similar to: Sending call to analog then to Vmail after timeout?

I had a Grandstream 286 at my home hitting my Asterisk box at the office, all worked well and I received phone calls fine until the device just up and died. I replaced this unit with an SPA-2000 because I have been impressed with the Sipura devices and decided to use them for most of my needs in the future. Problem is that my phone attached to the device rings shortly after power up of the

Hi, I am using a Netjet-s ISDN Card, and am having some trouble dialling out (Incoming Works Fine). TRUNK=Modem/ttyI0 exten => _90XXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:1}||Ttm) exten => _90XXXXXXXXX,2,Congestion I get the following when diallingout: -- Starting simple switch on 'Zap/2-1' -- Executing Dial("Zap/2-1", "Modem/ttyI0/04XXXXXXXX||Ttm") in new

Here is the situation: A T1 router going into an office which then plugs into the firewall box then into the switch. None of these devices support QOS.. Is there some sort of box/device that I can place between the T1 router and the firewall box which will allow me to prioritize voice traffic on this link? I can't change the T1 router to something that supports QOS because it has

I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. So far I have tested 4 providers which I will not mention here. I have found two

Can someone give me a clue as to why I keep hearing DTMF type beeps on my phone calls. It sounds exactly like someone on the other end is pushing a key on their phone but they are not! Has anyone ever heard of this before? It use to happen once in a while, today it's been happening a LOT and it's driving me batty.. -- Start Your Own ISP! http://www.YourOwnISP.com

Was using this monitor line to get soxmix to mix test-in.wav and test- out.wav into test.wav. exten => 1200,1,Monitor(wav|/tmp/test|m) When I start the conference, the * console shows this: monitor executing ( nice -n 19 soxmix "//tmp/test-in.wav" "//tmp/test- out.wav" "//tmp/test.wav" && rm -f "//tmp/test-"* ) & /tmp shows test-in.wav,

We are looking for a good firewall replacement which will basically do pot blocking and QOS. Our current solution just plain stinks.. We basically need to handle the traffic of a few web servers, mail server and asterisk box. The most traffic this device will need to handle is what can be shoved through a T1. I don't mind buying an appliance to get something solid but IP Cop just looks

hello, I''m running XEN 2.0 on IBM ThinkPad T23. Now the weird thing is that I get two different outputs from /sbin/lspci depending on whether I run 2.6.8.1-xen0 or 2.6.8.1-bproc. In particular the output from 2.6.8.1-xen0 seems to be missing those 4 lines 0000:00:01.0 PCI bridge: Intel Corp. 82830 830 Chipset AGP Bridge (rev 02) 0000:00:1e.0 PCI bridge: Intel Corp. 82801BAM/CAM PCI

hi, i am trying to do autoattendant but failing. as in the manual i inserted the background(welcome-mainmenu) file so that after the sound the caller can dial the extension he wants to call. i figured that the background sound wasn't coming in the asterisk. how do we do this without first loading the welcome message? for example after certain rings the caller can dial the extension no to

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Anyone using IAX.cc / Sixtel? Would love to hear experiences good or bad. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com

Can someone point me to some docs that explain this or give me a direction to go in. I have seen docs on this in the past but can't seem to dig em up now when I need them. Basically I want one Asterisk server to be the traffic cop and send some calls directly to ATA's and some calls to another Asterisk server, the other Asterisk server will then direct the calls to the end users ATA

Hi all, I'm stuggling with how to present calleds to a specific DDI (DID) with Music on hold whilst the call is hunted around 3 phones, then if not answered within a certain period forwarded to voicemail. So far I've got the queue working and the voicemail but not both together. Ive had a look on the wiki and the archives but can't spot anything that might point me in the right

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?

What I'm looking to achieve: Incoming calls to me extension will ring for 15 seconds. After that, I want the calls to forward to my cell phone and attempt to get through for another 30 seconds. After 30 seconds, I would like Asterisk to timeout the call, and goto my Asterisk voicemail. I've got the call forwarding down, but I'm using Nextel for cellular service, which loves give

I am looking to build a small PBX for an office that has 3 incoming analog lines and less than 10 extensions. For the Asterisk server I am going to use a small form factor PC with no-PCI slots so the FXO interface needs to be either FXO->SIP or USB. Can anyone make suggestions? I am looking at an AudioCodes MP114 FXO or possibly two Sangoma U100's but don't have experience with

Due to changes at the office, I'm finally getting around to setting up an AA to deal with incoming calls. One of the big changes is that we're dropping the old alphanumeric pager and will just send pages to our phones. I've got the outbound greeting message working in a test context no problem right now, but I'm kind of stuck on how to capture a DTMF sequence from a user and

Hi, I have a weird situation where I have a noop command putting the callerid of the caller on my asterisk console so I know who is calling as a test, but it is putting the callerid of my extension in instead of the callerid of the incoming line. My /etc/asterisk/zapata.conf is [channels] context=default ;switchtype=national usecallerid=yes cidsignalling=v23 cidstart=polarity hidecallerid=no

Hi, I had a FWD account set up with asterisk (using SIP) and it was working fine both ways. I switched to IAX2 and now I can't get incoming calls from FWD. People who call my FWD number get a "480 - user is not online" message without any traffic reaching my box. I can call FWD numbers fine over IAX2. It seems fwd isn't trying to place the call over IAX2 because it thinks