search for: simpletelecom

Hi List! any body use www.simpletelecom.com? I subscribe to www.simpletelecom.com for A-Z termination and paid US$15.00 and US$70.00 via credit card in two days, but my account has US$15.00 only. I checked my credit card from the bank and they said me the payment already paid to merchant. I've lost US$70.00 :( so anyone here has exp...

I'm trying to figure out if this is a nat problem. I have a private network behind a freebsd nat box. The * server is on a static nat, with a private ip of 10.139.10.165. I'm connecting with sjphone as the client from 10.139.10.159. I am calling out using simpletelecom. When connecting directly to simpletelecom using sjphone everything works fine. When I go through * I get disconnected after about 20 seconds. I cannot seem to get my settings correct, and I don't understand the debug logs enough to know what's happening. What I would like to know is wh...

Hi, We have Asterisk, running on a P4 box running Suse 9.1, making calls using IAX through SimpleTelecom and Nufone. What we are looking for is toll quality voice. The problem is that voice over calls routed through SimpleTelecom and nNufone occassionally breaks. We also have a digium card and the calls over the digium card using the Zaptel Interface have a very good quality. We have enough...

Anyone using Simpletelecom with *? I had a nice working system with them, my credit can out so I apply another $5 to continue testing. Since then nothing has worked. I always get: -- Executing SetCallerID("SIP/line1-74ac", ""myname"|<>|a") in new stack -- Executing Dial(&quo...

Hmmm.... Can't place calls... Can't access website... Neither of the 3 nameservers answer anything... Anyone heard/know something to explain all this?

I spent the better part of the day trying to figure out why my SIP calls going through * were just going dead after 20 seconds. I was sure it was a nat issue but now I'm not so sure anymore. I have * on a public ip and clients behind a nat. I was using simpletelecom to terminate my calls. I could connect fine if I went direct from client -> simpletelecom. If I used * as a proxy the audio just went dead after 20 seconds. A few minutes ago I tested the same setup using FWD instead of simpletelecom. Everything works, no dead audio. And of course iax didn&...

...e Sipura 2000 ATA to a Rogers Cable connection behind a Netgear router (NAT), the SIP clients aren't able to reach the Asterisk at all. We enabled the SIP debug in Asterisk, and it doesn't see any request coming from these SIP clients, and we also tried the to use a XTEN Lite to connect to Simpletelecom within this network and it fails to register as well. It seems to be a network configuration problem, but there isn't much log in the router that we can dignose as the Netgear router WR814 only logs TCP web requests. Does anyone know what the problem could be? Does Rogers Cable blocks SIP po...

We have another 500 beta openings in the SimpleConnect beta. SimpleConnect is a service for you to make IAX/SIP calls from * or any IAX/SIP agent. Beta participants get free calls to anywhere in the United States and Canada. If you want to become a beta tester, just go to https://secure.simpletelecom.com/order/ . No credit card is required. We're looking forward to your feedback. Sean SimpleConnect

...G[1121866688]: app_dial.c:998 dial_exec: Had to drop call because I couldn't make SIP/1400-45fb compatible with IAX2/simple/2 Phone: Cisco 7960 SIP image 7.3 IAX.conf [general] port=5036 bandwidth=low disallow=all allow= ulaw jitterbuffer=1 dropcount=1 register => henry831:ttcomaha@iax.simpletelecom.com [simple] type=peer host=iax.simpletelecom.com username=henry831 secret=XXXXXXXXXx context=default extensions.conf exten => _1NXXNXXXXXX,1,SetCallerID(5555555555) exten => _1NXXNXXXXXX,2,Dial,iax2/henry831@simple/${EXTEN}

...ed codecs. Call with G.711 sounds very choppy and cracking. Almost can't understand a word. Today I installed G.729 support into Asterisk but unbearable voice quality remains. It's a little bit better though. I have tested that Zyxel ATA with some commercial SIP providers (stanaphone, simpletelecom etc.) and the quality is OK. tcpdumping connection with both providers showed that G.711 alaw/ulaw was in use. Some Grandstream phones are also hooked into that same Asterisk pbx and using G.711. Those devices voice quality is superb. Firmware installed in Zyxel Prestige 2002 is V3.60(MD.2)C0 (...

...2999,1,VoicemailMain(${CALLERIDNUM}) [sourcekit-main] include=>sourcekit-sip exten => +19193233010,1,GoTo(sourcekit-sip,101,1) exten => _1NXXNXXXXXX,1,SetCallerID(9193233010) exten => _1NXXNXXXXXX,2,Dial(IAX2/netlogic/${EXTEN}) exten => _1NXXNXXXXXX,3,Dial(IAX2/username:secret@iax.simpletelecom.com/${EXTEN}) exten => _1NXXNXXXXXX,4,Hangup [netlogic] include=>sourcekit-main and, thr debug output from * CLI: Asterisk Ready. *CLI> iax2 debug IAX2 Debugging Enabled *CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00017ms SCall: 0...

...9;t either). Suggestions? I've tried setting dtmfmode to inband, rfc2833 and info in the iax.conf file for one of the providers (sixtel) with no apparent effects. Right now, I'm using sixtel and voipjet; they're the only two that can complete calls for me; I'm still trying to get simpletelecom and diamondcard to work for just a plain voice call. I'm using the latest CVS version of Asterisk (as of a few hours ago). - James __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrosp...

Are there any IP Providers that will pass Caller ID? Broadvoice used to but no they dont. THX

Compiled from CVS today and no more dropping outbound calls after 40 secs. :D (was using cvs from 3 days ago) (just thought I'd pass it along in case anyone is still strugling with broadvoice calls)

...2999,1,VoicemailMain(${CALLERIDNUM}) [sourcekit-main] include=>sourcekit-sip exten => +19193233010,1,GoTo(sourcekit-sip,101,1) exten => _1NXXNXXXXXX,1,SetCallerID(9193233010) exten => _1NXXNXXXXXX,2,Dial(IAX2/netlogic/${EXTEN}) exten => _1NXXNXXXXXX,3,Dial(IAX2/username:secret@iax.simpletelecom.com/${EXTEN}) exten => _1NXXNXXXXXX,4,Hangup [netlogic] include=>sourcekit-main and, thr debug output from * CLI: Asterisk Ready. *CLI> iax2 debug IAX2 Debugging Enabled *CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00017ms SCall: 0...

I posted this question a while back, and I'm posting again in hopes that someone has some ideas. Sorry if you've already seen this. When dialing out using a SIP or IAX provider (Broadvoice, SimpleTelecom, VoicePulse Connect) I often find that after the call is answered the first few seconds of audio are cut off (i.e. I don't hear the called party). This usually results in the called party saying "hello.... Hello???" until I hear them. Has anyone else experienced this problem and foun...