similar to: Cut leading digit?

In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset <--> (FXS)SPA3000(FXO) <--> PAP2 And set the Line1 dialplan on the SPA3k to '(<:@gw0>S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with

freebsd quite current ports tree 1.01 asterisk spa3k at 2.0.11(GWg) for calls in from the pstn side of an spa3k to asterisk, i get sporadic short beeps. they are not related to sip re-reg time, which is all that has occurred to me so far. calls in from the fxs side of the spa3k and out through nufone do not exhibit the beeps. calls from the fxs side of the spa3k out the fxo side do have the

Has anyone ever gotten * to work on commercial unixes such as HP-UX, Solaris, AIX? What about other architectures than x86? -- . -----BEGIN GEEK CODE BLOCK----- Version: 3.1 GCS/GIT> d-@ s+:+ a? C+++ BLHIS$ U+++ P+> L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+> h---- r+++ y++++ ------END GEEK CODE BLOCK------ . -------------- next part

Arrggghh. Tearing my hair out here. I'm trying to set up the spa3000 in the UK for my home, and want * to control the dial plan I've googled to no avail. I've read the manual to no avail. Can someone, please let me know what the parameters is the spa and * are to a) receive a call from the pstn b) make a call to the pstn from the phone attached I can make sip to sip calls (i.e. I

This is usually a very active list, but looking at my procmail log the last message I have received arrived on: >From asterisk-users-bounces@lists.digium.com Fri Jul 29 03:04:17 2005 Subject: Re: [Asterisk-Users] How can I use MySQL in the dialplan? Since that message there has been a gaping silence, any idea what is up, as I am sure seeing mail from everything else. Actually I

Hi all, I tried to set the calleridname of an incoming call to get different incoming labels displayed for different incoming numbers. This does work for hidden number-calls so I can set the displayed CIDName on my cisco7960 from "CID withheld" to "abc CID withheld" If the incoming CID isn't hidden it works to use SetCallerID but not to change only the CIDName with

Running a fairly recent subversion release of Asterisk, I'm running into a problem using labels (as opposed to priorities) with this application. Here is the dialplan segment: ; isolate gotoiftime bug with labels ;exten => 806,1,GotoIfTime(8:00-20:00,*,*,*?2:4) exten => 806,1,GotoIfTime(8:00-20:00,*,*,*?light:dark) exten => 806,n(light),noop(light) exten => 806,n,hangup exten

at my work we have a meridian 1. it has 6 lines from the outside world coming in to it. if someone calls our number 870-238-2111 they will always reach our resepsionist unless all 6 lines are in use, if all 6 are in use then the caller will get a busy signal. on our phone bill they are called "trunk lines". we have a seperate incoming line for our dsl. are these analog lines ?? to

Hello all, I have an ISDN termination box (TR1) that converts ISDN(Bri) to 2 normal analogue lines. The same number is assigned to these lines. These lines are connected to 2 spa3k registered to my asterisk box. When calls arrive, TR1 try to pass call to the first spa. If spa not takes the call immediately then try to pass to the other spa. The only configuration I found works is to put the

Asterisk has been working fine for me for several weeks using MGCP to a Adit600 for intra office calling. I have recently loaded h323 and the following errors occurs when starting asterisk. [chan_h323.so]Aug 10 09:09:18 WARNING[7824]: loader.c:258 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Aug 10 09:09:18 WARNING[7824]:

Hello, Thought people might be interested in this. I want my voicemails emailed to a person and not stored on my asterisk server. However, I want them to have a sequential number. I found that if I set the option delete=1 in my voicemail.conf file for the mailbox, then the numbering would keep being restarted. I wrote this shell script to fool Asterisk into numbering my voicemails sequentially

My SIP phones can dial out through Sipura SPA3k to POTS for local and 911 calls _but_ incoming POTS calls are being swallowup somehow. Am I on the right track with the code snippit below? sip.conf: --------- In sip.conf the following code is _supposed_ to ring the SIP phones when a POTS line call comes in through Sipuara to Asterisk. [spa3k-pstn-in] ; Pots-line-in from Sipura ; If

Hi, After reading this valuable forum and the voip-info wiki and follow all the steps , but my Cisco 12SP+ remains unregistered. These are my config files: skinny.conf [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 172.20.1.1 ; Address to bind to dateFormat = D-M-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 languaje=es allow = all ; disallow

Two asterisk boxes 150 miles apart, both cvs-head as of this morning (and since Sept 27th), connected via iax2 with low-utilized ds3 internet, C7960 calls exten on remote system (also C7960), and call goes to VM. No other calls in either system (eg, no load). Both boxes have iax config'ed as: trunk=yes allow=ilbc jitterbuffer=yes Recorded VM messages are very distorted. Changing only

Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. I see some that state they do but I also see reviews that say they don't. Doug

Hi, I am very new to Asterisk. I wanted to know how to retrive the Voicemails. I could see some voicemails assosiated with some extensions. Any ideas?? --------------------------------- How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos -------------- next part -------------- An HTML attachment was scrubbed... URL:

My apologies for the off-topic post ... No matter what settings I try, when I dial in to the SPA-3000 on the PSTN line, it picks up the call and immediately gives me a fast busy tone then hangs up. The info tab says under PSTN Line status: Last PSTN Disconnect Reason: PSTN Disconnect Tone which seems to indicate that the SPA thinks the caller has hung up. Since I am in Japan, it is possible

I tend to make it pause for 10 secs when loading the module as I have had a few occurances of loading before /dev/zap has been populated. Wouldn't trust the 2.6.12 kernel as far as I could virutally throw it. Has anyone had any problems with PCI-X systems? In particular call dropping? Regards Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Can anybody show me a working Sipura 3000 setup please? I need to setup one to my * box, ... What are the variants you can setup? Advantage - disadvantage. bye Ronald Wiplinger

Voxilla.com has a great config wizard for the SPA-3000 and * http://voxilla.com/spa3kasterisk.php I took the output from this wizard and dumped it on my test box with an SPA 3000 (with some mods to match my * contexts) and everything worked great. Calls from the PSTN to the spa3000 are routed to dialplan #8 on the spa3000, which dials * Both the FXO and FXS port register with * The SPA3000 is