similar to: Dumb Dialout Question

How do you enable call waiting on sip phones? Ive looked and googled and can only find call waiting pstn phones butnot for sip. Is their a way of setting this up within the dailplan?

I have a Polycom IP501 phone and have set it up to download the config from an FTP server, it did this once and now is in an endless loop of trying to contact the FTP server, failing, then rebooting. When I watch the FTP server logs it looks like the phone starts a session, ends it, starts it, ends it until the phone reboots. It is annoying like nothing I can describe! I have tried Windows 2003

For some reason my voice mail stopped working properly. I was able to go in as a new user, set the password and options and now can never log back in using the password I assigned the mailbox. I can log in through the web interface with that password fine, and the voicemail.conf looks fine but every time I try to check messages I get "Password incorrect please try again" until it

Is there a protocol I'm supposed to use here? It seems that people are asking 100 questions a day and SOMEONE is helping them, and I've posted this three times and not even an "I Don't Know". My third repost: Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has

Hi, I'm using the macro below in extensions.conf for most of my outbound calls. One issue with my current configuration is that when I make an outbound call it doesn't properly detect that my PSTN line (Zap/1) is busy with another call and then overflow to my outbound IAX connections. I think the root cause is that DIALSTATUS gets reported as BUSY. The debug output is below. My desired

I can dial from 601 to a public number. The public number rings. I pickup and hear nothing, while on 601 it keeps ringing. (BTW, is it right to say "ringing" on the active phone?) The *CLI> doesn't show me anything useful: Executing Macro("SIP/601-8238", "dial-pstn|88097880|") in new stack Executing SetGlobalVar("SIP/601-8238",

Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened. I set up inbound routing to ring my extension if a call comes in - and my extension rings but when I pick it up I get a dial tone. The whole time after I answer I hear the phone I originated the call on just ring and ring

I'm trying to figure out how to allow an extension to register more than once. For instance, I have all of these 4 line IP phones that I use with Asterisk and I would like to have a persons extension (say 101) ring at all four lines so that if the person is on the phone they can take another call, but it appears as though if you try to register the same extension more than once then the most

I got two phones on an ATA-186 (601, 602) and two phones on the TDM22B (603, 604). I have two lines on the TDM22B. I cannot figure out some of the problems: 1. 601 dials via ZAP/3-1 to local phone number at PSTN: ringing pickup on PSTN (empty) still ringing in the phone set 601 2. call from PSTN back: 601 picks up ... everything works !!! No caller id shows up 3. For testing I have only one

Hello, I'm having trouble getting this to work: I have a ring group that dials an extension and if no answer dials a cell phone. If the cell phone doesn't answer I want to go to voicemail or another extension. I have set the timeout to 15 seconds but it never actually works, it will just ring until the cell voice mail picks up. I'm using Asterisk@Home 2.8 and a TDM400P card.

Hi I am in the process of move a company from pstn to an asterisk setup. They had 2 pstn lines - only really needed a max of 2 previously. Now I have installed a tdm410 to handle the cross over from pabx to voip handset. this has been done, the tdm is now just used to provide a backup pstn line - only used as a last resort for outgoing calls - as its shared with a fax line. I use 2 voip

Hi, How could I possibly inform incoming callers that the number they'd dialed is monitored and recorded. I wanted that when a call-in or call-out is made, a playback will be played to inform caller & callee that thier line is monitored prior to start conversation. Thanks. Angel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best

Greetings, I'm working with a new CentOS 7 installation, moving a system up from CentOS 5 due to OpenSSL version 0.9.8e not meeting PCI Compliance requirements. However, while setting up the CentOS 7 environment one of the closed source applications is requiring 0.9.8. The software vendor has advised installing package openssl098e from yum; but I'm hesitant to do so from a

Greetings, I'm running Dovecot 2.0.9 on my CentOS 6 server, for several thousand mailboxes. Recently, I've had several reports of "my mailbox is suddenly empty, where'd my mail go?" I've enabled debug logging, but I'd like to make sure I have the best level of debug to see things like "delete message" commands? I've configured in logging:

On 10/21/2015 2:34 PM, Eero Volotinen wrote: > Remember that rhel/centos backports fixes, so just looking version > number is not reliable way to detect security issues. > > Eero Indeed, though I can say on CentOS 5 the required configuration to be PCI compliand is not valid in apache, and httpd will not start. -- ----------------------------------------------- - Nick Bright

Hi! Maybe I have just been looking on the wrong pages but there is a question that is very important for me. I already studied some Demo-Dialplans and made some basic experiences with Asterisk. But what I need to find out is how I can handle this. I am leaving my office and I want to tell asterisk to forward calls now to my mobile phone by just hitting a key (on my IP-Phone) or by using a

Hello, i have a simple but annoying problem. I have the following entry in /etc/asterisk/externsions.conf file: ---<Cut Here>--- exten => 10100,1,Wait(4) exten => 10100,2,Playback(transfer,noanswer) exten => 10100,3,Dial(${PHONE30},30,t) exten => 10100,4,Background(extension) exten => 10100,5,Background(is-curntly-unavail) exten => 10100,6,Voicemail(9999) exten =>

I'm still learning firewalld obviously, and I am having trouble groking the documentation to understand how to do this. I know I could do an iptables direct, but that doesn't seem like the "right" way to do it. What I'm trying to do is allow a specific service, only for a specific ip. Effectively, SNMP should be allowed form a specific IP address (the systems monitor).

On 17 Nov 2015 17:30, "Nick Bright" <nick.bright at valnet.net> wrote: > > On 11/17/2015 11:12 AM, Nick Bright wrote: >> >> firewall-cmd --zone=monitoring --add-source=1.2.3.4/32 >> firewall-cmd --zone=monitoring --add-service=snmp >> firewall-cmd --zone=monitoring --add-interface=ens192 >> firewall-cmd --runtime-to-permanent > > I went

Hello All, I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The softphone can call the other one but I can' t hear any voice. My configuration files are below: [root at localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend