search for: sip_nat_oneway_or_no_audio_asterisk

Hi all, I have set up Asterisk and I am able to register with my SIP provider and receive calls. When I try to register with Asterisk from outside I can place calls but tthe other person can't hear me. Have opened port 5060 UDP as well as port 10000 to 20000 UDP. Any ideas? Thanks, Christian

Ok, this is highly confusing. hestia*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 2944030 2944030 oneeighty_start No RFC3581 2944035 2944035 oneeighty_start No RFC3581 sip users (type=friend) are in sip.conf. I have nat=no

I have a asterisk server. Two SIP Soft XLites are connected to the server. I am able to make calls from one SIP Phones to the other SIP Phones and landlines successfully. The SIP Soft Phone on th eother side can hear my voice but I cannot hear their voice. They can call my local cell phone as well. Samething, they can hears my voice, I cannot hear their voice. The microphone and speakers are

I'm working on calls coming in to an asterisk box as H.323, and going out as SIP to a remote device (a VoiceMaster). The remote device is refusing the calls with SIP error 406 (Not Acceptable). I have attached the SIP debug output below. It looks like codecs overlaps - can anyone see why the call is being refused? (Note that I'm not registering with the remote SIP device, just

Hi! Problem: I can't hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) <-> Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone(Location B) I am having problems with sound, I have opened the

...selves by using STUN <tiki-index.php?page=STUN> and sending UDP keep-alive packets. Qualify <tiki-index.php?page=Asterisk+sip+qualify> sends keep-alive packets from Asterisk to the client on the inside." - however I can't get it to work http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html ----------------------------------------------------------------------------------- Here there is some detail about the NAT= option in sip.conf and firewall NAT types plus some understandable diagrams of why SIP & NAT is so much bother. http://www.voip-info.org/wiki-STUN ---------------...

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same