sr-users July 2005

sr-users@lists.kamailio.org

199 participants
544 discussions

by Sebastian Kühner

Hi there, Is there any way to forbid the users to registrate twice or more times to the same ser server with the same username (I want, that a user can only login once and if someone tries to login with the same username --> forbidden). Thanks! Sebastian

18 years, 12 months

[Serusers] ipphone---ser---asterisk---ser---pstn

by Iqbal

Hi I am setting up the following ipphone registered with ser, the username on ipphone belongs to special group called "asterisk" when he makes call, I append a prefix, and route his calls to asterisk. The prefix is appended because I have several "special" groups which all hit different extensions in asterisk. Anyhow this call gets to asterisk, where I then do a few mysql queries to find which group/company this user belongs to and then drop them into the correct context for their company. For any of these special users then need to dial 9[number] to route the call outside their corporate network, otherwise they can all dial internally using 3 digit numbers. The problem that I have got is that the calls get through okay, and both side can talk, but a) the call setup takes time b) the sip debug just does not seem correct, in fact its got way too much going on for my liking. setup, iphone ------ser------asterisk | | pstn GW ngrep -------- iphone invite to SER SER ---100 trying ---> ipphone SER --- INVITE ---> asterisk Asterisk ----> 100 trying ----> SER Atsreisk ----INVITE ----> SER SER -----100 trying -----> Asterisk SER -----INVITE ---PSTN GW PSTN GW ----100 trying -----> SER PSTN GW ----> 183 session progress ----> SER SER ---- 183 session progress ----> Asterisk Asterisk ----183--->ser ser .-----183 ---> ipphone PSTN GW ----200 OK ---> SER SER -----200 OK ----> Ast Ast ----ACK ----> SER SER ----ACK ---> GW AST ----OK ----> SER SER ----OK ----IPphone ipphone ----ACK ---> SER SER ------ACK ---AST Now around about here is where I think it should stop, cause it all seems to make sense...but heres where is starts to go wrong, I then get Ast ---INVITE ---> ser ser ----> 404 User Not found ----> ast ast ---ack ---> ser ser ---ack ---> gw ser ---invite --ast and various combos of this, but the call is going through. I am particularly concerned with the user not found part, since asterisk ip is trusted and the call does go through. Iqbal

18 years, 12 months

[Serusers] call limits?

by Adrian Georgescu

Try http://cdrtool.ag-projects.com On Jul 17, 2005, at 12:00 PM, serusers-request(a)lists.iptel.org wrote: > d to know how I can enforce a call limit for Hey guys, I need to know how I can enforce a call limit for my users.. if they already have x number of calls, I need to be able to block additional calls. Thanks, Chris Doyle

18 years, 12 months

[Serusers] Prepaid Again!

by Ah khng

Hi all, I'm a newbie as well (ser + Linux). Yet, this is another prepaid implementation question. Some told to use Asterisk astcc for prepaid application, so i plan to set the card number as the authentication name. Thus, the original ser database has to customize for integration with astcc database (at least auth_db). Can somebody help me in this for necessary changes? Or any opportunity to look at working database structure? Second, should i handling prepaid and postpaid in one ser proxy? if yes, what is the best approach? Thank Regards Seong

18 years, 12 months

[Serusers] unsolicited NOTIFY messages from Asterisk

by Subashini C V - CTD, Chennai

hi, i am getting unsolicited NOTIFY messages from Asterisk after the subscription. Is this type of NOTIFY messages is supported in any of the RFCs..? Because, the SIP stack with 3265 compliance, does not support any such NOTIFY messages and discarding those. I need the justification for this kind of NOTIFY messages sent by Asterisk. I need your valuable inputs... Thanks in Advance, Subashini DISCLAIMER This message and any attachment(s) contained here are information that is confidential, proprietary to HCL Technologies and its customers. Contents may be privileged or otherwise protected by law. The information is solely intended for the individual or the entity it is addressed to. If you are not the intended recipient of this message, you are not authorized to read, forward, print, retain, copy or disseminate this message or any part of it. If you have received this e-mail in error, please notify the sender immediately by return e-mail and delete it from your computer

18 years, 12 months

[Serusers] SERWEB edit NAME

by Kong

i had successfully install SERWEB 0.93, i am wondering how to let user edit their own first name and last name? if possible all other details that is in the subscriber field. thank you.

18 years, 12 months

[Serusers] Check_from() and check_to() issue

by Jon Mansey

Hi I seem to be having trouble with check_to() for registrations and check_from() for invites, even though I have them right after *_authorize and challenge calls. Maybe I don't fully understand the test that it does but it prevents registration from all my clients seemingly. What is the test failing on? Im getting 401 unauthorized. If I comment out the check_to() test, I register just fine. Could someone more fully explain the test criteria here so I can see where my problem is, thx. V0.9.3 Help appreciated, Jon

18 years, 12 months

[Serusers] SER Newbie Questions

by Lonnie Cumberland

Greetings All, I am new to the list so please forgive the newbie level questions, ok. Currently, we have 3 Asterisks PBX's locate throughout the world with our main webserver in the US and if I understand what SER can do for me then this might be the product that I need. We need something that will allow all of our users to have the same connection domains like (ie.. sip.examples.com) for example and which they can connect to SER which will re-route their connection to the closest PBX to handle their call at which point the PBX would take over the user connection and handle the call. Is this about correct as to what SER can do for us? If this is true, then is there some Turorials and HOWTO example documentation that I might read to see about getting SER set up on one of our contol servers? -- Thanks, Lonnie Cumberland OutStep Technologies Incorporated CELL: 313-333-2935 FAX: 619-639-2888

18 years, 12 months

[Serusers] Problem : Can SER process the reINVITE messages properly?

by Ricardo Martinez

Hello. I'm having problems trying to make SER, NAT'd endpoints and reINVITE work together. I was using the "gw-pstn3.07.cfg" file from onsip.org to do some tests, and this is what i have. In one side i have an Asterisk with an endpoint registered in it (let's call it A). In the other side i have a PAP2 under NAT (let's call it B). A ---------- Asterisk ----------- SER ----------- B (NAT'd) 200.0.0.7 200.0.0.6 200.0.0.5 10.0.0.4 When i make a call from "A" to "B" this is what i see (in terms of SDP). Looking from SER. A --------- Asterisk ------------ SER ------------ B (NAT'd) Public: 200.0.0.4 200.0.0.7 200.0.0.6 200.0.0.5 Inside: 10.0.0.1 INVITE c:200.0.0.6:19996 -------------------> INVITE c:200.0.0.5:35010 ----------------> Caller Via Called Status Duration Codec Type Traffic -------------------------------------------------------------------------- 200.0.0.6:19996 - 200.0.0.5:35010 - ?.?.?.?:? inactive 0'04" Unknown Audio 0/0/0 Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 So far is ok..........and the phone is answered OK c:10.0.0.1:16440 <---------------- (the phone is answered) OK c:200.0.0.5:35010 <--------------------- reINVITE c:200.0.0.7:19996 ---------------------> reINVITE c:200.0.0.7:19996 ----------------> OK c:10.0.0.1:16440 <---------------- OK c:10.0.0.1:16440 <--------------------- Finally according to the "session" information : Caller Via Called Status Duration Codec Type Traffic ---------------------------------------------------------------------------- ---------- 200.0.0.6:19996 - 200.0.0.5:35010 - 200.0.0.7:16420 inactive 0'26" G729 Audio 0/11.48k/11.48k Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 And the audio is only in one way. :( So. you can see the reINVITE message apparently is not being processed as a call to a NAT'd endpoint and therefore is not using the mediaproxy, you can see the second "OK" messsage has the invalid IP from the NAT'd user is in his sdp information. As i said it before i am using the gw-pstn configuration file from the onsip.org and as far as i can remember this configuration can handle the reINVITE? isn't I'm also using the last version of the mediaproxy (1.3.1). Can someone tell me what i'm doing wrong? Hope someone could help me here. Thanks in advance. Regards... Ricardo Martinez.-

18 years, 12 months

[Serusers] ftp.berlios.de

by Dinesh

Hello all, ser_0.9.3-0.2_i386.deb I installed the ser by using deb, but found out that the voicemail module is not there. Can someone tell me where I can find for this version? Thanks, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : dinesh(a)imcb.a-star.edu.sg WWW: www.imcb.a-star.edu.sg

18 years, 12 months

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sr-users July 2005