search for: nathelp

Hello, I'm currently using OpenSER as REGISTER server and Asterisk for the call routing. Do i need the OpenSER nathelper module if i want to offer (mostly) automatic NAT traversal to my users or does Asterisk have the same functionality? It seems that the nathelper module should be able to automatically traverse any NAT as long as the User-Agents use symmetric RTP. Further it is possible (in the ser.cfg) to automa...

Hi Friends, need to help. *I have problem about sip : SIP/2.0 401 Unauthorized* Is it require to nathelper module in kamailio ? *what can i write kamailio.cfg file when kamailio and Asterisk on same network?* Scenario is like as : ----------------------------- 1) kamailio server on 172.18.100.74 kamailio.cfg ( nathelpler module ) ----------------- loadmodule "nathelper.so" modparam("na...

Hi all I'm a newbie in asterisk.I ant to know how i ca configure asterisk to work with the nathelper module of openser to fix the nat problem! Thanks in advance! bets regards Serge --------------------------------- Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international.T?l?chargez la version b...

...dmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/nathelper.so" #loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/textops.so" #loadmodule "/usr/lib/ser/modules/maxfwd.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/...

...> >>> > >>> > >>> STUN is helpful, but as I understand it analyzes the situation and > >>> reports > >>> the configuration of a NAT. It doesn't help you keeping the NAT > >>> session open, > >>> as SER module nathelper or the FWD/Jasomi solution. > >>> Check here http://www.voip-info.org/wiki-SER+module+nathelper > >>> It's ugly, but what it does is sending UDP packets from the outside > >>> to the > >>> NAT to keep the ports open for incoming calls. NAT is a...

...ing to do in order to get SIP phones behind NAT working reasonably well without fiddling with the DSL router is to have some kind of far end nat traversal mechanism. Is there any way to do this with open source tools? I've seen somewhere that far end nat traversal can be achieved with SER + nathelper does the job... has anybody gotten this working in conjunction with Asterisk? Another question... Are you aware of a SIP ATA or phone that has some kind of VPN (i.e. PPTP) client embedded in? This would make the NAT problem go away nicely and provide added security... Cheers, Jean-Michel.

...ules/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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/xlog.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...

Hello i am trying to use this exmple with SER-0.9.3 but still NATED Clients are not working any other requirement http://www.voip-info.org/tiki-index.php?page=SER+example+NAThelper ----------------------------------------------------------- # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork...

One for the gurus.. I have seen there has been a lot of discussion about using SER with Asterisk.. This to me seemed like an over kill becasue it would basically be doing most of what Asterisk is doing anyway unless you create some weird and wonderful config in SER.. Anyway, I decided to go and have a quick read through the SER docs and in the section about NAT they say that the best way to

Why is it that the wiki indirectly recommends SER (or another proxy) out in front of Asterisk. If Asterisk can use radius, and provide the rest of AAA they why ? Incidentall\y, I'm not familiar with network configuration really, although I do understand most of the basics. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version:

...R CSeq: 101 INVITE Max-Forwards: 4 Remote-Party-ID: <sip:0000123456@109.147.41.48>;party=calling;screen=yes;privacy=off Timestamp: 1128586438 Contact: <sip:0000123456@109.147.41.48:53084> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 369 hint: NAThelper hint: SDP rewritten hint: usrloc applied hint: NAT... v=0 o=CiscoSystemsSIP-GW-UserAgent 5168 3221 IN IP4 209.47.41.61<http://209.47.41.61> s=SIP Call c=IN IP4 109.147.41.48 <http://109.147.41.48> t=0 0 m=audio 53870 RTP/AVP 0 8 18 3 101 c=IN IP4 109.147.41.48 <http://109.147.41.4...

hi i am configuring the SER on centos please can anybody knows abt this if yes please send me the step by spey ser+mysql_rtpproxy+nathelper i want all thies modules thanx and regards prabhakar.G _________________________________________________________________ Get the new Windows Live Messenger! http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline

I'm interested to hear how folks are handling NAT SIP routing issues "in the wild" for commercial use. Are you using a commerical SIP-aware NAT router solution? If so, what? Are you using a software SIP-proxy like SER or siproxd? If so, which? Do you set everything to "canreinvite=no" in sip.conf? Any comments about real-world implementations would be welcome. Thanks

...is because after using sipsak I've noticed maybe the reason that i don't get voice in one direction could be because at the SDP/SIP messages are references to the 192.168.1.50 (server internal ip) and not the ip from the provider. Is there a way to alter SDP messages to correct them??? SER nathelper could work with Asterisk? Or SIP ALG? (I'm trying to investigate this last one i've found mentions in faqs.org) Best regards, Helder -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050104/3cbc...

Hi: I live in a country where calls from landline phone to a mobile phones is more expensive than mobile to mobile. I have FXO card connected to the landline. All the calls from IAX goes through this interface to the pstn and mobile phones. I want to save money by transfering mobile calls through a mobile phone. Is there some interface between the FXO card and the mobile phone so asterisk can