sr-users September 2006

sr-users@lists.kamailio.org

184 participants
437 discussions

[Users] 1.1.x/permissions.html is not up to date
by Juha Heinanen 01 Sep '06

01 Sep '06

for some reason web page http://www.openser.org/docs/modules/1.1.x/permissions.html is not up to date. for example, 1.3.12. tag_col (string) 1.3.13. peer_tag_avp (string) are missing. they do exist in rel_1_1_0 source. -- juha

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[Serusers] extra things confusing auth_radius?
by Natambu Obleton 01 Sep '06

01 Sep '06

Ok. I have two clients setup. One is X-Lite and the other is a polycom 601. I am doing radius authentication, but for some reason it doesn't work with the polycom 601 phone. In the debug the only thing that is different about them is something in the uri. Aug 30 09:16:51 laplata ser[25968]: DEBUG: get_hdr_field: <To> [72]; uri=[sip:311@voip.fasttrackcomm.net] Works, but.. Aug 30 09:17:53 laplata ser[25970]: DEBUG: get_hdr_field: <To> [45]; uri=[sip:311@voip.fasttrackcomm.net;user=phone] <mailto:%5bsip:311@voip.fasttrackcomm.net;user=phone%5d> Doesn't. Like it doesn't even send an authentication request to my radius server. I think it is the ';user=phone' part that is confusing it? So what I want to do is remove that part from the URI as a test. I just added a subst to remove it, but still isn't working with radius. It acts like the realm is no longer local. Here is my config file: http://www.natambu.com/serconf.html Natambu Obleton Network Engineer FastTrack Communications nobleton(a)fasttrackcomm.net (970) 247-3366 office (970) 247-2426 fax

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[Serusers] SIP Contact header field issue incase of NAT
by Abdul Qadir 01 Sep '06

01 Sep '06

Hi All, I have problem connecting Nokia Phone with SER. The problem is When SER receives the registration request from the Nokia phone. The registration request contain the phone's private IP (192.xxx.xx.xx), since we use wifi for connection. Our proxy detects that since this is a private IP handles the NAT translation and determines the phone public IP address (e.g., 202.xxx.xx.xx, port xx). Our SIP proxy informs the phone that registration is successful but the contact information is changed from userid(a)192.xxx.xxx.xx to userid(a)202.xxx.xxx.xx, port xx. On recieving this changed contact info the Phone rejects 200 Ok. We tried to contact nokia people and they said that its part of RFC3261 to reject such contacts. After this I have also gone through RFC 3261 section "10.2.4" and it seems that phone is doing rite. Can any body tell me why SER is changing the contact address in case of NAT. It would be higly appreciated if you can give me some reference from some RFC or other document which I can show to nokia people. Best Regards, Abdul Qadir --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

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[Serusers] Bandwidth parameter modification (b:) from ser.cfg
by "M. Victoria Peláez" 01 Sep '06

01 Sep '06

Hi! I am a beginner with SIP/SER systems. I have been having a look into ser.cfg, in order to modify some SDP parameters, for instance, fix_nated_sdp to change connection IP, and I have found enough information to do it. But, now, I am traying to change b parameter, but cannot find where to do it (Which module manage it?). I am also looking for a document with a description of all bandwidth types (CT, AS). Could you tell me something about this stuff, please? Thank you very much in advance for your help. I am very sorry if this is a very silly question! Best regards, Victoria

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[Serusers] mediaproxy
by Javier Ramirez 01 Sep '06

01 Sep '06

Hi, I would like to know how to configure the cfg script, to configure the media proxy so that it can nat the calls between users that don't have public IP, but not when they have public IP. best Regards Javier

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[Users] openser routing lookup question
by Dave Logan 01 Sep '06

01 Sep '06

Hey all, I have what may or may not be a simple situation. I've read through dozens of forum entries, dozens more list mails, and a handful of "tutorials" for openser, and I'm still missing something I think. I'm not afraid of the documentation, but neither the "core cookbook" nor the modules documentation had enough detail that I could catch on to what's going on. Let me lay out my scenario, and maybe somebody can point to a particular document or solution I can follow up on. I have multiple asterisk boxes: four as media gateways with TDM trunks, one feature server (VM), and three registrars. So needless to say, I have a lot of dialplan in there just for calls that stay in my network. What I'd like is to introduce a single openser box in there, and for any call that any of those asterisk machines gets, it sends it to openser, which then just calls a simple script (which I've already more or less written) to know what to do. So for example, I own the 555-1000(-1099) DID block DID 555-1000 is registered to machine1 DID 555-1001 is registered to machine2 In openser, I'm envisioning a call like such (php style notation): my $number = 5551000 in openser: $newdestination = exec('lookupdestination($number)'); now $newdestination is 5551000@machine1 or my $number = 5551001 in openser: $newdestination = exec('lookupdestination($number)'); now $newdestination is 5551001@machine2 Then I can just route it in the normal openser way. So there, my lookupdestination script knows all the numbers in my central database and where they're located, and on an unknown number assumes it's outside our network so just sends it to the gateway as is (5552000@gw1) or so. Thanks for any pointers at all, dave -- Dave Logan http://www.digitalcoven.com/ "Do let's pretend that I'm a hungry hyena, and you're a bone!" - Alice

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[Users] Help in dailplan in asterisk
by raviprakash sunkara 01 Sep '06

01 Sep '06

Hi Users, I'm new to Asterisk, and I'm working with openSER , For Call Routing I'm using the OpenSER and for PBX, Voicemail and Conferencing I"m using the Asterisk. Now i'm planing for Voicemail, openser listen on 192.168.2.75:5060 Asterisk listen on 192.168.2.76:5060. in Extension.conf [from-sip[ exten => 9001,1,Ringing exten => 9001,2,Voicemail(9001) Just I wnat to leave the voicemail in voice mail box, Can anybody help me......... below message What it is ?................ Executing Ringing("SIP/9001-0984c1e8", "") in new stack -- Executing VoiceMail("SIP/9001-0984c1e8", "9001") in new stack Aug 31 19:31:10 ERROR[7869]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on (err 2002). Check debug for more info. Aug 31 19:31:10 WARNING[7869]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for '9001' Aug 31 19:31:10 WARNING[7869]: pbx.c:1700 pbx_extension_helper: No application 'Hungup' for extension (from-sip, 9001, 3) == Spawn extension (from-sip, 9001, 3) exited non-zero on 'SIP/9001-0984c1e8' -- Incoming call: Got SIP response 479 "Regretfully, we were not able to process the URI (479/SL)" back from 192.168.2.75 -- 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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sr-users September 2006