similar to: Nufone not taking GSM CALLS

Brian, Great job fixing up the list server... postings are happening very quickly now (within minutes)! Thanks, Rich

Hi, Anyone know if Asterisk and Digium Hardware supports Centos-3.1 which is clone of Redhat Enterprise 3.1 server.? -- Best regards, Frankie (fgravato@cfsdigital.com) mailto:nanog@cfsdigital.com

Folks, I've ordered a new account from Nufone last month. Transferred money to Nufone through their paypal account. I had communication with Nufone sales up until two weeks back. Since then there were no replies to my emails. I am afraid with this kind of unresponsiveness how one would run a reliable service with this company. Have no bad feeling with Jeremy as the author of widely used h323

I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse

Hi list I get this error message when I try to call to another city via IAX, in my iax.conf I have the codec iLBC but until now it works well, how can I solve this problem? -- Executing Dial("Zap/2-1", "IAX2/tuxtla/111110@default|60|Ttr") in new stack Feb 15 13:37:05 WARNING[360466]: chan_iax2.c:6266 iax2_request: Unable to create translator path for UNKN to GSM on

Hi all, I have a network with some asterisk in trunk with IAX2 and some SIP/ZAP phone connect to this *. In every call I need to use only alaw codec so in all conf file I have set disallow=all and allow=alaw. I try also to make some tuning of my environment removing unused codec and application. If I remove the codec_ulaw.so when I try to call I see this: [Oct 5 12:15:33] WARNING[16637]:

Hi List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes

Hi i was trying to call 17009978275 which is my Fwd line on my notebook from Asterisk and i keep getting this message on the console. -- Executing Dial("Zap/2-1", "IAX2/@iaxtel.com/17009978275@iaxtel") in new stack -- Called @iaxtel.com/17009978275@iaxtel WARNING[1150495040]: File chan_iax2.c, Line 4547 (socket_read): I don't know how to authenticate rob to

I recently plugged in Phone to my TDM400P Card to test out something I mostly use sip phones to interface with *. All of sudden I'm getting lot of line static noise coming of the card is there any settings I should look at or anything I need to do on the command line at this point I'm open to any ideas I'm running 0.7.1 on Redhat 9.0 machine. Any insight would be greatly appreciated.

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Notice Zap/g2 -- Executing Dial("SIP/chad.brown-d1ac", "Zap/g2/9528737") in new stack Aug 1 00:42:43 NOTICE[1200884528]: app_dial.c:714 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time Does anyone know why Asterisk wants to use group 2 regardless of how I am configured. Take a look at how I'm configured. Shouldn't

Hello All, Has anyone ever seen this before. This only happens when i'm on phone call -- Zap/2-1 is ringing -- SIP/2203-c48d is ringing -- SIP/2202-f2ad is ringing -- SIP/2204-11cd is ringing -- SIP/2205-ce62 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- SIP/2205-ce62 answered Zap/1-1 -- Hungup 'Zap/2-1' Jan

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

Hi I just signed up with voicepulse's voice connect service. then emailed me over configs for my extentions and iax i enter in all the info and when i start up * and do show registry it seems to be rejecting my login. Has anyone seen this before.. Any further insite will be greatly appreciated. thanks frankie (aim)cronparser (irc)crontibs 17006240093 -------------- next part

With the 1U Apple G5 server at a good reliability/storage/pricepoint level... got to thinking about compiling Asterisk on OS X.. at least for SIP phone call switching, voicemail, etc. Has anybody attempted this? Email me off list if this is too dev-heavy for the user list. Thanks, Ted W -----Original Message----- From: asterisk-users-request@lists.digium.com

Oops, I had some problems and was offline unable to remind you about the conference yesterday. LISTEN to recent recordings: http://x2z.eu/astusers.htm (Flash player, will autostart) THIS WEEK: Stephan Winterberg and Stephen Boche tell us more about the new book, whick looks like a great effort. A surprise visit from Jeremy, one of the pioneers of our community who started Nufone when someone

> -----Original Message----- > From: Asterisk@gtcus.com [mailto:Asterisk@gtcus.com] > Sent: Saturday, September 06, 2003 8:39 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] VONAGE or IP Dialtone > > > Thanks for the great feedback on these options. I am fairly > new at this and not familiar with the IAX/IAX2 capabilities > offered by

Does anybody know how to do an international call using NuFone. I realise this isn't really the place to ask, but NuFone appears to be closed for the weekend and would like to have a try at this before tomorrow. I assumed it would be '011' for an international line followed by country code but that doesn't seem to work. I am getting: -- Executing

I'm testing out my SNOM 200 phone by trying to call out through NuFone. When I do so, I don't hear an echo at all (in fact I can't hear myself through the phone) but the callee can hear an echo when she speaks. NuFone tells me their network is totally digital and so can't be involved in an echo. This is all well and good, but the echo is still there. Any suggestions? As a

hi, my * box is behind a NAT. i am using netgear router. it seems that my toll free number is unable to register with nufone because my * box is behind NAT firewall. the outgoing calls are working fine and nice. what can be done? cm configurations are fine. ===== Designs __________________________________ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/