similar to: confused about iax and voip providers termination

I've tried Idefisk and Cubix Softphones, and they both work fine, except for two issues: 1. Idefisk seems to have a longer delay between the time I can hit tones, and 2. Cubix, while can send DTMF faster, never actually connects to an Asterisk-dialed call -- I can't hear the party who answers. #2 has been asked but unanswered here:

SBC has an outage that is expected to last until tomorrow in our area. This has taken out our 5 POTS lines and our T1. I have signed up with EXGN for outbound calls and am using IAX. Calls ring through to the other party (my cell phone in this case) but Asterisk doesn't seem to think the call was answered. Ideas? -dave

Hello folx! I just started to play with *. I first installed it this past weekend on my Solaris 9 ultra 5 test box. Now I'm attempting to put it on a freshly built Linux box a mere few hours old. I've installed asterisk 1.2.7.1, libpri-1.2.2, zaptel 1.2.5 and the latest asterisk-sounds 1.2.1. This is running (or installed) on my Slackware 10.2 box running kernel 2.4.31. I'm testing

Dear list, I've been looking for a voip service provider with inexpensive and high- quality call service to China. However, the providers I've tried (voipjet, exgn, voxee) all have long to super-long latencies on calls to China. Has anyone found a service with good connections to China? Please share! John PS: I'm trying out this 'didww.com' service for the first time

Hello all, I'm looking to troubleshoot some echo issues and possibly "studdering" while using the speakerphone with the Intellitouch ITC3002. On the hardware end, i have a 2621 setup as the router with some policy maps for qos, and a 2900XL with cos priority set to 5. I have setup created 2 vlans currently (really small office), 1 for corp/data, and 2 voice vlan. Asterisk is on

I'm confused about VoIP termination service. Would this mean that I could have an Asterisk server in my basement and receive/call PSTN phones via a VoIP termination provider? The "call" out from my box would go over the Internet to a termination provider which would in turn convert it to analog over the PSTN to someone's plain old phone? And vise-versa, a plain old phone caller

I find that DTMF does not work reliably if jitterbuffer=on for certain IAX providers. For instance, DTMF tones are missed entirely about half the time when I dial into an exgn.net account. However, it always works fine for an unlimitel.ca account. Someone else has seen this too: http://bugs.digium.com/view.php?id=6011 Can anyone suggest a workaround (other than jitterbuffer=off)? - Mike

Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere <http://www.iconnecthere.com> Vonage <http://www.vonage.com> Teliax

Does anyone have any experience with any SIP or IAX providers that support E911? I'd love to convert entirely to Asterisk at my house, but the lack of emergency dialing has been a major hold-up for me. Thanks in advance for any suggestions! -- Kyle Sexton

If you connect an * system to a company like VoicePulse with there Connect service you are able to get incoming calls unlimited for 7.99/mo and outgoing at .0295/minute. VoicePulse also offers an unlimited business plan. Are you able to sign up with VoicePulse unlimited plan and connect an * system to it? Also, with VoicePulse connect you're able to make unlimited simultaneous calls

Hi all, I've been testing various codecs to eliminate "choppiness" that I sometimes get on my Asterisk IAX2 <> DSL <> provider (Exgn) connections, and Speex seems to work the best, so far - but Speex seems oddly unpopular. Can anyone share their experiences with Speex (good and bad)? Is anyone using it in a production environment? I like the variable bit rate and

Calling 7777 from a local extension on my local network, I get good voice quality from asterisk, and asterisk reliably recognizes my dtmf input. I set up a sipphone trunk (free) and called in to it via a separate sipphone account on another computer, and got slightly lower, but still good, audio quality. I set up a FWD trunk (free) and called in from the other computer, and got somewhat lower

I'm in Mexico an I'll like to know wish is the best IAX or SIP Termination provider. Im tring to start a small Pre-paid long distance service. Thanks Erick

Hello all, I am working on a creating some intelligent failover dial-plan logic and I'm running into something that I'd like some feedback on. Basically, it appears that if you place a call to an IAX2 peer that refuses the connection, or is unavailable, a NOANSWER dialstatus is returned. Example: -- Executing Macro("IAX2/cubix-19",

Anyone have a good provider for International (US/Canada at least) 800 termination/origination? I have a customer that had us port one of their 800 numbers and apparently didn't realize that they had published that number in Canada as well. Our current origination/termination provider can't handle Canadian inbound calls to that number, so I need to find another provider that can.

Hi All Has anyone used Cubix / Firefly successfully with Asterisk? When someone calls a Cubix softphone, Cubix never seems to answer the call correctly. The other person just hears ringing even though it has been answered. I am using IAX as the SIP support doesn't seem to 100% either. Idefisk works 100% on the same setup. Kind Regards Garth

Hi. I've searched the wiki and the newsgroup to no avail. We have a couple of grandstream phones and purchased seceral copies of the G729 codec for use with them. We registered the codec (first time) on a testing machine and it worked. We registered it on the production machine (second time) and it worked. Then the production system crashed. After reinstallation we had to

Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100505/5068aaab/attachment.htm

I've been losing patience with my current provider, a small company called Sellvoip. Their termination is good, and they are asterisk based, but they are understaffed and have no concept of customer service. So I'm shopping. I am interested in the opinions of others on the providers they work with. Here are my criteria, roughly in order a) Decent quality, low latency. In

Hi, I am looking for voip termination all over the world especially based on IAX or SIP. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040124/25e16736/attachment.htm