Re: [asterisk-users] Custom PHP for Call Files

1. The 'Total' line that is displayed with the 'ls -l' command output is NOT the total number of files, it is the total number of system blocks used by the files in the directory. 2. In order to truly understand the situation you need to understand the Linux file system permissions.,,, Every file

Re: [asterisk-users] Custom PHP for Call Files

On Mon, Dec 28, 2015 at 7:42 AM, Eherr wrote: > Thanks Dale! > > I will try your method in the one web directory. > > Also while I was waiting for a response, I decided to start from scratch > in a different web directory and rewrite my code for a database based >

Re: [asterisk-users] Help With Physical Layer

On Tue, Jun 30, 2015 at 3:34 AM, Tony Kasule timotsm...@gmail.com wrote: Hello, Anyone to help me with this issue? It has never worked :( On Wed, May 20, 2015 at 11:34 AM, Tony Kasule timotsm...@gmail.com wrote: Hello users, I have a Digium Te235 and asterisk 13 which have worked well

Re: [asterisk-users] Asterisk 13 logging to two places

The line you commented out was writing errors to /var/log/asterisk/messages The problem you are having is the logging to /var/log/messages via syslog. It appears that Asterisk is sending verbose logging out to syslog even though logger.conf does not have syslog configured. I am not sure why

Re: [asterisk-users] Asterisk 13 logging to two places

On Fri, Jun 26, 2015 at 2:13 PM, Steve Edwards asterisk@sedwards.com wrote: Please don't top-post. Sorry, Gmail got me there. On Fri, 26 Jun 2015, Dale Noll wrote: I turned on the messages that he had in the file again, all the logs were in /var/log/asterisk and it does not show

Re: [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]

I use a Perl script that monitors AMI events. It also checks the state of all queues and members and generates some basic HTML pages for monitoring the queues. It's not perfect, nor would I call it pretty, but it gets the job done. If you are interested, I can send it to you. Dale On Wed, Mar

Re: [asterisk-users] 4 Port PRI

When parsing the config file, all the current settings are applied when the 'channel = ' directive is encountered. So something like this will make the three remaining groups and set signalling on ports 1 3 as pri_cpe and ports 2 4 as pri_net. ; setting specific to Group 2 group=2

Re: [asterisk-users] broken pipe question

Jerry, asterisk_execute() event_list=0 ret=55 last_command='Action: Logoff' Response: Error[CR ][LF ]Message: Missing action in request[CR ][LF ][CR ][LF ] Shows an error. Perhaps the originate did not complete before the logoff was sent? I have never used original in AMI so it is just a

Re: [asterisk-users] broken pipe question

On Tue, Dec 16, 2014 at 1:04 PM, Jerry Geis ge...@pagestation.com wrote: I am running a heartbeat... Asterisk 11.15.0 - same behaviour is noticed on 1.4.43 also I issue a call through the API that does the below. just UserEvent and Hangup -- Executing [s@heartbeat:1]

Re: [asterisk-users] ISDN Cause Code 47 Errors

Thanks Richard and Andres. I had come to the same conclusion, however the provider was fairly snarky in saying is was my equipment. We were able to replace the Cisco 2800 with a Cisco 2900 series and the problem appears to have been resolved. Thanks again, I always appreciate another set of

[asterisk-users] ISDN Cause Code 47 Errors

We fairly recently switched service providers for our 4 PRI circuits. Since that time, we started to notice some failed inbound calls. These calls terminate with an ISDN cause code 47 'resource unavailble'. Most of the time I see this error on the first or second channel on the second span in a

Re: [asterisk-users] Testing 911 call

If there is a non-emergency number you can call and let them know you would like to do some test calls. This also allows you to schedule a time for testing when the PSAP is not as busy allowing for real calls to be handled. On Sun, May 5, 2013 at 11:15 AM, Mark Engelhardt

Re: [asterisk-users] Asterisk QSIG doesnt send the calling name to Nortel CS1000

giorno giovedì 2 maggio 2013, Dale Noll dn...@wi.rr.com ha scritto: I retired my Nortel switch a couple of years ago, but I do not believe I ever got Asterisk - Nortel to pass the CPND, just the number. If I remember correctly, I had to enter then names manually in Nortel (LD 95?) for display

Re: [asterisk-users] Asterisk QSIG doesnt send the calling name to Nortel CS1000

It is not something that Nortel ever really talked about. I had used their tool Meridian Admin Tool (MAT) which used an Ethernet connection to my switches to sent commands over the network. It did not take much to figure out that they we using an rlogin connection(tcpdump and wire shark are your

Re: [asterisk-users] Asterisk QSIG doesnt send the calling name to Nortel CS1000

I retired my Nortel switch a couple of years ago, but I do not believe I ever got Asterisk - Nortel to pass the CPND, just the number. If I remember correctly, I had to enter then names manually in Nortel (LD 95?) for display on the Nortel endpoints. On Tue, Apr 30, 2013 at 11:30 AM, Danilo

Re: [asterisk-users] ODBC dialplan looping problem

A select that returns 0 rows does not always mean error. If you make your SQL do a little more specific lookup, you can avoid the loop entirely. Instead of your: readsql=SELECT pin from users WHERE confid='${SQL_ESC(${CONF_ID})}' Something like: readsql=select if(count(value)=0,0,1) from users

Re: [asterisk-users] date - outgoing call

On Mon, Dec 10, 2012 at 11:02 PM, Joseph syscon...@gmail.com wrote: On 12/10/12 20:45, Steve Edwards wrote: On Mon, 10 Dec 2012, Joseph wrote: When a call comes in asterisk records the date correctly but when I cake a call out I get only something like: Date: 60 here is an example:

Re: [asterisk-users] IAX Trunk issue.

On 06/24/2012 07:53 PM, Mitchell Johnson wrote: I'm testing a few IAX trunk scenarios in a controlled lab. From server2 extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across the IAX trunk to server 1 (IP address 172.16.200.210). Instead of ringing the 6001 phone,

[asterisk-users] talkoff problem - relaxDTMF is off

About a month ago, we switched our PRIs from being run through a Nortel Meridan system to an Asterisk based PSTN gateway using a TE210P card. Since the cut over I have been getting reports of DTMF tones being heard by my internal users when on calls to/from the PSTN. I have confirmed via

[asterisk-users] SIP endpoints CANCEL when PRI receives Cause Code 31

We have an Asterisk server which connects to another Asterisk server acting as a PSTN gateway. This gateway machine has Digium TE210P card connected to a pair of PRIs. For the most part, all is working well, however there are some specific telephone numbers that my users have attempted to

Re: [asterisk-users] SIP endpoints CANCEL when PRI receives Cause Code 31

On 05/23/2012 02:59 PM, Richard Mudgett wrote: We have an Asterisk server which connects to another Asterisk server acting as a PSTN gateway. This gateway machine has Digium TE210P card connected to a pair of PRIs. For the most part, all is working well, however there are some specific

Re: [asterisk-users] Best practices to route calls according holidays

On 05/18/2012 07:57 AM, Olivier wrote: Hi, At the moment, I'm mostly using a Day/Night toggle button to let users deal with week-ends, holidays and opening hours. As Asterisk 1.8 introduces Calendar capabilities, I'm wondering if better alternatives now exist. Is it possible, safe, reliable

Re: [asterisk-users] Asterisk forward call

How about adding in a time check so that during certain hours Asterisk waits 8 seconds before answering, otherwise it answers right away. You could also setup a status variable within the AstDB to indicate immediate answer or delayed answer. Just some thoughts. Dale On 05/15/2012 05:40 PM,

Re: [asterisk-users] Custom Application recording problem

On 04/16/2012 04:09 PM, Billy Kaye wrote: Thanks Dale, Am not sure why it was working in 1.4 but for some reason it was ( Note : My Asterisk is running bundled with Elastix). But any your suggestion worked very fine. Glad to hear it. Now am having one problem how can define those

Re: [asterisk-users] Custom Application recording problem

Billy, I really should have had my coffee before answering you previous message. My head was in the wrong place (not saying where) and I sent you down the wrong path. Macro() is not the answer because of the WaitExten(). When WaitExten is used in a Macro(), it does not match within the

Re: [asterisk-users] Custom Application recording problem

On 04/16/2012 08:36 AM, Billy Kaye wrote: In my 1.4 asterisk I have a custom application that users call and make recordings which recording I save to a file with the caller Id. Below is the config file which works perfectly in 1.4 I am not going to say that your application doesn't work

Re: [asterisk-users] Phone Inventory

On 02/23/2012 08:49 AM, Danny Nicholas wrote: Here is a snippet that somebody smarter than I am can improve upon for a in `asterisk -rx sip show peers|cut -f1 -d/` ;do asterisk -rx sip show peer $a;done|grep Useragent for a in `asterisk -rx sip show peers|cut -f1 -d/` ;do asterisk -rx sip show

Re: [asterisk-users] [NAT] SSH vs. OpenVPN?

On 01/31/2012 06:57 AM, Gilles wrote: Thanks for the infos. So the only way to use SIP through locked-down NAT routers is to use OpenVPN, either with the few hardphones that support it or with a softphone on a computer. You can also setup OpenVPN to connect a remote subnet (remote office)

Re: [asterisk-users] Real T1 trunk group...

On 01/17/2012 05:16 AM, Louis Carreiro wrote: Well... It worked! I would have wrote back sooner but I got swamped with trying to move a business department to the new facility. Great! Glad to hear it is working for you :) I think my problem was that I fat fingered (actually, overlooked)

Re: [asterisk-users] Real T1 trunk group...

On 01/16/2012 04:48 AM, Louis Carreiro wrote: I've been banging my head against the wall for a while (almost 18 hours today alone) with this one... I migrated our incomming T1's from the Option 11 to our Asterisk box this morning. We have 1 local T1 and 2 long distance T1's. The local T1

Re: [asterisk-users] asterisk - AGI (perl) - sqlplus(oracle)

On 01/06/2012 10:30 AM, Ron Bergin wrote: Add a BEGIN {...} block prior to the use statements and in there redirect STDERR to a file. This will aloow you to capture compilation errors You should also add some debugging statements at key points in the script. Then run the script and review

Re: [asterisk-users] hwo to stok variable wiith menu

On 11/30/2011 11:13 AM, salaheddine elharit wrote: i have last question regarding this thread with exten = 3,n,MYSQL(Query resultid ${connid} insert into test ( option_name ) values ('${CALLERID(num)}')) i can store the phone number without issue i need also the date and hour fo call in the

Re: [asterisk-users] hwo to stok variable wiith menu

On 11/28/2011 08:24 AM, salaheddine elharit wrote: thank you for your help You are welcome. i would to ask you please, i want to store the phone number of the customer in the option_name column when he press 3 in context menu i have created a database aheevacss with user aheevaccs and

Re: [asterisk-users] hwo to stok variable wiith menu

On 11/25/2011 10:48 AM, salaheddine elharit wrote: thanks for your response i use mysql like a database and my question when the customer press 3 in context menu i want to stok this variable in a table in my database and i want to get this variable after could you please give an exemple like

Re: [asterisk-users] check if devices reachable in queue

On 11/21/2011 09:16 PM, Matt Hamilton wrote: Have you tried, instead of pre-processing the caller before calling Queue(), checking the ${QUEUESTATUS} variable. Even when the phones are UNREACHABLE, QUEUE is still

Re: [asterisk-users] check if devices reachable in queue

On 11/22/2011 08:59 AM, Matt Hamilton wrote: Thanks Dale, you pointed me in the right direction. Happy to be of assistance. I hope your project goes well. -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] check if devices reachable in queue

On 11/20/2011 02:49 PM, Matt Hamilton wrote: 2. if the devices/members in the queue are not reachable, I would like to forward him to a phone B. I'm looking for a fast/practical way of accomplishing the second one. In other words, before sending a call to a queue, I would like to see if

Re: [asterisk-users] Goto Queue, does not work, it should play message or any thing

On 11/15/2011 04:56 AM, bilal ghayyad wrote: Hi All; When the call coming via the E1 dahdi and I handle the call (as first step) by exten = 5631040,1,Goto(OrangeCMG,s,1) it is not working and the call will be disconnected instead of queued. But, when I handle the call (as first step) by

Re: [asterisk-users] DAHDI has broken by pbx :-(

On 11/11/2011 07:53 AM, Russell Brown wrote: Well... the other classic 'fix' seems to have worked. Power cycling everything and rebooting and the problem's gone away. That's a bit of a Curate's Egg as I don't know what the cause was but it gets me out of the mire for now. Many thanks for

Re: [asterisk-users] DAHDI has broken by pbx :-(

On 11/10/2011 12:33 PM, Russell Brown wrote: Yes it was to /etc/asterisk/chan_dahdi.conf (oh how I wish the problem was being caused by something that simple!). It was worth a shot. I have wasted time looking for solutions that were that simple. I am curious about the D-Channel not being

Re: [asterisk-users] Binding asterisk to two static IPs

On 10/12/2011 06:15 PM, ge...@riseup.net wrote: I solved it by having two physical connections to my network. Yes, I thought of this too. I used the second nic for the drbd-communication, but I think I will have to change this. If your networking equipment supports VLAN, you could add a

Re: [asterisk-users] Binding asterisk to two static IPs

Don't you think the problem will still occur that the answers from asterisk seem to come from the main address assigned to the NIC? Or isn't this possible because of the vlan? Thanks, Georg It should work. As far as the OS and routing tables are concerned, eth0 and eth0.42 are different

Re: [asterisk-users] Binding asterisk to two static IPs

On 10/13/2011 10:53 AM, ge...@riseup.net wrote: Just tested this, doesn't work. Asterisk ist still replying using the main-address associated to the NIC. In a previous posting, Jim Lucas proposed... --- snip --- I solved it by having two physical connections to my network. PBX E0 IP

Re: [asterisk-users] Queuing strategy

On 10/11/2011 04:24 AM, bilal ghayyad wrote: Dear all; I have three agents and I need the calls to be always send for agent1 and if he is busy then to be sent for agent2 and if he is busy then to be sent for agent3 and if all busy then to stay in the waiting until one of those three agents

Re: [asterisk-users] RESEND: Mixmonitor command parameter problem on Asterisk 1.8.4

I am not real familiar with the size of MixMonitor parameters, but just looking at the output, I would suggest you change the logic to call a script with a single argument. something like this, MixMonitor(${FILENAME},bW(2),/usr/local/bin/convert_to_mp3 ^{FILENAME}) ---

Re: [asterisk-users] using variables in the shell function

On 09/13/2011 07:49 PM, Israel Gottlieb wrote: is it possible to pas variables to the shell function Set(recordingavail=${SHELL(ls /var/lib/asterisk/sounds/custom/${TOPMENU})}) im trying to see if a file is available before playing the file or does anybody have a different idea but not using

Re: [asterisk-users] Inbound routes

This is how I have done it. In FreePBX, under the 'Setup' tab, choose 'Zap Channel DIDs'. Assign a DID (ex. 12345) to the channel(port). Then go to 'Inbound Routes' and create a route for the DID and set the destination to the appropriate extension. On 01/21/2011 08:21 AM, Vitor Carlos

Re: [asterisk-users] OpenVPN + SIP configuration?

I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance,

Re: [asterisk-users] nortel meridian question

Jerry Geis wrote: Hi all, I have asterisk 1.4.26 (and I tried 1.4.29) connected PRI all 23 lines and for the most part everything works. Dialing out on 23 lines to phones works fine. I have to use the Local channel to call the intercom system (from call files). If I only call 1

Re: [asterisk-users] question on nortel CS 1000 PBX and PRI connection to built in PA system

Hi, Is the Paging on the Nortel connected to a trunk port or is there some device connected as a physical extension that answers the call? Hopefully you have access to the nortel box. I am going to assume that you are not very familiar with the nortel command interface. If I am incorrect, I

Re: [asterisk-users] nortel cs 1000 swtich

Jerry Geis wrote: Anyone successully connected to nortel cs 1000 switch? Care to share you switch settings? I have asterisk 1.4.25, libpri 1.4.7, dahdi We tried national and the verizon guy said that wasnt working... We tried 5ess and we can get external calls - but internal calls we have no

Re: [asterisk-users] Generic question about PBX PRI installs

Jerry Geis wrote: The PBX guy seems to always complain about how he has MANY options and thats not enough information... What else am I supposed to supply this person. Are they not the PBX expert?... Anyway as example. the last customer I told the above information. He set up the PBX and

Re: [asterisk-users] Remote connection to an Asterisk server

Gary wrote: On the router, I've turned on DMZ to point to my Asterisk box's static IP address. My home (real world) IP address is static. The Problem: When I grab one of my Cisco 7940's and take it to my office, it does not see or register with my home Asterisk server after I change