similar to: Urgent help = RUBY & AGI

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

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, We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this natted network. We have the issue with calls to these SIP phones - no audio. It is probably the problem with port forwarding on router - but I am not sure how can I forward same sip ports (5004 to 5100) to two phones (nat addresses?)? Any help appreciated! Z. Zivanovic -------------- next part

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

Hi folks, we've tried contacting Jean-Marc Valin but email address bounces. We published yesterday an advisory about libfishsound, you can find it at the following URL: http://www.ocert.org/advisories/ocert-2008-2.html The issues seems to affect Speex (since the code is the same) versions <= 1.1.12. While the 1.2beta branch is not vulnerable we advise that you fix with a security release

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 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

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

Hello gentlemen, I am new here. I have a D-Link DPH-80S SIP phone (it's a non-US model), and I am trying to make it work with Asterisk. I tried versions 1.0.7 and yesterday's CVS and the behavior is the same. The phone registers with no problem, and can accept calls. But when I try to make outgoing call, there is a series of invite requests from the phone, to which asterisk responds

Hello sip show peers does not mark hosts as NAT even though sip.conf and sip_peers table has nat=yes. spitfire*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status voipuser.org/gdsm 216.127.66.119 N 255.255.255.255 5060 Unmonitored 5560/5560 192.168.4.5 D N A 255.255.255.255 5060

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 =>

Hello everyone. I need a quick help on how to capture who answered the call with agi. Here is an example: -- Zap/32-1 is ringing -- Zap/33-1 is ringing -- Zap/34-1 is ringing -- Zap/35-1 is ringing -- SIP/operator1-e77f answered Zap/23-1 So how can I capture this value and write it to mysql? I already have this:

I am trying this approach to see who picked the line: Here is what i am doing: EXEC DIAL SIP/ vaso &Zap/35||M(testing^30086) Macro: [macro-testing] exten => s,1,DumpChan() exten => s,2,AGI(whopicked.rb) exten => s,3,Hangup()

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

I have a really baffling problem. A couple of months ago I purchased a pair of GrandStream GSX-2000 phones for use with Asterisk. At first all was well. But recently I've noticed terrible sound quality problems. Basically the sound will "glitch" or stutter randomly from time to time. Now, what is interesting is that this happens even with the phone totally disconnected from any

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

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

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 I have been lurking for a while, but now have a small problem or 3. 1) I have my inbound line via sip from VOIPUSER.ORG and have a simple extension selection menu on my * box. Internally the DTMF tones are present, (for xlite and * on same LAN), however calling in via the sip line from a pstn doesn't register any tones in asterisk. I have tried all the different DTMFMODE settings in the