similar to: Problem with Music on hold

I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for testing. In addition I register a zoiper SIP soft phone. For the Grandstream I have busylevel=1 in sip.conf. If I place a call from the GXP280 to zoiper and then put that call on hold from the zoiper side and then call GXP280's extension, asterisk indicates the phone is ringing. As the GXP280 is a single line phone it

I sent this email a few days ago but did not see any responses to it: > I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for > testing. In addition I register a zoiper SIP soft phone. > > For the Grandstream I have busylevel=1 in sip.conf. > > If I place a call from the GXP280 to zoiper and then put that call on hold > from the zoiper side and then

I installed version 1.6.0.3-rc1 and my AMI application stopped working. I reinstalled 1.6.0.1 and it worked again. I reinstalled 1.6.0.3-rc1 and it stopped. Looks like a problem in the software to me. Following the same steps using the same code for the AMI and conf files for * I get bad behavior in 1.6.0.3-rc1 and good behavior in 1.6.0.1. I have this action: Action: Originate Channel:

Hi list, I'm trying to create an asterisk 1.8 rpm with SRTP. I found mention of a libsrtp rpm, <http://qutecom.ipex.cz/RPMS/srtp-1.4.4-1.i386.rpm > in these instructions, <http://www.voip-info.org/wiki/view/Asterisk+SRTP> but it is unreachable (by me, anyway). The libSRTP source is here, <http://srtp.sourceforge.net/download.html>. Has this already been packaged for

please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compile&run clients with srtp (linksys,grandstream,aastra, qutecom, eyebeam, ...) digium need feedback for srtp inclusion to 1.6.3.0 http://bugs.digium.com/view.php?id=5413 if you need additional info, i'm on jabber - cervajs at njs.netlab.cz thanks! --------------------------------------- Marek Cervenka

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, all! I'm being based on this document [1] to send and to receive calls using ekiga.net. But I'm seeing, in an Asterisk console, several messages of this type: [Feb 17 21:19:15] NOTICE[11875]: chan_sip.c:7715 sip_reg_timeout: -- Registration for 'danib2 at ekiga.net' timed out, trying again (Attempt #4775) -- Got SIP

Hi, I saw this problem mentioned before but the user appeared to be using the MP3 software with asterisk. I am using the native music on hold player in asterisk 1.2 and I too have a volume problem with music on hold. Is this controllable through the 'indications.conf'? I know this file controls frequency range for various sounds might it also control sound level or am I barking up

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm testing with a Grandstream BT200 telephone and, according to I read, it has a LED that blinks if for that extension messages were left. In "Voice Mail UserID", under the "ACCOUNT" tab, I put *100 that is the extension in which my Asterisk answer the voicemail service and if then I press MESSAGE button, the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! This is my first message to the list. Some time ago I'm user of Debian GNU/Linux and recently I also started to use CentOS GNU/Linux. Does anyone know how to enable the mark facility on CentOS? According to what I was seeing, unlike Debian GNU/Linux, which uses rsyslog, CentOS still uses the traditional syslog where the parameters to be

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm making a text mode installation of CentOS 5.5 on a single disk on which I intend to put a software RAID-1 degraded, thinking of adding another disk later on. But if I try to make the installation in a RAID-1 with a single disk, the system does not allow it. There is any way to do it? Thanks in advance for your replies. Regards,

Hi there, I just tried today's CVS: 4/23/2004 version and found a strange loise with music on hold. Basically, when on hold you hear very distorted music as if it was very loud. This is the exact same problem described last year at: http://lists.digium.com/pipermail/asterisk-users/2003-April/009735.html http://lists.digium.com/pipermail/asterisk-users/2003-May/011688.html No answers on

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I had the opportunity to test a Sangoma A200 card and I have some doubts that I would like to consult: I tried to detect the card and I had no success using the wctdm module with DAHDI. I guess this is because electronics is different because the TDM400P and OpenVox A400P cards have separate modules for each channel, while the Sangoma A200

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm testing a telephone connected to FXS port of a Sangoma A200 card. But I'm observing that callerid is not working. The configuration that I'm using in chan_dahdi.conf is the following one: - --------------------------------------------------------------------- ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit

Hi guys, I'm trying to write hangup causes from asterisk into the CDR record. Using version 1.4.24.1 at the moment, but no joy so far. Has anyone implemented this? Neeraj Chand Support Analyst Fiji Islands Australia T: +6793342526 T: +61388924326 M:+6799344012 New Zealand www.ocis.com.au T: +649 980 7022 -----Original Message----- From: asterisk-users-bounces at

Hello! I've configured Music on Hold in asterisk, the only, most certainly, stupid problem I have is, which DTMFs to send to activate and deactivate it. If I use the cli, I can establish a call with originate. With the "misdn send digit" command I can send a number of digits to the other party. But what are the combinations to put the other one on hold? Or do I have to use a

Hi, what software do i need to convert an mp3 to a g729 format? I have a portal where a user can upload their own MP3, but when a user is using a g729 codec, the music on hold has a crackly sound. using g711 it's very clear. so what i'd like to do is when they upload an MP3 i will make a copy on g729 format, so that asterisk can choose which file to play depending on what codec is

Hello, I've a problem. I've asterisk 1.6.0.5 version. And I've created callcenter, but agents registers to another SIP server. When agent tries transfer a client to another operator , pressing flash, I get this: [Aug 18 16:06:37] WARNING[5259]: chan_sip.c:5349 sip_indicate: Don't know how to indicate condition 9 [Aug 18 16:06:37] WARNING[5259]: channel.c:2858 ast_indicate_data:

For some reason our music on hold is intermittently skipping... running Asterisk 1.6.1.22 anybody know what could be causing this? I don't think it's an encoding problem because it plays fine sometimes. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110502/9e69a5c9/attachment.htm>

I'm running Connected to Asterisk 1.4.18-1. We have a T1 PRI and a SIP trunk coming in from our IVR. When people call in on the SIP trunk, I experience the following problem: I put the caller on hold, MOH starts. MOH quits after approx ten seconds. When I pick the call back up off of hold, they can't hear me for about 20 seconds (sometimes longer or not at all) but I can hear them. This

Hi there, We're experiencing the following problem on a handful of boxes and I'm wondering if anyone has any pointers for me. First up, hardware configs on boxes are the same: Dell 860, Dual Core Xeon, 1Gb RAM, 2x 160Gb SATA's in Software RAID 1. Various Trixbox versions - 2.2.3 and 2.4.2 to name two of them. Problem is that occasionally the music on hold speeds right up -