similar to: voice delay after call setup, outgoing calls

I have a DID with VoicePulse Connect, but the sound quality is horrible, it is often choppy and the caller's voice cuts out for 2-3 seconds at least once a minute, I have contacted VoicePulse many times, and they do not do anything about it! Does anyone have any similar problems? It isnt my Asterisk config because I have 0 problems using NuFone.

It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box

To whom it may concern, When dialing out an 800 number (888,866,877) through VoicePulse IAX you'll get a choppy sound. This is not due to a problem on your Asterisk or your line- the bad sound effect occurs in VoicePulse. (just spend lots of time finding that out) Assaf Benharoosh MCP, MCSA, MCSE ab@franticllc.com <mailto:ab@franticllc.com> Frantic, LLC. -------------- next part

Hello, I'm running Asterisk and using VoicePulse for IAX termination. I would like to have toll-free number assigned to my asterisk, any hints where I can get this number? VoicePulse does not offer toll-free numbers. Thanks, David

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

Hi, we are trying to install Junghann's quadBRI into Dell PowerEdge 2800 system without success. I don't know if the issue can be that Junghann's card fits 32-bit slot and Dell PE 2800 has only 3 PCI-X 64-bit slots. Can this be an issue? We get "CRC errors for HDLC frame" when the card is initialized. Any idea what can be wrong? 1/ We use latest bristuff packages. 2/

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

Hello, is it possible Asterisk's ChangeLog will contain a reference to appropriate bug number in bugzilla? This can be very handy. Thanks, David

Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup?

I have a very odd problem connecting to some websites from my CentOS 5 box Target websites: www.connecttech.com www.3ware.com (two of my HW vendors) I can usually get some kind of response, but if the content (download or page itself) is larger in size (downloads never pass 100K), then it hangs... When I fire-up wireshark, I get a lot of ougoing highlighted Checksum Errored packets but I

Hi, what new Dell servers are compatible and KNOWN to work with Digium TDM cards? I've looked at Digium's compatibility list at http://www.digium.com/index.php?menu=compatibility. Does this mean that other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with TDM cards? Can someone clarify this? Thanks -David

Just a quick couple of questions for ya'll. 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? I have had a terrible time getting a hold of anyone over there, and I need this functionality before I can migrate to * completely. 2) Are there currently any problems with inbound DID's? Everything is setup properly in *, but I am not able to receive inbound calls,

Hi, I am an experienced libvirt user on Fedora versions from F15 to F17. I have developped scripts to route trafic from outside on multiple interfaces/multiples IPs to multiple VMs, and back to affect each VM the required external IP address. I have servers with more than hundreds external IPs, and up to 4 VMs, each of them route trafic on different external IPs. I have servers with Fedora

Hello: I am new to the list. I am trying to set up asterisk with voicepulse. I have a voicepulse username + password, and SIP DID. When I login to voicepulse, I have this under my devices tab: Devices *Login:* Sysxxxxxxx *Password:* xxxxxxxxxx *Context:* VPWS *Connects to:* gw5.voicepulse.com My question is: Do I need a 2.4.x kernel? Currently I am running Debian/stable stock 2.2.x ? Has

We've got one of the Budgetone phones here, and we can call from any SIP phone, or an outside line TO this phone and the conversation sounds great for bothways, not a bad delay, no echo problem, etc. But when we pick up the Budgetone and dial an outside line or another SIP phone the person on the Budgeton just sounds really choppy and there is a slight delay. We've messed with

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

Hello, I want to optimize my registrations and calls of peers to my asterisk with the following options in sip.conf: ---///--- qualify = yes qualify = 500 qualifyfreq=5 registerattempts = 0 registertimeout = 10 maxexpiry = 60 minexpiry = 20 defaultexpiry = 600 ---///--- Can someone more experienced with these settings to help me to optimize connections from peers with mobile phone that using

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 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

Has any messed with getting Asterisk to work using the Voice Pulse Open Access plan? I currently have 2 numbers with Voice Pulse, 1 is a number that is assigned to their hardware device (Sipura SPA-2000), the other is a Open Access number that uses SIP from any device (you must authenticate with them). I want to be able to use the Open Access number on my Asterisk server here at home with no FXO