similar to: connecting Asterisk to NEC NEAX system

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, How can I make calls from Asterisk client to NEC NEAX 2400 traditional phone ? Is it possible to have a connection between Asterisk and NEC NEAX 2400, since NEC-NEAX2400 is an IP-PBX and supports SIP. Please help me to find a solution ;;; Thanks & Regards Ram Kumar Customer Support Engineer Barcode Gulf LLC Dubai , UAE Mobile : + 971 50 5594178 Email : Ramkumar@barcodegulf.net

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

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

Is there anyone out there who has tried to connect up an asterisk box to make and take calls through a NEC NEAX 2400 using Q.sig or anything like it? Can anyone tell me if it is possible? Thanks in advance, Phil Allred Brooklyn Law School

Greetings all, This is my first post to this forum, so please be kind. I am looking to integrate Asterisk as our primary voice mail server (IVS) for our NEC NEAX 2000 IVS. We currently use a 6-port Mitel system. The two communicate via analog TDM ports and a MCI serial connection. Is anyone familiar with setting up a NEAX 2000 with Asterisk as an IVR server? I know I would need six FXO and

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

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

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 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

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 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. 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]

Paste your extensions.conf section that is relevant. -Matthew ----- Original Message ----- From: "Greg - Cirelle Enterprises" <gcirino@cirelle.com> To: <asterisk-dev@lists.digium.com> Sent: Monday, December 27, 2004 4:32 PM Subject: [Asterisk-Dev] realtime voicemail > Let me clarify my last message. > > If I put in the wrong password I get polled > again for

i've been playing with 1.6 voicemail w/ IMAP storage. it seems to work fine. however once IMAP storage is enabled. everyone VM will use IMAP. is there a way to configure some users use IMAP and other users use traditional file base storage? -- 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'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. 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. 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:

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