001@192.168.162.187 SIP/2.0. Via: SIP/2.0/UDP 192.168.162.181;branch=z9hG4bK309b82ccf. Max-Forwards: 70. Content-Length: 337. To: "QE-A" <sip:990099001@yy.yyy.yyy.yy>;tag=f7983fe553e1941. From: "QE-B" <sip:990099002@yy.yyy.yyy.yy>;tag=5bf8b4fb69fca84. Call-ID: 7c933d3d19bb76c396b995202b18eb97(a)yy.yyy.yyy.yy. CSeq: 567882513 INVITE. Route: <sip:990099002@yy.yyy.yyy.yy:5060;nat=yes;ftag=f7983fe553e1941;lr=on>. P-Asserted-Identity: "QE-B" <sip:990099002@yy.yyy.yyy.yy>. Allow: INVITE. Allow: CANCEL. Allow: OPTIONS. Allow: BYE. Allow: REFER. Allow: INFO. Allow: UPDATE. Content-Type: application/sdp. Contact: "QE-B" <sip:990099002@192.168.162.181>. Supported: replaces. User-Agent: NeuralX MxSF/v3.2.6.26. . v=0. o=990099002 1136076203 1136076205 IN IP4 192.168.162.181. s=-. c=IN IP4 192.168.162.181. t=0 0. a=sendonly. m=audio 20006 RTP/AVP 0 101 8 18 . a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:101 0-15. a=fmtp:18 annexb=no. a=ptime:20. a=rtcp:20007 IN IP4 192.168.162.181. U yy.yyy.yyy.yy:5060 -> xxx.xx.xx.xx:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.162.181;branch=z9hG4bK309b82ccf;received=xxx.xx.xx.xx. To: "QE-A" <sip:990099001@yy.yyy.yyy.yy>;tag=f7983fe553e1941. From: "QE-B" <sip:990099002@yy.yyy.yyy.yy>;tag=5bf8b4fb69fca84. Call-ID: 7c933d3d19bb76c396b995202b18eb97(a)yy.yyy.yyy.yy. CSeq: 567882513 INVITE. Proxy-Authenticate: Digest realm="yy.yyy.yyy.yy", nonce="463f31b81a65dee3419ce3e4bfbecc253410cbf5". Server: Sip EXpress router (0.9.6 (i386/freebsd)). Content-Length: 0. Warning: 392 yy.yyy.yyy.yy:5060 "Noisy feedback tells: pid=49990 req_src_ip=xxx.xx.xx.xx req_src_port=62907 in_uri=sip:990099001@192.168.162.187 out_uri=sip:990099001@192.168.162.187 via_cnt==1". .

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