similar to: nathelper vs. asterisk

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

https://bugzilla.mindrot.org/show_bug.cgi?id=1489 Summary: ssh should normalize IP addresses before comparison Classification: Unclassified Product: Portable OpenSSH Version: 5.0p1 Platform: All OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: ssh AssignedTo:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 because of strange reasons my notebook sometimes crashes short after startup (but that's not ext3's fault, maybe mem?, when i wait several minutes it works without problems) the problem is that after 3 crashes at startup, when my notebook finally worked i got the msg: Sep 23 23:29:17 blackbox kernel: EXT3-fs warning (device ide0(3,3)):

No, you don't need a sound card. Do you have ztdummy loaded or zaptel device in your system? Regards, Gus ----- Original Message ----- From: "Chris Hariga" <contact@techselesta.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, October 19, 2003 8:19 PM Subject: [Asterisk-Users] Music on hold... > Hi, > > I need a sound card and mpg123 for music on

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"

Hi, I am faced with this dilema of asterisk not sending an ACK after it receives 200 OK from OpenSER (which is a response to a reinvite request sent by asterisk. Here is my setup Carrier<->OpenSER<->Asterisk1<->Asterisk2 A user is connected with Asterisk1 (through the carrier and OpenSER). On certain dtmf events the call is forwarded to Asterisk2 using the Dial command.

hello, i'm currently using asterisk (cvs-head) as PSTN gateway. the routing logic is mostly done in OpenSER. the problem is that i'm not able to transfer calls between the PSTN and another SIP peer (when the PSTN<>SIP connection goes over asterisk but the SIP<>SIP connection does not). there are 2 possibilities for asterisk to be part of the transfer: 1) asterisk receives

Hi, We are testing a SIP solution * + ser solution for a large implementation. All the clients are nated. When a client is dialing outside the domain (to a FWD sip account for example) all is perfect ! ;-) But ,when a call is done to a sip account, the client is ringing, then the caller can hear the nated client very well, but the nated client does'nt hear anything. RTP issue no ? I've

Hi List, After some research, it seems the only reasonable thing 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

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I

Hi all, Hope somebody can help-I really am stumped as to why this won't work. I realise that this isnt an Asterisk problem (Please dont bash me on the list) and I have emailed the SER list but I havent received a reply and maybe someone on this list can help...Once this problem is solved I am going to use Asterisk for voicemail etc with SER (I have it set up) I currently have SER set up and

hello, i've an asterisk box which is connected to an E1/PRI via a TE110P card. incoming calls from mobile phones where the number is transfered as a whole block work fine, but when dialing from an analog or ISDN line to the asterisk box there is a timeout of about 3-5 seconds. originally my incoming context looked like: exten => _X.,1,Dial(SIP/${EXTEN}@domain.tld) so i assumed that the

Hey All, Totem has stopped playing .wmv files recently. I don't know exactly when this happened, but I know I've played .wmv files not too very long ago. I've got 372 .wmv files all of which I've played at one time. When I try to launch any one of the .wmv files from file manager totem pops up but the controls are not functional. If I start totem from a terminal there is no

Hello Jerry, don't know why you need to upgrade gst-inspect plugin. But I did update gstreamer plugin last year. If you like to take a view http://centos.cms4all.org/repo/7/gstreamer/ cms4all.repo [cms4all-gstreamer] name=cms4all-gstreamer #baseurl=http://centos.cms4all.org/repo/7/gstreamer/ baseurl=file:///srv/repo/centos/7/gstreamer gpgcheck=0 enabled=1 This are private package, does it

Hi all, I try to make a call from my Openser(SIP Proxy) to the asterisk in different machine. I use my asterisk as a trunking gateway. I can make a call from my openser to some trunking gateway such as my cisco 5300 or welltech 5250. In the same method, I try to make a call to asterisk ( sip listen on udp 5060 ) I use ngrep on my asterisk machine and list as below. But I can't find any sip

Hi all, I have a need to rebuild the gstreamer and gst-plugins in CentOS 7. So I found the wiki for rebuilding a source package... I downloaded the vault.centos.org files, installed them with dependencies. I did the rpmbuild -ba for each of the four packages. All seemed to go well. No errors reported. However - the /usr/bin/gst-inspect* listing shows the files as OLD ls -l /usr/bin/gst-inspect*

I'm taking the liberty to announce this event on the Asterisk mailing list, as Asterisk and OpenSER form a valuable combination in SIP architectures. The second edition of OpenSER Summit will take place in San Jose, USA ,on the 17th of March, 2008, during VonX Spring 2008 pre-conference events. This is the first US edition of the OpenSER Summit - to learn more about the agenda and layout of

I am trying to add a package (I know its older - but it should work) gst-rtsp-server-0.10.8 to CentOS 7. The gstreamer 0.10 packages are included in C7 and they compile just fine. When I extract and try to compile gst-rtsp-server 0.10.8 the ./configure goes fine. but the make results in errors: make make all-recursive make[1]: Entering directory

Hi, I am trying to integrate asterisk with openser for a simple call. I am facing some issues with Asterisk. Below is the explanation: I have a UA1 sending invite to UA2 through Openser and Asterisk with the below sequence. Sequence is UA1->OpenSER->Asterisk->Openser->UA2 When Asterisk gets the INVITE, the INVITE contains two Via headers, one of the UA1 and

Trying to find a solution to a sticky problem here. We have 3 OpenSER systems. Phones register with the OpenSER systems, and after they authenticate the user, pass the registration info using OpenSER's send() command to all Asterisk boxes sitting behind them. Each asterisk system then knows about every phone. For this to work, I had to turn off authentication in Asterisk for both