similar to: Asterisk Disconnecting after 30sec when someone leaving VM

Hello All, I am trying to set up faxing using Asterisk@home 0.6. I have followed the instructions to the best of my knowledge. When a fax comes in, the system seems to detect OK but does ot manage to make the fax to pdf to email leap. Here is what I saw in the CLI when I tested. Any help would be appreciated. Thanks! Wiley -- Starting simple switch on 'Zap/2-1' -- Executing

DISA is currently disconnecting when I dial 8888 to access DISA. Below is my extensions.conf file from A@H and some lines which shows the disconnect. Should DISA be loaded as a module in modules.conf? When I do a 'show applications' i see that DISA is there. Help! -------------------------------------- ;Asterisk CLI as I placed a call from cell into the system. Playing

I'm setting up a new AAH 1.0 box to replace my AAH 0.6 box. Though on the new box, I've installed a generic ebay X100P. I don't have my livevoip or voicepulse accounts set up yet on the new box (can both boxes be registered at the same time?). I've set up one IP phone (SPA841) with the new box. I have my SBC POTS line plugged into the fxo card. I set up everything in AMP.

When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear F@510P) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power

I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version 1.400 and I am simply trying to configure into the "Extensions.conf" script an entry that will add to the "Auto-Attendant" a line that will allow a "Caller" to enter a "0" (Zero) will then ring the extension(s) of the "Operator" to speak directly with the "OPERATOR"

Docs? Polycom has docs? Where would one find this fabled land of... err I mean Polycom does stating what ftp servers are supported? Doug. -----Original Message----- From: Ken D'Ambrosio [mailto:ken@jots.org] Sent: Tuesday, March 07, 2006 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different business. I know that for a DID the routing is simple but I'm not seeing where I can match up a DID with a Zap channel. I'm currently looking into the zapata.conf file

Hello everyone, Please forgive the exclamation points but I have been battling this one off and on for about four days now. Sorry for the cross post. It all started with a box of IP 501s. I contacted my reseller and obtained the latest BootRom and SIP firmware. Unzipped, configured, copied over to my FTP server (running AstLinux, of course). The phone booted, so far so good. Updated

Hi listers, I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's. I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office. However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price. I would appreciate any input people have to offer. Thanks, James

Hi everybody, Hope your are not all on holyday because I've got a problem that is going to drive me crazy... I would like to remove some rows from a dataframe. The rows correspond to some specific indexes which I can get by looking at the name in the first column of my dataset. But I manage to get only the opposite of what I really want (function #1) #Function#1:

This is probably a stupid question.. How do I login to voicemail from the PSTN? I can dial *98 from within the system, but when dialing from the PSTN I have it set up to ring a dial group, then to an extensions vmail. During the extensions vmail prompts, I dial *98 and it sends me to the directory. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Nitesh, check out my config that I have for the Faktortel config in the asterisk@home sourceforge forum, you'll probably be able to work out how to set it up from there. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Wednesday, February 23, 2005 4:12 PM To:

Trouble in Parking Paradise! Good Day all, I have a situation that I have tracked as far as I can take it and am looking for assistance into the matter. My setup.. Asterisk 1.0.1 with the AMP config environment. When I have auto attendant answer the phone and I dial my extension "2204" The call come through and If I park it it gets place into extension 701 (the first parking

Hi, I'm trying to get capiECT working. I'd like to transfer call to another ISDN router connected extension and free channel from router to Asterisk. I get incoming call on CAPI and would liek to transfer it to dialed local extension - 400 in this case: [outbound-capi-local] exten => _4XX,1,NoOp(Transferring to local PBX ISDN number ${EXTEN} on msn CAPI/${CALLERIDNUM}) exten =>

I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP

Hi, I would like to create data.frame and increment their name to finally merge them. Does anybody has a solution ? Something like that ? n<-6 for (i in 1:n) { m[i]<-data.frame(name,value) m<-merge(m[i]:m[n]) } Nolwenn ******************************************** Nolwenn Le Meur INSERM U533 Facult? de m?decine 1, rue Gaston Veil 44035 Nantes Cedex 1 France Tel:

HI, I would like to know if it's possible to create a postscript file with multiple graphs . I'm creating some graphs by the means of a loop and I want to save each graph in the same file (splitting making the device by the number of graphs). Do I have to use the par(matrix()) option, the layout() function or the split.screen() one ? Thanks Nolwenn

Hi, ? I run the program testcall with one E1, it works fine; I receive DNIS and ANI for making calls and answering calls. ? When I start the Asterisk I receive call from outside correctly including DNIS and ANI, and receive the following messages: ? Sep? 7 10:29:59 WARNING[12167]: Answer Call Sep? 7 10:29:59 WARNING[12167]: MFC/R2 UniCall/2 Call control(5) Sep? 7 10:29:59 WARNING[12167]: MFC/R2

I installed the QuadBri card in my * server. I'm installing * on a RedHat 9 server I run the install.sh file. So far no problems. If I try to start /sbin/ztcfg -v -c /etc/zaptel.conf I will see the following error: Zaptel Configuration ====================== SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 3: CCS/ AMI Build-out:

Dear, Our iax clients's ip and port in the database were removed automatically, after 30 secs. the iax info is saved in odbc and postgresql . asterisk=# select * from iax_buddies where username='9706015'; name | username | type | secret | md5secret | dbsecret | transfer | inkeys | outkeys | auth | accountcode | amaflags | callerid | context | defaultip | host | language