- sr-users - kamailio.org

Re: [Serusers] Account module Help
by Greger V. Teigre 06 Jul '06

06 Jul '06

You probably haven't set the accounting flag in your loose_route handling. Do a test pretty early in your ser.cfg main route (before if(loose_route()) and set the appropriate flags. g-) ravi reddy wrote: > Thanks For your Response Mr. *Greger V. Teigre > > I did* it ,by checking the dictionaries by seeing > your messages in 2004 archives , > > but here i got a problem is that if i set "radius flag" in ser.cfg > near call invite then i am only getting AccStartTime but not > AccStop Time > > If i set flags near (cancel or Bye) iam getting only account stop > time and the acct start time is not showing , > > if i set two places even then iam getting only one time > > > *+----------------------+---------------------+---------------------+--------------------------+--------------------------+ > > | username | acctstarttime | acctstoptime | > callingstationid | calledstationid | > +----------------------+---------------------+---------------------+--------------------------+--------------------------+ > > | 32331001(a)81.21.34.37 <mailto:32331001@81.21.34.37> | 2006-07-04 > 18:45:50 | 2006-07-04 18:45:50 | sip:32331001@81.21.34.37 > <mailto:sip:32331001@81.21.34.37> | sip:22223333@81.21.34.37 > <mailto:sip:22223333@81.21.34.37> | > | 32331001(a)81.21.34.37 <mailto:32331001@81.21.34.37> | 2006-07-04 > 18:36:29 | 2006-07-04 18:36:29 | sip:32331001@81.21.34.37 > <mailto:sip:32331001@81.21.34.37> | sip:22223333@81.21.34.37 > <mailto:sip:22223333@81.21.34.37> | > | 32331001(a)81.21.34.37 <mailto:32331001@81.21.34.37> | 2006-07-04 > 18:17:35 | 2006-07-04 18:17:35 | sip:32331001@81.21.34.37 > <mailto:sip:32331001@81.21.34.37> | sip:22223333@81.21.34.37 > <mailto:sip:22223333@81.21.34.37> | > | 32331001(a)81.21.34.37 <mailto:32331001@81.21.34.37> | 2006-07-04 > 18:32:50 | 2006-07-04 18:32:50 | sip:32331001@81.21.34.37 > <mailto:sip:32331001@81.21.34.37> | sip:22223333@81.21.34.37 > <mailto:sip:22223333@81.21.34.37> | > +----------------------+---------------------+---------------------+--------------------------+--------------------------+ > > > so how i can get out of this trouble please help me ; > > > Thank you > regards > Ravi. > > > > * > On 7/4/06, *Greger V. Teigre* <greger(a)teigre.com > <mailto:greger@teigre.com>> wrote: > > I suggest you figure out your dictionary first (ref. your other > post). Without the attributes in place, you get nowhere. > Remember that there are several elements to radius functionality > in SER: > - standalone radius server > - dictionary file for radius server > - SER radius modules compiled and linked against radiusclient-ng > (library) > - the radiusclient.conf file defining the setup for the radius > client (used by SER radius modules) > - dictionary file for radiusclient-ng > > The attributes used are defined in SER modules. All attributes > must be found in the radiusclient.conf file (so that the > radiusclient can understand the attributes defined in the modules). > Then the radiusserver needs to understand all the attributes > (using it's own dictionary). > > It looks like this: > ser.cfg radiusauth or setflag (acc_flag) => calls radius module => > linked against radiusclient-ng.so => reads radiusclient.conf and > dictionary => sends requests on udp 1812 and 1813 => radius server > listens on ports and handles request responding back to > radiusclent, which returns data (or just ok) to the module > > g-) > > > ravi reddy wrote: >> Mr. Greger V.Teigre >> >> Thanks for your response , when i tried changing line as >> radius_log_flag the SER is showing there is no module like >> Radius_log_flag in acc module but when i keep like radius_flag >> its showing o.k >> but here my doubt is are the accounting messages will log in to >> radius??? >> my config file is accepting the radius_missed_flag... >> is this setting will log the messages in radacct???? >> >> >> Are you using FreeRadius for accounting ??? >> >> waiting for your reply >> >> Regards >> Ravi. >> >> >> >> On 7/4/06, *Greger V. Teigre* <greger(a)teigre.com >> <mailto:greger@teigre.com> > wrote: >> >> "To enable RADIUS accounting simply use radius_log_flag and >> radius_log_missed_flag parameters instead of log_flag and >> log_missed_flag. Mark transactions that should be logged with >> flags configured in the parameters." >> >> This means that you use modparam to set the flag values for >> these two and then use setflag(yourflag) where you want >> radius accounting to happen. >> g-) >> >> ravi reddy wrote: >> 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. >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> >> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >

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RE: [Serusers] SER / MediaProxy / B2BUA
by İlker Aktuna (Koç.net) 06 Jul '06

06 Jul '06

Hi everyone, I am looking for a freeware (or even opensource) sip->h323 solution which would turn my sip calls (from SER) to h323 (to any gateway that runs h323) Is there any free solution for this ? Also, I am looking for some free transcoding system which would turn my ILBC (codec) calls to g729. Is there any freeware transcoding solution ? Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/020C0603/04024D07/0307044A/036…> _____________________________________________________________________________________________________________________________________________ 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] Multiple SIP server
by Nhadie 06 Jul '06

06 Jul '06

Hi All, I setup 3 SIP servers: sip1 10.10.10.1 sip2 10.10.10.2 sip3 10.10.10.3 I setup the the DNS to do load balancing using SRV and also by defining sip.mydomain.com IN A 10.10.10.1 IN A 10.10.10.2 In A 10.10.10.3 the three servers are using a single database which is 10.10.10.1 but 10.10.10.2 and 10.10.10.3 are slaves for replicating 10.10.10.1 My only problem is that when 1 extension is registered on one sip and the other on another sip, they can't call each other. e.g. 1001(a)10.10.10.1 and 1002(a)10.10.10.2 and 1003(a)10.10.10.3 wont be able to call each other. How can I fix this? TIA Regards, Nhadie ________________________________________________ Message sent using UebiMiau 2.7

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[Serusers] Authentication prefix
by CM0002＠aol.com 05 Jul '06

05 Jul '06

I need to cut leading "00" and ad a prefix "123" to an outgoing call to allow authentication to provider. I tried several scripts but non of them works. Can somebody help me? Thanks from a SER novize! Regards Christian Last one: # send out 00 prefix to wholesale psnt termination VOIP if (uri=~"^sip:00[0-9].*@.*") { if (!is_user_in("From", "ld")) { sl_send_reply("403", "Payment required"); break; strip(2); prefix("123"); }; setflag(1); rewritehostport("64.xxx.xxx.76:5060"); if (!t_relay()) { sl_reply_error(); }; break; };

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[Serusers] Problem: can't load module vm.so
by Roberto Lopes 05 Jul '06

05 Jul '06

Good afternoon to all! If possible I'd like some help. I'm doing a school project about VoIP and till now everything has gone fine. In terms of calling everything is done... Now when I turned myself to add voicemail to SER I've been having some problems. I already managed to put voicemail working on Asterisk but that's only available for analogic-analogic, voip phone-analogic, softphone-analogic communications. So when i write down ser to put the server working this is the output: 0(14615) ERROR: load_module: could not open module </usr/local/lib/ser/modules/vm.so>: /usr/local/lib/ser/modules/vm.so: cannot open shared object file: No such file or directory 0(14615) parse error (43,13-14): failed to load module 0(14615) set_mod_param_regex: No module matching voicemail found | 0(14615) parse error (69,19-20): Can't set module parameter 0(14615) parse error (69,20-21): syntax error 0(14615) parse error (69,20-21): ERROR: bad config file (4 errors) The problem is that on the modules folder, vm.so doesn't exist. So through the source file I'd like to know how to make SER install it from the beginning of installation. I think i'd just have to do some changes in the makefile. But i've been losing a bit my patience on this. I hope someone can manage to help me. Thank you for your attention

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[Serusers] avp, selects question
by Luis Silva 05 Jul '06

05 Jul '06

Hi all! I'm trying to learn more about the new function added to ser, for manipulating the information inside the script. One problem that I found was when I tried to use %var1= "value1; %var1 +="value2; What I was trying to do was to get a value from a header field %auth_hdr = @hf_value.authorization; and then %auth_hdr += "something"; is it possible? (I got an error) then I tried another way $auth_hdr = @hf_value.authorization; %auth_hdr_something = "%$auth_hdr, something..."; and worked. but what is the difference between using that or using instead of $auth_hdr, use %auth_hdr. Can I do this: %auth_hdr_something = "%%auth_hdr, something..."; tks in advance

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[Serusers] PA Module
by Alois Schafferhofer 05 Jul '06

05 Jul '06

Hi, I have a problem with sending position coordinates with PUBLISH. If I send the x and y coordinates <x>01524.3441</x><y>04704.1020</y> via PUBLISH, SER cuts the last two digits (<x>1524.34</x><y>4704.1</y>). Why doesn't SER Notify the whole position? In which class does it cut the two digits? I hope, someone could help me. Best regards, Alois /usr/sbin/ser[11700]: handle publish request /usr/sbin/ser[11700]: get_pres_uri: _puri=sip:pda_user@fh-joanneum.at /usr/sbin/ser[11700]: handle_publish -4- p_uri= p_uri.len=23 /usr/sbin/ser[11700]: handle_publish -5- presentity=0x4069d1d8 /usr/sbin/ser[11700]: parse_pidf: sipuri=0x8216278:pres:pda_user@fh-joanneum.at contact=0x8216098:sip:pda_user@10.0.0.3 basic=0x82160b8:online location=(nil):(null) /usr/sbin/ser[11700]: parse_pidf: site=(nil):(null) floor=(nil):(null) room=(nil):(null) /usr/sbin/ser[11700]: parse_pidf: x=0x821be28:01524.3441 y=0x821be38:04704.1020 radius=(nil):(null) /usr/sbin/ser[11700]: find_presence_tuple: _p=0x4069d1d8 _p->tuples=0x4069d488 /usr/sbin/ser[11700]: publish_presentity_pidf: -1- /usr/sbin/ser[11700]: publish_presentity: -2- /usr/sbin/ser[11700]: publish_presentity: -3-: changed=1 /usr/sbin/ser[11700]: db_update_presentity starting: use_place_table=1 presid=4 /usr/sbin/ser[11700]: db_update_presentity: tuple- contact=sip:pda_user@10.0.0.3 len=21 basic=1 expires=1150916054 priority=0.500000 /usr/sbin/ser[11700]: db_update_presentity: cleaning contact from table /usr/sbin/ser[11700]: db_update_presentity: room= loc= /usr/sbin/ser[11700]: publish_presentity: -4- /usr/sbin/ser[11700]: publish_presentity: event_package=1 -1- /usr/sbin/ser[11700]: handle_publish -8- paerrno=0 /usr/sbin/ser[11700]: handled publish /usr/sbin/ser[11708]: timer_presentity: _p=0x4069d1d8 pda_user(a)fh-joanneum.at flags=1 watchers=0x4069d3a8 /usr/sbin/ser[11708]: notifying esca(a)fh-joanneum.at _p->flags=1 _w- event_package=1 _w->accept=2 _w->status=1 PUBLISH sip:pda_user@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK874819433..Max-Forwards: 70..From: <sip:pda_user@fhjoanneum .at>;tag=540877786..To: <sip:pda_user@fh-joanneum.at>..Call-ID: 996402808-569651771- 701518565..CSeq: 10 PUBLISH..Contact: <sip:pda_user@10.0.0.3>..Con tent-Type: application/cpim-pidf+xml..Content-Length: 396..Expires: 50..Event: presence..User- Agent: SIP .NET 1.0 evaluation version, www.independents oft.com....<?xml version="1.0" encoding="utf-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" entity="pres:pda_user@fh-joanneum.at"><tuple id="9da019 ef-d0ea-4de0-8ad3-edacae0ca89c"><status><basic>online</basic><geopriv><locationinfo>< civilAddress><x>01524.3441</x><y>04704.1020</y></civilAddress></ location-info></geopriv></status><contact priority="0.8">sip:pda_user@10.0.0.3</contact></tuple></presence>.. 10.15.200.57:5060 -> 10.15.200.136:5060 NOTIFY sip:esca@10.15.200.136 SIP/2.0..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bK766a.c7db8933.0..To: <sip:esca@fhjoanneum. at>;tag=z9hG4bK29818902. .From: <sip:pda_user@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-f2af..CSeq: 10 NOTIFY..Call-ID: 603541788327@10.15.200.136..Content-Length: 464..User-Agent: Sip EXpress router(0.9.3 (i386/linux))..Event: presence..Content-Type: application/pidf+xml..Subscription-State: active;expires=3479. ...<?xml version="1.0"?>..<!DOCTYPE presence PUBLIC "//IETF//DTD RFCxxxx PIDF 1.0//EN" "pidf.dtd">..<presence entity="sip:pda_user@fh-joanneum.at">..< tuple id="9r28r49">tid6e65cf14">.. <contact priority="0.500000">sip:pda_user@10.0.0.3</contact>.. <status>.. <basic>open</basic>.. <geopriv>< location-info><civilAddress> <x>1524.34</x>.. <y>4704.1</y>.. </civilAddress></location-info></geopriv>.. </status>.. </status>..</tupl e>..</presence>..

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[Serusers] SER / MediaProxy / B2BUA
by Daniel Salama 05 Jul '06

05 Jul '06

After reading this forum: http://www.voipuser.org/forum_topic_4468.html it made me wonder, whether or not you really need B2BUA if you already have Mediaproxy in your environment. I know the purpose of Mediaproxy is to help with NAT situations. However, given the fact that Mediaproxy is always in the media path, couldn't it be ALSO used to "terminate" a call in progress, the same way that B2BUA can? And by B2BUA I refer to the Asterisk B2BUA, all within the context of prepaid type services. Any comments? Thanks, Daniel

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[Users] How can i do accounting in such case
by tele 05 Jul '06

05 Jul '06

Hi all, In the traditional telephony who owned the call is the caller and is it possible for the caller to change the thorn without lost conversation but after 60sec if the call doesn't live, the central of caller disconnect the call. My test: I call from my UA to PSTN and during the call i disconnect my UA from IP connection. in my ethereal i can see now only RTP traffic from PSTN try to reach my UA, but after minutes nothing happen. note that i'm not using an rtpproxy right now. maybe with an rtpproxy it works and disconnect the call. my question is how can i generate accounting for the call if my UA disconnect the ethernet cable? the rtpproxy disconnect the call and generate the BYE for the accounting of the call. is it correct? hmm with a b2bua i can solve this issue but if i can do this only with openser is more better. anyone incurred if this kind of issue? thanks

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[Serusers] IP address spoofing
by Nick Hoffman 05 Jul '06

05 Jul '06

Hi guys. Say you have this setup, with an account for the caller on both Asterisk and SER: Caller -> SER -> Asterisk -> VoIP Provider -> Callee If the caller were to spoof SER's IP address and place a call directly to Asterisk (thus circumventing SER), what would happen? If the call was in fact setup, obviously the caller would not receive any audio from the callee. However, would the call be setup? When Asterisk responds to the caller's request and sends SIP packets back (to SER), would SER say "I don't know anything about this call! Asterisk, kill this call please."? Thanks for your input! -- 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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[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. _____________________________________________________________________________________________________________________________________________

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[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

3 4

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

04 Jul '06

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

2 2

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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[Serusers] Open Source Prepaid Card System
by Daniel Salama 03 Jul '06

03 Jul '06

Looking for recommendations for prepaid calling card billing system for SER (and possibly integrate with Asterisk for IVR purposes). Thanks, Daniel

2 1

Re: [Users] Debian Etch with MySQL module
by Jason Hoss 03 Jul '06

03 Jul '06

-- jh(a)tutpro.com wrote: From: Juha Heinanen <jh(a)tutpro.com> To: jason_hoss(a)quintum.com Cc: <users(a)openser.org> Subject: [Users] Debian Etch with MySQL module Date: Mon, 3 Jul 2006 22:57:16 +0300 libmysqlclient14 is available in sarge. you could try to force installation if you have with libmysqlclient15 installed. -- juha I have set up apt to pull from the ETCH repository by adding the following line to sources.list: deb http://www.openser.org/debian etch main This failure should not happen because the mysql installed is the base install for ETCH and the packages should work for ETCH. It seems that a recompile against this newer library should be done to make this package work for a fresh install of ETCH. Anyone have any suggestions on how to work around this? I would like to use force as a last resort. Jason Hoss

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[Users] Debian Etch with MySQL module
by Jason Hoss 03 Jul '06

03 Jul '06

2 1

[Serusers] RTP Proxy and SIP Proxy
by Gustavo Moreira 03 Jul '06

03 Jul '06

Hi, I World like to use SER with a point-to-multipoint sip softphone application This are my questions: 1- Is SER capable of handle that kind of signalling flow? 2- Is the RTP proxy able to fork RTP to more than one destination? Thanks.

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[Serusers] Acc.so module please help
by ravi reddy 03 Jul '06

03 Jul '06

Hi SER users I am trying to test SER+FreeRadius (for accounting) and iam getting the follwing errors in log messages ERROR MESSAGES:-( localhost /usr/local/sbin/ser[3503]: Maxfwd module- initializing localhost ser[3503]: ERROR: acc: can't get code for the Sip-Translated-Request-URI attribute localhost ser[3503]: init_mod(): Error while initializing module acc even then i changed the Attribute i am getting the same error please help regard this issue my SER.cfg looks like this loadmodule "/usr/local/lib/ser/modules/acc.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "radius_flag", 1 ) modparam("acc", "radius_missed_flag", 2) modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "service_type", 15) # -- 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) # ------------------------- 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 >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # 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("81.21.34.37", "subscriber")) { www_challenge("81.21.34.37", "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] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; }

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Re: [Serusers] problems with t_relay_to
by Zenone 03 Jul '06

03 Jul '06

Excuse me for my light explanation of my problem but your solution works! Thanks! Michel --------- Message d'origine -------- De: Andrey Kouprianov <andrey.kouprianov(a)gmail.com> A: serusers(a)iptel.org <serusers(a)iptel.org> Objet: Re: [Serusers] problems with t_relay_to Date: 03/07/06 13:00 > You did not specify your problem though... > > Anyway, try t_relay_to_udp() or t_relay_to_tcp() > > On 7/3/06, Zenone <zenone(a)telip.com> wrote: > > Hi, > > I have problems to make my ser work since two 't_relay_to' have been added > > in 'ser.cfg'. > > Here is my ser.cfg: > > # > > # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ > > # > > # simple quick-start config script > > # > > > > # ----------- 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/local/ser_directory/lib/ser/modules/mysql.so" > > > > loadmodule "/usr/local/ser_directory/lib/ser/modules/sl.so" > > loadmodule "/usr/local/ser_directory/lib/ser/modules/tm.so" > > loadmodule "/usr/local/ser_directory/lib/ser/modules/rr.so" > > loadmodule "/usr/local/ser_directory/lib/ser/modules/maxfwd.so" > > loadmodule "/usr/local/ser_directory/lib/ser/modules/usrloc.so" > > loadmodule "/usr/local/ser_directory/lib/ser/modules/registrar.so" > > loadmodule "/usr/local/ser_directory/lib/ser/modules/textops.so" > > > > # Uncomment this if you want digest authentication > > # mysql.so must be loaded ! > > loadmodule "/usr/local/ser_directory/lib/ser/modules/auth.so" > > loadmodule "/usr/local/ser_directory/lib/ser/modules/auth_db.so" > > > > modparam("tm", "fr_inv_timer", 90) > > # ----------------- 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) > > > > # ------------------------- request routing logic ------------------- > > > > # main routing logic > > > > route{ > > > > if (!mf_process_maxfwd_header("10")) { > > sl_send_reply("483","Too Many Hops"); > > break; > > }; > > > > if (msg:len >= 2048 ) { > > sl_send_reply("513", "Message too big"); > > break; > > }; > > > > > > if (!method=="REGISTER") record_route(); > > > > if (loose_route()) { > > append_hf("P-hint: rr-enforcedrn"); > > route(1); > > break; > > }; > > > > if (!uri==myself) { > > append_hf("P-hint: outboundrn"); > > route(2); > > break; > > }; > > > > if (uri==myself) { > > > > if (method=="REGISTER") { > > if (!www_authorize("192.168.0.120", "subscriber")) { > > www_challenge("192.168.0.120", "0"); > > break; > > }; > > save("location"); > > break; > > }; > > > > lookup("aliases"); > > if (!uri==myself) { > > append_hf("P-hint: outbound aliasrn"); > > route(2); > > break; > > }; > > > > t_on_failure("1"); > > lookup("location"); > > }; > > append_hf("P-hint: usrloc appliedrn"); > > route(1); > > } > > > > route[1] > > { > > if (uri=~"^sip:888@") { > > if (!t_relay_to( "192.168.0.120", "5070")) sl_reply_error(); > > break; > > }; > > t_on_failure("1"); > > t_relay(); > > } > > > > > > route[2] > > { > > if (!t_relay()) { > > sl_reply_error(); > > }; > > } > > > > > > failure_route[1] { > > if (!t_relay_to( "192.168.0.120", "5070")) > > { > > sl_reply_error(); > > }; > > } > > > > Thank you for your help! > > Michel > > > > ________________________________________________ > > Message sent using UebiMiau 2.7.8 > > > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > ________________________________________________ Message sent using UebiMiau 2.7.8

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[Serusers] problems with t_relay_to
by Zenone 03 Jul '06

03 Jul '06

Hi, I have problems to make my ser work since two 't_relay_to' have been added in 'ser.cfg'. Here is my ser.cfg: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- 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/local/ser_directory/lib/ser/modules/mysql.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/sl.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/tm.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/rr.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/usrloc.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/registrar.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/ser_directory/lib/ser/modules/auth.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/auth_db.so" modparam("tm", "fr_inv_timer", 90) # ----------------- 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) # ------------------------- request routing logic ------------------- # main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(2); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("192.168.0.120", "subscriber")) { www_challenge("192.168.0.120", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(2); break; }; t_on_failure("1"); lookup("location"); }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { if (uri=~"^sip:888@") { if (!t_relay_to( "192.168.0.120", "5070")) sl_reply_error(); break; }; t_on_failure("1"); t_relay(); } route[2] { if (!t_relay()) { sl_reply_error(); }; } failure_route[1] { if (!t_relay_to( "192.168.0.120", "5070")) { sl_reply_error(); }; } Thank you for your help! Michel ________________________________________________ Message sent using UebiMiau 2.7.8

2 1

[Serusers] set_timer called on a "detached" timer
by Nhadie 03 Jul '06

03 Jul '06

Hi All I'm seeing this error on my syslog set_timer called on a "detached" timer I trried to monitor when it occurs, it occurs when there are customers calling and I make a query on mysql, the query doesn't really put that muchload on the server, it returns it in 2 secs i just don't know why it's causing the detached time error. TIA Regards Nhadie ________________________________________________ Message sent using UebiMiau 2.7

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[Users] Hung up the callee after 32 sec
by raviprakash sunkara 03 Jul '06

03 Jul '06

hi User May the Problem in Router where the server located... OR UA client side Router ... Block the ACK session ... hung up automatically after 32 sec.... Or problem in Softphone setting ( x-lite 3.0 ) UAC 1 and 2 are witth 192.168.2.20. and 21 which are behind the NAT with public.... yy.y.y..y ( softphone) UAC 3 is softphne which is located in server network with ip 192.168.2.7 have public ip as xxx.xxx.x.xx ( SIP DOMAIN NAME)... CALL from UAC 1 to UAC 2 hungup the phone after 32 sec when it lefted . UAC 2 is hung upping and UAC1 is call establishing the Session.... CALL from UAC 2 to UAC 3 same as above.... CALL from UAC 3 to UAC 2 not hungup . this is mines openser.cfg from route main ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++===== _________________________________________________________________________ route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section and Acc Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.168.2.75:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); setflag(1); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); return; }; if (uri==myself) { if (method=="CANCEL") { route(3); return; } else if (method=="INVITE") { route(4); #route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; if (method=="CANCEL") { route(1); return; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); } #-------------------------------------------------------------------------------------------- # Handling Trusted Sources #-------------------------------------------------------------------------------------------- route[4] { if(allow_trusted()) { rewritehost("192.168.XX.XX"); #Our IP address t_relay_to(192.168.XX.XX:5060); #relaying trusted calls to its destination return; }; else { route(3); }; } #--------------------------------------------------------------------------------------------- # Reply Route #--------------------------------------------------------------------------------------------- onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } ----------------------------------------------------------------------------------------------------------------------------------------------------- -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology Kondapur, Hi-tech city, Hyderabad. www.hyperion-tech.com +91-9985077535

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Re: [Serusers] still battling ser
by Greger V. Teigre 03 Jul '06

03 Jul '06

Hi Nick, Well, honestly, I don't think many of the developers are on RedHat. They mostly use debian. The packaging for various distros are something that need to have a sponsor (i.e. somebody who actually create the packages, not necessary a developer). I'm not aware of a RedHat sponser, but I may be wrong. In your case however, you may need some extra help and I think Andrei is the person most likely to be of help (as this is a fairly specific problem). Expecting him to spend time on serusers to find postings like yours, is a bit of a strech, that's why I suggested a post to serdev. Sure, Andrei subscribes to serusers, but it's something about response time etc ;-) If you in addition would be interested in maintaining packages for future versions, that would be great! g-) PS! Yes, you can open a bug in the bugtracker: http://bugs.sip-router.org/ nick wrote: > Greger V. Teigre wrote: >> Nick, >> You may want to post this to serdev(a)iptel.org so you know developers >> see your trace. >> g-) >> > Hey, I don't want to bother anyone, but is there something else needed > on my part, do I need to open a bug in the bug tracker, or provide any > other information to > the devs?? > > It's been nearly two days, and nothing is in the bug tracker, and I > haven't seen much in the way of acknowledgement (not that I expect an > immediate solution) I just wonder if I need to do anything else... > > > Nick > > >

2 2

[Serusers] I got error message : [tls/tls_init.o] Error 1....Please help me..please
by Ferianto siregar 03 Jul '06

03 Jul '06

Dear all, When i try to compile again TLS by using Command ,TLS=1 make install, I got an error message. It is the error message : make: *** [tls/tls_init.o] Error 1 What should i do ?I am confused because this is the first I compile TLS in Openser. Please help me.. Note: Below is the message shown when i compile TLS support gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double - falign-loops -mcpu=athlon -DNAME='"openser"' -DVERSION='"1.0.1-tls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DOPENSSL_NO_KRB5 -DUSE_TLS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/ssl/include -c tls/tls_init.c -o tls/tls_init.o tls/tls_init.c:373:25: directives may not be used inside a macro argument tls/tls_init.c:372:56: unterminated argument list invoking macro "SSL_CTX_set_options" tls/tls_init.c: In function `init_ssl_ctx_behavior': tls/tls_init.c:376: `SSL_CTX_set_options' undeclared (first use in this function) tls/tls_init.c:376: (Each undeclared identifier is reported only once tls/tls_init.c:376: for each function it appears in.) tls/tls_init.c:376: parse error before ')' token make: *** [tls/tls_init.o] Error 1 Note: For The O/S I use Redhat 9 and for openser packet, I use Openser-1.0.1-tls. Thank you very much Regards, Ferianto --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

2 1

RE: [Users] Help in pstn.cfg
by Abdul Lateef 03 Jul '06

03 Jul '06

Hello, I did configure for PSTN and it is working well. Again i tried to use MySQL for user authuntication and i followed this tutorial http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+Op… But i am getting some error on REGISTER method: if (method=="REGISTER") { if (!www_authorize("localhost", "subscriber")) { www_challenge("localhost", "0"); return; }; save("location"); return; }; ERROR:: Jul 1 14:59:53 main openser[25713]: str2val: Error while converting integer value from string Jul 1 14:59:53 main openser[25713]: convert_row: Error while converting value Jul 1 14:59:53 main openser[25713]: convert_rows: Error while converting row #0 Jul 1 14:59:53 main openser[25713]: convert_result: Error while converting rows Jul 1 14:59:53 main openser[25713]: store_result: Error while converting result Thank You abdul __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 2

[Users] Error in Compiling TLS..please help...
by Ferianto siregar 03 Jul '06

03 Jul '06

Dear all, When i try to compile again TLS by using Command ,TLS=1 make install, I got an error message. It is the error message: make: *** [tls/tls_init.o] Error 1 What should i do else? I am confused because this is the first I compile it. I do hope anybody can give me any sugestion. Please help me.. Note: Below is the message shown when i compile TLS support gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double - falign-loops -mcpu=athlon -DNAME='"openser"' -DVERSION='"1.0.1-tls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DOPENSSL_NO_KRB5 -DUSE_TLS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/ssl/include -c tls/tls_init.c -o tls/tls_init.o tls/tls_init.c:373:25: directives may not be used inside a macro argument tls/tls_init.c:372:56: unterminated argument list invoking macro "SSL_CTX_set_options" tls/tls_init.c: In function `init_ssl_ctx_behavior': tls/tls_init.c:376: `SSL_CTX_set_options' undeclared (first use in this function) tls/tls_init.c:376: (Each undeclared identifier is reported only once tls/tls_init.c:376: for each function it appears in.) tls/tls_init.c:376: parse error before ')' token make: *** [tls/tls_init.o] Error 1 NB: For O/S i use Redhat 9 and for openser packet i use openser-1.0.1-tls.tar.gz Thank you very much Regards, Ferianto --------------------------------- Want to be your own boss? Learn how on Yahoo! Small Business.

2 1

[Serusers] CDRTool + SER/OpenSER
by Adrian Georgescu 02 Jul '06

02 Jul '06

Hi Daniel, You may use your own developed prepaid application in Asterisk / SER / OpenSERand obtain the maximum session time from CDRtool rating engine. The communication protocol is very simple, you have to call via a TCP socket in your b2bua the following function on starting the call: MaxSessionTime From=sip:123@example.com To=sip:0031650222@example.com Duration=7200 Lock=1\n and DebitBalance From=sip:123@example.com To=sip:0031650222 Duration=59 after the call has ended. You can see the available commands in the rating engine by connecting to it and issuing the help command: The modus operandi is described at http://cdrtool.ag-projects.com/ PREPAID.txt If you implement such prepaid application you can check how it may integrated with CDRTool at http://cdrtool.ag-projects.com/PrepaidEngine.pdf The siptrace module has been developed and tested only on OpenSER. The siptrace module is available now in CVS head and will be soon available in the next stable release of OpenSER. Adrian >>>>>>>> I'm evaluating the option of installing SER or OpenSER in my Asterisk environment along with FreeRadius and CDRTool. All this for both prepaid and postpaid applications. While reading the CDRTool README.txt file, I came across this text: "CDRTool makes rating functions available via the network so that you can use CDRTool rating engine for postpaid application or in combination with third-party B2BUAs for prepaid applications. The functions are available via TCP/IP network socket or as SOAP/XML webservice. A Call control module in OpenSER can perform the prepaid function based on money credit and rating per destination." and "CDRTool allows real-time web access to Call Details Records like the Radius tickets generated by SIP Express Router. The information can be combined with media information from MediaProxy while a separate OpenSER module can display full traces of all SIP packets the enter and exit the SIP Proxy. CDRs can be filtered and statistics can be built in real-time to summarize service usage based on various criteria." Does this mean that these features are ONLY available to OpenSER and not both SER and OpenSER? Also, for prepaid application I was thinking of using the Asterisk B2BUA tool along with other features like Codec translations. Can anyone comment on this? Thanks, Daniel

1 0

[Serusers] CDRTool + SER/OpenSER
by Daniel Salama 02 Jul '06

02 Jul '06

I'm evaluating the option of installing SER or OpenSER in my Asterisk environment along with FreeRadius and CDRTool. All this for both prepaid and postpaid applications. While reading the CDRTool README.txt file, I came across this text: "CDRTool makes rating functions available via the network so that you can use CDRTool rating engine for postpaid application or in combination with third-party B2BUAs for prepaid applications. The functions are available via TCP/IP network socket or as SOAP/XML webservice. A Call control module in OpenSER can perform the prepaid function based on money credit and rating per destination." and "CDRTool allows real-time web access to Call Details Records like the Radius tickets generated by SIP Express Router. The information can be combined with media information from MediaProxy while a separate OpenSER module can display full traces of all SIP packets the enter and exit the SIP Proxy. CDRs can be filtered and statistics can be built in real-time to summarize service usage based on various criteria." Does this mean that these features are ONLY available to OpenSER and not both SER and OpenSER? Also, for prepaid application I was thinking of using the Asterisk B2BUA tool along with other features like Codec translations. Can anyone comment on this? Thanks, Daniel

1 0

RE: [Users] Ok, Bye..
by Kelvin Williams 01 Jul '06

01 Jul '06

Here are some logs that I noticed when making a call. Don't know if they shed any light. Jul 1 17:13:56 sp /usr/local/sbin/openser[26727]: ERROR:parse_from_header: bad from header Jul 1 17:13:56 sp /usr/local/sbin/openser[26727]: ERROR:uac:restore_from_reply: failed to find/parse FROM hdr Jul 1 17:14:02 sp /usr/local/sbin/openser[26729]: ERROR:parse_from_header: bad from header Jul 1 17:14:02 sp /usr/local/sbin/openser[26729]: ERROR:uac:restore_from_reply: failed to find/parse FROM hdr Jul 1 17:14:05 sp /usr/local/sbin/openser[26725]: ERROR:parse_from_header: bad from header Jul 1 17:14:05 sp /usr/local/sbin/openser[26725]: ERROR:uac:restore_from_reply: failed to find/parse FROM hdr Jul 1 17:14:06 sp /usr/local/sbin/openser[26729]: ERROR:parse_from_header: bad from header Jul 1 17:14:06 sp /usr/local/sbin/openser[26729]: ERROR:uac:restore_from_reply: failed to find/parse FROM hdr Jul 1 17:14:06 sp /usr/local/sbin/openser[26725]: ERROR:parse_from_header: bad from header Jul 1 17:14:06 sp /usr/local/sbin/openser[26725]: ERROR:uac:restore_from_reply: failed to find/parse FROM hdr Jul 1 17:14:07 sp /usr/local/sbin/openser[26729]: ERROR:parse_from_header: bad from header Jul 1 17:14:07 sp /usr/local/sbin/openser[26729]: ERROR:uac:restore_from_reply: failed to find/parse FROM hdr Jul 1 17:14:08 sp /usr/local/sbin/openser[26725]: ERROR:parse_from_header: bad from header Jul 1 17:14:08 sp /usr/local/sbin/openser[26725]: ERROR:uac:restore_from_reply: failed to find/parse FROM hdr Jul 1 17:14:09 sp /usr/local/sbin/openser[26729]: ERROR:parse_from_header: bad from header Jul 1 17:14:09 sp /usr/local/sbin/openser[26729]: ERROR:uac:restore_from_reply: failed to find/parse FROM hdr Jul 1 17:14:10 sp /usr/local/sbin/openser[26725]: ERROR:parse_from_header: bad from header Jul 1 17:14:10 sp /usr/local/sbin/openser[26725]: ERROR:uac:restore_from_reply: failed to find/parse FROM hdr Jul 1 17:14:11 sp /usr/local/sbin/openser[26729]: ERROR: get_hdr_field: bad to header Jul 1 17:14:11 sp /usr/local/sbin/openser[26729]: ERROR: bad header field Jul 1 17:14:11 sp /usr/local/sbin/openser[26729]: ERROR:maxfwd:is_maxfwd_present : parsing MAX_FORWARD header failed! _____ From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Kelvin Williams Sent: Saturday, July 01, 2006 4:38 PM To: users(a)openser.org Subject: [Users] Ok, Bye.. I am encountering a problem when creating an outbound call using our PSTN Provider. I make the call, the phone rings (PSTN) and I can hear the other party answer. They however, cannot hear me, and shortly thereafter the call drops. I ngrep'd the call and found that they are sending me 200 OK's after the call is set up, and OpenSER appears to relay those back to my EMTA. However, I never see the ACK return from my EMTA. What could cause this problem? Inbound calls are not a problem. Thanks, Kw

1 0

[Users] Can't route calls to a PSTN gateway
by Hamid Ali Asgari 01 Jul '06

01 Jul '06

Hi, I am trying to route some SIP calls to my gateway but I get the following error: 0(17329) lookup(): '0033146' Not found in usrloc Bellow is my config, any thoughts? Thanks, Hamid # ----------- global configuration parameters ------------------------ debug=8 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) #/* 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/openser_fifo" listen = udp:10.20.20.18:5060 alias="mytestdomain.com" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc/ope nser/tls/user/user-cert.pem" #tls_private_key = "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc/ope nser/tls/user/user-privkey.pem" #tls_ca_list = "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc/ope nser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/openser/modules/mysql.so" #loadmodule "/usr/lib/openser/modules/sl.so" #loadmodule "/usr/lib/openser/modules/tm.so" #loadmodule "/usr/lib/openser/modules/rr.so" #loadmodule "/usr/lib/openser/modules/maxfwd.so" #loadmodule "/usr/lib/openser/modules/usrloc.so" #loadmodule "/usr/lib/openser/modules/registrar.so" #loadmodule "/usr/lib/openser/modules/textops.so" #loadmodule "/usr/lib/openser/modules/uri.so" #loadmodule "/usr/lib/openser/modules/auth.so" #loadmodule "/usr/lib/openser/modules/auth_radius.so" #loadmodule "/usr/lib/openser/modules/group_radius.so" #loadmodule "/usr/lib/openser/modules/avp_radius.so" #loadmodule "/usr/lib/openser/modules/acc.so" #loadmodule "/usr/lib/openser/modules/avpops.so" mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "avpops.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" #loadmodule "xlog.so" loadmodule "uri.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://root:hashtaloo.@localhost/openser") modparam("rr", "enable_full_lr", 1) modparam("auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("group_radius", "use_domain", 1) modparam("avpops", "avp_aliases", "day=i:101;time=i:102") # ------------------------- 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); } route(1); }; if(is_method("INVITE") && !has_totag()) { # set the acc flags setflag(1); setflag(2); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # 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("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; if (!radius_www_authorize("mytestdomain.com")) { www_challenge("mytestdomain.com", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); sl_send_reply("404", "Not Found"); exit; }; # calls to pstn if(uri=~"sip:00[1-9][0-9]+@") { if(is_method("INVITE") && !has_totag()) { if (!radius_is_user_in("From", "voip")) { sl_send_reply("403", "Forbidden PSTN"); exit; }; }; # Setting the gateway address rewritehostport("10.20.20.19:5060"); route(1); }; # load callee's avps if(avp_load_radius("callee")) { # check if user has time filter enabled if(avp_check("i:3", "eq/i:1")) { # print time in an avp avp_printf("i:100", "$Tf"); # extract day avp_subst("i:100/i:101", "/(.{3}) .+/*\1*/"); if(!avp_check("i:6", "fm/$day")) { sl_send_reply("403", "Forbidden - day"); exit; }; # extract 'hours:minutes' avp_subst("i:100/i:102", "/(.{10}) (.{5}):.+/\2/"); if((is_avp_set("i:4") && avp_check("i:4", "gt/$time")) || (is_avp_set("i:5") && avp_check("i:5", "lt/$time"))) { sl_send_reply("403", "Forbidden - time"); exit; }; }; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

2 1

Re: [Users] Ok, Bye..
by Daniel-Constantin Mierla 01 Jul '06

01 Jul '06

Hello, On 07/01/06 23:38, Kelvin Williams wrote: > > I am encountering a problem when creating an outbound call using our > PSTN Provider. > > I make the call, the phone rings (PSTN) and I can hear the other party > answer. They however, cannot hear me, and shortly thereafter the call > drops. I ngrep’d the call and found that they are sending me 200 OK’s > after the call is set up, and OpenSER appears to relay those back to > my EMTA. However, I never see the ACK return from my EMTA. What could > cause this problem? > your phone must send an ACK afte rreceiving the 200OK, otherwise the callee will tear down the call after aprox. 20sec. No audio problem could be if you are behind a NAT and you haven't configured the sip server to do the NAT traversal (via nathelper+rtpproxy or mediaproxy+python mediaproxy). Cheers, Daniel > Inbound calls are not a problem. > > Thanks, > > Kw > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[Users] Ok, Bye..
by Kelvin Williams 01 Jul '06

01 Jul '06

I am encountering a problem when creating an outbound call using our PSTN Provider. I make the call, the phone rings (PSTN) and I can hear the other party answer. They however, cannot hear me, and shortly thereafter the call drops. I ngrep'd the call and found that they are sending me 200 OK's after the call is set up, and OpenSER appears to relay those back to my EMTA. However, I never see the ACK return from my EMTA. What could cause this problem? Inbound calls are not a problem. Thanks, Kw

1 0

Re: [Serusers] How do I get rid of npdi=yes?
by Jiri Kuthan 01 Jul '06

01 Jul '06

I don't think that the problem in URI is presence of npdi. The problem is IMO that it is a SIP URDI without a domain -- I don't really see to where I would be able to send such a URI. -jiri At 19:53 23/06/2006, John Clements wrote: >I've looked around and have found one or two other posts asking about >this, but I have never seen an answer. The From is: > >=uri: <sip:5551212;npdi=yes@x.x.x.x:5060;dtg=SIP;user=phone> > >Which causes a number of issues. USRLOC no longer works: > >=lookup(): '5551212;npdi=yes' Not found in usrloc > >And when you send the call to an Asterisk server, Asterisk wigs out >because it truncates everything after the first ";" it encounters in the >URI. Does anyone have any idea how to get rid of it? I've tried subst(), >but could not get it to work. :-( > > > -John > > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

3 4

[Serusers] core dump....
by olivier.taylor 01 Jul '06

01 Jul '06

2 2

[Serusers] Logging weirdness with Aliases.
by sip 01 Jul '06

01 Jul '06

Okay... help me visualise something. Now, as far as I can tell, SER is doing exactly what it's supposed to be doing, but I can't figure out how to get it to do something different. I have an accounting section (my log flag is 1) that looks like this: if((method=="INVITE" && !has_totag()) || method=="BYE") { setflag(1); }; Basic, simple logging section. But when someone calls through from a DID to an alias in the aliases table, it of course logs the sip_to as the alias. Fine, I thought. I'll just add a lookup("aliases") before the setflag and all will be well, right? No. Then it doesn't seem to log anything at ALL. So what I'm trying to figure out is how to get SER to log a call coming into an alias as a call to the user who's aliased. I.e. a call is coming in from 1010(a)remote.proxy.com to 9999(a)proxy.com (which is really an alias to user 1234(a)proxy.com). Right now, SER is logging in the acc tables that the call went to sip:9999@proxy.com. I want to have it log that the call went to 1234(a)proxy.com and NOT that it went to the alias. This: if((method=="INVITE" && !has_totag()) || method=="BYE") { lookup("aliases"); setflag(1); }; gets me nothing at all. Ideas, anyone? I'm stumped. I'm sure it's something drastically simple that I'm just not getting. N.

2 5

[Serusers] I got error message : [tls/tls_init.o] Error 1....Please help me..please
by Ferianto siregar 01 Jul '06

01 Jul '06

Dear all, When i try to compile again TLS ( I have disabled Kerberos5) by using Command ,TLS=1 make install, I got an error message. It is : make: *** [tls/tls_init.o] Error 1 What should i do else? I am very confused. This is my first time to build TLS. Please help me..Please.. Note: Below is the message shown when i compile TLS support gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double - falign-loops -mcpu=athlon -DNAME='"openser"' -DVERSIonfiltered='"1.0.1-tls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DOPENSSL_NO_KRB5 -DUSE_TLS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/ssl/include -c tls/tls_init.c -o tls/tls_init.o tls/tls_init.c:373:25: directives may not be used inside a macro argument tls/tls_init.c:372:56: unterminated argument list invoking macro "SSL_CTX_set_options" tls/tls_init.c: In function `init_ssl_ctx_behavior': tls/tls_init.c:376: `SSL_CTX_set_options' undeclared (first use in this function) tls/tls_init.c:376: (Each undeclared identifier is reported only once tls/tls_init.c:376: for each function it appears in.) tls/tls_init.c:376: parse error before ')' token make: *** [tls/tls_init.o] Error 1 Thank you very much Regards, Ferianto --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs.Try it free.

1 0

[Users] I got error message : [tls/tls_init.o] Error 1....Please help me..please
by Ferianto siregar 01 Jul '06

01 Jul '06

Dear all, Thank you very much to Bogdan for helping me.Thanks When i try to compile again TLS (after I disable the Kerberos5) by using Command ,TLS=1 make install, I got an error message again It is the message: make: *** [tls/tls_init.o] Error 1 What should i do else? I am very confused. This is my first time to build TLS. Please help me..Please.. Note: Below is the message shown when i compile TLS support gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double - falign-loops -mcpu=athlon -DNAME='"openser"' -DVERSION='"1.0.1-tls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DOPENSSL_NO_KRB5 -DUSE_TLS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/ssl/include -c tls/tls_init.c -o tls/tls_init.o tls/tls_init.c:373:25: directives may not be used inside a macro argument tls/tls_init.c:372:56: unterminated argument list invoking macro "SSL_CTX_set_options" tls/tls_init.c: In function `init_ssl_ctx_behavior': tls/tls_init.c:376: `SSL_CTX_set_options' undeclared (first use in this function) tls/tls_init.c:376: (Each undeclared identifier is reported only once tls/tls_init.c:376: for each function it appears in.) tls/tls_init.c:376: parse error before ')' token make: *** [tls/tls_init.o] Error 1 Thank you very much. Regards, Ferianto --------------------------------- Want to be your own boss? Learn how on Yahoo! Small Business.

1 0

[Serusers] SER statistics
by Adrian A 30 Jun '06

30 Jun '06

Hi, I'm looking for a way to extract statistics from SER such as registrations or invites per second. I was thinking I could either log all of these to a file using xlog and then parse it, but it's not going to be pretty. Does anyone have any suggestions? Thanks.

2 1

[Users] DIsplay in INVITE non-US ascii?
by Greg Fausak 30 Jun '06

30 Jun '06

Hey Guys, Can somebody post an example of an INVITE with non-US characters in the From tag? What character set should be use for non-US callerid? -g -- Greg Fausak greg(a)thursday.com

2 1

[Users] Radius accounting on debian
by Jeff Williams 30 Jun '06

30 Jun '06

I have openser-tls and openser-tls-radius-modules install on a debian sarge box, but the radius options for the acc module seem not to work. galileo:~# /etc/init.d/openser restart There are errors in the configuration file. Please fix them first. Also make sure you have installed the debian packages that provide the additional modules that your configuration requires. cpl, jabber, mysql, postgres and radius are in separate packages. 0(14929) set_mod_param_regex: parameter <radius_config> not found in module <acc> 0(14929) parse error (34,20-21): Can't set module parameter 0(14929) set_mod_param_regex: parameter <radius_flag> not found in module <acc> 0(14929) parse error (35,19-20): Can't set module parameter 0(14929) set_mod_param_regex: parameter <radius_missed_flag> not found in module <acc> 0(14929) parse error (36,19-20): Can't set module parameter ERROR: bad config file (3 errors) Is radius support for the acc module not included in these packages? Do I need to grap the source? Jeff

3 4

[Users] Cisco radius auth
by Jeff Williams 30 Jun '06

30 Jun '06

Is there any way to get openser to do radius auth using attribute from the Cisco VSA Voice Implementation Guide <http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_programming…>? i.e. something like: Access-Request NAS-IP-Address = 203.56.98.120 NAS-Port-Type = Async User-Name = "62186234" cisco-h323-conf-id = "h323-conf-id=89437006 E20011DA B677BBCA D2EBCDD5" Called-Station-Id = "0011553334112401" Calling-Station-Id = "0295731743" User-Password = "<145>;<22><242>$<251><243>)u<183>h<225>y<173><150><193>" I would like to get openser to authenticate against an existing billing system if possible. Jeff

2 2

[Users] monitoring ser
by Giuseppe Parlato 30 Jun '06

30 Jun '06

Hi all, Have you ever tried to monitor ser using mrtg or what's up gold ? Is there any way to monitor this service using snmp? thanks Giuseppe

2 1

[Users] openser is using local ip for record route
by Kumar Subramanian 30 Jun '06

30 Jun '06

I am running openser behind the firewall. and advertised_address is set correctly to the external ip address. but record route seem to send the local/internal ip address. help is appreciated. Thx

2 1

[Users] Help in pstn.cfg
by Abdul Lateef 30 Jun '06

30 Jun '06

Hi guys, I am trying to route all calls from openser to our commercial SIP Proxy server. I modify PSTN.cfg file from examples folder and i tried to call from one sip dialer but there is no any request from openser to my SIP Proxy server. and the dialer is registered and trying to call without any error. I was try to look openser logs but there is not error or warrning logs. Same time i did capture from Ethereal and found "Time Out" from openser to sip dialer. Here is my pstn.cfg configuration:: loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" # -- acc params -- modparam("acc", "log_level", 1) # that is the flag for which we will account -- don't forget to # set the same one :-) modparam("acc", "log_flag", 1 ) # ------------------------- request routing logic ------------------- # main routing logic route{ if (method=="REGISTER") { save("location"); return; }; if (loose_route()) { t_relay(); return; }; if (method=="INVITE") { record_route(); }; if (!uri=~"sip:\+?[0-9]+@.*") { if (method=="INVITE") { sl_send_reply("403", "Call cannot be served here"); } else { # forward(uri:host, uri:port); }; return; }; #end of (!uri=~"sip:\+?[0-9]+@.*") setflag(1); if (method=="INVITE") { if (!uri=~"sip:00[1-9][0-9]+@.*") { sl_send_reply("403", "Forbidden"); return; }; #End if (uri=~"sip:00[1-9][0-9]+@.*") }; #End of (method=="INVITE") rewritehostport("192.168.0.10:5060"); # forward the request now if (!t_relay()) { sl_reply_error(); return; }; } #end of route -------- Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: abdulzu(a)hotmail.com GoogleTalk: lateef.np(a)gmail.com YM!: abdul_zu Doha Qatar http://www.hatif.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] xten to sipura compatbility issue
by Raymond Chen 30 Jun '06

30 Jun '06

Dear all, I have set up both xten and sipura, I want to test onnet calls between them. I am fine with xten eyebeam to xten eyebeam, and sipura to sipura, both scenario works fine. when I do xten to sipura 2000, it rings and when pick up, xten sends BYE right after ACK. I've tried 1.0.1 verson and the CVS HEAD, happens to both version. did anyone encountered the same problem? thanks Ray ps. debug attached -- U 61.144.140.110:42075 -> 210.184.138.8:5066 INVITE sip:02061241205@210.184.138.8:5066 SIP/2.0. To: <sip:02061241205@210.184.138.8>. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Via: SIP/2.0/UDP 192.168.1.168:9801;branch=z9hG4bK-d87543-443309010-1--d87543-;rport. Call-ID: c47ef44f950e7a56. CSeq: 1 INVITE. Contact: <sip:3068878@192.168.1.168:9801>. Max-Forwards: 70. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 3004t stamp 16741. Content-Length: 234. . v=0. o=- 21525734 21525766 IN IP4 192.168.1.168. s=eyeBeam. c=IN IP4 192.168.1.168. t=0 0. m=audio 9802 RTP/AVP 18 101. a=alt:1 1 : 8743E761 095E081B 192.168.1.168 9802. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. # U 210.184.138.8:5066 -> 61.144.140.110:42075 SIP/2.0 100 trying -- your call is important to us. To: <sip:02061241205@210.184.138.8>. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Via: SIP/2.0/UDP 192.168.1.168:9801;branch=z9hG4bK-d87543-443309010-1--d87543-;rport=42075;received=61.144.140.110. Call-ID: c47ef44f950e7a56. CSeq: 1 INVITE. Server: OpenSer (1.1.0-pre2-notls (i386/linux)). Content-Length: 0. . # U 210.184.138.8:5066 -> 219.136.10.86:64189 INVITE sip:862061241205@219.136.10.86:64189 SIP/2.0. Record-Route: <sip:210.184.138.8:5066;lr=on;ftag=6a33d80f>. To: <sip:02061241205@210.184.138.8>. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Via: SIP/2.0/UDP 210.184.138.8:5066;branch=z9hG4bK72b9.90fc56c.0. Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-443309010-1--d87543-;rport=42075. Call-ID: c47ef44f950e7a56. CSeq: 1 INVITE. Contact: <sip:3068878@61.144.140.110:42075>. Max-Forwards: 69. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 3004t stamp 16741. Content-Length: 273. P-hint: Proxied. P-hint: usrloc applied. . v=0. o=- 21525734 21525766 IN IP4 192.168.1.168. s=eyeBeam. c=IN IP4 210.184.138.8. t=0 0. m=audio 35130 RTP/AVP 18 101. a=alt:1 1 : 8743E761 095E081B 192.168.1.168 9802. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=direction:active. a=nortpproxy:yes. # U 219.136.10.86:64189 -> 210.184.138.8:5066 SIP/2.0 100 Trying. To: <sip:02061241205@210.184.138.8>. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Call-ID: c47ef44f950e7a56. CSeq: 1 INVITE. Via: SIP/2.0/UDP 210.184.138.8:5066;branch=z9hG4bK72b9.90fc56c.0. Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-443309010-1--d87543-;rport=42075. Record-Route: <sip:210.184.138.8:5066;lr=on;ftag=6a33d80f>. Server: Sipura/SPA2000-2.0.13(g). Content-Length: 0. . # U 219.136.10.86:64189 -> 210.184.138.8:5066 SIP/2.0 180 Ringing. To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Call-ID: c47ef44f950e7a56. CSeq: 1 INVITE. Via: SIP/2.0/UDP 210.184.138.8:5066;branch=z9hG4bK72b9.90fc56c.0. Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-443309010-1--d87543-;rport=42075. Record-Route: <sip:210.184.138.8:5066;lr=on;ftag=6a33d80f>. Server: Sipura/SPA2000-2.0.13(g). Content-Length: 0. . # U 210.184.138.8:5066 -> 61.144.140.110:42075 SIP/2.0 180 Ringing. To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Call-ID: c47ef44f950e7a56. CSeq: 1 INVITE. Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-443309010-1--d87543-;rport=42075. Record-Route: <sip:210.184.138.8:5066;lr=on;ftag=6a33d80f>. Server: Sipura/SPA2000-2.0.13(g). Content-Length: 0. . # U 61.144.140.110:46935 -> 210.184.138.8:5066 SIP/2.0 404 Not Found. To: sip:61.144.140.110:46935;tag=ab59191034906d25i0. From: sip:keepitopen@broadz.net;tag=75ca79a5. Call-ID: c60b3302-3e11e12-c8(a)210.184.138.8. CSeq: 1 OPTIONS. Via: SIP/2.0/UDP 210.184.138.8:5066;branch=0. Server: Sipura/SPA2002-3.1.2(a). Content-Length: 0. . # U 219.136.10.86:64189 -> 210.184.138.8:5066 SIP/2.0 180 Ringing. To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Call-ID: c47ef44f950e7a56. CSeq: 1 INVITE. Via: SIP/2.0/UDP 210.184.138.8:5066;branch=z9hG4bK72b9.90fc56c.0. Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-443309010-1--d87543-;rport=42075. Record-Route: <sip:210.184.138.8:5066;lr=on;ftag=6a33d80f>. Server: Sipura/SPA2000-2.0.13(g). Content-Length: 0. . # U 210.184.138.8:5066 -> 61.144.140.110:42075 SIP/2.0 180 Ringing. To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Call-ID: c47ef44f950e7a56. CSeq: 1 INVITE. Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-443309010-1--d87543-;rport=42075. Record-Route: <sip:210.184.138.8:5066;lr=on;ftag=6a33d80f>. Server: Sipura/SPA2000-2.0.13(g). Content-Length: 0. . # U 219.136.10.86:64189 -> 210.184.138.8:5066 SIP/2.0 200 OK. To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Call-ID: c47ef44f950e7a56. CSeq: 1 INVITE. Via: SIP/2.0/UDP 210.184.138.8:5066;branch=z9hG4bK72b9.90fc56c.0. Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-443309010-1--d87543-;rport=42075. Record-Route: <sip:210.184.138.8:5066;lr=on;ftag=6a33d80f>. Contact: 862061241205 <sip:862061241205@192.168.10.244:8177>. Server: Sipura/SPA2000-2.0.13(g). Content-Length: 238. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 12432 12432 IN IP4 192.168.10.244. s=-. c=IN IP4 192.168.10.244. t=0 0. m=audio 43712 RTP/AVP 18 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 210.184.138.8:5066 -> 61.144.140.110:42075 SIP/2.0 200 OK. To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Call-ID: c47ef44f950e7a56. CSeq: 1 INVITE. Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-443309010-1--d87543-;rport=42075. Record-Route: <sip:210.184.138.8:5066;lr=on;ftag=6a33d80f>. Contact: 862061241205 <sip:862061241205@219.136.10.86:64189>. Server: Sipura/SPA2000-2.0.13(g). Content-Length: 255. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. P-hint: Reply-Proxied. . v=0. o=- 12432 12432 IN IP4 192.168.10.244. s=-. c=IN IP4 210.184.138.8. t=0 0. m=audio 35132 RTP/AVP 18 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. a=nortpproxy:yes. # U 61.144.140.110:42075 -> 210.184.138.8:5066 ACK sip:862061241205@219.136.10.86:64189 SIP/2.0. To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Via: SIP/2.0/UDP 192.168.1.168:9801;branch=z9hG4bK-d87543-708924014-1--d87543-;rport. Call-ID: c47ef44f950e7a56. CSeq: 1 ACK. Route: <sip:210.184.138.8:5066;lr;ftag=6a33d80f>. Contact: <sip:3068878@192.168.1.168:9801>. Max-Forwards: 70. User-Agent: eyeBeam release 3004t stamp 16741. Content-Length: 0. . # U 210.184.138.8:5066 -> 219.136.10.86:64189 ACK sip:862061241205@219.136.10.86:64189 SIP/2.0. To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Via: SIP/2.0/UDP 210.184.138.8:5066;branch=z9hG4bK72b9.90fc56c.2. Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-708924014-1--d87543-;rport=42075. Call-ID: c47ef44f950e7a56. CSeq: 1 ACK. Contact: <sip:3068878@61.144.140.110:42075>. Max-Forwards: 69. User-Agent: eyeBeam release 3004t stamp 16741. Content-Length: 0. P-hint: Proxied. P-hint: rr-enforced. . # U 61.144.140.110:42075 -> 210.184.138.8:5066 BYE sip:862061241205@219.136.10.86:64189 SIP/2.0. To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Via: SIP/2.0/UDP 192.168.1.168:9801;branch=z9hG4bK-d87543-956249433-1--d87543-;rport. Call-ID: c47ef44f950e7a56. CSeq: 2 BYE. Route: <sip:210.184.138.8:5066;lr;ftag=6a33d80f>. Contact: <sip:3068878@192.168.1.168:9801>. Max-Forwards: 70. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: eyeBeam release 3004t stamp 16741. Content-Length: 0. . # U 210.184.138.8:5066 -> 219.136.10.86:64189 BYE sip:862061241205@219.136.10.86:64189 SIP/2.0. To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Via: SIP/2.0/UDP 210.184.138.8:5066;branch=z9hG4bK42b9.d7afa485.0. Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-956249433-1--d87543-;rport=42075. Call-ID: c47ef44f950e7a56. CSeq: 2 BYE. Contact: <sip:3068878@61.144.140.110:42075>. Max-Forwards: 69. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: eyeBeam release 3004t stamp 16741. Content-Length: 0. P-hint: Proxied. P-hint: rr-enforced. . # U 219.136.10.86:64189 -> 210.184.138.8:5066 SIP/2.0 200 OK. To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Call-ID: c47ef44f950e7a56. CSeq: 2 BYE. Via: SIP/2.0/UDP 210.184.138.8:5066;branch=z9hG4bK42b9.d7afa485.0. Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-956249433-1--d87543-;rport=42075. Server: Sipura/SPA2000-2.0.13(g). Content-Length: 0. . # U 210.184.138.8:5066 -> 61.144.140.110:42075 SIP/2.0 200 OK. To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0. From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f. Call-ID: c47ef44f950e7a56. CSeq: 2 BYE. Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-956249433-1--d87543-;rport=42075. Server: Sipura/SPA2000-2.0.13(g). Content-Length: 0. 0(2815) receive_msg: cleaning up 0(2815) SIP Request: 0(2815) method: <INVITE> 0(2815) uri: <sip:02061241205@210.184.138.8:5066> 0(2815) version: <SIP/2.0> 0(2815) parse_headers: flags=2 0(2815) DEBUG:parse_to:end of header reached, state=10 0(2815) DBUG:parse_to: display={}, ruri={sip:02061241205@210.184.138.8} 0(2815) DEBUG: get_hdr_field: <To> [33]; uri=[sip:02061241205@210.184.138.8] 0(2815) DEBUG: to body [<sip:02061241205@210.184.138.8> ] 0(2815) Found param type 232, <branch> = <z9hG4bK-d87543-443309010-1--d87543->; state=6 0(2815) Found param type 235, <rport> = <n/a>; state=17 0(2815) end of header reached, state=5 0(2815) parse_headers: Via found, flags=2 0(2815) parse_headers: this is the first via 0(2815) After parse_msg... 0(2815) preparing to run routing scripts... 0(2815) parse_headers: flags=100 0(2815) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(2815) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2815) parse_headers: flags=80 0(2815) parse_headers: flags=ffffffffffffffff 0(2815) DEBUG: get_hdr_body : content_length=234 0(2815) found end of header 0(2815) DEBUG:check_content_type: type <application/sdp> found valid 0(2815) parse_headers: flags=ffffffffffffffff 0(2815) parse_headers: flags=200 0(2815) find_first_route: No Route headers found 0(2815) loose_route: There is no Route HF 0(2815) DEBUG: add_param: tag=6a33d80f 0(2815) DEBUG:parse_to:end of header reached, state=29 0(2815) DBUG:parse_to: display={3068878}, ruri={sip:3068878@210.184.138.8} 0(2815) grep_sock_info - checking if host==us: 13==13 && [210.184.138.8] == [210.184.138.8] 0(2815) grep_sock_info - checking if port 5066 matches port 5066 0(2815) DEBUG:avpops:load_avps: loaded avps = 0 0(2815) rewrite_uri: Rewriting Request-URI with 'sip:862061241205@219.136.10.86:64189' 0(2815) DEBUG:avpops:load_avps: loaded avps = 0 0(2815) parse_headers: flags=ffffffffffffffff 0(2815) parse_headers: flags=ffffffffffffffff 0(2815) DEBUG:check_content_type: type <application/sdp> found valid 0(2815) parse_headers: flags=40 0(2815) force_rtp_proxy2: proxy reply: 35130 210.184.138.8 0(2815) DEBUG: t_newtran: msg id=69 , global msg id=40 , T on entrance=0xffffffff 0(2815) parse_headers: flags=ffffffffffffffff 0(2815) parse_headers: flags=78 0(2815) t_lookup_request: start searching: hash=39719, isACK=0 0(2815) DEBUG: RFC3261 transaction matching failed 0(2815) DEBUG: t_lookup_request: no transaction found 0(2815) DEBUG:tm:t_relay: new INVITE 0(2815) parse_headers: flags=ffffffffffffffff 0(2815) check_via_address(61.144.140.110, 192.168.1.168, 1) 0(2815) check_via_address: doing dns lookup 0(2815) WARNING:vqm_resize: resize(0) called 0(2815) DEBUG:tm:_reply_light: reply sent out. buf=0x81631c0: SIP/2.0 1..., shmem=0xb60aff08: SIP/2.0 1 0(2815) DEBUG:tm:_reply_light: finished 0(2815) DEBUG: mk_proxy: doing DNS lookup... 0(2815) parse_headers: flags=2000 0(2815) parse_headers: flags=ffffffffffffffff 0(2815) clen_builder: content-length: 273 (273) 0(2815) check_via_address(61.144.140.110, 192.168.1.168, 1) 0(2815) check_via_address: doing dns lookup 0(2815) DEBUG: add_to_tail_of_timer[4]: 0xb60b048c 0(2815) DEBUG: add_to_tail_of_timer[0]: 0xb60b049c 0(2815) SER: new transaction fwd'ed 0(2815) DEBUG:tm:UNREF_UNSAFE: after is 0 0(2815) DEBUG:destroy_avp_list: destroying list (nil) 0(2815) receive_msg: cleaning up 0(2815) SIP Reply (status): 0(2815) version: <SIP/2.0> 0(2815) status: <100> 0(2815) reason: <Trying> 0(2815) parse_headers: flags=2 0(2815) DEBUG:parse_to:end of header reached, state=10 0(2815) DBUG:parse_to: display={}, ruri={sip:02061241205@210.184.138.8} 0(2815) DEBUG: get_hdr_field: <To> [33]; uri=[sip:02061241205@210.184.138.8] 0(2815) DEBUG: to body [<sip:02061241205@210.184.138.8> ] 0(2815) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(2815) Found param type 232, <branch> = <z9hG4bK72b9.90fc56c.0>; state=16 0(2815) end of header reached, state=5 0(2815) parse_headers: Via found, flags=2 0(2815) parse_headers: this is the first via 0(2815) After parse_msg... 0(2815) forward_reply: found module nathelper, passing reply to it 0(2815) forward_reply: found module tm, passing reply to it 0(2815) DEBUG: t_check: msg id=70 global id=69 T start=0xffffffff 0(2815) parse_headers: flags=22 0(2815) parse_headers: flags=8 0(2815) DEBUG: t_reply_matching: hash 39719 label 207998729 branch 0 0(2815) DEBUG:tm:REF_UNSAFE: after is 1 0(2815) DEBUG: t_reply_matching: reply matched (T=0xb60b0370)! 0(2815) DEBUG: t_check: msg id=70 global id=70 T end=0xb60b0370 0(2815) DEBUG:tm:reply_received: org. status uas=100, uac[0]=0 local=0 is_invite=1) 0(2815) parse_headers: flags=80 0(2815) Found param type 234, <received> = <61.144.140.110>; state=6 0(2815) Found param type 232, <branch> = <z9hG4bK-d87543-443309010-1--d87543->; state=6 0(2815) Found param type 235, <rport> = <42075>; state=16 0(2815) end of header reached, state=5 0(2815) parse_headers: Via found, flags=80 0(2815) parse_headers: this is the second via 0(2815) DEBUG: get_hdr_body : content_length=0 0(2815) found end of header 0(2815) DEBUG:tm:t_should_relay_response: T_code=100, new_code=100 0(2815) DEBUG:tm:relay_reply: branch=0, save=0, relay=-1 0(2815) DEBUG: add_to_tail_of_timer[1]: 0xb60b049c 0(2815) DEBUG:tm:UNREF_UNSAFE: after is 0 0(2815) DEBUG:destroy_avp_list: destroying list (nil) 0(2815) receive_msg: cleaning up 0(2815) SIP Reply (status): 0(2815) version: <SIP/2.0> 0(2815) status: <180> 0(2815) reason: <Ringing> 0(2815) parse_headers: flags=2 0(2815) DEBUG: add_param: tag=3b3dbe3871bbfa37i0 0(2815) DEBUG:parse_to:end of header reached, state=29 0(2815) DBUG:parse_to: display={}, ruri={sip:02061241205@210.184.138.8} 0(2815) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02061241205@210.184.138.8] 0(2815) DEBUG: to body [<sip:02061241205@210.184.138.8>] 0(2815) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(2815) Found param type 232, <branch> = <z9hG4bK72b9.90fc56c.0>; state=16 0(2815) end of header reached, state=5 0(2815) parse_headers: Via found, flags=2 0(2815) parse_headers: this is the first via 0(2815) After parse_msg... 0(2815) forward_reply: found module nathelper, passing reply to it 0(2815) forward_reply: found module tm, passing reply to it 0(2815) DEBUG: t_check: msg id=71 global id=70 T start=0xffffffff 0(2815) parse_headers: flags=22 0(2815) parse_headers: flags=8 0(2815) DEBUG: t_reply_matching: hash 39719 label 207998729 branch 0 0(2815) DEBUG:tm:REF_UNSAFE: after is 1 0(2815) DEBUG: t_reply_matching: reply matched (T=0xb60b0370)! 0(2815) DEBUG: t_check: msg id=71 global id=71 T end=0xb60b0370 0(2815) DEBUG:tm:reply_received: org. status uas=100, uac[0]=100 local=0 is_invite=1) 0(2815) parse_headers: flags=80 0(2815) Found param type 234, <received> = <61.144.140.110>; state=6 0(2815) Found param type 232, <branch> = <z9hG4bK-d87543-443309010-1--d87543->; state=6 0(2815) Found param type 235, <rport> = <42075>; state=16 0(2815) end of header reached, state=5 0(2815) parse_headers: Via found, flags=80 0(2815) parse_headers: this is the second via 0(2815) DEBUG: get_hdr_body : content_length=0 0(2815) found end of header 0(2815) DEBUG:tm:t_should_relay_response: T_code=100, new_code=180 0(2815) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(2815) old size: 476, new size: 410 0(2815) build_res_from_sip_res: copied size: orig:245, new: 179, rest: 231 msg= SIP/2.0 180 Ringing To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0 From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f Call-ID: c47ef44f950e7a56 CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-443309010-1--d87543-;rport=42075 Record-Route: <sip:210.184.138.8:5066;lr=on;ftag=6a33d80f> Server: Sipura/SPA2000-2.0.13(g) Content-Length: 0 0(2815) DEBUG:tm:relay_reply: sent buf=0x81631c0: SIP/2.0 1..., shmem=0xb60ae728: SIP/2.0 1 0(2815) DEBUG: add_to_tail_of_timer[1]: 0xb60b049c 0(2815) DEBUG:tm:UNREF_UNSAFE: after is 0 0(2815) DEBUG:destroy_avp_list: destroying list (nil) 0(2815) receive_msg: cleaning up 1(2817) DEBUG: timer routine:4,tl=0xb60b048c next=(nil) 0(2815) SIP Reply (status): 0(2815) version: <SIP/2.0> 0(2815) status: <404> 0(2815) reason: <Not Found> 0(2815) parse_headers: flags=2 0(2815) DEBUG: add_param: tag=ab59191034906d25i0 0(2815) DEBUG:parse_to:end of header reached, state=29 0(2815) DBUG:parse_to: display={}, ruri={sip:61.144.140.110:46935} 0(2815) DEBUG: get_hdr_field: <To> [49]; uri=[sip:61.144.140.110:46935] 0(2815) DEBUG: to body [sip:61.144.140.110:46935] 0(2815) get_hdr_field: cseq <CSeq>: <1> <OPTIONS> 0(2815) Found param type 232, <branch> = <0>; state=16 0(2815) end of header reached, state=5 0(2815) parse_headers: Via found, flags=2 0(2815) parse_headers: this is the first via 0(2815) After parse_msg... 0(2815) forward_reply: found module nathelper, passing reply to it 0(2815) DEBUG:nathelper:sipping_rpl_filter: reply for SIP natping filtered 0(2815) DEBUG:destroy_avp_list: destroying list (nil) 0(2815) receive_msg: cleaning up 0(2815) SIP Reply (status): 0(2815) version: <SIP/2.0> 0(2815) status: <180> 0(2815) reason: <Ringing> 0(2815) parse_headers: flags=2 0(2815) DEBUG: add_param: tag=3b3dbe3871bbfa37i0 0(2815) DEBUG:parse_to:end of header reached, state=29 0(2815) DBUG:parse_to: display={}, ruri={sip:02061241205@210.184.138.8} 0(2815) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02061241205@210.184.138.8] 0(2815) DEBUG: to body [<sip:02061241205@210.184.138.8>] 0(2815) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(2815) Found param type 232, <branch> = <z9hG4bK72b9.90fc56c.0>; state=16 0(2815) end of header reached, state=5 0(2815) parse_headers: Via found, flags=2 0(2815) parse_headers: this is the first via 0(2815) After parse_msg... 0(2815) forward_reply: found module nathelper, passing reply to it 0(2815) forward_reply: found module tm, passing reply to it 0(2815) DEBUG: t_check: msg id=73 global id=71 T start=0xffffffff 0(2815) parse_headers: flags=22 0(2815) parse_headers: flags=8 0(2815) DEBUG: t_reply_matching: hash 39719 label 207998729 branch 0 0(2815) DEBUG:tm:REF_UNSAFE: after is 1 0(2815) DEBUG: t_reply_matching: reply matched (T=0xb60b0370)! 0(2815) DEBUG: t_check: msg id=73 global id=73 T end=0xb60b0370 0(2815) DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 is_invite=1) 0(2815) parse_headers: flags=80 0(2815) Found param type 234, <received> = <61.144.140.110>; state=6 0(2815) Found param type 232, <branch> = <z9hG4bK-d87543-443309010-1--d87543->; state=6 0(2815) Found param type 235, <rport> = <42075>; state=16 0(2815) end of header reached, state=5 0(2815) parse_headers: Via found, flags=80 0(2815) parse_headers: this is the second via 0(2815) DEBUG: get_hdr_body : content_length=0 0(2815) found end of header 0(2815) DEBUG:tm:t_should_relay_response: T_code=180, new_code=180 0(2815) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(2815) old size: 476, new size: 410 0(2815) build_res_from_sip_res: copied size: orig:245, new: 179, rest: 231 msg= SIP/2.0 180 Ringing To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0 From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f Call-ID: c47ef44f950e7a56 CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-443309010-1--d87543-;rport=42075 Record-Route: <sip:210.184.138.8:5066;lr=on;ftag=6a33d80f> Server: Sipura/SPA2000-2.0.13(g) Content-Length: 0 0(2815) DEBUG:tm:relay_reply: sent buf=0x81631c0: SIP/2.0 1..., shmem=0xb60ae728: SIP/2.0 1 0(2815) DEBUG: add_to_tail_of_timer[1]: 0xb60b049c 0(2815) DEBUG:tm:UNREF_UNSAFE: after is 0 0(2815) DEBUG:destroy_avp_list: destroying list (nil) 0(2815) receive_msg: cleaning up 0(2815) SIP Reply (status): 0(2815) version: <SIP/2.0> 0(2815) status: <200> 0(2815) reason: <OK> 0(2815) parse_headers: flags=2 0(2815) DEBUG: add_param: tag=3b3dbe3871bbfa37i0 0(2815) DEBUG:parse_to:end of header reached, state=29 0(2815) DBUG:parse_to: display={}, ruri={sip:02061241205@210.184.138.8} 0(2815) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02061241205@210.184.138.8] 0(2815) DEBUG: to body [<sip:02061241205@210.184.138.8>] 0(2815) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(2815) Found param type 232, <branch> = <z9hG4bK72b9.90fc56c.0>; state=16 0(2815) end of header reached, state=5 0(2815) parse_headers: Via found, flags=2 0(2815) parse_headers: this is the first via 0(2815) After parse_msg... 0(2815) forward_reply: found module nathelper, passing reply to it 0(2815) forward_reply: found module tm, passing reply to it 0(2815) DEBUG: t_check: msg id=74 global id=73 T start=0xffffffff 0(2815) parse_headers: flags=22 0(2815) parse_headers: flags=8 0(2815) DEBUG: t_reply_matching: hash 39719 label 207998729 branch 0 0(2815) DEBUG:tm:REF_UNSAFE: after is 1 0(2815) DEBUG: t_reply_matching: reply matched (T=0xb60b0370)! 0(2815) DEBUG: t_check: msg id=74 global id=74 T end=0xb60b0370 0(2815) DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 is_invite=1) 0(2815) parse_headers: flags=ffffffffffffffff 0(2815) Found param type 234, <received> = <61.144.140.110>; state=6 0(2815) Found param type 232, <branch> = <z9hG4bK-d87543-443309010-1--d87543->; state=6 0(2815) Found param type 235, <rport> = <42075>; state=16 0(2815) end of header reached, state=5 0(2815) parse_headers: Via found, flags=ffffffffffffffff 0(2815) parse_headers: this is the second via 0(2815) DEBUG: get_hdr_body : content_length=238 0(2815) found end of header 0(2815) DEBUG:check_content_type: type <application/sdp> found valid 0(2815) parse_headers: flags=40 0(2815) DEBUG: add_param: tag=6a33d80f 0(2815) DEBUG:parse_to:end of header reached, state=29 0(2815) DBUG:parse_to: display={3068878}, ruri={sip:3068878@210.184.138.8} 0(2815) force_rtp_proxy2: proxy reply: 35132 210.184.138.8 0(2815) parse_headers: flags=ffffffffffffffff 0(2815) parse_headers: flags=80 0(2815) parse_headers: flags=80 0(2815) DEBUG:tm:t_should_relay_response: T_code=180, new_code=200 0(2815) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(2815) parse_headers: flags=2000 0(2815) parse_headers: flags=ffffffffffffffff 0(2815) clen_builder: content-length: 255 (255) 0(2815) old size: 888, new size: 862 0(2815) build_res_from_sip_res: copied size: orig:888, new: 862, rest: 0 msg= SIP/2.0 200 OK To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0 From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f Call-ID: c47ef44f950e7a56 CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-443309010-1--d87543-;rport=42075 Record-Route: <sip:210.184.138.8:5066;lr=on;ftag=6a33d80f> Contact: 862061241205 <sip:862061241205@219.136.10.86:64189> Server: Sipura/SPA2000-2.0.13(g) Content-Length: 255 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp P-hint: Reply-Proxied v=0 o=- 12432 12432 IN IP4 192.168.10.244 s=- c=IN IP4 210.184.138.8 t=0 0 m=audio 35132 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=nortpproxy:yes 0(2815) DEBUG: add_to_tail_of_timer[2]: 0xb60b03b8 0(2815) DEBUG:tm:relay_reply: sent buf=0x81633c8: SIP/2.0 2..., shmem=0xb60b2060: SIP/2.0 2 0(2815) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(2815) DEBUG:tm:UNREF_UNSAFE: after is 0 0(2815) DEBUG:destroy_avp_list: destroying list (nil) 0(2815) receive_msg: cleaning up 0(2815) SIP Request: 0(2815) method: <ACK> 0(2815) uri: <sip:862061241205@219.136.10.86:64189> 0(2815) version: <SIP/2.0> 0(2815) parse_headers: flags=2 0(2815) DEBUG: add_param: tag=3b3dbe3871bbfa37i0 0(2815) DEBUG:parse_to:end of header reached, state=29 0(2815) DBUG:parse_to: display={}, ruri={sip:02061241205@210.184.138.8} 0(2815) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02061241205@210.184.138.8] 0(2815) DEBUG: to body [<sip:02061241205@210.184.138.8>] 0(2815) Found param type 232, <branch> = <z9hG4bK-d87543-708924014-1--d87543->; state=6 0(2815) Found param type 235, <rport> = <n/a>; state=17 0(2815) end of header reached, state=5 0(2815) parse_headers: Via found, flags=2 0(2815) parse_headers: this is the first via 0(2815) After parse_msg... 0(2815) preparing to run routing scripts... 0(2815) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(2815) parse_headers: flags=100 0(2815) get_hdr_field: cseq <CSeq>: <1> <ACK> 0(2815) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2815) parse_headers: flags=80 0(2815) parse_headers: flags=ffffffffffffffff 0(2815) DEBUG: get_hdr_body : content_length=0 0(2815) found end of header 0(2815) parse_headers: flags=200 0(2815) is_preloaded: No 0(2815) grep_sock_info - checking if host==us: 13==13 && [219.136.10.86] == [210.184.138.8] 0(2815) grep_sock_info - checking if port 5066 matches port 64189 0(2815) grep_sock_info - checking if host==us: 13==13 && [219.136.10.86] == [210.184.138.8] 0(2815) grep_sock_info - checking if port 5066 matches port 64189 0(2815) check_self: host != me 0(2815) grep_sock_info - checking if host==us: 13==13 && [210.184.138.8] == [210.184.138.8] 0(2815) grep_sock_info - checking if port 5066 matches port 5066 0(2815) after_loose: Topmost route URI: 'sip:210.184.138.8:5066;lr;ftag=6a33d80f' is me 0(2815) parse_headers: flags=200 0(2815) found end of header 0(2815) find_next_route: No next Route HF found 0(2815) after_loose: No next URI found 0(2815) DBG:rr:run_rr_callbacks: callback id 0 entered 0(2815) DEBUG:uac:restore_from: getting 'vsf' Route param 0(2815) DEBUG:uac:restore_from: Route param 'vsf' not found 0(2815) parse_headers: flags=ffffffffffffffff 0(2815) parse_headers: flags=ffffffffffffffff 0(2815) ERROR: extract_body: message body has length zero 0(2815) ERROR: force_rtp_proxy2: can't extract body from the message 0(2815) DEBUG: t_newtran: msg id=75 , global msg id=74 , T on entrance=0xffffffff 0(2815) parse_headers: flags=ffffffffffffffff 0(2815) parse_headers: flags=78 0(2815) t_lookup_request: start searching: hash=39719, isACK=1 0(2815) parse_headers: flags=38 0(2815) DEBUG: add_param: tag=6a33d80f 0(2815) DEBUG:parse_to:end of header reached, state=29 0(2815) DBUG:parse_to: display={3068878}, ruri={sip:3068878@210.184.138.8} 0(2815) DEBUG: t_lookup_request: e2e proxy ACK found 0(2815) DEBUG:tm:REF_UNSAFE: after is 1 0(2815) DEBUG:tm:t_relay: forwarding ACK 0(2815) DEBUG:tm:t_relay: it's an end2end ACK 0(2815) DEBUG:tm:UNREF_UNSAFE: after is 0 0(2815) DEBUG: mk_proxy: doing DNS lookup... 0(2815) check_via_address(61.144.140.110, 192.168.1.168, 1) 0(2815) check_via_address: doing dns lookup 0(2815) Sending: ACK sip:862061241205@219.136.10.86:64189 SIP/2.0 To: <sip:02061241205@210.184.138.8>;tag=3b3dbe3871bbfa37i0 From: 3068878<sip:3068878@210.184.138.8>;tag=6a33d80f Via: SIP/2.0/UDP 210.184.138.8:5066;branch=z9hG4bK72b9.90fc56c.2 Via: SIP/2.0/UDP 192.168.1.168:9801;received=61.144.140.110;branch=z9hG4bK-d87543-708924014-1--d87543-;rport=42075 Call-ID: c47ef44f950e7a56 CSeq: 1 ACK Contact: <sip:3068878@61.144.140.110:42075> Max-Forwards: 69 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 0 P-hint: Proxied P-hint: rr-enforced . 0(2815) orig. len=470, new_len=556, proto=1 0(2815) DEBUG:destroy_avp_list: destroying list (nil) 0(2815) receive_msg: cleaning up 1(2817) DEBUG: timer routine:1,tl=0xb60b049c next=(nil) 0(2815) SIP Request: 0(2815) method: <BYE> 0(2815) uri: <sip:862061241205@219.136.10.86:64189> 0(2815) version: <SIP/2.0> 0(2815) parse_headers: flags=2 0(2815) DEBUG: add_param: tag=3b3dbe3871bbfa37i0 0(2815) DEBUG:parse_to:end of header reached, state=29 0(2815) DBUG:parse_to: display={}, ruri={sip:02061241205@210.184.138.8} 0(2815) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02061241205@210.184.138.8] 0(2815) DEBUG: to body [<sip:02061241205@210.184.138.8>] 0(2815) Found param type 232, <branch> = <z9hG4bK-d87543-956249433-1--d87543->; state=6 0(2815) Found param type 235, <rport> = <n/a>; state=17 0(2815) end of header reached, state=5 0(2815) parse_headers: Via found, flags=2 configuration : if (nat_uac_test("19")) { if (!search("^Record-Route:")) { fix_nated_contact(); # Rewrite contact with source IP of signalling if (method=="REGISTER") { fix_nated_register(); } else if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via append_hf("P-hint: Proxied\r\n"); setflag(10); # Mark as NATed }; }; if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); exit(); }; if (method=="REGISTER") { if (src_ip==222.88.222.67 || src_ip==210.184.138.4) { save("location"); exit(); } if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit(); }; save("location"); if (isflagset(10)) { if (!(src_ip==222.88.222.67 || src_ip==210.184.138.4)) { #add_path_received(); append_branch("sip:222.88.222.67:80"); t_replicate("sip:210.184.138.4:80"); } } else { if (!(src_ip==222.88.222.67 || src_ip==210.184.138.4)) { #add_path(); append_branch("sip:222.88.222.67:80"); t_replicate("sip:210.184.138.4:80"); } } return; }; route[1] { # if client or server know to be behind a NAT, enable relay if (isflagset(10)) { 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"); t_on_failure("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); return; }; }

2 2

Re: [Users] Using # for Sip 2 Sip calls
by Glenn Dalgliesh 30 Jun '06

30 Jun '06

Well I would becarefull using # since some UA's use # to terminate digit input and dial..... Not positive but I think * would be a better choice. --------------------- Sent with ChatterEmail True push email for the Treo Smartphone www.chatteremail.com -----Original Message----- From: Kenny Chua <strain15(a)yahoo.com> Date: Wednesday, Jun 28, 2006 10:56 pm Subject: [Users] Using # for Sip 2 Sip calls Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls? I came up with something like this: lookup("aliases"); if (uri=~"^sip:#[0-9]*@"){ xlog("Sip 2 SIP\n"); route(4); route(1); return; }; Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you. --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail Beta. --0-591390942-1151549737=:48905 Content-Type: text/html; charset=iso-8859-1 Content-Transfer-Encoding: 8bit Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls? I came up with something like this:     lookup("aliases");     if (uri=~"^sip:#[0-9]*@"){         xlog("Sip 2 SIP\n");         route(4);         route(1);         return;     }; Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you. <hr size=1>Do you Yahoo!? Get on board. <a href="http://us.rd.yahoo.com/evt=40791/*http://advision.webevents.yahoo.com/handr…">You're invited</a> to try the new Yahoo! Mail Beta. --0-591390942-1151549737=:48905-- [Users] Using # for Sip 2 Sip callsKenny Chua <strain15(a)yahoo.com>To: users(a)openser.org

3 2

[Users] Last SEMS version compatible with openSER ?
by Christophe Irles 30 Jun '06

30 Jun '06

Hi, I'm wondering if the last version of SEMS is still compatible with OpenSER 1.0.X or more ? Is someone using it there ? are the conferencing and voice mail feature working well ? Thanks, Christophe

3 2

[Serusers] monitoring ser
by Giuseppe Parlato 30 Jun '06

30 Jun '06

Hi all, Have you ever tried to monitor ser using mrtg or what's up gold ? Is there any way to monitor this service using snmp? thanks Giuseppe

4 4

[Serusers] Call quality
by Mike Cassidy 30 Jun '06

30 Jun '06

Can anyone point me in the right direction for resources that could help me to improve call quality? I have everything up and running ok, but the quality of the calls is a bit ropey at the moment. Any help here would be much appreciated. Thanks in advance, Mike

3 2

Re: [Users] Using # for Sip 2 Sip calls
by Kenny Chua 30 Jun '06

30 Jun '06

Ok, I've just used this routing, it just shows thats its trying and the call never goes through. Here is my openser.cfg with the data from ngrep. Not sure why the call wont go through: debug=3 fork=yes log_stderror=no listen=192.168.1.6 alias=test.com port=5060 children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" 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/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); t_relay(); exit; } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); return; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); return; }; if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (uri=~"^sip:\*[0-9]*@"){ xlog("Sip 2 Sip\n"); strip(1); #strip away the * route(4); route(1); return; }; if (!lookup("location")){ if (uri=~"^sip:[0-9]*@") { xlog("PSTN Gateway\n"); route(4); route(5); return; }; sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- xlog("Routed to route 5\n"); rewritehostport("pstn.gateway:5060"); avp_write("i:45", "inv_timeout"); route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } U 2006/06/29 14:33:43.120386 192.168.1.1:1027 -> 192.168.1.6:5060 INVITE sip:201@test.com SIP/2.0..Record-Route: <sip:192.168.1.6;f tag=1535693657;lr=on>..Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bK038c.2054 5ab4.0..Via: SIP/2.0/UDP 70.xxx.xxx.xx:5060;received=192.168.1.1;rport=5060 ;branch=z9hG4bK5F856FB99EE741348558D3F5BDFB4199..From: Kenny <sip:200@test.com>;tag=1535693657..To: <sip:*201@test.com>..Contact: <sip:200@192.168.1.1:5060>..Call-ID: 0A64050F-C53B-4751-94FD-4658CB6CF37F@1 92.168.1.30..CSeq: 33944 INVITE..Max-Forwards: 69..Content-Type: applicatio n/sdp..User-Agent: X-Lite release 1105x..Content-Length: 325....v=0..o=200 268371887 268371936 IN IP4 70.189.13.171..s=X-Lite..c=IN IP4 192.168.1.6..t =0 0..m=audio 21088 RTP/AVP 0 8 3 98 97 101..a=rtpmap:0 pcmu/8000..a=rtpmap :8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex /8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv..a=no rtpproxy:yes.. U 2006/06/29 14:33:43.120906 192.168.1.6:5060 -> 192.168.1.1:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 192.168.1.6;bra nch=z9hG4bK038c.20545ab4.0;received=192.168.1.1..Via: SIP/2.0/UDP 70.xxx.xxx.xxx:5060;received=192.168.1.1;rport=5060;branch=z9hG4bK5F856FB99EE74134855 8D3F5BDFB4199..From: Kenny <sip:200@test.com>;tag=1535693657..To: <sip:*201@test.com>;tag=eb9822f1ed1b15a932bf98c097b3aaca.4dc7..C all-ID: 0A64050F-C53B-4751-94FD-4658CB6CF37F@192.168.1.30..CSeq: 33944 INVI TE..Proxy-Authenticate: Digest realm="test.com", nonce="44a42c439 d432ec2bff44d80632d0ec98b8d4d6e"..Server: OpenSer (1.0.1 (i386/linux))..Con tent-Length: 0..Warning: 392 192.168.1.6:5060 "Noisy feedback tells: pid=7 010 req_src_ip=192.168.1.1 req_src_port=1027 in_uri=sip:201@test.com out_uri=sip:201@test.com via_cnt==2".... --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs.Try it free.

2 1

[Users] Hung up incoming calls
by raviprakash sunkara 30 Jun '06

30 Jun '06

Hi , A small issues, But i seems to be challenging issuses to all, That... UAC are behind the NAT.. are hunging up automatically.. That caller is not hung up, callee is hung up after 32 seconds. And missed and rejecting call are working fine. I think , ACK session time out and is not requesting to openser . problem in openser.cfg with NAT or X-lite 3.0 softphones.. Problem in confiuration setting in softphones. Or Problem with router Gateways ( NETGEAR) from outbound to inbound. > Please any one give me suggestions or clues to resolve my issuess. > > > > > > > > > > > > > > > > > > > Bye Happpen Weakend ,,,,, Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer && openSER) Hyperion Technology www.hyperion-tech.com

1 0

[Serusers] Configuring Sipura SPA942 for use with ffee IPTEL.ORG service
by David 30 Jun '06

30 Jun '06

Hi, I have registered for a free iptel.org sip account. I have a Sipura SPA942 sip phone and I am trying to find basic configuration information for setting up a sip phone for use with iptel.org's free service. I have scoured the site and cannot seem to find anything. Can someone provide the basic information or point me to the correct documentation on setting up a hard phone for use on iptel.org? Thanks,

2 3

[Serusers] Create a new module
by Kavvalaki Alexandra 30 Jun '06

30 Jun '06

Hello, Can anyone tell me how can i create a new module? Thank you

3 2

[Serusers] Suggestion Required
by Abdul Lateef 30 Jun '06

30 Jun '06

Hi friends, I am new in OpenSER, I want to use it only for SIP Proxying with freeradius. I made plan to install openser with freeradius on Virtual Server to get only 100 cuncurent calls. 1- Is it possible to install on Virtual Server? 2- Which Codecs or used, because i want to calculate the bandhwidth according to the codec? 3- What RAM should be used to handel 100 cuncurent calls? 4- It can accept h323-credit-time from radius to control max credit call time? I will appriciate for your kind of suggestion. Regards, __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] (no subject)
by Kavvalaki Alexandra 30 Jun '06

30 Jun '06

Hello, Can anyone tell me how can i create a new module? Thanks

1 0

[Serusers] Assure a BYE is received ??
by Adrian Georgescu 30 Jun '06

30 Jun '06

I do now know your setup but you only need to add some extra columns to your radacct table. Except for the time required to apply the sql changes (your table will lock and you cannot write to it) I am not aware of any impact on a standard freeradius installation. Adrian >>>> Ok, it means that I'll just make the RTP pass through mediaproxy by enabling the use of it on non-NATed UA's. Just a question, I have an existing RADIUS accounting setup, would there be a problem if I implemented the accounting feature of mediaproxy? I'm afraid because written in the docs in the distro i need to patch the radacct table, will it affect the existing one or alter something? Thanks, Ryan

3 6

[Serusers] crash info - DBG: tcp_main_loop: dead child 2 (shutting down?)
by Hakan YASTI 30 Jun '06

30 Jun '06

Dear All, I user ser-0.9.3_1. Today it has crashed and I got the following logs from debug.log. What can be the problem? Best Regards, Hakan. Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: SIP Reply (status): Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: version: <SIP/2.0> Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: status: <100> Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: reason: <Trying> Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: parse_headers: flags=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: Found param type 232, <branch> = <z9hG4bK846c.8bba40d.0>; state=16 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: end of header reached, state=5 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: parse_headers: Via found, flags=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: parse_headers: this is the first via Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: After parse_msg... Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: grep_sock_info - checking if host==us: 11==11 && [84.51.32.21] == [84.51.32.21] Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: grep_sock_info - checking if port 5060 matches port 5060 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: forward_reply: found module tm, passing reply to it Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: t_check: msg id=257654 global id=257653 T start=0xffffffff Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: parse_headers: flags=17 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: Found param type 232, <branch> = <z9hG4bKacHxBfRNa>; state=16 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: end of header reached, state=5 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: parse_headers: Via found, flags=17 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: parse_headers: this is the second via Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: end of header reached, state=9 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: get_hdr_field: <To> [43]; uri=[sip:009xxxxxxxxx@mydomain;user=phone] Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: to body [<sip:009xxxxxxxxxx@mydomain;user=phone>^M ] Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: get_hdr_field: cseq <CSeq>: <1> <INVITE> Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: parse_headers: flags=4 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: t_reply_matching: hash 50760 label 218409912 branch 0 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: t_reply_matching: reply matched (T=0x2877aa78)! Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: t_check: msg id=257654 global id=257654 T end=0x2877aa78 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: reply_received: org. status uas=100, uac[0]=0 local=0 is_invite=1) Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: ->>>>>>>>> T_code=100, new_code=100 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: relay_reply: branch=0, save=0, relay=-1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: add_to_tail_of_timer[1]: 0x2877aba0 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG:destroy_avp_list: destroying list 0x0 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: receive_msg: cleaning up Jun 26 14:37:56 globalser /usr/local/sbin/ser[666]: DBG: tcp_main_loop: dead child 2 (shutting down?) Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: Memory status (pkg): Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: qm_status (0x8103e40): Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: heap size= 1027664 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: used= 807328, used+overhead=1018224, free=9440 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: max used (+overhead)= 1026976 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: dumping all alloc'ed. fragments: Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: Memory status (pkg): Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 0. N address=0x8108fe8 frag=0x8108fe0 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: qm_status (0x8103e40): Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 1. N address=0x8109078 frag=0x8109070 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: heap size= 1027664 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 2. N address=0x8109108 frag=0x8109100 size=32 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: used= 809296, used+overhead=1020912, free=6752 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 3. N address=0x8109138 frag=0x8109130 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: max used (+overhead)= 1028288 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 4. N address=0x8109158 frag=0x8109150 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: dumping all alloc'ed. fragments: Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 5. N address=0x8109178 frag=0x8109170 size=112 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: 0. N address=0x8108fe8 frag=0x8108fe0 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 6. N address=0x81091f8 frag=0x81091f0 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: 1. N address=0x8109078 frag=0x8109070 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 7. N address=0x8109218 frag=0x8109210 size=112 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: 2. N address=0x8109108 frag=0x8109100 size=32 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 8. N address=0x8109298 frag=0x8109290 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: 3. N address=0x8109138 frag=0x8109130 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 9. N address=0x81092b8 frag=0x81092b0 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: 4. N address=0x8109158 frag=0x8109150 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 10. N address=0x81092d8 frag=0x81092d0 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: 5. N address=0x8109178 frag=0x8109170 size=112 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 11. N address=0x81092f8 frag=0x81092f0 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: 6. N address=0x81091f8 frag=0x81091f0 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[645]: 71. N address=0x810a6f8 frag=0x810a6f0 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[645]: 73. N address=0x810a818 frag=0x810a810 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[645]: 74. N address=0x810a8a8 frag=0x810a8a0 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[645]: 75. N address=0x810a938 frag=0x810a930 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[645]: 76. N address=0x810a9c8 frag=0x810a9c0 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[645]: 77. N address=0x810aa58 frag=0x810aa50 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[645]: 78. N address=0x810aae8 frag=0x810aae0 size=128 used=1

2 1

Re: [Users] Cisco radius auth
by Glenn Dalgliesh 30 Jun '06

30 Jun '06

What is your existing billing system? Many radius based platforums support digest auth. The cisco portion take some coding to make work but can be done. --------------------- Sent with ChatterEmail True push email for the Treo Smartphone www.chatteremail.com -----Original Message----- From: Jeff Williams <jeffw(a)globaldial.com> Date: Thursday, Jun 29, 2006 9:55 pm Subject: Re: [Users] Cisco radius auth Jeff Williams wrote: Is there any way to get openser to do radius auth using attribute from the Cisco VSA Voice Implementation Guide <http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_programming…>? i.e. something like: > Access-Request NAS-IP-Address = 203.56.98.120 NAS-Port-Type = Async User-Name = "62186234" cisco-h323-conf-id = "h323-conf-id=89437006 E20011DA B677BBCA D2EBCDD5" Called-Station-Id = "0011553334112401" Calling-Station-Id = "0295731743" User-Password = "<145>;<22><242>$<251><243>)u<183>h<225>y<173><150><193>" > I would like to get openser to authenticate against an existing billing system if possible. Mmm, I realised after sending this that since SIP does digest auth and doesn't send a password as such, this is probably not possible. Jeff _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Serusers] Assure a BYE is received ??
by Adrian Georgescu 30 Jun '06

30 Jun '06

There is no guarantee that a BYE will be received from an end-point. One possible solution is to use MediaProxy 1.7.2 in combination with SER. It will correctly terminate calls that have no BYE, by updating the radacct table AcctSessionTime column based on the last time RTP was relayed. Adrian ===== Hi, Use for logging: if (method=="BYE" || method=="CANCEL") { log(1, "SER: BYE"); setflag(1); } and record_route() to be sure that "BYE" wil visit your server. Dani On Mon, 26 Jun 2006 17:42:24 +0800 Ryan Pagquil <rpagquil at philonline.com> wrote: > Hi, > How can I be sure that all calls are terminated by a BYE message? I > have some instances that a BYE message is not logged by SER. Is there > a way to fix this? What could be the possible scenarios that causes > missing BYE's? > > Thanks, > Ryan >

4 10

[Users] Using # for Sip 2 Sip calls
by Glenn Dalgliesh 29 Jun '06

29 Jun '06

Give this a try.. also if issues still exist include log messages from a call attempt with the ngrep. If you get a 407 and your client is getting it back you should see a second invite including the auth info. debug=3 fork=yes log_stderror=no listen=192.168.1.6 alias=test.com port=5060 children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" 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/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); t_relay(); exit; } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); return; }; if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { xlog("Invite section\n"); route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(3); route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- xlog("Route 3\n"); if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } if (uri=~"^sip:\*[0-9]*@"){ xlog("Sip 2 Sip\n"); strip(1); #strip away the * lookup(aliases); lookup(locations); route(4); route(1); return; } if (!lookup("location")){ if (uri=~"^sip:[0-9]*@") { xlog("PSTN Gateway\n"); route(4); route(5); return; }; sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- xlog("Routed to route 5\n"); rewritehostport("pstn.gateway:5060"); avp_write("i:45", "inv_timeout"); route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; }

1 0

Re: [Users] Using # for Sip 2 Sip calls
by Kenny Chua 29 Jun '06

29 Jun '06

So I took your advice and decided to use * to identify sip 2 sip calls. However, theres something wrong with my routing. I added route(6) to get authorize. Because when I try to dial sip to sip I get 407 proxy authentication required. Still after adding route(6), I still get the 407 proxy authentication required message. What is wrong? Route (1) is just the default message handler This is what I have: route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (uri=~"^sip:\*[0-9]*@"){ xlog("Sip 2 Sip\n"); strip(1); #strip the * because we dont need it route(4); route(6); route(1); return; }; if (!lookup("location")){ if (uri=~"^sip:[0-9]*@") { # International PSTN xlog("PSTN Gateway\n"); route(4); route(5); return; }; sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- xlog("Routed to route 5\n"); rewritehostport("pstn.gateway:5060"); avp_write("i:45", "inv_timeout"); route(1); } route[6] { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: Hi, that's right. For example SIPURA ATAs with two lines but online one terminal use # for line selection.... you better use a digit that does not overlap with the PSTN dialling plan. regards, bogdan Glenn Dalgliesh wrote: >Well I would becarefull using # since some UA's use # to terminate digit input and dial..... Not positive but I think * would be a better choice. >--------------------- >Sent with ChatterEmail >True push email for the Treo Smartphone >www.chatteremail.com > > >-----Original Message----- >From: Kenny Chua >Date: Wednesday, Jun 28, 2006 10:56 pm >Subject: [Users] Using # for Sip 2 Sip calls > >Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls? > > I came up with something like this: > lookup("aliases"); > if (uri=~"^sip:#[0-9]*@"){ > xlog("Sip 2 SIP\n"); > route(4); > route(1); > return; > }; > > Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you. > > >--------------------------------- >Do you Yahoo!? > Get on board. You're invited to try the new Yahoo! Mail Beta. >--0-591390942-1151549737=:48905 >Content-Type: text/html; charset=iso-8859-1 >Content-Transfer-Encoding: 8bit > >Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls? I came up with something like this: lookup("aliases"); if (uri=~"^sip:#[0-9]*@"){ xlog("Sip 2 SIP\n"); route(4); route(1); return; }; Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you. > --------------------------------- Sneak preview the all-new Yahoo.com. It's not radically different. Just radically better.

2 1

[Serusers] sip phone for solaris?
by Cesc 29 Jun '06

29 Jun '06

Hi, Any experiences? ideally it would come pre-compiled, but anything will be good enough (own compilation ... Java ... ) Cesc

2 1

[Users] avp problems switching from 1.0.1 to 1.1.x
by tele 29 Jun '06

29 Jun '06

Hi, My billing system doesn't correlate START/STOP accounting packets and use Cisco-VSA like attributes. I need to do an accounting compatible with my billing system from OpenSER so i've done a dirty method like that: i know it's not really performance but it's the only way i can do that. The problem is with OpenSER 1.1.x that generate accounting with the attribute filled with the same value of "disconnect_time". example: h323-setup-time = '1151511700' h323-connect-time = '1151511700' h323-disconnect-time = '1151511700' the xlog debug output of openser1.1.x i have added is with correct attribute values. with OpenSER 1.0.1 everything works fine. CONFIGS: with OpenSER 1.1.x: modparam("acc", "radius_extra", "h323-setup-time=$avp(setup_time); h323-connect-time=$avp(connect_time); h323-disconnect-time=$avp(disconnect_time)") modparam("avpops","avp_aliases","setup_time=s:setup_time") modparam("avpops","avp_aliases","connect_time=s:connect_time") modparam("avpops","avp_aliases","disconnect_time=s:disconnect_time") if (is_method("INVITE")) { avp_write("$Ts","$avp(setup_time)"); avp_db_store("$from","$avp(setup_time)"); } else if (is_method("ACK")) { avp_write("$Ts","$avp(connect_time)"); avp_db_store("$from","$avp(connect_time)"); } else if (is_method("BYE")) { avp_write("$Ts","$avp(disconnect_time)"); avp_db_store("$from","$avp(disconnect_time)"); }; if (is_method("BYE")) { avp_db_load("$from","$avp(setup_time)"); avp_db_load("$from","$avp(connect_time)"); avp_db_load("$from","$avp(disconnect_time)"); xlog("L_INFO","Before accounting SETUP_TIME[$avp(setup_time)] CONNECT_TIME[$avp(connect_time)] DISCONNECT_TIME[$avp(disconnect_time)]"); setflag(3); setflag(5); avp_db_delete("$from","$avp(setup_time)"); avp_db_delete("$from","$avp(connect_time)"); avp_db_delete("$from","$avp(disconnect_time)"); }; With OpenSER 1.0.1: modparam("acc", "radius_extra", "h323-setup-time=$avp($setup_time); h323-connect-time=$avp($connect_time) ; h323-disconnect-time=$avp($disconnect_time)") modparam("avpops","avp_aliases","setup_time=s:setup_time") modparam("avpops","avp_aliases","connect_time=s:connect_time") modparam("avpops","avp_aliases","disconnect_time=s:disconnect_time") if (is_method("INVITE")) { avp_write("0","i:1"); avp_printf("i:1", "$Ts"); avp_copy("i:1","$setup_time"); avp_db_store("$from","$setup_time"); avp_delete("i:1"); } else if (is_method("ACK")) { avp_write("0","i:1"); avp_printf("i:1", "$Ts"); avp_copy("i:1","$connect_time"); avp_db_store("$from","$connect_time"); avp_delete("i:1"); } else if (is_method("BYE")) { avp_write("0","i:1"); avp_printf("i:1", "$Ts"); avp_copy("i:1","$disconnect_time"); avp_db_store("$from","$disconnect_time"); avp_delete("i:1"); }; if (is_method("BYE")) { avp_db_load("$from","$setup_time"); avp_db_load("$from","$connect_time"); avp_db_load("$from","$disconnect_time"); xlog("L_INFO","Before accounting SETUP_TIME[$avp($setup_time)] CONNECT_TIME[$avp($connect_time)] DISCONNECT_TIME[$avp($disconnect_time)]"); setflag(3); setflag(5); avp_db_delete("$from","$setup_time"); avp_db_delete("$from","$connect_time"); avp_db_delete("$from","$disconnect_time"); };

4 7

[Users] loose_route goes strict - Stray colon in Record-Route
by G.Jacobsen 29 Jun '06

29 Jun '06

Further to below problem regarding BYEs not loose_routed. I see now that openser sends the invite with a wierd colon at the end of "127.0.0.1:" in record route. The same colon is of course also present in the BYEs which loop. Is it possible that OS gets confused by this when receiving the BYE ? What causes this - How can I work around this ? Thanks a lot for your help in the matter. I am using openser 1.0.1 Gerry Inivite sent by OS: INVITE sip:7336058@127.0.0.1:5061 SIP/2.0 Max-Forwards: 10 Record-Route: <sip:127.0.0.1:;r2=on;ftag=as58f53672;lr=on> Record-Route: <sip:101.101.101.101;r2=on;ftag=as58f53672;lr=on> Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK39c3.dcf37ad6.0 Via: SIP/2.0/UDP 202.202.202.202:5060;branch=z9hG4bK2955936f From: "7654321" <sip:7654321@XXXXXXXXXXXX>;tag=as58f53672 To: <sip:7654321@XXXXXXXXXXXX> Contact: <sip:7654321@202.202.202.202> Call-ID: 5cdc877611378bf05dc78ef3035be7fb@xxxxxxxxxxxxxx CSeq: 103 INVITE User-Agent: Asterisk PBX Date: Thu, 29 Jun 2006 07:23:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 240 -----Original Message----- From: G.Jacobsen [mailto:g_jacobsen@yahoo.co.uk] Sent: Donnerstag, 29. Juni 2006 01:04 To: Devel@Openser. Org Subject: loose_route goes strict with BYEs Gentlemen, I have openser sitting on the same machine as another UA - openser is on port 5060 the other on 5061. BYEs by the UA are not loose_routed to the originating client - causing the BYEs to loop on openser with 513 message too big. I am not sure whether this is related - but a similar problem seems to have been touched here before - still cant find the solution. http://openser.org/pipermail/devel/2006-April/002540.html I have put all aliases with the port 5060 now - still doesnt help. If I read the debug correctly openser does strict routing on this line - why ? "rewrite_uri: Rewriting Request-URI with 'sip:111.111.111.111;r2=on;ftag=109880961431220;lr=on' " I would be most grateful if someone could explain which parameter causes the BYE to go strict. TIA Gerry Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: SIP Request: Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: method: <BYE> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: uri: <sip:1234567@sip.somedomain.com:5060> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: version: <SIP/2.0> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=2 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: add_param: tag=109880961431220 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG:parse_to:end of header reached, state=29 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: get_hdr_field: <To> [55]; uri=[sip:1234567@sip.somedomain.com:5060] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: to body [<sip:1234567@sip.somedomain.com:5060>] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: Found param type 235, <rport> = <n/a>; state=6 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: Found param type 232, <branch> = <z9hG4bK719885386>; state=16 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: end of header reached, state=5 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: Via found, flags=2 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: this is the first via Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: After parse_msg... Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: preparing to run routing scripts... Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=100 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: get_hdr_field: cseq <CSeq>: <2> <BYE> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: add_param: tag=1957747793 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG:parse_to:end of header reached, state=29 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=200 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: is_preloaded: No Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if host==us: 14==9 && [sip.somedomain.com] == [127.0.0.1] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if port 5060 matches port 5060 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if host==us: 14==13 && [sip.somedomain.com] == [111.111.111.111] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if port 5060 matches port 5060 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if host==us: 14==9 && [sip.somedomain.com] == [127.0.0.1] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if port 5060 matches port 5060 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if host==us: 14==13 && [sip.somedomain.com] == [111.111.111.111] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if port 5060 matches port 5060 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if host==us: 9==9 && [127.0.0.1] == [127.0.0.1] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if port 5060 matches port 5060 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: grep_sock_info - checking if host==us: 9==9 && [127.0.0.1] == [127.0.0.1] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DBG:rr:run_rr_callbacks: callback id 0 entered Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG:uac:restore_from: getting 'vsf' Route param Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG:uac:restore_from: Route param 'vsf' not found Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: after_strict: Next hop: 'sip:111.111.111.111;r2=on;ftag=109880961431220;lr=on' is loose router Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=ffffffffffffffff Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: get_hdr_body : content_length=0 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: found end of header Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: rewrite_uri: Rewriting Request-URI with 'sip:111.111.111.111;r2=on;ftag=109880961431220;lr=on' Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: after_strict: The last route URI: 'sip:111.111.111.111;r2=on;ftag=109880961431220;lr=on' Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=ffffffffffffffff Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=58 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: rc_get_seqnbr: couldn't open sequence file /var/run/radius.seq: Permission denied Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=58 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: ACC: request accounted: method=BYE, i-uri=sip:1234567@sip.somedomain.com:5060, o-uri=sip:111.111.111.111;r2=on;ftag=109880961431220;lr=on, call_id=2BCF5051-F28C-4CCA-AE6B-D895607B668E(a)81.198.36.24, from=<sip:7654321@sip.somedomain.com:5060>;tag=1957747793, code=200 ok Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: t_newtran: msg id=2 , global msg id=1 , T on entrance=0xffffffff Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=ffffffffffffffff Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: parse_headers: flags=78 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: t_lookup_request: start searching: hash=51354, isACK=0 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: RFC3261 transaction matching failed Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: t_lookup_request: no transaction found Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DBG: trans=0x405797d8, callback type 1, id 0 entered Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: mk_proxy: doing DNS lookup... Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: check_via_address(127.0.0.1, 127.0.0.1, 0) Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: add_to_tail_of_timer[4]: 0x405798f4 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: SIP Request: Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG: add_to_tail_of_timer[0]: 0x40579904 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: method: <BYE> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: SER: new transaction fwd'ed Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: uri: <sip:111.111.111.111;r2=on;ftag=109880961431220;lr=on> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: DEBUG:destroy_avp_list: destroying list (nil) Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: version: <SIP/2.0> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19818]: receive_msg: cleaning up Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: parse_headers: flags=2 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG: add_param: tag=109880961431220 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG:parse_to:end of header reached, state=29 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG: get_hdr_field: <To> [55]; uri=[sip:1234567@sip.somedomain.com:5060] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG: to body [<sip:1234567@sip.somedomain.com:5060>] Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: Found param type 232, <branch> = <z9hG4bKa98c.f4c21e03.0>; state=16 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: end of header reached, state=5 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: parse_headers: Via found, flags=2 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: parse_headers: this is the first via Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: After parse_msg... Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: preparing to run routing scripts... Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: parse_headers: flags=100 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: Found param type 235, <rport> = <5061>; state=6 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: Found param type 232, <branch> = <z9hG4bK719885386>; state=16 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: end of header reached, state=5 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: parse_headers: Via found, flags=100 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: parse_headers: this is the second via Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: get_hdr_field: cseq <CSeq>: <2> <BYE> Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG:maxfwd:is_maxfwd_present: value = 69 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG: add_param: tag=1957747793 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG:parse_to:end of header reached, state=29 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: parse_headers: flags=200 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: DEBUG: get_hdr_body : content_length=0 Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: found end of header Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: find_first_route: No Route headers found Jun 29 00:32:16 gatesip1 /usr/local/sbin/openser[19830]: loose_route: There is no Route HF ___________________________________________________________ All new Yahoo! Mail "The new Interface is stunning in its simplicity and ease of use." - PC Magazine http://uk.docs.yahoo.com/nowyoucan.html

2 1

[Users] openser adding a voxee route
by Kumar Subramanian 29 Jun '06

29 Jun '06

we use SER as the SIP proxy and have a route/voice termination with voxee. We want to share/use this route from our ip phones. Is there a way to add this route to ser or operser. voxee requires a registration with the userid & password provided by voxee. So, when a call is made , SER would have to rewrite the INVITE so that the originator has the username/password of the voxee account that had we had setup. In the eyes of voxee the calls would be coming in from one user(the account that we opened). Please let me know how to achieve this editing openser.cfg, using modules etc Thanks

2 1

[Users] disconnect reason code
by tele 29 Jun '06

29 Jun '06

Hi all, Is there a pseudo variables with the disconnect code for a calls? I need to catch the value of the attribute Sip-Response-Code but for another attribute like: modparam("acc","radius_extra","h323-disconnect-cause=$code"); how can i do this? regards :tele

1 0

[Serusers] Problem with ser 0.9.6 on CentOS 4.3 x86_64
by nick 29 Jun '06

29 Jun '06

I'm running CentOS 4.3 x86_64 on a dual opteron 246 system with 1 gig of ram, kernel 2.6.9-34.ELsmp. Ser runs fine with quite a few modules, but when I enable acc.so, I immediately get a segfault. I ran it under gdb and was able to obtain this information: #0 0x0000002a9578516f in insert_tmcb (cb_list=0x0, types=1, f=0x2a955675c7 <acc_onreq>, param=0x0) at t_hooks.c:99 #1 0x0000002a95785494 in register_tmcb (p_msg=0x0, t=0x0, types=1, f=0x2a955675c7 <acc_onreq>, param=0x0) at t_hooks.c:166 #2 0x0000002a955674c7 in mod_init () at acc_mod.c:308 #3 0x0000000000430d79 in init_mod (m=0x5d5228) at sr_module.c:474 #4 0x0000000000430ce8 in init_mod (m=0x5d52e8) at sr_module.c:471 #5 0x0000000000430ce8 in init_mod (m=0x5d53a8) at sr_module.c:471 #6 0x0000000000430ce8 in init_mod (m=0x5d5468) at sr_module.c:471 #7 0x0000000000430ce8 in init_mod (m=0x5d5528) at sr_module.c:471 #8 0x0000000000430ce8 in init_mod (m=0x5d55e8) at sr_module.c:471 #9 0x0000000000430ce8 in init_mod (m=0x5d56a8) at sr_module.c:471 #10 0x0000000000430ce8 in init_mod (m=0x5d5768) at sr_module.c:471 #11 0x0000000000430e05 in init_modules () at sr_module.c:497 #12 0x0000000000416f3c in main (argc=3, argv=0x7fbffffa18) at main.c:1577 --------------------------------------------- Starting program: /usr/local/sbin/ser -V version: ser 0.9.6 (x86_64/linux) flags: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197.2.1 2005/07/25 16:56:24 andrei Exp $ main.c compiled on 09:19:06 Jun 28 2006 with gcc 3.4.5 Program exited normally. --------------------------------------------------- I also did a bt full: (gdb) bt full #0 0x0000002a9578516f in insert_tmcb (cb_list=0x0, types=1, f=0x2a955675c7 <acc_onreq>, param=0x0) at t_hooks.c:99 cbp = (struct tm_callback *) 0x2a96010a78 #1 0x0000002a95785494 in register_tmcb (p_msg=0x0, t=0x0, types=1, f=0x2a955675c7 <acc_onreq>, param=0x0) at t_hooks.c:166 cb_list = (struct tmcb_head_list *) 0x0 #2 0x0000002a955674c7 in mod_init () at acc_mod.c:308 load_tm = 0x2a9578d4a0 <load_tm> #3 0x0000000000430d79 in init_mod (m=0x5d5228) at sr_module.c:474 No locals. #4 0x0000000000430ce8 in init_mod (m=0x5d52e8) at sr_module.c:471 No locals. #5 0x0000000000430ce8 in init_mod (m=0x5d53a8) at sr_module.c:471 No locals. #6 0x0000000000430ce8 in init_mod (m=0x5d5468) at sr_module.c:471 No locals. #7 0x0000000000430ce8 in init_mod (m=0x5d5528) at sr_module.c:471 No locals. #8 0x0000000000430ce8 in init_mod (m=0x5d55e8) at sr_module.c:471 No locals. #9 0x0000000000430ce8 in init_mod (m=0x5d56a8) at sr_module.c:471 No locals. #10 0x0000000000430ce8 in init_mod (m=0x5d5768) at sr_module.c:471 No locals. #11 0x0000000000430e05 in init_modules () at sr_module.c:497 No locals. #12 0x0000000000416f3c in main (argc=3, argv=0x7fbffffa18) at main.c:1577 cfg_stream = (FILE *) 0x6cd010 c = -1 r = 4224491 tmp = 0x39d1014c40 "" tmp_len = 0 port = 0 proto = 0 options = 0x46d4e8 "f:cm:b:l:n:N:rRvdDETVhw:t:u:g:P:G:i:x:" ret = -1 seed = 4256706975 rfd = 7

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RE: [Users] Using subst() of Textops module
by Leonid Fainshtein 29 Jun '06

29 Jun '06

Thank you, Bogdan. I'm going to learn textops sources. Leonid -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Wednesday, June 28, 2006 8:25 PM To: Leonid Fainshtein Cc: users(a)openser.org Subject: Re: [Users] Using subst() of Textops module Hi Leonid, that wasn't a mistake - "t" si short format to "To". "\u" is a marker for current RURI - is not a pseudo variable, it's something strictly related to subst ops. Unfortunately this is not highly documented. regards, bogdan Leonid Fainshtein wrote: >Hi, >In the textops module doc >(http://openser.org/docs/modules/1.0.x/textops.html#AEN124) there is an >example that demonstrates how to replace URI in the header "To" with >the message URI: > > ># replace the uri in to: with the message uri (just an example) if ( >subst('/^To:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/t:\1\u\2/ig') ) {}; > > >After I've fixed a small mistake (instead of /t: I put /To:) it worked. >But I don't understand how does it work. It looks like "u" is >substituted by URI from R-URI but I didn't find such core variable. >Could somebody explain me? > >Thank you in advance, >Leonid Fainshtein > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

1 0

[Serusers] CDRTool / asterisk billing based
by Adrian Georgescu 29 Jun '06

29 Jun '06

Hi Concy, CDRTool is well documented, here are the links: http://download.dns-hosting.info/CDRTool/README.txt http://download.dns-hosting.info/CDRTool/INSTALL.txt http://download.dns-hosting.info/CDRTool/RATING.txt There is no free support from AG Projects because of limited time resources but it is functional, documented and licensed GPL without restrictions. There is a mailing list setup somewhere by a user of this mailing lists but I forgot where it is, if anyone knows I can add it to the documentation. Regards Adrian ============== You can try out these two tools TraBas and Istpp Message: 15 Date: Thu, 22 Jun 2006 17:53:30 +0200 (CEST) From: <hgaillac-sip at yahoo.fr> Subject: [Serusers] [Asterisk-Users] CDRTool / asterisk billing based on realtime To: asterisk-users at lists.digium.com Cc: serusers at iptel.org Message-ID: <20060622155330.80131.qmail at web25515.mail.ukl.yahoo.com> Content-Type: text/plain; charset=iso-8859-1 Hello, I read ag-projects it does'not provide support for its cdrtool pseudo gpl licensed . It's one of the bad open source project even seen . No documentation ...

3 2

[Serusers] waiting bug in "sc moni"
by Ramin NIkaeen 29 Jun '06

29 Jun '06

Friends, Can someone tell me what the "waiting" under the "Transaction Statistics" means when you execute "sc moni 1"? Does it show the number of SIP transactions in "waiting" state? What is this "waiting" state? It seems that there is a bug in my system because when there are currently no calls on SER, waiting shows some non-zero number and it continues to increase as time goes on and never goes back to zero. Does anyone know where this counter is decremented. I traced through the tm transaction module and it seems to me that in the function tm_stats, in file t_stats.h, "s_waiting[process_no]++" increments this counter. But where is decremented when a transaction is completed? I very much appreciate your help ramin ------------------------------------------------------------------------ ---------------------------------------- Here is the "sc moni 1" output on my SER server when there are no active calls: [cycle #: 1; if constant make sure server lives and fifo is on] Server: Sip EXpress router (0.9.2 (i386/linux)) Now: Wed Jun 28 16:49:26 2006 Up Since: Tue Jun 27 11:21:33 2006 Up time: 106073 [sec] Transaction Statistics Current: 0 (13 waiting) Total: 3436 (0 local)

3 2

[Serusers] Q registering ua using IP or fqdn?
by Mindaugas Kuprys 29 Jun '06

29 Jun '06

Hi, Intresting thing, I'm able to register UA then I'm using SER FQDN but not then using IP address in UA config? This should be so? Mindaugas

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[Serusers] 2 questions regarding user location
by Kyriakos 29 Jun '06

29 Jun '06

Hi, I have two questions regarding user location. 1) Is there a way of getting the user location info for manipulating it in ser.cfg ? Something like lookup but without rewriting anything. Basically what I want to do is check if the caller and the caller are NATed and belong to a certain same network then I should not use media proxy. The criterion for the same network would be that both sides are originating from the same public ip. Thus I would have to retrieve the "received" field value for the calee and compare it with the ip contained in the contact header field of the caller in the INVITE . Perhaps there is also another way to do this? 2) What happens when I have multiple phones registered under one user? How lookup(location) works when I have more than one ips in location table for the user? During tests I had a UA behind NAT registering to SER with an ip assigned from the dhcp. After a while I had the UA assigned a different ip from the dhcp and the Register on SER under the same user name. The problem was that during registration SER would respond back to the UA for requesting authorisation credentials but when it was time to send the 200 OK message for registration , it would send it only to the previous location of the UA and not the one it sent the REGISTER message from with of course failure to register because UA would not get the OK from SER. I noticed that in Contact header of the 200 message there were more than one return ips (the old one and the new one), Of course as soon as the old location timed out the registration would go ok. Thanks KM

2 3

[Users] Using # for Sip 2 Sip calls
by Kenny Chua 29 Jun '06

29 Jun '06

Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls? I came up with something like this: lookup("aliases"); if (uri=~"^sip:#[0-9]*@"){ xlog("Sip 2 SIP\n"); route(4); route(1); return; }; Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you. --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail Beta.

1 0

[Users] Last Hop Detection
by Tavis P 28 Jun '06

28 Jun '06

I'm trying to find a way to detect if a BYE message that is being routed between two proxies is at the last hop before being sent to the endpoint. I tried checking for a existence of the "Route" header field, but i gather its removed on the way out so is present until its routed I also tried comparing the value of the "Max-Forwards" header field to an arbitrary integer but i was unable to find a way to do that either The purpose is to avoid double accounting messages that traverse two proxies in our network, is anyone aware of a solution to this problem? tavis

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[Users] Re: [Devel] My TLS Compiling Error..Please help me...
by Bogdan-Andrei Iancu 28 Jun '06

28 Jun '06

Hi Ferianto, that's a quite know problem with some openssl packages - you need to disable the kerberos support into it, otherwise it will not compile properly. regards, bogdan Ferianto siregar wrote: > Dear all, > > I try to compile the TLS Support in my openser. I use > openser-1.0.1-tls and the operating system is Redhat. > I have done some commands like: > 1. localhost@root # export TLS=1 > 2. Uncomment (enable) TLS =1 in Makefile > 3. localhost@root # TLS=1 make all > > I do those task based on the tutorial that i have read from > www.openser.org/docs/tls.html. > But, when i run the make command, TLS=1 make all, there is an error > message. The error message is > > make: *** [main.o] Error 1 > > Does someone can help me to solve this problem? What should i do ? > Please help me.... > > NB: this is the message that i have got when run the make command > > >> In file included from /usr/include/openssl/ssl.h:179, > > >> from tls/tls_init.h:27, > > >> from main.c:112: > > >> /usr/include/openssl/kssl.h:72:18: krb5.h: No such file or directory > > >> In file included from /usr/include/openssl/ssl.h:179, > > >> from tls/tls_init.h:27, > > >> from main.c:112: > > >> /usr/include/openssl/kssl.h:132: parse error before "krb5_enctype" > > >> /usr/include/openssl/kssl.h:134: parse error before "FAR" > > >> /usr/include/openssl/kssl.h:135: parse error before '}' token > > >> /usr/include/openssl/kssl.h:147: parse error before > "kssl_ctx_setstring" > > >> /usr/include/openssl/kssl.h:147: parse error before '*' token > > >> /usr/include/openssl/kssl.h:148: parse error before '*' token > > >> /usr/include/openssl/kssl.h:149: parse error before '*' token > > >> /usr/include/openssl/kssl.h:149: parse error before '*' token > > >> /usr/include/openssl/kssl.h:150: parse error before '*' token > > >> /usr/include/openssl/kssl.h:151: parse error before "kssl_ctx_setprinc" > > >> /usr/include/openssl/kssl.h:151: parse error before '*' token > > >> /usr/include/openssl/kssl.h:153: parse error before "kssl_cget_tkt" > > >> /usr/include/openssl/kssl.h:153: parse error before '*' token > > >> /usr/include/openssl/kssl.h:155: parse error before "kssl_sget_tkt" > > >> /usr/include/openssl/kssl.h:155: parse error before '*' token > > >> /usr/include/openssl/kssl.h:157: parse error before "kssl_ctx_setkey" > > >> /usr/include/openssl/kssl.h:157: parse error before '*' token > > >> /usr/include/openssl/kssl.h:159: parse error before "context" > > >> /usr/include/openssl/kssl.h:160: parse error before > "kssl_build_principal_2" > > >> /usr/include/openssl/kssl.h:160: parse error before "context" > > >> /usr/include/openssl/kssl.h:163: parse error before > "kssl_validate_times" > > >> /usr/include/openssl/kssl.h:163: parse error before "atime" > > >> /usr/include/openssl/kssl.h:165: parse error before > "kssl_check_authent" > > >> /usr/include/openssl/kssl.h:165: parse error before '*' token > > >> /usr/include/openssl/kssl.h:167: parse error before "enctype" > > >> In file included from tls/tls_init.h:27, > > >> from main.c:112: > > >> /usr/include/openssl/ssl.h:909: parse error before "KSSL_CTX" > > >> /usr/include/openssl/ssl.h:931: parse error before '}' token > > >> make: *** [main.o] Error 1 > > > Thank you. > Sincerely yours, > > Ferianto > > ------------------------------------------------------------------------ > Do you Yahoo!? > Get on board. You're invited > <http://us.rd.yahoo.com/evt=40791/*http://advision.webevents.yahoo.com/handr…> > to try the new Yahoo! Mail Beta. > >------------------------------------------------------------------------ > >_______________________________________________ >Devel mailing list >Devel(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/devel > >

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[Users] Problem with forking
by Kenny Chua 28 Jun '06

28 Jun '06

Hi all, I was wondering what is the max children I can spawn. When I set children = 32 I got this error when doin openserctl moni: Error opening openser's FIFO /tmp/openser_fifo Make sure you have line fifo=/tmp/openser_fifo in your config Seems like openser or my system can't support that many children? I read that in the core cookbook that max children was 32. Wonder why it crashed on my system. Its a Dell Server, specs should be Pentium 4 3 Ghz with 1 GB ram. I think this spec should be enough to support 32 children. Here is the debug error. 51(4998) ERROR:domain_db_init: cannot initialize database connection 51(4998) 52(5000) new_connection: Too many connections 52(5000) ERROR: child_init(50): Error while connecting database 52(5000) ERROR:init_mod_child: Error while initializing module usrloc, rank 50 52(5000) init_children failed ERROR: domain:child_init(): Unable to connect to the database What am I suppose to do to fix it? I'm using mysql database. Is it because mysql needs to support more connection? How am I suppose to change that? Thank you --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

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[Users] My TLS Compiling Error..Please help me...
by Ferianto siregar 28 Jun '06

28 Jun '06

Dear all, I try to compile the TLS Support in my openser. I use openser-1.0.1-tls and The operating system is Redhat. I have done some commands like: 1. localhost@root # export TLS=1 2. Uncomment (enable) TLS =1 in Makefile 3. localhost@root # TLS=1 make all I do those task based on the tutorial that i have read from www.openser.org/docs/tls.html. But, when i run the make command, TLS=1 make all, there is an error message. The error message is make: *** [main.o] Error 1 Does someone can help me to solve this problem? What should i do ? Please help me.... NB: this is the message that i have got when run the make command >> In file included from /usr/include/openssl/ssl.h:179, >> from tls/tls_init.h:27, >> from main.c:112: >> /usr/include/openssl/kssl.h:72:18: krb5.h: No such file or directory >> In file included from /usr/include/openssl/ssl.h:179, >> from tls/tls_init.h:27, >> from main.c:112: >> /usr/include/openssl/kssl.h:132: parse error before "krb5_enctype" >> /usr/include/openssl/kssl.h:134: parse error before "FAR" >> /usr/include/openssl/kssl.h:135: parse error before '}' token >> /usr/include/openssl/kssl.h:147: parse error before "kssl_ctx_setstring" >> /usr/include/openssl/kssl.h:147: parse error before '*' token >> /usr/include/openssl/kssl.h:148: parse error before '*' token >> /usr/include/openssl/kssl.h:149: parse error before '*' token >> /usr/include/openssl/kssl.h:149: parse error before '*' token >> /usr/include/openssl/kssl.h:150: parse error before '*' token >> /usr/include/openssl/kssl.h:151: parse error before "kssl_ctx_setprinc" >> /usr/include/openssl/kssl.h:151: parse error before '*' token >> /usr/include/openssl/kssl.h:153: parse error before "kssl_cget_tkt" >> /usr/include/openssl/kssl.h:153: parse error before '*' token >> /usr/include/openssl/kssl.h:155: parse error before "kssl_sget_tkt" >> /usr/include/openssl/kssl.h:155: parse error before '*' token >> /usr/include/openssl/kssl.h:157: parse error before "kssl_ctx_setkey" >> /usr/include/openssl/kssl.h:157: parse error before '*' token >> /usr/include/openssl/kssl.h:159: parse error before "context" >> /usr/include/openssl/kssl.h:160: parse error before "kssl_build_principal_2" >> /usr/include/openssl/kssl.h:160: parse error before "context" >> /usr/include/openssl/kssl.h:163: parse error before "kssl_validate_times" >> /usr/include/openssl/kssl.h:163: parse error before "atime" >> /usr/include/openssl/kssl.h:165: parse error before "kssl_check_authent" >> /usr/include/openssl/kssl.h:165: parse error before '*' token >> /usr/include/openssl/kssl.h:167: parse error before "enctype" >> In file included from tls/tls_init.h:27, >> from main.c:112: >> /usr/include/openssl/ssl.h:909: parse error before "KSSL_CTX" >> /usr/include/openssl/ssl.h:931: parse error before '}' token >> make: *** [main.o] Error 1 Thank you. Sincerely yours, Ferianto --------------------------------- Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta.

1 0

[Users] how to t_replicate to multiple proxy
by Raymond Chen 28 Jun '06

28 Jun '06

Dear all, do anyone know how to t_replicate to multiple proxy, I tried putting two t_replicate statement back to back but it only replicate to the first proxy, not the second. t_replicate("sip:xxx.xxx.xxx.xxx:5060"); t_replicate("sip.yyy.yyy.yyy.yyy:5060"); thanks Ray

3 2

[Users] is OpenSER for me?
by Saeed Tariq 28 Jun '06

28 Jun '06

Hi, 1. I have few questions, I am using Cisco gateways to terminate and originate calls, Can I do the same with openSER? 2. I've installed openser 1.0.1 where with Mysql support, where can I do the configuration like routing etc. 3. From where I can get Web interface for openser 1.0.1 Thanks Best Regards.

2 1

[Users] more on accounting
by tele 28 Jun '06

28 Jun '06

Hi all, I'm using OpenSER v1.0.1 and i have done an accounting compatible with my billing system. Now i have problem with detection of flow direction for billing reason. i'm talking about openser 1.0.x: the problem is when the BYE come from callee. i know the problem is solved in openser1.1.x with the modparam("acc", "detect_direction", 1) i have read in modules documentation of 1.0.x that i can use is_direction() of RR module for detect call flow direction but it says that this must be called after loose_route() so for the accounting i must call setflag before loose_route() but what can i do if i want detect call flow before of exec setflag? Hmmm i think that i must install OpenSER 1.1.x and test with it :tele

1 1

[Users] Multiple Sockets + Multiple Processors: Any known issues?
by Carsten Bock 28 Jun '06

28 Jun '06

Hi Everybody, I have some problems with a System which has several network interfaces and multiple network interfaces. The system basically receives requests on the public socket/ip and sends them, via the private socket to some proxies, which "do the work". It's a kind of load-balancer based on OpenSER 1.0.1 and the Dispatcher Module from 1.1 / CVS. Everything works fine, but sometimes i see some ICMP Messages (Destination not reachable / Port not reachable) on this "loadbalancer"; as a response to requests from the Proxies to the loadbalancer. Has anybody seen this issue before? Any ideas, what's wrong? Any known issues? Are there any known problems with multiple interfaces? Any known issues with more than one processor? I've copied my configuration below, any thing in capital letters are Parameters from my M4 file... The system is not really on high load, has 64 children and should easily handle the load... (Dual Intel Xeon(a)2.80GHz, 1 Gig of Ram) Any help would be appreciated... Thanks in advance, Carsten debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=CHILD_PROCS check_via=yes dns=off rev_dns=off listen=DISPATCHER_EXTERNAL_IP listen=DISPATCHER_INTERNAL_IP port=5060 sip_warning=0 disable_core_dump=yes # Use the FIFO fifo="/tmp/openser_fifo" # for more info: sip_router -h # ------------------ module loading ---------------------------------- loadmodule "/lib/openser/modules/maxfwd.so" loadmodule "/lib/openser/modules/sl.so" loadmodule "/lib/openser/modules/tm.so" loadmodule "/lib/openser/modules/xlog.so" loadmodule "/lib/openser/modules/dispatcher.so" loadmodule "/lib/openser/modules/textops.so" loadmodule "/lib/openser/modules/rr.so" loadmodule "/lib/openser/modules/mccs_dispatcher.so" # ----------------- setting module-specific parameters --------------- # TM Module: Timeout for a request with no provisional Answers modparam("tm", "fr_timer", PROXY_TIMEOUT) # No double Record-Route; we do it manually modparam("rr", "enable_double_rr", 0) # Dispatcher: Forward Register Requests modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") # Dispatcher: Enable Failover-Support modparam("dispatcher", "flags", 2) route { ################################################################################################################## # ï¿1/2berprï¿1/2fung auf zuviele Forwards. ############################################################################################################### if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","To Many Hops"); drop(); return; }; ################################################################################################################## # Loose-Routing (RFC3261) ############################################################################################################### # Record-Route-Header entfernen if (loose_route()) { if(uri=~"sip:.*[@]INTERNAL_IP_REGEX") { # Send via the internal Interface: force_send_socket(DISPATCHER_INTERNAL_IP:5060); } else { # Send via the external Interface: force_send_socket(DISPATCHER_EXTERNAL_IP:5060); } if (!t_relay()) { sl_reply_error(); } return; } ################################################################################################################## # Record-Route, damit alle Nachrichten ï¿1/2ber diesen SIP-Proxy laufen ############################################################################################################### if (method == "INVITE") record_route(); ################################################################################################################## # Relay requests from the Gateways ############################################################################################################### if (OTHER_PROXY_QUERY) { if (uri != myself) { # Send via the external Interface: force_send_socket(DISPATCHER_EXTERNAL_IP:5060); # Und das Paket entsprechend weiterleiten if (!t_relay_to_udp("MCCS_GLOBAL_DISPATCHER_PRIMARY", "MCCS_GLOBAL_DISPATCHER_PRIMARY_PORT")) { log(1, "Not possible to relay to Dispatcher\n"); # Fehler melden sl_reply_error(); } return; } else { # Anfrage fï¿1/2r ein PSTN-Out-Gateway if (uri=~"sip:mg_out_.*") { # "mg_out_" entfernen strip(7); # PSTN-Out Mediagateway wï¿1/2hlen (Round-Robin) ds_select_domain("2", "4"); # Send via the external Interface: force_send_socket(DISPATCHER_EXTERNAL_IP:5060); # Anfrage fï¿1/2r einen Mediaserver } else if (uri=~"sip:media_.*") { # "media_" entfernen strip(6); # Mediaserver wï¿1/2hlen (Round-Robin) ds_select_domain("3", "4"); # Send via the internal Interface: force_send_socket(DISPATCHER_INTERNAL_IP:5060); } else { sl_send_reply("503", "Service not available"); drop(); return; } # On Failure: Choose next Proxy t_on_failure("1"); # Relay to the Proxy if (!t_relay()) { sl_reply_error(); } return; } } ################################################################################################################## # Routing zu den SIP-Proxys ############################################################################################################### # Send via the internal Interface: force_send_socket(DISPATCHER_INTERNAL_IP:5060); # Proxy wï¿1/2hlen (Round-Robin, SIP-Proxies) ds_select_domain("1", "4"); # On Failure: Choose next Proxy t_on_failure("1"); # Relay to the Proxy if (!t_relay()) { sl_reply_error(); } return; } failure_route[1] { # Choose another gateway if (t_check_status("408") && t_local_replied("all")) { if (ds_next_domain()) { # On Failure: Choose next Proxy t_on_failure("1"); # Relay to Proxy t_relay(); } else { t_reply("503", "Service not available"); return; } } }

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[Users] Ack is Losting .........
by raviprakash sunkara 28 Jun '06

28 Jun '06

Hi Users... The UAC's behind the NAT is hung up the softphones.. Automatically... I think Ack is Losting .. So the Session is Looses.. here ...... in this :----> nat_uac_test(paramter) what parameters should placed here.... 3 or 10 or 1 Same fo fix_naed_sdp().. --------------------------- opense,cfg.... route { route(2); if(!method=='ReGISTER') { record_route(); }; if(uri==myself) { ..... ...... .... } route(1); } route[1] { log("Testing-------> route[1] \n"); if (subst_uri('/(sip:.*);nat=yes/\1/')){ setflag(6); }; if (isflagset(5)||isflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ log("Testing-------> route[2] \n"); # force_rport(); if (nat_uac_test("19")) { log("Testing-------> nat_uac-test_19\n"); if (method=="REGISTER") { log("Testing-------> nat _ reg \n"); fix_nated_contact(); } else if (method=="INVITE") { log("Testing-------> nat_cantact \n"); fix_nated_sdp("1"); # fix_nated_contact(); }; force_rport(); setflag(5); }; } route[3] { log("Testing-------> route 3 \n"); if (is_method("BYE|CANCEL")) { log("Testing-------> Bye\n"); acc_db_request("Stop","acc"); unforce_rtp_proxy(); } else if (is_method("INVITE")){ log("Testing-------> Invite in route [3\n"); # fix_nated_sdp("3"); acc_db_request("Start","acc"); force_rtp_proxy(); t_on_failure("1"); } else if(is_method("ACK")) { log("Testing-------------------> ACK\n"); force_rtp_proxy(); }; if (isflagset(5)) { # fix_nated_sdp("3"); search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); }; t_on_reply("1"); } failure_route[1] { if (isflagset(6) || isflagset(5)) { acc_db_request("Stop","acc"); unforce_rtp_proxy(); } } onreply_route[1] { log("Testing-------> t_on_reply\n"); if ((isflagset(5) || isflagset(6)) && status=~"(183)|(2[0-9][0-9])" && (!search("^Content-Length:\ 0"))) { log("Testing ---> 183 200 \n"); force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); # if (isflagset(6)) { ## fix_nated_sdp("1"); # fix_nated_contact(); # } exit; } -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

2 1

[Users] Is there a way to leave a route block and return to it where you left?
by Brandon Price 28 Jun '06

28 Jun '06

I have a route block that performs callerid name lookups for incoming calls. I want to leave this as a separate route block, and not include it into the incoming route for simplicity of development. I basically want to hop out of route 2 down to route 101, then hop back to route 2 under the line that is pointed to. Any clues,? can this even be done or does the structure of the tm (being stateless) limit it? Is there a way to hop to a certain line number of a route like say route(2,4); going to route 2 line 4. I dunno. ############################################################### # INCOMING ############################################################### route[2] { xlog("INVITE Received (incoming): $fu -> $ru"); setflag(1); strip(1); route(101); <--- do more stuff ......................................... ######################################################################## ################################## # CNAM ######################################################################## ################################# route[101] { xlog("Running cnam query"); # do some stuff.... return; };

2 1

[Users] 404 Not found
by Kenny Chua 28 Jun '06

28 Jun '06

Hello all, I've registered my SPA 3000 on my openser server. It logs in and everythin works fine. However I was wondering why do I get this 404 Not found when I do ngrep -qt port 5060. What is causing this? And how do I get rid of it? U 2006/06/28 01:24:52.587457 192.168.1.101:5060 -> 192.168.1.6:5060 NOTIFY sip:192.168.1.6 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.101:5060;branch= z9hG4bK-9291d204..From: Kenny Sipura <sip:203@192.168.1.6>;tag=6824a583236f 93f1o0..To: <sip:192.168.1.6>..Call-ID: 3297ad4b-b2ceeec9@192.168.1.101..CS eq: 413 NOTIFY..Max-Forwards: 70..Event: keep-alive..User-Agent: Linksys/SP A3000-3.1.10(GWd)..Content-Length: 0.... U 2006/06/28 01:24:52.598448 192.168.1.6:5060 -> 192.168.1.101:5060 SIP/2.0 404 User Not Found..Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG 4bK-9291d204..From: Kenny Sipura <sip:203@192.168.1.6>;tag=6824a583236f93f1 o0..To: <sip:192.168.1.6>;tag=eb9822f1ed1b15a932bf98c097b3aaca.d130..Call-I D: 3297ad4b-b2ceeec9@192.168.1.101..CSeq: 413 NOTIFY..Server: OpenSer (1.0. 1 (i386/linux))..Content-Length: 0..Warning: 392 192.168.1.6:5060 "Noisy fe edback tells: pid=4969 req_src_ip=192.168.1.101 req_src_port=5060 in_uri=s ip:192.168.1.6 out_uri=sip:192.168.1.6 via_cnt==1".... **********Openser.cfg********************************** debug=3 fork=yes log_stderror=no listen=192.168.1.6 alias=test.com port=5060 children=32 dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" 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/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); t_relay(); exit; } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { xlog("Does not have to tag, FORBIDDEN\n"); sl_send_reply("403", "Forbidden"); exit; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); xlog("Forcing rtpport\n"); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); return; }; if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); xlog("Unforce RTP proxy2\n"); }; xlog("Dunno why forbidden\n"); sl_reply_error(); xlog("Forbidden4\n"); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); xlog("Setflag cos behind NAT\n"); } lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (uri=~"^sip:00[0-9]*@") { # International PSTN xlog("International PSTN\n"); strip(2); route(4); route(5); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); xlog("NAT Traversal Section\n"); } } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- xlog("Routing to PSTN gateway\n"); rewritehostport("pstn.gateway:5060"); avp_write("i:45", "inv_timeout"); route(1); xlog("Routed to PSTN gateway\n"); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

3 2

[Users] modules API
by Simon Morvan 28 Jun '06

28 Jun '06

Hello there, Simple question : What is the "rank" passed to the per-child module initialization function ? And why is it sometimes negative ? -- Simon Morvan.

2 1

[Serusers] CDRTool / asterisk billing based
by Concy Pereira 28 Jun '06

28 Jun '06

Hi Harry You can try out these two tools TraBas and Istpp Message: 15 Date: Thu, 22 Jun 2006 17:53:30 +0200 (CEST) From: <hgaillac-sip(a)yahoo.fr> Subject: [Serusers] [Asterisk-Users] CDRTool / asterisk billing based on realtime To: asterisk-users(a)lists.digium.com Cc: serusers(a)iptel.org Message-ID: <20060622155330.80131.qmail(a)web25515.mail.ukl.yahoo.com> Content-Type: text/plain; charset=iso-8859-1 Hello, I read ag-projects it does'not provide support for its cdrtool pseudo gpl licensed . It's one of the bad open source project even seen . No documentation ... A real bad project !! Don't waste time with that Anybody could advise me a billing system based on cdr records with realtime support cdr_obdc.so? Regards Harry

1 0

[Users] Can't rewritehostport to PSTN gateway
by Kenny Chua 28 Jun '06

28 Jun '06

Hi everybody, I'm having problems rewritehostport to PSTN gateway. It gives me the 403 Forbidden error. Here is the error: SEND TIME: 130027089 SEND >> ACK sip:0013373038435@test.com SIP/2.0 Via: SIP/2.0/UDP70.xxx.xxx.xxx:5060;rport;branch=z9hG4bK437FA12BFB124DA68EB569C02212327D From: Kenny <sip:200@test.com>;tag=1187743606 To: <sip:0013373038435@test.com>;tag=eb9822f1ed1b15a932bf98c097b3aaca.befd Contact: <sip:200@70.xxx.xxx.xxx:5060> Call-ID: 67A1BB1A-C737-4C24-B7AF-19E9C92C2B47(a)192.168.1.30 CSeq: 9401 ACK Max-Forwards: 70 Content-Length: 0 SEND TIME: 130027095 SEND >> 70.xxx.xxx.xxx:5060 INVITE sip:0013373038435@test.com SIP/2.0 Via: SIP/2.0/UDP70.xxx.xxx.xxx:5060;rport;branch=z9hG4bKF16D7EAC63D6463A8624A802F5BB4908 From: Kenny <sip:200@test.com>;tag=1187743606 To: <sip:0013373038435@test.com> Contact: <sip:200@70.xxx.xxx.xxx:5060> Call-ID: 67A1BB1A-C737-4C24-B7AF-19E9C92C2B47(a)192.168.1.30 CSeq: 9402 INVITE Proxy-Authorization: Digest username="200",realm="test.com",nonce="44a21eb77c572fc09155f89521700490cb8a87ef",response="75921f5f225efeaeb0177e3353769190",uri="sip:0013373038435@test.com" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105x Content-Length: 308 v=0 o=200 130026958 130027078 IN IP470.xxx.xxx.xxx s=X-Lite c=IN IP4 70.xxx.xxx.xxx t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv RECEIVE TIME: 130027172 RECEIVE << 70.xxx.xxx.xxx:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 70.xxx.xxx.xxx:5060;rport=5060;branch=z9hG4bKF16D7EAC63D6463A8624A802F5BB4908;received=192.168.1.1 From: Kenny <sip:200@test.com>;tag=1187743606 To: <sip:0013373038435@test.com> Call-ID: 67A1BB1A-C737-4C24-B7AF-19E9C92C2B47(a)192.168.1.30 CSeq: 9402 INVITE Server: OpenSer (1.0.1 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.6:5060 "Noisy feedback tells: pid=4989 req_src_ip=192.168.1.1 req_src_port=5060 in_uri=sip:0013373038435@test.com out_uri=sip:13373038435@pstn.gateway:5060 via_cnt==1" RECEIVE TIME: 130027371 RECEIVE << 70.xxx.xxx.xxx:5060 SIP/2.0 403 Forbidden To: <sip:0013373038435@test.com>;tag=3360460030-876466 From: Kenny <sip:200@test.com>;tag=1187743606 Call-ID: 67A1BB1A-C737-4C24-B7AF-19E9C92C2B47(a)192.168.1.30 CSeq: 9402 INVITE Contact: sip:13373038435@65.xxx.xxx.xxx:5060 Via: SIP/2.0/UDP 70.xxx.xxx.xxx:5060;received=192.168.1.1;rport=5060;branch=z9hG4bKF16D7EAC63D6463A8624A802F5BB4908 Content-Length: 0 SEND TIME: 130027372 SEND >>70.xxx.xxx.xxx:5060 ACK sip:0013373038435@test.comSIP/2.0 Via: SIP/2.0/UDP 70.xxx.xxx.xxx:5060;rport;branch=z9hG4bKF16D7EAC63D6463A8624A802F5BB4908 From: Kenny <sip:200@test.com>;tag=1187743606 To: <sip:0013373038435@test.com>;tag=3360460030-876466 Contact: <sip:200@70.xxx.xxx.xxx:5060> Call-ID: 67A1BB1A-C737-4C24-B7AF-19E9C92C2B47(a)192.168.1.30 CSeq: 9402 ACK Max-Forwards: 70 Content-Length: 0 ********************************Openser.cfg******************************************** debug=3 fork=yes log_stderror=no listen=192.168.1.6 alias=test.com port=5060 children=32 dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" 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/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); t_relay(); exit; } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { xlog("Does not have to tag, FORBIDDEN\n"); sl_send_reply("403", "Forbidden"); exit; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); xlog("Forcing rtpport\n"); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); return; }; if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); xlog("Unforce RTP proxy2\n"); }; xlog("Dunno why forbidden\n"); sl_reply_error(); xlog("Forbidden4\n"); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); xlog("Setflag cos behind NAT\n"); } lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (uri=~"^sip:00[0-9]*@") { # International PSTN xlog("International PSTN\n"); strip(2); route(4); route(5); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); xlog("NAT Traversal Section\n"); } } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- xlog("Routing to PSTN gateway\n"); rewritehostport("pstn.gateway:5060"); avp_write("i:45", "inv_timeout"); route(1); xlog("Routed to PSTN gateway\n"); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

2 1

[Users] Hung up problem
by raviprakash sunkara 28 Jun '06

28 Jun '06

Hi Users,,,, small issues in openser.. UAC 1 , UAC 2 are behind the NAT.. UAC 3 in NAT When I Make call UAC 2 <=================> UAC 1 , After 32 second its automatically hung up the Softphone. Same like .......... whe n UAC 1 <====================================> UAC 2 . When stranger interesting to all Users in OPENSER Developers in this world.. UAC 3 <============================> UAC 1 . it doesn't Hung up the phone But voice is not Hearing .. and UAC 1 <======================> UAC 3 its Hung up automatically ..... voice is clear... listening .. Where will be the problem either in softphone ( x-lite 3,0) or openser.cfgnat .... I thing the TM parameters is holding the Time . to process the call ... please help me................ -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

2 1

[Serusers] Anyway to steal Media IP/Port from RTPProxy or MediaProxy
by Hoa Thai Duy 28 Jun '06

28 Jun '06

Hi List I want to get to c= and m= value after use_media_proxy or force_rtp_proxy (after real RTP flow between UA and media/rtpproxy) I want this in order to steal this pair of information, and bypass the RTPProxy/MediaProxy and use this information for UA to talk with other application server (eg. Asterisk) Pls. help Brgds

2 5

[Serusers] compatibility with openser modules
by Andy Thomas 28 Jun '06

28 Jun '06

Does anyone know if I can use openser modules with ser? For example I would like to use the UAC module from openser in my ser 0.9.6 installation

2 1

[Users] how i start AVPops module implementation
by Tanzeel serfaraz 28 Jun '06

28 Jun '06

Hi all users; plz help me out how can i start AVPops module implementation using OPENSER 1.0.0- thanks, Tanzeel --------------------------------- Sneak preview the all-new Yahoo.com. It's not radically different. Just radically better.

2 1

[Serusers] [Fwd: query about radius]
by olivier.taylor 27 Jun '06

27 Jun '06

Please stop to annoy people by contacting them thru google talk and private mail Mr Aman Sachdeva. Seems that i am not alone to complain about that. Olivier

1 0

[Serusers] P-Asserted-Identity
by Adrian Georgescu 27 Jun '06

27 Jun '06

Hi Dani, The closest idea I have in mind is to load into an avp the result of a mysql query, which returns a random number or entries from a database sorted randomly. But for this you to work you would need to use OpenSER devel branch, not sure if equivalent SER functionality exists. See http://openser.org/docs/modules/devel/avpops.html#AEN268 Adrian >>>>>>> Like Adrian said: i'm not intend to test any IMS platform. So, Adrian, do you have any ser module witch can do that ? Dani

2 1

Re: [Users] keep alive using RE-INVITE
by Javier Ramirez 27 Jun '06

27 Jun '06

Bogdan, I don't understand how to make this with openser. but I understand the keepalive mechanism can you help me? best regards Javier Ramirez ----- Original Message ----- From: Bogdan-Andrei Iancu To: Javier Ramirez Sent: Wednesday, June 21, 2006 1:35 PM Subject: Re: [Users] keep alive using RE-INVITE Javier, I guess the RFC3261 or google (SIP + reINVITE) are the best source for looking into. regards, bogdan Javier Ramirez wrote: > Bogdan, > I read that a lot of person use the RE-INVITE to keepalive the call > but I don´t understand this. > please, can you help me ? > best regards > Javier Ramirez > > > ----- Original Message ----- > *From:* Bogdan-Andrei Iancu <mailto:bogdan@voice-system.ro> > *To:* Javier Ramirez <mailto:jvr_78@yahoo.com.ar> > *Cc:* users(a)openser.org <mailto:users@openser.org> > *Sent:* Thursday, June 08, 2006 7:14 AM > *Subject:* Re: [Users] keep alive using RE-INVITE > > Hi Javier, > > this is quite impossible to do - by sending re-INVITEs from a middle > point, you will break the CSEQ consistency between the end points. > Within the dialog, the cseq must increase from message to message > - if > you send a re-invite to point A with cseq n+1, after a while point > B may > sent a request with the same cseq since it was not aware that some > party > in the middle changed the number. > > the best way to do it is if you have a end2end keepalive mechanism as > Session-Timer. > > regards, > bogdan > > Javier Ramirez wrote: > > > Hi ! > > I looking for a mechanism to keep alive, and I see that is used the > > RE-INVITE, but > > How to use the Re-INVITE to a keep alive mechanism ? > > exist another way to implement another keep alive mechanism ? > > > > best regards > > Javier ramirez > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > __________ NOD32 1.1586 (20060608) Information __________ > > This message was checked by NOD32 antivirus system. > http://www.eset.com > __________ NOD32 1.1611 (20060620) Information __________ This message was checked by NOD32 antivirus system. http://www.eset.com

2 3

[Users] Question
by mmathur 27 Jun '06

27 Jun '06

hi.. i want to know that from where i can get an Openser Development kit... waiting for your response... Thanks regards, Mayank Mathur

4 6

[Users] difference between the rtpproxy and mediaproxy
by raviprakash sunkara 27 Jun '06

27 Jun '06

Hi Users,, Till I'm in confuse ,,, That I use openser , UAC are Behind NAT and Not Behind NAT . I'm Using the rtpproxy... But Confuse here which is one the following the Besr . Is rtpproxy or mediaproxy... In here ... When I using rtpproxy... 1) openser server is not requesting and respondes the 'BYE " 2 ) two back the Voice is clearly listen in softphones and after that voice is hearing Please.. -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

2 1

[Users] tm-module status rewrite
by Andreas Granig 27 Jun '06

27 Jun '06

Hi, Under heavy load, I sometimes get log messages like these: ERROR:tm:t_should_relay_response: status rewrite by UAS: stored: 408, received: 100 ERROR:tm:t_should_relay_response: status rewrite by UAS: stored: 408, received: 180 It doesn't seem to be critical because the calls are established successfully, so is this really an error? If not, can we decrease the log priority or even only print it in debug mode? Andy

2 1

RE: [Users] Reload Config-File
by Douglas Garstang 27 Jun '06

27 Jun '06

> -----Original Message----- > From: martin(a)campus-merseburg.de [mailto:martin@campus-merseburg.de] > Sent: Friday, April 28, 2006 5:40 AM > To: users(a)openser.org > Subject: Re: [Users] Reload Config-File > > > > Am 28.4.2006 schrieb "martin" <martin(a)campus-merseburg.de>: > > > > >Is there a way to reload the configfile of openser without > restarting it? > >the hup signal doesnt work. > > > > Ok. Nevermind. > Q: Is it possible to reload OpenSER's configuration file with > a signal? > > A: No. To reload the configuration file, OpenSER must be restarted. > > Would be a nice "carrier-grade" feature. I second that. It would definitely be nice.

2 1

[Serusers] P-Asserted-Identity
by Adrian Georgescu 27 Jun '06

27 Jun '06

The test is against regulatory bodies in some countries that allow only certain CLIs to be presented. IMSs is step farther away in the pipeline Adrian ///////// Hi Dani, Im not entirely sure about that random number problem you are trying to solve, but I think it's not currently possible. You may try looking into avops module though... Yet I have another couple of questions: 1. Are you trying to test an IMS platform? 2. If, yes, then which one? Andrey.

2 1

[Users] isdsturiset
by OpenSer Users 27 Jun '06

27 Jun '06

hello, I am interested to know if I can test for the IP address ot the "next hoop" after loose_route(). I see that isdsturiset() does not take any parameter. for the moment I use something like if(search("^Route:.*IP_ADDRESS_TO_SEARCH_FOR.*") { ... do something here ... } is there an another, nicer, way to do this ? thanks, razvan radu

1 0

[Users] Not hung upping the softphones...
by raviprakash sunkara 27 Jun '06

27 Jun '06

Hi , User,,, great issue to me ... please users help into my issue 1) the softphones are not hung upping ... Disconnecting... that means Behind the Nat client send Bye request .. But openser server is responseing the Bye .. and Ack to server... my openser.cfg route{ log("Testing-------> Starting\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; log("Testing-------> NAT detection into route 2 \n"); route(2); if (!method=="REGISTER") { log("Testing-------> method != REG\n"); record_route(); }; if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { log("Testing-------> uri==my\n"); if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (isflagset(5)) { setflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); }; save("location"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found...... ! "); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { log("Testing-------> route[1] \n"); if (subst_uri('/(sip:.*);nat=yes/\1/')){ setflag(6); }; if (isflagset(5)||isflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ log("Testing-------> route[2] \n"); force_rport(); if (nat_uac_test("19")) { log("Testing-------> nat_uac-test_19\n"); if (method=="REGISTER") { log("Testing-------> nat _ reg \n"); fix_nated_register(); } else { log("Testing-------> nat_cantact \n"); fix_nated_contact(); }; setflag(5); }; } route[3] { log("Testing-------> route 3 \n"); if (is_method("BYE|CANCEL")) { log("Testing-------> Bye\n"); acc_db_request("Stop","acc"); unforce_rtp_proxy(); } else if (is_method("INVITE")){ log("Testing-------> Invite in route [3\n"); acc_db_request("Start","acc"); force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isflagset(6) || isflagset(5)) { acc_db_request("Stop","acc"); unforce_rtp_proxy(); } } onreply_route[1] { log("Testing-------> t_on_reply\n"); if ((isflagset(5) || isflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isflagset(6)) { fix_nated_contact(); } exit; } -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

1 0

R: [Users] CANCEL
by D'Addelfio Davide 27 Jun '06

27 Jun '06

Thanks, now it works fine!!! Davide -----Messaggio originale----- Da: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Inviato: lunedì 26 giugno 2006 20.08 A: Norman Brandinger Cc: users(a)openser.org; D'Addelfio Davide Oggetto: Re: [Users] CANCEL Hi, see http://openser.org/docs/modules/stable/acc.html#AEN239. regards, bogdan Norman Brandinger wrote: > Hi, > > Below is a little <snip></snip> of my config. > > Note that the "define's" are preprocessed by M4. I found that as the > config file grows in size, using human readable names in place of > numbers greatly enhances productivity. > > By the way, did you check out: > http://openser.org/docs/modules/stable/acc.html ? > > To quote: > > 1.5.4. |report_cancels| (integer) > By default, CANCEL reporting is disabled -- most accounting > applications are happy to see INVITE's cancellation status. Turn on if > you explicitly want to account CANCEL transactions. > > Regards, > Norm > > ... > define(`YES', 1) > define(`FLAG_ACC', 6) > define(`FLAG_ACC_DB', 7) > ... > modparam("acc", "report_cancels", YES) > modparam("acc", "db_flag", FLAG_ACC_DB) > modparam("acc", "log_flag", FLAG_ACC) > ... > #---------------------------------------------------------------- > # Setup Accounting > #---------------------------------------------------------------- > if (is_method("BYE|REFER|CANCEL")) { > setflag(FLAG_ACC); > setflag(FLAG_ACC_DB); > }; > ... > > > D'Addelfio Davide wrote: > >> >> Hi to all, >> >> i can't understand why my openser doesn't capture the Cancel and >> doesn't store it into database, could anyone help me in some way? >> Please, I'm desperate...!!! >> >> Attached my cfg file. >> >> Thanks >> >> Davide >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> ------------------------------------------------------------------------ >> >> No virus found in this incoming message. >> Checked by AVG Free Edition. >> Version: 7.1.394 / Virus Database: 268.9.3/374 - Release Date: 6/23/2006 > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[Users] Openser password change
by Kenny Chua 27 Jun '06

27 Jun '06

Hello everybody, Have anyone of you changed the openser and openserro password? I've actually changed it in mysql. And then of course changed it in my openser.cfg. However after that, everything just fell apart. What else am I missing? If I do change those 2 passwords, where else am I suppose to change it? Another question is, have anyone of you ever had the problem where openser just don't want to start. I've found out it's not starting whenever I typed openserctl moni or ps. It tells me openser_fifo not found. Then I use monit to restart it. Still nothing happens. Created a fake openser_fifo in /tmp, still nothing worked. After awhile of leaving it on, openser finally decides to start. Not sure what triggered it. This problem just happens sometimes. My system is, CentOS 4.3, openser 1.0.1, rtpproxy, monit 4.8.1. Thank you. --------------------------------- Ring'em or ping'em. Make PC-to-phone calls as low as 1¢/min with Yahoo! Messenger with Voice.

3 2

[Serusers] accounting question
by Nhadie 27 Jun '06

27 Jun '06

Hi All, Should I also account BYE's with sip_status not equal to 200? if i don't i might lose some minutes that I will alreadybe billed by the gteway and if I do customers might tell me that those calls are not completed. Hope you guys can help me. Thanks Regards, Nhadie ________________________________________________ Message sent using UebiMiau 2.7

2 1

[Serusers] Assure a BYE is received ??
by Ryan Pagquil 27 Jun '06

27 Jun '06

Hi, How can I be sure that all calls are terminated by a BYE message? I have some instances that a BYE message is not logged by SER. Is there a way to fix this? What could be the possible scenarios that causes missing BYE's? Thanks, Ryan

3 3

Re: [Serusers] How do I get rid of npdi=yes?
by Jiri Kuthan 26 Jun '06

26 Jun '06

At 15:43 24/06/2006, John Clements wrote: >> You have a broken UA. > >Where do you get that? Nothing is broken, its the Number Porting flag the >TDM sends. If anything, Asterisk and SER are broken because they do not >recognize it :-) Hi John, I am wondering why you think that SER is broken? IMHO, as Juha pointed out your URI may be a 'valid' result of standard-recommended way of translating Tel to Sip URIs (Rfc3261, 19.1.6). Now since the same specification does not syntactically recognize parameters with the userpart of a URI, your username is matched in usr-loc database against what it is (some number, followed by semicolon, ndpdi and some more characters). I agree that most likely such lookup does not make much sense (unless you really have usernames with semicolon in it :-))). To me, the recommendation in rfc3261 seems not entirely thought thru -- one obviously didn't want to mix one parameter set with another. However processing the username-embedded parameters hasn't been addressed, making it as a result a subject of local URI policy. SER's subst_uri can fix it for you, i.e., implement your policy 'throw all parameters from userpart', -jiri p.s. -- recall my previous comment about lack of domain, I just 'misparsed' at some of the characters in the URI. >And its subst_uri() that gets rid of it. I found it last night. Thanks >though. > > -John > > >> AFAIK it's not possible with From. SER processes messages and always >> keep the original message. When you do subst etc, you basically tell SER >> how to change the message before sending a reply or forwarding. The ruri >> can be changed (but still the original uri is kept, i.e. revert_uri()) >> The lookup function will use the From in the original message, >> regardless of what you do. >> You have a broken UA. >> g-) >> >> John Clements wrote: >>> I've looked around and have found one or two other posts asking about >>> this, but I have never seen an answer. The From is: >>> >>> =uri: <sip:5551212;npdi=yes@x.x.x.x:5060;dtg=SIP;user=phone> >>> >>> Which causes a number of issues. USRLOC no longer works: >>> >>> =lookup(): '5551212;npdi=yes' Not found in usrloc >>> >>> And when you send the call to an Asterisk server, Asterisk wigs out >>> because it truncates everything after the first ";" it encounters in the >>> URI. Does anyone have any idea how to get rid of it? I've tried subst(), >>> but could not get it to work. :-( >>> >>> >>> -John >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >> >> > > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

Re: [Users] "513 Message too big" for the Invite
by Bogdan-Andrei Iancu 26 Jun '06

26 Jun '06

Hi Reddy, what you mean by shutting down? any logs? regards, bogdan Reddy, Muralidhar wrote: >Hello Bogdan, > > Thanks for your help. > It is working now. > > But after 8-10 calls Proxy shuts down. Do I need to increase the >memory? > >Thanks, >Murali >Phone: 301-515-3716 >e-mail: mreddy(a)ti.com > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Monday, June 26, 2006 5:37 AM >To: Reddy, Muralidhar >Cc: users(a)openser.org >Subject: Re: [Users] "513 Message too big" for the Invite > >Hi Reddy, > >probably the message exceeds the 4096 bytes configured in the script. >try to increase the value or to disable the check (# if......#endif does > >not work - try # or // for each line or the C like /*...*/)... > >regards, >bogdan > >Reddy, Muralidhar wrote: > > > >>Hello, >> >>Everything works fine if I run Proxy without TLS disabled. >> >>I tried both TCP/UDP, it works fine. >> >>For TLS, Registration went fine however Proxy responds "513 Message >>too big" to the Invite. >> >>I don't see any wrong headers in the message. >> >>The length is almost same for TCP and TLS invite message, the only >>change is sip and sips. >> >>Any 'C' file or config file I need to modify? >> >>

1 0

[Users] internal server error " 500 ".
by raviprakash sunkara 26 Jun '06

26 Jun '06

Hi users, I'm cracking my head with "500 :-I'm terribly sorry, server error occurred (1/SL) " . What's mean this... When I trying to calls from softphone to softphone " its says above message '. openser server ip 192.168.2.4 sip -domain is xxx.xxx.xxx.xxx softphones ip are 192.168.2.21 and 20 which behind the NAT ( i.e not in server location ,its some where place ) please check the marked one in debug. SIP Request: 0(29266) method: <INVITE> 0(29266) uri: <sip:104@24.98.157.209> 0(29266) version: <SIP/2.0> 0(29266) parse_headers: flags=2 0(29266) Found param type 235, <rport> = <n/a>; state=6 0(29266) Found param type 232, <branch> = <z9hG4bK1EA935B35FA14629985951EBA8B3F547>; state=16 0(29266) end of header reached, state=5 0(29266) parse_headers: Via found, flags=2 0(29266) parse_headers: this is the first via 0(29266) After parse_msg... 0(29266) preparing to run routing scripts... 0(29266) LOG:- step1 0(29266) parse_headers: flags=100 0(29266) DEBUG:parse_to:end of header reached, state=9 0(29266) DEBUG: get_hdr_field: <t> [25]; uri=[sip:104@xxx.xxx.xxx.xxx] 0(29266) DEBUG: to body [<sip:104@xxx.xxx.xxx.xxx> ] 0(29266) get_hdr_field: cseq <CSeq>: <3472> <INVITE> 0(29266) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(29266) parse_headers: flags=200 0(29266) DEBUG: get_hdr_body : content_length=149 0(29266) found end of header 0(29266) find_first_route: No Route headers found 0(29266) loose_route: There is no Route HF 0(29266) LOG:- method not reg 0(29266) DEBUG: add_param: tag=612182576 0(29266) DEBUG:parse_to:end of header reached, state=29 0(29266) parse_headers: flags=200 0(29266) find_first_route: No Route headers found 0(29266) loose_route: There is no Route HF 0(29266) grep_sock_info - checking if host==us: 13==11 && xxx.xxx.xxx.xxx] == [192.168.2.4] 0(29266) grep_sock_info - checking if port 5060 matches port 5060 0(29266) grep_sock_info - checking if host==us: 13==13 && xxx.xxx.xxx.xxx] == [xxx.xxx.xxx.xxx] 0(29266) grep_sock_info - checking if port 5060 matches port 5060 0(29266) LOG:- uri method myself 0(29266) LOG:- INVITE & ACK IN URI 0(29266) parse_headers: flags=ffffffffffffffff 0(29266) DEBUG:check_content_type: type <application/sdp> found valid 0(29266) record_route(): Double attempt to record-route 0(29266) parse_headers: flags=58 0(29266) parse_headers: flags=ffffffffffffffff 0(29266) DEBUG:check_content_type: type <application/sdp> found valid 0(29266) parse_headers: flags=40 0(29266) force_rtp_proxy2: proxy reply: 35002 0(29266) DEBUG: t_newtran: msg id=5 , global msg id=0 , T on entrance=0xffffffff 0(29266) parse_headers: flags=ffffffffffffffff 0(29266) parse_headers: flags=78 0(29266) t_lookup_request: start searching: hash=43828, isACK=0 0(29266) DEBUG: RFC3261 transaction matching failed 0(29266) DEBUG: t_lookup_request: no transaction found 0(29266) DBG: trans=0xb6123100, callback type 1, id 0 entered 0(29266) DEBUG:tm:t_relay: new INVITE 0(29266) parse_headers: flags=ffffffffffffffff 0(29266) check_via_address(61.17.248.68, 192.168.2.21, 0) 0(29266) WARNING:vqm_resize: resize(0) called 0(29266) DEBUG:tm:_reply_light: reply sent out. buf=0x81213a0: SIP/2.0 1..., shmem=0xb61247a8: SIP/2.0 1 0(29266) DEBUG:tm:_reply_light: finished 0(29266) DEBUG: mk_proxy: doing DNS lookup... 0(29266) parse_headers: flags=1000 0(29266) parse_headers: flags=ffffffffffffffff 0(29266) clen_builder: content-length: 187 (187) 0(29266) check_via_address(61.17.248.68, 192.168.2.21, 0) 0(29266) DEBUG: add_to_tail_of_timer[4]: 0xb612321c 0(29266) DEBUG: add_to_tail_of_timer[0]: 0xb612322c 0(29266) SER: new transaction fwd'ed 0(29266) rewrite_uri: Rewriting Request-URI with 'sip:104@192.168.2.20 :5060' 0(29266) lookup(): '104' Not found in usrloc 0(29266) grep_sock_info - checking if host==us: 12==11 && [192.168.2.20] == [192.168.2.4] 0(29266) grep_sock_info - checking if port 5060 matches port 5060 0(29266) grep_sock_info - checking if host==us: 12==13 && [192.168.2.20] == [xxx.xxx.xxx.xxx] 0(29266) grep_sock_info - checking if port 5060 matches port 5060 0(29266) grep_sock_info - checking if host==us: 12==11 && [192.168.2.20] == [192.168.2.4] 0(29266) grep_sock_info - checking if port 5060 matches port 5060 0(29266) grep_sock_info - checking if host==us: 12==13 && [192.168.2.20] == xxx.xxx.xxx.xxx] 0(29266) grep_sock_info - checking if port 5060 matches port 5060 0(29266) check_self: host != me 0(29266) LOG:_ uri in uri myself 0(29266) parse_headers: flags=ffffffffffffffff 0(29266) DEBUG: t_newtran: msg id=5 , global msg id=5 , T on entrance=0xb6123100 0(29266) ERROR: t_newtran: transaction already in process 0xb6123100 0(29266) parse_headers: flags=ffffffffffffffff 0(29266) check_via_address(61.17.248.68, 192.168.2.21, 0) 0(29266) ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) 0(29266) LOG:- ENeter into route 1 0(29266) DEBUG: t_newtran: msg id=5 , global msg id=5 , T on entrance=0xb6123100 0(29266) ERROR: t_newtran: transaction already in process 0xb6123100 0(29266) parse_headers: flags=ffffffffffffffff 0(29266) check_via_address(61.17.248.68, 192.168.2.21, 0) 0(29266) ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) 0(29266) DEBUG:destroy_avp_list: destroying list (nil) 0(29266) receive_msg: cleaning up 0(29266) SIP Request: 0(29266) method: <ACK> 0(29266) uri: <sip:104@24.98.157.209> 0(29266) version: <SIP/2.0> 0(29266) parse_headers: flags=2 0(29266) Found param type 235, <rport> = <n/a>; state=6 0(29266) Found param type 232, <branch> = <z9hG4bK1EA935B35FA14629985951EBA8B3F547>; state=16 0(29266) end of header reached, state=5 0(29266) parse_headers: Via found, flags=2 0(29266) parse_headers: this is the first via 0(29266) After parse_msg... 0(29266) preparing to run routing scripts... 0(29266) parse_headers: flags=8 0(29266) DEBUG: add_param: tag=30901ad8c164fe584bedac02134d4858.65cc 0(29266) DEBUG:parse_to:end of header reached, state=29 0(29266) DEBUG: get_hdr_field: <t> [67]; uri=[sip:104@24.98.157.209] 0(29266) DEBUG: to body [<sip:104@24.98.157.209>] 0(29266) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 0(29266) DEBUG:destroy_avp_list: destroying list (nil) 0(29266) receive_msg: cleaning up 0(29266) SIP Request: 0(29266) method: <ACK> 0(29266) uri: <sip:104@24.98.157.209> 0(29266) version: <SIP/2.0> 0(29266) parse_headers: flags=2 0(29266) Found param type 235, <rport> = <n/a>; state=6 0(29266) Found param type 232, <branch> = <z9hG4bK1EA935B35FA14629985951EBA8B3F547>; state=16 0(29266) end of header reached, state=5 0(29266) parse_headers: Via found, flags=2 0(29266) parse_headers: this is the first via 0(29266) After parse_msg... 0(29266) preparing to run routing scripts... 0(29266) parse_headers: flags=8 0(29266) DEBUG: add_param: tag=30901ad8c164fe584bedac02134d4858.65cc 0(29266) DEBUG:parse_to:end of header reached, state=29 0(29266) DEBUG: get_hdr_field: <t> [67]; uri=[sip:104@24.98.157.209] 0(29266) DEBUG: to body [<sip:104@24.98.157.209>] 0(29266) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 0(29266) DEBUG:destroy_avp_list: destroying list (nil) ............................................................................... openser.cfg route { log("LOG:- step1\n"); if(!mf_process_maxfwd_header("13")) { sl_send_reply("483","Too many hops are there "); exit; }; if(msg:len>max_len) { sl_send_reply("513","Messages are Too Big ... !"); exit; }; # loose_route(); if(!method=="REGISTER") { log("LOG:- method not reg \n"); record_route(); }; if(loose_route()) { log("LOG:- loose route\n"); acc_rad_request("200"); t_relay(); exit; }; if(uri==myself) { log("LOG:- uri method myself\n"); if(method=="REGISTER") { log("LOG:- REG IN URI MY SELF \n"); if(!www_authorize("xxx.xxx.xxx.xxx.","subscriber")) { log("LOG :- REG IN URI AUTH\n"); www_challenge("xxx.xxx.xxx.xxx","0"); exit; }; save("location"); setflag(6); force_rport(); exit; }; if(method=="INVITE"||method=="ACK") { log("LOG:- INVITE & ACK IN URI \n"); # if (uri =~ "^sip:1].@*"){ # log("LOG:-rewrite port and host\n"); log("LOG:- Aftere the rewite \n "); fix_nated_sdp("1"); record_route(); acc_rad_request("Start"); force_rtp_proxy(); t_on_reply("1"); t_relay(); }; if(method=="BYE"||method=="CANCEL") { setflag(2); acc_rad_request("200"); unforce_rtp_proxy(); }; if(!lookup("location")) { log("LOG:- NOT IN LOCATION \n"); sl_send_reply("404","sorry guy ... urs friend Not found"); exit; }; }; lookup("aliases"); # if (!uri==myself) { # log("LOG:_ uri in uri myself \n"); # append_hf("P-hint: outbound alias\r\n"); # route(1); # }; if(!t_relay()) { sl_reply_error(); }; # lookup("aliases"); # exit; route(1); # exit; } route[1] { log("LOG:- ENeter into route 1\n"); if(isflagset(6)) { force_rtp_proxy(); }; t_on_reply("1"); if(!t_relay()) { sl_reply_error(); }; } onreply_route[1] { log("LOG :- reply_route \n"); if(status=~"2[0-9][0-9]" && isflagset(6)) { log("LOG:- staus 183| 200 \n"); fix_nated_contact(); force_rtp_proxy(); }; } -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

2 1

[Users] Problems with reverse DNS resolving in comp_ip
by Andreas Sikkema 26 Jun '06

26 Jun '06

Hi, Today we had an issue on one of our openser machines. Some CPEs that were trying to register came from an IP address that for some reason would not (or very slowly, or both) resolve to a DNS name. Therefore openser was very slow to respond to any incoming SIP message, resulting in loads of unreachable customers. About one third of the registration attempts was succeeding and I think hardly any call attempts succeeded. What lead us to the DNS problem was that we increased the loglevel from 3 (our normal loglevel) to 7 and we noticed lots of "comp_ip: could not rev_resolve ip address xxx.xxx.xxx.xxx" messages. Why is the function eval_elem() in route.c calling comp_ip which in turn wants to resolve an ip address? Is it really necessary? Naturally I will try to find out why the dns resolver was taking so long, but I'm having trouble finding out from the source why resolving an ip address is necessary. Thanks! -- Andreas (who's very sleepy, it took quite a while to figure this one out)

2 1

[Users] UA not receiving BYE from SPA 3000
by Kenny Chua 26 Jun '06

26 Jun '06

I'm currently having this problem: SPA 3000 calls UA(xlite). The call goes through and everything is fine. When SPA 3000 hangs up. UA never hangs up. Seems like its not receiving the BYE. However, if the UA hangs up, everything is fine. This is some error I get: localhost /usr/local/sbin/openser[3236]: ERROR: t_newtran: transaction already in process 0xb62f1980 Jun 23 23:07:43 localhost /usr/local/sbin/openser[3236]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) This is how I handle my BYE in openser.cfg: if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); t_relay(); }; if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); return; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; t_relay(); route(1); return; }; I'm using rtpproxy on the same system. Any ideas how to fix this issue? Also, I was wondering how do I set rtpproxy to start on startup? Thank you --------------------------------- Yahoo! Groups gets better. Check out the new email design. Plus theres much more to come.

2 1

R: [Users] difference between return, break and exit.
by D'Addelfio Davide 26 Jun '06

26 Jun '06

Thanks for quick reply. Do you think in this case is bettr to use return or break function? if (method=="INVITE") { route(7); return; } else if (method=="ACK") { route(8); return; } else if (method=="CANCEL") { route(9); return; } else if (method=="BYE") { route(10); return; }; if (loose_route()) { xlog("L_INFO","Eseguo loose_route"); # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); return; }; if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("138.132.167.240", "subscriber")) { # www_challenge("138.132.167.240", "0"); # exit; #}; save("location"); #exit; return; }; }; Thanks Davide ________________________________ Da: raviprakash sunkara [mailto:sunkara.raviprakash.feb14@gmail.com] Inviato: lunedì 26 giugno 2006 14.21 A: D'Addelfio Davide Oggetto: Re: [Users] difference between return, break and exit. Hi Davide , openser 0.9 is having the break,,, now in 1.0.1 and 1.1.0 is using.... now.. return is also used ... But the difference b/n the return and exit.. is In retuern the function is + ve or - ve return type.. where exit .... to gettinh out the condition function; bettter to use only exit function... On 6/26/06, D'Addelfio Davide <Davide.D'Addelfio(a)italtel-consultant.com> wrote: Hi to all, i need to understand the difference in usage between return, break and exit. Could someone help me, maybe with some example? Thanks Davide _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

2 1

[Users] CANCEL
by D'Addelfio Davide 26 Jun '06

26 Jun '06

Hi to all, i can't understand why my openser doesn't capture the Cancel and doesn't store it into database, could anyone help me in some way? Please, I'm desperate...!!! Attached my cfg file. Thanks Davide

3 2

[Serusers] P-Asserted-Identity
by Dani Popa 26 Jun '06

26 Jun '06

Hi, How can i add P-Asserted-Identity with an external application ? Any ideeas ? thanks in advance. Dani

3 3

RE: [Users] Out of Memory
by Douglas Garstang 26 Jun '06

26 Jun '06

Bogdan, Don't appear to have ever received that message. To answer your questions: We are not using ENUM at all. We aren't using our own module for DNS lookups. I will try the latest cvs. Doug > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Monday, June 26, 2006 8:33 AM > To: Douglas Garstang > Cc: Users(a)openser.org > Subject: Re: [Users] Out of Memory > > > See the list. I sent you a reply : > http://www.openser.org/pipermail/users/2006-June/005321.html > > regards, > bogdan > > Douglas Garstang wrote: > > >>-----Original Message----- > >>From: Douglas Garstang > >>Sent: Thursday, June 22, 2006 12:09 PM > >>To: Bogdan-Andrei Iancu > >>Cc: Users(a)openser.org > >>Subject: RE: [Users] Out of Memory > >> > >> > >> > >> > >>>-----Original Message----- > >>>From: Douglas Garstang > >>>Sent: Wednesday, June 21, 2006 12:48 PM > >>>To: Bogdan-Andrei Iancu > >>>Cc: Users(a)openser.org > >>>Subject: RE: [Users] Out of Memory > >>> > >>> > >>>Bogdan, > >>> > >>>I finally managed to upload the memory dumps to pastebin. > Links are: > >>>Hopefully you can access them. > >>> > >>>Memory dump right after OpenSER was started: > >>>http://pastebin.com/723872 > >>>Memory dump after OpenSER running for 20min: > >>>http://pastebin.com/723890 > >>>Memory dump after OpenSER running for 50min: > >>>http://pastebin.com/723902 > >>> > >>>The problem started to occur between the 20 and 50 minute > >>>samples. About 20 calls had been processed in that time. > >>> > >>> > >>So... I'm wondering if anything has been determined with this... > >> > >> > > > >Hi Bogdan. Did you manage to find anything with this? > > > >Douglas. > > > > > >

2 1

RE: [Users] Out of Memory
by Douglas Garstang 26 Jun '06

26 Jun '06

> -----Original Message----- > From: Douglas Garstang > Sent: Thursday, June 22, 2006 12:09 PM > To: Bogdan-Andrei Iancu > Cc: Users(a)openser.org > Subject: RE: [Users] Out of Memory > > > > -----Original Message----- > > From: Douglas Garstang > > Sent: Wednesday, June 21, 2006 12:48 PM > > To: Bogdan-Andrei Iancu > > Cc: Users(a)openser.org > > Subject: RE: [Users] Out of Memory > > > > > > Bogdan, > > > > I finally managed to upload the memory dumps to pastebin. Links are: > > Hopefully you can access them. > > > > Memory dump right after OpenSER was started: > > http://pastebin.com/723872 > > Memory dump after OpenSER running for 20min: > > http://pastebin.com/723890 > > Memory dump after OpenSER running for 50min: > > http://pastebin.com/723902 > > > > The problem started to occur between the 20 and 50 minute > > samples. About 20 calls had been processed in that time. > > So... I'm wondering if anything has been determined with this... Hi Bogdan. Did you manage to find anything with this? Douglas.

2 1

[Serusers] SER + B2BUA
by roger leszczynski 26 Jun '06

26 Jun '06

Anyone here use ser + b2bua for billing reasons? if so can you provide the config files? ..... i want to be able to capture cdrs even if i restart ser.

6 6

RE: [Serusers] Facing problems while downloading SEMS
by Aimable Habiyakare 26 Jun '06

26 Jun '06

George, As per this link http://www.iptel.org/drupal/sems Please use only following SERi source tarball:ser-0.9.6-sems_src.tar.gz <http://ftp.iptel.org/pub/sems/ser-0.9.6-sems_src.tar.gz> The link I gave has also some sample config files. For the VoiceXML support, I don't know but I think it does not - yet. Thanks Aimable ________________________________ From: Benjamin.George(a)t-systems.com [mailto:Benjamin.George@t-systems.com] Sent: Monday, June 26, 2006 7:45 AM To: Aimable Habiyakare Cc: serusers(a)iptel.org Subject: RE: [Serusers] Facing problems while downloading SEMS Hi Aimable, Thanks a lot buddy! I got whatever I wanted from that link. Now let me compile it properly and test it with my application. By the way I would like to ask you two more things: 1. SER 0.9.6 is already running on my m/c and my question is, before installing SEMS, any other modules have to be incorporated in to the SER? 2. Whether SER support Voice XML (VXML)? Regards, Benjamin. ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Aimable Habiyakare Sent: Friday, June 23, 2006 6:21 PM To: George, Benjamin Cc: serusers(a)iptel.org Subject: RE: [Serusers] Facing problems while downloading SEMS Hi George You can go here http://ftp.iptel.org/pub/sems/ you will find every thing you need Aimable ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Benjamin.George(a)t-systems.com Sent: Friday, June 23, 2006 2:37 PM To: serusers(a)lists.iptel.org Subject: [Serusers] Facing problems while downloading SEMS Hi, I am planning to integrate SEMS (media server) with SER to develop some conferencing and IVR applications on top of that. While I was trying to checkout SEMS code from SVN repository using SVN: or HTTP: it is not getting downloaded. It was throwing some connection failure error. The commands which I tried are: svn checkout svn://svn.berlios.de/sems/trunk, and svn checkout http://svn.berlios.de/svnroot/repos/sems/trunk So it will be great if somebody can provide me some pointer on: 1. How can I checkout SEMS source successfully? 2. Is there any SEMS source tarball readily available to download? 3. Is there any SEMS ready to install binary on debian platform is available? 4. Whether SEMS support Voice XML (VXML)? Regards, Benjamin.

1 0

[Users] difference between return, break and exit.
by D'Addelfio Davide 26 Jun '06

26 Jun '06

Hi to all, i need to understand the difference in usage between return, break and exit. Could someone help me, maybe with some example? Thanks Davide

1 0

RE: [Serusers] ERROR: bad config file (1 errors)
by mukesh 26 Jun '06

26 Jun '06

hi ram, thanks. I've got the Mysql problem solved. new one arises !! [root@oracle2 sbin]# ./ser 0(7598) WARNING: fix_socket_list: could not rev. resolve 202.80.40.141 0(7598) WARNING: fix_socket_list: could not rev. resolve 202.80.40.141 Listening on udp: 202.80.40.141 [202.80.40.141]:5060 udp: 127.0.0.1 [127.0.0.1]:5060 tcp: 202.80.40.141 [202.80.40.141]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 Aliases: tcp: localhost:5060 tcp: localhost.localdomain:5060 tcp: oracle:5060 tcp: oracle2.hungama.com:5060 udp: localhost:5060 udp: localhost.localdomain:5060 udp: oracle:5060 udp: oracle2.hungama.com:5060 *: 202.80.40.141:* *: ims.hungama.com:* stateless - initializing [root@oracle2 sbin]# 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 0(0) ERROR: udp_init: bind(4, 0x810bffc, 16) on 202.80.40.141: Address already in use #this is error below is few lines from my ser.cfg, may help u to find the error # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) listen=202.80.40.141 listen=127.0.0.1 # hostname matching an alias will satisfy the condition uri==myself". alias=ims.mysip.com alias=202.80.40.141 /* 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 tdfsdfis if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.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) any clu ?? regards Mukesh _____ From: ram [mailto:talk2ram@gmail.com] Sent: Monday, June 26, 2006 11:19 AM To: mukesh Subject: Re: [Serusers] ERROR: bad config file (1 errors) Hi Mukesh better get in to onsip.org download ser-0.97-pre edit make file in the exclude remove mysql and save make all make install should solve ur problem ram On 6/23/06, mukesh <mukesh(a)hungama.com> wrote: hi olivier, in fact I have downloaded ser-0.9.6_linux_i386.tar.tar from iptel.org <http://iptel.org/> . then I have put the ser-0.9.6_linux_i386.tar.tar file in root(/) directory and executed following command [root@oracle2 /]# tar -xvzf ser-0.9.6_linux_i386.tar.tar do still need to recompile it ? .....I think complation had to be done in case I had the source code. regards Mukesh _____ From: olivier.taylor [mailto:olivier.taylor@gmail.com] Sent: Friday, June 23, 2006 3:16 PM To: mukesh Cc: olivier.taylor(a)phonext.com <mailto:olivier.taylor@phonext.com> ; serusers(a)lists.iptel.org Subject: Re: [Serusers] ERROR: bad config file (1 errors) it seems that you didn't compile ser with mysql support. go to the makefile et have a look at exclude_modules?= delete the mysql entry and recompile Olivier mukesh a écrit : hi olivier, below is the log -----------------------------------log starts here-------------------------------------------- Jun 23 14:51:48 oracle2 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.12: cannot open shared object file: No such file or directory Jun 23 14:51:48 oracle2 ser: parse error (37,13-14): failed to load module -----------------------------------log ends here-------------------------------------------- and line no. 37 is loadmodule "/usr/local/lib/ser/modules/mysql.so" please suggest some solution. regards Mukesh _____ From: olivier.taylor [mailto:olivier.taylor@gmail.com] Sent: Friday, June 23, 2006 12:45 PM To: mukesh Cc: olivier.taylor(a)phonext.com; serusers(a)lists.iptel.org Subject: Re: [Serusers] ERROR: bad config file (1 errors) yes, sure :) /var/log/messages you will find the informations you need concerning that error, at least, the line of your cfg where the error occurs. Olivier mukesh a écrit : Thanks you olivier. Can you be little more specific, where this message file is located ? Regards Mukesh -----Original Message----- From: olivier.taylor [mailto:olivier.taylor@gmail.com] Sent: Thursday, June 22, 2006 9:55 PM To: mukesh Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] ERROR: bad config file (1 errors) see the message file, you will find there the line wich is not correct on your ser.cfg Olivier mukesh a écrit : Hi I am getting "ERROR: bad config file (1 errors)" while truing to execute the SER. And I could not figure out whats wrong with this line loadmodule "/usr/local/lib/ser/modules/mysql.so" If I comment it then everything is file. Following is my ser.cfg file ---------------------------------ser.cfg star------------------------------ # # $Id: ser.cfg,v 1.25.2.1 <http://1.25.2.1/> 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) listen=192.87.41.149 <http://192.87.41.149/> #this address has been changed # hostname matching an alias will satisfy the condition uri==myself". alias=sip.myserver.com <http://sip.myserver.com/> /* 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" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" #added by mukesh # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") #added by mukesh 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) # ------------------------- 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 >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # 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("sip.myserver.com <http://sip.myserver.com/> ", "subscriber")) { www_challenge("ser.hungama.com <http://ser.hungama.com/> ", "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] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } ---------------------------------ser.cfg ends------------------------------ Any help/suggestion is highly appreciated. Regards Mukesh _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _____ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] first posting, got a few problems..
by nick 26 Jun '06

26 Jun '06

To explain my situation, I've got a dual opteron 246 system with 1gig of ram and 2 80gig disks in raid1 (software raid), I'm running CentOS 4.3 x86_64 with kernel 2.6.9-34.ELsmp I'm running ser 0.9.6-6 (the -6 may just be an artifact from the fedora core rpm.. I dunno) (compiled from ser-0.9.6_src.tar.gz) Since no one seems to makes rpms for redhat or centOS (the same thing obviously) I went and made my own. I grabbed a fedora core 6 src rpm, and modified the specfile to my own liking (basically I took out the stuff related to postgres), the compile went fine, as far as I can tell, no errors were big enough to stop the compilation at least, and the rpms installed just fine. All the modules are in the right place, and my ser.cfg seems to be correct (I don't get any more errors during startup), however, I do get a segfault :( here's the attached output from /log/messages Jun 23 12:20:18 sipserver ser: Listening on Jun 23 12:20:18 sipserver ser: udp: 192.168.1.93 [192.168.1.93]:5060 Jun 23 12:20:18 sipserver ser: tcp: 192.168.1.93 [192.168.1.93]:5060 Jun 23 12:20:18 sipserver ser: Aliases: Jun 23 12:20:18 sipserver ser: tcp: pc-00093:5060 Jun 23 12:20:18 sipserver ser: udp: pc-00093:5060 Jun 23 12:20:18 sipserver ser: Jun 23 12:20:18 sipserver kernel: ser[3682]: segfault at 0000000000000000 rip 0000002a9589ef92 rsp 0000007fbffff850 error 4 Jun 23 12:20:18 sipserver ser: ser startup succeeded Obviously ser isn't running, does anyone have any idea where I can start looking to find out what dies/died/went wrong with the compile? Thanks, Nick

4 9

[Serusers] Checking Active Calls
by Ryan Pagquil 26 Jun '06

26 Jun '06

Hi, I'm implementing SER in a prepaid environment. Now we are getting active calls from the radacct tables of RADIUS since we are using RADIUS auth and acct. We are just assuming that those entries without an acctstoptime and sessiontime are active calls. Is there a way for me monitor if the call (entry in the radacct table) is really active? Or is there any tool that I can use to monitor active RTP sessions from gateway to UA? Thanks, Ryan

2 1

[Users] "513 Message too big" for the Invite
by Reddy, Muralidhar 26 Jun '06

26 Jun '06

Hello, Everything works fine if I run Proxy without TLS disabled. I tried both TCP/UDP, it works fine. For TLS, Registration went fine however Proxy responds "513 Message too big" to the Invite. I don't see any wrong headers in the message. The length is almost same for TCP and TLS invite message, the only change is sip and sips. Any 'C' file or config file I need to modify? Openser.conf: debug=9 # debug level (cmd line: -dddddddddd) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=yes dns=no rev_dns=no port=5061 children=4 fifo="/tmp/openser_fifo" disable_tls = 0 listen = tls:158.218.105.138:5061 tls_verify = on tls_require_certificate = on tls_method = TLSv1 tls_certificate = "/home/mreddy/tools/cert.pem" tls_private_key = "/home/mreddy/tools/priv.pem" tls_ca_list = "/home/mreddy/tools/root_cert_fluffyCA.pem" #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/textops.so" route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; #if 1 if (msg:len >= 4096 ) { sl_send_reply("513", "Message too big"); exit; }; #endif # 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); }; if (!uri==myself) { route(1); }; # 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") { save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } Thanks, Murali Reddy Texas Instruments Murali Reddy Texas Instruments Germantown,MD Phone: 301-515-3716 e-mail: mreddy(a)ti.com <mailto:mreddy@ti.com>

2 1

[Serusers] How do I get rid of npdi=yes?
by John Clements 26 Jun '06

26 Jun '06

I've looked around and have found one or two other posts asking about this, but I have never seen an answer. The From is: =uri: <sip:5551212;npdi=yes@x.x.x.x:5060;dtg=SIP;user=phone> Which causes a number of issues. USRLOC no longer works: =lookup(): '5551212;npdi=yes' Not found in usrloc And when you send the call to an Asterisk server, Asterisk wigs out because it truncates everything after the first ";" it encounters in the URI. Does anyone have any idea how to get rid of it? I've tried subst(), but could not get it to work. :-( -John

7 9

Re: [Users] AVPops Arrays?
by Daniel-Constantin Mierla 26 Jun '06

26 Jun '06

On 06/26/06 03:19, Kelvin Williams wrote: > > I’m contemplating the storage of a lot of values (15 or so) for users > using AVPops. My question would be the MySQL server and any potential > bottleneck it may create. > > Is it possible to perform one database hit and extract all of the AVPs > into an array which could then be referenced throughout the transaction? > with the cvs heade version you can use avp_db_query(). Cheers, Daniel > Thanks, > > Kw > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

RE: [Serusers] Facing problems while downloading SEMS
by Benjamin.George＠t-systems.com 26 Jun '06

26 Jun '06

Hi Aimable, Thanks a lot buddy! I got whatever I wanted from that link. Now let me compile it properly and test it with my application. By the way I would like to ask you two more things: 1. SER 0.9.6 is already running on my m/c and my question is, before installing SEMS, any other modules have to be incorporated in to the SER? 2. Whether SER support Voice XML (VXML)? Regards, Benjamin. _____ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Aimable Habiyakare Sent: Friday, June 23, 2006 6:21 PM To: George, Benjamin Cc: serusers(a)iptel.org Subject: RE: [Serusers] Facing problems while downloading SEMS Hi George You can go here http://ftp.iptel.org/pub/sems/ <http://ftp.iptel.org/pub/sems/> you will find every thing you need Aimable _____ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Benjamin.George(a)t-systems.com Sent: Friday, June 23, 2006 2:37 PM To: serusers(a)lists.iptel.org Subject: [Serusers] Facing problems while downloading SEMS Hi, I am planning to integrate SEMS (media server) with SER to develop some conferencing and IVR applications on top of that. While I was trying to checkout SEMS code from SVN repository using SVN: or HTTP: it is not getting downloaded. It was throwing some connection failure error. The commands which I tried are: svn checkout svn://svn.berlios.de/sems/trunk, and svn checkout http://svn.berlios.de/svnroot/repos/sems/trunk So it will be great if somebody can provide me some pointer on: 1. How can I checkout SEMS source successfully? 2. Is there any SEMS source tarball readily available to download? 3. Is there any SEMS ready to install binary on debian platform is available? 4. Whether SEMS support Voice XML (VXML)? Regards, Benjamin.

1 0

[Users] AVPops Arrays?
by Kelvin Williams 26 Jun '06

26 Jun '06

I'm contemplating the storage of a lot of values (15 or so) for users using AVPops. My question would be the MySQL server and any potential bottleneck it may create. Is it possible to perform one database hit and extract all of the AVPs into an array which could then be referenced throughout the transaction? Thanks, Kw

1 0

[Serusers] CDRTool +Asterisk + Ser
by hgaillac-sip＠yahoo.fr 24 Jun '06

24 Jun '06

Hello, I try to configure CDRTool + Asterisk . After having fixed a lot of files I get too many problems with this software. First I get this error why ? Error: Table 'asterisk_cdr' was not locked with LOCK TABLES The basic soap client does not connect to the soap server to get the version . harry ___________________________________________________________________________ Yahoo! Mail réinvente le mail ! Découvrez le nouveau Yahoo! Mail et son interface révolutionnaire. http://fr.mail.yahoo.com

2 1

RE: [Users] Modifying from_uri
by Kelvin Williams 24 Jun '06

24 Jun '06

Disregard.. Found it the moment I hit send. UAC. _____ From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Kelvin Williams Sent: Saturday, June 24, 2006 5:41 PM To: users(a)openser.org Subject: [Users] Modifying from_uri We are deploying in a cable environment, and for sake of ease we are registering all SIP UAs by their MAC address. And associating phone numbers with dbaliases. It all works great until we try to send a phone call to the PSTN. I need to know the way to modify the from_uri to change it from the MAC of the device to the actual phone number which we will retrieve using avp or so. Thanks in advance. Kw

1 0

[Users] Modifying from_uri
by Kelvin Williams 24 Jun '06

24 Jun '06

We are deploying in a cable environment, and for sake of ease we are registering all SIP UAs by their MAC address. And associating phone numbers with dbaliases. It all works great until we try to send a phone call to the PSTN. I need to know the way to modify the from_uri to change it from the MAC of the device to the actual phone number which we will retrieve using avp or so. Thanks in advance. Kw

1 0

[Users] errors in xlog-output
by Thomas Balsfulland 24 Jun '06

24 Jun '06

hello together, i am using openser v1.0.1 as a simple proxy. for logging i use xlog like xlog("L_ERR","[$Tf]: PSTN -> SIP: new Call [$fU] to [$tU]\n"); with the setting modparam("xlog", "buf_size", 16384). but in the logfile i see many error-messages: 5(29423) ERROR: warning_builder: buffer size exceeded 5(29423) WARNING: warning skipped -- too big is this a problem of xlog? anyone knows how can i fix this? thank you for help. regards thomas -- thomas balsfulland tbals(a)ctrl-c.de zwischen mut und dummheit liegt nur ein grat der sich durch das vorbereitetsein unterscheidet

2 1

Re: FW: [Users] Out of Memory
by Bogdan-Andrei Iancu 24 Jun '06

24 Jun '06

Douglas, indeed, it looks like there is a a memory leak in using the DNS resolver. It is used by ENUM module and by the SIP resolver. So, are you using ENUM module for lookups? Do you have some module of your own using the DNS resolver? I took a look into the SIP resolver and I found some possible cases leading to memory leak, but there were some error handling corner cases......only if you have some very bogus DNS results...anyhow try to update from CVS and see if the leak is still present. regards, bogdan Douglas Garstang wrote: >Bogdan, > >I guess I didn't realise the dumping was done on a per process basis. >I've sent you a complete daemon.log file this time. You will see that many of the processes are reporting out of memory. >I sent a USR1 signal to pid's 21131, 21127 and 21126 (which where all reporting out of memory). > >Douglas. > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Thursday, June 22, 2006 12:10 PM >To: Douglas Garstang >Cc: Users(a)openser.org >Subject: Re: [Users] Out of Memory > > >Hi Douglas, > >I when over the dumps you get, but there is no increasing memory >consumption in time (some number of used mem chunks). Also the first >line shows: > Memory status (pkg): > qm_status (0x8121ba0): > heap size= 1048576 > used= 21908, used+overhead=248632, free=799944 > >so only 25% is used from the 1M available. > >Please check that you are actually sending the SIGUSR1 signal to the >process that generated the memory error (it's about the per-process >memory and not the shared one). So, wait for the error to occur, send >the signal to the reporting process and please send me the log (no need >to post again on web). > >regards, >bogdan > > >Douglas Garstang wrote: > > > >>Bogdan, >> >>I finally managed to upload the memory dumps to pastebin. Links are: >>Hopefully you can access them. >> >>Memory dump right after OpenSER was started: http://pastebin.com/723872 >>Memory dump after OpenSER running for 20min: http://pastebin.com/723890 >>Memory dump after OpenSER running for 50min: http://pastebin.com/723902 >> >>The problem started to occur between the 20 and 50 minute samples. About 20 calls had been processed in that time. >> >>Douglas. >> >> >> >> >> >>>-----Original Message----- >>>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>>Sent: Wednesday, June 21, 2006 10:14 AM >>>To: Douglas Garstang >>>Cc: Users(a)openser.org >>>Subject: Re: [Users] Out of Memory >>> >>> >>>As written in the wiki, if you do not know how to interpret >>>the result, >>>you should post them on a ftp or http server and send the link to the >>>list - also, important is to mention in which context did you get the >>>mem dump - is the shutdown dump, a run time dump... >>> >>>regards, >>>bogdan >>> >>> >>>Douglas Garstang wrote: >>> >>> >>> >>> >>> >>>>Once I get the memory status, how do I interpret the results? >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>>>-----Original Message----- >>>>>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>>>>Sent: Wednesday, June 21, 2006 9:43 AM >>>>>To: Douglas Garstang >>>>>Cc: Users(a)openser.org >>>>>Subject: Re: [Users] Out of Memory >>>>> >>>>> >>>>>Hi Douglas, >>>>> >>>>>it was signal SIGUSR1 and not SIGHUP - I strongly advice you >>>>>to go through >>>>> http://openser.org/dokuwiki/doku.php?id=memory >>>>>note you have to recompile and reinstall in order to be able to do >>>>>memory debug. >>>>> >>>>>also does not matter how much mem. your system has - openser >>>>>will use by >>>>>default only 1M per process. Refer to the wiki for how to >>>>> >>>>> >>>>> >>>>> >>>increase it. >>> >>> >>> >>> >>>>>Regards, >>>>>Bogdan >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >> >> >> >> > > >

1 0

[Serusers] radius dictionary.ser missing attributes?
by John Clements 23 Jun '06

23 Jun '06

I configured radius accounting and added the contents of /etc/ser/dictionary.ser to /etc/radiusclient/dictionary and get the following errors when starting radiusd: rlm_acct_unique: Cannot find attribute 'Client-IP-Address' in dictionary So I took the one out of: /usr/src/ser/ser-0.9.7-pre1/modules/acc/etc/sip_dictionary and appended it to the original dicitionary and received the same error. Does anyone know what I may be doing wrong? -John

2 2

[Users] I receive "513 Message too big" for the Invite
by Reddy, Muralidhar 23 Jun '06

23 Jun '06

Hello, Everything works fine if I run Proxy without TLS disabled. I tried both TCP/UDP, it works fine. For TLS, Registration went fine however Proxy responds "513 Message too big" . I don't see any wrong headers in the message. The length is almost same for TCP and TLS invite message, the only change is sip and sips. Any C file I need to modify? Openser.conf # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=9 # 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=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. lineset sip register_timeout: -R) port=5061 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support disable_tls = 0 listen = tls:158.218.105.138:5061 tls_verify = on tls_require_certificate = on tls_method = TLSv1 tls_certificate = "/home/mreddy/tools/cert.pem" tls_private_key = "/home/mreddy/tools/priv.pem" tls_ca_list = "/home/mreddy/tools/root_cert_fluffyCA.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #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/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/openser/modules/auth.so" #loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 2) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) #modparam("usrloc|auth_db", "db_url", "dbtext:///tmp/openserdb") # -- 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) # ------------------------- 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"); exit; }; #if 1 if (msg:len >= 4096 ) { sl_send_reply("513", "Message too big"); exit; }; #endif # 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); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # 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("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } Thanks, Murali Reddy Texas Instruments

1 0

[Serusers] Facing problems while downloading SEMS
by Benjamin.George＠t-systems.com 23 Jun '06

23 Jun '06

Hi, I am planning to integrate SEMS (media server) with SER to develop some conferencing and IVR applications on top of that. While I was trying to checkout SEMS code from SVN repository using SVN: or HTTP: it is not getting downloaded. It was throwing some connection failure error. The commands which I tried are: svn checkout svn://svn.berlios.de/sems/trunk, and svn checkout http://svn.berlios.de/svnroot/repos/sems/trunk So it will be great if somebody can provide me some pointer on: 1. How can I checkout SEMS source successfully? 2. Is there any SEMS source tarball readily available to download? 3. Is there any SEMS ready to install binary on debian platform is available? 4. Whether SEMS support Voice XML (VXML)? Regards, Benjamin.

2 1

[Serusers] oops, forgot to click reply to all...
by nick 23 Jun '06

23 Jun '06

-------- Original Message -------- Subject: Re: [Serusers] first posting, got a few problems.. Date: Fri, 23 Jun 2006 15:51:06 +0200 From: nick <nick(a)mobilia.it> To: Atle Samuelsen <clona(a)cyberhouse.no> References: <449BCB37.5030908(a)mobilia.it> <20060623113401.GM86417(a)mystic.cyberhouse.no> Atle Samuelsen wrote: > Hi nick > > Try to do the following : > > ulimit -c unlimited > > that will make it coredump instead. > then, next time you start it, it should make a core file ser.core or > something like that. > > When that´s done, do : > > gdb �-c ser.core(orwhatthecorefileiscalled)ser > > this will open gdb, so you can run a backtrace. > > type : > bt > > and you will get the backtrace. Past it here, and maybe somebody can see > what´s wrong :-) > > Best regards, > > Atle > > > * nick <nick(a)mobilia.it> [060623 12:29]: > >>To explain my situation, I've got a dual opteron 246 system with 1gig of ram and 2 80gig disks in raid1 (software raid), I'm running CentOS 4.3 x86_64 with kernel 2.6.9-34.ELsmp I'm running ser 0.9.6-6 (the >>-6 may just be an artifact from the fedora core rpm.. I dunno) (compiled from ser-0.9.6_src.tar.gz) >> >> >>Since no one seems to makes rpms for redhat or centOS (the same thing obviously) I went and made my own. I grabbed a fedora core 6 src rpm, and modified the specfile to my own liking (basically I took out >>the stuff related to postgres), the compile went fine, as far as I can tell, no errors were big enough to stop the compilation at least, and the rpms installed just fine. All the modules are in the right >>place, and my ser.cfg seems to be correct (I don't get any more errors during startup), however, I do get a segfault :( >> >>here's the attached output from /log/messages >> >>Jun 23 12:20:18 sipserver ser: Listening on >>Jun 23 12:20:18 sipserver ser: udp: 192.168.1.93 [192.168.1.93]:5060 >>Jun 23 12:20:18 sipserver ser: tcp: 192.168.1.93 [192.168.1.93]:5060 >>Jun 23 12:20:18 sipserver ser: Aliases: >>Jun 23 12:20:18 sipserver ser: tcp: pc-00093:5060 >>Jun 23 12:20:18 sipserver ser: udp: pc-00093:5060 >>Jun 23 12:20:18 sipserver ser: >>Jun 23 12:20:18 sipserver kernel: ser[3682]: segfault at 0000000000000000 rip 0000002a9589ef92 rsp 0000007fbffff850 error 4 >>Jun 23 12:20:18 sipserver ser: ser startup succeeded >> >>Obviously ser isn't running, does anyone have any idea where I can start looking to find out what dies/died/went wrong with the compile? >> >>Thanks, >> >>Nick >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> > > Can you tell me where it should create this coredump? I did the ulimit -c unlimited and restarted ser both manually and with the init.d command, but neither command seems to have created a file for me to parse...

1 0

RE: [Serusers] Facing problems while downloading SEMS
by Aimable Habiyakare 23 Jun '06

23 Jun '06

Hi George You can go here http://ftp.iptel.org/pub/sems/ you will find every thing you need Aimable ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Benjamin.George(a)t-systems.com Sent: Friday, June 23, 2006 2:37 PM To: serusers(a)lists.iptel.org Subject: [Serusers] Facing problems while downloading SEMS Hi, I am planning to integrate SEMS (media server) with SER to develop some conferencing and IVR applications on top of that. While I was trying to checkout SEMS code from SVN repository using SVN: or HTTP: it is not getting downloaded. It was throwing some connection failure error. The commands which I tried are: svn checkout svn://svn.berlios.de/sems/trunk, and svn checkout http://svn.berlios.de/svnroot/repos/sems/trunk So it will be great if somebody can provide me some pointer on: 1. How can I checkout SEMS source successfully? 2. Is there any SEMS source tarball readily available to download? 3. Is there any SEMS ready to install binary on debian platform is available? 4. Whether SEMS support Voice XML (VXML)? Regards, Benjamin.

1 0

[Serusers] ERROR: bad config file (1 errors)
by mukesh 23 Jun '06

23 Jun '06

Hi I am getting "ERROR: bad config file (1 errors)" while truing to execute the SER. And I could not figure out whats wrong with this line loadmodule "/usr/local/lib/ser/modules/mysql.so" If I comment it then everything is file. Following is my ser.cfg file ---------------------------------ser.cfg star------------------------------ # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) listen=192.87.41.149 #this address has been changed # hostname matching an alias will satisfy the condition uri==myself". alias=sip.myserver.com /* 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" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" #added by mukesh # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") #added by mukesh 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) # ------------------------- 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 >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # 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("sip.myserver.com", "subscriber")) { www_challenge("ser.hungama.com", "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] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } ---------------------------------ser.cfg ends------------------------------ Any help/suggestion is highly appreciated. Regards Mukesh

3 8

[Serusers] openser.pid exists but doesn't exist for openserctl
by Shenen Shenen 23 Jun '06

23 Jun '06

Hi!I've created openser.pid with openser -P /var/run/openser.pid but the file is empty and if I start openserctl restart it says that doesn't exist the openser.pid file..... Which values have I to set enter the openser.pid?(I put them editing the file....)

1 0

[Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723
by Adrian Georgescu 23 Jun '06

23 Jun '06

Yes ilker, MediaProxy relays IP packets at layer 3. It does not interpret the content of the RTP packets (like used codecs) nor does it do any conversion, it's a simple IP packet relay, which works for any type of media session negotiated between the user agents. Adrian ====== I think mediaproxy can't do transcoding. Can anyone confirm this ? Thanks, ilker

1 0

[Serusers] Need your Help -please
by S.A.S. Lorensuhewa 23 Jun '06

23 Jun '06

Daer All, I am trying to configure my Sip server about a month now. Still I have problem of connecting a UA to the Sip Server. Folowing two activities I have done and activity two confirm that client is trying to connect to the Sip server @ 192.248.48.38. But still user is not registering. In my sip server I have created one account for testing as follows. serctl add sip1 sip1ppp sip1(a)sip.ruh.ac.l I will also attach my ser.cfg file. Please somebody check the .cfg file and let me know any changes I need or any other solution for my problem Thanks Aruna 1. serctl ul show ---------------- ===Domain list=== ---Domain--- name : 'aliases' size : 512 table: 0xb6153ed8 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ---Domain--- name : 'location' size : 512 table: 0xb6151e78 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ===/Domain list=== [root@SipServer ~]# 2. ngrep -W byline sip1 port 5060 -q --------------------------------- U 192.248.48.25:5060 -> 192.248.48.38:5060 REGISTER sip:sip.ruh.ac.lk SIP/2.0. Via: SIP/2.0/UDP 192.248.48.25:5060;rport;branch=z9hG4bKAFD579D50D4A4A9BB71882B15F53078D. From: aruna <sip:sip1@sip.ruh.ac.lk>;tag=2467586724. To: aruna <sip:sip1@sip.ruh.ac.lk>. Contact: "aruna" <sip:sip1@192.248.48.25:5060>. Call-ID: E1A26030F76346DDAF92D1E10E897349(a)sip.ruh.ac.lk. CSeq: 65515 REGISTER. Expires: 1800. Max-Forwards: 70. User-Agent: X-Lite release 1103m. Content-Length: 0. . ---------------------------------------------------------- S. A. S. Lorensuhewa Head, Department of Computer Science University of Ruhuna, Matara, Sri Lanka Telephone: 0094 (0)41 2227019 Fax: 0094 (0)41 2222683 Email: lorensuhewa(a)cc.ruh.ac.lk, aruna(a)cc.ruh.ac.lk

3 5

RE: [Serusers] Mediaproxy problem - accounting
by İlker Aktuna $Koç.net$ 23 Jun '06

23 Jun '06

1 0

[Serusers] Mediaproxy problem - accounting
by Adrian Georgescu 23 Jun '06

23 Jun '06

hi Ilker, Pl check if you have the mysql python package installed. On Debian the name of the packet is python-mysqldb ii python-mysqldb 1.2.1c3-4 A Python interface to MySQL also make sure you have the extra fields in the radacct table. Adrian ============= Hi, I am trying to use dbaccounting with mediaproxy. Undfortunately it doesn't seem to send anything to mysql db. How can I trace where the problem is ? My mediaproxy accounting config is : [Accounting] dbaccounting = on [Database] user = ser password = heslo host = 127.0.0.1 database = radius table = radacct Thanks, ilker

1 0

[Serusers] problem to install the the ser
by vijay tiwari 23 Jun '06

23 Jun '06

i have to install SIP Express Router i had downlod the package ser-0.8.11-0.i386.rpm its show installed but show any thing please help me out Name and version of your operating system Linux aster 2.6.11-1.1369_FC4 #1 Thu Jun 2 22:55:56 EDT 2005 i686 i686 i386 GNU/Linux package ser-0.8.11-0.i386.rpm Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.) M- 09871633711 --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.

1 0

[Serusers] Beneficial Hardware For SER
by Nick Hoffman 23 Jun '06

23 Jun '06

Hi guys. Does SER receive greater benefit from CPU, RAM or harddrive upgrades? I have SER running on a P4 3GHz with 1GB of RAM and 7200RPM drives. I've searched through the serusers and serdev mailing lists for this information, but haven't been able to find a definitive answer. Cheers, -- 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.

2 1

[Serusers] Forward number to IP
by John Clements 23 Jun '06

23 Jun '06

Is there any way I can forward all calls to 555-1212 to server a.b.c.d? No phone actually connects to my SER server, they connect to other servers. I would just like to be able to have the SER server forward calls to these servers. I can do it by editing the ser.cfg, but would like to be able to do it using the database or radius. All I have found though is it using user accounts. Any ideas? -John

2 1

[Serusers] Stuck with libmysqlclient.so.10 and it is a problem.
by Chahn John Kim 23 Jun '06

23 Jun '06

Hi, all. I am looking for your expert help. I am building ser 0.8.12 on my fedora core 4, just because it is the highest version provided by iptel, while debian has all the newest goodies to .9+. In that, ser's mysql.so links to libmysqlclient.so.10 and this creates problems; *** After ser starts, it immediately shuts down when it loads auth_db.so. due to "authentication protocol" mismatch with mysql server 4.12 which is shipped with Fedora core. Is there 3rd party ser mysql module that uses libmysqlclient.so.14 somewhere? Or is it possible to compile newer verion source to work on fedora? Basically I am looking for ser mysql module that calls so.14 and can run on fedora. Thanks in advance. -- Chahn John Kim Koreanet.US Tel: 847-564-0698 email: johnk(a)koreanet.us johnk(a)netcomusa.us

3 3

[Users] radius accounting problem
by tele 23 Jun '06

23 Jun '06

hi all, reading acc modules README: * There is no session accounting -- OpenSER maintains no sessions. If one needs to correlate INVITEs with BYEs for example for purpose of billing, then it is better done in the entity which collects accounting information. Otherwise, SIP server would have to become sessions-stateful, which would very badly impact its scalability. so it's impossible to do radius accounting for billing purpose? i need a B2BUA for that? any hints? anyway i'm trying radius acct this is my problem: 0(25586) ACC: transaction answered: call_id=d3cd352603256e75@ua1, totag=360701152, from="222" <sip:222@mysip>;tag=d288f8fc7ea68a04, i-uri=sip:111@ss:5060, method=BYE, o-uri=sip:111@ss:5060, fromtag=d288f8fc7ea68a04, code=200, to=<sip:111@mysip>;tag=360701152, uid=n/a, userpart=111 0(25586) ERROR: acc_rad_request: radius-ing failed i've activated all possible debug but i received only this message about radius. i'm trying to activate debug for radiusclient this is my setup: FreeBSD 6.1 OpenSER 1.0.1 radiusclient-0.5.2 thank you

6 12

[Serusers] WINFO
by Alois Schafferhofer 23 Jun '06

23 Jun '06

Hi. If I send a Subscribe with event=presence.winfo, SER answers with "Unsupported document format". In my config file, I have the line "watcherinfo_notify", 1. Where is the problem? I can't find an error in my Subscribe message. Regards, Alois SUBSCRIBE sip:pda_user@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK7288678..Max-Forwards: 70..From: <pda_user(a)fh-joanneum.at> ;tag=1085963678..To: <pda_user(a)fh-joanneum.at>..Call-ID: 1239455995-305180140- 863656382..CSeq: 2 SUBSCRIBE..Contact: <pda_user(a)10.0.0.3>..Content-Leng th: 0..Accept: application/pidf+xml..Allow-Events: application/watcherinfo+xml..Expires: 3600..Event: presence.winfo..User-Agent: ... SIP/2.0 415 Unsupported document format for given package..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK7288678..From: <pda_user(a)fh-joanneum.at>;tag=1 085963678..To: <pda_user(a)fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28- c331..Call-ID: 1239455995-305180140-863656382..CSeq: 2 SUBSCRIBE..Server : Sip EXpress router (0.10.99-dev35-pa-4.1 (i386/linux))..Content-Length: 0..Warning: 392 10.0.0.11:5060 "Noisy feedback tells: pid=10832 req_src_ip= 10.0.0.3 req_src_port=5060 in_uri=sip:pda_user@fh-joanneum.at out_uri=sip:pda_user@fhjoanneum. at via_cnt==1".

3 2

[Users] Re: could not rev. resolve IP......
by Daniel-Constantin Mierla 22 Jun '06

22 Jun '06

On 06/21/06 13:14, Shenen Shenen wrote: > I've removed the listen parameters....fork was alway set like > fork=yes....with openser command I've this; > [root@asterisk12 ~]# openser > 0(3357) WARNING: fix_socket_list: could not rev. resolve > 192.168.251.12 <http://192.168.251.12> > 0(3357) WARNING: fix_socket_list: could not rev. resolve > 192.168.251.200 <http://192.168.251.200> > 0(3357) WARNING: fix_socket_list: could not rev. resolve > 192.168.251.12 <http://192.168.251.12> > 0(3357) WARNING: fix_socket_list: could not rev. resolve > 192.168.251.200 <http://192.168.251.200> > Listening on > udp: 127.0.0.1 <http://127.0.0.1> [127.0.0.1 > <http://127.0.0.1>]:5070 > udp: 192.168.251.12 <http://192.168.251.12> [ > 192.168.251.12 <http://192.168.251.12>]:5070 > udp: 192.168.251.200 <http://192.168.251.200> > [192.168.251.200 <http://192.168.251.200>]:5070 > tcp: 127.0.0.1 <http://127.0.0.1> [ 127.0.0.1 > <http://127.0.0.1>]:5070 > tcp: 192.168.251.12 <http://192.168.251.12> > [192.168.251.12 <http://192.168.251.12>]:5070 > tcp: 192.168.251.200 <http://192.168.251.200> > [192.168.251.200 <http://192.168.251.200>]:5070 > Aliases: > tcp: localhost:5070 > udp: localhost:5070 > > stateless - initializing > 0(0) Maxfwd module- initializing > 0(0) AUTH module - initializing > 0(0) AUTH_DB module - initializing > [root@asterisk12 ~]# 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 > 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 > 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 > 0(0) ERROR: tcp_init: bind(7, 0x811b12c, 16) on 127.0.0.1:5070 > <http://127.0.0.1:5070> : Address already in use > > > > > ........so, where could be the problem now? as you can see from the list of ip addresses found by openser, you have no network interface configured with 192.168.251.11. It is why you got errors highlighted in the previous messages. Now, it seems that there is another process using port 5070. Check if it is openser already running. Do "netstat -alut" and see if some application is running on port 5070. Cheers, Daniel > > On 6/21/06, *Daniel-Constantin Mierla* <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>> wrote: > > > > On 06/21/06 12:44, Shenen Shenen wrote: > > No,this isn't the problem;I've changed the port on my openser like > > 5070, and asterisk is set like 5060 in sip.conf, but I've always the > > same error; > > > > Jun 21 03:33:43 asterisk12 openser[19037]: ERROR: udp_init: bind(6, > > > > 0x8119a1c, 16) on 192.168.251.11 <http://192.168.251.11> > <http://192.168.251.11/> < > > http://192.168.251.11/>: Cannot assign > > requested address > > > > .....what could be? > > > do you have a network interface configured with the IP > 192.168.251.11 <http://192.168.251.11>? > Do 'ifconfig' to check it out. > > You can remove the 'listen' parameters from your config file and > openser > will bind to available interfaces -- do not forget to set fork=yes, > otherwise openser will listen to the first interface it founds, > and that > is usually 127.0.0.1 <http://127.0.0.1>. > > Daniel > > > > > > > > > > > On 6/21/06, *Daniel-Constantin Mierla* <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro> > > <mailto:daniel@voice-system.ro <mailto:daniel@voice-system.ro>>> > wrote: > > > > You can safely ignore reverse resolve DNS messages, they are > warnings > > meaning no hostname was found set for those IP addresses - > openser > > tries > > to get them in the aliases list. The problem comes from line: > > > > Jun 21 03:33:43 asterisk12 openser[19037]: ERROR: udp_init: > bind(6, > > 0x8119a1c, 16) on 192.168.251.11 <http://192.168.251.11> > <http://192.168.251.11 <http://192.168.251.11>> < > > http://192.168.251.11/>: Cannot assign > > requested address > > > > This means that you do not have the IP 192.168.251.11 > <http://192.168.251.11> > > <http://192.168.251.11> configured on the > > system. Do you have asterisk and openser listening on the > same port > > (5060)? You must use different ports. > > > > Cheers, > > Daniel > > > > > > On 06/21/06 10:55, Shenen Shenen wrote: > > > Hi!This my openser.cfg;I've 2 asteriskathome with openser > > installed;I > > > want to copy the authentificated sip registration from one > box to > > > another and have always the same Sip registered in all 2 > box, so if > > > one box falls down the other can run without reinitialized > all the > > > sips (like Phoner..cubix...and also IP phones...)....I'm > using > > > vrrpd...12 is the master, 11 is the slave, 200 is the virtual > > IP(this > > > point to the master in the first case..and all Sip proxies > like the > > > phoner or cubix points to the master..(I can set only 1 > proxy.. > > > 192.168.251.200 <http://192.168.251.200> > <http://192.168.251.200> > > <http://192.168.251.200/> ..in x-lite I can set more > > > proxies and I'vent problems..)(for this I must use openser..) > > > > > > > > > This is the openser on the "slave" 192.168.251.11 > <http://192.168.251.11> > > <http://192.168.251.11> <http://192.168.251.11/> > > > > > > > > > # SCRIPT PER COPIARE LO STATO DELLE REGISTRAZIONI DEI SIP DAL > > .11 al .12 > > > > > > # ----------- global configuration parameters > > ------------------------ > > > > > > debug=3 # debug level (cmd line: -dddddddddd) > > > fork=no > > > log_stderror=no # (cmd line: -E) > > > listen= 192.168.251.12 <http://192.168.251.12> > <http://192.168.251.12> > > <http://192.168.251.12/> > > > listen=192.168.251.11 <http://192.168.251.11> > <http://192.168.251.11> < > > http://192.168.251.11/> > > > listen=192.1668.251.200 > > > port=5060 > > > children=5 > > > dns=no > > > rev_dns=no > > > # ------------------ module loading > > ---------------------------------- > > > > > > loadmodule "modules/mysql/mysql.so" > > > loadmodule "modules/sl/sl.so" > > > loadmodule "modules/tm/tm.so" > > > loadmodule "modules/maxfwd/maxfwd.so" > > > loadmodule "modules/usrloc/usrloc.so" > > > loadmodule "modules/registrar/registrar.so" > > > loadmodule "modules/auth/auth.so" > > > loadmodule "modules/auth_db/auth_db.so" > > > > > > # ----------------- setting module-specific parameters > > --------------- > > > > > > # digest generation secret; use the same in backup server; > > > # also, make sure that the backup server has sync'ed time > > > modparam("auth", "secret", "alsdkhglaksdhfkloiwr") > > > > > > # ------------------------- request routing logic > > ------------------- > > > > > > # main routing logic > > > > > > route{ > > > > > > # initial sanity checks -- messages with > > > # max_forwars==0, or excessively long requests > > > if (!mf_process_maxfwd_header("10")) { > > > sl_send_reply("483","Too Many Hops"); > > > return; > > > }; > > > if (len_gt( max_len )) { > > > sl_send_reply("513", "Message too big"); > > > return; > > > }; > > > > > > # 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") { > > > > > > # verify credentials > > > if (!www_authorize("", "subscriber")) { > > > www_challenge("", "0"); > > > return; > > > }; > > > > > > # if ok, update contacts and ... > > > save("location"); > > > # ... if this REGISTER is not a replica from our > > > # peer server, replicate to the peer server > > > if (!src_ip== 192.168.251.12 <http://192.168.251.12> > <http://192.168.251.12> > > < http://192.168.251.12/>) { > > > t_replicate("192.168.251.12 <http://192.168.251.12> > <http://192.168.251.12> > > <http://192.168.251.12/>", "5060"); > > > }; > > > return; > > > }; > > > # do whatever else appropriate for your domain > > > log("non-REGISTER\n"); > > > }; > > > } > > > > > > > > > > > > This is the openser on the "master" 192.168.251.12 > <http://192.168.251.12> > > <http://192.168.251.12> > > > <http://192.168.251.12/ > > > > > > > > > > > > > > > > > > > # SCRIPT PER COPIARE LO STATO DELLE REGISTRAZIONI DEI SIP DAL > > .12 al .11 > > > > > > # ----------- global configuration parameters > > ------------------------ > > > > > > debug=3 # debug level (cmd line: -dddddddddd) > > > fork=no > > > log_stderror=yes # (cmd line: -E) > > > listen= 192.168.251.12 <http://192.168.251.12> > <http://192.168.251.12> < > > http://192.168.251.12/> > > > listen= 192.168.251.11 <http://192.168.251.11> > <http://192.168.251.11> > > <http://192.168.251.11/> > > > listen=192.1668.251.200 > > > port=5060 > > > children=5 > > > dns=no > > > rev_dns=no > > > > > > # ------------------ module loading > > ---------------------------------- > > > > > > loadmodule "modules/mysql/mysql.so" > > > loadmodule "modules/sl/sl.so" > > > loadmodule "modules/tm/tm.so" > > > loadmodule "modules/maxfwd/maxfwd.so" > > > loadmodule "modules/usrloc/usrloc.so" > > > loadmodule "modules/registrar/registrar.so" > > > loadmodule "modules/auth/auth.so" > > > loadmodule "modules/auth_db/auth_db.so" > > > > > > # ----------------- setting module-specific parameters > > --------------- > > > > > > # digest generation secret; use the same in backup server; > > > # also, make sure that the backup server has sync'ed time > > > modparam("auth", "secret", "alsdkhglaksdhfkloiwr") > > > > > > # ------------------------- request routing logic > > ------------------- > > > > > > # main routing logic > > > > > > route{ > > > > > > # initial sanity checks -- messages with > > > # max_forwars==0, or excessively long requests > > > if (!mf_process_maxfwd_header("10")) { > > > sl_send_reply("483","Too Many Hops"); > > > return; > > > }; > > > if (len_gt( max_len )) { > > > sl_send_reply("513", "Message too big"); > > > return; > > > }; > > > > > > # 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") { > > > > > > # verify credentials > > > if (!www_authorize("", "subscriber")) { > > > www_challenge("", "0"); > > > return; > > > }; > > > > > > # if ok, update contacts and ... > > > save("location"); > > > # ... if this REGISTER is not a replica from our > > > # peer server, replicate to the peer server > > > if (!src_ip== 192.168.251.11 <http://192.168.251.11> > <http://192.168.251.11> > > < http://192.168.251.11/>) { > > > t_replicate("192.168.251.11 <http://192.168.251.11> > <http://192.168.251.11> > > <http://192.168.251.11/>", "5060"); > > > }; > > > return; > > > }; > > > # do whatever else appropriate for your domain > > > log("non-REGISTER\n"); > > > }; > > > } > > > > > > If I run the following command... I've problems with > resolution > > of IP > > > addresses... how can I resolve this problem?????????? > > > > > > 100000000000000000000000000000000000000000000000000000 > > > thanks.....................help me please!!!!!I'm becoming > crazy... > > > > > > > > > [ root@asterisk12 ~]# tail -f /var/log/messages > > > Jun 21 03:33:43 asterisk12 openser: WARNING: > fix_socket_list: could > > > not rev. resolve 192.168.251.11 <http://192.168.251.11> < > http://192.168.251.11> > > <http://192.168.251.11/> > > > Jun 21 03:33:43 asterisk12 openser: WARNING: > fix_socket_list: could > > > not rev. resolve 192.168.251.200 <http://192.168.251.200> > <http://192.168.251.200> > > <http://192.168.251.200/> > > > Jun 21 03:33:43 asterisk12 openser[19037]: Maxfwd module- > > initializing > > > Jun 21 03:33:43 asterisk12 openser[19037]: AUTH module - > > initializing > > > Jun 21 03:33:43 asterisk12 openser[19037]: AUTH_DB module - > > initializing > > > Jun 21 03:33:43 asterisk12 openser[19037]: INFO: udp_init: > > SO_RCVBUF > > > is initially 110592 > > > Jun 21 03:33:43 asterisk12 openser[19037]: INFO: udp_init: > SO_RCVBUF > > > is finally 221184 > > > Jun 21 03:33:43 asterisk12 openser[19037]: INFO: udp_init: > SO_RCVBUF > > > is initially 110592 > > > Jun 21 03:33:43 asterisk12 openser[19037]: INFO: udp_init: > SO_RCVBUF > > > is finally 221184 > > > Jun 21 03:33:43 asterisk12 openser[19037]: ERROR: > udp_init: bind(6, > > > 0x8119a1c, 16) on 192.168.251.11 <http://192.168.251.11> < > http://192.168.251.11> > > <http://192.168.251.11/>: Cannot > > > assign requested address > > > Jun 21 03:34:01 asterisk12 crond(pam_unix)[19040]: session > > opened for > > > user root by (uid=0) > > > Jun 21 03:34:01 asterisk12 crond(pam_unix)[19040]: session > > closed for > > > user root > > > > > > > > >

1 0

[Users] mediaproxy.ini with 2 networks
by Andrew And 22 Jun '06

22 Jun '06

I have 2 alias IP (one IP to real network and another IP to network 10.0.0.0) in Proxy SIP, in ProxyIP (file mediaproxy.ini) I need to put the IP. How can I put this 2 IPs in ProxyIP (file mediaproxy.ini)?

1 0

[Users] Re: Final ACK getting wrong IP address
by Justin Pearce 22 Jun '06

22 Jun '06

Hey Guys, I did a bit more poking at the problem. It seems almost like the call process just breaks down in the signaling. All the packets seem to get to their destination up to the last ACK packet before the two clients start broadcasting their audio and video to one another. All other SIP packets after that seem to also wind up with the private IPs put on them. I put together the following scripting to be put down in the route section: #This conditional is to try and redeem ACKs that some how managed to get a private address attached to them. #This is basically a hack to try and "normalize" the packets coming through and pray they work. if(method=="ACK" || method=="BYE" || method=="INFO"){ if(uri=~"^sip:192\.168\.[0-9]+\.[0-9]+[:]*[0-9]*" || uri=~"^sip:10\.[0-9]+\.[0-9]+\.[0-9]+[:]*[0-9]*" || uri=~"^sip:172\.(16|17|18|19|20|21|22|23|24|25|26|27|28|29|30|31)\.[0-9] +\.[0-9]+[:]*[0-9]*") { #Rewrite the R-URI of the packet to the To-URI in the To header. subst_uri('/^sip:[0-9]+\.[0-9]+\.[0-9]+\.[0-9]+[:]*[0-9]*$/$tu/ig'); #Force the server to perform a lookup on the new URI and try and route it properly. lookup("location"); #Strip out the Record-Route remove_hf("Record-Route"); #There may be more magic needing to be inserted here for additional mangling of the packet. #<insert cuisinart here> }; }; This, while not pretty, at least routes the packets, but the clients are being picky about the things. Client A and Client B register to the server Client A calls Client B...Call Fails Client B calls Client A after the previous call...Call Connects All other calls back and forth are fine until the clients unregister. Could there be something I'm missing? I dunno if I hit on a bug or something, but it seems rather odd that everything up to that one point is fine. Thanks, Justin Pearce Information Technology/Programming Price Video Productions JustinP(a)PriceVideo.com 361-572-3810 800-733-3810 Fax: 361-572-3894 www.PriceVideo.com

1 0

[Users] help
by Justin Pearce 22 Jun '06

22 Jun '06

Justin Pearce Information Technology/Programming Price Video Productions Justin(a)PriceVideo.com <mailto:Justin@PriceVideo.com> 361-572-3810 800-733-3810 Fax: 361-572-3894 www.PriceVideo.com

1 0

[Users] FWD's Outbound Proxy?
by Enky 22 Jun '06

22 Jun '06

Someone to know how the FWD's Outbound Proxy (fwdnat.pulver.com:5082) works? And what software they use? I have made some tests with different NATs and it works perfect with all of them. Is it possible to build such kind of Outbound Proxy, based on OpenSER? I will appreciate any help or comment.

2 1

[Users] Seeking OpenSER Guru
by Kelvin Williams 22 Jun '06

22 Jun '06

I'm seeking an OpenSER Guru who I could pay (via PayPal) to answer a few questions by telephone today. Thanks, Kw Kelvin(a)williamschadwell.com

1 0

RE: [Users] Out of Memory
by Douglas Garstang 22 Jun '06

22 Jun '06

Bogdan, I finally managed to upload the memory dumps to pastebin. Links are: Hopefully you can access them. Memory dump right after OpenSER was started: http://pastebin.com/723872 Memory dump after OpenSER running for 20min: http://pastebin.com/723890 Memory dump after OpenSER running for 50min: http://pastebin.com/723902 The problem started to occur between the 20 and 50 minute samples. About 20 calls had been processed in that time. Douglas. > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, June 21, 2006 10:14 AM > To: Douglas Garstang > Cc: Users(a)openser.org > Subject: Re: [Users] Out of Memory > > > As written in the wiki, if you do not know how to interpret > the result, > you should post them on a ftp or http server and send the link to the > list - also, important is to mention in which context did you get the > mem dump - is the shutdown dump, a run time dump... > > regards, > bogdan > > > Douglas Garstang wrote: > > >Once I get the memory status, how do I interpret the results? > > > > > > > >>-----Original Message----- > >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >>Sent: Wednesday, June 21, 2006 9:43 AM > >>To: Douglas Garstang > >>Cc: Users(a)openser.org > >>Subject: Re: [Users] Out of Memory > >> > >> > >>Hi Douglas, > >> > >>it was signal SIGUSR1 and not SIGHUP - I strongly advice you > >>to go through > >> http://openser.org/dokuwiki/doku.php?id=memory > >>note you have to recompile and reinstall in order to be able to do > >>memory debug. > >> > >>also does not matter how much mem. your system has - openser > >>will use by > >>default only 1M per process. Refer to the wiki for how to > increase it. > >> > >>Regards, > >>Bogdan > >> > >> > >> > >

2 1

RE: [Users] Out of Memory
by Douglas Garstang 22 Jun '06

22 Jun '06

> -----Original Message----- > From: Douglas Garstang > Sent: Wednesday, June 21, 2006 12:48 PM > To: Bogdan-Andrei Iancu > Cc: Users(a)openser.org > Subject: RE: [Users] Out of Memory > > > Bogdan, > > I finally managed to upload the memory dumps to pastebin. Links are: > Hopefully you can access them. > > Memory dump right after OpenSER was started: > http://pastebin.com/723872 > Memory dump after OpenSER running for 20min: > http://pastebin.com/723890 > Memory dump after OpenSER running for 50min: > http://pastebin.com/723902 > > The problem started to occur between the 20 and 50 minute > samples. About 20 calls had been processed in that time. So... I'm wondering if anything has been determined with this...

1 0

[Serusers] CDRTool / asterisk billing based on realtime
by hgaillac-sip＠yahoo.fr 22 Jun '06

22 Jun '06

Hello, I read ag-projects it does'not provide support for its cdrtool pseudo gpl licensed . It's one of the bad open source project even seen . No documentation ... A real bad project !! Don't waste time with that Anybody could advise me a billing system based on cdr records with realtime support cdr_obdc.so? Regards Harry ___________________________________________________________________________ Yahoo! Mail réinvente le mail ! Découvrez le nouveau Yahoo! Mail et son interface révolutionnaire. http://fr.mail.yahoo.com

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[Serusers] query about acc module
by Aman Sachdeva 22 Jun '06

22 Jun '06

sir, i am install ser sucessfully and my two user connect and they can talk easilly using xlite(softphone),but i have faced one problem that i could not store the information about call information in acc table of ser database. i store the user location information in the database but not call inforation.i load all the module which is needed to accounting table such as(acc &tm). i am sending my configuration file. # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- 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" #SER_SQL_URI="mysql:*//ser:heslo@localhost/ser"* # ------------------ module *loading* ---------------------------------- # Uncomment this if you want to *use* SQL database loadmodule "/usr/local/ser-0.9.6/modules/mysql/mysql.so" loadmodule "/usr/local/ser-0.9.6/modules/sl/sl.so" loadmodule "/usr/local/ser-0.9.6/modules/tm/tm.so" loadmodule "/usr/local/ser-0.9.6/modules/rr/rr.so" loadmodule "/usr/local/ser-0.9.6/modules/maxfwd/maxfwd.so" loadmodule "/usr/local/ser-0.9.6/modules/usrloc/usrloc.so" loadmodule "/usr/local/ser-0.9.6/modules/registrar/registrar.so" loadmodule "/usr/local/ser-0.9.6/modules/textops/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/ser-0.9.6/modules/auth/auth.so" loadmodule "/usr/local/ser-0.9.6/modules/auth_db/auth_db.so" loadmodule "/usr/local/ser-0.9.6/modules/acc/acc.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) # -- acc params -- # *set* the reporting *log* level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion *status* will be reported modparam("acc", "log_flag", 1 ) modparam("usrloc|acc|auth_db|group|msilo|uri","db_url","mysql:*//ser:heslo@localhost/ser")* # ------------------------- 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 >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # 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(); setflag(1); # 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("192.168.1.7", "subscriber")) { www_challenge("192.168.1.7", "0"); break; }; setflag(1); *save*(*"location"*); record_route(); break; }; lookup("*alias*es"); 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; }; }; setflag(1); append_hf("P-hint: usrloc applied\r\n"); record_route(); route(1); } route[1] { # send it out now; *use* stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } i am waiting for ur replly Regards aman sachdeva sachdeva.aman(a)gmail.com

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RE: [Serusers] Mediaproxy problem - accounting
by İlker Aktuna (Koç.net) 22 Jun '06

22 Jun '06

What is a working configuration for this ? Thanks... ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Thursday, June 22, 2006 4:21 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem - accounting Hi, I am trying to use dbaccounting with mediaproxy. Undfortunately it doesn't seem to send anything to mysql db. How can I trace where the problem is ? My mediaproxy accounting config is : [Accounting] dbaccounting = on [Database] user = ser password = heslo host = 127.0.0.1 database = radius table = radacct Thanks, ilker _____________________________________________________________________________________________________________________________________________ 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. _____________________________________________________________________________________________________________________________________________ <http://387555.sigclick.mailinfo.com/sigclick/02010207/07074903/000C074E/002…> _____________________________________________________________________________________________________________________________________________ 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] openser error with resolve IP and Address already in use
by Shenen Shenen 22 Jun '06

22 Jun '06

I've this error with openser; [root@asterisk11 ~]# openser 0(3176) WARNING: fix_socket_list: could not rev. resolve 192.168.251.11 0(3176) WARNING: fix_socket_list: could not rev. resolve 192.168.251.11 Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.251.11 [ 192.168.251.11]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.251.11 [ 192.168.251.11]:5060 Aliases: tcp: asterisk11.local:5060 udp: asterisk11.local:5060 stateless - initializing [root@asterisk11 ~]# 0(0) Maxfwd module- initializing 0(0) AUTH module - initializing 0(0) AUTH_DB module - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) ERROR: tcp_init: bind(6, 0x811b0a4, 16) on 127.0.0.1:5060 : Address already in use and this is my openser.cfg; # SCRIPT PER COPIARE LO STATO DELLE REGISTRAZIONI DEI SIP DAL .11 al .12 # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=no # (cmd line: -E) listen=192.168.251.12 listen=192.168.251.11 listen=192.1668.251.200 port=5060 children=5 dns=no rev_dns=no # ------------------ module loading ---------------------------------- loadmodule "modules/mysql/mysql.so" loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/usrloc/usrloc.so" loadmodule "modules/registrar/registrar.so" loadmodule "modules/auth/auth.so" loadmodule "modules/auth_db/auth_db.so" # ----------------- setting module-specific parameters --------------- # digest generation secret; use the same in backup server; # also, make sure that the backup server has sync'ed time modparam("auth", "secret", "alsdkhglaksdhfkloiwr") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); return; }; # 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") { # verify credentials if (!www_authorize("", "subscriber")) { www_challenge("", "0"); return; }; # if ok, update contacts and ... save("location"); # ... if this REGISTER is not a replica from our # peer server, replicate to the peer server if (!src_ip== 192.168.251.12) { t_replicate("192.168.251.12", "5060"); }; return; }; # do whatever else appropriate for your domain log("non-REGISTER\n"); }; } I'm using openser on asteriskAtHome, I've changed the port on asterisk like 5065...but I've this problems....I've also make this: export SIP_DOMAIN="192.168.251.12" and the others...with and without the " " I've warmings and 1 error....the IP are all up and works...so what could be? 0(0) ERROR: tcp_init: bind(6, 0x811b0a4, 16) on 127.0.0.1:5060 : Address already in use 100000000000000000000000000000000000000000000000000000000000 thanks

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[Users] Openser as proxy
by Jimmy Montano 22 Jun '06

22 Jun '06

Greetings we are running a SIP server provided by Portaone, don't know what they did to the config but their SIP server (I guess it's IPtel's SER) but it doesn't allow eyebeam softphone to use peer-to-peer presence info. It's there any way for us to use Openser as a proxy to Portaone's SER so that users may place call and also have presence information available? Thanks in advance Jimmy

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[Serusers] Mediaproxy problem - accounting
by İlker Aktuna (Koç.net) 22 Jun '06

22 Jun '06

Hi, I am trying to use dbaccounting with mediaproxy. Undfortunately it doesn't seem to send anything to mysql db. How can I trace where the problem is ? My mediaproxy accounting config is : [Accounting] dbaccounting = on [Database] user = ser password = heslo host = 127.0.0.1 database = radius table = radacct Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/060F0F04/0D034500/0403014D/0A1…> _____________________________________________________________________________________________________________________________________________ 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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[Users] audio problem in NAT helper..
by raviprakash sunkara 22 Jun '06

22 Jun '06

Hi Users, I'm cracking my head with audio problem.. I installed openser 1.0.1 in Linux + freeRadius + RtpProxy + mysql When I make a call ( which are behind the nat ) the audio is not hearing and audio signal of Softphone is blinking up and down.. In Debug output I didn't see any SIP reply or sip request for ACK.. I thing so that, If client nehind the NAT didn't receive the ACK. this may be problem... SIP/2.0 200 Ok v: SIP/2.0/UDP 192.168.2.21:5060;received=61.17.248.68 ;rport=4007;branch=z9hG4bK9615579CC17F47929BB2FEF3F33F4BFC f: ravi <sip:1003@24.98.157.209>;tag=145161195 t: <sip:104@24.98.157.209>;tag=641313084 m: <sip:104@61.17.248.68:5060> Record-Route: <sip:192.168.2.4;ftag=145161195;lr=on> i: 839BBA5C-84B5-4D09-ACF1-B19FA0EB3C59(a)192.168.2.21 CSeq: 54483 INVITE c: application/sdp Server: X-Lite release 1105x Content-Length: 166 v=0 o=104 86794725 86799157 IN IP4 192.168.2.20 s=X-Lite c=IN IP4 192.168.2.4 t=0 0 m=audio 35034 RTP/AVP 1 a=rtpmap:1 pcmu/8000 a=sendrecv a=nortpproxy:yes --------------------- > this is problem openSER server didn't get request ACK I think So.. What may the problem... plz help me -- Thanks and Regards w ith cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

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Re: [Serusers] Caller-ID & Remote-Party-ID example
by Greger V. Teigre 22 Jun '06

22 Jun '06

I'm not very familiar with freeradius, but in general there is only one dictionary, sometimes with some sort of #include-support if you want to split up the dictionary. g-) Kostas Marneris wrote: > thanks for your help. > I hadn't understand all the Radius issues.. > > I should add the following line on FreeRadius Dictionary . > ATTRIBUTE Sip-Rpid 213 string # Proprietary, auth_radius > > Is this the proper way or should I have a separate dictionary file ? > > > thank you. > > > Greger V. Teigre wrote: > >> The SIP-Rpid attribute will be returned as part of the RADIUS auth if >> present (0.9.x). In 0.10.x I believe it is SIP-AVP=rpid:value >> This is the "saved value". >> g-) >> >> Kostas Marneris wrote: >> >> >>> Hello all, >>> >>> I'm trying to get 'caller-id with Remote-Party-ID' work, >>> but I think I'm missing something.. >>> >>> What is the "saved value of the SIP URI received from radius" ? >>> (according to auth/README file). >>> >>> >>> What is supposed to be the RADIUS attribute/value in RADIUS user profile ? >>> >>> >>> Can someone send an example ? >>> >>> >>> thank you. >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >>> >> >> > >

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RE: [Serusers] How to send instant message programmatically
by srinivas.sriram＠wipro.com 22 Jun '06

22 Jun '06

Hello Ilker, I am writing to address of the sip message to fifo and from the fifo call sipp instead of mysql. But , I do not now how to send message from sipp when the user is online. can you please help in sending message from sipp to user agent(to address). Thanks & Regards Sriram Srinivas ________________________________ From: İlker Aktuna (Koç.net) [mailto:ilkera@koc.net] Sent: Monday, June 19, 2006 7:06 PM To: Srinivas Sriram (WT01 - IP-Multimedia Carrier & Ent Networks) Subject: RE: [Serusers] How to send instant message programmatically Hi Sriram, I am also interested in sending missed calls as IM when the user becomes online. Do you know how to write missed calls to a mysql table ? If that's done, we can send messages via sipp when the user is online. Please share with me your findings... Regards, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of srinivas.sriram(a)wipro.com Sent: Monday, June 19, 2006 3:37 PM To: serusers(a)iptel.org Subject: RE: [Serusers] How to send instant message programmatically Hi Greger I have checked with sipp , with that we can send messages. But What is my requirement is i have write the code for constructing SIP "MESSAGE" message for a given sip uri. This I have to call form one of the ser module. Total what I am trying is to send a IM message when there is a missed call for a particular SIP uri. In SER there is a code for writing into the data base of the sip uri , when there is a missed call for that Uri. At this point we have to send IM message. Thank you, Regards, Sriram Srinivas. ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, June 19, 2006 5:35 PM To: Srinivas Sriram (WT01 - IP-Multimedia Carrier & Ent Networks) Cc: serusers(a)iptel.org Subject: Re: [Serusers] How to send instant message programmatically You are better off generating the message outside SER. SER is not a UAS. Have a look at sipsak or sipp (both found at sourceforge, I think). g-) srinivas.sriram(a)wipro.com wrote: Hello, I am using SER-0.9.6. How to send instant message programmatically, how to build sip message and which function can by used(like t_uac, t_request). Can you help where and which to be changed in ser code. Thank you, Regards, Sriram Srinivas. 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 ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers 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 <http://387555.sigclick.mailinfo.com/sigclick/000A0702/02074C00/0305054B/051…> _____________________________________________________________________________________________________________________________________________ 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. _____________________________________________________________________________________________________________________________________________ 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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[Users] Any way to steal Media IP/Port from RTPProxy or MediaProxy
by Hoa Thai Duy 22 Jun '06

22 Jun '06

Hi List I want to get to c= and m= value after use_media_proxy or force_rtp_proxy (after real RTP flow between UA and media/rtpproxy) I want this in order to steal this pair of information, and bypass the RTPProxy/MediaProxy and use this information for UA to talk with other application server (eg. Asterisk) Pls. help Brgds

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[Users] Differentiating between SIP and PSTN invites
by Kenny Chua 22 Jun '06

22 Jun '06

Hello, I was wondering, how should I implement this. I would like the user to press the #key first if they would like to dial SIP 2 SIP calls and just leave PSTN calls as it is? How am I suppose to do that? I have this problem too, which whenever I hangup, the other side doesn't gets disconnected. What should I check for to enable hang up on both sides if one sides suddenly hangs up? Thank you. --------------------------------- Want to be your own boss? Learn how on Yahoo! Small Business.

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RE: [Serusers] checking source ip and dest ip
by İlker Aktuna (Koç.net) 22 Jun '06

22 Jun '06

Thank you; that was what I'm looking for. How can I match i.e IP block 10.224.0.0 - 10.255.255.255 ? Is there an easy way or should I list them with || as in : (uri=~"^.*@10\.224\..* <mailto:uri=~^.*@10\.224\..*> " || uri=~"^.*@10\.225\..*" || .....) Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Wednesday, June 21, 2006 12:50 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Check the request uri after lookup, right before t_relay(). (uri=~"^.*@10\..*" <mailto:^.*@10\..*> ) As long as you don't have nated clients (and thus don't use fix_nated_register()) the request uri will contain the actual IP address. g-) İlker Aktuna (Koç.net) wrote: Hi, In fact my network is not 10.0.0.0 ; that was just an example (bad example). The network addresses are specific public network addresses (lets say 3 different C Class network) and has no relation to NAT. That's why I need to check the IP adrresses (from and to) of the SIP message before deciding if it should use mediaproxy or not. Any idea how to do that ? Soruce address can be checked as : if (src_ip="195.1.1.+") { } but that won't be same for destination address as "dst_ip" is the SER server ip. How can I check the ip address of the client that the SIP message will be forwarded ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 10:26 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Yes, and your network is in the RFC1918 (private) address space, so you can use the same logic (even without change as you want to proxy when the address is on your network). g-) İlker Aktuna (Koç.net) wrote: In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not. what do you mean by RFC1918 addresses ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 1:40 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses. The problem has already been solved, why not reuse? ;-) g-) İlker Aktuna (Koç.net) wrote: Hi, I need to check source ip of a SIP message and also destination IP to which the message will be sent. i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0 is it possible ? Thanks, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not. what do you mean by RFC1918 addresses ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 1:40 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses. The problem has already been solved, why not reuse? ;-) g-) İlker Aktuna (Koç.net) wrote: Hi, I need to check source ip of a SIP message and also destination IP to which the message will be sent. i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0 is it possible ? Thanks, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, In fact my network is not 10.0.0.0 ; that was just an example (bad example). The network addresses are specific public network addresses (lets say 3 different C Class network) and has no relation to NAT. That's why I need to check the IP adrresses (from and to) of the SIP message before deciding if it should use mediaproxy or not. Any idea how to do that ? Soruce address can be checked as : if (src_ip="195.1.1.+") { } but that won't be same for destination address as "dst_ip" is the SER server ip. How can I check the ip address of the client that the SIP message will be forwarded ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 10:26 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Yes, and your network is in the RFC1918 (private) address space, so you can use the same logic (even without change as you want to proxy when the address is on your network). g-) İlker Aktuna (Koç.net) wrote: In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not. what do you mean by RFC1918 addresses ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 1:40 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses. The problem has already been solved, why not reuse? ;-) g-) İlker Aktuna (Koç.net) wrote: Hi, I need to check source ip of a SIP message and also destination IP to which the message will be sent. i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0 is it possible ? Thanks, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not. what do you mean by RFC1918 addresses ? Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/070C0F04/02074D00/0507084D/051…> _____________________________________________________________________________________________________________________________________________ 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

[Users] help me .
by raviprakash sunkara 22 Jun '06

22 Jun '06

hi user... What means " t_newtrans :- transaction already in process " and " I'm terribly sorry, server error occurred ". How to resovle this.... Please help me... -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

1 0

[Users] OpenSER - one year old
by buxalex＠libero.it 22 Jun '06

22 Jun '06

Hello to everybody, I would like to say.... Happy Birhtday Openser!!!! :O) Sometimes I got answers with delay, sometimes I spent nights trying to understand how to configure a module, or even a single statement, somrtimes I needed to read ten times the same document because it was unclear related to my knolwledge and still I want to thank the community for this great product. Every day I try to read all the emails coming from the community, and every day I learn a little bit more (threads, ideas, suggestions,solutions...). By the way, there is any open source for a sip softphone in windows (minisip even doesnt start on my laptop with windows 2000) :o)) thanks Alessandro

2 1

[Serusers] Mediaproxy has nothing to do with codecs..
by Dave 22 Jun '06

22 Jun '06

Can someone please clarify whether the statement in Subject is correct? Does mediaproxy not simply take a RTP stream and "proxy" it in both directions without really caring about codecs? Codec negotiation is done between endpoints and mediaproxy does not come in the picture. If the above is true, does it mean that if two SIP user agents support video and we use mediaproxy, it won't be an issue? Please educate me the what if any relationship mediaproxy has with codecs... Dave __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

Re: [Users] 499 Not Acceptable here
by Kenny Chua 22 Jun '06

22 Jun '06

Thanks Andreas, that was really the problem. Really got to watch out for codecs. --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.

1 0

Re: [Users] keep alive using RE-INVITE
by Javier Ramirez 21 Jun '06

21 Jun '06

Bogdan, I don't understand how to make this with openser. but I understand the keepalive mechanism can you help me? best regards Javier Ramirez ----- Original Message ----- From: Bogdan-Andrei Iancu To: Javier Ramirez Sent: Wednesday, June 21, 2006 1:35 PM Subject: Re: [Users] keep alive using RE-INVITE Javier, I guess the RFC3261 or google (SIP + reINVITE) are the best source for looking into. regards, bogdan Javier Ramirez wrote: > Bogdan, > I read that a lot of person use the RE-INVITE to keepalive the call > but I don´t understand this. > please, can you help me ? > best regards > Javier Ramirez > > > ----- Original Message ----- > *From:* Bogdan-Andrei Iancu <mailto:bogdan@voice-system.ro> > *To:* Javier Ramirez <mailto:jvr_78@yahoo.com.ar> > *Cc:* users(a)openser.org <mailto:users@openser.org> > *Sent:* Thursday, June 08, 2006 7:14 AM > *Subject:* Re: [Users] keep alive using RE-INVITE > > Hi Javier, > > this is quite impossible to do - by sending re-INVITEs from a middle > point, you will break the CSEQ consistency between the end points. > Within the dialog, the cseq must increase from message to message > - if > you send a re-invite to point A with cseq n+1, after a while point > B may > sent a request with the same cseq since it was not aware that some > party > in the middle changed the number. > > the best way to do it is if you have a end2end keepalive mechanism as > Session-Timer. > > regards, > bogdan > > Javier Ramirez wrote: > > > Hi ! > > I looking for a mechanism to keep alive, and I see that is used the > > RE-INVITE, but > > How to use the Re-INVITE to a keep alive mechanism ? > > exist another way to implement another keep alive mechanism ? > > > > best regards > > Javier ramirez > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > __________ NOD32 1.1586 (20060608) Information __________ > > This message was checked by NOD32 antivirus system. > http://www.eset.com > __________ NOD32 1.1611 (20060620) Information __________ This message was checked by NOD32 antivirus system. http://www.eset.com

1 0

[Users] FW: ACK problems
by Glenn Dalgliesh 21 Jun '06

21 Jun '06

It seems that for some reason openser(555.111.555.111:5060) will not forward the ACK packet in this scenario. Config at bottom. ACKs from other clients seem to get forwarded just fine. Have tried both 1.0.1 and CVS-head same results.. Thanks 111.222.111.222:5060 Openser (Statefull) sending us a call 555.111.555.111:5060 Openser (Statefull) 77.159.77.105:9193 Eyebeam Client ACK is in SIP Message 9 which openser should then send to 77.159.77.105:9193 SIP MESSAGE 1 111.222.111.222:5060() -> 555.111.555.111:5060() UDP Frame 1 21/Jun/06 16:13:12.8910 TimeFromPreviousSipFrame=0.0000 TimeFromStart=0.0000 INVITE sip:9999999019@555.111.555.111:5060 SIP/2.0 Record-Route: <sip:9999999019@111.222.111.222:5060;nat=yes;ftag=783386-fdb1078445624;lr=on > Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 To: <sip:9999999019@111.222.111.222:5060> Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE Server: Sansay VSX 2.1 Contact: <sip:9999992264@666.111.82.202:5060;transport=udp> Max-Forwards: 69 Content-Type: application/sdp Content-Length: 283 v=0 o=sansay-VSX 10 10 IN IP4 206.57.80.14 s=session controller c=IN IP4 111.222.111.222 t=0 0 m=audio 63226 RTP/AVP 18 0 8 100 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=ptime:20 ________________________________________ SIP MESSAGE 2 555.111.555.111:5060() -> 111.222.111.222:5060() UDP Frame 2 21/Jun/06 16:13:12.8933 TimeFromPreviousSipFrame=0.0023 TimeFromStart=0.0023 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 To: <sip:9999999019@111.222.111.222:5060> Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE Server: OpenSer (1.0.1 (i386/linux)) Content-Length: 0 Warning: 392 555.111.555.111:5060 "Noisy feedback tells: pid=16606 req_src_ip=111.222.111.222 req_src_port=5060 in_uri=sip:9999999019@555.111.555.111:5060 out_uri=sip:9999999019@77.159.77.105:9193 via_cnt==2" ________________________________________ SIP MESSAGE 3 555.111.555.111:5060() -> 77.159.77.105:9193() UDP Frame 3 21/Jun/06 16:13:12.8934 TimeFromPreviousSipFrame=0.0001 TimeFromStart=0.0024 INVITE sip:9999999019@77.159.77.105:9193 SIP/2.0 Record-Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> Record-Route: <sip:9999999019@111.222.111.222:5060;nat=yes;ftag=783386-fdb1078445624;lr=on > Via: SIP/2.0/UDP 555.111.555.111;branch=z9hG4bKdbc7.30da3132.0 Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 To: <sip:9999999019@111.222.111.222:5060> Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE Server: Sansay VSX 2.1 Contact: <sip:9999992264@666.111.82.202:5060;transport=udp> Max-Forwards: 68 Content-Type: application/sdp Content-Length: 283 P-hint: usrloc applied v=0 o=sansay-VSX 10 10 IN IP4 206.57.80.14 s=session controller c=IN IP4 111.222.111.222 t=0 0 m=audio 63226 RTP/AVP 18 0 8 100 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=ptime:20 ________________________________________ SIP MESSAGE 4 77.159.77.105:9193() -> 555.111.555.111:5060() UDP Frame 4 21/Jun/06 16:13:12.9413 TimeFromPreviousSipFrame=0.0479 TimeFromStart=0.0503 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 555.111.555.111;branch=z9hG4bKdbc7.30da3132.0 Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 Record-Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> Record-Route: <sip:9999999019@111.222.111.222:5060;lr;nat=yes;ftag=783386-fdb1078445624> Contact: <sip:9999999019@77.159.77.105:9193> To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 0 ________________________________________ SIP MESSAGE 5 555.111.555.111:5060() -> 111.222.111.222:5060() UDP Frame 5 21/Jun/06 16:13:12.9426 TimeFromPreviousSipFrame=0.0013 TimeFromStart=0.0516 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 Record-Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> Record-Route: <sip:9999999019@111.222.111.222:5060;lr;nat=yes;ftag=783386-fdb1078445624> Contact: <sip:9999999019@77.159.77.105:9193> To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 0 ________________________________________ SIP MESSAGE 6 77.159.77.105:9193() -> 555.111.555.111:5060() UDP Frame 7 21/Jun/06 16:13:13.9012 TimeFromPreviousSipFrame=0.9586 TimeFromStart=1.0102 Extra Information: Packet is not a complete SIP message ________________________________________ SIP MESSAGE 7 77.159.77.105:9193() -> 555.111.555.111:5060() UDP Frame 8 21/Jun/06 16:13:15.1016 TimeFromPreviousSipFrame=1.2003 TimeFromStart=2.2106 SIP/2.0 200 OK Via: SIP/2.0/UDP 555.111.555.111;branch=z9hG4bKdbc7.30da3132.0 Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 Record-Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> Record-Route: <sip:9999999019@111.222.111.222:5060;lr;nat=yes;ftag=783386-fdb1078445624> Contact: <sip:9999999019@77.159.77.105:9193> To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 235 v=0 o=- 519699122 519699160 IN IP4 77.159.77.105 s=eyeBeam c=IN IP4 77.159.77.105 t=0 0 m=audio 6902 RTP/AVP 0 101 a=alt:1 1 : BA47AC39 000000CF 77.159.77.105 6902 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ________________________________________ SIP MESSAGE 8 555.111.555.111:5060() -> 111.222.111.222:5060() UDP Frame 9 21/Jun/06 16:13:15.1037 TimeFromPreviousSipFrame=0.0021 TimeFromStart=2.2127 SIP/2.0 200 OK Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 Record-Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> Record-Route: <sip:9999999019@111.222.111.222:5060;lr;nat=yes;ftag=783386-fdb1078445624> Contact: <sip:9999999019@77.159.77.105:9193> To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 235 v=0 o=- 519699122 519699160 IN IP4 77.159.77.105 s=eyeBeam c=IN IP4 77.159.77.105 t=0 0 m=audio 6902 RTP/AVP 0 101 a=alt:1 1 : BA47AC39 000000CF 77.159.77.105 6902 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ________________________________________ SIP MESSAGE 9 111.222.111.222:5060() -> 555.111.555.111:5060() UDP Frame 10 21/Jun/06 16:13:15.2282 TimeFromPreviousSipFrame=0.1245 TimeFromStart=2.3372 ACK sip:9999999019@555.111.555.111:5060 SIP/2.0 Record-Route: <sip:111.222.111.222;ftag=783386-fdb1078445624;lr=on> Via: SIP/2.0/UDP 111.222.111.222;branch=0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK78338620-bdb1078445624 Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 ACK Server: Sansay VSX 2.1 Max-Forwards: 69 Content-Length: 0 ________________________________________ SIP MESSAGE 10 77.159.77.105:9193() -> 555.111.555.111:5060() UDP Frame 11 21/Jun/06 16:13:15.5921 TimeFromPreviousSipFrame=0.3639 TimeFromStart=2.7011 SIP/2.0 200 OK Via: SIP/2.0/UDP 555.111.555.111;branch=z9hG4bKdbc7.30da3132.0 Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 Record-Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> Record-Route: <sip:9999999019@111.222.111.222:5060;lr;nat=yes;ftag=783386-fdb1078445624> Contact: <sip:9999999019@77.159.77.105:9193> To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 235 v=0 o=- 519699122 519699160 IN IP4 77.159.77.105 s=eyeBeam c=IN IP4 77.159.77.105 t=0 0 m=audio 6902 RTP/AVP 0 101 a=alt:1 1 : BA47AC39 000000CF 77.159.77.105 6902 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ________________________________________ SIP MESSAGE 11 555.111.555.111:5060() -> 111.222.111.222:5060() UDP Frame 12 21/Jun/06 16:13:15.5942 TimeFromPreviousSipFrame=0.0020 TimeFromStart=2.7032 SIP/2.0 200 OK Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 Record-Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> Record-Route: <sip:9999999019@111.222.111.222:5060;lr;nat=yes;ftag=783386-fdb1078445624> Contact: <sip:9999999019@77.159.77.105:9193> To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 235 v=0 o=- 519699122 519699160 IN IP4 77.159.77.105 s=eyeBeam c=IN IP4 77.159.77.105 t=0 0 m=audio 6902 RTP/AVP 0 101 a=alt:1 1 : BA47AC39 000000CF 77.159.77.105 6902 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ________________________________________ SIP MESSAGE 12 111.222.111.222:5060() -> 555.111.555.111:5060() UDP Frame 13 21/Jun/06 16:13:15.6761 TimeFromPreviousSipFrame=0.0819 TimeFromStart=2.7850 ACK sip:9999999019@555.111.555.111:5060 SIP/2.0 Record-Route: <sip:111.222.111.222;ftag=783386-fdb1078445624;lr=on> Via: SIP/2.0/UDP 111.222.111.222;branch=0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK78338620-bdb1078445624 Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 ACK Server: Sansay VSX 2.1 Max-Forwards: 69 Content-Length: 0 -*-*-*-*-*-*-*-*-*-*-*- route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # 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("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

1 0

FW: [Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723
by Richard C. Thompson 21 Jun '06

21 Jun '06

As far as I know mediaproxy does just that, it proxies the call but doesn't change the codec so if it comes in g729 it goes out the same way. It doesn't do transcoding. Most of my calls are g729 and you can see these calls by doing a sessions.py in the mediaproxy server Richard C. Thompson VP Technology, CCNA Virtual Communications, LLC -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kouprianov Sent: Wednesday, June 21, 2006 4:31 AM To: serusers(a)iptel.org Subject: Re: [Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723 How about making clients change the default codec to smth else? On 6/21/06, Benjamin.George(a)t-systems.com <Benjamin.George(a)t-systems.com> wrote: > > > > Hi, > > In my understanding, you can't configure media proxy for g729 as it won't > support g729 codec. It is a patented codec and SER media proxy as well as > SEMS supports only the patent free codecs like g711a/u or iLBC, etc. As a > result of this you will get some choppy sound only if you use g729. So I > feel, you have to modify the code in order to incorporate g729 with media > proxy. > > Regards, > > Benjamin. > > > ________________________________ > > > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of ravi > reddy > Sent: Wednesday, June 21, 2006 1:02 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723 > > > > > Hi , > I am using ser-0.9.6 and mediaproxy-1.4.2 and the are working > fine til now but when i use the sip-sip phone (or) sip - pstn in the peak > hours i cant hear voice very clearly its choppy sound , > > when i tried to call grandstream sip phone to pstn i made > grandstream phone to use only g-729 but mediaproxy is showing session in > active > > can any body tell me how to change mediaproxy's g-711a to some thing like > g-729 or g-723 so that i can hear clear voice in peak hours too > > > thanks you very much > > regards > ravi. > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

FW: [Serusers] calls status at SER
by Richard C. Thompson 21 Jun '06

21 Jun '06

I use a mediaproxy server with ser so if you do a sessions.py it will show you all the current calls Richard C. Thompson VP Technology, CCNA Virtual Communications, LLC _____ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Atif Rasheed Sent: Wednesday, June 21, 2006 8:26 AM To: serusers(a)iptel.org Subject: [Serusers] calls status at SER Hi all, We are doing stateful routing from SER. I need to know how can we check simultanous calls going through SER, total no. of calls processed etc. Like we check current channels in asterisk or we see statistics in gnugk. appriciate the help. Thank you -- Atif Rasheed

1 0

[Users] mediaproxy.ini (more than 1 IP)
by Andrew And 21 Jun '06

21 Jun '06

When I am using mediaproxy I need to write the server address IP in ProxyIP of the file mediaproxy.ini. But when I have 2 or more alias in my interface of the server how can I put more than 1 IP?

1 0

[Serusers] location information
by Alois Schafferhofer 21 Jun '06

21 Jun '06

Hi, I have a problem with sending position coordinates with PUBLISH. If I send the x and y coordinates <x>01524.3441</x><y>04704.1020</y> via PUBLISH, SER cuts the last two digits (<x>1524.34</x><y>4704.1</y>). Maybe, someone could tell me in which class or in which method this happens and what I have to do. Best regards, Alois /usr/sbin/ser[11700]: handle publish request /usr/sbin/ser[11700]: get_pres_uri: _puri=sip:pda_user@fh-joanneum.at /usr/sbin/ser[11700]: handle_publish -4- p_uri= p_uri.len=23 /usr/sbin/ser[11700]: handle_publish -5- presentity=0x4069d1d8 /usr/sbin/ser[11700]: parse_pidf: sipuri=0x8216278:pres:pda_user@fh-joanneum.at contact=0x8216098:sip:pda_user@10.0.0.3 basic=0x82160b8:online location=(nil):(null) /usr/sbin/ser[11700]: parse_pidf: site=(nil):(null) floor=(nil):(null) room=(nil):(null) /usr/sbin/ser[11700]: parse_pidf: x=0x821be28:01524.3441 y=0x821be38:04704.1020 radius=(nil):(null) /usr/sbin/ser[11700]: find_presence_tuple: _p=0x4069d1d8 _p->tuples=0x4069d488 /usr/sbin/ser[11700]: publish_presentity_pidf: -1- /usr/sbin/ser[11700]: publish_presentity: -2- /usr/sbin/ser[11700]: publish_presentity: -3-: changed=1 /usr/sbin/ser[11700]: db_update_presentity starting: use_place_table=1 presid=4 /usr/sbin/ser[11700]: db_update_presentity: tuple- >contact=sip:pda_user@10.0.0.3 len=21 basic=1 expires=1150916054 priority=0.500000 /usr/sbin/ser[11700]: db_update_presentity: cleaning contact from table /usr/sbin/ser[11700]: db_update_presentity: room= loc= /usr/sbin/ser[11700]: publish_presentity: -4- /usr/sbin/ser[11700]: publish_presentity: event_package=1 -1- /usr/sbin/ser[11700]: handle_publish -8- paerrno=0 /usr/sbin/ser[11700]: handled publish /usr/sbin/ser[11708]: timer_presentity: _p=0x4069d1d8 pda_user(a)fh-joanneum.at flags=1 watchers=0x4069d3a8 /usr/sbin/ser[11708]: notifying esca(a)fh-joanneum.at _p->flags=1 _w- >event_package=1 _w->accept=2 _w->status=1 PUBLISH sip:pda_user@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK874819433..Max-Forwards: 70..From: <sip:pda_user@fhjoanneum .at>;tag=540877786..To: <sip:pda_user@fh-joanneum.at>..Call-ID: 996402808-569651771- 701518565..CSeq: 10 PUBLISH..Contact: <sip:pda_user@10.0.0.3>..Con tent-Type: application/cpim-pidf+xml..Content-Length: 396..Expires: 50..Event: presence..User- Agent: SIP .NET 1.0 evaluation version, www.independents oft.com....<?xml version="1.0" encoding="utf-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" entity="pres:pda_user@fh-joanneum.at"><tuple id="9da019 ef-d0ea-4de0-8ad3-edacae0ca89c"><status><basic>online</basic><geopriv><locationinfo>< civilAddress><x>01524.3441</x><y>04704.1020</y></civilAddress></ location-info></geopriv></status><contact priority="0.8">sip:pda_user@10.0.0.3</contact></tuple></presence>.. 10.15.200.57:5060 -> 10.15.200.136:5060 NOTIFY sip:esca@10.15.200.136 SIP/2.0..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bK766a.c7db8933.0..To: <sip:esca@fhjoanneum. at>;tag=z9hG4bK29818902. .From: <sip:pda_user@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-f2af..CSeq: 10 NOTIFY..Call-ID: 603541788327@10.15.200.136..Content-Length: 464..User-Agent: Sip EXpress router(0.9.3 (i386/linux))..Event: presence..Content-Type: application/pidf+xml..Subscription-State: active;expires=3479. ...<?xml version="1.0"?>..<!DOCTYPE presence PUBLIC "//IETF//DTD RFCxxxx PIDF 1.0//EN" "pidf.dtd">..<presence entity="sip:pda_user@fh-joanneum.at">..< tuple id="9r28r49">tid6e65cf14">.. <contact priority="0.500000">sip:pda_user@10.0.0.3</contact>.. <status>.. <basic>open</basic>.. <geopriv>< location-info><civilAddress> <x>1524.34</x>.. <y>4704.1</y>.. </civilAddress></location-info></geopriv>.. </status>.. </status>..</tupl e>..</presence>..

1 0

[Users] 499 Not Acceptable here
by Kenny Chua 21 Jun '06

21 Jun '06

Hi all, I'm somehow getting this message when I try to dial from SIP 2 SIP. It says 499 Not acceptable here. Any help will be appreciated. Also, I was wondering how do I setup rtpproxy on another machine. How am I suppose to link it from my openser machine to the other server. RECEIVE TIME: 18379350 RECEIVE << 70 .xxx.xxx.xxx:5060 SIP/2.0 488 Not Acceptable Here Call-ID: AD7821E1-85B3-47DB-9EC8-ACF50039FC3E(a)42.77.149.86 From: "Test "<sip:200@test.org>;tag=1917233790 To: <sip:203@test.org>;tag=2d5d4 CSeq: 50639 INVITE Via: SIP/2.0/UDP 130.70.11.16:5060;rport;branch=z9hG4bK9847221F640D42E0B8F2F08835F57F28 Record-Route: <sip:192.168.1.5> Content-Length: 0 Openser.cfg debug=3 fork=yes log_stderror=no listen=192.168.1.5 alias=test.org port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" 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/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); return; }; if (method=="INVITE") { if (!proxy_authorize("test.org","subscriber")) { proxy_challenge("test.org","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); return; }; if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (uri=~"^sip:00[0-9]*@") { # International PSTN xlog("International PSTN\n"); strip(2); route(4); route(5); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- xlog("Routing to PSTN gateway\n"); rewritehostport("pstn.net:5060"); avp_write("i:45", "inv_timeout"); route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com --------------------------------- Yahoo! Sports Fantasy Football 06 - Go with the leader. Start your league today!

2 1

RE: [Users] Out of Memory
by Douglas Garstang 21 Jun '06

21 Jun '06

Oh, well you know what, I don't have access to system with a publically available ftp or http server. I'm trying to use pastbin.com, but they seem to be having a problem right now. > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, June 21, 2006 10:14 AM > To: Douglas Garstang > Cc: Users(a)openser.org > Subject: Re: [Users] Out of Memory > > > As written in the wiki, if you do not know how to interpret > the result, > you should post them on a ftp or http server and send the link to the > list - also, important is to mention in which context did you get the > mem dump - is the shutdown dump, a run time dump... > > regards, > bogdan > > > Douglas Garstang wrote: > > >Once I get the memory status, how do I interpret the results? > > > > > > > >>-----Original Message----- > >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >>Sent: Wednesday, June 21, 2006 9:43 AM > >>To: Douglas Garstang > >>Cc: Users(a)openser.org > >>Subject: Re: [Users] Out of Memory > >> > >> > >>Hi Douglas, > >> > >>it was signal SIGUSR1 and not SIGHUP - I strongly advice you > >>to go through > >> http://openser.org/dokuwiki/doku.php?id=memory > >>note you have to recompile and reinstall in order to be able to do > >>memory debug. > >> > >>also does not matter how much mem. your system has - openser > >>will use by > >>default only 1M per process. Refer to the wiki for how to > increase it. > >> > >>Regards, > >>Bogdan > >> > >> > >> > >

2 1

[Users] user 'admin' and mysql
by Mark Kent 21 Jun '06

21 Jun '06

Hello, I just noticed that openser_mysql.sh creates the username "admin" with the default openserrw password in the subscriber table. This seems to introduce a security hole where a well-known username and password pair would exist on most virgin openser installations. Is there a good reason to have that entry in the "subscriber" table? Is it used anywhere? Now I know that we're supposed to change the mysql access passwords, but I have to admit that I didn't think to change a password actually emebedded IN the data of the mysql database. Did I miss a critical security note somewhere alerting me to this default user? Thanks, -mark

4 5

RE: [Users] Out of Memory
by Douglas Garstang 21 Jun '06

21 Jun '06

Once I get the memory status, how do I interpret the results? > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, June 21, 2006 9:43 AM > To: Douglas Garstang > Cc: Users(a)openser.org > Subject: Re: [Users] Out of Memory > > > Hi Douglas, > > it was signal SIGUSR1 and not SIGHUP - I strongly advice you > to go through > http://openser.org/dokuwiki/doku.php?id=memory > note you have to recompile and reinstall in order to be able to do > memory debug. > > also does not matter how much mem. your system has - openser > will use by > default only 1M per process. Refer to the wiki for how to increase it. > > Regards, > Bogdan > > Douglas Garstang wrote: > > >I posted this last week. Didn't get much of a reply. > > > >I have three identical systems all running identical > versions of OpenSER 1.1.x > >One of the systems is showing memory problems. The system > has 4GB of RAM. > >I am only loading a few modules: > > > >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/textops.so" > >loadmodule "/usr/local/lib/openser/modules/xlog.so" > >loadmodule "/usr/local/lib/openser/modules/dispatcher.so" > > > >The system runs for a short period of time and processes > almost no calls before the errors start to occur. > >One of the developers said I could send a HUP signal to > OpenSER to get memory stats. Tried that. It didn't work. > > > >Jun 21 09:29:35 apollo /usr/local/sbin/openser[15623]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 426 > >Jun 21 09:29:35 apollo /usr/local/sbin/openser[15623]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 426 > >Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: > ERROR: dns_cname_parser: out of memory > >Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: > ERROR: get_record: rdata parse error > >Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: > ERROR: get_record > >Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: > ERROR: build_req_buf_from_sip_req: out of memory > >Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: > ERROR:tm:print_uac_request: no pkg_mem > >Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: > ERROR:tm:t_forward_nonack: failure to add branches > >Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 442 > >Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 442 > >Jun 21 09:29:37 apollo /usr/local/sbin/openser[15618]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:38 apollo /usr/local/sbin/openser[15617]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:38 apollo /usr/local/sbin/openser[15619]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:39 apollo /usr/local/sbin/openser[15620]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:39 apollo /usr/local/sbin/openser[15621]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:39 apollo /usr/local/sbin/openser[15624]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:40 apollo /usr/local/sbin/openser[15622]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:40 apollo /usr/local/sbin/openser[15618]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:40 apollo /usr/local/sbin/openser[15617]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:41 apollo /usr/local/sbin/openser[15620]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:41 apollo /usr/local/sbin/openser[15623]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:41 apollo /usr/local/sbin/openser[15621]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:41 apollo /usr/local/sbin/openser[15624]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:42 apollo /usr/local/sbin/openser[15622]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:42 apollo /usr/local/sbin/openser[15618]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:42 apollo /usr/local/sbin/openser[15617]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > >Jun 21 09:29:43 apollo /usr/local/sbin/openser[15619]: > ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 > > > >Doug. > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > >

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[Users] database access from a module
by Simon Morvan 21 Jun '06

21 Jun '06

Hello there, Is it possible to use database functions provided by mysql or postgresql modules from a module i'm writing ? I don't see any documented functions. -- Simon Morvan.

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[Users] Out of Memory
by Douglas Garstang 21 Jun '06

21 Jun '06

I posted this last week. Didn't get much of a reply. I have three identical systems all running identical versions of OpenSER 1.1.x One of the systems is showing memory problems. The system has 4GB of RAM. I am only loading a few modules: 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/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/dispatcher.so" The system runs for a short period of time and processes almost no calls before the errors start to occur. One of the developers said I could send a HUP signal to OpenSER to get memory stats. Tried that. It didn't work. Jun 21 09:29:35 apollo /usr/local/sbin/openser[15623]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 426 Jun 21 09:29:35 apollo /usr/local/sbin/openser[15623]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 426 Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: ERROR: dns_cname_parser: out of memory Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: ERROR: get_record: rdata parse error Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: ERROR: get_record Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: ERROR: build_req_buf_from_sip_req: out of memory Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: ERROR:tm:print_uac_request: no pkg_mem Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: ERROR:tm:t_forward_nonack: failure to add branches Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 442 Jun 21 09:29:37 apollo /usr/local/sbin/openser[15624]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 442 Jun 21 09:29:37 apollo /usr/local/sbin/openser[15618]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:38 apollo /usr/local/sbin/openser[15617]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:38 apollo /usr/local/sbin/openser[15619]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:39 apollo /usr/local/sbin/openser[15620]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:39 apollo /usr/local/sbin/openser[15621]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:39 apollo /usr/local/sbin/openser[15624]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:40 apollo /usr/local/sbin/openser[15622]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:40 apollo /usr/local/sbin/openser[15618]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:40 apollo /usr/local/sbin/openser[15617]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:41 apollo /usr/local/sbin/openser[15620]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:41 apollo /usr/local/sbin/openser[15623]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:41 apollo /usr/local/sbin/openser[15621]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:41 apollo /usr/local/sbin/openser[15624]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:42 apollo /usr/local/sbin/openser[15622]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:42 apollo /usr/local/sbin/openser[15618]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:42 apollo /usr/local/sbin/openser[15617]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Jun 21 09:29:43 apollo /usr/local/sbin/openser[15619]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 379 Doug.

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[Users] Radius AccountingStop strangeness (replaced calling directions on BYE)
by Pletli Antal 21 Jun '06

21 Jun '06

Hi guys! I'm trying the acc module in radius mode. When the caller (A) calls the callee (B) and (A) hangs up the conversation then the AccountingStop message is correct: the caller-station-id = (A) and the called-station-id = (B). But when (A) calls the (B) and (B) hangs up the conversation, the AccountingStop is incorrect because in stop message the caller-station-id = (B) and the called-station-id = (A) Is this a bug or feature? I think this is because of BYE message header values, but this is'nt a correct behavior. Is there any solution to correct this problem? Best regards, Antal Pletli

2 1

[Users] IP authentication in openser using database
by Tanzeel serfaraz 21 Jun '06

21 Jun '06

Hi users , i want to know that how can i give access to initiate a call to those which are my trusted user without authentication.(trusted user have trusted ip )\ regards Tanzeel --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.

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[Serusers] [Windows Messenger SIP 5.1] Problem with "Send an instant message"
by Malik NakaMura 21 Jun '06

21 Jun '06

Hi, My configuration : - Windows Messenger 5.1 SIP installed on 2 PCs (Windows XP) - Ser v0.9.6 installed on Mandrake 10.1 The problem is that I have this error : "The following message could not be delivered to all recipients : my message" when I send an instant message, but the voice call works very well... with Ethereal I see that the instant message receiver send these requests to Ser : - Trying - OK - BYE why does the receiver send BYE whereas the transaction is not finished ? my traces from ethereal : from Ser(192.168.10.137) to Receiver(192.168.10.127) : INVITE sip:192.168.10.127:10835 SIP/2.0 from Receiver(192.168.10.127) to Ser(192.168.10.137) : SIP/2.0 100 Trying SIP/2.0 200 OK BYE sip:192.168.10.137;ftag=6a036c30432a4e0dab11dda94916e9ea;lr=on SIP/2.0 from Ser(192.168.10.137) to Receiver(192.168.10.127) : ACK sip:192.168.10.127:10835 SIP/2.0 MESSAGE sip:192.168.10.127:10835 SIP/2.0 SIP/2.0 200 OK from Receiver(192.168.10.127) to Ser(192.168.10.137) : SIP/2.0 481 Call Leg/Transaction Does Not Exist thanks in advance, Best regards, Malik

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[Serusers] binding has expired
by Giuseppe Parlato 21 Jun '06

21 Jun '06

Hi I get really often this log error from the same user user, Binding '67800072','sip:6780000072@151.8.212.151:5060' has expired why? Jay

4 4

[Serusers] calls status at SER
by Atif Rasheed 21 Jun '06

21 Jun '06

Hi all, We are doing stateful routing from SER. I need to know how can we check simultanous calls going through SER, total no. of calls processed etc. Like we check current channels in asterisk or we see statistics in gnugk. appriciate the help. Thank you -- Atif Rasheed

1 0

[Users] mediaproxy
by Roberto Pioli 21 Jun '06

21 Jun '06

I want use openser with mediaproxy and I want to use rtp proxy in every situation. Do I have to call use_mediaproxy() before any t_relay?and where do I have to call end_media_session()? Roberto

1 0

[Serusers] Problem in exporting function
by Shashwat Gupta 21 Jun '06

21 Jun '06

Hello all, I am very new to ser I am using 8.14 version of ser ... I build up ser from source code available from CVS repositry. Actually I wanted to make a simple module for logging of Source name and host name that passes through ser (just for learning purpose) ...... I build up the module and placed it in ser's source module directory and recompiled the source .....Now problem is that when I try to call one exported function from ( i.e. siplog) from ser.cfg .... It gives me following error message ...... 0(12672) parse error (89,8-9): unknown command, missing loadmodule? when I did ser -dd -c I got following 0(14407) find_export: <siplog> not found 0(14407) find_export: <siplog> not found 0(14407) parse error (90,9-10): unknown command, missing loadmodule? So I think siplog() is not exported ....... could not find the reason why it is so because I am exporting it ........... My source code is following ----- #include "../../sr_module.h" #include <stdio.h> char * str_param; int int_param; MODULE_VERSION static int mod_init(void) ; static int siplog(struct sip_msg *msg,char *str,char *str2); static int mod_init(void) { fprintf(stdout,"initializing siplog module"); return 0; } static cmd_export_t cmds[] = { {"siplog",siplog,1,0,REQUEST_ROUTE}, {0,0,0,0,0} }; struct module_exports exports = { "siplog", /*module name */ cmds, /*cmd-name registered by this module */ 0, mod_init, 0, 0, 0, 0 }; static int siplog(struct sip_msg*msg,char*str,char*str2) { /*the basic logic of module goes here*/ } My ser.cfg file is as follows--- # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- 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/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/print.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" #added by annirudh loadmodule "/usr/local/lib/ser/modules/pa.so" #added by shashwat loadmodule "/usr/local/lib/ser/modules/siplog.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) # ------------------------- 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 >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; print(); siplog(); # 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] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } I would be very thankful for any kind of help .... Thanks ------------------------------------------------------------------------------------------------------------------------------------------------------------- result of nm siplog.so | grep ' T ' --------------------> 000009ec T _fini 00000748 T _init result of uname -a ------------> Linux server1.ims 2.4.20-8 #1 Thu Mar 13 17:54:28 EST 2003 i686 i686 i386 GNU/Linux reslut of ser -V ------------>flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.168.4.3.2.1 2004/10/22 17:21:32 andrei Exp $ main.c compiled on 16:06:05 Jun 19 2006 with gcc 3.2 -- regards shashwat gupta

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Re: [Serusers] How to start a VOIP business ?
by Roberto Pereyra 21 Jun '06

21 Jun '06

Argentine roberto 2006/6/21, ram <talk2ram(a)gmail.com>: > > Hi > > where are you located ? > > > ram > > > > On 6/21/06, Roberto Pereyra <pereyra.roberto(a)gmail.com> wrote: > > > > Subcription to any International whole sale VOIP > > carrier > > > > > > > > Please, somebody knows any to try ?. > > roberto pereyra > > 2006/6/21, Greger V. Teigre <greger(a)teigre.com>: > > > > There was an old SNMP module way back, but nobody wanted to maintain > it... > > It is most definitely a need. > > I know people use various monitoring tools like nagios. > > For version 0.10.x (CVS head), the new XMLRPC interface can be used for > > monitoring also, but it's more of a pull than push... > > g-) > > > > > > Benjamin.George(a)t-systems.com wrote: > > > > > > > > > > Hi, > > > > > > > > I doubt there are some monitoring/management tools already available or > > packed with SER. I am not very sure, whether SER components support SNMP > or > > not. But you can develop your own application by considering which are all > > the parameters (for example: Bandwidth usage, BHCA, call logs, etc) has to > > be monitored in a communication environment apart from normal system > > parameters like CPU usage, Memory consumption, Threads, I/O, etc. > > > > > > > > Normally what happens is, whoever supplies the network components provide > > the performance management/monitoring tools or adapters/api to integrate > > with other industry leading tools or support for SNMP. > > > > > > > > Or otherwise, you can use any good products which are available in the > > market (but they are not et-al freeware! L L). > > > > Some of tools are: > > > > BMC Patrol > > HP Open view > > CA Unicenter > > Mercury > > ProactiveNet > > > > Etc… > > > > > > > > If you have any idea about any free tools which can be used to monitor SER > > environment, please do let me know that. I am also curious to know/get > such > > a tool. > > > > > > > > Regards, > > > > Benjamin. > > > > ________________________________ > > > > > > From: ram [mailto:talk2ram@gmail.com] > > Sent: Tuesday, June 20, 2006 2:26 PM > > To: George, Benjamin > > Subject: Re: [Serusers] How to start a VOIP business ? > > > > > > > > > > Hi > > > > > > > > > > > > can you tell me where can get below said tools ? > > > > > > > > > > > > Any good performance management/monitoring tool (normally part of OSS) to > > measure and manage the performance of the entire infrastructure to assure > > zero/minimum downtime > > > > > > > > > > > > ram > > > > > > ________________________________ > > > > _______________________________________________ Serusers > > mailing list Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > -- > Ing. Roberto Pereyra > ContenidosOnline > Looking for Linux Virtual Private Servers ? Click here: > http://www.spry.com/hosting-affiliate/scripts/t.php?a_aid=426&a_bid=56 > _______________________________________________ > > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > -- Ing. Roberto Pereyra ContenidosOnline Looking for Linux Virtual Private Servers ? Click here: http://www.spry.com/hosting-affiliate/scripts/t.php?a_aid=426&a_bid=56

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RE: [Serusers] How to start a VOIP business ?
by Benjamin.George＠t-systems.com 21 Jun '06

21 Jun '06

Hi, I doubt there are some monitoring/management tools already available or packed with SER. I am not very sure, whether SER components support SNMP or not. But you can develop your own application by considering which are all the parameters (for example: Bandwidth usage, BHCA, call logs, etc) has to be monitored in a communication environment apart from normal system parameters like CPU usage, Memory consumption, Threads, I/O, etc. Normally what happens is, whoever supplies the network components provide the performance management/monitoring tools or adapters/api to integrate with other industry leading tools or support for SNMP. Or otherwise, you can use any good products which are available in the market (but they are not et-al freeware! :-( :-(). Some of tools are: 1. BMC Patrol 2. HP Open view 3. CA Unicenter 4. Mercury 5. ProactiveNet Etc... If you have any idea about any free tools which can be used to monitor SER environment, please do let me know that. I am also curious to know/get such a tool. Regards, Benjamin. _____ From: ram [mailto:talk2ram@gmail.com] Sent: Tuesday, June 20, 2006 2:26 PM To: George, Benjamin Subject: Re: [Serusers] How to start a VOIP business ? Hi can you tell me where can get below said tools ? Any good performance management/monitoring tool (normally part of OSS) to measure and manage the performance of the entire infrastructure to assure zero/minimum downtime ram

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