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Commit-ID: d1596f8b90d41f5382d6550c03826bed0ce82ffa Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=d1596f8b90d41f5382d6550c03826bed0ce82ffa Author: Herbert Xu <herbert at gondor.apana.org.au> AuthorDate: Mon, 26 Mar 2018 18:33:49 +0800 Committer: Ben Hutchings <ben at decadent.org.uk> CommitDate: Fri, 25 Jan 2019 02:57:21 +0000 [klibc] redir: Fix typo in

Commit-ID: 4265f8d559e294cc39afce8cc6849341db751b0b Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=4265f8d559e294cc39afce8cc6849341db751b0b Author: Herbert Xu <herbert at gondor.apana.org.au> AuthorDate: Mon, 26 Mar 2018 18:33:49 +0800 Committer: Ben Hutchings <ben at decadent.org.uk> CommitDate: Sat, 28 Mar 2020 21:42:54 +0000 [klibc] dash: redir: Fix typo

There are times when I need to call a client from my cell and I want a recording of the call. I was trying to put into an * a way of doing that. Below is what I'm using in my extensions.conf exten=> X,1,Read(num,"/var/lib/asterisk/sounds/mtas/10digit",10,,,5) exten=> X,2,SayDigits(${num}) exten=> X,3,Background(/var/lib/asterisk/sounds/mtas/verify) exten=>

I have the below script that doesn't seem to be working. I don't know if I have something in the script wrong that I am just missing. Or if I don't have the php.ini set correctly for emailing This is the CLI output -- Executing [4099XXXXXX at port3_real:1] Goto("DAHDI/50-1", "newhire,s,1") in new stack -- Goto (newhire,s,1) -- Executing [s at

Other than having stripping out IPs this is what I am receiving for my voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works fine calls that come in on my PRI. BUT at least from this VOIP source the To field which is my RDNIS information for these calls, doesn't actually fill into ${CALLERID(rdnis). But as you can see I'm getting the information. My question is, Does

I have three tables customers, indentities and people. I hav ecreated models for them. The are connected with foreign keys and has_many/belongs_to in this fashion: customers has_many: identities identities belongs_to: customer identities has_many: people people belongs_to: identities Now, I have a form in which I would like to create db entries for the three tables...

First off this is not my work for extensions.conf it is modified from http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl ogin-to-standard-dialplan-methods-part-1/ So credit to Leif Madsen <http://www.leifmadsen.com> But as to my question [AgentLogin] ;A replaced version of AgentCallbackLogin() using a GoSub() ; exten =>

Trying to configure IAX for use I think I have everything set right. But my IAX phone wont connect. When I run wireshark I'm seeing this Note if above screenshot from wireshark does not show here is a link for it: http://img402.imageshack.us/i/tempe.jpg/ I've tried a variety of setups in my IAX.conf (they all end up with the same issue, tried just bindaddr=0.0.0.0 with

It might be worth mentioning the voip call is coming from a number we have thru bandwidth.com in case anyone uses them. James Shigley From: James A. Shigley Sent: Wednesday, December 02, 2009 3:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Variable Name needed That wasn't it either. I tried a few other likely fields from

AGI is executable. =20 Then run 'agi debug' from the asterisk cli, place a call and see what was send and receive from your agi =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A. Shigley Sent: April-23-09 12:26 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] AGI PHP script =20 I have the

Alright I know how to do basic IVR in *. But what I'm working on trying to do now is a survey. I've found very little things out there on google or the archives for how to do surveys with the * ivr. Here is more or less what I'm trying to accomplish 1. Call comes in Plays Greeting 2. Starts Survey 3. Ask Q1, Record the answer (voice responses) repeat this

Alright I have a SIP phone located off premises with a very annoying issue. Well I say a sip phone it is actually two phones hooked to a Cisco Spa 2102 Link: http://www.cisco.com/en/US/products/ps10026/index.html Each phone being a different line/extension. Alright either line can ALWAYS make outbound calls no issue. The problem is on the Inbound side. I'm completely stumped as

is executable. Then run 'agi debug' from the asterisk cli, place a call and see what was send and receive from your agi From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A. Shigley Sent: April-23-09 12:26 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] AGI PHP script I have the below

Hello All - I've been poking around the past few weeks, trying to familiarize myself with all of this. I am new to Linux, VoIP and Asterisk -- to be complete. This is my first exposure to all of these technologies. I installed AsteriskNow on my old dual Pentium 833mhz Dell PowerEdge 2400 and the install went well. I can log in and poke around in Linux and I even configured the box to be

Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [root at asterisk asterisk]# cat

hello hpa, please pull for the dash update git pull git://brane.itp.tuwien.ac.at/~mattems/klibc.git maks with this changes: Alexey Gladkov (1): Check return code for getgroups and fwrite Herbert Xu (17): Remove unnecessary truncation in _STPUTC Always call conv_escape_str in echocmd Fix \c spillage across echo commands Release 0.5.3. Make eval with empty