similar to: Forward to my phones the domain of the CALLERID in incoming URI calls

Hi, I write the CDR of my Asterisk 1.2.17 server in MySQL database using cdr_addon_mysql.so. Now I'm trying to write some custom values to userfield column by the SET(CDR(USERFILED)=SOME_TEXT) sintax, but nothing gets writeen in MySQL cdr table!! Why? I'm I skeeping something or what? Taking a look at the URL:

Is it possible to force the Dial function to skip to the next priority if it doesn't find the server of the called contact within a few seconds? I know I can use: Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]) where I can use some short timeout in the "timeout" option, but if I do so, when some call is well succeeded, it will only ring for that

Hi, Can Asterisk write to multiple MySQL databases in different machines, at the same time, as a backup scheme? If it does, where can that be configured? In res_mysql.conf file? Does anyone ever made it? Regards, Ricardo. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi is it possible to set up in the dialplan (on in sip.conf, or something else) the hostname of the outgoing uri call? This is my scenario: - CCM integrated with Asterisk via h323 - SIP user registerd to Asterisk - Asterisk is behind NAT - Asterisk ip is 10.10.10.2 - SIP user view Asterisk as 10.10.15.2 (Asterisk is behind NAT) When the CCM calls the SIP user the call works perfectly. The

Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem

> From: Ricardo Carvalho <rjcarvalho@reit.up.pt> > Subject: [asterisk-users] How to match wild card inside a GoToIf? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > > How can I match wildcards inside a GoToIf? > > I have something like this, but it doesn't work: > > [default] > exten =>

Hi, I use Asterisk 1.2.17 in my site, and now I'm trying to configure ENUM lookup in my server. When someone calls a number that has multiple ENUM entries, randomly Asterisk seems to fail to return a correct answer, and dial by ENUM fails. I've goggled a bit on this, but didn't get any good conclusion. There is some RFC Compliant ENUM Macro that can be used that is announced

Dear all, I'm trying to configure Asterisk to be able to ask the caller to enter a given password in order to continue dialplan execution. I've tested this feature using the Authenticate application like this: exten => _X./5219,1,Answer exten => _X./5219,2,Authenticate(1234,a) exten => _X./5219,3,Playback(pin-number-accepted) exten => _X./5219,4,Dial(SIP/${EXTEN},120)

Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the available patch found at URL http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3 that is announced to support it too. With both Asterisk versions, I've sent with success FAXes between two FAX machines each one attached to an ATA interface, both registered in

Hi all, Does ENUMLOOKUP can query multiple DNS servers without having to replicate the same code in which the only thing replaced is the server? If I use ENUMLOOKUP(${exten}), Asterisk will parse enum.conf file to find the list of DNS servers in order of preference to be queried, but, I pretend to use something like this: ${ENUMLOOKUP(+${ARG1:2:},sip,c) which does not seam to care about

I would like to setup my Asterisk server to process an incoming SIP URI and redirect all requests to a specific context. Example: (1) using a sip phone I'd like to be able to call: sip:somedomain.com *or* sip:someone@somedomain.com (2) i'd like my asterisk server to answer the call and route it to the context=in-from-sipclient which would play thru some DP actions Can anyone give

Hello, I'd like to implement some public sip uri's that poeple can call into and get an echo test. Is there a way to force a codec so that users can test various codecs? Something like: echo-test at example.com (negotiates whatever codec, is there a way to figure out what codec was negotiated and tell the user) echo-test-g711 at example.com (forces g711) echo-test-g729 at

Hello, I am having a problem with the lmtp delivery of mail addressed to multiple recipients. ?The first recipient receives the mail correctly but the next recipients do not. ?ltmp apparently tries to create a link from the subsequent recipients' mail store to the first recipient mail store and fails. This returns 451 smtp codes to the sender which then resends at some later time. Please see

I have two users on my network, Mary and Bob, who work together in a shared share. They both belong to the group Accounting. Bob is a savvy Linux user who accesses the share via NFS4. Mary toils away using Windows accessing the share via the Samba server. Mary will create a directory on the share and dump a number of files in which Bob and Mary will split the load. Bob, being a LInux user, will

If we get an incoming call I can edit the callerID provided to add the leading '90' and set the name so that sales calls can be identified according to the number called. If however the callerID is unavailable then setting the callerID name or number fails (it shows as unavailable on the phone). This is the call log from such an incoming call without callerID. == Spawn extension (voip,

Hi, Since we're moving from a legacy PABX that has been serving one of our customers for more than 15 years, we'd like this process to require no "human habits" change among the users. Software: Asterisk 1.4.22 Hardware: Polycom phones (mainly 430/601) Here are the "problems": We did configure call groups, pickup groups, ... - When someone picks up a call from

Hallo, I have got an existing data frame and want to add a new column. The existing data frame was created like this: > df <- rbind( c("Fred", "Mary", 4), c("Fred", "Mary", 7), + c("Fred", "Mary", 9), c("Barney", "Liz", 3), + c("Barney", "Liz", 5) ) > df

Hi I have an organism directory that contains two folders galGal3 and hg19 and many other files. orgDir = '/home/mary/org' When I try to use list.dir() function, it gives me the same answer, no matter what is the value of full.names argument. > list.dirs(path = indexDir, full.names = FALSE)[1] "/home/mary/org" [2] "/home/mary/org/galGal3" [3]

Dear R Helpers, I am missing something very elementary here, and I don't seem to get it from the help pages of the ave, seq and seq_along functions, so I wonder if you could offer a quick help. To use an example from an earlier post on this list, I have a dataframe of this kind: dat = data.frame(name = rep(c("Mary", "Sam", "John"), c(3,2,4))) dat$freq =

Here is the situation that I would like to create: Call comes in Receptionist sees that the caller ID is Jenny <8675309> Receptionist picks up phone and transfers call to Batman Batman looks at his phone and sees that the caller ID is Jenny <8675309> I can't seem to figure out how to forward the caller ID. Is this possible with Asterisk?