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37461 discussions

[Serusers] SER in the message path
by Nick Hoffman 05 Jul '06

05 Jul '06

Several sources have told me that, once a call has been established, SER does not stay in the message path between the caller and callee. However, after examining some packet traces in Ethereal, I'm a little confused. I have the following setup: Caller -> SER -> Asterisk -> VoIP Provider -> Callee When I start tcpdump on Asterisk and place a call to the PSTN, the only devices that Asterisk is sending data to and receiving data from are SER and the VoIP provider. Doesn't this mean that SER does in fact stay in the SIP message path for the duration of a call? Thanks for explaining this, -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

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RE: [Serusers] Mediaproxy problem - how to disable P2P rtp connection
by İlker Aktuna (Koç.net) 05 Jul '06

05 Jul '06

Thanks, Is there any way to disable sending "media attribute (a): alt:1" from the client ? And how does it decide to use this alternate media connection ? P.S: I use Eyebeam client. ilker -----Original Message----- From: Alexandr Dubovikov [mailto:shurik@start4.info] Sent: Tuesday, July 04, 2006 2:14 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Mediaproxy problem - how to disable P2P rtp connection On Tue, Jul 04, 2006 at 01:47:33PM +0300, ?lker Aktuna (Ko?.net) wrote: > Hi, > > is there any solution for the problem described in my following email ? > > Thanks, > ilker > > ________________________________ > > From: ?lker Aktuna (Ko?.net) > Sent: Tuesday, June 20, 2006 11:53 AM > To: serusers(a)iptel.org > Subject: RE: [Serusers] Mediaproxy problem - how to disable P2P rtp > connection > > > Hi, > > I have found the problem. (in fact it was not a problem) The client > was using the alternate media connection information which is peer-2-peer. > How can I prevent P2P connection ? > I need to remove "media attribute (a): alt:1" from the SIP INVITE messages, but how ? replace and substr will work everywere, but after it you should to recallculate content-length of SDP. > > Thanks, > ilker > > Wbr, -- Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info AD1-UANIC * ICQ: 122351182 * http://www.start4.info <http://387555.sigclick.mailinfo.com/sigclick/00090603/04034D00/0103014A/031…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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[Serusers] Account module Help
by ravi reddy 04 Jul '06

04 Jul '06

Hi SER Users , I tried to install SER with basic configurations and also with onsip config file with these SER is working pretty good ,Now i want to test accounting on radius server i installed Radius server with sql module and it is listening on 1813 for accounting , i am using radiusclient-ng-0.5.2 and i append dictionary.ser to main dictionary file. every thing is o.k Now , how can I get the accounting details in radius server is there any specific configuration to do that ? in ser.cfg i make some changes as shown in SER_RADIUS HOW TO's but i dont understand the way to move ........... :-( so any body can please tell me an out line scenario how to send account packets to radius server so that i will work on that ; Thanks in Advance Regards' Ravi.

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[Serusers] debugging mode ser
by Giuseppe Parlato 04 Jul '06

04 Jul '06

Hi all, how can I debug SER or start ser in debugging mode ? ser -E -ddd.. is the only way ? thnak you Giuseppe

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[Serusers] Open Source Prepaid Card System
by Concy Pereira 04 Jul '06

04 Jul '06

Hi Daniel, Can I use Trixbox as a Prepaid calling card system?.Trixbox comes with everything ready including the A2billing Message: 11 Date: Mon, 3 Jul 2006 13:55:01 -0700 (PDT) From: Dave <ddx66(a)yahoo.com> Subject: Re: [Serusers] Open Source Prepaid Card System To: Daniel Salama <lists(a)infoway.net>, Mailing List SER User Mailing List <serusers(a)iptel.org> Message-ID: <20060703205501.68418.qmail(a)web32102.mail.mud.yahoo.com> Content-Type: text/plain; charset=iso-8859-1 I recommend A2Billing - Great software and integrates easily w/ SER. --- Daniel Salama <lists(a)infoway.net> wrote: > Looking for recommendations for prepaid calling card > billing system > for SER (and possibly integrate with Asterisk for > IVR purposes). > > Thanks, > Daniel >

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RE: [Serusers] Mediaproxy problem - how to disable P2P rtp connection
by İlker Aktuna (Koç.net) 04 Jul '06

04 Jul '06

Hi, is there any solution for the problem described in my following email ? Thanks, ilker ________________________________ From: İlker Aktuna (Koç.net) Sent: Tuesday, June 20, 2006 11:53 AM To: serusers(a)iptel.org Subject: RE: [Serusers] Mediaproxy problem - how to disable P2P rtp connection Hi, I have found the problem. (in fact it was not a problem) The client was using the alternate media connection information which is peer-2-peer. How can I prevent P2P connection ? I need to remove "media attribute (a): alt:1" from the SIP INVITE messages, but how ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 6:13 PM To: serusers(a)iptel.org Subject: RE: [Serusers] Mediaproxy problem Hi, Please help me with the following problem. Do I need SRV dns records even if I don't use load balancing ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, Please help me with the following problem. Do I need SRV dns records even if I don't use load balancing ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, I have found the problem. (in fact it was not a problem) The client was using the alternate media connection information which is peer-2-peer. How can I prevent P2P connection ? I need to remove "media attribute (a): alt:1" from the SIP INVITE messages, but how ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 6:13 PM To: serusers(a)iptel.org Subject: RE: [Serusers] Mediaproxy problem Hi, Please help me with the following problem. Do I need SRV dns records even if I don't use load balancing ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, Please help me with the following problem. Do I need SRV dns records even if I don't use load balancing ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker <http://387555.sigclick.mailinfo.com/sigclick/0B0B0305/04074D01/0A02054C/030…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

2 1

[Serusers] Error in loading Module Please Help
by ravi reddy 04 Jul '06

04 Jul '06

Hello , I am using SER-0.9.6 and FreeRadius-1.1.1 (for getting accouting logs) i made necessary dictionary appends but when i tried to start SER iam getting these messages :-( These two errors it is showing 0(0) ERROR: acc: can't get code for the User-Name attribute 0(0) init_mod(): Error while initializing module acc Please tell me where iam doing mistake. Thank You Regards Ravi

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[Users] Performance troubleshooting
by Dimo 04 Jul '06

04 Jul '06

Hi, I am having performance issues with openser running on Fedora. Here is the scenario: I have a openser which is sending calls to two asterisks, doing radius accounting and local syslog accounting, the database for user location is a local MySQL and the asterisks are load-balanced through dispatcher. i have modules: loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/avp_radius.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/options.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/auth_radius.so" loadmodule "/usr/local/lib/openser/modules/dispatcher.so" Now the problem is that if i use sipp to send INVITE with 10-20 call attempts per seccond at one time ser just timeouts without answering for 1-2 seconds and sipp has to retransmit the packets. The timeout which sipp waits is 500ms so it should be enough. I have something like on 500 call attempts i get 700 retransmits. I will start stripping the config to troubleshoot but i was wondering if you can give me some idea what the problem might be or some good advice on troubleshooting, where should i look, and if you have any bes practices to recommend. BR, Dimitar

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[Users] Limited credit
by Jeff Williams 04 Jul '06

04 Jul '06

Is it possible to support limited credit customers with openser? Jeff

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Re: [Serusers] RTP Proxy and SIP Proxy
by Andrey Kouprianov 04 Jul '06

04 Jul '06

Hi! This is just my own opinion. See inline > Hi, > > I World like to use SER with a point-to-multipoint sip softphone > application > That figures you need a conferencing capability. > This are my questions: > > 1- Is SER capable of handle that kind of signalling flow? SER has little to do with it. All depends on how the clients will generate their INVITEs, which SER will handle appropriately. > > 2- Is the RTP proxy able to fork RTP to more than one destination? > RTPproxy (as well as mediaproxy) is not a conferencing server but merely a node that helps NATed clients talk to each other, i.e. RTPproxy is a relay node. That's all. If you are looking for a conferencing software for audio, I would recommend using Asterisk. As for vdo and other multimedia, I have a little idea because, myself, Im struggling to find it. > > Thanks. Regards, Andrey.

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