similar to: Asterisk and festival

Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial("SIP/Phone4-e85b", "SIP/612@fwd.pulver.com|90|Ttr") in new stack -- Called 612@fwd.pulver.com -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)"

Hi, all I have couple of SIP phones and they are in [from-sip] context. I also have an IAX2 phone. I have put this one in [iax-user] context. I want to make calls between SIP and IAX2 phones. If I put them all in same context all is fine, however when they are in different contexts they will not call each other and I will get message (in * CLI) that particular extension does not exist in a

I'm trying to get two linux machines with kphone-3.11 two communicate with each other over asterisk. I have them configured correctly on asterisk to use sip channels, but when I call from one phone to the other I don't any voice communication between the phones. According to the phones I'm connected, but according to asterisk, I get the following message: -- Executing

I have the following asterisk configuration (sip.conf) : [General] externip=82.79.81.3 localnet=192.168.10.0 localmask=255.255.255.0 [Phone1] type=friend host=dynamic nat=yes qualify=yes context=sip callerid="Phone1" <1> disallow=all allow=gsm [Phone2] type=friend host=dynamic qualify=yes context=sip callerid="Phone2" <2> disallow=all allow=gsm [Phone3]

Hey guys, I'm looking for a programming or Development Team/Company to do some custom coding for Asterisk. What we need is not exactly simple. In fact, I'm not sure the extent of the coding as far as technical terms go at all. Currently we have a "call center" with 4 phones. There will be a total of 8 people using the phones. Obviously, no more than 4 people will use

Hi! I have two, not too related questions: - the probably simpler one: if anyone can help me out using a Cisco 7905G with chan_sccp? I did already managed to get it working with a SIP image, I'd just like to see it work with this one as well. It's probably something I screw up with the configuration, as the phone registers, only I don't get any lines with it, although I have it

All, I'm having trouble getting the X100P working. Lsmod shows : zaptel 179808 0 I did a . # modprobe zaptel and here is my zaptel.conf (comments omitted) __SNIP__ fxsks=1 loadzone = us defaultzone=us __SNIP__ Here is zapata.conf __SNIP__ [trunkgroups] [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 usecallerid=yes hidecallerid=no

Hi, I've been playing round with Asterisk on Redhat 9 (2.4 Kernel) and was experiencing bad echo problems using Budgetone 100's when calling analogue lines in uk (Isdn4Linux / Digi Datafire). Calls to other ISDN and mobile network seemed ok although not much testing done. I've tried installing Asterisk on a faster processor (P4 3.0 GHZ) with a 2.6.8-24 kernel to see if that helps.

I'm new in hire so Hello to everyone! I'm beginner user of Asterisk CVS-HEAD-10/01/04-14:31:34 . I just installed it on my Mandrake Linux 10.0 kernel 2.6.3-4mdk with sample configuration (used make samples). I would like to make phone connections between X-Lite (SIP) installed on computers in LAN. How to make this? I was reading manual, and tried to make changes in sip.conf but this all

I have an X100P used, at present, largely for outgoing calls. It shares the single incoming POTS line with a number of analog phones. Is it possible to talk the X100P (Zap/1) to answer a ringing call only if I ask it to? I'd like to use only the SIP phone in my office, but let the analog phones continue to work in the rest of the house (until I can afford FXS cards anyway..) I can force

Dear all, I've been taking my baby-steps toward setting up an Asterisk phone system in my office, as also between my home and office (connected by DSL). I'm have a rough time getting two * boxes talk IAX over a LAN. I don't know what I am doing wrong, but am attaching my iax.conf and extensions.conf on both the boxes. Does anyone see it? ------config files start------ site-0

Hey, how can I implement a dial plan like the following: incoming call: 1. ring phones 1,2,3 monday to friday between 9:00 and 20:00; if no answer after 15 sec also ring phones 4 and 5 2. ring phone 1 monday to friday between 0:00-9:00 and 20:00-24:00; if no answer after 20 sec also ring phones 2 and 3 3. ring phone 1 saturday and sunday all day I do not need a in detail answer for each of the

Hi, I have an FXO card connected to my phone line which works in Asterisk as Zap/1. Is there any way of detecting whether something else is on the line before picking up on this channel? For example, I dont want to pick up and dial out on the line if someone is on it using another phone (which is connected directly to the line, rather than through Asterisk). Also, when an incoming call comes

Hi, I need to setup a demo for asterisk and need some help here please. The demo is connecting to Asterisk a Cisco 7970 SIP (ver. &.0) and a SIP client on HP iPAQ via a wireless hotspot. I need to configure both with the same extension with a shared line like in Cisco CallManager. This way if the extension is called both iPAQ and the IP phone ring and the user gets to pick up using either.

My analog card, uses a PCI slot and a 12V power connector, is configured with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS ports but I can't dial out from them. Is extensions.conf where I need to make changes? [root at robin asterisk]# cat chan_dahdi.conf [trunkgroups] [channels] [phone](!) usecallerid = yes hidecallerid = no callwaiting = yes usecallingpres = yes

Hello. I have another issue. When I call in, everything is processed correctly, including voicemail, but I don't hear any ringing/ringback. exten => s,1,Zapateller(answer|nocallerid) exten => s,2,NoOp exten => s,3,Playback(pls-wait-connect-call) exten => s,4,Dial(${PHONE1}&${PHONE2}&${PHONE3}&${PHONE4},15,Ttm) exten => s,5,Answer exten => s,6,Wait(1) exten

Hello All, This could be more a routing/linux problem, but I'm wondering if can be handled somehow within *. I have a client that has a contract with a VoIP provider. This contract limits number of phones registered using same IP to 3. There are 3 ways to handle this: 1. Few ATA connected to 3 phones + network 2. Linux router that lies between * box and network. Virual interfaces, iptables

>From this list I found that I could use SetGroup and CheckGroup to do what I wanted. But I'm not quite sure how I do it. The case is that I have 3 user groups, and one main group. The main group is for all the incoming calls from external phones. The main group should be allowed to have 3 calls at the time. The 3 user groups are internal groups that I want to limit by ony having one

I'm using * at home and I planned on having * let the answering machine in my kitchen to the "general" voicemail getting. However, about 6s into the call * will hang up the line. I found a post about OHT somethingorother, so I can probably work around it, but I'd like to know what's happening and if there's a better way around. Thanks! -- -M There are 10 kinds of

This may be obvious but I have not found the answer in the archives or web searching. I am in the process of transitioning to Asterisk. While I have two systems connected to the same PSTN line, I want to configure Asterisk to not answer an incoming call. Is this a setting that you would have in the zapata.conf file? -------------- next part -------------- An HTML attachment was scrubbed... URL: