similar to: Voicemail 2

Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated

Hi, How can I retrieve those voicemails using my ip phone? and how will i confiugre it on asterisk? Please help I'm very new in asterisk. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Do?a Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

Our smallest network of systems has only four computers connected via Gigabit Ethernet.? The oldest and most stable platform is an eight year old Dell E520 running CentOS 6.8.? We often try out applications on this Dell/CentOS machine before moving them to other systems on our other networks. Last night, one of our users decided to create a single, 228GB home directory tar archive on an empty,

Hi to All, I've an Asterisk CVS Head working with Mysql. My problem is that instead of ANSWERED or something like, into the CDR database records, I find only numbers. This is also a problem to let ASTPP works, infact I receive an error: ERROR - ERROR - ERROR - ERROR - ERROR DISPOSITION NOT MATCHED and the call has no cost. Any suggestions? Thanks -- .:FaberK:.

Hi, I'm using voipbuster at work, and I've got 2 questions: 1) Is it possible to send faxes using voipbuster connex? 2) Is it possible to cut off or cover the voice that say the charge per minute?(I've payed the '5' euro, and from that moment I've got it!). Of course I understand that is to let me know how much I'm going to spend, but I do not like it, expecially when

Hi, I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Thanks, Adrien -- Adrien Laurent - CIO www.modulis.ca 514-284-2020 ext 202 adrien@modulis.ca

I have seen *some* similar activity in different machines through the years and it *always* turns out to be a hardware issue. If this machine is particularly old, I would be suspicious of that. Linc Fessenden ________________________________________ From: CentOS [centos-bounces at centos.org] on behalf of m.roth at 5-cent.us [m.roth at 5-cent.us] Sent: Wednesday, December 07, 2016 1:51 PM To:

Hi folks, we use meetme application with pin so when a customer joins he's prompted for his name. Then the voice say:"press one to accept the recording..." My question is, is it possible to cut off that request to"press one"? Thanks to all -- .:FaberK:.

Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .:FaberK:.

when i send calls from an asterisk box to a voip provider the call fails and give me these messages: *CLI> Sep 23 21:32:32 WARNING[14595]: chan_sip.c:6890 handle_response: Forbidden - wrong password on authentication for INVITE to '"asterisk" <sip:asterisk@195.112.214.99:5070>;tag=as19e688a1' -- SIP/call-0f60 is circuit-busy == Everyone is busy/congested at this

Hi folks, I've just updated my * and I noticed that from the update the uniqueid field into mysql, is not written and ASTPP do not charge the calls. I got an eye to cdr_mysql.c and I found that at line 212, into one insert query, uniqueid is missing. But I can be wrong. In any case, somebody got same problem? Any suggestions? Thanks to all. -- .:FaberK:. -------------- next part

Dear All, I am new to R and am trying to learn how to create functions using R. Below is code which calculates Lin's Concordance Coefficient. After I calculate the coefficient I want to create a scatter plot which annotates the coefficient along with preceding text onto the plot. The below code doesn't seem to work. If I use only the object 'lincc' on the text command it works

Hi, I've a problem with a cisco 827-4v and asterisk (1.0.9) acting as sip-to-pstn GW. The issue is that when a call comes in from the pstn, asterisk correctly contacts the router, which in turns send a "183 Session progress". Obviously, asterisk thinks that the telephone is not ringing (because it expects a "180 Ringing") and we have no ringback on the pstn side. Putting a

Hello Guys, first of all, I'm very new with asterisk. I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7 Now I'm trying with asterisk-oh323 I've already installed pwlib, oh323 and I've already set the variables. Now, when I try to "make" asterisk-oh323 I receive this error messagge: for x in wrapper asterisk-driver; do make -C $x all ||

maybe there is a bug regarding the use of nis to mount the user's home directory at the login or my misconfiguration. After the CentOS 6.4 (64bit) installation I checked for the latest samba version on the official repository using yum: the latest version (that was already installed) is samba- 3.6.9-151.el6. >From "man smb.conf" I have seen that "nis homedir" is not yet

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

Hi all, I've got an Asterisk box with 1 Sangoma A102 and 1 Ericsson PBX. I need to use Asterisk as E1 line for the Ericsson PBX. How do I have to connect them? I'm trying to connect the Sangoma to the Ericsson, but RED alarms remain. Any suggestions? Thanks -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi to all, I've just started to see that Asterisk-gui from Digium. Does anybody know, when the first official-realese will be released? Thanks to all -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071012/09197260/attachment.htm

Hi guys, I'm using Zyxel Prestige 2602R, as router/SIP-ua with my architecture SER+Asterisk. Normally, everything is fine. In these days I'm experiencing some problems: some guests said me that, if he call everything is right, but if is called, he cannot hear the caller. Immediately, I though into an RTP-Proxy problem, but is not. Then I saw that message appear on the Asterisk CLI, during

Hi I have set up an Asterisk System with One XLite Phone and when i call the trunk line or receive calls via a trunk line (FXO generic X100P) i'm getting one way Voice. I can hear the called party - but they cannot hear me... Any ideas - is t a NAT issue or is it something to do with the generic X100P card. How does one sort this problem -- Mark D'Cruz D'Cruz Consulting