similar to: sip probe syntax

I am facing one way audio problem in sip trunking between asterisk and avaya. +-------------+ +----+ | avaya sip |-------| P1 | +-------------+ +----+ | | | +-------------+ | Asterisk | WAN

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

I am not that good at regex and it's use in Asterisk. I am running Asterisk 1.4.13 Currently I have this in my extensions.conf for incoming calls on our house phone line: [housemenu] exten => s,1,GotoIF($["${CALLERID(num)}" = "815xxxxxxx"]?s|12); 815xxxxxxx is our home phone number, when caller id fails or is missing that is what is recorded. I want to expand this

I am setting up my * for at home office and still have analog phones attached and answer from those analog phones and not necessarily through the pbx. I found that with the X100P cards, they see the 2nd ring and will be ready to answer the line. I used a Wait to pause and allow another 2 rings before * answers. But found that if we answer the line after the 2nd ring and before the 4th, * still

Hello, If my understanding is correct, these days it seems that many ISDN BRI lines are configured in energy saving mode in which signalling D-channel is "dropped" until a new call comes in. Is it possible to replicate this behaviour with Asterisk (when Asterisk is in NT mode and is seen as a public ISDN by another PBX, for instance) ? If not, would you it would be a useful addition to

Hi I have come across a problem, with my tdp410 and soekris board (basically pc on a chip amd geode cpu). I am using the box as a firewall/asterisk box. The problem occurs when I drop ppp and I get dead loop dectiotn going, I seem to lose interrupts and get lots of messages in syslog from wctdm24xx saying missed interrupt increasing latency its out lined here

I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -------------- next part

Hello, I was wondering if anyone has any insights on the best way to automatically monitor an asterisk box to check it is constantly available and processing calls. Many thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more

I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x "restart gracefully" However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? Thanks

Hi Which is the best interface card to connect PSTN line with Asterisk. Can somebody please help. My intention is to route the incoming PSTN calls to internal IP Phones through Asterisk and Vice versa. The Asterisk is in LAN and is reachable from all the IP phones in the LAN. Thanks Manoj -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a way to program an FXO device to totally ignore incoming calls? I want to put an FXO on a Fax line so that 911 calls can be sent via that line, but all other activity on the line is between the Fax machine and the phone company. Perhaps munge the ring tone detect if nothing else? Cary

I'm looking into getting a new phone and wondering what the difference in functionality is between a single line phone with call waiting and a real 2 line phone (either a real SIP phone or an analog 2 line phone and a 2 port ATA) is. Why would I want the real 2 lines vs. just being able to take an incoming call via call-waiting? Cheers, b. -------------- next part -------------- A non-text

I want to interface asterisk with a legacy pbx that has around 23 extensions through my 8 fxs card, how do i work around this? Hint: I have already terminated 8 extensions from the legacy PBX, i was thinking whether i can peer the extensions from the PBX i.e like 5 extensions be peered to one extension connecting to the fxs? How can i do this? Thanks in advance, -------------- next part

Is there a way I can change the ring cadence on FXS lines on a system using a Digium Wildcard TDM2400 card? I recently deployed a new phone system, and the customer has a few POTS phones in areas where they don't have data network services, so we're using the FXS lines to provide dialtone at those outbuildings. The old phone system would ring those phones with a short ring-short

Dear list, I am trying to configure the nagios plugin called check_sip. I just read the README file included with the plugin. I follow the readme steps to configure the plugin, without success. In the nagios web interface I can see (No output!) In the status information column. If I run the chech_sip plugin from a linux console, I get /usr/local/nagios/libexec# ./check_sip -u

I beg to differ. Digium is hiding from the real world and somebody is going take the software and run with it. My customers lost in excess of $50.000 and cut my pay in half, because of hackers. The hackers figured out how to scan every asterisk for weak passwords or open ports, and bang them real good. We need two things: a) disable in sip.conf the reply for INVITES that have wrong user

Does anyone have some decent Nagios scripts out there that do more than monitor the proc itself? Rather than reinvite the wheel, figured I'd ask. I already saw the one on the wiki. Matt

I am trying to get distinctive ringing going again with these phones, depending on the outside line the call comes in on. I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) But

My PSTN line doesn't allways hang up properly after it goes to voicemail. The problem occurs when a caller hangs up during the initial greeting. Even though the hangup accured, voicemail continues to record, usually a fast busy and/or a teleco generated "please hangup now" message. After the voicemail.conf 'maxmessage=180' expires the line simply stays offhook. The hardware

I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13 with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog phones via Adtran chan bank. When I went to * 1.4.13, the Uniden phones stopped being able to login to voicemail. All phones are on same lan with Asterisk. I get 'Login incorrect'