similar to: testing simultaneous calls

I am posting here since you guys are my last hope. I am trying to configure a Polycom Soundpoint IP 335 with MWI. Is there any way to eliminate the scrolling messages and Msgs softkey? I am trying to get it where it's just the light that indicates the new messages. I don't know if Asterisk has to send a different notification or what have you. Thanks, --Eric -------------- next

Hi Richard, Thank you for the reply Correct, I did mean 13.15. I set no optimize and better backtrace through "make menuselect" and the output is now [Aug 28 21:41:16] ERROR[17171][C-0000392d]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x21962b0 (0) Got 26 backtrace records #0: [0x61923f] main/utils.c:2475

Hello, I've recently setup a small load test against an instance of Asterisks. I've tested on asterisk 13.5 and 14.6 with the same results. I am using PJSIP. My dial plan is, [test] exten => 1001,1,Answer exten => 1001,n,MusicOnHold(15) exten => 1001,n,Hangup I am using SIPP to test. I can share XML if desired but it simply waits on the line while music plays for 8

I'd like to be able to save the choices made in menuselect in a way that they can be tracked in a CM system and applied to a later release of Asterisk using an automated tool like Ansible. What's the best way to do that?

Hello asterisk-users, We've been having intermittent issues with chan_sip - it stops responding to cli requests, trying to reload chan_sip from cli doesn't seem to have any effect, initiated calls carry on for a short period, but no new SIP requests are processed ('sip show channels' hangs forever, server stops responding to SIP OPTIONS, or any other SIP messages). We have updated

Hello. I have a dataset with about 500,000 observations, most of which are not unique. The first 10 observations look like 901000000000100000010100101011002 901101101110100000010100101011002 901000000000100000010100000001002 901000000000100000010101001011002 901000000000100000010101010011002 901000000000100000010100110101002 901000000000100000010100101011002 900000000000100000010010101011002

Hi All; I am looking for a good Outbound Dialer and to be practical with possibility to do modification on it, the outbound dialer should send the calls to the agent when the agent is logged in as long the agent is belong to the queue (or let us say the skill group of this campaign). Any one can guide me? If I can build this using the AMI, so I appreciate if anyone did it before me so I can use

http://sipp.sourceforge.net/ Anyone care to throw this at Asterisk to see what happens? I would, but I am having significant temporal shortfalls recently due to the apparent warping of the space/time continuum when I answer the phone with clients/associates. It seems that entire days pass by before I hang up... very odd, and very counter-productive to getting good Asterisk work done. JT

I trying to compile R-.1.7.1 under OS X. Configure worked fine; then make crashes with: /sw/lib/libg2c.a(err.o) definition of common _f__formatted (size 4) /sw/lib/libg2c.a(err.o) definition of common _f__hiwater (size 4) /sw/lib/libg2c.a(err.o) definition of common _f__putn (size 4) /sw/lib/libg2c.a(err.o) definition of common _f__reading (size 4) /sw/lib/libg2c.a(err.o) definition of common

The Asterisk Development Team would like to announce the release of Asterisk 16.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.1.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hi, I'm having a bit of a problem with storing voicemail messages in an odbc database. I *think* I've got everything configured correctly but messages are stored on the asterisk server instread of in the database. System info 64 bit redhat RHEL 5.1 Asterisk 1.4.26 unixODBC installed used makemenuselect to instal res_odbc and cdr_odbc Back end database DB2 Database name voiceml

Hey all, I tried to do some searching but I found snippets and I am having trouble putting it all together. I want to have an option off the IVR that plays back the announcement for the day. At the end of the message, I want the caller to get kicked back to the previous menu. The conditions are that I want the recorder to dial a feature code that prompts him to record the message. He

Hi All, I have installed newt and newt_devel but dahdi_tool will not compile/install. I'm trying this with dahdi-linux-complete-2.5.0.2+2.5.0.2. does anyone have any suggestions as to what I am doing wrong? Regards David. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, As an exercice, I installed sipp on the same box as a Asterisk 11.4 instance (to keep network equipements out of the equation). I'm focusing on the maximum number of new calls this Asterisk instance can deal with. Here is the dialplan (AEL) I'm playing with: _X. => { Verbose(0,Incoming call from ${CALLERID(num)} to ${EXTEN} in ${CONTEXT} - case A);

Hi, I keep getting the following on my slug no matter what version I try (1.2, 1.4, 1.6, 1.6svn): ... make[2]: Leaving directory `/usr/portage/distfiles/svn/trunk/menuselect/mxml' gcc -o menuselect menuselect.o strcompat.o menuselect_stub.o mxml/libmxml.a menuselect.o: file not recognized: File format not recognized collect2: ld returned 1 exit status distcc[19221] ERROR: compile (null) on

Hi List; I am facing a problem relaed to menuselect when I am trying to compile zaptel -1.4.2.1, the error as following: [root at localhost zaptel-1.4.2.1]# make linux26 make[1]: Entering directory `/usr/src/asterisk-1.4.4/zaptel-1.4.2.1/menuselect' make[2]: Entering directory `/usr/src/asterisk-1.4.4/zaptel-1.4.2.1/menuselect' make[3]: Entering directory

Hi All, I'm in the process of writing an install script and I would like to change some settings for the install process but I don't want the user to go into menuselect and make the changes manually. Is there a way to make the changes to menuselect from the CLI? As an example, selecting the iLBC codec. menuselect codec ilbc on Regards David. -------------- next part -------------- An

Hi all, I thought I'd share I script I made (based on some of Leif's works) that lets you download, compile and install Asterisk all in one go; and then removed the dev tools used. We use it quite a bit to provision systems using Ansible, but it is easier than remembering everything every time even if you are using a shell. At the moment I have scripts for Centos 7 and Asterisk 13, but

Please forgive me if this has been answered in another post. I''ve looked around but couldn''t find a solution. I bought the book "Agile Web Development with Rails" and think I''m going to really like it. I went through the first chapter with no problems. However, as I begin developing the database on page 57, I get a consitent error: Before updating scaffolding

hi all i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0 any suggestions ? make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' make[1]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... [CC]