similar to: NOT able to call on local extensions while successfully call on external mobile no.(using SONETEL account)

Hi I need help This is the error: Really destroying SIP dialog 'NDMxOWRmYTRhMWVkMGFhMjllMzU4YmNmNjQwN2NlM2Y.' Method: SUBSCRIBE -- Executing [00919885497796 at internal:1] Set("SIP/1001-0000000b", "CALLERID(num)=8790771141") in new stack -- Executing [00919885497796 at internal:2] Dial("SIP/1001-0000000b", "SIP/00919885497796 at sonetel")

Prof Nash, Prof Galanos Is it possible to use a generic code stub in front of packages that use optimx to improve optimx use or curtail it according to the requirements? Best Regards Amit +91 7899381263 ________________________________________________________________________ Please request Skype as available 5th Year FPM (Ph.D.) in Finance and Accounting Area Indian Institute

Hello, I'm Shivang Bansal, a 3rd year Computer Science Engineering undergraduate at Institute of Engineering & Technology in Lucknow, India. This mail is an expression of my interest for Google Summer of Code program of this year. I want to apologize for getting in so late. Actually I would have contacted earlier, but sudden demise of my Grandfather disabled me in doing so. I am

Hi, i still got a little Problem (Thunderbird cant copy my mail to "Gesendet") but besides that i can send and receive mail. ive used: http://wiki.dovecot.org/HowTo/SimpleVirtualInstall there it says that /etc/dovecot/passwd should look like: test:{PLAIN}pass bill:{PLAIN}secret timo at example.com:{PLAIN}hello123 dave at example.com:{PLAIN}world234 joe at elsewhere.org:{PLAIN}whee

While using Ubuntu 10.10 + Dovecot 1.2.12 + Postfix 2.7.1-1: To enable virtual accounts, I am using the following /etc/dovecot/auth.d/virtualsomename.auth file: passdb passwd-file { args = /etc/dovecot/passwd } userdb static { args = uid=vmail gid=vmail home=/home/vmail/%u } EOT cat /etc/dovecot/passwd looks like this: test:{PLAIN}pass bill:{PLAIN}secret timo at

Dear list; I'm trying for forward some calls to an others asterisk using IAX2 protocol. But My asterisk can forward some calls and for others it hangs up automaticaly. Before my asterisk was working perfectly, i do not know what is happening!! When i try directly zoiper with my provider's asterisk it works perfectly. Here is the output from the cli when i made a call that asterisk hangs

Hello list, i need help. My problem is that when I want to call (using ISDN phone or internal SIP client) via the Sip provider a real phone number (ISDN phone or internal SIP Asterisk >> SIP ), I get a ring tone. When I now decide to hang up (e.g. if nobody answers), the called telephone continues to ring almost forever. the sip.conf: [2563105] accountcode = 2563105 username =

Hi all, I have this weird problem. I'm running asterisk 1.0.3 on Debian Sid (official debian package). We have 2 fritz ISDN cards. All is working great. Till I called the bank. It rings one time and then gives me the congestion tone. Here is what I see on the CLI (phone nr obfuscated for privacy reasons): -- Executing Dial("SCCP/michiel-00000004",

hi there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help

I've noticed unstable behavior of nlminb on some Linux systems. The problem can be reproduced by compiling R-3.5.2 using gcc-8.2 and running the following snippet: f <- function(x) sum( log(diff(x)^2+.01) + (x[1]-1)^2 ) opt <- nlminb(rep(0, 10), f, lower=-1, upper=3) xhat <- rep(1, 10) abs( opt$objective - f(xhat) ) < 1e-4 ## Must be TRUE The example works perfectly when

Hello list, i need help. My problem is that when I want to call (using ISDN phone or internal SIP client) via the Sip provider a real phone number (ISDN phone or internal SIP Asterisk >> SIP ), I get a ring tone. When I now decide to hang up (e.g. if nobody answers), the called telephone continues to ring almost forever. the sip.conf: [2563105] accountcode = 2563105 username =

Greetings everyone, I'm having some issues connecting a Asterisk box to a old ISDN PBX. Everything works fine but the undetermined digits rules. For instance, if I have _00X. and I start dialing for instance 0035..., Asterisk just get the 4 first numbers and starts dialing 0035. I've tried adding: stop_tone_after_first_digit = yes append_digits2exten = yes Although I

Hi, I'm setting up dovecot pop3 deamon for a small set of mailbox. I want to be able to set a small set of mailbox simply, so I've used "passdb passwd-file" + virtual mailboxes with "userdb static". My problem is that it seems that postfix accept and create mailbox for every address I post. It is configured to use local delivery through dovecot's deliver. How

Hi All, For a 1.4 version asterisk, whats the recommended mechanism for dialling with ENUM lookup? At the moment I user SIPbroker, but am getting tired of it hanging on certain numbers, so I was thinking about implementing it myself. I've seen various vo-ip.info pages (http://www.voip-info.org/wiki/view/Asterisk+cmd+EnumLookup) talking about the func ENUMLOOKUP instead of EnumLookup

On 2017-04-18 08:17 PM, Pete Mundy wrote: >> On 19/04/2017, at 7:58 am, D'Arcy Cain <darcy at VybeNetworks.com >> <mailto:darcy at VybeNetworks.com>> wrote: >> >> <snip> >> Everything looks the same as another one that works except for two >> things. The one that works doesn't have the "Probation passed" lines. >> I am

Try this for CHAN_SIP: same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the mailbox same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we have a mailbox defined log into it If you are using PJSIP it's more complex same => n,Set(EndPoint=${CHANNEL(endpoint)}) ; Get the peer same =>

Hello, I'm trying to create a H.323 trunk between Asterisk 11 and Avaya. I have done this before between Asterisk 1.6 and Avaya but had some issues placing external calls from the Asterisk to the Public network which is connected to Avaya. I'm trying to create that trunk on Asterisk 11 because the 1.6 is outdated and has no support. On the Asterisk side I have Aastra 6731i SIP phones

hello, I have been using asterisk now for about 2 years now on a RH8.0 it is our main call gateway. I have on the box 3 T1 TDM cards connected to 2 Rhino channel banks (FXS) and 1 CAC Access bank I (FXO) with so many softphones and ATA 186s. It has been working good till today some few hours ago. i just discovered that there were no dialtone on the phones. Asterisk did not spit out any error, it

Asterisk 11.1.0 One queue with strategy=leastrecent. (Full queues.conf below.) Occasionally (several times today), a caller will get "stuck" in the queue - there are operators available to take the call, but the caller stays in the queue for a long time. Any idea what might cause this, or where I can start looking to debug it? I'm going to start digging through the queue log

Hello, dear R-ers! I have a data frame: x<-data.frame(a=c(4,2,4,1,3,4),b=c(1,3,4,1,5,0),c=c(NA,2,5,3,4,NA),d=rep(NA,6),e=rep(NA,6)) x When x$a==1, I would like to replace NAs in columns d and e with 8 and 9, respectively When x$a != 1, I would like to replace NAs in columns d and e 101 and 1022, respectively. However, I only want to do it for rows 2:5 - while ignoring what's happening in