search for: voipus

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 ',...) retry if r.message.include?('BUSY')...

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 context for the s...

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

...> actual format = alaw, > host prefs = (alaw), > priority = mine -- Executing [0033661681 at fax-out:1] NoOp("IAX2/iaxmodem-1", "we are at fax-out") in new stack -- Executing [0033661681 at fax-out:2] Dial("IAX2/iaxmodem-1", "SIP/voipuser/0033661681") in new stack -- Called voipuser/008675533661681 -- SIP/voipuser-081f99c0 is making progress passing it to IAX2/iaxmodem-1 [Mar 28 04:01:00] NOTICE[16748]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50)...

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 number. It doesn't get sent to...

.../wengophoneng/wengo/rev12359/trunk/" for peer 2001 -- Executing [100 at my-phones:1] VoiceMail("SIP/2001-08225788", "2000") in new stack -- <SIP/2001-08225788> Playing 'vm-intro' (language 'en') -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host sip.voipuser.org, port 5060 -- <SIP/2001-08225788> Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/2000/tmp/OayHq7 format: wav, 0x821ebd0 -- User hung up The recordin...

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 => <account>:<secret>@sip.voipuser.org/<TelNo> and [TelNo] type=friend context=external nat=yes username=<account> secret=<secret> host=sip.voipuser.org fro...

...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 sip.voipuser.org, port 5060 -- Accepting AUTHENTICATED call from 127.0.0.1: > requested format = alaw, > requested prefs = (), > actual format = alaw, > host prefs = (alaw), > priority = mine -- Executing [003366168...

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 ________________________________...

...anyone can shed some light on the possible problem or where to look for more info it would be greatly appreciated. -- Accepting AUTHENTICATED call from 172.xx.xx.xx, requested format = 1024, actual format = 1024 -- Executing Dial("IAX2/xxxx@xxxx/1", "SIP/00441223xxxxxx@voipuser|40|r") in new stack -- Called 00441223xxxxxx@voipuser Jan 18 11:19:21 WARNING[27087]: chan_sip.c:6811 handle_response: Forbidden - wrong password on authentication for INVITE to '""xxxx xxxx" <sip:xxxxxxxx@217.xxx.xxx.xxx>;tag=as6471df4d' -- SIP/voipu...

...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 left on my *@home 0.6 box. I don't have any pstn attached the the box, and I am running FWD, voipuser, and alg as providers for various routes and redundancy. I can find a number of providers for sending SMS via pstn to BT, but nothing to reliably use online SMS services. I saw a few for outbound that were in beta. Basically can anyone recommend an SMS client for internet only usage in the UK....

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 t...

...ay start earlier so please check the site, IRC, Twitter or Facebook for the exact start time. If any of you are planning to be there, please email me if you'd care to have coffee or something stronger. Also, if you happen to be in the vicinity of Paris, we have an invitation left. Site: http://voipusersconference.org IRC Freenode.net #vuc Twitter: http://twitter.com/voipusers Facebook: http://facebook.com/voipusers

...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 => s,n,Verbose(1|callpres is ${CALLINGPRES}) exten => s,n,Dial(SIP/${phonenum}@voipuser,60) -- Executing [s at incoming:3] Verbose("Zap/1-1", "1|incoming number is Zap/1-1 calling to s routing to ") in new stack -- Executing [s at incoming:4] Verbose("Zap/1-1", "1| callidall is ") in new stack callidall is -- Executing [s at...

...is new asterisk no longer matches the nonce that it sent to this phone (check_auth is called with NULL third argument). This does look like a bug in the phone firmware. However, the phone can successfully initiate calls via several commercial and community providers. I tried iconnecthere.com and voipuser.org and it works! Now, the question: could it be possible to make the phone work with asterisk? Any ideas? I can send the log on request (it is rather big). Thanks. Eugene P.S. I also have a problem calling from asterisk to iconnecthere but I'll report it in a separate message. ---------...

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 Unmonitored 5561/5561 192.168.4.5 D N A 255.255.255.255 5061 Unmonitored 4561/4561...

...ernal source. (It isn't the power, as the two phones are in different buildings in different towns. Besides, both are connected to UPSs). To me this seems to indicate a fault with the phones themselves. But for both of them to develop the same fault at the same time seems odd. I asked on the voipuser.org forum if anybody else had had similar problems, but everybody who responded said all was well with their phones. But given the wider reach of this list, I thought I'd ask here as well. Both phones have the latest firmware from the GS website. Does anyone have any ideas? Has anyone had...

...ce any deja vu.] I have a * server acting as an IVR system. The calls come in via IAX. After a period of about 40 seconds of silence (either waiting for the caller to dial an extension, or with the audio paused in controlplayback), the call hangs up. All I see in the CLI is "Hungup 'IAX2/voipuser@...'" I know it isn't a timeout in * because they give a more useful message - even so I've now set the absolute, digit and response timeouts to be large, and created t and T extensions, but it's still happening! I have 3 inbound phone numbers acquired from 3 different place...

PSI System Admin-Message-ID: <112664837810704203@ns1.psinetworks.net> Hi list members, I'm sure this question has been posted before but I am still unable to find the answer. I have a TDM 400P line card and I would like to set it up to IGNORE the distinctive ring pattern that I have for a fax machine. Many thanks Brad