similar to: Quality Suffers on Outgoing Only

So I went to go compile the Zaptel library from the HEAD CVS and I get some really really odd errors which don't make any sense. I've attached the console output ... any idea why this is going on and how to fix this? Thanks, Sam Bacsa ---------------- SNIP -------------------- [root@pbx zaptel]# make gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB

I am running * 1.0.5 and have been having lots of problems with outgoing calls and their sound quality. I am using ULAW for the codec and sixtel for termination. Basically the problem is that portions of the call seem to be lost and replaced with silence. Sometimes I can't hear the person talking othertimes they can't hear me. This situation comes and goes throughout the call. Bandwidth

Hi there, I support a large number of enterprise users who contractually must connect to our support center via a 4G VOIP connection. I simply want to be able to auto detect all poor quality calls in realtme (as they are being made), play a message and drop the call - without user intervention. All decent call quality calls will be allowed through - to be handled by support staff. Its a

Hello speex team, Where can I download test samples (*.wav) from? I could play the samples from http://www.speex.org/samples/, now I want to encode and decode few samples myself. :-) I want to encode and decode them with all possible options (VBR, VAD, DTX, COMP, NOENH, PACKETLOSS, etc). Speex is wonderful. Thank you. :-) -Basawaraj Send instant messages to your online friends

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

Hi all! I know this is quite offtopic, but I found nothing in google nor in the net-snmp mailing list, and I know there''s a lot of people here graphing with snmp+rrdtool. I installed net-snmp 5.1.2-6.1 in Debian to produce graphs for some ethernet interfaces. It seems it''s not reporting correct values for them. Here''s my snmpd.conf (very simple) and the output from the

Hi, I''m new to XEN and have the following Problem: If I shutdown my domU the network of the dom0 leaks. A ping from another server to my xen dom0 shows a Packetloss > 80%. Then if I restart the domU after some seconds the ping show a loss from 0%. I retryed that 3 times but it is really hard to go to the dom0 via ssh and recreate the domU if the packetloss is that high. I used google

Hi there, I have two ethernet connections of 2Mbit/s each and I''m trying to add them together to one 4Mbit/s connection but I cannot get more than approximate 1Mbit/s! My setup: I have a LAN (10.2.18.0/24), connected to a larger network (10.0.0.0/8) by two WAN-connections with 2Mbit/s each. On each end I have a Linux router. I first setup the routers to use TEQL with one of the

I'm using speex under symbian (8000 hz, 16 bit) narrow band. The phones API only give me a buffer of 4096 bytes in recording.To reproduce audio I must fill up the buffer of the same dimension. 4096 isn't a multiple of 320. I want encode the audio in streaming. The solution that I adopt to encode is: - Divide 4096-256 bytes in 12 frames of 320 bytes. - Therefore the frame number 13 is

Hi guys, I have the following configuration: 10 SIP softphones <--> Asterisk <--> PSTN Audio is always good on SIP softphone side, but callers from PSTN side *sometimes* complain that the audio quality is bad (and volume low). The QoS is turned on on the computers where SIP softphone is installed, and the tos setting in sip.conf is set to 0x18. The interesting thing is that usually

Author: rmesta Repository: /hg/zfs-crypto/gate Revision: a5e1262c14c8b8b2e86d10d72c8edef705b390f5 Log message: 6346204 NFSv4 client suffers undetected write errors to full file system Files: update: usr/src/uts/common/fs/nfs/nfs4_vnops.c

Hi all, I am not sure if this is the right list. I implemented a network store based on blktap, a IO consumes100 ms, however tapdisk stops every 10 seconds, as a result previous io''s completion will be sent to kernel until 10 seconds later. This makes the performance suffer. Am I using blktap wrong ? Thanks, Yongqiang -- Best Wishes Yongqiang Yang

I agree with the in-effective of TCP transmission, but I wonder if the the UDP packet is dropped, the tinc VPN itself wouldn’t retransmit, and if the upper level application doesn’t handle the packet loss well, will this be the problem? Or the upper level application have very limited tolerance to packet loss(like RDP application, I guess if the packet loss go to certain threshold, the connection

All, A bit of background: I work on a QA API on a network that is very choppy (a lot of network interrupts), and we use ssh to do a large part of this automation. This leads to some problems: ssh connections seem to be sensitive to network state, becoming unusable if the choppiness reaches a certain threshold, and either timing out or disconnecting if this happens. Anyways, I stumbled across

Greetings, I have a Sipura SPA-1001. When I make outgoing calls, I have very jittery sound. Incoming calls work fine. This wasn't the case a few months ago, I am running head as of yesterday. Any suggestions? Thanks, Erik

I'm having slight problems with outgoing audio quality on Zap channels. People hear an interrupted voice. Can anyone help..? Regards, Fabrizio Mazzoni Macron SPA

Hello, I have an Asterisk running with a Sangoma A200 card with Hardware Echo cancelling connected to the UK PSTN. If a PSTN call comes in, voice both ways is OK, however if an outgoing call over the PSTN is made I can hear the other party OK but they can not, they can barely understand what I am saying, my voice is unclear fading and skipping. Internal SIP and IAX2 calls are OK,

Thanks Dragos, I assume I will be setting those parameters during initialization of encoder right? Question is, if connection gets too lossy, how will opus adapt to it? Can it automatically shift bitrate down to minimize impact? Mark from IRC suggests that the app has to be aware of the losses and change it on the fly. Has anybody on the list tried this? Kelvin Chua On Wed, Mar 4, 2015 at 5:53

I've watched the dialogue about how asterisk has to manipulate the packets from an IAX2 connection to a SIP client. That said, I'm wondering if a previous problem I've been trying to diagnose could be related to that process. In short, here's how I describe it: Outbound: SIP 7960 > Asterisk > IAX2 Audio is perfect both directions. Inbound: IAX2 > Asterisk > SIP

Hello all, Since a while some of our SIP users complain about gaps (sometimes multiple seconds of silence) in the RX audio stream (direction pbx -> phones). Our configuration is an Asterisk with two Wildcard TE410P cards that are connected with E1 PRI's to an external server running callcenter software. The SIP users use Snom and Polycom phones. The gaps are not present in