similar to: size and quality of audio clips effect the playback??

Hi, When I start asterisk with asterisk -vvvvc it works preaty well, but I get the following warnings: Warning, flexibel rate not heavily tested! WARNING[1074494176]: File res_musiconhold.c, Line 462 (moh_register ): Unable to open pseudo channel for timing... Sound may be choppy. WARNING[1074494176]: File res_musiconhold.c, Line 462 (moh_register ): Unable to open pseudo channel for timing...

Greetings I was getting bad static crackle on a phone, so I reload from the latest CVS and did a make clean ; make install on zaptel, libpri and asterisk Now I get this error [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) May 28 13:59:42 WARNING[16384]: chan_skinny.c:2541 reload_config: Unable to get our IP address, Skinny disabled Urgent handler [chan_oss.so] => (OSS

hi list, when I run asterisk, appears the following: .... WARNING[1074459808]: File chan_oss.c, Line 346 (setformat): Requested 8000 Hz, got 8178 Hz -- sound may be choppy WARNING[1074459808]: File chan_oss.c, Line 974 (load_module): XXX I don't work right with non-full duplex sound cards XXX WARNING[1133735216]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource

Hi, I get the following message when I try to run Asterisk. This happened after I changed some of the configuration files: extensions.conf, zaptel.conf and zapata.conf. I have both TDM40B and X100P cards and am trying to use them together. Please let me know if there is something else I should do. Thnx, Gladys [chan_local.so] => (Local Proxy Channel) == Registered channel type

pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw

Thanks for reading this. I've been pulling hair for days trying to resolve this, and any help someone can give me would be very much appreciated. I have an Asterisk box that is basically a P4-3GHz, a Digium-recommended SuperMicro X5SSE-GM motherboard, 2GB RAM, 250GB IDE hard-drive with UDMA, a SoundBlaster Live! 24 sound card, a Digium Wildcard TE110P, and a Digium Wildcard TDM400P with 4

Try to use 2.6.10 kernel not 2.6.12.3 kernel Coz I got some problems with the kernel 2.6.12, I had the same Digium Wildcard TE110P, Good luck ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Robert Christian Inviato: marted? 26 luglio 2005 19.13 A: asterisk-users@lists.digium.com Oggetto:

Hello I have a rather limit hardware where i try to run Asterisk, 1 GHz VIA C3. Everything works fine except music playback on MOH. I have encoded the music in same codac as as the voicestream (ulaw), ulaw is used end to end. MOH fades in and out with varius volume, and is very choppy. What I don't quite understand is when I put the music file as "you are first in line" it

Oh, wise Centaurs, On the last install (Suse) I was running on this old notebook (Dell i600m) the audio worked fine. Specifically, Realplayer and onsite players (like you get at NPR and other sites) and youtube played just fine. But with this current install (CentOS 5.3, fully updated) all of those sources just mentioned play choppy. It's like an echo so severe that it's impossible to

I'm having an issue with an asterisk install that anything recorded (in .gsm format) and all of the pre-recorded .gsm files are choppy. All calls into the asterisk box is fine and any voice mails left in a box are fine as well. It's just the playback of any recorded message and any of the pre-recorded files. Anyone have an idea what might be going on? The only problem is the playback of

Hi All, I've just setup a test Asterisk system that allows incoming/outgoing calls via an ISDN card (l4i) and incoming/outgoing calls via SIP (iconnecthere). I have two SIP Softphones (Xten X-Lite) for making and receiving calls. When receiving an incoming call via the ISDN interface the sound quality is fine for the Softphone user (i can hear the caller perfectly), but the person

Hi, using asterisk 1.2.5 with mysql in a centos 4.2 (2.6 kernel) no hardware interfaces installed gives me this error. Im a bit new to this so any help will be appreciated. == Parsing '/etc/asterisk/musiconhold.conf': Found Mar 26 00:58:49 WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. [chan_oss.so] => (OSS

I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the

Hello, I am posting this after spending hours digging through the list archives. Problem : When asteirsk plays a voice prompt, the voice clip is really choppy. I figure that this is something to with the sound card, the timing of playback etc. But cannot seems to find an answer. Here is the Notice which appear when voice prompt is played. NOTICE[1217602880]: File sched.c, Line 209

I've spent the past week experimenting with Asterisk@Home 2.6, and then Asterisk 1.2.6 individually, on VMWare Workstation 5.5. I have an entirely IP (hard & soft)phone setup (IAX and SIP) so I have no requirements to support any Digium PCI cards, etc. All in Asterisk works extremely well except for one thing: Playback of sounds (GSM format) such as an ivr greetings, sound terrible.

Long story short, I have a small LAN, a Red Hat 8.0 fileserver, a Win95 fileserver and 2 workstations, one Win98SE and the other is Red Hat 8.0. Using Samba as the common file system thread, I can access and play an MP3 file from the Win95 server using the RH workstation and it works fine. I can access and play an MP3 from Linux or Win95 server using the Windows workstation and that works

Any users of the VocTel VOIP service? (Canadian) How have you found the quality (Choppy / smooth audio)? Any problems registering? (I have been unable to register for hours) After reading about the collapse of a big USA VOIP provider, I'm curious Thanks, OCG -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have asterisk-0.4.0 running. When I make a call between an ATA186 and Asterisk using ulaw or alaw codec, all is fine. I installed the CVS version and tried the same thing but the voice is choppy. The installation was done on the same linux server. The stats on the ATA186 show no packet loss but a great number of "late packets". The stats when running version *-.0.4.0 do not

Hi All, I'm running Asterisk on the following vendor_id : GenuineIntel model name : Celeron (Coppermine) cpu MHz : 668.202 cache size : 128 KB with 192 MB Ram Audio coming from Asterisk (the demo ) is excellent when using a SIP phone on the LAN to Asterisk, and when dialling in from outside via ISDN to Asterisk. However, when connecting from SIP phone to SIP

Hello. I've just installed asterisk-1.4.21.2 zaptel-1.4.12.1 chan_ss7-1.0.10 libpri-1.4.7 I am using Sangoma A104 card with wanpipe-3.2.7.1 drivers. My OS: Ubuntu 8.04 Server Kernel: 2.6.24-16-server I am getting a choppy GSM playback and too many defunct AGI processes when channel closes. i am using Perl or PHP, also 'agi-test.agi' going to defunct too... I was able to playback GSM