similar to: Incompatible destination (88) Error Message

I have connected asterisk 1.0.7 with Avaya Definity via E1 with a TE100P Digium Card. Inbound calls are working perfectly and I dont have any problem. But when I try to make an outgoing call with my softphone (xlite) I am getting the following messages. Hungup 'Zap/13-1' Executing Dial("SIP/IZ-bc0a", "Zap/g1/3118") in new stack Called g1/3118 Channel 0/1, span 1 got

summary: I can successfully ncvar_put(...) data to a file, but when I try to ncvar_get(...) the same data, I get > Error in if (nc$var[[li]]$hasAddOffset) addOffset = nc$var[[li]]$addOffset else addOffset = 0 : > argument is of length zero How to fix or debug? details: R code @ https://github.com/TomRoche/GEIA_to_NetCDF successfully (if crudely) uses R packages={ncdf4, maps, fields}

hello everyone, I have a trixbox server with an E1 card(not digium).It connects to an AVAYA pbx use E1. It works fine.But when i change the E1 card to digium TE220B,there is a problem. When sip extension A(on trixbox) call PSTN extension B(on avaya),A must wait longtime before B start to ring.From the log I find there are two times call. I don't know why the first request be rejected

ok ok I have set up a public directory under /home/public on a linux machine configured as a Samba Server. The configuration of the /etc/smb.conf file is like this: [public] comment = Public directory path = /home/public browseable = Yes writeable = Yes I can see the directory from a w98 client and can open/run the files. But I do

Hello, I have one question regarding *. Default configuration for asterisk is to keep configuration(s) in ordinary text based config files. My question is simple: is it possible to keep those config info (at least, to start from - sip.conf, extensions.conf and voicemail.conf) on a database, which asterisk access via ODBC. If it is possible, I'd appreciate if someone points me where I can

Hi group, anyone can explain me the exact difference between pri value in zapata.conf ? ; PRI Dialplan: Only RARELY used for PRI. ; ; unknown: Unknown ; private: Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN If I use it, I also must use prilocaldialplan = local ? Thanks Giordano -------------- next

Hello! I have a problem with integrate() in my function nctspa(). Integrate produces an error message "evaluation of function gave a result of wrong length". I don't know what that means. Could anyone suggest me what is wrong with my function? These are the examples of function calls that work OK: nctspa(a=1:10,n=5) nctspa(a=1:10, n=5, mu=2, theta=3, renorm=0) This does not work:

Hello! I have a problem with integrate() in my function nctspa(). Integrate produces an error message "evaluation of function gave a result of wrong length". I don't know what that means. Could anyone suggest me what is wrong with my function? These are the examples of function calls that work OK: nctspa(a=1:10,n=5) nctspa(a=1:10, n=5, mu=2, theta=3, renorm=0) This does not work:

Yes, I can make a call on that extension from other definity phone, if you mean it. -----Original Message----- From: Ken Godee [mailto:ken@perfect-image.com] Sent: 30 ÉÀÌÑ 2004 Ç. 19:14 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing Roman Bessyadovskii wrote: > Hi All. > > I connect

HI >i have a problem please help me to solve it: http://r.789695.n4.nabble.com/file/n4639434/aj.pdf aj.pdf >i want to calculate the vecteur a[j] where j: 1...8 >this is the code in R: >aj.fun <- function(j, i, X, z, E, beta0, beta1){ + n <- length(X) + iX <- order(X) + iz <- order(z) + e1 <- -(beta)*z[ iz[1:(i - 1)] ] + numer <- E[j] - sum( X[ iX[1:(i - 1)] ]

Hello, I'm using Asterisk stable (1.0.3) with Asterisk-oh323 (0.6.5). Everything is working fine, well, except that : when a call is made from an h323 device (gnomemeeting for example), the caller does not hear any ringing at all, he suddenly hears the person who answers the phone. That can be quite disturbing for the users. Any help would be very welcome. thank you. Yves

Hello all, I recently noticed the realTime effort and must say it is a nice idea! I would appreciate any help to get it running .. I downloaded the code & patches and succefully patched my asterisk (CVS-HEAD-11/29/04-12). - created a DB called asterisk, and a table sip using the schema supplied at http://bugs.digium.com/bug_view_page.php?bug_id=0002613. - entered an entry: insert into

Hello! Is it possible to connect Siemens HiPath 4000 to Asterisk? What equipment required on Siemens side? I mean IP not E1. Sorry for asking here. Siemens-related websites use "salesperson language". There is no technical information.

I am hoping someone could shed some light and point me in the right direction? I'm attempting to get callerid work between an Avaya Definity PBX and Asterisk via TE110P connected via T1/PRI Crossover PRI. From the Definity side I've searched endlessly and came with an example which we modeled as close as we can, but still no luck. While doing PRI intense debug span 1 in I see a couple

Only a warning, and doesn't seem to do anything bad. But I can't seem to figure out what these warnings mean? -- Requested transfer capability: 0x00 - SPEECH [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: k [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: o [Feb 17

I'm testing an ivr implementation (first time) using: exten => 620,1,Wait,1 exten => 620,2,Answer exten => 620,3,DigitTimeout,5 exten => 620,4,ResponseTimeout,10 exten => 620,5,Background(npi-greeting) ; "Thanks for calling press 1 for" exten => 1,1,Goto(npi-directory,s,1) For initial testing, I've arbitrarily mapped this onto ext 620 (will change that later

I am always fascinating by good programming techniques. R contains a lot of very good examples I have been learning from. Since I am using some functions from package "wmtsa", I though I could borrow the elegant example, in the documentation of function "wavBestBasis" to compute the entropy from the Wavelet Transform coefficients. In the following I have pasted the excerpt,

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Hi, I'm facing a strange problem. My setup is: Alice cellphone <--GSM--><--ISDN--> Asterisk <-- ISDN --><--GSM--> Bob cellphone When Alice calls Asterisk which forwards the incoming call to Bob, sometimes Bob sees Alice's number, sometimes he sees a default CallerID (which happens to match the dialed number and the ANI). For various reasons, Bob really needs to

Hello Julien, Am Mon, 14 Jan 2019 22:15:47 +0100 schrieb Julien dupont <marcelvierzon at gmail.com>: > ** Test 1 ** > On VPN_office I use 'tcpdump -npi any icmp and host 192.168.1.3' > When pinging 192.168.1.1 from client 1, with no success, I see no packet > passing. Sorry - the tcpdump command should end with "192.168.1.1" instead of