sr-users November 2006

sr-users@lists.kamailio.org

189 participants
307 discussions

[Users] one way audio
by rilawich ango 08 Nov '06

08 Nov '06

Hi all, Anyone experiences an one way audio problem when a call is connected to the called party? I am using openser with mediaproxy to provide services to our customer. However, some of our customer complain about the one way audio problem (calling party can hear called party but not vice versa). The system works to most of the customer without that problem. I wonder why there are some customers experience that problem. I assume there is the port problem during RTP packet transmission. Am I right? In the mediaproxy setting, the portRange is 2000:8000. Do I need to increase it? Any method to trace the problem if my assumption is wrong? Thanks.

1 0

[Users] Forwarding requests
by Harry Harcrow 07 Nov '06

07 Nov '06

_____ From: Harry Harcrow [mailto:harry.harcrow@verizonbusiness.com] Sent: Tuesday, November 07, 2006 3:36 PM To: 'users(a)openser.org' Subject: Forwarding requests 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

1 0

[Serusers] ser accounting Issue
by ram 07 Nov '06

07 Nov '06

Hi i have SER running with mysql i have a question related to accounting when the calls are live, when i reboot the server i wont see the Bye messages, this kind of situation, how can i terminate the call i mean send bye message to all pending calls and account them when i server come back any help on this issue iam not sure if this issue already addressed Ram

1 0

[Users] photos from OpenSER BoF
by Daniel-Constantin Mierla 07 Nov '06

07 Nov '06

Hello, just to get a feeling about how the event is going, we uploaded few pictures, since the wireless is very crowded, bandwidth is quite limited -- more will come in the next days, along with the slides. We are very please to have more that 75% attendees from the first days, the room for BoF session proved very small, hopefully tomorrow the places in summit room will be enough -- we got a lot of new people interested in the summit from VoN event. http://www.openser.org/pictures/openser-bof/ Cheers, Daniel

1 0

[Serusers] error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy
by vijay tiwari 07 Nov '06

07 Nov '06

hi all we i call to one UA to another UA i found this error. error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy thanks vijay --------------------------------- Sponsored Link Free Uniden 5.8GHz Phone System with Packet8 Internet Phone Service

1 0

[Users] OpenSER Summit @ VoN Berlin - ready to go
by Daniel-Constantin Mierla 07 Nov '06

07 Nov '06

Hello everybody, OpenSER summit is ready to start in one day. Everything is in place, although some fine tunings will be there all the time. Next days the activity on mailing lists may decrease a bit, as many of OpenSER users are joining the event. To check the event's schedule and details: http://www.voneurope.com/expo.html http://www.openser.org The event will not be broadcasted on the web, the presentations will be collected and published after the event. Photos and briefing will be available as well. Cheers, Daniel

3 2

Re: [Serusers] new major SER pre-release, Ottendorf, is out for testing
by sip 07 Nov '06

07 Nov '06

Well it's about time!!! ;) Seriously though, thanks for all the hard work you've all put in on this release. I'm looking forward to getting the chance to install and play with it, and (most importantly for me) documenting as much as I can about how to migrate from a production 0.9.X solution to the new code -- something I imagine is going to be tricky as much of our actual web/management code will need to be rewritten. I can't express really how much SER and the Iptel team have changed my life for the better. Thanks again for everything. N. On Mon, 06 Nov 2006 12:02:24 +0100, Jiri Kuthan wrote > Ladies and gentlemen, girls and boys, > > the new testing pre-release of the best SIP proxy server ever, SER, > is out! > > This major pre-release, codenamed Ottendorf, includes fundamental > improvements to the software. The improvements allow SER users to > achieve significantly higher managability and reliability by better > data mode, interop features, security, etc. They recollect > voluminous feedback we have received from iptel customers and open > source users. In particular, the new features include but are not > limited to: - improved data model which eliminates any possible ambiguities. > - key under-the-hood changes with impact on performance: timer subsystem > overhaul (now fast, with fine resolution, high accuracy and > variability), TCP processing overhaul (higher scalability) - key > reliability changes: DNS cache and IP blacklisting - key interop > changes: support for STUN-bis, presence server aligned to latest > IETF development, sanity module for early detection of invalid messages > - more management: management features are now available via XML-RPC > remotely, as well as via an alternative binary protocol - more > security: ratelimit module, more check types in permissions module - > scripting features: select operator in combination with revised AVPs, > named flags and routes, more types of routes (onsend, timer, > branch) - applications: presence has been greatly updated to keep > pace with development in the IETF/SIMPLE (includes asynchronous subscription > authorization, extended PIDF); web interface includes many new managibility > features (multidomain, templates and multilanguage to name some). New > scripting flexibility allows safely for features like individual ringing > timers and direct access to SIP message elements. > - new tools: Nils's script execution tree tool > http://lists.iptel.org/pipermail/serdev/2006-October/008124.html > and refurbished ser_ctl > see the following links for some additional details: > - http://www.iptel.org/ser/doc/010whatsnew > - http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/NEWS?rev=HEAD > > Accompanying software, SEMS and SERWEB have been greatly overhauled > as well. SEMS now features improved performance (single-threaded > model now), service composition via python-IVR-scripting and B2BUA, > adaptive playout buffer for better voice quality, and more. SERWEB > features new multilevel administrative access system, multidomain > support, per-domain customization, remote xml-rpc access, > multilanguage support, and more. The new command-line utility, > SER_CTL, provides also many great improvements over the previous tool. > > Interop > ------- > Shortly, we have been keeping the bar high through our participation > in the IETF and SIPITs. SER implements latest interop changes from > the IETF (STUN-bis, presence + XCAP, to name the most important ones) > and both new features and legacy features were extensively and > successfully tested middle of October in SIPit. > > Next Steps > ----------- > Keep in mind, this is a pre-release for testing purposes. It is now > call on users to test the software hard and provide feedback. In the > meantime, the codebase remains frozen and we only commit patches > fixing various bugs. Once we get good confidence that Ottendorf has > stabilized and is being adequately documented, we will declare the > final release. No date has been determined yet. Also, we consider > changing CVS to SVN. > > Again, and this is very important: the final release will be as good > as the feedback we obtain for this pre-release. Please do spend your > time sending solid feedback back to [serusers]. > > Roadmap > ------- > We keep rolling. We are still debating with our generous > contributors what is the next possible good things to work on, the > topics include but are not limited to as of now: - documentation. - > databases are important. LDAP and BDB, DB-API revision. - interop: > identity, TLS revision - advanced scripting (scripting engine, vars > returns, AVPs) - uncountable improvements (futex, dev/rtc, > tm/refcnts, tcp fd cache, .....) > > Trivia > -------- > The release name Ottendorf is coming from a place in Saxonian > Switzerland, where many iptel fellows and their friends spent a > weekend during this Indian summer. http://www.iptel.org/jan/gallery/saxonian_switzerland_2006/ > http://www.iptel.org/~andi/2006-09-22-Ottendorf/ > http://www.iptel.org/~vlada/saxon_switzerland/index.html > http://www.ohlmeier.de/albums/ottendorf_2006/index.html > > Acknowledgments > --------------- > This grand achievement is result of extremely many folks who did lot > of hard work. Out of folks affiliated with iptel, Andrei and Jan > spent immense effort on moving SER forward, with great support from Hallik > (command-line tool), Michal, Miklos, Nils, Tomas, Vaclav (presence), > Vlada (new STUN). In this context, I would like to thank to our > company for the great support. Recently, Bogdan Pintea greatly > helped to make the upcoming release better and more stable under > sponsorship of his company, iptego. transnexus provided osp support. > > SEMS has been greatly improved thanks to Stefan and Raphael. > Credits for serweb go to Karel. > > Same importantly, SER has been immensely benefitting of contributors, > to name in alphabetic order at least the 'top-of-the-iceberg' > fellows: Bogdan Pintea, Cesc Santa, Greg Fausak, Greger Teigre, > Hendrik Scholz, <http://tutpro.com/>Juha Heinänen, Klaus Darilion, > Maxim Sobolev, Peter Nixon, Uli Abend. > > Great support on the mailing list has been provided by Greger V. > Teigre, Java Rockx, Klaus Darilion, Zeus Ng, and more. > > Simply, this list is too short and incomplete to capture all the > generous contributions that came in -- my appologies for everyone > else I have missed. > > Warnings and Disclaimers > ------------------------ > This is a TESTING version. We are encouraging people to give it intense > testing and provide us with feedback, so that we can properly assess > what it takes to declare it final. The volume and number of changes > is big, caution is advised. > > All GPL warnings and disclaimers apply. > > This is SER, which is not the derivate software known as openser. We > have no ideas of in what extent and timeline derivate projects follow > (or not) SER. > > ------------------------------------------------------------------------------- > > To Give it a Try > ----------------- > warning: this is just a 'jump-start' for black-belt users. It is not > a replacement for documentation or a detailed step-by-step installation > guide. It assumes knowledge of operating system, SIP, and SER. > > PREPARATION > - make sure mysql is running. In the default configuration, SER > assumes it to be running on localhost, if this does not hold, you > need to change ser.cfg. - make sure python is installed (needed for > new command-line utilities) SER INSTALLATION FROM CVS - Download > from CVS/HEAD using instructions at http://www.iptel.org/downloads - > Compile it: make include_modules="mysql xmlrpc" all - Install it > the default way of your administrative privileges allow to: make > include_modules="mysql xmlrpc" install - Download command-line tool > (see more documentation under http://cvs.berlios.de/cgi- > bin/viewcvs.cgi/*checkout*/ser/serctl/README?rev=HEAD) - cvs co serctl > - Initiate database: > ./scripts/mysql/ser_mysql.sh create > START > - Start it. Call: ser > - Begin provisioning users with the new serctl (the capital word are > just place holders) # ser_ctl domain add DOMAINNAME # ser_ctl > user add USERNAME@DOMAINNAME -p PASSWORD # If you want to have > Asserted Identity header for your user # - ser_attr add uid=UID asserted_id="PID" > # If you want to have gateway support > # - ser_db add attr_types name=gw_ip rich_type=string raw_type=2 \ > description="The gateway IP for the default ser.cfg" default_flags=33 > # - ser_attr add global gw_ip=GATEWAY-IP > - more things you may wish to do: > - familiarize yourself with current ser.cfg > - send us feedback. > > -- > Jiri Kuthan http://iptel.org/~jiri/ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

2 2

[Users] Re: [Semsdev] SEMS users meeting at VON Europe 2006 (Berlin)
by Stefan Sayer 07 Nov '06

07 Nov '06

Hello, we would like to meet tomorrow, Wednesday, at 13:00 in the exhibition area. I hope to see those of you who are at VON Europe and interested in SEMS to discuss its development. Regards Stefan Raphael Coeffic wrote: > > Hello everybody, > > Stefan Sayer and I are proud to invite you to a SEMS users meeting at the > VON > Europe 2006 in Berlin on wednesday 8th november. It could a great chance > to > discuss all your favorite features and gives us the possibility to know > what > is important to you. > > At the moment, there is still no meeting point that have been agreed upon. > So > if you have any good ideas, including plans for the evening, please make > your > suggestions on the list. > > Regards > Raphael. > > -- > Dipl. Inf. Raphael Coeffic > > iptelorg GmbH > Am Borsigturm 11 > 13507 Berlin > Germany > > rco(a)iptel.org > www.iptelorg.com > > T +49-30-3251-3218 > F +49-30-6908-8248 > _______________________________________________ > Semsdev mailing list > Semsdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/semsdev > -- Stefan Sayer Media Services Development iptego GmbH Am Borsigturm 40 13507 Berlin Germany stefan.sayer(a)iptego.de www.iptego.de _______________________________________________ Semsdev mailing list Semsdev(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/semsdev

1 0

[Serusers] Using different domains
by Nick Hoffman 07 Nov '06

07 Nov '06

Hi guys. I just noticed that when user agents register, the domain in their SIP URI does not have to equal their domain in the subscribers database table. For instance, I have 2 domains in the domain table: sip.altcall.com sip.voxpak.com The username "foobar" in my subscribers table has the domain "sip.altcall.com", but can register and make calls as foobar(a)sip.voxpak.com . Why is this? Cheers, -- Nick E: nick.hoffman(a)voxpak.com P: +61 7 5591 3588 F: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

3 2

[Users] Connect two OpenSER...
by Steffen Witt 07 Nov '06

07 Nov '06

Hello, I have two OpenSER servers that I want to "connect" with each other: Phone "a" <---> Server "A" <------------> Server "B" <---> Phone "b" Phone "a" should call phone "b" without adding the domain during call initiation and vice versa. Phone numbers from "A" starts with "5xxxx" and from "B" with "6xxxx". I have tried to edit openser.cfg e.g. at server "A": route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; ... # TRY TO USE AVP # if (uri=~"69000@") { avp_printf("$avp(s:new_uri)","sip:69000@IP_OF_SERVER_B;transport=udp"); avp_pushto("$ru","$avp(s:new_uri)"); t_relay(); }; # TRY TO USE FORWARD # # if (to_uri=~"69000@") { # forward("udp:IP_OF_SERVER_B:5060"); # exit; # }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; ... route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } The call receives phone "b" but there will be no connection established. I get immediatelly an error on server "A" side... "Status: 500 I'm terribly sorry, server error occured (1/SL)". Any ideas how to configure it correctly so that I can reach my phone on the other site? What rules do I have to create? What functions to use? Are there any examples available (besides the source files)? Thanks for your help! Best regards, Steffen

2 1

← Newer
1
...
22
23
24
25
26
27
28
...
31
Older →

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users November 2006