search for: u3000

...-- Executing Dial("SIP/3001-ec1c", "SIP/3000|20") in new stack -- Called 3000 -- Got SIP response 488 "Not Acceptable Here" back from 206.222.193.92 == No one is available to answer at this time -- Executing VoiceMail("SIP/3001-ec1c", "u3000") in new stack == Parsing '/etc/asterisk/voicemail.conf': Found -- Playing 'vm/3000/unavail' My sip.conf looks like: [general] port = 5060...

...led 3000 Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum retries exceeded on call 033f41c2187409b13ca364502ea9434e@206.222.193.101 for seqno 102 (Critical Request) == No one is available to answer at this time -- Executing VoiceMail2("SIP/3008-9a9b", "u3000") in new stack -- Playing 'voicemail/default/3000/greet' (language 'en') -- Playing 'vm-isunavail' (language 'en') but the phone doesn't ring. The 7960 is registered and can place outbound calls. Same with multiple 7960's. Did I miss a mandator...

...: 4[Feb 26 18:14:42] WARNING[4581]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'sip' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [3000 at centos-context:3] VoiceMail('IAX2/iax1Centos-1', 'u3000 at default') in new stack[Feb 26 18:14:42] WARNING[4581]: app_voicemail.c:2850 leave_voicemail: No entry in voicemail config file for '3000' == Auto fallthrough, channel 'IAX2/iax1Centos-1' status is 'CHANUNAVAIL'Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX...

...quot;SIP/3000|60|tr") in new stack -- Called 3000 Aug 25 10:16:13 NOTICE[4092]: chan_sip.c:1806 auto_congest: Auto-congesting SIP/3000-d838 -- SIP/3000-d838 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing VoiceMail("Zap/1-1", "u3000") in new stack -- Playing '/var/spool/asterisk/voicemail/default/3000/unavail' (language 'en') -- Playing 'vm-intro' (language 'en') == Spawn extension (from-pots-internal, 3000, 3) exited non-zero on 'Zap/1-1' -- Executing Hangup(&qu...

...>> >retries >> > exceeded on call 033f41c2187409b13ca364502ea9434e@206.222.193.101 for >> > seqno 102 >> > (Critical Request) >> > == No one is available to answer at this time >> > -- Executing VoiceMail2("SIP/3008-9a9b", "u3000") in new stack >> > -- Playing 'voicemail/default/3000/greet' (language 'en') >> > -- Playing 'vm-isunavail' (language 'en') >> > >> >but the phone doesn't ring. The 7960 is registered and can place >> >outb...

...not find one with some quick googling): U0009-U000D (Control characters, containing TAB, CR and LF) U0020 SPACE U0085 NEL U00A0 NBSP U1680 OGHAM SPACE MARK U180E MONGOLIAN VOWEL SEPARATOR U2000-U200A (different sorts of spaces) U2028 LSP U2029 PSP U202F NARROW NBSP U205F MEDIUM MATHEMATICAL SPACE U3000 IDEOGRAPHIC SPACE would it be possible for R to treat these all in the same way? (Or does it already but my R is misconfigured?) Hadley -- http://had.co.nz/

Hello I am still having problems with IAXTELL and FWD configuration. I get the following when I dial 17009965342 which is set as an example to dial to FWD people. 1+700+99+ 5 digit number. I have placed XXXXX where my passwords are. CLI> Executing Dial("Zap/14-1", "IAX/abatista:xxxxxx@iaxtel.com/917009965342@iaxtel") in new stack -- Calling using options

...lled 3000 Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum retries exceeded on call 033f41c2187409b13ca364502ea9434e@206.222.193.101 for seqno 102 (Critical Request) == No one is available to answer at this time -- Executing VoiceMail2("SIP/3008-9a9b", "u3000") in new stack -- Playing 'voicemail/default/3000/greet' (language 'en') -- Playing 'vm-isunavail' (language 'en') but the phone doesn't ring. The 7960 is registered and can place outbound calls. Same with multiple 7960's. Did I miss a mandator...

I have an X100P used, at present, largely for outgoing calls. It shares the single incoming POTS line with a number of analog phones. Is it possible to talk the X100P (Zap/1) to answer a ringing call only if I ask it to? I'd like to use only the SIP phone in my office, but let the analog phones continue to work in the rest of the house (until I can afford FXS cards anyway..) I can force

[3000] type=friend dtmfmode=INFO insecure=yes canreinvite=no auth=plaintext host=dynamic allow=ulaw [2000] type=friend dtmfmode=INFO insecure=yes canreinvite=no auth=plaintext host=dynamic allow=ulaw i have declared these two users 3000 and 2000. they are registering successfully. problem is that

Hi, I have an FXO card connected to my phone line which works in Asterisk as Zap/1. Is there any way of detecting whether something else is on the line before picking up on this channel? For example, I dont want to pick up and dial out on the line if someone is on it using another phone (which is connected directly to the line, rather than through Asterisk). Also, when an incoming call comes

...00,3,Hangup exten => 1000,102,Voicemail,b1000 exten => 1000,103,Hangup exten => 2000,1,Dial,Zap/2|20 exten => 2000,2,Voicemail,u2000 exten => 2000,3,Hangup exten => 2000,102,Voicemail,b2000 exten => 2000,103,Hangup exten => 3000,1,Dial,Zap/3|20 exten => 3000,2,Voicemail,u3000 exten => 3000,3,Hangup exten => 3000,102,Voicemail,b3000 exten => 3000,103,Hangup exten => _NXXXXXX,1,Dial(Zap/4/${EXTEN}|20,t) [incoming] exten => s,1,Wait(1) exten => s,2,Dial(Zap/g1|20,t) ; Calls the first available channel in group 1 exten => s,3,Voicemail,u9000 ; Directs...

...WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum > retries > exceeded on call 033f41c2187409b13ca364502ea9434e@206.222.193.101 for > seqno 102 > (Critical Request) > == No one is available to answer at this time > -- Executing VoiceMail2("SIP/3008-9a9b", "u3000") in new stack > -- Playing 'voicemail/default/3000/greet' (language 'en') > -- Playing 'vm-isunavail' (language 'en') > > but the phone doesn't ring. The 7960 is registered and can place > outbound calls. Same with multiple 7960's. &...

Hi... I have the next problem.. I have a FXO card with i can make calls but i cant recive calls. At the consol, i get the next error: -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/2-1 WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): Ring/Off-hook in strange state 6 on channel

hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM

...1,Dial(SIP/3000,20) ; ; Priority 2 send the caller to voicemail, and gives the "u"navailable ; message for user 2000, as recorded previously. The only way out ; of voicemail in this instance is to hang up, so we have reached ; the end of our priority list. ; exten => 3000,2,Voicemail(u3000) ; ; If the Dialed number in priority 1 above results in ; a "busy" code, then Dial will jump to 101 + (current priority) ; which in our case will be 101+1=102. This +101 jump is built ; into Asterisk and does not need to be defined. ; exten => 3000,102,Voicemail(b3000) exten => 3...