search for: vnaks

Some more info: Just tried this on a server without using any NAT and no port forwarding, no masquerading, and I still have the same problem. So there goes that idea. I do not know what this VNAK error means. By the way, I am using the latest version (1.2.6) of asterisk, have also tried other versions with the same problem [1.0.9 (Ubuntu Breezy) and 1.0.7 (Debian Sarge) and 1.2.1 (Ubuntu

...terisk version shown here is a completely stock, CVS version from just a few days ago. The "outboard" Asterisks are somewhat modified but also re-synchronized with CVS within the last week. Also, all Asterisks have iax jitterbuffer=no. So, my questions are: 1. What do the excessive VNAKs indicate? Some type of communication error? NAT-related perhaps? 2. Does the 20,000+ jitter have something to do with the audio sounding terrible? 3. Why is there jitter at all if all Asterisks have their IAX2 jitter buffers turned off? 4. Is there any significance to the "Username (no...

Hi Chaps, Is there a way to detect/highlight poor quality voice calls going through an asterisk server? Was thinking of picking up a cdr record or some other variable showing poor quality on calls when the internet is having issues. Is there any qos or poor audio quality variables available? Cheers, Taff. ___________________________________________________________ Yahoo! Answers - Got

Can anyone tell me why I keep getting these messages from IAXTEL? It does appear to register since I get lines like this: 2005-04-30 04:26:42 VERBOSE[1644]: -- Registered to '69.73.19.178', who sees us as 67.182.152.242:4569 But what is this? I don't think IAXTEL is working for me, since I can't dial 800 #s through it when I copy the iaxtel.com instructions. 2005-05-20

I was wondering if anyone had seen this problem before and/or could offer any insight into what the trouble might be: I have an Asterisk machine that it set up as a mutual friend with another one (in another state... about 150ms away). Calls between the two fail to get established approximately 50% of the time. When a call works, everything is fine. When one fails, however, I see a large

Hi, I use a simple scheme: SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk----> SIP video phone B (h264/Asterisk 11.7.0) When calls from A to B and vice versa drop on pickup. On B side: [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit

Hi, I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to connect to a 1.2.5 box for PSTN. There are 15 users on the remote server, all connecting via SIP softphones. For some reason, there is an increasing number of calls where the callee does not get any audio although the caller can hear them perfectly. This happens between 5% and 10% of the time. If they hang up and call

...debug, I never see any response on the console from *. However, the phone shows a response with INVAL. Seems like an odd response to a ping request. I believed it should get an ACK. Is that wrong? If I should get an ACK, what could I have messed up to prevent * from sending the ACK? I also see VNAKs on the * console. I believed that a VNAK basically meant "you sent me a voice frame at the wrong time." Yet I don't see any voice frames being sent before the VNAK. I wouldn't be involved with these details, but * keeps deciding that the phones are unreachable, and I can't...

Hello, I'm experiencing some problems with running IAX2 protocol on quite reliable link with G729A codec. My customer has 2mb FR link to the Internet used in about 20%. Ping statistics: 50 packets transmitted, 49 received, 2% packet loss, time 49496ms rtt min/avg/max/mdev = 9.308/13.126/33.307/4.851 ms Everything would be great, but the quality isn't good enough. I have 2mb/512kb DSL

Hello, I have a TDM400P with 2x2 configuration of FXOs and FXSs. I set a test extension of '444' to dial out a specific zap trunk and call a local #. First time I call out to '444' everything works fine. If I hang up the call, and within 10 seconds dial the same number again, I get a fast busy. Seems it isn't letting go of the trunk or something, and I don't have a

I'm taking inbound calls on an * server, then transferring them to a second * server where they join a MeetMe conference. If I have 'notransfer=yes' set on the first * server it works fine, but if I allow the transfer (call then shifts to be between the DID provider and the second server), the call is dropped 3-5 minutes later. There is no firewall on my end, and the two

Hi all, I am trying to connect to a softphone application using an Iax channel on Asterisk 1.4.30. I can do outbound calls, from softphone to asterisk, but not inbound from asterisk to softphone. I get the following Debug: ---------------------------------------------------------------------- ---------------------------------------------------------------------- Tx-Frame Retry[000] -- OSeqno:

I recently switched from BroadVoice to VoicePulse Connect on my Asterisk box. Too many Meetme quality complaints (whether real or perceived). I had to make a choice to use IAX2 or SIP with VoicePulse. I first tried to go with SIP because I already had it working and all of our devices are SIP. Problem is that every time I turn my back, the Asterisk registration with the VoicePulse SIP server

hallo, i am really confused how nativ briging is working with asterisk, i use a asterisk server as central server and register another asterisk and an iaxcomm client to the server, all three have public ips on the internet. somtimes, when i call from iaxcomm to my asterisk, the calls go peer to peer (i can see it with tcpdump) but sometimes the get routed through the central asterisk server