similar to: Call forwarding...

I am trying to implement a follow-me script (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. From my main menu, I want the extension 300 to execute the script as follows: exten => 300,1,dial(sip/200,20) exten => 300,2,playback(pls-wait-connect-call) exten =>

Greetings, I have implemented the following command to allow CNAM to be delivered to my users. exten => 9969,1,Wait(1) This works great! However it has spawned a new problem. When this is implemented into a full dial plan. If a user is set to DND or sends a call to Voicemail by hitting deny the caller gets a busy. Below is a result of the calls. With the Wait(1) statement -- Executing

hello all, i have already configure sip.conf and dialplan. i done the follow me script. first problem: i want to call(with kphone) someone at my extension, i must dial the extension number. i can't dial their username. 20531603@192.168.8.125 (work) mustafa@192.168.8.125 (call fail) is it possible to do that?? second problem: if i want to call another number (not my

I have the following scenario: PSTN gateway (202.180.nnn.nnn) -> OpenSER 1.0.1 (147.202.nnn.nnn) -> Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, often (but not always) Asterisk repeatedly sends a SIP 200 OK message and eventually hangs up the call. sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

I have a remote installation that connects via IAX from my office pbx. When I call an extension on the remote pbx, after the dial period, the call is terminated. Nothing I do in configuration of that extension seems to matter: -- Executing NoOp("IAX2/netconcepts@nnn.nnn.nnn.nnn:4569-5", ""Dial 710"") in new stack -- Executing

I recently upgradeded dovecot on one of our servers from version 1.0.10 to version 1.1.3. Ever since, we've been seeing occasional errors similar to this sequence (with the username and IP addresses elided): Sep 30 00:09:56 alcor dovecot: pop3-login: Login: [4954], XXXX, NNN.NNN.NN.NNN Sep 30 00:09:56 alcor dovecot: wrapper[5006]: pop3, XXXX, NNN.NNN.NN.NNN Sep 30 00:09:56 alcor

I've attempted to change my smb password on a remote NT PDS, but it always fails with resolve_name: Attempting lmhosts lookup for name SERVER<0x20> getlmhostsent: lmhost entry: 127.0.0.1 localhost resolve_name: Attempting host lookup for name SERVER<0x20> Connecting to nnn.nnn.nnn.nnn at port 139 error connecting to nnn.nnn.nnn.nnn:139 (Connection refused) unable to connect to

Possible Mediatrix 1204 fxo sip gateway workaround Need some feedback from experienced * users relative to this workaround please please please. Problem: The mediatrix 4-port fxo gateway does not provide any mechanism for * to select which "port" an outbound pstn call will use. (See lots of previous posts over the past four days for more detail if needed.) Our reseller has been

Hi Asterisk-Users, We have a problem with queues.conf / extensions.conf queues.conf file reads like ... member => SIP/8399 extensions.conf reads like ... exten => 8399, 1, SetCIDNum(${AccountNumber}|a) exten => 8399, 2, Dial(SIP/8399,10,Ttrf) When somebody calls to the queue, we observed that it is not going through extensions.conf (previous two lines) That mean's it is not

hi. I am trying to get print working and have a hard time indeed... My goal is to replace our nt4 print server with a linux/samba one as we have problems serving printers and drivers to win2000 workstations (and I don't want to set up a win2000 server for printing, and have the same problem _again_ when switching to XP ;-) The situation so far : - Debian woody - Samba 2.2.4-1 - Winbind

I have a client host that I don't have access to now, which attempts to establish ssh connection back to my BSD server using the private key. Client runs this command: /usr/bin/ssh -i ~/.ssh/my_key_rsa -o "ExitOnForwardFailure yes" -p $HPORT $HUSER@$HOST -R $LPORT:localhost:$LPORT -N On the server debug log looks like this: Connection from NNN.NNN.NNN.NNN port 43567 debug1: HPN

Dear List, I'm trying to get diff/removed data and it's offset out. So I write a functions in receive_data. When I run backup, I found there is a weird behavior which I don't understand. i = recv_token(f_in, &data) will receive (i = -1, offset2 = 0) some where in the middle of the transfer procedure. That's to say, it's going to transfer the first data block from sender,

Hello List. I am using rsync to pull html files from a shared drive to 2 web server boxes to keep the files synchronized. However, I have a few files I do not want rsync to copy over because they contain information specific (IP address) to the box hosting them. Rsync seems to intermittently ignore the exclude statement and copies them from time to time. Is there a way to absolutely prevent

Hi! I'm running 1.1rc4 on a system and this happens occasionally: --8<-- mail.info; dovecot: auth(default): client in: AUTH 1 PLAIN service=imap lip=NN.NN.NN.NN rip=NNN.NN.NNN.NN lport=143 mail.info; dovecot: auth-worker(default): pam(XXXXXXXXXXXX,NNN.NN.NNN.NN): lookup service=imap kern.alert; kernel: grsec: From NN.NN.NN.NN: denied resource overstep by requesting

I tried to get callerid working the normal way but the cid is never passed to the phone. It doesn't work untill I set SetCIDNum(0${PRI_NETWORK_CID}) in extensions.conf which I found in the wiki: http://www.voip-info.org/tiki-print.php?page=Asterisk+zaphfc Is this intended behaviour, or still a bug? It does work but it only shows one zero even though I have nationalprefix = 0

>> The call that gets dropped had a retransmission of INVITE from UAC >> to UAS (and therefore retransmission of 200 OK from UAS to UAC). >> There is nothing wrong with the re-transmission as such, but I >> noticed a potential bug in Asterisk in the way it responds to an >> INVITE retransmission. Asterisk is bumping up the session version >> number in

Can someone recommend a method to answer the following type of question: Suppose I have a coin with a probability hhh of coming up heads (and 1-hhh of coming up tails) I plan on flipping the coin nnn times (for example, nnn = 500) What is the expected probability or frequency of a run of rrr heads* during the nnn=500 coin flips? Moreover, I would probably (excuse the pun) want the answer for a

Hi, I'm currently using chan_capi-cm-0.6, with the following capi.conf: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 language=de [ISDN1] msn=8304490 incomingmsn=8304490 isdnmode=msn group=1 controller=1 softdtmf=1 context=demo echosquelch=1 echocancel=yes echotail=64 callgroup=1 devices=2 Each user has a different numer, e.g. 83044910, 83044911, 83044912 and so

Hello there, I have a client connecting to dovecot and comunicating like this: * OK Dovecot ready. * AUTHENTICATE MYAUTH + base64 challange base64_response {NNN} * NO Invalid base64 data in continued response The problem is that dovecot seems to reject the response because of the {NNN} in the end of the string. If I remove the {NNN} it authenticates just fine. Any idea of whether it's

Hello, I have a little problem with signalling. An E100p is connected to an Alcatel PBX, wich has an E1 to the outside. Located in Germany. zapata.conf: switchtype=euroisdn pridialplan=local prilocaldialplan=local overlapdial=yes signalling=pri_cpe .... With asterisk 1.0.2 I can call from a SIP phone to a phone connected to the Alcatel and the SIP number is correctly displayed at the caller.