similar to: X-lite not hanging up / DTMF not present through voipuser.org

Hi all, does someone knows what happen with voipuser.org web site and services? Registration failed since more than 24 hours and no access to the web site :-( Regards -- Daniel

Hi all I have been googling and wiki-ing and have found a number of potential solutions to my questions, but I don't want to have to play about for too long and risk messing up my * box now I've just got it working, if one of you kind folk could offer your 2 penneth, (being a Brit I'll have none of this cents business ;] ). I want to send an SMS message whenever I get a voicemail

Hi, I am simulating the sending of fax using sendfax through voip to reach an Asteria server via ZAP/1 ( PSTN phone line ) which then route call to a fax machine at ZAP/2. It seems like I am not able to establish any handshake with the physical fax machine using the sendfax program. Does anyone know why that happens and how to fix it? The scenario will be deployed in remote location in the

Here's something that should be easy for RUBY pro's. Here is a script: 1.times do r = $agi.exec('DIAL', SIP/voipuser&Zap/32&Zap/33&Zap/34&Zap/35) r = $agi.get_variable('DIALSTATUS') # $agi.set_variable(' WHOANSWERED

> -----Original Message----- > From: Mark Charlton [mailto:asterisk@mcwebtree.com] > Plus if you send your users to VoicemailMain(${CALLERIDNUM}) > they don't hear > it at all. > They just get "enter password". Yup. If you do that, the only time they hear it is during the initial setup call (if you have "forcename=yes" or

Hi All, Asterisk 1.4.12 on CentOS 5 Yesterday and today I got the following warnings in /var/log/asterisk/messages: WARNING[2066] chan_sip.c: Got authentication request (401) on unknown REGISTER to '<sip:account at sip.voipuser.org>;tag=d8f15e1f30efddd35168b07dba9d540e.3922' The corresponding bits in sip.conf are: register =>

Hi I have been trying for a while to find a way to get an SMS send when I receive a voicemail into my asterisk system. I don't want to send an SMS if the caller doesn't leave a message. I have voicemail.conf set up to email and delete. 302 => 302,Website Sales,sip@example.com,,attach=yes|delete=yes However I can't seem to find a way to test is a message is left. I have tried

JID Pri S Owner Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host

Hi. I'm new here so I hope this is a sensible question/sensible place for it. I have a PSTN to IAX phone number with voipuser.org that I'm using to test an IVR service. The only trouble is that after approximately 40 seconds of silence (e.g. after background playback of a menu prompt) the call gets cut off. Is this a common problem? I've already set the ResponseTimeout to a big

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

Dear friends, Asterisk's voicemail functions work fine for me, but I am having difficulty storing the voice messages inside mysql. My real-time CDR recording works so I assume the odbc connection is fine. The voicemail.conf I have is : [general] format = wav attach = yes dbuser=root dbpass=sqlpass dbhost=localhost dbname=asterisk odbcstorage=asterisk odbctable=voicemessages Asterisk shows

Hello, I am using voipuser.org service, and am trying to make a SIP call. Everything seems to work fine, except I can't hear anything on my end. When I make a SIP call, the other party can hear me, but I can't hear anything. I am using asterisk + Digium TDM board with an FXO port where I connect a regular telephone. Can anyone assist? I believe I have some asterisk

Hello R users! I have several data frames where some of the variables have many missing observations. For example, Q1 in one of my data frames has over 66% of its observations missing. I have tried imputation with mice but it does not work for all the data frames and I get the following message or a similar message to this: iter imp variable 1 1 Q1 Q2 Q3 Q4 Q5 Q6 Q7 Q8 Q9 Q10 Q11

Hi, I am new to Asterisk and I am having a setup problem that I am trying to resolved for the last couple days without any success. I am pretty much desperated on this issue and I don't know why. Can someone please kindly help me to troubleshoot this? I can't hear any audio from Asterisk when running Playback or VoiceMail tests. I have my Asterisk server ( running on Debian,

Hi I believe this is due to the way US phone systems work, however I'm going to ask anyway. In the UK there are several providers who provide national rate PSTN => Voip gateways which are free to receive calls on, (for the recipient), the caller pays the cost of calling. E.g 0844 0870 etc. I am looking for a US provier who offers the same sort of system. I don't call the US but I

I've just ordered a Siemens Gigaset C450 IP cordless IP/DECT phone, given that it's supported by asterisk http://www.voipuser.org/review_41.html However, I see that a slightly better Gigaset S450 IP is available for only a slight price premium. Are there any user experiences with the S450 IP? -- Eugen* Leitl <a href="http://leitl.org">leitl</a> http://leitl.org

Can you clarify what the nature of the security restriction is? If you can't run the R installer then how it is that you could run R? That would still involve running an external exe even if it came in a zip file. Could it be that the restriction is not on running exe files but on downloading them? If that is it then there are obvious workarounds (rename it not to have an exe externsion or

Hi, I'm trying to get h323 communication working between asterisk (0.7.1) and Innovaphone Gatekeeper + innovaphone phones. chan_323 installed OK with currently recommended pwlib_1.5.2 and openh323_1.12.2. Registration asterisk with the gatekeeper works OK, externsion for my test(sip) phone gets registered with gatekeeper. when establishing a call between a h323 phone and asterisk I run into

Hi all, I am using Digium PCI board to receive PSTN call through regular phone line. It is no problem for me to receive calls, but I am not able to obtain the Caller ID if the calls are from the phone line. exten => s,1,Answer() exten => s, n, Verbose(1|incoming number is ${CHANNEL} calling to ${EXTEN} routing to ${phonenum} ) exten => s,n, Verbose(1|callid is ${CALLID(num)}) exten