similar to: Problems with Simpletelecom and *

I'm having a very annoying problem with access my asterisk system from work. Our phone system here only produces very very short DTMF tones. The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE => CALLUK.COM 0870 => IAX Inbound The

Hello all. I'm having an odd problem getting * and sipgate to work together. From Sipgate support I have gotten this repsonse to my query: ===== Your Asterisk is registering incorrectly with our servers. It registers like this: sip:s@217.XXX.XXX.XXX:5076 The "s" should be your SIP ID. Anything else is rejected. I don't know where you can find this setting, but from our

I'm working on an IVR that gives the users the option (number 5 in the main menu) to dial by extension: exten => 5,1,Set(TIMEOUT(digit)=5) ; Dial Extension exten => 5,2,Set(TIMEOUT(response)=10) exten => 5,3,Background(LCL/prompt-60) exten => 5,4,WaitExten(15) When going option 5 you can dial some extensions such as 2802, it goes to the extension (all extens start with 28 on the

Hello all. I'm having a problem debugging an IVR I'm building. I can't see any reason this shouldn't be working. Firstly the asterisk version is: Asterisk SVN-trunk-r7230 built by root @ localhost.localdomain on a i686 running Linux on 2006-02-17 22:44:48 UTC Basically the problem is this. While the playbacks are happening you can push any one of the options and to happily

I've noticed that my * box isn't logging all that it use to / should. I'm running version Asterisk CVS-v1-0-03/07/05-22:42:03, prior versions would log everything including connections to voicemail and such. This version of (or more likely my configs) seems only to be logging certain things. It logs most calls to both Master.csv and MySQL, but there are still loads of calls

Hello, I'm having a problem I can't seen to figure out. In a nut shell, I have asterisk running with 4 accounts configured. All accounts work fine for local calling to each other and voicemail. However, only 1 account can make outgoing calls. All the others fail with the following error. If anyone can shed some light on the possible problem or where to look for more info it

Does anyone here use either Gradewell or inWeb for service? They are both UK based. I'm trying to get a couple of inbound IAX2 based numbers from both of them to work with no luck at all. The one thing that sets these guys apart from the rest of companies offering inbound numbers is they tie the account to the IP of the asterisk server. Neither use register lines in iax.conf, there appears

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

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

Hi, all. The asterisk setup is working fine, receiving calls via broadvoice "initially". ? When call comes in via broadvoice number, asterisk picks it up and routes correctly, as long as the call came in within ~2 min from the previous one. In other words, as long as a call comes in within ~2 min since the previous one, asterisk will answer the call. However, if the call comes in

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 experience with them? are they a SCAM? Thanks! </Madhawa>

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

Hello, Dovecot log is showing too many POP3 RETR entries which are identical lines. I also suspect that it is causing high pop traffic eating most of the network bandwidth. Here are some of the lines out of 11009 in a day. Such pattern is observed only for few users. dovecot version is 2.1.17. ============== Mar 20 00:00:07 pi3 dovecot: pop3(user at example.com): Disconnected: Logged out

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

A quick status report on how 1.0rc8 behaved in service for a few hours with several hundred simultaneous users, at a site very new to dovecot. Oh, and a question at the end. Summary: Reasonable for a first shot but one significant problem, requiring backing off. Background: We have a long-established UW-IMAP service for a user population of about 20,000 based on a few Linux (Redhat) machines

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

Hi: We have got SIP clients connecting to our Asterisk fine with a DSL connection behind router (NAT), but when we bring the 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

What the heck does this mean? This is the first time I've seen this. Calls were going through ok for a couple weeks now. Dec 6 09:22:24 WARNING[1121866688]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/1400-45fb(4) to IAX2/simple/2(256) Dec 6 09:22:24 WARNING[1121866688]: app_dial.c:998 dial_exec: Had to drop call because I couldn't make SIP/1400-45fb

Hi, Does anyone have suggestions hooking Zyxel Prestige 2002 to Asterisk? I have tested Zyxel Prestige with both supported 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

Anyone out there using NetLogic DIDs? And have inbound working? I got outbound working, but no joy so far with inbound. Here are the relevant parts from my conf files: iax.conf [general] tos=lowdelay jitterbuffer=no register => username:secret@zoot.netlogic.net [netlogic] type=friend host=dynamic context=sourcekit-main auth=plaintext username= secret= disallow=all allow=ulaw allow=all