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[Serusers] PA module SER 2_0_0 - no xcap query with auth parameter set to xcap
by tzieleniewski 03 Jul '07

03 Jul '07

Hi, I am trying to use the xcap authorization for PA during the SUBSCRITPION message servicing. I have the following code which work fine when auth parameter is set to 'none': if(lookup_user("$tu.uid", "@ruri")) { if (!t_newtran()) sl_reply_error(); drop; } (handle_subscription("registrar")) ... } but when I switch to auth 'xcap' I cant see any query send to XCAP server simulated on apache according to the presence handbook. Please point me what do I miss. Thanks Cheers Tomasz

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[Serusers] xl_mod_init warning
by Todd Lewis 02 Jul '07

02 Jul '07

I using the example configuration file that comes with ser-2.0-rc1 No matter what I do, I cannot figure out what's causing this warning /usr/local/ser/sbin/ser[43254]: Maxfwd module- initializing /usr/local/ser/sbin/ser[43254]: WARNING: xl_mod_init: more IP 10.0.0.6 not used I have two ip's assigned to the server, using one Nic card Static IP 10.0.0.6 Virtual IP 10.0.0.4 NicCard em0 I have tried -------------------- listen=10.0.0.4 fork=yes/no --------------------- -------------------- listen-10.0.0.6 fork=yes/no -------------------- -------------------- listen=10.0.0.4 listen=10.0.0.6 fork=yes/no -------------------- -------------------- listen=em0 listen=10.0.0.4 fork=yes/no -------------------- -------------------- listen=em0 fork=yes/no -------------------- I always get the same error message. Any ideas?

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Re: [OpenSER-Users] FreeRADIUS-CDRTool Rel. 1.0
by Dan-Cristian Bogos 02 Jul '07

02 Jul '07

Hi, my code has no connection to any BBUA. It is just providing the maximum call duration information back to OpenSER which can later do what u want it to do with this info. I personally use yate (yate.null.ro) to provide call cut when the calls are going out on PSTN, by adding an extra SIP Header in OpenSER and then later process it in Yate to timeout the calls. I guess there are unlimited possibilities later, if u have the information available in OpenSER AVPs, including creating your own timer module which sends a session cancelation order to RTPProxy (heard that it is possible to kill the calls with RTPProxy, but never done it). Let me know if I was clear enough. Cheers, DanB On 7/2/07, A.M. <agentm(a)themactionfaction.com> wrote: > > On Jul 2, 2007, at 11:10 , Dan-Cristian Bogos wrote: > > > All, > > > > as a personal gratitude for all the efforts involved in opensource > > projects, especially in OpenSER community, I would like to contribute > > myself with a bit of code, investing in something I considered useful > > for this community. > > > > Hereby, I inform u that the first release of FreeRADIUS-CDRTool > > connector module is available for download and use at sourceforge.net. > > Project link: http://sourceforge.net/projects/frad-cdrtool/ > > > > Basically, FreeRADIUS-CDRTool is a FreeRADIUS module written in python > > (therefore one might need to compile rlm_python in FreeRADIUS to make > > it work), able to connect on CDRTool Engine via the Telnet socket (the > > only way I found recommended by CDRTool folks) and perform different > > actions via Prepaid API, like return maximum value for the call, > > instruct CDRTool to lock accounts already in a call, instruct > > unlocking at end of the call and balance debiting, etc. This module > > should be used together with OpenSER radius authentication and > > accounting. > > If proper configured, FreeRADIUS-CDRTool can authorize calls for > > CDRTool Prepaid users. At the starting of each call the module will > > return the maximum call duration allowed, information which can be > > processed later in a bbua scenario to timeout the calls reaching 0 > > balance, and total credit available at starting of the call, all > > through SIP-AVP Radius Attributes. > > Hi- thanks for the information. How does your code interact with the > b2bua? Which b2buas are supported? > > Cheers, > M >

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[OpenSER-Users] FreeRADIUS-CDRTool Rel. 1.0
by Dan-Cristian Bogos 02 Jul '07

02 Jul '07

All, as a personal gratitude for all the efforts involved in opensource projects, especially in OpenSER community, I would like to contribute myself with a bit of code, investing in something I considered useful for this community. Hereby, I inform u that the first release of FreeRADIUS-CDRTool connector module is available for download and use at sourceforge.net. Project link: http://sourceforge.net/projects/frad-cdrtool/ Basically, FreeRADIUS-CDRTool is a FreeRADIUS module written in python (therefore one might need to compile rlm_python in FreeRADIUS to make it work), able to connect on CDRTool Engine via the Telnet socket (the only way I found recommended by CDRTool folks) and perform different actions via Prepaid API, like return maximum value for the call, instruct CDRTool to lock accounts already in a call, instruct unlocking at end of the call and balance debiting, etc. This module should be used together with OpenSER radius authentication and accounting. If proper configured, FreeRADIUS-CDRTool can authorize calls for CDRTool Prepaid users. At the starting of each call the module will return the maximum call duration allowed, information which can be processed later in a bbua scenario to timeout the calls reaching 0 balance, and total credit available at starting of the call, all through SIP-AVP Radius Attributes. I have tested the module in a Debian Linux environment, python2.4 and FreeRADIUS 1.6, CDRTool version 5.1.4, on low call load. I would not recommend yet using it in a production environment till we get some stability feedback. Have fun! Kind regards, DanB

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[OpenSER-Users] mediaproxy StopRecordSerializer when closing channel
by Dzejms 02 Jul '07

02 Jul '07

Hi, I have this common problem Jul 2 16:03:25 localhost mediaproxy[5566]: error: uncaptured python exception, closing channel <rtphandler.CommandHandler connected xxx.xxx.xxx.xxx:46948 at -0x5f92e794> (exceptions.NameError:global name 'StopRecordSerializer' is not defined [asyncore.py|readwrite|94] [asyncore.py|handle_read_event|391] [/usr/local/mediaproxy/modules/rtphandler.py|handle_read|441] [/usr/local/mediaproxy/modules/rtphandler.py|process|569]) Mediaproxy version is 1.8.2 it's running on debian 4.0 etch, witch python 2.4 (also tried 2.5), i have python-mysqldb module installed, also python-dev-all . Problem still exist. When i'm using 1.7.2 version od mediaproxy, it seems to be ok. I'm also using 1.8.2 mediaproxy version on debian sarge with python 2.3 where everything seems to be ok. Any ideas? Broken debian packages? Dzejms.

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[OpenSER-Users] manipulate INVITES
by Patrick Miccio 02 Jul '07

02 Jul '07

Hello everybody, since this is my first post and I am kinda new to OpenSER, bare with me ;) I have been trying to manipulate invites for the last 2 days and cant find the proper way to reach what is expected. my goal: send invites to specific PBX (Asterisk for example) with only the desired extension and not the full username, this is based on a avp_check if that UAC needs only extensions, some do, some don't ;) Here is what I do: First off all I extract the real username (in subscribers) and extension from the R-URI and place it in "$avp(callee_user)" and "$avp(callee_exten)" ############# my solution No1 is "working" avp_pushto("$ru/username", "$avp(callee_user)"); avp_copy("$avp(callee_exten)","$avp(callee_tmp)/gd"); if( subst('/^To:(.*)sip:[^@]*(@[a-zA-Z0-9.]+.*)$/To:\1sip:$avp(callee_tmp)\2/ig') ){}; -> callee_user is 012345678 -> callee_exten is 200 -> $ru is sip:012345678@xxx.xxx.xxx.xxx;user=phone -> subst [ To: <sip:012345678200@xxx.xxx.xxx.xxx;user=phone> ] with [To: <sip:200@xxx.xxx.xxx.xxx;user=phone> ] Those 3 lines work "fine", the testing Asterisk PBX gets the call correctly. The problem is that I should not change the "To:" part, it will make problems with call forwards etc... ############# my solution No.2 NOT working. if (subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(callee_tmp)@\2;/i')){$}; doesn't work because OpenSER of course cant find 200 in the userloc.... ############# Generally I would think that changing the INVITE uri itself like in solution No.2 is the best way, couldn't find any proper way to make that work though. Does anyone have an idea how to realize this properly, or even if this makes sense ;) ? thx in advance, Patrick.

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[OpenSER-Users] help with using avp...
by liran tal 02 Jul '07

02 Jul '07

Hey everyone, I'm using OpenSER 1.1.1 SVN version. I'd like to save the Authorization Digest Username as it appears in header into a variable (avp) so that later I can use it to set an accounting attribute in the radius. >From the SIP Packet the information is given like this: Authorization: Digest username="dsdsds", realm="1.2.3.4", algorithm=MD5, uri="sip:1.2.3.4", nonce="xxx" Where I would like to grab the username part "dsdsds". >From the docs I could only conclude that doing it would be something like this: avp_write("$hdr[Authorization]","s:12"); avp_subst("$avp(s:12)", "/.*username=(.*),/\1/"); I'm not even sure about that piece of code, if someone can help... Thanks, Lir.

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[Serusers] RLS module in ser 2_0_0 - subscriptions handling
by tzieleniewski 02 Jul '07

02 Jul '07

Hi, I am using the RLS module for RLS service and resource list subscriptions. Ive done everything according to the presence handbook. When SER sends NOTIFY messages to the subscribing UAs every instances of the subscribed resources is in the pending state. How are those subscriptions handled by ser?? Do I have to include presence_b2b molude so that those subscriptions in the pending state would be handled?? Thanks in advance Cheers Tomasz

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[Serusers] routing rtpstream through nat within vpn
by Martin Drasar 02 Jul '07

02 Jul '07

Hello everyone, I have following scenario: +----------+ | VPN | |+--------+| || MRCP || |+--------+| | || | |+--------+| +---------+ || Proxy ====| Solaris | |+--------+| +---------+ +----------+ I want both sip messages and rtp stream to go through proxy between mrcp servers and solaris box. Currently I am transferring sip messages using call forwarding (avpops module) and trying to use rtpproxy for rtp stream forwarding. For rtpproxy to work I have set up NAT on proxy, but still have issues. This is the invite packet sent from proxy to MRCP server: INVITE sip:UAS@10.1.1.100:5060 SIP/2.0 Record-Route: <sip:10.0.0.1;ftag=3d423348;lr=on> <=## Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKe359.5e010b95.0 <=## Via: SIP/2.0/UDP 10.0.0.6:5070;branch=z9hG4bK-d8754z-5212be016a2d3c2a-1---d8754z-;rport=5070 Max-Forwards: 16 Contact: <sip:sender@10.0.0.6:5070> To: <sip:synthesizer@10.0.0.1:5060;transport=UDP> <=## From: <sip:sender@10.0.0.6:5070>;tag=3d423348 Call-ID: MTZkYWE2MGYxNTgyNmY0YTRmNzkwNjAyNmU5OGQ0ZGU. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY Content-Type: application/sdp Content-Length: 282 v=0 o=- 1183369050 1183369050 IN IP4 10.0.0.6 s=- t=0 0 m=application 35050 TCP/MRCPv2 <=## c=IN IP4 10.0.0.1 <=## a=connection:new a=setup:active a=cmid:1 a=resource:speechsynth m=audio 53384 RTP/AVP 0 c=IN IP4 10.0.0.1 <=## a=mid:1 a=recvonly a=direction:active a=nortpproxy:yes Lines with "<=##" are the lines with which I think I have problem. Because mrcp servers receive 'To' and 'Via' with address that is outside vpn, they cannot reply. As a result I send x times invite and end with 408 timeout. Also there is problem with line "m=application 35050 TCP/MRCPv2". rtpproxy changes that port, but according to mrcp specification I need it to be exactly 9. Even when changing PORT_MIN and PORT_MAX both to 9, I sometimes get port 10. Could you please advice me what to do? Am I trying in right direction? Will I have to alter the sources to make it work like I need to? Thanks for your answers Martin

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Re: [Serusers] How SER communicate to other SIP server
by roayu 02 Jul '07

02 Jul '07

Hi, Olaf! I'm not sure what functions that I can use to include inside ser.cfg. Could you pls let me know where I can see all those functions like sl_send_reply(), so that I can use them. Thanks. Best regards, Roa Yu -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu Sent: Friday, June 29, 2007 8:47 AM To: Olaf Bergmann Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server Hi, Olaf! I'll have a look and try on it. Thanks. Best regards, Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Thursday, June 28, 2007 5:34 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > Hi, Olaf! > > Ok, but which section that I need to concentrate on the most? I've search > through the whole documentation and I can't find the configuration/setting > on the ser.cfg file. Really appreciate on your help... Look at the "Hello World ser.cfg" <http://siprouter.onsip.org/doc/gettingstarted/ch06.html>. For unknown SIP domains, route[1] does the DNS-based forwarding mentioned before. You should give it a try and see how it works. HTH, Olaf _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[OpenSER-Users] not able to run openser
by Vishwadeepak Agarwal 02 Jul '07

02 Jul '07

hi all, I have installed openser. when i gave ./openser.init start i got this Command Monit to start OpenSER... ./openser.init: line 64: /usr/local/bin/monit: No such file or directory when i tried to run ./openser i got this ERROR: bad config file (16 errors) can anyone help? thanks in advance regards, Vishwadeepak ______________________________________________________________________

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[Serusers] Efficient config and routing of 100's of prefix's
by Shaun Hofer 29 Jun '07

29 Jun '07

Hi, To terminate a call to the PSTN, my SER box has to route between multiple PSTN gateways depending on the destination phone number/prefix. For instance, -Use gateway1 if the destination begins with any of these prefixes: 07550[0-4] 075509 07551[0-2] 075514 075515[7-8] 0755187 0755337 07553[7-9] 075551[0-1] 07555[2-8] 07556[0-4] etc.. -Use gateway2 if the destination begins with any of these prefixes: 0730[0-2] 07303[0-6] 0731[0-2] 073131 07313[5-9] 073140 07314[4-6] 0731[6-7] 0732[1-7] 07329[1-2] 07329[5-6] etc.. -Use gateway3 if etc... -Use gatewayX if etc... Obviously, one or a small number of regular expressions can't efficiently cover all of these prefixes. However, doing it like this isn't ideal because ser.cfg would be huge and cumbersome. Also, the prefixes for each gateway may change over time: if ( (uri =~ "^sip:07550[0-4]") || (uri =~ "^sip:075509") || ... ) { route to gateway1 } if ( (uri =~ "^sip:0730[0-2]") || (uri =~ "^sip:07303[0-6]") || ... ) { route to gateway2 } if ( ... ) { route to gatewayX } What's the most efficient way of doing this with SER? Thanks in advanced -Shaun

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[Serusers] Difficult NAT issues :external UAs behind NAT and internal UAs on private IPs but _not_ NATted
by Michael Grigoni 29 Jun '07

29 Jun '07

Greetings: I really need some fresh eyes on my ser.cfg and help with this difficult NAT issue: our internal UAs are on private IPs, registered with ser running on the border router (it has aliases for the private IP internal LANs and the public IP on which it listens). Calls between UAs on public IPs and our internal UAs (on their private IPs) work well. Calls between UAs behind NAT (from the Internet) and our internal UAs (on their private IPs) have no audio from us to them. Only private IP addresses in routes from the Internet must be rewritten for NAT; our internal UAs on their private IPs must be routed without change (which works if the foreign UA isn't behind NAT). We are running '0.8.99-dev19 openbsd/sparc' and 'rtpproxy 1.21 2004/08/13' Our 'ser.cfg', a packet dump of a call from an Internet UA behind NAT ('adams.dmp') and a segment of the logfile ('serlog') are at: http://www.cybertheque.org/sw/ser/configs All help is truly appreciated. Michael Grigoni Cybertheque Museum

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[OpenSER-Users] Integrating XCAP-lite with openser
by ashok.akumar＠wipro.com 29 Jun '07

29 Jun '07

Hi all, I have downloaded XCAP-lite to support XCAP. I already have openser running. But the problem is how to integrate XCAP-lite with openser? I guess for that I need to configure config.php file in xcap-root directory of XCAP-lite. Can anyone tell me how to configure config.php or what i need to do for integrating XCAP-lite with openser? Any help will be appreciated. Rgds, Ashok The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[OpenSER-Users] mysql.so not included in 0.9.5
by Rizwan Hisham 29 Jun '07

29 Jun '07

Hi all, I installed the openser 0.9.5. when i tried to include mysql module in openser.cfg, it gave me this error. 0(8531) ERROR: load_module: could not open module </usr/local/lib/openser/modules/mysql.so>: /usr/local/lib/openser/modules/mysql.so: cannot open shared object file: No such file or directory 0(8531) parse error (28,13-14): failed to load module I checked the source directory. It has the mysql module files included in its source then why it did not compile them and place mysql.so in the required directory. This is how i installed: -->make -->make modules -->make all -->make install Anybody with answeres ??? -- Best Regards Rizwan Hisham Software Engineer AXVOICE Inc. www.axvoice.com

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Subject=[OpenSER-Users] openser.cfg debug howto
by Patrick Miccio 29 Jun '07

29 Jun '07

just use: /path/to/openser -c -f /path/to/openser.cfg cheers, Patrick.

1 0

[OpenSER-Users] Re: A problem with CDRTool
by Dan-Cristian Bogos 29 Jun '07

29 Jun '07

Hi Jacky, looks like your rating engine is not running. Did u install and start it? I am using debian packages for cdrtool and it is automatically installed. If u used same, all u must do is edit the /etc/defaults/cdrtool to enable both ratingengine as well as memcached and then start cdrtool with /etc/init.d/cdrtool start. Hope that helps. Cheers, DanB

1 0

[OpenSER-Users] accounting parallel invite
by yoshiki 29 Jun '07

29 Jun '07

hello ! I am weak in English. Please teach it about a accounting request-uri on parallel invite. UAC ----> OpenSER ----> UAS1 | ----> UAS2 1. UAC call to UAS. e.g. from: UAC(a)test.com -> to: UAS(a)test.com 2. avpop module in OpenSER search UAS. 3. hit UAS1 and UAS2. 4. I enumerate contact address that was found from a location table to request-URI. 5. OpenSER relay to UAS1 and UAS2. 6. UAS1 pick up call. 7. acc module account to database src_user UAC dst_user UAS1 But I want to account parallel number (UAS). How may you write in UAS to database ? -------------------------------------- Start Yahoo! Auction now! Check out the cool campaign http://pr.mail.yahoo.co.jp/auction/

2 1

[OpenSER-Users] Openser as reverse proxy
by hdpml 29 Jun '07

29 Jun '07

Dear Mailinglist, I am a rookie in SER and joined this mailinglist a few minutes ago. Hope you can help me with my question. I found ser at my search for a sip / rtp proxy. I want to install the following scenario: client -> Internet or WLAN -> SER proxy + rtpproxy(DMZ) -> Asterisk (Internal LAN) Is it possible with openser that the authentication is passed through to asterisk and openser accept the call if the user is valid? My idea is, not to have any userdata on the SER proxy. I just want that the proxy forwards the request to the asteriskserver and checks if the REGISTER is OK or not. If the user is invalid, the proxy should deny requests from this user to the asteriskserver. If this will be possible, is it possible to integrate encryption of RTP and SIP too? Like a http reverseproxy? Thank you very much. Dominic HDPnet GmbH Erwin-Rohde-Str. 18 69120 Heidelberg Geschaeftsfuehrer: Marc Hermann Registergericht: Mannheim HRB 337012 Sitz: Heidelberg Umsatzsteuer ID Nr.: DE 211 257 470 www.hdpnet.de Diese E-Mail enthaelt vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtuemlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet.

2 1

[OpenSER-Users] The Via Header of CANCEL Message
by fancy 29 Jun '07

29 Jun '07

According to RFC 3665 section 3.8 and RFC 3261 section 9, there is only one via header in CANCEL message (message F11, F13 at page 72 of RFC 3665). But when I ran the similar scenario(through one proxy), I got two via header in CANCEL message such as following: CANCEL sip:UA12@node11.under.test.com;transport=udp SIP/2.0 Record-Route: <sip:192.0.50.50;lr=on;ftag=11380> Via: SIP/2.0/UDP ss.under.test.com;branch=z9hG4bKd068.25909417.0 Via: SIP/2.0/UDP node.under.test.com:5060;received=192.0.10.10;branch=z9hG4bKPUA4295483 Max-Forwards: 69 From: UA11 <sip:UA11@under.test.com>;tag=11380 To: <sip:UA12@under.test.com> Call-ID: 11380(a)tom.example.com CSeq: 4 CANCEL Content-Length: 0 P-hint: usrloc applied How can I remove the 2nd via header? Thank you Best Regards, Fangyu Ling

2 1

Re: [OpenSER-Users] Openser 1.2.1 with FreeRADIUS 1.1.6 with MySQL 5.0.41
by Neeraj Gupta 29 Jun '07

29 Jun '07

1 0

[OpenSER-Users] A problem with CDRTool
by Lee Jacky 29 Jun '07

29 Jun '07

Hi,all. I have met a problem with CDRTool. I am using openser1.1+CDRTool5.1.4. When I open CDRTool's web page,and click the Rating link,then I get lots of records, but the following warning is also showed: Warning: fsockopen() [function.fsockopen]: unable to connect to IP-Address:9024 in /usr/local/apache/htdocs/CDRTool/rating_lib.phtml on line 2492 | Cannot connect to rating engine IP-Address:9024 Had anyone met this before? Any tips will be appreciated. ___________________________________________________________ Mp3疯狂搜-新歌热歌高速下 http://music.yahoo.com.cn/?source=mail_mailbox_footer

1 0

Re: [Serusers] How SER communicate to other SIP server
by roayu 29 Jun '07

29 Jun '07

Hi, Olaf! I'll have a look and try on it. Thanks. Best regards, Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Thursday, June 28, 2007 5:34 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > Hi, Olaf! > > Ok, but which section that I need to concentrate on the most? I've search > through the whole documentation and I can't find the configuration/setting > on the ser.cfg file. Really appreciate on your help... Look at the "Hello World ser.cfg" <http://siprouter.onsip.org/doc/gettingstarted/ch06.html>. For unknown SIP domains, route[1] does the DNS-based forwarding mentioned before. You should give it a try and see how it works. HTH, Olaf

1 0

Re: [Serusers] add prefix to source
by Szasz Szabolcs 28 Jun '07

28 Jun '07

Hi! I know the prefix() and the strip() functions, but how can I change the from_uri (caller)? Thanks for the help. > > ----- Original Message ----- > From: "SIP" <sip(a)arcdiv.com> > To: "Szasz Szabolcs" <sszasz(a)digicomm.ro> > Sent: Monday, June 25, 2007 12:34 AM > Subject: Re: [Serusers] add prefix to source > > >> Use the prefix() function. >> >> i.e. to add the prefix 555 onto the beginning of a URI, >> >> prefix("555"); >> >> N. >> >> >> Szasz Szabolcs wrote: >>> Hi all >>> I need to add a prefix to the source uri/number. >>> How ca I do this? Can you help me guys? >>> Thanks a lot >>> Szasz Szabolcs >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> -- >> This message was scanned for spam and viruses by BitDefender. >> For more information please visit http://linux.bitdefender.com/ >> >> > -- This message was scanned for spam and viruses by BitDefender. For more information please visit http://linux.bitdefender.com/

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[Serusers] Error 483 response to ACK and BYE
by SIP 28 Jun '07

28 Jun '07

In response to a BYE and an ACK from some (not all) servers that send to our proxy, our server's spitting back a 483 'Too many hops' reply. It doesn't happen with all peers, and it certainly doesn't happen locally... but it does happen with some, and I'm not entirely sure as to why. The 483 block at the beginning is normal: if (msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; Exchange looks like this: U 198.65.166.131:5060 -> 63.64.65.66:5060 BYE sip:1101XXXXXXX@63.64.65.66:5060 SIP/2.0. Record-Route: <sip:198.65.166.131;ftag=gpp0q468oh;lr>. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK9b05.b75fbe14.0. Via: SIP/2.0/UDP 192.168.1.51:2054;received=63.XX.XX.XX;branch=z9hG4bK-wdtud2nffq88;rport=2054. Route: <sip:63.64.65.66;ftag=gpp0q468oh;lr=on>. From: "User One" <sip:1747XXXXXXX@their.proxy.server:5060>;tag=gpp0q468oh. To: <sip:1101XXXXXXX@their.proxy.server;user=phone>;tag=as7fbbcb66. Call-ID: 3c267038e7ef-586eg1emcfsi@snom190. CSeq: 2 BYE. Max-Forwards: 16. Contact: <sip:1747XXXXXXX@192.168.1.51:2054;line=oxd9zlst;nat=yes>. User-Agent: snom190/3.60x. Content-Length: 0. RemoteIP: 63.XX.XX.XX. P-hint: rr-enforced. P-NATed-URI: YES (1). P-RTP-Proxy: YES (1). U 63.64.65.66:5060 -> 198.65.166.131:5060 SIP/2.0 483 Too Many Hops. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK9b05.b75fbe14.0. Via: SIP/2.0/UDP 192.168.1.51:2054;received=63.XX.XX.XX;branch=z9hG4bK-wdtud2nffq88;rport=2054. From: "User One" <sip:1747XXXXXXX@their.proxy.server:5060>;tag=gpp0q468oh. To: <sip:1101XXXXXXX@their.proxy.server;user=phone>;tag=as7fbbcb66. Call-ID: 3c267038e7ef-586eg1emcfsi@snom190. CSeq: 2 BYE. Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 63.64.65.66:5060 "Noisy feedback tells: pid=3488 req_src_ip=63.64.65.66 req_src_port=5060 in_uri=sip:1101XXXXXXX@63.64.65.66:5060 out_uri=sip:1101XXXXXXX@63.64.65.66:5060 via_cnt==18". What would be causing this? Is it because the IP address is being used in the URI to us as opposed to the domain (I tried adding the IP to the domain table and to an alias line alternately, but it didn't fix things) ? It's irksome in that it only happens with certain peers and not others, so there's something in the way we're handling messages from them that's not right or different, but since we handle all incoming the same way, I'm at a loss as to why it works with some but not with others. N.

4 9

[OpenSER-Users] Any demo for stateful proxy setting with 301/302 forwarded?
by Lucian Romi 28 Jun '07

28 Jun '07

Hi, I'm using openser basic config now. And I'm totally a freebie here. I want to set up a system with handle 301/302 forward for INVITE sending from openser, so it must be a stateful proxy and seems like there must be something in ser.cfg to support this. If don't, then openser will forward 301 and 302 all the way back, that's not I want. I'm really appreciate any help. Thanks!!!!

2 2

[OpenSER-Users] Openser 1.2.1 with FreeRADIUS 1.1.6
by Neeraj Gupta 28 Jun '07

28 Jun '07

2 4

[OpenSER-Users] NEW MODULE: Datagram implementation for MI
by Bogdan-Andrei Iancu 28 Jun '07

28 Jun '07

Hi, thanks to Ancuta Onofrei, we added today a new transport implementation for the Management Interface (MI). This is Datagram (via unix and network sockets) and it completes the migration to MI (after fifo), but not the last from the transport implementations :). It is in alpha stage, so more testing is needed. NOTE that all MI commands are available via datagram interface; also the syntax of the commands and replies is exactly as for FIFO. For more info, please see the online documentation: http://www.openser.org/docs/modules/1.2.x/mi_datagram.html or ask on the mailing list. Any feedback - bugs or improvements - are welcomed. Also I want to bring in discussion a related subject - dropping the old unix sockets implementation from the core. Before doing it I would like to know who is still using it and if there are any issues with migrating to the new implementation. Regards, Bogdan

4 9

[OpenSER-Users] openser.cfg debug howto
by Rizwan Hisham 28 Jun '07

28 Jun '07

Hi all, I am a new user of openser.i just installed and configured it to send and recieve calls. im having some problems with scripting. Whenever there is a problem in openser.cfg, it just displays the number of errors. is there anyway to know at what line the error is and also what kind of error. As im a new user and so i dont know the syntax and symantics of the scripting very well, im not very good at finding the error by looking at the file. Hopefully this is possible -- Best Regards Rizwan Hisham Software Engineer AXVOICE Inc. www.axvoice.com

1 0

[OpenSER-Users] Passing information/Headers
by Kelvin Williams 28 Jun '07

28 Jun '07

I need to pass information between OpenSER and Asterisk for a couple applications I have created. I figured the best way to do this would be to append a header field and have either OpenSER or Asterisk peel it off and take a look at what it says. However, I've found that Asterisk some times gets perturbed by the additional header field. Is it possible/wise to create your own headers? What other way could I pass information (used for scripts in Asterisk/OpenSER) between the two servers? Thanks, kw

2 1

Re: [Serusers] How SER communicate to other SIP server
by roayu 28 Jun '07

28 Jun '07

Hi, Olaf! Ok, but which section that I need to concentrate on the most? I've search through the whole documentation and I can't find the configuration/setting on the ser.cfg file. Really appreciate on your help... Thanks. Best regards, Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Thursday, June 28, 2007 4:20 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > Hi, Olaf! > > Is there any settings that I need to do on SER or ser.cfg? Sure. At this point, you really should read <http://www.iptel.org/ser/doc/gettingstarted> before proceeding. Best regards, Olaf

2 1

[OpenSER-Users] wengophone and openser
by Radu Spineanu 28 Jun '07

28 Jun '07

Hi, I have a working basic openser + asterisk setup. Twinkle works OK and wengophone doesn't. It tries to connect and then after a while times out. While debugging I noticed a lot of OPTIONS ping requests are going unanswered by openser because it contains an user in the uri. Registration seems to be working OK. I've attached the ngrep output that shows this. Has anyone got wengophone to work with openser? -- Radu Spineanu interface: eth0 (217.73.174.0/255.255.255.224) filter: (ip or ip6) and ( port 5060 ) # U 82.79.243.82:5060 -> 217.73.174.13:5060 REGISTER sip:217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=123456789..Route: <sip:217.73.174.13;lr>..From: nobo dy <sip:nobody@217.73.174.13>;tag=123456789..To: <sip:nobody@217.73.174.13>..Call-ID: 000001@ping..Contact: <sip:nobody@82.79.243.82>..C Seq: 2 REGISTER..Content-Length: 0.... # U 217.73.174.13:5060 -> 82.79.243.82:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 82.79.243.82:5060;rport=5060;branch=123456789..From: nobody <sip:nobody@217.73.174.13>;tag=12 3456789..To: <sip:nobody@217.73.174.13>;tag=5e02e7658def22f683939a852e493344.3f4e..Call-ID: 000001@ping..CSeq: 2 REGISTER..WWW-Authentic ate: Digest realm="217.73.174.13", nonce="46823bd8e5f68588801e0655bbc907e2425ee03f"..Server: OpenSER (1.2.1-notls (x86_64/linux))..Conte nt-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 REGISTER sip:217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1094221745..Route: <sip:217.73.174.13:5060;lr >..From: <sip:1001@217.73.174.13>;tag=1065314925..To: <sip:1001@217.73.174.13>..Call-ID: 807187534@82.79.243.82..CSeq: 1 REGISTER..Conta ct: <sip:1001@82.79.243.82:5060>..Max-Forwards: 70..User-Agent: wengo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 2940..Content-Length: 0.... # U 217.73.174.13:5060 -> 82.79.243.82:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 82.79.243.82:5060;rport=5060;branch=z9hG4bK1094221745..From: <sip:1001@217.73.174.13>;tag=106 5314925..To: <sip:1001@217.73.174.13>;tag=5e02e7658def22f683939a852e493344.4bcc..Call-ID: 807187534@82.79.243.82..CSeq: 1 REGISTER..WWW- Authenticate: Digest realm="217.73.174.13", nonce="46823bd9c467d1377ec65d343c7dfc7bd40513c3"..Server: OpenSER (1.2.1-notls (x86_64/linux ))..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 REGISTER sip:217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK2031845783;xxx-nat-type=sym..Route: <sip:217. 73.174.13:5060;lr>..From: <sip:1001@217.73.174.13>;tag=1065314925..To: <sip:1001@217.73.174.13>..Call-ID: 807187534@82.79.243.82..CSeq: 2 REGISTER..Contact: <sip:1001@82.79.243.82:5060>..Authorization: Digest username="1001", realm="217.73.174.13", nonce="46823bd9c467d137 7ec65d343c7dfc7bd40513c3", uri="sip:217.73.174.13", response="2e3d2c8bb188495863d4306f5cd5ba5c", algorithm=MD5..Max-Forwards: 70..User-A gent: wengo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 2940..Content-Length: 0.... # U 217.73.174.13:5060 -> 82.79.243.82:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 82.79.243.82:5060;rport=5060;branch=z9hG4bK2031845783;xxx-nat-type=sym..From: <sip:1001@217.73.174.13>; tag=1065314925..To: <sip:1001@217.73.174.13>;tag=5e02e7658def22f683939a852e493344.142c..Call-ID: 807187534@82.79.243.82..CSeq: 2 REGISTE R..Contact: <sip:1001@82.79.243.82:5060>;expires=2940..Server: OpenSER (1.2.1-notls (x86_64/linux))..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... # U 82.79.243.82:5060 -> 217.73.174.13:5060 OPTIONS sip:ping@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5060;rport;branch=z9hG4bK1028071012..From: Radu <sip:1001@217.73.1 74.13>;tag=675779892..To: <sip:ping@217.73.174.13>..Call-ID: 1333741103@82.79.243.82..CSeq: 20 OPTIONS..Max-Forwards: 70..User-Agent: we ngo/v1/wengophoneng/wengo/rev0/trunk/..Expires: 120..Accept: application/sdp..Content-Length: 0.... exit 14 received, 0 dropped

3 2

Re: [Serusers] How SER communicate to other SIP server
by roayu 28 Jun '07

28 Jun '07

Hi, Olaf! Is there any settings that I need to do on SER or ser.cfg? Thanks. Best Regards, Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Thursday, June 28, 2007 3:04 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > Hi, Olaf! > > Thanks for your reply. Could you please further explain on the connectivity > between those to DIFFERENT sip servers? If I wanna call using a softphone to > call from SER to SIPserver B, how should I make the call? In a typical scenario, you configure your softphone to use SER as outbound proxy (registrar, etc.) and let it route outgoing SIP requests according to its configuration. When calling foreign domains (i.e., the callee has a domain that is different from your softphone's), a simple t_relay would do. More elaborate routing policies could be based on numeric "dial plans" etc. to forward SIP requests according to your needs. Best regards, Olaf

2 1

[OpenSER-Users] Does SER/OpenSER has the ability to continue searching for other routes despite 4xx/5xx/6xx responses?
by Jason Ma 28 Jun '07

28 Jun '07

Hi all, Does SER/OpenSER has the ability to continue searching for other routes despite 4xx/5xx/6xx responses?

2 1

Re: [Serusers] How SER communicate to other SIP server
by roayu 28 Jun '07

28 Jun '07

Hi, Olaf! Thanks for your reply. Could you please further explain on the connectivity between those to DIFFERENT sip servers? If I wanna call using a softphone to call from SER to SIPserver B, how should I make the call? Thanks. Best Regards, Roa Yu -----Original Message----- From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de] Sent: Wednesday, June 27, 2007 3:10 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How SER communicate to other SIP server roayu wrote: > I would like to find out is there anyway that the SER server can > communicate to other SIP servers? You could use SIP for this purpose. It has turned out to work pretty well since all SIP servers implement it by design ;-) > For instance, I have an account at SER > and another account at SIPserver B. I would like to call from SER to > SIPserver B, and vice versa. Is it workable? Or any additional devices > needed in between these 2 SIP servers? If yes, could anyone give me some > clues or guide to do so. You might want to have a look at the documents section at <http://iptel.org> , especially <http://www.iptel.org/ser/doc/gettingstarted>. HTH, Olaf

2 1

[Serusers] Floating point exception
by bozhang＠nec-labs.com 28 Jun '07

28 Jun '07

Hi, I download the ser software and I can compile it successfully on my machine with Fedora core 4. But when I tried to run it with the 'hello world ser.cfg' configuration file, it reports "Floating point exception" error message. Any body had similar error before? or any idea what is happening? I tried to copy the compiled binary to other machines with FC4 , all gave me the "Floating point exception" error. any input will be appreacited! -Bo

2 1

Re: [OpenSER-Users] About the opensip server.
by Henning Westerholt 28 Jun '07

28 Jun '07

On Mittwoch, 27. Juni 2007, gaoqiang.qian(a)nokia.com wrote: > Yes, I have the database in the msyql server by the openser_mysql.sh > create command, I can see it. > But I am not sure with the openser_mysql.sh create, is it enough for > adding those two users openserro and openserrw in the database? Normally this should be sufficient. But better check to be safe. Please reply also to the mailing list, that others can see your questions and the answers too. Henning

2 1

[OpenSER-Users] totally new
by Andres Paglayan 27 Jun '07

27 Jun '07

Hi, I did a little reading through the documentation at openser.org looking for something like a "tutorial" and couldn't find one, is there any kind of tutorial like resource available? Thanks, Andres Paglayan --"Harmony is more important than being right" Bapak

2 1

[OpenSER-Users] multiple NAPTR records in ENUM module
by Jonathan Creasy 27 Jun '07

27 Jun '07

Is there a way to have OpenSER try each NAPTR record when multiple are returned? 4.3.2.1.5.5.5.5.5.5.1.mysite.enum. 0 IN NAPTR 100 15 "u" "E2U+sip" "!^.*$!sip:+15555551234@server2" . 4.3.2.1.5.5.5.5.5.5.1.mysite.enum. 0 IN NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:+15555551234@server1" . 4.3.2.1.5.5.5.5.5.5.1.mysite.enum. 0 IN NAPTR 100 25 "u" "E2U+sip" "!^.*$!sip:+15555551234@server3" . That would route the calls to server1 (order 100, preference 10) but if server1 is not responding I would like it to move on and try server2. -Jonathan

2 7

[Users] Malformed CANCEL request from Asterisk possibly impacting OpenSER?
by David Coulthart 27 Jun '07

27 Jun '07

We are currently experiencing problems with our production OpenSER (1.1.0) environment where the load on the server is steadily increasing and the OpenSER response time increases drastically (e.g., 30+ seconds for an internal call to ring). Capturing the network traffic when this happens we have noticed that our Asterisk (1.2.10) voicemail server is sending malformed CANCEL requests to our OpenSER proxy. We are not certain whether this is a cause or an effect of the load problems on the OpenSER server. Here is the SIP data from the bad CANCEL request: Session Initiation Protocol Request-Line: CANCEL :u43974@128.59.59.215:5069 SIP/2.0 Method: CANCEL [Resent Packet: False] Message Header Via: SIP/2.0/UDP 128.59.59.215:5069;branch=z9hG4bK6734835b;rport Transport: UDP Sent-by Address: 128.59.59.215 Sent-by port: 5069 Branch: z9hG4bK6734835b RPort: rport Route: <sip:128.59.59.199;lr=on;ftag=4FC702F4-7F812557> From: <sip:43974@sip.columbia.edu;user=phone> SIP from address: sip:43974@sip.columbia.edu To: "Jim Rios" <sip:47891@sip.columbia.edu>;tag=4FC702F4-7F812557 SIP Display info: "Jim Rios" SIP to address: sip:47891@sip.columbia.edu SIP tag: 4FC702F4-7F812557 Contact: <sip:u43974@128.59.59.215:5069> Contact Binding: <sip:u43974@128.59.59.215:5069> URI: <sip:u43974@128.59.59.215:5069> SIP contact address: sip:u43974@128.59.59.215:5069 Call-ID: c9a0135b-41273fb6-f7b6cea9(a)160.39.14.100 CSeq: 101 CANCEL Sequence Number: 101 Method: CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Accompanying this bad CANCEL request are the following error messages recorded by OpenSER to syslog when it attempts to process it: May 29 15:34:07 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26) May 29 15:34:07 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069> May 29 15:34:07 cocoa openser[892]: xl_get_ruri: ERROR while parsing the R-URI May 29 15:34:08 cocoa openser[892]: c9a0135b-41273fb6-f7b6cea9(a)160.39.14.100: route[0] CANCEL r-uri <<null>> from <sip:43974@sip.columbia.edu;user=phone> to <sip:47891@sip.columbia.edu> May 29 15:34:08 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26) May 29 15:34:08 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069> May 29 15:34:08 cocoa openser[892]: loose_route: Error while parsing Request URI May 29 15:34:08 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26) May 29 15:34:09 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069> May 29 15:34:09 cocoa openser[892]: WARNING: do_action:error in expression May 29 15:34:09 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26) May 29 15:34:09 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069> May 29 15:34:09 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26) May 29 15:34:10 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069> May 29 15:34:10 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26) May 29 15:34:10 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069> May 29 15:34:10 cocoa openser[892]: alias_db_lookup: Error while parsing Request-URI While the malformed packet is likely the fault of the Asterisk server (we see no packets from OpenSER to Asterisk with the sip protocol missing from the URI), we are wondering if OpenSER errors out in a bad way when it tries to process this packet, possibly entering into some sort of loop that is causing the load to go through the roof. It is also unclear to us why the Asterisk voicemail server would be responding with a cancel. In our packet capture we see no record of a request initiated by the Asterisk server that it would be sending a CANCEL for. Any insight that could be offered into this problem would be greatly appreciated. Thx, David Coulthart

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Re: [OpenSER-Users] About the opensip server.
by Henning Westerholt 27 Jun '07

27 Jun '07

On Mittwoch, 27. Juni 2007, gaoqiang.qian(a)nokia.com wrote: > When I start the debug model, I have got some error of connecting mysql > database, it said something like > "mysql:openserro.openserro@localhost/openser" not found in the pool. > Could you tell me what could be the reason? > I can not find the achieves about this issues in the mail achieves. > Could you point out me some links? Hello Qiang, have you setup a mysql database and create the right tables with the database setup script? You find more informations about openser in the openser wiki on www.openser.org. If you want to use openser in a production environment, you should install a recent version, e.g. the latest stable release 1.2.1. Cheers, Henning

1 0

[OpenSER-Users] About the opensip server.
by gaoqiang.qian＠nokia.com 27 Jun '07

27 Jun '07

> Hi, > I am working in Nokia and now response for setup a openser sip server > providing sip proxy and registrar for one of our project. After I have > install the openser Sip server by following all the steps from CVS > installation guide. When I start the sip server with openser start > command, I always got: > 'openser' start: /etc/init.d/openser > Openser failed to start > Strating open sip router: > Listen on > Udp: > ... > Aliases: > Tcp: > ... > It seems the starting is failed. So could you tell me what could be > the reason? Or could you please tell me is there someway I can test if > my openser is running now? > Thanks. >

3 3

[Serusers] How SER communicate to other SIP server
by roayu 27 Jun '07

27 Jun '07

Hi, there! I would like to find out is there anyway that the SER server can communicate to other SIP servers? For instance, I have an account at SER and another account at SIPserver B. I would like to call from SER to SIPserver B, and vice versa. Is it workable? Or any additional devices needed in between these 2 SIP servers? If yes, could anyone give me some clues or guide to do so. Thank you. Best Regards, Roa yu

2 1

[OpenSER-Users] XCAP lite: http proxy setting
by vinayakumar.bendi＠wipro.com 27 Jun '07

27 Jun '07

Hi, Am trying to check 'XCAP server simulator for OpenSER presence agent' using an XCAP (client) test-tool from seagull. When I send an XCAP PUT request to XCAP server the request fails during schema validation. Have enabled libxml traces in xcap-lib.php and found the following error .. Jun 26 18:18:26 presence XCAP[26336]: <pre> Jun 26 18:18:26 presence XCAP[26336]: DOMDocument::schemaValidate() Generated Errors! Jun 26 18:18:26 presence XCAP[26336]: Warning 1549: failed to load external entity "http://www.w3.org/2001/xml.xsd" on line 0 Jun 26 18:18:26 presence XCAP[26336]: Warning 3084: Element '{http://www.w3.org/2001/XMLSchema}import': Failed to locate a schema at location 'http://www.w3.org/2001/xml.xsd'. Skipping the import. in file /var/www/xcap-root/schemas/resource-lists.xsd on line 7 Jun 26 18:18:26 presence XCAP[26336]: Error 3004: attribute use (unknown), attribute 'ref': The QName value '{http://www.w3.org/XML/1998/namespace}lang' does not resolve to a(n) attribute declaration. in file /var/www/xcap-root/schemas/resource-lists.xsd on line 73 Jun 26 18:18:26 presence XCAP[26336]: Error: failed to validate resource-lists against resource-lists.xsd It appears that the validation fails when trying to load an HTTP URL (for xml.xsd). I am running this in a network which has a gateway to internet via a proxy. Any clues to get around this problem will be appreciated? I have tried downloading xml.xsd locally and updating references in resource-lists.xsd but haven't had enough luck in getting it work either :-( Jun 26 18:16:53 presence XCAP[26321]: <pre> Jun 26 18:16:53 presence XCAP[26321]: DOMDocument::schemaValidate() Generated Errors! Jun 26 18:16:53 presence XCAP[26321]: Error 201: Namespace prefix xs on schema is not defined in file file:///var/www/xcap-root/schemas/xmlschema/xml.xsd on line 1 Jun 26 18:16:53 presence XCAP[26321]: Error 201: Namespace prefix xs on annotation is not defined in file file:///var/www/xcap-root/schemas/xmlschema/xml.xsd on line 2 ... Kind regards, Vinaya The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[Users] Setting the media IP:port without Nathelper
by Bruno Machado 27 Jun '07

27 Jun '07

Hi all. There is a way to change the media IP and port without the module Nathelper? I don't want to use RTPProxy or Media Proxy. Let's suppose that the received INVITE has the headers: c=IN IP4 192.168.0.56 t=0 0 m=audio 20900 I want to set up the 'c' header to, for example, 192.168.0.100 and port 40000. Is it possible? Thanks. Bruno ____________________________________________________________________________________ Novo Yahoo! Cadê? - Experimente uma nova busca. http://yahoo.com.br/oqueeuganhocomisso

4 4

[Serusers] How SER communicate to other SIP server
by roayu 27 Jun '07

27 Jun '07

Hi, there! I would like to find out is there anyway that the SER server can communicate to other SIP servers? For instance, I have an account at SER and another account at SIPserver B. I would like to call from SER to SIPserver B, and vice versa. Is it workable? Or any additional devices needed in between these 2 SIP servers? If yes, could anyone give me some clues or guide to do so. Thank you. Best Regards, Roa yu

1 0

[Serusers] SER Accounting.
by Robert Zorop 26 Jun '07

26 Jun '07

Hi, all. I know this is a little bit out off Topic. I'm trying to install CDRTool for CDR normalization and billing with SER. I get the things allmost working (freeradius + MySQL + SER) But i can't get the CDRs normalized with price. Can anybody please help me or post a CDRTool global.inc working file? . Thanks a lot in advance any help will be highly apreciated!

2 1

[Serusers] Ser and asterisk as voicemail
by Riccardo Cupardo 26 Jun '07

26 Jun '07

2 1

[OpenSER-Users] SIP softphones which support TLS
by Vikas CG. 26 Jun '07

26 Jun '07

Hi all, Can any one please suggest some free SIP softphones which support TLS connection. Thanks, Vikas C.G. +91-9844502090 Span Infotech (I) Pvt.Ltd, Bangalore-40. SEI CMM - Level 4 & Certified ISO 9001:2000 www.spansystems.com <http://www.spansystems.com>

2 1

[OpenSER-Users] totag in 200 cancelling reply
by Alex Hermann 26 Jun '07

26 Jun '07

Hello, Situation is : UAC -> Asterisk1 -> OpenSER -> Asterisk2 OpenSER is using tm module (t_relay and friends). When the UAC cancels an INVITE, the CANCEL is propagated to Asterisk2 which replies to the INVITE with a 487 and to the CANCEL with 200 OK. Both replies have the same ToTag, as rfc3261 9.2 suggests (SHOULD). The 487 reply is forwarded to Asterisk1 by OpenSER with the ToTag unaltered, but the 200 OK is forwarded as "200 cancelling" with a different ToTag (created by OpenSER). This messes up the transaction matching on Asterisk1 (which uses pedantic=yes in the sip.conf and has to stay that way) Is there a way to force OpenSER to use the ToTag of the incoming reply (the winning branch?) instead of it creating a new one? -- Greetings, Alex Hermann

2 1

[Serusers] Routes to SIP broker
by Andrey Kuprianov 26 Jun '07

26 Jun '07

Hi, Are you guys planning on creating an outgoing route through sipbroker.com? Andrey.

2 3

[Serusers] Rfc 3327 path
by Stefan Ljung (TN/EAB) 26 Jun '07

26 Jun '07

Hi As I understand it rfc3327 is not supported in SER 0.9.6 . When I read the documentation for SER 2.0 I could not find anything written on the rfc3327 subject. But for SER 1.1.X the support for rfc3327 seem to be implemented. But SER 1.1.X has no stable release, right? Am I right ? Please advice as to which version of SER I should turn to have this service in my SIP domain.

2 1

[OpenSER-Users] CDRTOOLS+RADIUS+OPENSER
by ram 26 Jun '07

26 Jun '07

Hi can some have comment success rate of above combination deployement, stability and other issues, planning to deploy in live environment ram

1 0

[OpenSER-Users] Radius auth change attribute
by Richard Timsit 26 Jun '07

26 Jun '07

Hi, this was already asked in this mailinglist in Dec 2006, is a way to rewrite radius auth request packet before sending to radius ? I need to change User-Name attribute of the authentication request. Something new on this ? Thank you. -- Richard Timsit <Richard.Timsit(a)epfl.ch> EPFL

1 0

[OpenSER-Users] openser 1.2.1 core dumps on 302 redirect
by Brandon Price 26 Jun '07

26 Jun '07

With debug=99 I get ... 0(31783) DEBUG:t_check_status: checked status is <302> 0(31783) +15183024234 has a forward 0(31783) DEBUG:uac_redirect:get_redirect: resume branch=0 0(31783) DEBUG:uac_redirect:get_redirect: checking branch=0 (added=0) 0(31783) DEBUG:uac_redirect:get_redirect: branch=0 is a redirect (added=0) 0(31783) parse_headers: flags=7 0(31783) Found param type 232, <branch> = <z9hG4bK68c33491>; state=16 0(31783) end of header reached, state=5 0(31783) parse_headers: Via found, flags=7 0(31783) parse_headers: this is the second via 0(31783) DEBUG: get_hdr_body : content_length=0 0(31783) found end of header 0(31783) DEBUG:uac_redirect:sort_contacts: <sip: 13143212222(a)eastern.testdomain.net> q=10 0(31783) DEBUG:uac_redirect:shmcontact2dset: adding contact <sip: 13143212222(a)eastern.testdomain.net> 0(31783) ******* setting for branch 0 flags 0 0(31783) parse_headers: flags=78 Segmentation fault (core dumped) // Where is this core dump being written? related openser.cfg snippet: if(t_check_status("302")){ xlog("$tU has a forward"); get_redirects("*", "redirected"); xlog("forwarding to $tU"); t_relay("5.6.7.8"); exit; }; Any help is as always, greatly appreciated.

2 1

[OpenSER-Users] Debian Etch with MySQL module
by Gerardo Gonzalez 26 Jun '07

26 Jun '07

Regards, I am running Debian Etch and added the repository for OpenSER to the sources.list file. I installed OpenSER but when I try to install openser-mysql-module, I get the following error: The following packages have unmet dependencies. openser-mysql-module: Depends: libmysqlclient14 but it is not installable. First, I added the following repository: deb http://www.openser.org/debian etch main There's no repository etch. I was looking into http://www.openser.org/debian/dist and I saw the following directories: /sarge /stable I got the next error: Err http://www.openser.org etch/main Packages 404 Not Found Can anybody help me? Thanks -- GJGM

2 1

[OpenSER-Users] Permissions Module
by Kelvin Williams 26 Jun '07

26 Jun '07

Does anyone have Openser 1.2, with PostgreSQL 8.2.4 and the Permissions module working with at least one entry in the address table? If anyone has Openser 1.2 and PostgreSQL 8.2.4 running in combination, could you see if you can duplicate this problem? Any time any entry is in the address table, OpenSER will not start, reporting this error: openser=# select * from address; id | grp | ip_addr | mask | port ----+-----+---------+------+------ (0 rows) Starts fine! openser=# insert into address (grp, ip_addr, mask, port) values (0, '192.168.1.100',32, 5060); INSERT 0 1 openser=# select * from address; id | grp | ip_addr | mask | port ----+-----+---------------+------+------ 5 | 0 | 192.168.1.100 | 32 | 5060 (1 row) openser=# Jun 23 16:55:47 gaia /usr/local/sbin/openser[24163]: ERROR: permissions: address_reload(): Database problem Jun 23 16:55:47 gaia /usr/local/sbin/openser[24163]: permissions:init_addresses(): Reload of address table failed Jun 23 16:55:47 gaia /usr/local/sbin/openser[24163]: Error while initializing allow_address function Jun 23 16:55:47 gaia /usr/local/sbin/openser[24163]: init_mod(): Error while initializing module permissions Thanks in advance, kw

3 3

Re: [Serusers] Please help its urgent
by Jiri Kuthan 26 Jun '07

26 Jun '07

This indeed looks very much like a memory leek. Report it please with information as described in the memory leak section of http://www.iptel.org/ser/problem_reporting Anyhow -- upgrading to the latest version would be probably the safest thing to move ahead, I think that for many including yourself use of historical version is suboptimal use of time. -jiri At 10:43 21/06/2007, Arun Kumar wrote: >error: >-------- > sip ser[17671]: ERROR: sip_msg_cloner: cannot allocate memory > sip ser[17671]: ERROR: new_t: out of mem: > sip ser[17671]: ERROR: t_newtran: new_t failed >sip ser[17671]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) >sip ser[17671]: Warning: sl_send_reply: I won't send a reply for ACK!! >sip ser[17669]: ERROR: sip_msg_cloner: cannot allocate memory >sip ser[17669]: ERROR: new_t: out of mem: >sip ser[17669]: ERROR: t_newtran: new_t failed >sip ser[17669]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) >sip ser[17669]: Warning: sl_send_reply: I won't send a reply for ACK!! >sip ser[17669]: ERROR: sip_msg_cloner: cannot allocate memory >sip ser[17669]: ERROR: new_t: out of mem: >sip ser[17669]: ERROR: t_newtran: new_t failed >sip ser[17669]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) > >this is my latest mem info: ># free > total used free shared buffers cached >Mem: 2074376 2019056 55320 0 75972 1517308 >-/+ buffers/cache:425776 1648600 >Swap: 2031608 0 2031608 > >thanks > >On 6/21/07, Hendrik Scholz <<mailto:hendrik.scholz@freenet-ag.de>hendrik.scholz(a)freenet-ag.de > wrote: >Hi! > >Arun Kumar wrote: >> Hi All, > >quite an appropriate greeting for both serusers at Iptel and OpenSER. > >> please can some one tell me is there any memory leak with ser-0.9.6. It >> runs fine for few days and then starts some memory related problems in >> that case I've to restart my ser. > >from one of your previous mails I got this: ># free >total used free shared buffers cached >Mem: 2074376 1740724 333652 0 72328 1280192 >-/+ buffers/cache: 388204 1686172 >Swap: 2031608 0 2031608 > >If you take a close look at the numbers you'll see that (minus >buffers/cache) you have 1.6GB of memory left. > > >> I've 2GB of physical RAM in my machine >> with Quad Core cpu. but now I've only 70MB free out of 2 GB in max 1 day >> It will start to give me again error. > >Which error? >Please paste a few (reoccuring) messages from your SER logs. > >Cheers, >Hendrik > >-- >Hendrik Scholz >VoIP Systems Engineer > >---------------------------------------------------------------------------- >Telefon: + 49 (0) 431 90 20 552 >Telefax: + 49 (0) 431 90 20 559 >E-Mail: <mailto:hendrik.scholz@freenet.ag>hendrik.scholz(a)freenet.ag >Website: <http://www.freenet.de>www.freenet.de; <http://www.mobilcom.de>www.mobilcom.de >---------------------------------------------------------------------------- >freenet Cityline GmbH >Ein Unternehmen der freenet AG >Hamburger Chaussee 2-4 >24114 Kiel >---------------------------------------------------------------------------- >Geschäftsführer: Eckhard Spoerr, Axel Krieger >Amtsgericht Kiel, HRB 6202 > > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[OpenSER-Users] The Via Header of CANCEL Message
by 凌芳瑜 26 Jun '07

26 Jun '07

According to RFC 3665 section 3.8 and RFC 3261 section 9, there is only one via header in CANCEL message (message F11, F13 at page 72 of RFC 3665). But when I ran the similar scenario(through one proxy), I got two via header in CANCEL message such as following: CANCEL sip:UA12@node11.under.test.com;transport=udp SIP/2.0 Record-Route: <sip:192.0.50.50;lr=on;ftag=11380> Via: SIP/2.0/UDP ss.under.test.com;branch=z9hG4bKd068.25909417.0 Via: SIP/2.0/UDP node.under.test.com:5060;received=192.0.10.10;branch=z9hG4bKPUA4295483 Max-Forwards: 69 From: UA11 <sip:UA11@under.test.com>;tag=11380 To: <sip:UA12@under.test.com> Call-ID: 11380(a)tom.example.com CSeq: 4 CANCEL Content-Length: 0 P-hint: usrloc applied How can I remove the 2nd via header? Thank you Fangyu Ling

1 0

[OpenSER-Users] problem with asterisk and ser
by Frank Cases 25 Jun '07

25 Jun '07

SER 1 ip (1.2.3.4) Asterisk 2 ip's same nic as aliases.. same subnet as ser.. (1.2.3.5,1.2.3.6) If I call asterisk from SER and asterisk has 2 ips When I call asterisk . SER (1.2.3.4 ) INVITE to Asterisk (1.2.3.5) Asterisk answer (100 Trying ) from another ip Asterisk back (100 Trying ) to SER from (1.2.3.6) to (1.2.3.4) Anyway to fix ?

1 0

[OpenSER-Users] pronunciation?
by A.M. 25 Jun '07

25 Jun '07

I wasn't able to find this question answered in the FAQ: What is the preferred English-language pronunciation of "OpenSER"? -M

2 1

[Serusers] SER RTCP
by Arun Kumar 25 Jun '07

25 Jun '07

Hi how do I enable RTCP in ser ? thanks arun

1 0

[OpenSER-Users] Usrloc db_mode = 3 (DB_ONLY)
by Graham Markall 25 Jun '07

25 Jun '07

Hi, I am just getting started with openser (using openser-1.2.1) and have run across a problem I cannot seem to solve. When using db_mode=2 for the usrloc module, everything works fine. However, when db_mode is set to 3, calls are not able to be made - it seems that lookup("location") always fails. I see this in the log: Jun 25 16:35:17 cobalt1 /usr/local/sbin/openser[10084]: check_response(): Authorization is OK Jun 25 16:35:17 cobalt1 /usr/local/sbin/openser[10084]: check_username(): Digest username and URI username match Jun 25 16:35:17 cobalt1 /usr/local/sbin/openser[10084]: DEBUG:usrloc:db_load_urecord: aor not found in DB Jun 25 16:35:17 cobalt1 /usr/local/sbin/openser[10084]: lookup(): '1002' Not found in usrloc Having checked the mysql query log, the query that was run is: 130 Query select contact,expires,q,callid,cseq,flags,cflags,user_agent,received,path,sock et,methods, last_modified from location where username='1001'order by q If I run this query myself at the mysql command prompt, I get exactly one row returned: +-----------------------------------------------------+----------------- ----+-------+-------------------------------+-------+-------+--------+-- ----------------------------+----------+------+---------------------+--- ------+---------------------+ | contact | expires | q | callid | cseq | flags | cflags | user_agent | received | path | socket | methods | last_modified | +-----------------------------------------------------+----------------- ----+-------+-------------------------------+-------+-------+--------+-- ----------------------------+----------+------+---------------------+--- ------+---------------------+ | sip:1001@192.168.0.25:5062;transport=udp;user=phone | 2007-06-25 17:34:59 | -1.00 | bd348eeee82e4349(a)192.168.0.25 | 20092 | 0 | 0 | Grandstream GXP2000 1.1.2.27 | NULL | NULL | udp:172.16.1.1:5060 | 8063 | 2007-06-25 16:34:59 | +-----------------------------------------------------+----------------- ----+-------+-------------------------------+-------+-------+--------+-- ----------------------------+----------+------+---------------------+--- ------+---------------------+ 1 row in set (0.00 sec) Having followed through the code, in modules/usrloc/udomain.c (about line 550), things seem to be going wrong for me in this bit of code: if (ul_dbf.query(_c, keys, 0, vals, columns, (use_domain)?2:1, 13, order, &res) < 0) { LOG(L_ERR, "ERROR:usrloc:db_load_urecord: db_query failed\n"); return 0; } if (RES_ROW_N(res) == 0) { DBG("DEBUG:usrloc:db_load_urecord: aor not found in DB\n"); ul_dbf.free_result(_c, res); return 0; } I tried to find out why this is happening by attaching with gdb, but when I try to view res or res->n, I get a message that variable res is currently unavailable - I'm not sure if this is something to do with the way I'm using gdb though. When I step through, RES_ROW_N(res) == 0 must be true as it enters the code within the {} instead of skipping over it. Could anyone point me in the right direction as to why this doesn't work please? Many thanks, Graham Markall

1 0

[OpenSER-Users] XCAP Client on OpenSER
by Toni Barata 25 Jun '07

25 Jun '07

Hi all, Openser works pretty well as Presence Server, but on this development stage (version 1.2.x) the communication with XCAP Server is done via mysql (xcap_xml table). The integration with XCAP-Lite works this way and works Ok. However, SER has a library with XCAP-Client interface and we have started an integration of that library on the presence module of Openser. With this standard approach becomes possible the integration of the Openser Presence Server with, not only the XCAP-Lite, but also with others XCAP Servers, e.g., OpenXDM (https://openxdm.dev.java.net/). To have this integration accomplish get into the following steps: Libxcap Instalation: tar xvzf ser-2.0.0+cvs20070315_src.tar.gz cd ser-2.0.0+cvs20070315/lib/cds/ make install cd ../xcap/ make install export LD_LIBRARY_PATH=/usr/local/lib ( execute before starting openser) Run the attached patch: cd openser-1.2.x (openser's root) cp ../xcap_interface.diff . (Copy the patch to openser's root) patch -p1 <xcap_interface.diff (apply the patch ) Compile Openser, or if it is already compilied, just compile and install the presence module: make modules=modules/presence modules cp modules/presence/presence.so /usr/local/lib/openser/modules/ Add the XCAP configuration parameters on the Openser configuration file (openser.cfg): "... modparam("presence", "xcap_server", "http://x.x.x.x") modparam("presence", "xcap_root", "xcap-root") modparam("presence", "xcap_app_usage", "org.openmobilealliance.pres-rules") modparam("presence", "xcap_document_name", "presrules") modparam("presence", "xcap_auth_username", "openser") modparam("presence", "xcap_auth_password", "openserrw") ..." The integration was done with the following versions: openser-1.2.x-svn-20070517-221101.tgz (http://www.openser.org/downloads/snapshots/openser-1.2.x/) ser-2.0.0+cvs20070315_src.tar.gz (http://ftp.iptel.org/pub/ser/daily-snapshots/testing/) With this modification, when the Presence Servers need to consult a client's pres-rules it can do a XCAP Get instead of a MySql Query. This approach is closer to the OMA Presence SIMPLE + OMA XDM standards. Try it, test it and give us some feedback about it. If you have difficulties on the installation or configuration please contact us. Best Regards, Toni

1 0

[OpenSER-Users] OpenSER Attributes sending to RADIUS
by liran tal 25 Jun '07

25 Jun '07

Hey everyone, After quite a bit of google digging I've came across Dimitry's mail that cleared up some things that weren't so obvious in the documentation of the ACC module. He gave the example of using the following code: modparam("acc", "radius_extra", "Sip-UA=$hdr(User-Agent);Sip-Via=$hdr(Via[*]); Sip-SDP=$rb") Which in his case makes use of the Sip-UA, Sip-Via and Sip-SDP attributes but in the dictionary.sip file provided with openser/radiusclient-ng these attribute aren't present, there are other sets of attributes but not these... And so, I tried adding myself some custom attributes to both dictionary files, the one on the freeradius server and the other on openser's radiusclient-ng server and tried for example this: ATTRIBUTE Sip-Src-IP 900 string ATTRIBUTE Sip-Src-Port 901 string ATTRIBUTE Sip-Hdr-Contact 902 string ATTRIBUTE Sip-Hdr-UA 903 string ATTRIBUTE Sip-Hdr-From 904 string ATTRIBUTE Sip-Hdr-Media 905 string And then I put in openser's cfg: modparam("acc", "radius_extra", "Sip-Src-IP=$si") But it doesn't work, the accounting record on the freeradius server doesn't show the Sip-Src-IP attribute. If I change the modparam to use Sip-User-Realm which is defined in the original dictionary.sip on both servers then THAT IS added to the accounting record... my question is why? And the second question, is there a more extensive dictionary.sip file? How is it possible to add my own custom attributes as well? Thanks, Liri.

1 0

RE: [OpenSER-Users] RE: [Users] out of memory - please help
by Papadopoulos Georgios 25 Jun '07

25 Jun '07

> -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Monday, June 25, 2007 9:28 AM > To: Papadopoulos Georgios > Cc: Bogdan-Andrei Iancu; users(a)openser.org > Subject: Re: [OpenSER-Users] RE: [Users] out of memory - please help > > > > Papadopoulos Georgios wrote: > > Why did this only appear on the first receiver child? I > would expect > > that all children receive an equal amount of traffic so they should > > all run out of memory more or less at the same time. Except > if when a > > message is received, Openser finds the first available > child and does > > this by trying child 1,2,3... Is this the case? > > The distribution between the UDP listener threads is not done > by openser but by the OS. openser will open port 5060 and > then fork the process "children=..." times. Then, all > processes will read from the same port. > Then how do you explain that only the first child ran out of memory? Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[OpenSER-Users] openser routing question
by Radu Spineanu 25 Jun '07

25 Jun '07

Hi, I've been trying to setup a openser + asterisk server following the tutorial here: http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+Op… and with that openser configuration example. The only difference in my setup is that I am using openser 1.1. For some reason it doesn't seem to be working, the call is not established. This is the debugging output: /usr/sbin/openser[26006]: SIP Request: /usr/sbin/openser[26006]: method: <INVITE> /usr/sbin/openser[26006]: uri: <sip:101@217.73.174.13> /usr/sbin/openser[26006]: version: <SIP/2.0> /usr/sbin/openser[26006]: parse_headers: flags=2 /usr/sbin/openser[26006]: Found param type 235, <rport> = <n/a>; state=6 /usr/sbin/openser[26006]: Found param type 232, <branch> = <z9hG4bKapseucta>; state=16 /usr/sbin/openser[26006]: end of header reached, state=5 /usr/sbin/openser[26006]: parse_headers: Via found, flags=2 /usr/sbin/openser[26006]: parse_headers: this is the first via /usr/sbin/openser[26006]: After parse_msg... /usr/sbin/openser[26006]: preparing to run routing scripts... /usr/sbin/openser[26006]: parse_headers: flags=100 /usr/sbin/openser[26006]: DEBUG:maxfwd:is_maxfwd_present: value = 70 /usr/sbin/openser[26006]: parse_headers: flags=10 /usr/sbin/openser[26006]: DEBUG:parse_to:end of header reached, state=10 /usr/sbin/openser[26006]: DBUG:parse_to: display={}, ruri={sip:101@217.73.174.13} /usr/sbin/openser[26006]: DEBUG: get_hdr_field: <To> [25]; uri=[sip:101@217.73.174.13] /usr/sbin/openser[26006]: DEBUG: to body [<sip:101@217.73.174.13>^M ] /usr/sbin/openser[26006]: DEBUG: add_param: tag=hsiim /usr/sbin/openser[26006]: DEBUG:parse_to:end of header reached, state=29 /usr/sbin/openser[26006]: DBUG:parse_to: display={"Radu Spineanu"}, ruri={sip:100@217.73.174.13} /usr/sbin/openser[26006]: parse_headers: flags=200 /usr/sbin/openser[26006]: get_hdr_field: cseq <CSeq>: <797> <INVITE> /usr/sbin/openser[26006]: DEBUG: get_hdr_body : content_length=303 /usr/sbin/openser[26006]: found end of header /usr/sbin/openser[26006]: find_first_route: No Route headers found /usr/sbin/openser[26006]: loose_route: There is no Route HF /usr/sbin/openser[26006]: grep_sock_info - checking if host==us: 13==13 && [217.73.174.13] == [217.73.174.13] /usr/sbin/openser[26006]: grep_sock_info - checking if port 5060 matches port 5060 /usr/sbin/openser[26006]: grep_sock_info - checking if host==us: 13==13 && [217.73.174.13] == [217.73.174.13] /usr/sbin/openser[26006]: grep_sock_info - checking if port 5060 matches port 5060 /usr/sbin/openser[26006]: DEBUG: has_totag: no totag /usr/sbin/openser[26006]: DEBUG: has_totag: no totag /usr/sbin/openser[26006]: is_user_in(): User is not in group 'voicemail' /usr/sbin/openser[26006]: rewrite_uri: Rewriting Request-URI with 'sip:101@82.79.243.82:5068' /usr/sbin/openser[26006]: parse_headers: flags=ffffffffffffffff /usr/sbin/openser[26006]: DEBUG: t_newtran: msg id=2 , global msg id=0 , T on entrance=0xffffffffffffffff /usr/sbin/openser[26006]: parse_headers: flags=ffffffffffffffff /usr/sbin/openser[26006]: parse_headers: flags=78 /usr/sbin/openser[26006]: t_lookup_request: start searching: hash=18702, isACK=0 /usr/sbin/openser[26006]: DEBUG: RFC3261 transaction matching failed /usr/sbin/openser[26006]: DEBUG: t_lookup_request: no transaction found /usr/sbin/openser[26006]: DEBUG:tm:t_relay: new INVITE /usr/sbin/openser[26006]: parse_headers: flags=ffffffffffffffff /usr/sbin/openser[26006]: check_via_address(217.73.174.16, 82.79.243.82, 0) /usr/sbin/openser[26006]: WARNING:vqm_resize: resize(0) called /usr/sbin/openser[26006]: DEBUG:tm:_reply_light: reply sent out. buf=0x5e6458: SIP/2.0 1..., shmem=0x2b05574b9fb8: SIP/2.0 1 /usr/sbin/openser[26006]: DEBUG:tm:_reply_light: finished /usr/sbin/openser[26006]: DEBUG: mk_proxy: doing DNS lookup... /usr/sbin/openser[26006]: check_via_address(217.73.174.16, 82.79.243.82, 0) /usr/sbin/openser[26006]: DEBUG: add_to_tail_of_timer[4]: 0x2b05574b7ba0 /usr/sbin/openser[26006]: DEBUG: add_to_tail_of_timer[0]: 0x2b05574b7bc0 /usr/sbin/openser[26006]: SER: new transaction fwd'ed /usr/sbin/openser[26006]: DEBUG:tm:UNREF_UNSAFE: after is 0 /usr/sbin/openser[26006]: DEBUG:destroy_avp_list: destroying list (nil) /usr/sbin/openser[26006]: receive_msg: cleaning up /usr/sbin/openser[26012]: DEBUG: timer routine:4,tl=0x2b05574b7ba0 next=(nil) /usr/sbin/openser[26012]: DEBUG: retransmission_handler : request resending (t=0x2b05574b79c0, INVITE si ... ) /usr/sbin/openser[26012]: DEBUG: add_to_tail_of_timer[5]: 0x2b05574b7ba0 /usr/sbin/openser[26012]: DEBUG: retransmission_handler : done amy /usr/sbin/openser[26012]: DEBUG: timer routine:5,tl=0x2b05574b7ba0 next=(nil) amy /usr/sbin/openser[26012]: DEBUG: retransmission_handler : request resending (t=0x2b05574b79c0, INVITE si ... ) amy /usr/sbin/openser[26012]: DEBUG: add_to_tail_of_timer[6]: 0x2b05574b7ba0 amy /usr/sbin/openser[26012]: DEBUG: retransmission_handler : done [..] Jun 24 15:38:47 amy /usr/sbin/openser[26012]: DEBUG: timer routine:0,tl=0x2b05574b7bc0 next=(nil) /usr/sbin/openser[26012]: DEBUG: final_response_handler:stop retr. and send CANCEL (0x2b05574b79c0) /usr/sbin/openser[26012]: DEBUG:tm:t_should_relay_response: T_code=100, new_code=408 /usr/sbin/openser[26012]: DEBUG:tm:t_pick_branch: picked branch 0, code 408 /usr/sbin/openser[26012]: DEBUG:tm:relay_reply: branch=0, save=0, relay=0 /usr/sbin/openser[26012]: parse_headers: flags=ffffffffffffffff /usr/sbin/openser[26012]: check_via_address(217.73.174.16, 82.79.243.82, 0) /usr/sbin/openser[26012]: DEBUG: add_to_tail_of_timer[4]: 0x2b05574b7ae8 /usr/sbin/openser[26012]: DEBUG: add_to_tail_of_timer[0]: 0x2b05574b7b08 /usr/sbin/openser[26012]: DEBUG:tm:relay_reply: sent buf=0x5e4fb0: SIP/2.0 4..., shmem=0x2b05574b9fb8: SIP/2.0 4 /usr/sbin/openser[26012]: DEBUG: final_response_handler : done /usr/sbin/openser[26012]: DEBUG: timer routine:7,tl=0x2b05574b7ba0 next=(nil) /usr/sbin/openser[26006]: SIP Request: /usr/sbin/openser[26006]: method: <ACK> /usr/sbin/openser[26006]: uri: <sip:101@217.73.174.13> /usr/sbin/openser[26006]: version: <SIP/2.0> /usr/sbin/openser[26006]: parse_headers: flags=2 /usr/sbin/openser[26006]: Found param type 235, <rport> = <n/a>; state=6 /usr/sbin/openser[26006]: Found param type 232, <branch> = <z9hG4bKapseucta>; state=16 /usr/sbin/openser[26006]: end of header reached, state=5 /usr/sbin/openser[26006]: parse_headers: Via found, flags=2 /usr/sbin/openser[26006]: parse_headers: this is the first via /usr/sbin/openser[26006]: After parse_msg... /usr/sbin/openser[26006]: preparing to run routing scripts... /usr/sbin/openser[26006]: DEBUG : sl_filter_ACK: to late to be a local ACK! /usr/sbin/openser[26006]: parse_headers: flags=100 /usr/sbin/openser[26006]: DEBUG:maxfwd:is_maxfwd_present: value = 70 /usr/sbin/openser[26006]: parse_headers: flags=10 /usr/sbin/openser[26006]: DEBUG: add_param: tag=90f6b651c515331ae78169ff327ec50d-6648 /usr/sbin/openser[26006]: DEBUG:parse_to:end of header reached, state=29 /usr/sbin/openser[26006]: DBUG:parse_to: display={}, ruri={sip:101@217.73.174.13} /usr/sbin/openser[26006]: DEBUG: get_hdr_field: <To> [67]; uri=[sip:101@217.73.174.13] /usr/sbin/openser[26006]: DEBUG: to body [<sip:101@217.73.174.13>] /usr/sbin/openser[26006]: DEBUG: add_param: tag=hsiim /usr/sbin/openser[26006]: DEBUG:parse_to:end of header reached, state=29 /usr/sbin/openser[26006]: DBUG:parse_to: display={"Radu Spineanu"}, ruri={sip:100@217.73.174.13} /usr/sbin/openser[26006]: parse_headers: flags=200 /usr/sbin/openser[26006]: get_hdr_field: cseq <CSeq>: <797> <ACK> /usr/sbin/openser[26006]: DEBUG: get_hdr_body : content_length=0 /usr/sbin/openser[26006]: found end of header /usr/sbin/openser[26006]: find_first_route: No Route headers found /usr/sbin/openser[26006]: loose_route: There is no Route HF /usr/sbin/openser[26006]: grep_sock_info - checking if host==us: 13==13 && [217.73.174.13] == [217.73.174.13] /usr/sbin/openser[26006]: grep_sock_info - checking if port 5060 matches port 5060 /usr/sbin/openser[26006]: grep_sock_info - checking if host==us: 13==13 && [217.73.174.13] == [217.73.174.13] /usr/sbin/openser[26006]: grep_sock_info - checking if port 5060 matches port 5060 /usr/sbin/openser[26006]: rewrite_uri: Rewriting Request-URI with 'sip:101@82.79.243.82:5068' /usr/sbin/openser[26006]: parse_headers: flags=ffffffffffffffff /usr/sbin/openser[26006]: DEBUG: t_newtran: msg id=3 , global msg id=2 , T on entrance=0xffffffffffffffff /usr/sbin/openser[26006]: parse_headers: flags=ffffffffffffffff /usr/sbin/openser[26006]: parse_headers: flags=78 /usr/sbin/openser[26006]: t_lookup_request: start searching: hash=18702, isACK=1 /usr/sbin/openser[26006]: DEBUG: RFC3261 transaction matched, tid=apseucta /usr/sbin/openser[26006]: DEBUG:tm:REF_UNSAFE: after is 1 /usr/sbin/openser[26006]: DEBUG: t_lookup_request: transaction found (T=0x2b05574b79c0) /usr/sbin/openser[26006]: DEBUG: cleanup_uac_timers: RETR/FR timers reset /usr/sbin/openser[26006]: DEBUG: add_to_tail_of_timer[2]: 0x2b05574b7a40 /usr/sbin/openser[26006]: DEBUG:tm:UNREF_UNSAFE: after is 0 /usr/sbin/openser[26006]: DEBUG:destroy_avp_list: destroying list (nil) /usr/sbin/openser[26006]: receive_msg: cleaning up /usr/sbin/openser[26012]: DEBUG: timer routine:0,tl=0x2b05574b7b08 next=(nil) /usr/sbin/openser[26012]: DEBUG: timer routine:4,tl=0x2b05574b7ae8 next=(nil) /usr/sbin/openser[26012]: DEBUG: timer routine:2,tl=0x2b05574b7a40 next=(nil) /usr/sbin/openser[26012]: DEBUG: wait_handler : removing 0x2b05574b79c0 from table /usr/sbin/openser[26012]: DEBUG: delete transaction 0x2b05574b79c0 /usr/sbin/openser[26012]: DEBUG: wait_handler : done And this an ngrep output: U 217.73.174.16:5063 -> 217.73.174.13:5060 INVITE sip:101@217.73.174.13 SIP/2.0..Via: SIP/2.0/UDP 82.79.243.82:5063;rport;branch=z9hG4bKlejwtozv..Max-Forwards: 70..To: <sip:101@217 .73.174.13>..From: "Radu Spineanu" <sip:100@217.73.174.13>;tag=lewnx..Call-ID: pfysegloyvynryc@82.79.243.82..CSeq: 305 INVITE..Contact: < sip:100@82.79.243.82:5063>..Content-Type: application/sdp..Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO..Suppor ted: replaces,norefersub,100rel..User-Agent: Twinkle/1.0.1..Content-Length: 303....v=0..o=100 246106612 367660894 IN IP4 82.79.243.82..s= -..c=IN IP4 82.79.243.82..t=0 0..m=audio 8012 RTP/AVP 98 97 8 0 3 101..a=rtpmap:98 speex/16000..a=rtpmap:97 speex/8000..a=rtpmap:8 PCMA/8 000..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:20.. # U 217.73.174.13:5060 -> 217.73.174.16:5063 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 82.79.243.82:5063;rport=5063;branch=z9hG4bKlejwtozv;received=217.73. 174.16..To: <sip:101@217.73.174.13>..From: "Radu Spineanu" <sip:100@217.73.174.13>;tag=lewnx..Call-ID: pfysegloyvynryc@82.79.243.82..CSeq : 305 INVITE..Server: OpenSer (1.1.0-notls (x86_64/linux))..Content-Length: 0..Warning: 392 217.73.174.13:5060 "Noisy feedback tells: pi d=26007 req_src_ip=217.73.174.16 req_src_port=5063 in_uri=sip:101@217.73.174.13 out_uri=sip:101@82.79.243.82:5068 via_cnt==1".... # U 217.73.174.13:5060 -> 82.79.243.82:5068 INVITE sip:101@82.79.243.82:5068 SIP/2.0..Record-Route: <sip:217.73.174.13;lr=on;ftag=lewnx>..Via: SIP/2.0/UDP 217.73.174.13;branch=z9hG4 bK4a11.a2772f67.0..Via: SIP/2.0/UDP 82.79.243.82:5063;received=217.73.174.16;rport=5063;branch=z9hG4bKlejwtozv..Max-Forwards: 69..To: <si p:101@217.73.174.13>..From: "Radu Spineanu" <sip:100@217.73.174.13>;tag=lewnx..Call-ID: pfysegloyvynryc@82.79.243.82..CSeq: 305 INVITE..C ontact: <sip:100@82.79.243.82:5063>..Content-Type: application/sdp..Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INF O..Supported: replaces,norefersub,100rel..User-Agent: Twinkle/1.0.1..Content-Length: 303..P-hint: usrloc applied....v=0..o=100 246106612 367660894 IN IP4 82.79.243.82..s=-..c=IN IP4 82.79.243.82..t=0 0..m=audio 8012 RTP/AVP 98 97 8 0 3 101..a=rtpmap:98 speex/16000..a=rtpmap :97 speex/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=pt ime:20.. # U 217.73.174.13:5060 -> 82.79.243.82:5068 INVITE sip:101@82.79.243.82:5068 SIP/2.0..Record-Route: <sip:217.73.174.13;lr=on;ftag=lewnx>..Via: SIP/2.0/UDP 217.73.174.13;branch=z9hG4 bK4a11.a2772f67.0..Via: SIP/2.0/UDP 82.79.243.82:5063;received=217.73.174.16;rport=5063;branch=z9hG4bKlejwtozv..Max-Forwards: 69..To: <si p:101@217.73.174.13>..From: "Radu Spineanu" <sip:100@217.73.174.13>;tag=lewnx..Call-ID: pfysegloyvynryc@82.79.243.82..CSeq: 305 INVITE..C ontact: <sip:100@82.79.243.82:5063>..Content-Type: application/sdp..Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INF O..Supported: replaces,norefersub,100rel..User-Agent: Twinkle/1.0.1..Content-Length: 303..P-hint: usrloc applied....v=0..o=100 246106612 367660894 IN IP4 82.79.243.82..s=-..c=IN IP4 82.79.243.82..t=0 0..m=audio 8012 RTP/AVP 98 97 8 0 3 101..a=rtpmap:98 speex/16000..a=rtpmap :97 speex/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=pt ime:20.. # Initially I did the setup folowing http://www.openser.org/dokuwiki/doku.php/asterisk:realtime-integration which worked but didn't have voicemail redirection support. Thanks and sorry for the long mail, Radu Spineanu

1 1

[OpenSER-Users] Insert a Header into ACK message
by 凌芳瑜 25 Jun '07

25 Jun '07

Dear All: I've tried many methods to add Max-Forwarsd header into ACK message (message 8), all methods I tried were fail. According to RFC 3261 8.1.1.6, Max-Forwards must exist in this ACK message. Can anyone tell me how to add a header into ACK message? Thank you very much. UA11 : NUT(Proxy 1) : UA12 | : | : | |-------:------->| : | 1. INVITE |<------:--------| : | 2. 407 |-------:------->| : | 3. ACK | : | : | |-------:------->| : | 4. INVITE | : |-------:------->| 5. INVITE |<------:--------| : | 6. 100 | : | : | | : |<------:--------| 7. 486 (Busy Here) | : | : | | : |-------:------->| 8. ACK | : | : | |<------:--------| : | 9. 486 | : | : | |-------:------->| : |10. ACK | : | : | Best Regards, Fangyu Ling

4 9

[OpenSER-Users] LCR and replies GW
by Frank Cases 25 Jun '07

25 Jun '07

Here it is.. We are trying to keep the prefix, in a header or a avp or something.. So we want to use the prefix to log a certain gw used.. Usually calls come from A to SER to other Term boxes. The problem is once I load_gw() then I cannot use the values. EX: on invite I do some checking , then load_gw.. If gw is box X.. then we add a prefix 2312 lets say. Upon reply I cannot get that value back even in $ruri Although I can set anything in the invite before leaving . Also tried add_rr_param but no go.. Any ideas ? Basically on : onreply_route[1] { if(is_method("INVITE") && ($rs==183) ) { # DO STUFF I need to access the gateway it used from here.. } }

2 1

[Serusers] SMS support
by Andrey Kuprianov 25 Jun '07

25 Jun '07

Hi, Has anyone ever tried using an SMS module and actually sent a message? Andrey.

2 2

[Serusers] XCAP authorization - how to set the username in the xcap root without domain part
by tzieleniewski 25 Jun '07

25 Jun '07

Hi, How can configure pa module in such a way that it will look for the particular user presence document in the "...pres-rules/users/user/.." instead of "../pres-rules/users/user@domain/..". Cheers tomasz

2 3

Re: [OpenSER-Users] how to calculate PDD and ASR
by raviprakash sunkara 25 Jun '07

25 Jun '07

On 6/25/07, ram <talk2ram(a)gmail.com> wrote: > > Google for more information to learn > > any way for your disctionary > > Post Dialing Delay PDD > If the reqeust is Delay.. then can u do , redialing by using the avp modules. This is take place on failover case... Automatic Speech Recognition *ASR* > This ASR , openSER can't do ASR , u need to forward this to Asterisk or PBX servers... On 6/25/07, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote: > > > > > > > > On 6/25/07, ram <talk2ram(a)gmail.com > wrote: > > > > > > Hi > > > > > > iam using openser with mysql accounting > > > > > > can some one assist me how to calculate ASR and PDD > > > > > > > can abbreviate the ASR and PDD . > > > > > > for the calls > > > > > > recomend some documentation to read would be great > > > > > > ram > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > -- > > Thanks &Regards > > Ravi Prakash Sunkara > > > -- Thanks &Regards Ravi Prakash Sunkara

2 1

[Serusers] Help cant connect using natconfig
by Kirk McCalla 25 Jun '07

25 Jun '07

Hello all I have set up ser and I am able to use the sample config to get ser running within my NAT. Now I am trying to get external users to connect to my ser which is behind a nat device. Whenever I use the sample config I can make calls inside my netkwork. whenever I use the nat config that I have below. I cannot even get my local devices to speak to each other. Can someone help here. Whenever I start ser with this config this is what ther system does. ser dead but subsys locked <serusers(a)lists.iptel.org> This is the config that I am using # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # !! Nathelper loadmodule "/usr/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- request routing logic ------------------- # main routing logic 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"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

2 1

[Serusers] Odd error message....
by SIP 25 Jun '07

25 Jun '07

We've never seen this one come through before, so I'm thinking it's in some way related to the cortelco US that's being used. Jun 24 10:15:26 proxy ser[7081]: ACC: call missed: method=INVITE, i-uri=sip:1101XXXXXXX@CLIENT.IP.ADDRESS:60001, o-uri=sip:1101XXXXXXX@CLIENT.IP.ADDRESS:60001, call_id=53ca8e9573a1aa535700119a21ce693c(a)CLIENT.IP.ADDRESS, from=<sip:1315XXXXXXX@our.proxy.com>;tag=as67cfdba3, code=410 Gone Jun 24 10:15:26 proxy ser[7073]: ERROR: parse_uri: uri too short: <2.0> (3) Jun 24 10:15:26 proxy ser[7073]: is_user_in(): Error while parsing URI Jun 24 10:15:26 proxy ser[7073]: ERROR: parse_uri: uri too short: <2.0> (3) Jun 24 10:15:26 proxy ser[7073]: ERROR: parse_sip_msg_uri: bad uri <2.0> Jun 24 10:15:26 proxy ser[7073]: loose_route: Error while parsing Request URI Jun 24 10:15:26 proxy ser[7073]: ERROR: parse_uri: uri too short: <2.0> (3) Jun 24 10:15:26 proxy ser[7073]: ERROR: parse_sip_msg_uri: bad uri <2.0> Jun 24 10:15:26 proxy ser[7073]: WARNING: do_action:error in expression Jun 24 10:15:26 proxy ser[7073]: ERROR: parse_uri: uri too short: <2.0> (3) Jun 24 10:15:26 proxy ser[7073]: ERROR: parse_sip_msg_uri: bad uri <2.0> This is on a response to the reinvite BACK to the local user (1101XXXXXXX) from the PSTN (1315XXXXXXX). The actual block looks like: U CLIENT.IP.ADDRESS:60001 -> OUR.PROXY.IP.ADDRESS:5060 SIP/2.0 410 Gone. Via: SIP/2.0/UDP OUR.PROXY.IP.ADDRESS:5060;branch=z9hG4bKffae.1a137353.0. Via: SIP/2.0/UDP OUR.PSTN.GATEWAY.ADDRESS:5090;branch=z9hG4bK338d81c7;rport=5090. From: <sip:1315XXXXXXX@our.proxy.com>;tag=as67cfdba3. To: "David Jarrett" <sip:1101XXXXXXX@our.proxy.com>;tag=53c99d61. Contact: <sip:1101XXXXXXX@CLIENT.IP.ADDRESS:60001>. Call-ID: 53ca8e9573a1aa535700119a21ce693c(a)CLIENT.IP.ADDRESS. CSeq: 102 INVITE. Content-Length: 0. I'm really not sure which part of the URI is too short (or, to be honest, why SER spits out that message). Clearly '2.0' is not a valid URI... but I'm not entirely sure where it's getting that as the URI. Ideas? We're running 0.9.6. N.

2 1

[OpenSER-Users] RE: [Users] out of memory - please help
by Papadopoulos Georgios 25 Jun '07

25 Jun '07

Hi Bogdan, The server has been running for about 18 hours without any "out of memory" messages. Usually these appeared after a few hours. I think it is fixed, but I will keep you posted if it reappears. Thanks a lot! I guess the same fix needs to go in subst_uri() and maybe also add a check if(result->s!=0) ? Why did this only appear on the first receiver child? I would expect that all children receive an equal amount of traffic so they should all run out of memory more or less at the same time. Except if when a message is received, Openser finds the first available child and does this by trying child 1,2,3... Is this the case? Last (related?) question: There is a thread that the first receiver child starts, what is its purpose? Thank you George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Thursday, June 21, 2007 4:32 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] out of memory - please help > > Hi George, > > I made a fix related to subst_user() - could you apply it > and give it a try? > > Thanks and regards, > Bogdan > > Papadopoulos Georgios wrote: > > Hi Bogdan, > > > > I am using all of the functions that you mentioned. I am also using > > subst(). Please, let me know if you need more info. > > > > best regards > > > > George > > > > > > > >> -----Original Message----- > >> From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >> Sent: Thursday, June 21, 2007 1:46 PM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [Users] out of memory - please help > >> > >> Hi George, > >> > >> looking over the log, it might be something wrong with > >> subst_str() function - I see a lot of memory allocated by this > >> function (3267 mem chunks). > >> > >> now, I have to trace the function using the subst_str() function > >> :)..So, are you using from script one of the following functions: > >> subst_uri() > >> subst_user() > >> avp_subst() > >> > >> regards, > >> bogdan > >> > >> Papadopoulos Georgios wrote: > >> > >>> Hello all, > >>> > >>> We have an ongoing issue with Openser running out of > memory after a > >>> few hours. It is very consistent which makes me believe it > >>> > >> is a memory > >> > >>> leak. Can someone look at the log file? I have done a mem dump on > >>> process 8573 which is the one that ran out of memory. Here > >>> > >> is the log: > >> > >>> _http://www.real.gr/files/openser.20070620.log.gz_ > >>> It always happens on the first receiver child. Our current > >>> > >> solution is > >> > >>> to restart Openser every few hours. I would like to solve > >>> > >> this at its > >> > >>> root, so please let me know if you need more info. > >>> My Openser is 1.2.x from svn, revision 2333. > >>> > >>> thank you for any help > >>> > >>> George > >>> > >>> > >>> > >>> > >>> Disclaimer > >>> > >>> The information in this e-mail and any attachments is > >>> > >> confidential. It > >> > >>> is intended solely for the attention and use of the named > >>> addressee(s). If you are not the intended recipient, or person > >>> responsible for delivering this information to the intended > >>> > >> recipient, > >> > >>> please notify the sender immediately. Unless you are the intended > >>> recipient or his/her representative you are not > authorized to, and > >>> must not, read, copy, distribute, use or retain this > message or any > >>> part of it. E-mail transmission cannot be guaranteed to be > >>> > >> secure or > >> > >>> error-free as information could be intercepted, corrupted, lost, > >>> destroyed, arrive late or incomplete, or contain viruses. > >>> > >>> > >>> > >> > --------------------------------------------------------------------- > >> - > >> > >>> -- > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users(a)openser.org > >>> http://openser.org/cgi-bin/mailman/listinfo/users > >>> > >>> > >> > > > > > >

3 2

[OpenSER-Users] Help with subst()...
by Sajith T S 25 Jun '07

25 Jun '07

Hi, This looks simple, but I just can't wrap my (feeble) brain around it: Can anyone give a subst() regexp and replacement string to snip out "+" characters in the From header field's display name as well as user part of the URI? (There's a PSTN gateway that sends in caller's number with a "+" prefix and our users do not like it since they can't call back without manually editing out the "+"...) Thanks, Sajith.

1 0

[OpenSER-Users] how to calculate PDD and ASR
by ram 25 Jun '07

25 Jun '07

Hi iam using openser with mysql accounting can some one assist me how to calculate ASR and PDD for the calls recomend some documentation to read would be great ram

1 0

[OpenSER-Users] Presence
by KUMAR 25 Jun '07

25 Jun '07

Hi all, Can anyone please suggest me some good place to learn presence in OpenSER using SIMPLE (except the RFC). thank you kumar

1 0

[Serusers] add prefix to source
by Szasz Szabolcs 24 Jun '07

24 Jun '07

Hi all I need to add a prefix to the source uri/number. How ca I do this? Can you help me guys? Thanks a lot Szasz Szabolcs -- This message was scanned for spam and viruses by BitDefender. For more information please visit http://linux.bitdefender.com/

1 0

[Serusers] Out-of-office-reply: Serusers Digest, Vol 50, Issue 24
by David.Monahan＠aepona.com 24 Jun '07

24 Jun '07

I am on vacation and will not have access to email until I return to the office on 4th July.

1 0

[OpenSER-Users] User-Agent
by Christian Heidinger 24 Jun '07

24 Jun '07

Is it possible to modify the User-Agent/Server [Server: OpenSER (1.2.1-notls(x86_64/linux)).] field that OpenSER sends out? We use a variety of open source projects in production and prefer to sheild their true identities for security reasons (ie. preventing us from becoming the next victim of the newest exploit).

2 1

[Serusers] Cannot register with iptel.org
by Andrey Kuprianov 23 Jun '07

23 Jun '07

Hi, Im not able to register my latest X-lite phone with iptel.org. It times out. Any idea why? Any additional setup parameters i need to know of? Andrey.

2 1

[OpenSER-Users] using remove_hf and append_hf in both route and failure route
by Mik Cheez 22 Jun '07

22 Jun '07

I use remove_hf and append_hf to modify the RPID in the Request_Route, which works fine, but then if I try it in Failure_Route I end up with two RPID headers. Is it possible the revert_uri() is doing something to it? If not, can you just not do a remove/append like this? I checked to make sure the remove_hf is returning true. I triple checked to make sure my variable contains only one header. Here are some snippets from my config: route { ... ... if (method=="INVITE") { perl_exec("command", "0"); /*sets rpid and ouri vars*/ if (is_avp_set("$avp(rpid)")){ if(is_present_hf("Remote-Party-ID")){ remove_hf("Remote-Party-ID"); } append_hf("$avp(rpid)"); } avp_pushto("$ru","$avp(ouri)"); ... ... failure_route[1]{ if(method == "INVITE" ....blah blah blah revert_uri(); perl_exec("command", "1"); /*sets rpid and ouri vars*/ if (is_avp_set("$avp(rpid)")){ if(is_present_hf("Remote-Party-ID")){ remove_hf("Remote-Party-ID"); } append_hf("$avp(rpid)"); } avp_pushto("$ru","$avp(ouri)"); ... ... Best regards, Mik Cheez

2 3

[OpenSER-Users] RE: [Users] out of memory - please help
by Papadopoulos Georgios 22 Jun '07

22 Jun '07

Hi Bogdan, Bad news, I am afraid. I got another "out of memory". I did another memory dump on process 10714 this time. Actually there is one dump that I did earlier to see if there were any segments allocated by subst_str. And I did a second dump after the "out of memory" happened. The log file is: http://www.real.gr/files/openser.20070622.log.gz thank you George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Friday, June 22, 2007 1:44 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] out of memory - please help > > Hi George, > > Papadopoulos Georgios wrote: > > Hi Bogdan, > > > > The server has been running for about 18 hours without any "out of > > memory" messages. Usually these appeared after a few hours. > I think it > > is fixed, but I will keep you posted if it reappears. Thanks a lot! > > > That is super! I will commit the patch on SVN. > > I guess the same fix needs to go in subst_uri() and maybe > also add a > > check if(result->s!=0) ? > > > well, in subst_uri() there is no need because the result->s > chunk is transferred to msg->new_uri (and freed later from > there); in subst_user(), the result->s chunk was copied by > the SET_USER_T action, so we need to free it also. > > Why did this only appear on the first receiver child? I > would expect > > that all children receive an equal amount of traffic so they should > > all run out of memory more or less at the same time. Except > if when a > > message is received, Openser finds the first available > child and does > > this by trying child 1,2,3... Is this the case? > > > this scheduling is done by kernel and there is no guarantee > of a uniform distribution in case of low traffic. > > Last (related?) question: There is a thread that the first receiver > > child starts, what is its purpose? > > > you mean a process (there are no threads in openser). it > might be a listener of mi_fifo. > > Regards, > Bogdan Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 2

[OpenSER-Users] DB_Extra and AVP Problem
by Alexandre dos Reis Ferreira 22 Jun '07

22 Jun '07

Hello All, I have a problem with db extra and AVP. Some times don’t log the values of AVP in the database. Script. Openser Version 1.2.1. ... modparam("acc", "db_extra", "from_uri=$fu;i_uri=$ou;o_uri=$ru;cod_op=$avp(s:cod_op);ip_do_originador=$si;flag_followme=$avp(s:flag_follow_me)") ... route[2] { ... $avp(s:cod_op) = $(rU{s.select,0,#}); xlog("L_ALERT", "Test_01 RURI=$rU AVP=$avp(s:cod_op) To=$tu IP=$si\n"); ... } Example using the scene of a call. For RURI = sip:06#022xxxxxxxx@domain.com I have the following result in logs: => Log file Jun 21 19:20:33 ser01-brt openser-1.2.1[14952]: Test_01 RURI=06#022xxxxxxxx AVP=06 To=sip:022xxxxxxxx@domain IP=x.x.x.x => Insert in database ( The values of cod_op and flag_follow_me is null) T 10.5.0.19:33875 -> 10.5.0.22:3306 [AP] insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,i_uri,o_uri,cod_op,ip_do_originador,flag_followme ) values ('INVITE','a7ba357706b2e329','as10700de1','1cefcf0885d9adbf(a)192.168.1.19','200','OK','2007-06-21 19:20:33','sip:6101@domain,'sip:022xxxxxxxx@domain','sip:06#022xxxxxxxx@x.x.x.x:5060;transport=udp',' ','x.x.x.x',' ') The problem is intermittent and occurs in 15% of the calls. Does anybody know how can i resolve this question? Best regards, Areis. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.472 / Virus Database: 269.9.4/860 - Release Date: 21/6/2007 17:53

1 0

[Serusers] sip and rtp forwarding problem
by Martin Drasar 22 Jun '07

22 Jun '07

Hello everyone, I am a complete beginner with ser and I would like to ask you for a help. I have such scenario. There are MRCP servers (ASR and TTS) accessible only through PPTP vpn. Because I couldn't find PPTP client for Solaris, there is only one possible solution for me - to have linux box connect to vpn and act as a proxy and forward SIP and rtp packets between MRCP servers and Solaris. +-------------+ | PPTP VPN | | +-------+ | | | MRCP | | | +-------+ | | || | | || | | +---------+ | +-----------+ | |Linux box|=====|Solaris box| | +---------+ | +-----------+ +-------------+ As there is no way to make MRCP servers register with proxy, I was thinking about using call forwarding... When the MRCP servers are not registered with proxy (everytime) proxy should then forward the sip messages and rtp stream to their IP address (in fact I will be making call to unregistered UA, that proxy doesn't know but I do). Problem is that does not seem like a clean solution to me. Can you please give me some pointers how to solve this issue? Thanks in advance -- Martin Drasar, Developer / Analyst OptimSys, s.r.o. drasar(a)optimsys.cz Tel: +420 541 143 065 Fax: +420 541 143 066 http://www.optimsys.cz

2 2

[Serusers] tm matching forked replies with received parameter in branch
by samuel 22 Jun '07

22 Jun '07

Hi all, Just a question about transaction matching in SER 0.9.7: In a forked request, SER receives a reply with a ;received=IP parameter after the branch parameter in the Via header and I don't know if this can affect parallel forking because it does not CANCEL the other branches.... Thanks, Samuel.

3 4

Re: [Serusers] Quick resend of INVITEs
by Stephen Helfen 22 Jun '07

22 Jun '07

Nope...not in this case. Here, the originating UA sends one INVITE. My SER is sending the multiple INVITEs before it even receives a 100 Trying from the far end. It does not happen on every call...there seems to be no pattern as to when it does happen.Thanks.SteveDate: Fri, 22 Jun 2007 09:35:36 +0200From: greger(a)teigre.comTo: steve_helfen(a)hotmail.comCC: serusers(a)iptel.orgSubject: Re: [Serusers] Quick resend of INVITEs The originating UA may be where to look. g-) Stephen Helfen wrote: Sometimes my ser proxy resends an INVITE 67 milliseconds after the original INVITE without receiving any response from the recipient. Essentially, what would normally be a 1 second timer for a resend is vastly shortened with no real explanation. If there is a Trixbox at the other end, the conversation results in a dropped call as the Trixbox continues to send repeated 200 Oks, but does not acknowledge the ACKs to those messages. The Asterix supposedly has a fix for this issue that is not included in the Trixbox. What I am trying to determine, though, is WHY the double invites are sent in the first place. Thanks, Steve Live Earth is coming. Learn more about the hottest summer event - only on MSN. Check it out! _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Make every IM count. Download Windows Live Messenger and join the i’m Initiative now. It’s free. http://im.live.com/messenger/im/home/?source=TAGWL_June07

2 1

[Users] presence
by KUMAR 22 Jun '07

22 Jun '07

Hi all, I am using openser-1.2.1+RTPProxy. I am using XLite as UA. I want to be able to store the buddy list of the users at the server. For this, I am using the presence module. But I cannot get it working. I have created the presence related tables in mysql. And those tables do show some information in them. I have attached my openser.cfg and the message dump from wireshark. The messages have been captured at the UA end. I also get the following message at the server. PRESENCE: handle_subscribe:Missing or unsupported event header field value 2(12452) PRESENCE:handle_subscribe: ERROR occured 2(12452) WARNING: script writer didn't release transaction Am I missing something basic here? Please reply. Thank you Kumar

8 18

[Serusers] need help!
by Mukesh Prajapati 22 Jun '07

22 Jun '07

hi, I m trying to add my-sql support to my SER. i install SER with .deb package. I uncommented the lines regarding my-sql.so module.... now when i try to start server it's giving error like "bad configuration file" can anybody halp me in this regard? thnx mukesh --------------------------------- We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list.

4 3

[Serusers] dns srv load balancing
by Mário Ferreira 22 Jun '07

22 Jun '07

Hi, Does anyone know if is possible for SER to perform load balancing according to DNS SRV priority and weight fields? It seems that SER ignores these fields. With use_dns_cache turned on, SER uses always one of the SRV entries, and only tries other if use_dns_failover is turned on and the first destination fails. With use_dns_cache turned off, SER uses all entries on a random way, depending of which entry appears first on the answer section. Is it necessary some configuration for SER to load balance according SRV records or this is not yet supported? regards, Mário Ferreira

2 1

[Serusers] Call between SER proxy servers of different domain
by Ranganath B 22 Jun '07

22 Jun '07

Hi, Was trying to make calls between 2 SIP soft phones which are registered with 2 different SER SIP proxy servers running on different machines but it fails always. When I make a call from Soft Phone-A registered with SER-A to Soft Phone-B registered with SER-B it fails. Soft Phone-A receives 407 response to its INVITE from SER-B and always fails to authenticate. Please provide me inputs on how to solve this problem. I have tried even adding Soft Phone-A's username and password in SER-B but none of them worked. I have another doubt about "aliases" in SER. Can I make some effective use of this and solve the problem mentioned above? Thanks in advance, Ranganath

2 1

[Serusers] Quick resend of INVITEs
by Stephen Helfen 22 Jun '07

22 Jun '07

Sometimes my ser proxy resends an INVITE 67 milliseconds after the original INVITE without receiving any response from the recipient. Essentially, what would normally be a 1 second timer for a resend is vastly shortened with no real explanation. If there is a Trixbox at the other end, the conversation results in a dropped call as the Trixbox continues to send repeated 200 Oks, but does not acknowledge the ACKs to those messages.The Asterix supposedly has a fix for this issue that is not included in the Trixbox.What I am trying to determine, though, is WHY the double invites are sent in the first place.Thanks,Steve _________________________________________________________________ Live Earth is coming. Learn more about the hottest summer event - only on MSN. http://liveearth.msn.com?source=msntaglineliveearthwlm

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[Serusers] FreeBSD xml problem
by Todd Lewis 21 Jun '07

21 Jun '07

I was having a problem compiling SER in FreeBSD. Some of the libraries are not referenced correctly. Here are the notes from the install. Compiled against Sems 2.0-rc1 and FreeBSD 6.2 all applications installed from ports. command used to compile gmake group_include="standard standard-dep" all To get some modules to compile properly, I did the following. For XMLPRC ln -s /usr/local/lib/libxml2.a /usr/lib/libxml2.a ln -s /usr/local/lib/libxml2.la /usr/lib/libxml2.la ln -s /usr/local/lib/libxml2.so /usr/lib/libxml2.so ln -s /usr/local/include/libxml2 /usr/include/libxml2 For Modules Depending on XCAP (xcap presence pa rls) ln -s /usr/local/include/iconv.h /usr/include/iconv.h Still can't get EVAL module to compile tried the following. mkdir /usr/includes/uuid ln -s /usr/local/sys/uuid.h /usr/include/uuid/uuid.h Got Error gmake[1]: Entering directory `/usr/local/ser/src/ser-2.0.0-rc1/ modules/eval' gcc -shared -Wl,-O2 -Wl,-E eval.o -luuid -o eval.so /usr/bin/ld: cannot find -luuid gmake[1]: *** [eval.so] Error 1 gmake[1]: Leaving directory `/usr/local/ser/src/ser-2.0.0-rc1/modules/ eval'

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[Serusers] Proxy for SDP Multipart Packets
by Edson 21 Jun '07

21 Jun '07

Hi... First I try MediaProxy, but it has problems with multipart SDP packets... It LOGs: error: mediaproxy/checkContentType(): invalid Content-Type for SDP message error: mediaproxy/getSDPMessage(): content type is not `application/sdp' error: use_media_proxy(): failed to get the SDP message Then I try RTPProxy... but it also didn't work. It gaves me: ERROR:check_content_type: invalid type for a message ERROR: extract_body: content type mismatching ERROR: force_rtp_proxy2: can't extract body from the message We are using a Cisco 2600 for IPIPGW (our PSTN GWs are on the H323 side). This messages appears for any call destinated to PSTN. Can anybody point me how to solve this problem? The solution based on MediaProxy had worked until last month... Edson.

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[Users] How does Openser Presence Server works?
by li wang 21 Jun '07

21 Jun '07

Dear all, There is one question for OpenSer Presence module i don't understand. I see that the Presence server takes action base on the received SIP message SUBSCRIBE/PUBLISH, I want to know which stack SIP are u using in Openser? Because i look around the SIP Servlet API standard, it doesn't contain the method like doPublish(), doUpdate(). How do u do to realise the Presence module with Openser? Thanks, li _________________________________________________________________ Ten : Messenger en illimité sur votre mobile ! http://mobile.live.fr/messenger/ten/

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[OpenSER-Users] RTPProxy or MediaProxy with Multipart SDP packets?
by Edson 21 Jun '07

21 Jun '07

Hi, guys... As the subject suggest, I'm facing a problem with Multipart SDP packets comming from a Cisco IPIPGW. I'm using MediaProxy, but reading an old e-mail on the list, it pointed that this was do a kind-of RFC race condition. RFC say that GW should respect multipart requirements according to Proxies request, but also say that Proxies should handle Multipart messages... Any way, my question is if RTPProxy lacks from the same 'problem' as MediaProxy? Can RTPProxy handle multipart SDP packets? Edson.

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[OpenSER-Users] RE: [Users] ERROR: sip_msg_cloner: cannot allocate memory
by Papadopoulos Georgios 21 Jun '07

21 Jun '07

Hi Bogdan, I followed Henning's advice and I increased the shared memory to 128MB (-m 128). I haven't seen this ERROR since then. However this is an error that I have only seen once. Maybe during those three minutes something very unusual happened. Regards, George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Thursday, June 21, 2007 4:51 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org; Henning Westerholt > Subject: Re: [Users] ERROR: sip_msg_cloner: cannot allocate memory > > HI George, > > any feedback on this? Henning was asking you to see if there > is a memory outage or leak...any result? > > regards, > bogdan > > Papadopoulos Georgios wrote: > > Hello all, > > > > I am running Openser 1.2 for several days and whithin a period of > > three minutes I got a lot of these messages: > > Jun 8 10:27:41 ser1 openser[23287]: ERROR: new_t: out of mem: > > Jun 8 10:27:41 ser1 openser[23287]: ERROR: t_newtran: new_t failed > > Jun 8 10:27:41 ser1 openser[23287]: ERROR: new_t: out of mem: > > Jun 8 10:27:41 ser1 openser[23287]: ERROR: t_newtran: new_t failed > > Jun 8 10:27:42 ser1 openser[23279]: ERROR: sip_msg_cloner: cannot > > allocate memory Jun 8 10:27:42 ser1 openser[23279]: ERROR: > new_t: out > > of mem: > > Jun 8 10:27:42 ser1 openser[23279]: ERROR: t_newtran: new_t failed > > Jun 8 10:27:42 ser1 openser[23279]: ERROR: sip_msg_cloner: cannot > > allocate memory Jun 8 10:27:42 ser1 openser[23279]: ERROR: > new_t: out > > of mem: > > Jun 8 10:27:42 ser1 openser[23279]: ERROR: t_newtran: new_t failed > > Jun 8 10:27:42 ser1 openser[23279]: ERROR:tm:relay_reply: cannot > > alloc reply shmem Jun 8 10:27:42 ser1 openser[23287]: > > ERROR:tm:relay_reply: cannot alloc reply shmem Jun 8 10:27:42 ser1 > > openser[23287]: ERROR:tm:_reply_light: cannot allocate shmem buffer > > Jun 8 10:27:42 ser1 openser[23279]: ERROR:tm:_reply_light: cannot > > allocate shmem buffer They only appeared in the log for a > few minutes > > and then they stopped. > > After this happened I got the following statistics and ps info. > > [root@ser1 sbin]# ./openserctl fifo get_statistics all > > core:rcv_requests = 34449564 core:rcv_replies = 14967742 > > core:fwd_requests = 78714 core:fwd_replies = 15461 > core:drop_requests > > = 49337 core:drop_replies = 0 core:err_requests = 0 > core:err_replies = > > 250 core:bad_URIs_rcvd = 457 core:unsupported_methods = 471 > > core:bad_msg_hdr = 389075 shmem:total_size = 33554432 > shmem:used_size > > = 4467768 shmem:real_used_size = 4731144 shmem:max_used_size = > > 33052304 shmem:free_size = 28823288 shmem:fragments = 9383 > > sl:1xx_replies = 0 sl:2xx_replies = 16962574 sl:3xx_replies = 2306 > > sl:4xx_replies = 4501596 sl:5xx_replies = 12366570 > sl:6xx_replies = 0 > > sl:sent_replies = 33833046 sl:sent_err_replies = 275 > sl:received_ACKs > > = 49337 tm:received_replies = 14952031 tm:relayed_replies = > 14702762 > > tm:local_replies = 69373 tm:UAS_transactions = 14671506 > > tm:UAC_transactions = 0 tm:2xx_transactions = 14551357 > > tm:3xx_transactions = 0 tm:4xx_transactions = 113514 > > tm:5xx_transactions = 9327 tm:6xx_transactions = 1 > > tm:inuse_transactions = 248 usrloc:registered_users = 2668 > > usrloc:location-users = 2668 usrloc:location-contacts = 2844 > > usrloc:location-expires = 53356 registrar:max_expires = 600 > > registrar:max_contacts = 10 registrar:default_expire = 600 > > registrar:accepted_regs = 14722245 registrar:rejected_regs = 43947 > > > > [root@ser1 sbin]# ./openserctl ps > > Process:: ID=0 PID=23265 Type=attendant > > Process:: ID=1 PID=23279 Type=receiver child=0 sock= > > xxx.xxx.xxx.xxx:5060 > > Process:: ID=2 PID=23284 Type=receiver child=1 sock= > > xxx.xxx.xxx.xxx:5060 > > Process:: ID=3 PID=23287 Type=receiver child=2 sock= > > xxx.xxx.xxx.xxx:5060 > > Process:: ID=4 PID=23290 Type=receiver child=0 sock= > > yyy.yyy.yyy.yyy:5060 > > Process:: ID=5 PID=23292 Type=receiver child=1 sock= > > yyy.yyy.yyy.yyy:5060 > > Process:: ID=6 PID=23293 Type=receiver child=2 sock= > > yyy.yyy.yyy.yyy:5060 > > Process:: ID=7 PID=23297 Type=receiver child=0 sock= > > zzz.zzz.zzz.zzz:5060 > > Process:: ID=8 PID=23300 Type=receiver child=1 sock= > > zzz.zzz.zzz.zzz:5060 > > Process:: ID=9 PID=23303 Type=receiver child=2 sock= > > zzz.zzz.zzz.zzz:5060 > > Process:: ID=10 PID=23306 Type=receiver child=0 sock= > > qqq.qqq.qqq.qqq:5060 > > Process:: ID=11 PID=23309 Type=receiver child=1 sock= > > qqq.qqq.qqq.qqq:5060 > > Process:: ID=12 PID=23310 Type=receiver child=2 sock= > > qqq.qqq.qqq.qqq:5060 > > Process:: ID=13 PID=23311 Type=receiver child=0 sock= > > vvv.vvv.vvv.vvv:5060 > > Process:: ID=14 PID=23318 Type=receiver child=1 sock= > > vvv.vvv.vvv.vvv:5060 > > Process:: ID=15 PID=23321 Type=receiver child=2 sock= > > vvv.vvv.vvv.vvv:5060 > > Process:: ID=16 PID=23324 Type=timer > > Does anyone know what might be causing these errors? > > > > Thank you > > > > George > > > > > > > > > > Disclaimer > > > > The information in this e-mail and any attachments is > confidential. It > > is intended solely for the attention and use of the named > > addressee(s). If you are not the intended recipient, or person > > responsible for delivering this information to the intended > recipient, > > please notify the sender immediately. Unless you are the intended > > recipient or his/her representative you are not authorized to, and > > must not, read, copy, distribute, use or retain this message or any > > part of it. E-mail transmission cannot be guaranteed to be > secure or > > error-free as information could be intercepted, corrupted, lost, > > destroyed, arrive late or incomplete, or contain viruses. > > > > > ---------------------------------------------------------------------- > > -- > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

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[Users] ERROR: sip_msg_cloner: cannot allocate memory
by Papadopoulos Georgios 21 Jun '07

21 Jun '07

Hello all, I am running Openser 1.2 for several days and whithin a period of three minutes I got a lot of these messages: Jun 8 10:27:41 ser1 openser[23287]: ERROR: new_t: out of mem: Jun 8 10:27:41 ser1 openser[23287]: ERROR: t_newtran: new_t failed Jun 8 10:27:41 ser1 openser[23287]: ERROR: new_t: out of mem: Jun 8 10:27:41 ser1 openser[23287]: ERROR: t_newtran: new_t failed Jun 8 10:27:42 ser1 openser[23279]: ERROR: sip_msg_cloner: cannot allocate memory Jun 8 10:27:42 ser1 openser[23279]: ERROR: new_t: out of mem: Jun 8 10:27:42 ser1 openser[23279]: ERROR: t_newtran: new_t failed Jun 8 10:27:42 ser1 openser[23279]: ERROR: sip_msg_cloner: cannot allocate memory Jun 8 10:27:42 ser1 openser[23279]: ERROR: new_t: out of mem: Jun 8 10:27:42 ser1 openser[23279]: ERROR: t_newtran: new_t failed Jun 8 10:27:42 ser1 openser[23279]: ERROR:tm:relay_reply: cannot alloc reply shmem Jun 8 10:27:42 ser1 openser[23287]: ERROR:tm:relay_reply: cannot alloc reply shmem Jun 8 10:27:42 ser1 openser[23287]: ERROR:tm:_reply_light: cannot allocate shmem buffer Jun 8 10:27:42 ser1 openser[23279]: ERROR:tm:_reply_light: cannot allocate shmem buffer They only appeared in the log for a few minutes and then they stopped. After this happened I got the following statistics and ps info. [root@ser1 sbin]# ./openserctl fifo get_statistics all core:rcv_requests = 34449564 core:rcv_replies = 14967742 core:fwd_requests = 78714 core:fwd_replies = 15461 core:drop_requests = 49337 core:drop_replies = 0 core:err_requests = 0 core:err_replies = 250 core:bad_URIs_rcvd = 457 core:unsupported_methods = 471 core:bad_msg_hdr = 389075 shmem:total_size = 33554432 shmem:used_size = 4467768 shmem:real_used_size = 4731144 shmem:max_used_size = 33052304 shmem:free_size = 28823288 shmem:fragments = 9383 sl:1xx_replies = 0 sl:2xx_replies = 16962574 sl:3xx_replies = 2306 sl:4xx_replies = 4501596 sl:5xx_replies = 12366570 sl:6xx_replies = 0 sl:sent_replies = 33833046 sl:sent_err_replies = 275 sl:received_ACKs = 49337 tm:received_replies = 14952031 tm:relayed_replies = 14702762 tm:local_replies = 69373 tm:UAS_transactions = 14671506 tm:UAC_transactions = 0 tm:2xx_transactions = 14551357 tm:3xx_transactions = 0 tm:4xx_transactions = 113514 tm:5xx_transactions = 9327 tm:6xx_transactions = 1 tm:inuse_transactions = 248 usrloc:registered_users = 2668 usrloc:location-users = 2668 usrloc:location-contacts = 2844 usrloc:location-expires = 53356 registrar:max_expires = 600 registrar:max_contacts = 10 registrar:default_expire = 600 registrar:accepted_regs = 14722245 registrar:rejected_regs = 43947 [root@ser1 sbin]# ./openserctl ps Process:: ID=0 PID=23265 Type=attendant Process:: ID=1 PID=23279 Type=receiver child=0 sock= xxx.xxx.xxx.xxx:5060 Process:: ID=2 PID=23284 Type=receiver child=1 sock= xxx.xxx.xxx.xxx:5060 Process:: ID=3 PID=23287 Type=receiver child=2 sock= xxx.xxx.xxx.xxx:5060 Process:: ID=4 PID=23290 Type=receiver child=0 sock= yyy.yyy.yyy.yyy:5060 Process:: ID=5 PID=23292 Type=receiver child=1 sock= yyy.yyy.yyy.yyy:5060 Process:: ID=6 PID=23293 Type=receiver child=2 sock= yyy.yyy.yyy.yyy:5060 Process:: ID=7 PID=23297 Type=receiver child=0 sock= zzz.zzz.zzz.zzz:5060 Process:: ID=8 PID=23300 Type=receiver child=1 sock= zzz.zzz.zzz.zzz:5060 Process:: ID=9 PID=23303 Type=receiver child=2 sock= zzz.zzz.zzz.zzz:5060 Process:: ID=10 PID=23306 Type=receiver child=0 sock= qqq.qqq.qqq.qqq:5060 Process:: ID=11 PID=23309 Type=receiver child=1 sock= qqq.qqq.qqq.qqq:5060 Process:: ID=12 PID=23310 Type=receiver child=2 sock= qqq.qqq.qqq.qqq:5060 Process:: ID=13 PID=23311 Type=receiver child=0 sock= vvv.vvv.vvv.vvv:5060 Process:: ID=14 PID=23318 Type=receiver child=1 sock= vvv.vvv.vvv.vvv:5060 Process:: ID=15 PID=23321 Type=receiver child=2 sock= vvv.vvv.vvv.vvv:5060 Process:: ID=16 PID=23324 Type=timer Does anyone know what might be causing these errors? Thank you George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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[OpenSER-Users] RE: [Users] out of memory - please help
by Papadopoulos Georgios 21 Jun '07

21 Jun '07

Hi Bogdan, Yes I know, that's where I got stuck too... Anyway, probably I can get rid of avp_subst(), but not immediately. I will let you know as soon as possible. subst_user() is heavily used in the script, so it has to stay... Is subst() also a suspect or not? thanks George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Thursday, June 21, 2007 3:41 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] out of memory - please help > > George, > > I was trying to reduce the investigation options focusing > only on the functions you are using....but useless :) you are > using all of them :).... > > is there any way to remove couple of them and test again? > > regards, > bogdan > > Papadopoulos Georgios wrote: > > Hi Bogdan, > > > > I am using all of the functions that you mentioned. I am also using > > subst(). Please, let me know if you need more info. > > > > best regards > > > > George > > > > > > > >> -----Original Message----- > >> From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >> Sent: Thursday, June 21, 2007 1:46 PM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [Users] out of memory - please help > >> > >> Hi George, > >> > >> looking over the log, it might be something wrong with > >> subst_str() function - I see a lot of memory allocated by this > >> function (3267 mem chunks). > >> > >> now, I have to trace the function using the subst_str() function > >> :)..So, are you using from script one of the following functions: > >> subst_uri() > >> subst_user() > >> avp_subst() > >> > >> regards, > >> bogdan > >> > >> Papadopoulos Georgios wrote: > >> > >>> Hello all, > >>> > >>> We have an ongoing issue with Openser running out of > memory after a > >>> few hours. It is very consistent which makes me believe it > >>> > >> is a memory > >> > >>> leak. Can someone look at the log file? I have done a mem dump on > >>> process 8573 which is the one that ran out of memory. Here > >>> > >> is the log: > >> > >>> _http://www.real.gr/files/openser.20070620.log.gz_ > >>> It always happens on the first receiver child. Our current > >>> > >> solution is > >> > >>> to restart Openser every few hours. I would like to solve > >>> > >> this at its > >> > >>> root, so please let me know if you need more info. > >>> My Openser is 1.2.x from svn, revision 2333. > >>> > >>> thank you for any help > >>> > >>> George > >>> > >>> > >>> > >>> > >>> Disclaimer > >>> > >>> The information in this e-mail and any attachments is > >>> > >> confidential. It > >> > >>> is intended solely for the attention and use of the named > >>> addressee(s). If you are not the intended recipient, or person > >>> responsible for delivering this information to the intended > >>> > >> recipient, > >> > >>> please notify the sender immediately. Unless you are the intended > >>> recipient or his/her representative you are not > authorized to, and > >>> must not, read, copy, distribute, use or retain this > message or any > >>> part of it. E-mail transmission cannot be guaranteed to be > >>> > >> secure or > >> > >>> error-free as information could be intercepted, corrupted, lost, > >>> destroyed, arrive late or incomplete, or contain viruses. > >>> > >>> > >>> > >> > --------------------------------------------------------------------- > >> - > >> > >>> -- > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users(a)openser.org > >>> http://openser.org/cgi-bin/mailman/listinfo/users > >>> > >>> > >> > > > > > >

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