search for: theperson

...swer at this time (1:0/0/0) -- Executing Voicemail("SIP/4560-4e18", "u4560") Mar 25 04:46:23 WARNING[28472]: app_voicemail.c:1713 invent_message: fn [/var/spool/asterisk/voicemail/default/4560/greet] Mar 25 04:46:23 WARNING[28472]: file.c:480 ast_openstream_full: File en/vm-theperson Mar 25 04:46:23 WARNING[28472]: file.c:485 ast_openstream_full: File vm-theperson Mar 25 04:46:23 WARNING[28472]: file.c:489 ast_openstream_full: File vm-theperson does not exist in any format Mar 25 04:46:23 WARNING[28472]: file.c:490 ast_openstream_full: File vm-theperson does not exist in any fo...

Hi *, I have set up a very simple asterisk configuration where I intend to be redirected to the voicemail whenever I dial 100 with my kphone SIP client. The problem is that asterisk can not find the stream 'vm-theperson'. I have made a non-standard installation (since I am just testing), and that file is located in /mnt/tr2/fake_root/installed/usr/local/var/lib/asterisk/sounds. 1. How can I tell asterisk where to look for the streams? 2. I have traced (with strace) all 6 threads created by asterisk to find o...

...of the build process. This is the output of the build when the error occurs: done rm -f /tmp/asterisk/var/lib/asterisk/mohmp3/sample-hold.mp3 mkdir -p /tmp/asterisk/var/spool/asterisk/voicemail/default/1234/INBOX :> /tmp/asterisk/var/spool/asterisk/voicemail/default/1234/unavail.gsm for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isunavail; do \ cat /tmp/asterisk/var/lib/asterisk/sounds/$x.gsm >> /tmp/asterisk/var/spool/asterisk/voicemail/default/1234/unavail.gsm ; \ done :> /tmp/asterisk/var/spool/asterisk/voicemail/default/1234/busy.gsm for x in vm-theperson digits/1...

Hi all, 2 days ago, managed to install rhat and asterisk, starting with 2 sip phones, simple config. CLI reports : "playing 'vm-theperson' (language 'no')" when transferred to voicemail after timeout, or "playing 'vm-password' (language 'no')" when dialing into voicemail extension. But, no sound, nothing heard in phones. First time, first problem, what am I missing, please help rgds Fr...

I have quite an old version of Chameleon Mail, currently the prompts played when leaving a message are ? -- Executing VoiceMail("SIP/209-3b0e", "u5") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/5' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message I...

...utput: //WORKING -- Executing [13479012223 at vm:1] NoOp("SIP/n1-00000281", "13479012050") in new stack -- Executing [13479012223 at vm:2] VoiceMail("SIP/n1-00000281", "13479012050 at default|u") in new stack -- <SIP/n1-00000281> Playing 'vm-theperson' (language 'en') -- <SIP/n1-00000281> Playing 'digits/1' (language 'en') -- <SIP/n1-00000281> Playing 'digits/9' (language 'en') -- <SIP/n1-00000281> Playing 'digits/0' (language 'en =============================...

...4838bc8", "CHANUNAVAIL,1") in new stack -- Goto (macro-phone,CHANUNAVAIL,1) -- Executing [CHANUNAVAIL at macro-phone:1] VoiceMail("SIP/ipkall-trunk-14838bc8", "1651xxxxxxx at default,u") in new stack -- <SIP/ipkall-trunk-14838bc8> Playing 'vm-theperson.gsm' (language 'en') == Spawn extension (macro-phone, CHANUNAVAIL, 1) exited non-zero on 'SIP/ipkall-trunk-14838bc8' in macro 'phone' == Spawn extension (mydids, 165xxxxxx, 1) exited non-zero on 'SIP/ipkall-trunk-14838bc8' Sometimes it works properly and the...

Hey guys, I have stdexten macro dialplan and I have to handle those who doesn't have voicemail box setup. Right now if someone call and if person unavailable the it's just hangup that call. I want it say "person doest have vm setup yet." smthing like that. How should I handle this in my dialplan ? -- Sent from my iPhone

...quot;, "SIP/sipset") in new stack == Everyone is busy at this time -- Executing VoiceMail("SIP/sippc-b5f6", "b421") in new stack == Parsing '/etc/asterisk/voicemail.conf': == Parsing '/etc/asterisk/voicemail.conf': Found -- Playing 'vm-theperson' -- Playing 'digits/4' -- Playing 'digits/2' -- Playing 'digits/1' -- Playing 'vm-isonphone' -- Playing 'vm-intro' == Spawn extension (default, 421, 102) exited non-zero on 'SIP/sippc-b5f6' Any help is appreciated. Thanks....

...", "0?200:300") in new stack -- Goto (macro-get-vmcontext,s,300) -- Executing NoOp("IAX2/vmail-ottawa-peer-5", "") in new stack -- Executing VoiceMail("IAX2/vmail-ottawa-peer-5", "2502@default|u") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/5' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'vm-isunavail'...

I have googled this one to death, and can't find anything. I added a number of new users to my asterisk (current CVS) system. I am using the "Voicemail2" family. I added entries in extensions.conf and voicemail.conf for my new users, and I have tested leaving and retrieving new voicemails for them. All of this works fine. But if one of the new users tries to "Administer

...t Here is what the log shows: -- Zap/1-1 is ringing -- Nobody picked up in 20000 ms -- Hungup 'Zap/1-1' -- Executing Ringing("Zap/49-1", "") in new stack -- Executing VoiceMail("Zap/49-1", "u100") in new stack -- Playing 'vm-theperson' (language 'en') May 2 18:36:45 WARNING[163851]: chan_zap.c:1193 zt_set_hook: zt hook failed: Device or resource busy -- Starting simple switch on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Playing 'digits/1' (language 'en') -- Playing 'digits/...

...i: Dial return value was 0 and dialstring was SIP/203|15|tr -- AGI Script dialparties.agi completed, returning 0 -- Executing Wait("SIP/202-ed3d", "1") in new stack -- Executing VoiceMail("SIP/202-ed3d", "u203") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/3' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro'...

Hi All, Anyone know if theres a way to share a Cisco 7940 between hot-desk users? My phones get their setup via SIP .cnf files, that load at boot via tftp, so I'm assuming the configs a failry static. However if I want a phone to be hot-desked, I could have different users sitting there. Is there any concept of "logging on" in these environments? Cheers, Adrian

...ERBOSE[6554] logger.c: -- Nobody picked up in 20000 ms [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Executing Voicemail("SIP/PACK501-480b08c0", "103 at enge-local|u") [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- <SIP/PACK501-480b08c0> Playing 'vm-theperson' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- <SIP/PACK501-480b08c0> Playing 'digits/1' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- <SIP/PACK501-480b08c0> Playing 'digits/0' (language 'en') [2...

...xecuting Dial("SIP/67.23.212.25-0814f830", "SIP/334|20") in new stack -- Called 334 -- SIP/334-26f8 is ringing -- Nobody picked up in 20000 ms -- Executing VoiceMail("SIP/67.23.212.25-0814f830", "u334") in new stack -- Playing 'vm-theperson' (language 'en') == Spawn extension (default, 4084, 2) exited non-zero on 'SIP/67.23.212.25-0814f830' I have dropped this question at the asterisk user list some days ago, but it's being ignored... (or nobody has the answer) Can anybody shed some light on this ? Warm...

...lled 200 -- Started music on hold, class 'default', on SIP/201-1e08 -- SIP/200-b925 is ringing -- Stopped music on hold on SIP/201-1e08 -- Nobody picked up in 20000 ms -- Executing VoiceMail("SIP/201-1e08", "su200") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'beep' (lan...

...>> /var/log/asterisk/call_log") in new stack -- Executing [s at from-pstn:4] Answer("DAHDI/1-1", "") in new stack -- Executing [s at from-pstn:5] VoiceMail("DAHDI/1-1", "5000,u") in new stack -- Stopped music on hold on DAHDI/1-1 -- Playing 'vm-theperson.gsm' (language 'my') -- Playing 'digits/5.gsm' (language 'my') -- Playing 'digits/0.gsm' (language 'my') -- Playing 'digits/0.gsm' (language 'my') -- Playing 'digits/0.gsm' (language 'my') -- Playing 'vm-isunavail.gsm...

...efault,stdexten-NOANSWER,1) -- Executing [stdexten-NOANSWER at default:1] VoiceMail("SIP/20321-00000002", "20545,u") in new stack > 0x7f66440060b0 -- Probation passed - setting RTP source address to 10.128.1.47:7078 -- <SIP/20321-00000002> Playing 'vm-theperson.ulaw' (language 'en') -- <SIP/20321-00000002> Playing 'digits/2.ulaw' (language 'en') -- <SIP/20321-00000002> Playing 'digits/0.ulaw' (language 'en') -- <SIP/20321-00000002> Playing 'digits/5.ulaw' (language 'en...

...quot;, "SIP/904|90|t") in new stack -- Called 904 -- SIP/904-3622 is ringing -- Local/904@sanset-1df8,1 is ringing -- Nobody picked up in 22000 ms -- Executing VoiceMail("IAX2/comcentrixs-uplink-6", "u944@sanset") in new stack -- Playing 'vm-theperson' (language 'de') == Spawn extension (sanset, 904, 3) exited non-zero on 'Local/904@sanset-1df8,2' Mar 28 21:55:06 WARNING[4187]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x819ea38', 10 retries! -- Playing 'digits/9' (language 'de...