similar to: strange Music on Hold between SNOM, Grandstream and Asterisk

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

I have an * setup with sip phones and sip fxo gateway. When a sip phone places a sip/fxo call on hold, MOH is very choppy. It looks like RTP has a real problem with timing if it is not receiving RTP packets. If the outside call that is placed on hold is not generating any audio, the sip/fxo gateway does not send * RTP packets. Is this valid? Is this a problem with the sip/fxo gateway or a problem

Does anyone know how to maximize music on hold quality on calls inbound from PSTN? I know that it is common to have choppy and static sounding music on hold when connecting via PSTN but how can that be minimized? I assume that the bitrates, type of music, etc can minimize the effects. Does anyone have any experience in this area? Do you know where I should look for more information?

My new asterisk install seems to be running fine - including playing all prompts etc without error. However, when placing someone on hold they here choppy music (first second or so) then quiet. I see the errors below. What is causing this? (Note that I am running AsteriskWin32). Thanks, Mike May 17 17:26:02 VERBOSE[3708]: -- Executing Answer("SIP/2434-263c", "") in

Product Review Mediatrix 1204 4-Port SIP FXO Gateway Firmware: v2.4.10.69 - US Version US Retail: ~$750, Street Price: ~$450. The Mediatrix 1204 SIP FXO gateway is equipped with four RJ11 pstn jacks and one RJ45 Ethernet jack on its rear panel. It terminates the four pstn lines in either Loop Start or Ground Start mode, handles incoming CallerID, and generates either Dial Tone (back towards the

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

Hello to all. I do not know what is causing choppy music on hold when call comes in through E1 card (PRI).. but this channel info is somehow strange.. We use Alaw over PRI (and I think it's format number 8), But why is WriteFormat at 2 ????? Thanks! show channel Zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: 1136667936.0 Caller

I don't know everything that's going on as someone else has been working on the project, but it hasn't really been going anywhere, so I had some questions. We've got some Snom 320s with Asterisk 1.2.9.1 (I believe). All was well (with a previous release), but the phones started to get real choppy. We are also running a softphone at this location and it was fine. The SIP qualify

I'm using the default mp3 files that ship with Asterisk: fpm-calm-river.mp3 fpm-world-mix.mp3 If they were variable bit rate, I think I would see a warning about 'varibel' or similar...is anyone else able to get these files to work? -Mike- ________________________________ From: Sander [mailto:crombeen@rommelweb.nl] Sent: Tuesday, May 17, 2005 6:34 PM To: 'Asterisk

I just got a grandstream SIP phone Here is my sip.conf for the phone [mlh] type=friend insecure=yes username=mlh secret=mlh host=dynamic canreinvite=no The phone as the default config on it. If I use the phone to call a Zap interface (a tdm card) the voice sounds all choppy. If I use the phone to call a x100p card, it does not dial what I dial (no DTMF) I don't know

Hi all, I have my asterisk setup as following: IP 2 x E1 x-lite <-------> Asterisk -------> PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data

I am experiencing choppy sound when I bridge from a SIP peer to an IAX peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am experiencing choppy sound from the SIP peer to the IAX peer but not vice-versa. I know that this is not a bandwidth issue because I don't have choppy sound (with the same codec) when bridging IAX->IAX peers or SIP->SIP peers. My timing source is

Hi folks, in order to establish backward compatibility we made an image that automatically detects if the other side does not support RFC3264. Please try it out, we would be very interested if this image is a progress! http://snom.com/download/share/snom200-2.03b-SIP.bin Thanks, CS

OK, background/config. running * (show version reports 0.9.0) on Mandrake 9.2 (kernel: 2.4.22-32mdk) with a dual 800mhz PIII with 256M Ram 4port FXO digium card, no IRQ sharing I can find (cat /proc/pci & cat /proc/interrupts), vmstat reports a minimum of 80+% CPU idle when problem occurs. connect to a Grandstream 101 (GS) via vpn (no nat). Link has 100ms - 150ms ROUND TRIP latency

Hey all, I''ve got an interesting one for anyone who''s up for a challenge. Essentially, I have a very choppy effect, that almost looks like timeouts are overloaded or interfering or something, that only occurs when sortables are on the same page as "standard" effects. Here''s what I''m doing: I have a menu that slides in and out on the right side of

Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw => ulaw is choppy, ulaw => alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with asterisk 1.2.13? -Benoit-

Hi folks, I'm experiencing some problems with sound through phpAGI ... What I'm trying to do is a menu, doing some database lookups and so ... But sometimes the sound become too choppy ... just sometimes .. like 1 of 5 calls ... but is a big percentage ... And I have my current menu on the dialplan that I have no problems with it ... I'm using .gsm for both but different

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

Just come across a problem - we have sent out heaps of PAP-2 ATA's and just discovered that when joined in a conference they are choppy on the up leg (so the other users in the conference will hear them with a choppy sound) but the down leg is perfectly fine (so the end user can hear the conference participants perfectly). I have tested the same setup with different brands of ATA's

Hello I have asterisk running in Red Hat 9 with a cheap HFC card on i4l. I have choppy sound problems sometimes, and echo problems often. I am using a 2 port Grandstream ATA, Grandstream BT and a Grandstream GPX-2000 I read that changing to BriStuff will fix the echo problems, but have also read other users say that the only way they solved the echo/choppy sound problems was using a Fritz ISDN