sr-users December 2005

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158 participants
313 discussions

[Serusers] Announcement: New FTP Server with SIP related software
by Jan Janak 04 Dec '05

04 Dec '05

Hello, We have a new FTP server containing the collection of your favorite SIP and RTP related software. The FTP server is reachable either using FTP at ftp://ftp.iptel.org (use anonymous username and your email address as password), or through the web interface at http://ftp.iptel.org ftp.iptel.org is a public FTP server dedicated to freely available SIP (Session Initiation Protocol, RFC3261) based software. Here you can find everything that you will need to implement robust and highly scalable SIP based server systems. Among others, this archive contains: * SIP Express Router (SER) * RTP proxy server for NAT traversal (RTPProxy) * Web based administration interface (SERWeb) * SIMPLE based instant messaging and presence server with support for XCAP * voicemail and announcement server (SEMS) We also keep a local copy of IETF RFCs and Internet drafts. Should you have any problems, requests, or suggestions for improvements, do not hesitate to contact us by e-mail at <ftpmaster(a)iptel.org>. The contents of the FTP server will be updated and expanded regularly, it is not limited to iptel.org or SER related software. Jan.

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[Serusers] UAs and SER behind NAT
by Ladislav Andel 04 Dec '05

04 Dec '05

Hello, I'm sorry for this basic question, but I have SER running on my LAN with (RFC1918) addresses with UAs on the same network. Then there are UAs registering from behind NAT to this SER on the LAN network. (SER is not listening on any public IP and it's just used in the private network) I came across this issue. If a UA on the LAN network is registering to SER the logic of ser.cfg recognize the UA is behind NAT after nat_uac_test. If I want to introduce NAT traversal with RTPproxy for this private network nat_uac_test will always return that the UA is behind NAT because it has RFC1918 IP address. The RTPproxy will be used at all times for UAs on the network or even I introduce STUN for NATed UAs I know this is not a common usage of SER because it is invented to be used on the global internet but I was just wondering is there any possibility not to use RTPproxy at all times for UAs on the private network? Maybe this is a silly question but it would help me for my testing purposes with SER where I will not have to use my setup with usage of public IPs. Thank you for any thoughts. Ladislav

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[Users] Creating permenant contacts with FIFO ul_add
by Doug Meredith 04 Dec '05

04 Dec '05

I am trying to use FIFO to manage permanent contacts. I can create contacts now, but can't figure out what value to specify for the expiry. A positive value represents a real expiry in seconds. Negative numbers, "Never", and "Permanent" all result in "400 Invalid expires format". A zero results in "500 Error while adding contact". The information at http://www.openser.org/dokuwiki/doku.php?id=fifo_relay#ul_add is more of an example. It doesn't really explain the arguments. Does anyone know how to specify that a contact is permanent? The command "openserctl ul add" is able to create permanent contacts, so it must be possible. Doug -- Doug Meredith (doug.meredith(a)systemguard.com) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com

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[Users] FIFO response "400 ul_add: replicate expected"
by Doug Meredith 04 Dec '05

04 Dec '05

I am using OpenSER 1.0. When I send the following to the FIFO: ----------------- :ul_add:ans location 81(a)dev.skyridge.com sip:34@sirloin.skyridge.com 3600 1 ----------------------- I get the response "400 ul_add: replicate expected". I have done a google search without finding anything helpful. Can anyone explain what this means? Doug -- Doug Meredith (doug.meredith(a)systemguard.com) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com

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[Users] dispatcher relaying instead of dipatching?
by Rick Richardson 03 Dec '05

03 Dec '05

On my test network I have set up a dispatcher server that points to two proxy servers that hit an external database. I have a simple route script on the dispatcher that should be routing requests to the two proxy servers. dispatcher .51 caller .99 recipient .100 proxy1 .52 proxy2 .53 I'm making call requests from .99 to .100, with .51 used as the proxy server. The problem I'm getting, is that the INVITE is being sent directly to the recipient without passing through the proxys, the dispatcher is passing the message directly. At least i think that is what is happening. Here is the sip INVITE: INVITE sip:rick@192.168.0.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.99:5060;rport;branch=z9hG4bK837723126 From: <sip:rick@192.168.0.51>;tag=679508646 To: <sip:rick@192.168.0.100> Call-ID: 125992616(a)192.168.0.99 CSeq: 20 INVITE Contact: <sip:rick@192.168.0.99:5060> Max-Forwards: 5 User-Agent: Linphone-1.0.1/eXosip Subject: Phone call Expires: 120 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 352 So it seems to me that the invite request is being sent to .51 and it is forwarding directly to .100 without dispatching the message to .52 or .53 here is my dispatcher config: # $Id: dispatcher.cfg,v 1.1.1.1 2005/06/13 16:47:37 bogdan_iancu Exp $ # sample config file for dispatcher module # debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL0 children=2 check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) port=5060 fifo="/tmp/openser_fifo" # for more info: sip_router -h # ------------------ module loading ---------------------------------- loadmodule "//lib/openser/modules/xlog.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/dispatcher.so" # ----------------- setting module-specific parameters --------------- # -- dispatcher params -- modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") # modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("2", "2"); forward(uri:host,uri:port); # t_relay(); }

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[Serusers] Failure route weirdness....
by sip 03 Dec '05

03 Dec '05

Okay... I'm a little stumped on what's going on here, so I'll start with a general explanation before I bother posting my (likely hard to read) ser.cfg. I'm having a weird issue when calling from X-Lite to a SNOM 320 phone (I haven't tried the SNOM 190 with a different firmware yet, but I don't see that it would make much difference). I have a failure route which handles voicemail and such, and I have the invite timer set to 27 seconds (roughly 4.5 standard US rings). Now, when I call from X-Lite to X-Lite, or from Leadtek 8051 to X-Lite, or from SNOM to X-lite, or from SNOM to SNOM, or from Leadtek to Leadtek, etc, etc... everything works EXACTLY as it's supposed to. I have log messages telling me when the failure route is being entered, and it hits it at right on 27 seconds. However, when I call TO the SNOM phone from any other device besides another SNOM phone, I never hit the failure route until a 408 timeout at 58 seconds. Now, I can't think of a reason for a UA to overwrite the invite timer on SER or even how such overall weirdness would occur, but it's 58 seconds every time like clockwork.. error 408. At that point, it enters the failure route and does its processing (forwarding the call to voicemail as far as SER is concerned but not as far as the UA is concerned... which just receives an error message). Any ideas? Also... when I dial SNOM to SNOM, the SIP method I get in the ACC is PRACK (no INVITE). Can someone fill me in on what a PRACK is? Is there something I should be looking at to figure out why calling one UA behaves differently than other UAs? N.

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[Serusers] SER + Radius
by Sait Karalar 03 Dec '05

03 Dec '05

Hi list, I am so new to list SER. I just tried to I want to authenticate SIP users via RADIUS and want accounting with radius. also if possible, ask route to radius also. but I did not manage it ;( any simple sample? I read documents etc. no good sample found. -- gMail : google mail Try it! It is the best http://www.gmail.com

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[Users] Features support cfg file ?
by Paul van Schagen 03 Dec '05

03 Dec '05

Hello, I have setup openser and I have a request to incorporate an unconditional forward feature which can be turned on by dialing a "*21+ number" and can be turned of by dialing #21#. Can anyone provide me with an example piece of .cfg file that can do this ? best regards Paul

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[Users] Handeling Diversion - expired?
by Helge Waastad 03 Dec '05

03 Dec '05

Hi, I have a question more in a general aspect. Since the draft-levy-sip-diversion-08 has expired, should a Proxy strip such headers? According to RFC3261,the UAS MUST ignore headers-fields not defined in the specification or any supported extension... I've realized that some systems need Diversion headers for voicemail (I know Asterisk doesn't, but Cisco Unity does) Any comments? -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

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[Serusers] STUN support
by Jan Henkins 02 Dec '05

02 Dec '05

Hello all, Is there anything that needs to be done to the ser cfg file to support STUN? If so, could you please share that config snippet? Thanks in advance. -- Regards, Jan Henkins Cell: +27-84-951-4334

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sr-users December 2005