similar to: IMAP voicemail storage.

hi folks. i have the following puzzle: when i call certain cell phone# using a regular phone & POTS. the called cell phone co. usually return a message such as phone travel out of range or phone is busy etc. if the phone is unreachable. now when i have the following setup: sip phone -> asterisk -> PRI -> phone co. i call the same cell# and if it's unavailable. the PRI return

hi folks. we've experienced some weird problems lately. we have about 600 SIP phone on a single system running *1.4.26.2 for about a month. recently there was massive UNREACHABLE messages like this one showed up: chan_sip.c: Peer '2699' is now UNREACHABLE! Last qualify: 1252 then they all became reachable again in a few seconds. sometimes it last for couple minutes. but sometimes

does anyone get a hold of the SPA-941 Provisioning Guide? i tried call Sipura's tech support, seems like none of them heard of the term "remote provisioning". they kept refering me to their web site which i've check thoroughly, and could not find any documentations on the SPA-941. finally they gave me a phone number to call, which appears to be a fax machine. that's when i

hi. i connected Asterisk to an NEC NEAX system with a crossover T1 cable and the Digium TE405P using E&M wink signaling. the connection's ok. however when dialing from the NEC to the Asterisk. most of the time the Asterisk only sees the first digit of the dialed number(which is 4 digits). some time if i dialed the 4 digits very fast it might get through. seems like there's a timming

i recently bought an SP4000 conference phone but having problem provisioning it using ftp, every time it just hangs at "Updating initial configuration..." screen. when i switch it to tftp, it'll work fine. i though it was bootrom/firmware issue so i upgrade it to bootrom 3.2.2/sip 2.0.1 but it makes no difference. any thoughts? p.s. i'm using debian sarge proftpd 1.2.10 and the

hi guys. i'm writing some simple applications for the cisco 7970 services button. i read the asterisk wiki and it mention there's a CMXML_App_Guide.pdf file but there's nowhere can i find a link for it. does anybody know where can i find it? regards. -- Edwin Lam <edwin.lam at officegeneral.com> Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370

hi folks. i'm experimenting with iaxmodem + hylafax using DID to determine where to send the fax to it's final destination. however i have difficulties passing the DID information from iaxmodem to hylafax. in extensions.conf: exten => _XXXX,1,Dial(IAX2/iaxmodem0/${EXTEN}|20|r) exten => _XXXX,n,Dial(IAX2/iaxmodem1/${EXTEN}|20|r) exten => _XXXX,n,Busy exten => _XXXX,n,Hangup

hi does any body know which version combination of spandsp/tx_fax/rx_fax will work with * 1.2.24? i tried different combo. they're either seg fault during runtime or won't compile. very frustrated :/ p.s. i know. hylafax/iaxmodem is far more stable. but i have specific reasons to use rx_fax. -- Edwin Lam <edwin.lam at officegeneral.com> Systems Engineer, Office General, Inc.

hi folks. is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? suppose i want to do something like this: exten => 1234,1,dial(SIP/1234) exten => 1234,2,<do something> but when the dial command hangs up normally, line 2 won't get executed. -- Edwin Lam <edwin@officegeneral.com> Systems Engineer, Office

hi folks. i have a Digium TE220 PCI-E 2 port T1/E1 controller installed in an IBM x3400 server. i load the wct4xxp driver seems ok. but when i execute "ztcfg -vvv" command. the kernel panic. i tried zaptel 1.2.21 & 22. they have the same result. following is my zaptel.conf: loadzone=cn defaultzone=cn span=1,1,0,ccs,hdb3 span=2,0,0,esf,b8zs bchan=1-15 dchan=16 bchan=17-31

hi folks. does any one have experience setting up E1 PRI in Shanghai, China? it works fine when we use SIP phone to dial out, however when using forward function on the same phone, it seems like it's dialing out but there's actually no respond from the phone company (China Telecom) and eventually the dial command will timed out. here's our PRI portion of zapata.conf:

our * crashed twice in a month with segmentation fault & a core dump. here's the stack trace: #0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6 #1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6 #2 0xb7e17090 in strdup () from /lib/tls/libc.so.6 #3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at

hi folks. we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and we start having problems of dropping calls (actually the calls wasn't dropped it just the sound was muted for about 5-10 seconds, but most users will think the call dropped and hangup/redial). i've check the console output. there was a lot of messages like the following: Sep 28 15:00:49 NOTICE[8182]:

hi folks. i've been experimenting with SILK codec and meet with some success on incorporating it in pjsip (an open source sip client). now i'm trying to do the same thing on Asterisk. any documentations, pointers, etc i should look into? any help is appreciated. -- Edwin Lam <edwin.lam at officegeneral.com> Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283

i've been using *1.2 w/ realtime static zapata in mysql table fine. but after i upgraded to 1.4. it seems like the zapata table doesn't load correctly. i have to go in the console and use the "zap restart" to get the zap channels register. is this sounds like a bug or something i'm missing when upgrading to 1.4? -- Edwin Lam <edwin.lam at officegeneral.com> Systems

hi folks. i have several remote sites with total of 200 sip phones connect to our Asterisk server. i want to minimize bandwidth usage and thinking about getting a Digium TC400B transcoder card. what are your experience with it? how's the quality? also if there are 120 active channels in used. will the 121 person able to make calls? will it support more channels if i put 2 cards in the system?

hi folks. i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having problems with fax. after receiving fax with the ReceiveFAX app. everything seems ok. the .tiff file was there, phone line seems to hang up. then asterisk will crash. any ideas? also i looked in the log file. this is what before it crashed: [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found

hi folks. recently some of our customers complained about bad voice quality on the phone system. i looked at the logs and found a lot of these: [2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:45] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:54] NOTICE[11305]

hi. could somebody explain how exactly the following parameters in zapata.conf work: pridialplan prilocaldialplan internationalprefix nationalprefix localprefix privateprefix unknownprefix the wiki & comments doesn't quite explain them. and phone companies are absolutely no help. i've setup systems in the US & China with trial & error until it works. now i'm setting up a

I'm running Asterisk 1.4.18 and having a problem with the clearglobalvars option. I have a NIGHT_SERVICE variable which I initially set equal to off. I then have an extension they can dial which will toggle that variable. My problem is when you enter the CLI and type "reload", it resets to "off" again. I've tried setting the clearglobalvars=no as well as just