similar to: spa-1001 not getting a dial tone on my pbx

hello my friend is trying to get his spa-1001 (sipura) 1001 connected to my asterisk box. he reset his spa-1001 to factory defaults I emailed him the voip-info page I found on google and yes I did look on google anyways he isn't able to get the thing to connect to it eg getting a dial tone, he did install x-lite and it worked fine with that am running asterisk@home 1.0 can some one please tell

hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email: hanksmith4@earthlink.net gmail: hanksmith5@gmail.com msn messenger: hanksmith4@earthlink.net aim: hanksmith5 skype: hanksmith5 -------------- next part -------------- An HTML attachment was scrubbed... URL:

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

Dear R users I have a list, as follows: > intvl.period.myrs $Devonian [1] 4.8 4.2 9.5 5.7 $Ordovician [1] 7.2 5.1 10.2 1.9 $Silurian [1] 4.7 3.0 7.8 2.0 3.3 1.6 2.6 2.7 I want to write a loop that will sum up the values in each part, and give me a vector containing the (in this case 3) summed values this is what I have so far: for (i in 1:length(names(intvl.periods.myrs)) {

Yo all, As there has been some intrest, here's my updated version: I post it to "-dev" as well as "-users", as it may be of intrest to both. Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate set of features. Currently, my implementation supports call- forward unconditional, on no answer

Dear all, After struggling for some time with *apply() and eva() without success, I decided to ask for help. I have 3 lists labeled with, each contains 3 different interpolation functions with identical names: > names(missgp0) [1] "spl.1mb" "spl.2mb" "spl.5mb" > > names(missgp1) [1] "spl.1mb" "spl.2mb" "spl.5mb" > >

The following transfer led to a crash of asterisk, without leaving a core or any utterances in messages or debug file. It looks like the zombie which was created during the MASQ-transfer was not cleaned up... But why did it start a Dial??? And... why does Asterisk die when this happens?? Thanks!!! Michiel -- Zap/32-1 answered Zap/6-1 -- Stopped music on hold on Zap/6-1 -- Starting

Hi All, I'm upgrading some PBX's from 1.2 to 1.4 and having a bit of trouble with converting the Realtime application to the REALTIME function. I have the method down and understand simplistically what is going on, at least enough to get my old 1.2 apps to run in 1.4 functions. I do not understand why change from the app to the func? What the benefits? To me, the app seemed so

Ia had a server overload today because someone did a call forward to their own extension. To do a call forward I write a key called CFWD with the extensi?n number and number to dial . The main script tests if the key/value exists and dials the number stored in the database. What is an easy way to prevent dumb people from creating a loop? -- Telecomunicaciones Abiertas de M?xico S.A. de

Hi all, Has anyone already written something which allows saving and loading the internal DB settings? All users CFWD and speeldial settings are stored in the DB in my setup which makes it a pain to restart Asterisk.... Looking at showtree in db.c (why isn't that exposed in the CLI?) It shouldn't be too difficult, but I don't want to reinvent the wheel. On the same track, I am also

Hi all, I've got a provisioning file that I use to configure Cisco SPA112's. I'm wanting to get this file to do 3 things for me. I want to turn T.38 on, Call forwarding off, and Call waiting, off for both lines. but it's not working. This is what I'm using to enable T.38 for line 1. <FAX_Enable_T38_1_>Yes</FAX_Enable_T38_1_>

hello, i have trouble with nat + sip outgoing call.when make an outgoing call to a sip gateway, i have no sound. i have 2 sip gateway, one is asterisk. asterisk is on public ip and private ip other sip gateway is on public ip phone are cisco and grandstream on private ip on the same subnet as asterisk. phone are connected by sip to asterisk (i have try with or without nat=yes) incoming call

I can't seem to find the download link for the windows serve can some one send me the windows server off list or direct me to a link where I can get it? thanks hank ----- Original Message ----- From: "Lorenzo Prince" <lorenzo@princenet.sytes.net> To: <hanksmith4@attbi.com> Sent: Thursday, May 15, 2003 2:33 PM Subject: Re: [vorbis] sgreaming in ogg format? (fwd)

----- Original Message ----- From: "hank" <hanksmith4@earthlink.net> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Tuesday, June 28, 2005 10:52 PM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? >I am using asterisk@home 1.0 > my mp3 is called > mp3 > it has nothing before it

http://chan-sccp.berlios.de/ 20050713 ftp://ftp.berlios.de/pub/chan-sccp/chan_sccp-20050713.tar.bz2 I didn't have a spare 7960 to use this week, so maybe some line issue is still present. - fixed a memory leak on database updates (dnd, cfwd*) - fixed old memory leak on unload (now unload chan_sccp.so and load chan_sccp.so work. It does reload the config when asterisk is running) - socket

Hi all! I am fitting a Poisson model, using the following command: > fit2<-glm(canc~id1+year1+time+lnpa,family=poisson) where 'id1', 'year1' and 'time' are factors. I defined them with: > id1<-C(factor(id1), treatment) and 'lnpa' is a continuous variable. The 'summary' function gives me all the effects estimates, that is, for id1, I

Hello, I'm configured Sipura-3000 to forward IP calls to PSTN number on no answer (In User1 tab Cfwd No Ans Dest: 123456@gw0) IPPhone ---IP---> Sipura-3000 ---PSTN---> PSTN User Generally it works fine, but Sipura sends back SIP OK to IPPhone just prior to dialing to PSTN number. How to configure Sipura to detect that the remote side on PSTN picks up the phone and only then to

> Ia had a server overload today because someone did a call forward > to their own extension. To do a call forward I write a key called CFWD > with the extensi?n number and number to dial . The main script tests if > the key/value exists and dials the number stored in the database. What > is an easy way to prevent dumb people from creating a loop? Right after you have

Could this possibly mean that any person who has CF set should never be available as CF Destination. Simple db entry/check can have this done. On Tue, Jun 2, 2015 at 5:34 PM, <dk at donkelly.biz> wrote: > > > > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *Kevin Larsen > *Sent:* Tuesday, June

On Tuesday 02 Jun 2015, Carlos Chavez wrote: > Ia had a server overload today because someone did a call forward > to their own extension. To do a call forward I write a key called CFWD > with the extensi?n number and number to dial . The main script tests if > the key/value exists and dials the number stored in the database. What > is an easy way to prevent dumb people from