sr-users March 2009

sr-users@lists.kamailio.org

92 participants
149 discussions

by Alex Balashov

Thanks, everyone! Very useful and excellent new features. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

15 years, 1 month

[Kamailio-Users] sqlops problem

by BERGANZ François

I follow the module instructions and I do : sql_query("ca", "select server from servers where etat_ping=oui & etat_asterisk=oui", "nb_srv"); $var(i) = 0; $var(s)=$dbr(nb_srv=>[server,$var(i)]); ### line 166 I have that error: ERROR:sqlops:sql_parse_index: bad number <server> ERROR:sqlops:pv_parse_dbr_name: invalid index in [[server,$var(i)]] ERROR:core:pv_parse_spec: pvar "dbr" has an invalid name param [nb_srv=>[server,$var(i)]] ERROR:core:pv_parse_spec: wrong char [)/41] in [$dbr(nb_srv=>[server,$var(i)])] at [29 (5)] CRITICAL:core:yyerror: parse error in config file, line 166, column 110-140: unknown script variable Anyone know why? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

15 years, 1 month

[Kamailio-Users] How to change request uri with Header value?

by Ramu

Hi All, I have situation where I need to change the request uri with To header value. Scenario: OpenServ will receive INVITE as follows INVITE sip:2000@open-ser-ip SIP/2.0. To: "2000"<sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com>>. Now I would like to change the request uri as follows: INVITE sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com> SIP/2.0. To: "2000"<sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com>>. and will do a lookup location. Ramu

15 years, 1 month

Re: [Kamailio-Users] Handling CANCEL and 487

by Daniel-Constantin Mierla

Hello, ok. I am cc-ing to mailing list to close the discussion thread. The issue was actually caused by a mis-configuration. Cheers, Daniel On 03/02/2009 05:53 PM, Juan Perez wrote: > Thank you Daniel, I applied that code and it is working now. > I appreciate it a lot. > cheers > jp > > ------------------------------------------------------------------------ > *From:* Juan Perez <jperezsip2008(a)yahoo.com> > *To:* Daniel-Constantin Mierla <miconda(a)gmail.com> > *Sent:* Friday, February 27, 2009 1:27:55 PM > *Subject:* Re: [Kamailio-Users] Handling CANCEL and 487 > > thank you a lot, I will do it and let you know > question: if the call is not cancelled, after the 200 OK there is an > ACK, why that ACK does not have any prob? > > > ------------------------------------------------------------------------ > *From:* Daniel-Constantin Mierla <miconda(a)gmail.com> > *To:* Juan Perez <jperezsip2008(a)yahoo.com> > *Sent:* Friday, February 27, 2009 11:36:17 AM > *Subject:* Re: [Kamailio-Users] Handling CANCEL and 487 > > > > On 02/27/2009 06:32 PM, Juan Perez wrote: > > thank you a lot, I am attaching the config file > in the main route block you have nothing that process the ack, so it > gets drop. > > Try to add: > > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > t_relay(); > exit; > } else { > exit; > } > } > > Daniel > -- Daniel-Constantin Mierla http://www.asipto.com

15 years, 1 month

[Serusers] fix_nated_contact onreply

by samuel

I have seen lots of default config files where in the reply route only after checking the message (client_nat_test(1)) fix_nated is called. Why is not called when the NAT flag is set upong lookup_XX? In ser-oob I think the reply_route should include the case of a user called behind a NAT and the reply is not fixed due to some router in the middle. Will it hurt including fix_nated_contact in the case of checking the flag? Friday thoughts... Samuel

15 years, 1 month

Re: [Kamailio-Users] how to know if my gw servers are active ?

by BERGANZ François

For the database, asterisk and SER work on the sameone. in sip.conf create one for SER and insecure=invite. It accept all from SER In the SER, configure the auth_db module to check the username col and password col. SER is a wall between client and asterisk(s). The same SER can be a wall between asterisk and provider. It is just a gymnastic! To test, it, the best is to have all in the same UC. And have one virtual networkInterface per Asterisk and SER. Cordialement, BERGANZ François http://www.acropolistelecom.net  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -----Message d'origine----- De : Abdul Hakeem [mailto:alhakeem100@hotmail.com] Envoyé : lundi 2 mars 2009 12:14 À : 'BERGANZ François' Objet : RE: [Kamailio-Users] how to know if my gw servers are active ? Hi, Thanks for the prompt reply. My comments are below: -----Original Message----- From: BERGANZ François [mailto:francois@acropolistelecom.net] Sent: 02 March 2009 07:59 To: alhakeem(a)ipextelecom.net Cc: users(a)lists.kamailio.org Subject: RE: [Kamailio-Users] how to know if my gw servers are active ? Abdul Hakeem, I use Asterisk-SER for another way. SER authenticate users and asterisk(s) accept all SIP frames from the SER. >>How did you create the users databases on Asterisk and Ser ? Also, how did you configure Asterisk to forward registration requests to Ser ? If you could send me sample Asterisk/Ser configuration, I'll be much grateful. Then, asterisk(s) forward the call to the provider. When a call come from the provider, asterisk forward it to the SER and the SER just do a lookup(location)! >>What happens if a call comes in to SER directly, & it has to do a quick lookup to see which Asterisk the user is currently located ? The reason I ask is because for example User-A with e164 number/DID +447931800952 might be registered to Asterisk-X now and 10 minutes later register with Asterisk-Y, Did you implement some refresh mechanism on the SER database ?. Because in this case inbound IP calls are hitting SER first before being redirected to the appropriate Asterisk server. I look forward to your reply. Regards, Abdul Hakeem Cordialement, BERGANZ François ? Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -----Message d'origine----- De : Abdul Hakeem [mailto:alhakeem@gmail.com] Envoyé : samedi 28 février 2009 09:41 À : 'BERGANZ François' Cc : users(a)lists.kamailio.org Objet : RE: [Kamailio-Users] how to know if my gw servers are active ? Hello, How did you setup your Asterisk to update SER/OpenSER/Kamalito's database ? I am trying to setup a solution for 2 Asterisk or more servers to allow dynamic and not static user registration. i.e. any user can register with any of the Asterisk servers. Let's say if UserA register with Asterisk1, this will reflect on SER's database where to route the call. In this scenario of UserA now register with Asterisk4 5 minutes later, SER's database will be automatically updated and calls to UserA will be automatically routed to Asterisk4. What I am hoping to achieve is not to do this via manual intervention. Any this will be much appreciated. Cheers, Abdul Hakeem -----Original Message----- From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of BERGANZ François Sent: 27 February 2009 10:24 To: 'Daniel-Constantin Mierla' Cc: users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] how to know if my gw servers are active ? Yes, my asterisk update a database to help the SER Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -----Message d'origine----- De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] Envoyé : vendredi 27 février 2009 11:19 À : BERGANZ François Cc : 'Henning Westerholt'; users(a)lists.kamailio.org Objet : Re: [Kamailio-Users] how to know if my gw servers are active ? On 02/27/2009 12:10 PM, BERGANZ François wrote: > I need to dispatch to the asterisk which have less users connected > How do you get the asterisk with the less users connected? You have some data in a db table? Cheers, Daniel > > Cordialement, > BERGANZ François > P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. > > > -----Message d'origine----- > De : Henning Westerholt [mailto:henning.westerholt@1und1.de] > Envoyé : vendredi 27 février 2009 11:07 À : BERGANZ François Cc : > users(a)lists.kamailio.org Objet : Re: [Kamailio-Users] how to know if > my gw servers are active ? > > On Friday 27 February 2009, BERGANZ François wrote: > >> If I cant use dispatcher module? >> Because, I don’t use the module algo to choose the dst. >> > > Well, then perhaps you need to use another solution, or adapt the dispatcher > code to your needs. May i ask what is the problem with existing algorithms? > What exactly do you want to achive? > > Cheers, > > Henning > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla http://www.asipto.com _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

15 years, 1 month

[Kamailio-Users] how to know if my gw servers are active ?

by BERGANZ François

hello, I have a list of servers in my database. I need that openser send OPTIONS to servers. for example, I could have a col 'active' to know if each server is active or not. In fact, I need to do an other style of dispatch, and, I cant use the dispatcher module! is there a solution for that ? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

15 years, 1 month

[Kamailio-Users] CANCEL over nat

by alexander merkulov

i have async rtp proxy setup on multihomed host. config for transfer cancel and invites: if(lookup("location")){ log("found loc"); fix_nated_contact(); $avp(i:11)=$fU; $var(fr)="sip:"+$avp(i:11)+"@69.70.173.195"; xlog("replace $var(fr)"); if (method == "BYE" || method == "CANCEL"){ unforce_rtp_proxy(); # force_rtp_proxy("AIEOC"); }else{ uac_replace_from("$avp(i:11)","$var(fr)"); if(force_rtp_proxy("FAIEOC")) t_on_reply("2"); } route(1); exit(); } cancel is routed ok, ATA get cancel in form of (ip changed): RECEIVING FROM: 69.70.xxx.xx:5060 CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK6e35.c727f225.0 From: "Unknown" <sip:Unknown@69.70.173.195 <sip%3AUnknown(a)69.70.173.195> >;tag=as704e6b0d Call-ID: 735a38dd3d77f4f20f29f28b335f11f4(a)192.168.1.180 To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>> CSeq: 102 CANCEL Max-Forwards: 70 asterisk Content-Length: 0 but ignore it. same with any other ATA type(including softphones) can u please explaine me what is incorrect in CANCEL? cancel produced by asterisk.CALL-id is correct. here is corespondoing invite 16:06:56.0 RECEIVING FROM: 69.70.173.195:5060 INVITE sip:34249@193.110.78.12:8484 SIP/2.0 Record-Route: <sip:69.70.173.195;r2=on;lr=on;ftag=as704e6b0d;last_from=bWVyYWxtZXJhbG1ldWFwdGRqaHVvZXJgaQ--> Record-Route: <sip:192.168.2.170;r2=on;lr=on;ftag=as704e6b0d;last_from=bWVyYWxtZXJhbG1ldWFwdGRqaHVvZXJgaQ--> Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK6e35.c727f225.0 Via: SIP/2.0/UDP 192.168.1.180:5060;rport=5060;branch=z9hG4bK49fa4d6d From: "Unknown" <sip:Unknown@69.70.173.195 <sip%3AUnknown(a)69.70.173.195> >;tag=as704e6b0d To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>> Contact: <sip:Unknown@192.168.1.180:5060> Call-ID: 735a38dd3d77f4f20f29f28b335f11f4(a)192.168.1.180 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Sun, 01 Mar 2009 14:05:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 447 v=0 o=root 8300 8300 IN IP4 69.70.173.195 s=session c=IN IP4 69.70.173.195 t=0 0 m=audio 10336 RTP/AVP 0 8 110 97 18 4 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes -- Merkulov Alexander

15 years, 1 month

[Kamailio-Users] dial the prefix #31# to mask the number

by ALAEDDINE abbech

Hi When i dial #31# 011111with a sip phone i have an INVITE in OPENSER with a number %2331%23 011111. Who know how can i send an INVITE whith prefix #31# using a phone or softphone. thanks,

15 years, 1 month

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sr-users March 2009