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19 participants
37571 discussions

[Serusers] bridging SER peering other Sip Servers
by Joao Pereira 28 Oct '05

28 Oct '05

Hello to all, I have my SER in bridging mode (with two IPs in two non routable networks), with rtpproxy and nathelper: rtpproxy -l 10.0.0.135/193.136.2.205 And the clients in the private and public networks can speak each other with no problems. But in the calls from the private network to external SERs (for example: user(a)iptel.org), the speech only passes one way (from my private IPclient to the user(a)iptel.org) Can someone tell me what could be the problem? These are my routing lines: if (lookup("interno")) { if (dst_ip == 10.0.0.135){ if (force_rtp_proxy("FAII")) t_on_reply("1"); } if (dst_ip == 193.136.2.205){ if (force_rtp_proxy("FAEI")) t_on_reply("1"); } } else if (lookup("externo")) { if (dst_ip == 10.0.0.135){ if (force_rtp_proxy("FAIE")) t_on_reply("1"); } if (dst_ip == 193.136.2.205){ if (force_rtp_proxy("FAEE")) t_on_reply("1"); } } else { sl_send_reply("403", "Utilizador nao Encontrado"); break; }; (....) Thats my external routing section: if( !uri=~"@fccn.pt"){ if (!method=="REGISTER") # internal IPs must use RTP proxy to call user(a)iptel.org if (dst_ip == 10.0.0.135){ if (force_rtp_proxy("FAIE")) t_on_reply("1"); route(1); break; } record_route(); t_on_reply("3"); t_relay(); break; } (....) onreply_route[1] { if (!(status=~"183" || status=~"200")) break; force_rtp_proxy("FA"); }

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[Serusers] how can i avoid multiple logins?
by Ryan Pagquil 28 Oct '05

28 Oct '05

Hi, How can I avoid multiple logins using the same username? I want to secure the users so that only the authentic user can only log on? Is there any modules or commands to make this? For example user1 is logged I don't want other people to log also using user1 since user1 is currently logged on. Thanks in advance, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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[Users] steps to build rpm packages
by Daniel-Constantin Mierla 28 Oct '05

28 Oct '05

Hello, I try to summarize basic steps for generating the RPMs (exemplified for RedHat). The RPM experts can update/correct these guidelines. a) an archive with the sources will be available for download (openser-1.0.0_src.tar.gz) once everything is prepared b) download it in /usr/src/redhat/SOURCES c) copy the spec file that is designed for your distribution in same folder (for RedHats is better to try first openser.spec.fedora) as openser.spec d) edit openser.spec as follows: ** Release: must be set to a string that identifies the distro (e.g., for fedora core 3.0, it should be "Release: fc3.0") ** update the packager name and address to yours ** comment the packages that you cannot generate (e.g., jabber, radius) - there are several sections for each package, which can be easily identified by the name of the package -- you have to comment the whole section e) copy the init file (either openser.init or openser.init.SuSE, from sip-server/packaging/rpm) in /usr/src/redhat/SOURCES f) cd /usr/src/redhat/SOURCES g) rpmbuild -ba openser.spec For SuSE there might be some small differences, but basically is the same process. These guidelines are for those to have root access. If not, then you have to follow first http://dag.wieers.com/howto/bits/rpm-build-user.php (more help via google searching after "building rpm as user"). If everything goes fine, you should get at the end something like (e.g., I built openser base and mysql package): Wrote: /usr/src/redhat/SRPMS/openser-1.0.0-rhel3.0.src.rpm Wrote: /usr/src/redhat/RPMS/i386/openser-1.0.0-rhel3.0.i386.rpm Wrote: /usr/src/redhat/RPMS/i386/openser-mysql-1.0.0-rhel3.0.i386.rpm These are the files we need. If possible, please make them available on a ftp or http site and send me the address to take them. I will notify you back onces I got them, to remove them from your location, if you wish. Otherwise, I will accept attachments to email (at this address), but let me know first that you are going to send them, to be prepared :-) Cheers, Daniel

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[Serusers] How does user agents send log-off request?
by Ryan Pagquil 28 Oct '05

28 Oct '05

Hi, How does User agents sends log-off signal to SER? I know that for the user agent to log in it needs to send a REGISTER request, what request should the user agent need to send SER to be able to log off and be removed from the location table? Could you please explain how the user agent does this... Thanks in advance, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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[Serusers] question on Adding aliases
by Eric Haskins 28 Oct '05

28 Oct '05

We are trying to get inbound DID's working on our SER box. Outbound is working perfectly as we are doing pre-paid. I ran this command [root@voip1 ~]# serctl alias add 6049094251 sip:5002@216.x.xxx.xx domain unknown: use usernames with domain or set default domain in SIP_DOMAIN [root@voip1 ~]# I get the same result if I put our domain in place of the IP address. We had someone else set this box up for us and I am having problems reaching them. 5002 is is the username of my test ATA logged into the system I can make outgoing calls to the PSTN What am I doing wrong?? I have been reading Archives allday trying to grasp this. The DID provider is directing that DID at the IP of this SER box Eric Haskins Lead Software Developer eric(a)v1010.com

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RE: [Serusers] end the session if no media
by Hakan YASTI 28 Oct '05

28 Oct '05

Yes, I know about it. I can see the active sessions from the session.py. But do not know how to end the seesion, if there is no rtp session and the bye has not come. My session con troller does not check the rtp session because the media is proxied at SER. As I know, may be I am wrong, the ser is managing the media proxy. Ser says to media proxy to start when a call is proxied, and end when the call is finished. Do you suggest a SBC that can do this function..? Thanks, Best Regards, ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Thursday, October 27, 2005 9:20 AM To: Hakan YASTI; serusers(a)lists.iptel.org; ritesh.j(a)net4.in Subject: Re: [Serusers] end the session if no media session.py will show if the session is active. Also when a call is put on hold, there will be no rtp stream. I don't know if your SBC handles that. Or is your question something else? g-) ----- Original Message ----- From: Hakan YASTI <mailto:hakan.yasti@teletek.net> To: Greger V. Teigre <mailto:greger@teigre.com> ; serusers(a)lists.iptel.org ; ritesh.j(a)net4.in Sent: Wednesday, October 26, 2005 09:18 AM Subject: RE: [Serusers] end the session if no media Hi Teigre, Thanks for your reply and kind suggestions. In fact, I am using a session controller for sip calls. And it alerts me via email when a sessiıon is reached for about 60 minutes. At that time, I check it manually and kil the session if the session is abondened. By sending BYE messages to both legs. From the PSTN side I get the reply what I assume ( 200 OK ) but from the sip side I get No transaction exist. Because the session was ended before and session controller couldn’t get the BYE message for some reasons. ( Network failure or something else ). What I am looking for that, from the media proxy, I see the active rtp session when I use media proxy for the calls. And I assume that, when the call has finished and couldn’t send or get BYE message, there will be no rtp stream going on. May be I am wrong for this. If rtp session is still open when the a side hunp up and couldn’t send bye , there is nothing to do except killing the session with sending bye messages to both sides, because there is no way to understand the call is active or not. Best Regards, Hakan ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Wednesday, October 26, 2005 10:00 AM To: Hakan YASTI; serusers(a)lists.iptel.org; ritesh.j(a)net4.in Subject: Re: [Serusers] end the session if no media Hakan and Ritesh, It seems to me that you are both asking about the same thing. This topic has been discussed many times on serusers. Search google (site:mail.iptel.org) for "BYE stateful sipsak", I'm pretty sure you will get some info. Also, this FAQ could help: http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#q2 g-) ----- Original Message ----- From: Hakan YASTI <mailto:hyasti@teletek.net> To: serusers(a)lists.iptel.org Sent: Thursday, October 13, 2005 09:13 AM Subject: [Serusers] end the session if no media Hi, I wonder, is there any way, to end the session when the media has ended. Sometimes, we have problem at BYE message. For some reasons, the bye is not coming and may be it is coming wrong. At that time, the media is ended but the call is seen as active. I am using ser and using media proxy. Any suggestion for this issue ? Best Regards, Hakan YASTI NGN Engineer Teletek Telekomünikasyon Hizmetleri A.Þ. Ayazma Dere Cad. Akþit Plaza No:12 Fulya, Beþiktaþ ISTANBUL - TURKEY Tel : +90 212 310 22 34 Fax : +90 212 227 8700 GSM : +90 533 669 06 84 E-mail: hakan.yasti(a)teletek.net <mailto:hakan.yasti@teletek.net> http://www.teletek.net ________________________________ _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers Disclaimer: This e-mail (including any attachments) may contain confidential and/or privileged information. Copying, disclosure or distribution of the material in this e-mail without our companys' authority is strictly forbidden. If you are not the intended recipient (or have received this e-mail in error), please notify the sender and delete it from your system immediately. Our Company makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. Any opinions expressed in this message are those of the author and may not necessarily reflect the opinions of Our Company. Yasal Uyarı: Bu e-posta'nin içerdigi bilgiler (ekleri de dahil olmak üzere) gizlidir. Şirketimizin onayı olmaksızın içeriği kopyalanamaz, üçüncü kişilere açıklanamaz veya iletilemez . Bu mesajın gönderilmek istendiği kişi değilseniz (ya da bu e-posta’yı yanlışlıkla aldıysanız), lütfen yollayan kişiyi haberdar ediniz ve mesajı sisteminizden derhal siliniz. Şirketimiz bu mesajın içerdigi bilgilerin doğruluğu veya eksiksiz olduğu konusunda bir garanti vermemektedir. Bu nedenle bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından, saklanmasından sorumlu değildir. Bu mesajın içeriği yazarına ait olup, Şirketimizin görüşlerini içermeyebilir. Disclaimer: This e-mail (including any attachments) may contain confidential and/or privileged information. Copying, disclosure or distribution of the material in this e-mail without our companys' authority is strictly forbidden. If you are not the intended recipient (or have received this e-mail in error), please notify the sender and delete it from your system immediately. Our Company makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. Any opinions expressed in this message are those of the author and may not necessarily reflect the opinions of Our Company. Yasal Uyarı: Bu e-posta'nin içerdigi bilgiler (ekleri de dahil olmak üzere) gizlidir. Şirketimizin onayı olmaksızın içeriği kopyalanamaz, üçüncü kişilere açıklanamaz veya iletilemez . Bu mesajın gönderilmek istendiği kişi değilseniz (ya da bu e-posta’yı yanlışlıkla aldıysanız), lütfen yollayan kişiyi haberdar ediniz ve mesajı sisteminizden derhal siliniz. Şirketimiz bu mesajın içerdigi bilgilerin doğruluğu veya eksiksiz olduğu konusunda bir garanti vermemektedir. Bu nedenle bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından, saklanmasından sorumlu değildir. Bu mesajın içeriği yazarına ait olup, Şirketimizin görüşlerini içermeyebilir.

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[Serusers] Beer on Wednesday 26th
by Atle Samuelsen 28 Oct '05

28 Oct '05

Hi Guys And Girls, Fellow Serusers, Greger and Me where thinking of trying to join the norwegian troops at Beer Palace on Aker Brygge at 5pm, Upcoming Wednesday. Anybody Intrested in Joining for a Beer or 5? Best Regards Atle

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[Serusers] Removal of email
by voip learner 28 Oct '05

28 Oct '05

Dear sir, Please remove me from the list of the emails. Iam getting hundredsof emails and the mail folder is getting bulk. Kindly do it ASAP. Regards __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Re: Open source billing solution for SER (VoIP)
by Eric Haskins 27 Oct '05

27 Oct '05

>Hello experts, > > I was trying to set up a VoIP system with SER as my signalling >proxy and was able to get lot of useful info from this mailing list. >Thank you. > I would like to know if there exists an open source billing >solution which can fit seemlessly with SER and can handle realtime >rating ( in a prepaid scenario). My initial study on this gave me the >following link with a list of open source billing solutions. I would >also like to know some reivews on these from experts like you. Since I >have very less time to bring my project to life. > >http://www.voip-info.org/wiki/view/Open+Source+Billing+Systems > > >Thanks a lot in advance. > >Regards, >Jithu We have a working system we built that allows for rates to be preset in a db. It calculates start and stop using acc table. I dont know if we will release as gpl but if you want to see the backend functionality contact me directly. It is already setup to work in prepaid eric(a)v1010.com Eric

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[Users] TCP-alias and NAT - one more try to get an answer
by Alexander Ph. Lintenhofer 27 Oct '05

27 Oct '05

Hi All, I just wanted to ask you once again about the TCP-alias riddle. I found out, that there is a problem with the combination of fix_nated_contact(), force_tcp_alias() and NAT: Imagine following situation: Alice behind NAT: socket 172.16.0.6:2421 Nat-Box translates this to 192.168.0.13:6007 The Outbound-Proxy of Alice is 192.168.0.1 Bob is registered with 192.168.1.1 1.) Her INVITE: ==================================================================== INVITE sip:bob@192.168.1.1:2331;transport=tcp;line=wxqurd1s SIP/2.0 [...] Via: SIP/2.0/TCP 172.16.0.6:2421;received=192.168.0.13; branch=z9hG4bK-wm9jcstcboys;rport=6007 From: "Alice" <sip:alice@atlanta.com>;tag=sufzmxi0us To: "Bob" <sip:bob@biloxi.com>;tag=c9550czwtn [...] Contact: <sip:alice@172.16.0.6:2421;transport=tcp;line=fyyuh6tl> [...] ==================================================================== 2.) fix_nated_contact() doesn't work with TCP (look at nathelper.c). force_tcp_alias() now creates following tuple as TCP-alias: 192.168.0.13:6007 to 192.168.0.13:2421 Reason: The TCP-alias is not built solely from the Via-header as suggested in the draft. The portnumber is taken from the Via-header and the IP-address is taken from the source of the incoming datagram. I read it in the sourcecode and assured it by contacting Andrei! 3.) So as a result of the notfixed Contact-header of Alice's INVITE the BYE of Bob is addressed to 172.16.0.6:2421. But no TCP-alias exists for this socket :-( 4.) I made following test by rewriting the Contact-header.... ==================================================================== if (method=="INVITE") { replace("Contact: <sip:alice@172.16.0.6:2421;transport=tcp;", "Contact: <sip:alice@192.168.0.13:2421;transport=tcp;"); } ==================================================================== ...with success. Now TCP-alias works as you can see on my Ethereal-trace below! Compare the destination port of the packet to the destination port of the RURI! ==================================================================== [...] Transmission Control Protocol, Src Port: 5060 (5060), Dst Port: 6007 (6007), ... Session Initiation Protocol Request-Line: BYE sip:alice@192.168.0.13:2421;transport=tcp;line=fyyuh6tl SIP/2.0 Message Header [...] From: "Bob" <sip:bob@biloxi.com>;tag=kcsveifugd To: "Alice" <sip:alice@atlanta.com>;tag=ricaq5cy15 Contact: <sip:bob@192.168.1.1:2331;transport=tcp;line=wxqurd1s> [...] ==================================================================== Another solution: Comment the lines in nathelper.c which force the return in case of TCP or TLS. Now all works well! But why?????????????? regards, Philipp

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