sr-users July 2010

sr-users@lists.kamailio.org

64 participants
87 discussions

userblacklist Module - Documentation fault?

by "Nicolas Rüger"

Hello, in the documentation of the Module "Userblacklist" seems not really clear to me. Please clarify my doubts. Refering to: http://www.kamailio.org/docs/modules/3.0.x/modules_k/userblacklist.html in "4.1. check_user_blacklist" is written: "[...]If a match is found and it is not set to whitelist, false is returned. Otherwise, true is returned.[...]" Question: 1.) Is "not whitelisted" equal to "blacklisted"? 2.) Why is "false" returned if the blacklist_check is positive? 3.) Is "true" even returned if there's no match? I'd have expected the following: if (user_is_blacklisted) then return TRUE; else return FALSE; Can you please clarify the behavior? Thank you. Regards, Nicolas Example that seems clear to me: =============================== In "4.2. check_user_whitelist" is written: "[...]If a match is found and it is set to whitelist, true is returned. Otherwise, false is returned.[...]" cause it means to me: if (user_is_whitelisted) then return TRUE; else return FALSE; -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

13 years, 8 months

Meeting in Chicago, Cluecon 2010, next week

by Daniel-Constantin Mierla

Hello, I will give a presentation about latest Kamailio - SIP Router news and usage together with FreeSwitch Media Server at Cluecon 2010 (http://www.cluecon.com), next week in Chicago. If you are around and want to meet during the event or out of it, drop me an email. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

13 years, 8 months

Kamailio Transformation Assignment to an AVP

by Brandon Armstead

Hello, I have the following transformation on the SDP Body: $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9 ]+)\015\012(.*)$/\3/s}) However when I assign this to an AVP, i.e. $avp(s:sdp-payloads) = $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9 ]+)\015\012(.*)$/\3/s}); I receive back a NULL result / transformation "regex does not match" is what I receive with high verbose syslog on kamailio. HOWEVER. When I do something like this: xlog("L_INFO", "Payloads Available: $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9 ]+)\015\012(.*)$/\3/s})"); It executes properly -- transformation regex matches and the payloads are displayed as expected. If anyone can provide any insight as to what I may be doing wrong it would be greatly appreciated. P.S. Kamailio SVN Revision # 2:5906M (1.5.2-notls). Also I CC'ed devel list (as I do believe this may be a bug). Thanks! Sincerely, Brandon Armstead

13 years, 8 months

Re: [SR-Users] how to set up iptel.org sip outbound

by Daniel-Constantin Mierla

Hello, please keep the mailing list cc-ed, private emails related to mailing lists discussions will not be answered. On 7/29/10 1:46 AM, j w wrote: > Hi Daniel, > > Thanks for your reply. Regarding set up outbound from ATA. I have > check with the Linksys tech support. They checked and said the > setting that they can do is all done, I have check with the outbound > provider. > > Can your outbound service allow me to dial out with a landline phone > via ATA. Would you please give me some setting information for your > outbound. Then it is not my service, I pointed you to a mailing list that is more appropriate for this subject. What I can give you as hints, try to put the ata in a network hub and sniff the sip traffic (with ngrep, wireshark, tcdump, etc...) to see if there is some attempt to send registrations/calls to sip server. Also, check maybe the device has some troubleshooting tool embedded (i.e., dedicated web page for logs). Cheers, Daniel > > Much obliged. > > Joseph > > ------------------------------------------------------------------------ > Date: Wed, 28 Jul 2010 10:54:24 +0200 > From: miconda(a)gmail.com > To: w7269826(a)hotmail.com > CC: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] how to set up iptel.org sip outbound > > Hello, > > On 7/19/10 8:35 PM, j w wrote: > > Hi, > > I registered iptel.org for an account and set up my Linksys > SPA3102 ATA gateway. The inbound is working but outbound cannot > dial out. > > First, the mailing list for iptel.org service is: > http://lists.iptel.org/mailman/listinfo/services > > You should get better answers there. > > Can any one help me: > 1. how to set up the sip outbound in order for me to dial out via ATA. > > > This has to do with ATA configuration, not much related to the server > side. You should have a user manual for that device. > > 2. how to get Canada phone number. > > > I don't think iptel.org is offering this, if you need for incoming, > you probably can get it from somewhere else and forward it to you > iptel.org accoung. > > Thanks a lot. > > I am not sure how to retrieve the post/mailing list, would you > please forward me your help reply to my > e-mail:w7269826@hotmail.com <mailto:e-mail:w7269826@hotmail.com>. > > > This is a public mailing list based on subscriptions. You have to > subscribe via web at: > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > Enter your email address, you will get a confirmation email with > further instructions. Once you follow them, you will receive all > emails sent to mailing list. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > -- Daniel-Constantin Mierla http://www.asipto.com/

13 years, 8 months

urgent :geographic number problem

by hala alramli

our geographic number cant receive calls from outbound (from normal landlines number) but it receive from our system numbers just ?where is the problem,is it from our server or from the company that provide us with this geographic number? please help me regards

13 years, 9 months

urgent :geographic number problem

by hala alramli

13 years, 9 months

multiple instances on different ports

by Robert R

Hi, I am trying to run multiple different instances of openser on the same server using different ports. Regardless of receiving INVITE on any of the ports (5061, 5062, ...) I want openser always to proxy the message to port 5060. invite:5061 ---> openser --> invite:5060 invite:5062 ---> openser --> invite:5060 By adding two listen ports as follow, openser sends the traffic to the same outgoing port as the incoming port: listen=udp:209.16.123.160:5061 listen=udp:209.16.123.160:5062 Is there a way to hard code the outgoing port? or I should overwrite the port? Thanks, R

13 years, 9 months

Missing fields in ACC

by David

Hello, I am doing db logging, and my ACCs are being added to the table, but it is missing the leg_src and leg_dst columns. I set the leg_src and leg_dst columns, I do this : modparam("acc", "multi_leg_info", "leg_src=$avp(s:caller);leg_dst=$avp(s:dialed)") In my routing script on the initial INVITE I do this : $avp(s:dialed) = $tu; $avp(s:caller) = $fu; xlog("L_WARNING", "Call Tracked - leg_dst=$avp(s:dialed) leg_src=$avp(s:caller)\n"); The xlog shows the leg_src and leg_dst, but they do not appear in the leg_dst and leg_src columns of my acc table. I have no warnings or errors ( other than "ERROR: ctl: could not delete unix socket /tmp/ser_ctl: Operation not permitted" ), but that's not worrysome. I do not use t_newtran(), t_check_trans() is called before I set the values, but this does not create a transaction. After the above code is called, I call t_relay(); Any idea what might cause the ACC record to be added without the multi_leg_info ? BTW, the avpops table in the database is a ENGINE=BLACKHOLE. As far as I can tell, the ACC is the same as my Kamailio 1.5.3 setup, but it would seem in migrating to 3.0, something broke, any ideas ? Thanks, David

13 years, 9 months

no signal in phone line ,server slow

by hala alramly

hello, we have a problem with our kamailio server sometimes it being very slow ,and i cant hear the signal in the phone line for a while then it fix him self ,when i return to syslog every thing normal, what is the reasons ? regards, hala _________________________________________________________________ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969

13 years, 9 months

how to set up iptel.org sip outbound

by j w

Hi, I registered iptel.org for an account and set up my Linksys SPA3102 ATA gateway. The inbound is working but outbound cannot dial out. Can any one help me: 1. how to set up the sip outbound in order for me to dial out via ATA. 2. how to get Canada phone number. Thanks a lot. I am not sure how to retrieve the post/mailing list, would you please forward me your help reply to my e-mail:w7269826@hotmail.com. Much obliged. Joseph _________________________________________________________________ MSN Dating: Find someone special. Start now. http://go.microsoft.com/?linkid=9734384

13 years, 9 months

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sr-users July 2010