similar to: IAX best practices

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

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I am experiencing a situation where I am getting intermittent choppy audio. Here is the network layout: Termination provider -> IAX2 over the Internet -> 20Mb fiber connection -> router -> Asterisk My ATA connection goes into the router between the fiber and the Asterisk server on another interface here is the layout from me to Asterisk: Sipura ATA (SPA1001 running

Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on "Teliax" before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi All Could a VPN be used to help with SIP Tunneling and QoS issues. State 1: Two IP Networks Connected via the Public Internet transmitting VoIP Traffic Say a VoIP User and VoIP Termination Provider. Each side can put QoS onto their part, but if QoS does NOT exist between them then call quality will be bad anyhow. State 2: Same as above except a VPN tunnel is setup between each side. Thus

So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me a phone at my hotel rooms, etc. During the day or late at night the thing works great - best ATA I've ever used. However, in the mid-evening (when many business travellers are at the hotel room doing work), the outgoing audio channel gets so choppy that the person on the other end can't make me out clearly.

All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure

I'm running Asterisk 1.2.13 on a Solaris 10 X86 box behind an IPCop firewall on a 5Mbps down/512 up cable connection. I'm having sound quality problems when users call in for voicemail and with music on hold. The sound is choppy and muffled while souding pretty good for calls inside the network. I'd appreciate some pointers as to where to start looking to improve things. I've

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

Greetings - I''m new to QoS, so please be gentle (and yes, I''ve RTFM, though I don''t understand every bit of it) Here''s the thing; I''ve tried several scripts--simple and complex--for classifying my Vonage traffic into a high-priority queue, but no matter what I do it doesn''t seem to work. Right now I''m using the HTB script

Am 22.06.2020 um 17:01 schrieb Telium Technical Support: > I don't know if there was a prior email with more details, but.... > > Latency is as important as speed. Have you checked latency between your device and pop? What about QoS at your location, and does your ITSP support/respect QoS? That's a very good idea... Could you suggest me how can I check it? The Gateway is a

My ADSL connection speed is 512Kb up and 128Kb down. When making calls from Asterisk to IAX and back to the Asterisk, the sound is choppy and 20% of voice messages was lost. What is the production bandwidth requirement per internet call. I understand there is no guarantee of QoS but at least a benchmark to follow. -- David Kwok Iaxtel/FWD # 17001813482 -------------- next part

Hi, Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? Ta. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040421/3d91c7f6/attachment.htm

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

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

Spinning off from another topic...what are people using for QoS / Shaping? I'm using Wondershaper script with OK results...but I'd like better. Ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091008/96b14a3e/attachment.htm

We have a small office installation running over a cable modem. (8M down, 500k up confirmed with numerous speed test sites) When a single call is up, call quality is fine. When a second call is up, outbound audio is immediately choppy. We're using ulaw, and confirmed that traffic with 2 calls is <175kbps in/out. (IAX connection out) Asterisk doesn't report any dropped frames, the

Hi all, I'm trying to get my head around our Asterisk network configuration. We've been using it for about 2 years now (home office) and it works great. Its Asterisk 1.4.2 with SIP through external provider(s). We have the Asterisk server behind our IPCop firewall, and have a dedicated IP address that comes to the firewall from our ISP (Cox) and that is routed to our Asterisk box

I just signed up with livevoip for 800 DID and have very choppy audio. From PSTN to my asterisk is ok but asterisk to PSTN is terrible. I am using IAX and was assigned to server iax01.nyc.*. I do not believe it is a bandwidth problem on my end and I have no problems using iax with gafachi. I opened a ticket with livevoip but no response yet. Would I be better off using sip with them? Is there