similar to: Dial strings with ; characters

I'm having problems in dialing numbers over SIP that include characters from the UK international phone number conventions. I have my contacts in Outlook, with the numbers represented as: +<countrycode> (<area code>) <numberpart> <numberpart> eg: +44 (20) 7834 1234 or: +1 (801) 555 1234 I'm using the SJphone softphone, doing my testing through the Stanaphone

Due to the format of the message coming from the H323 channels included w/ Asterisk we were unable to use our gatekeeper. For a quick solution we tried the OH323 channel drivers and can receive inbound calls from the parent gatekeeper. We are trying to do a dial to gatekeeper... I am trying exten => 5551212,1,Wait,2 exten => 5551212,2,Dial,OH323/5551212 But I am not sure if this is the

Hello, I've begun working on an AGI in PERL to provide the ability to dial a string, similar to composing text messages on cell phones. The current idea is that it would allow you to use a telephone to enter a string of text, (number 2 can be used for letters a, b, c, number 3 for d, e, f, Etc), and one could review the text thus far, backspace over characters, and "commit"

Can someone please explain how to match a + character in a dial plan (so that I can swap it for the "00" country escape code). In Europe at least the + is a common shortcut for the international prefix (which is "00" in my country). However, my trunk chokes on the + character and all my speed-dials are setup with a + at the start of them... Trying to fix the phone rather

Hello Michael, Here is the BackTrace of the program which i forgot to attach BACKTRACE OF Asterisk -vvc #0 0x42074d60 in _int_realloc () from /lib/tls/libc.so.6 #1 0x420738c4 in realloc () from /lib/tls/libc.so.6 #2 0x47c7da89 in PAbstractArray::SetSize(int) () from /home/sip/pwlib/lib/libpt_linux_x86_r.so.1.5 #3 0x47c7cf4d in PContainer::SetMinSize(int) () from

Michael, Thanks a bunch, I downloaded from inaccessnetworks.com thinking that it is the latest :). Ok I will upgrade it. just for the record, following worked. exten => _87.,1,Dial(OH323/H323:${EXTEN:1}@16.52.153.206) Cheers Sathya Date: Fri, 28 Nov 2003 11:28:59 +0200 From: Michael Manousos <manousos@inaccessnetworks.com> Organization: inAccess Networks To:

Hi, I am trying to consume a web service that returns a JSON string, however, when I run > fromJSON(s) Error: '\/' is an unrecognized escape in character string starting "http:\/" because the string s contains a single backslash, which is the JSON escape character, but R expects 2 backslashes, I think. Is there anyway to preprocess the JSON string returned from the web

In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd like to do something possibly unique with the formatting of extensions in my dial plan, and am having trouble. We use VoicePulse connect, which gives us local DID for inbound and outbound calls (even though DTMF tones are not working in Voice Pulse Connect at the moment). To dial local numbers, you have to

Hello everyone, I am reading a HTML table from a website with readHTMLTable() from the XML package: > library(XML) > moose = readHTMLTable("http://www.decisionmoose.com/Moosistory.html", header=FALSE, skip.rows=c(1,2), trim=TRUE)[[1]] > moose V1 V2 V3 1 07.02.2010 SWITCH to Long Bonds\n (BTTRX)

Hello R-users, I have code that gives me the important variables from an analysis. I need to input these variables into a different analysis. To do this, I need to modify them slightly... 1) remove all numbers at the end of the variables, 2) remove all periods. I tried to do it with the awkward code below. It works to remove all the numbers, but when I try to remove the period everything

Is there a Pause or Wait character for dial tone command for the Dial command? Like in a modem Dial string? I'm having issues with the FXO Analog line not returning dial tone quick enough for * to recognize. Arick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040905/a4a1388f/attachment.htm

Hi Paddy! This used to be 80 characters total (including all characters like channel type, '&' and '/'. Had the same issue in the past where I extended that in the code and recompiled. From what I understand there is basically no longer a hard limit in Dial since the recent change in the latest versions other than a single device must not exceed this but you can concatenate

There seems to be a problem in print.default with some ISO-latin1 characters (the chars AFTER ASCII in western Europe...) if they appear in strings. (no problem if they are part of a function comment, see below). Some of the characters lead to 4 character Hex-codes being printed instead: "û" ## ^u prints as "0xFB" If you use the funny characters in comments of functions,

Dunno why your phone isn't allowing you do negotiate g711u but I can tell you how to upgrade the firmware. I called them on Thursday for myself and they gave me the following tftp server address for which to program my phone. 4.3.153.50 Load this into your phone's tftp area and reboot it. It'll go out to the net and check the firmware revision and change it if required. I've done

On systems even key systems it is customary to have an 'internal' dial tone. Since Asterisk simply ignores the 9 and keeps the tone going it is hard to tell for some 'new users' if they can make a call. My first idea was to change the generated dial tone via source. Then if the user presses 9 go to a different context where I would record about 30 seconds of the normal dial

exten => s,1,Answer() exten => s,n,Background(astcc-please-enter-your) exten => s,n,Background(zip-code) exten => s,n,WaitExten(5) exten => s,n,Read(NUMBER,,5) exten => s,n,SayDigits(${NUMBER}) exten => 22042,n,Dial(SIP/sipvendor/11111111111,120,A(ginger3)) exten => 22601,n,Dial(SIP/sipvendor/11111111111,120,A(ginger3)) ; xxxxx/ winchester exten =>

Hi all, I've just connected a cellsocket to my * server so that I can keep call costs low for outgoing calls to mobile phones. I'm using an old fashioned 5110 cellsocket (because I have a few of these phones and I think that they are very reliable). For those that don't know the cellsockets they convert a mobile phone into a POTS line. The cellsocket has a funny dialtone but that

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-4ff7' -- Executing

Hello all, This new version (0.6.1) of asterisk-oh323 fixes the "one-way audio" problem of the previous release. Download from the usual location: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael.