similar to: Call waiting on X100P

I did cvs update on asterisk, zaptel, libpri as of today (November 5, 2003). I also did 'make clean' on each of them. My previous version of asterisk was cvs of September 15, 2003. No other changes have been made to my system other that these updates. when running 'make asterisk' the following error appears term.c:55: conflicting types for `term_color'

Hello my isp has blocked outgoing and incoming connection for port 5060 . I have ssh access to server so i want to send all traffic from port 5091 to port 5060 of asterisk .so i tried iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to 127.0.0.1:5060 Now my softphone is able to register with asterisk but it isnt able to make any calls . bindport = 5091 in my sip.conf under

Hey list, I'm having problems to get running *. I don't have any digium hardware yet. I just want to perfrom some tests using SIP. I compiled asterisk and zaptel with ztdummy enabled on Fedora Core 3. When I try to start ztdummy I get the following message: localhost# modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line0: Unable to open master device /dev/zap/ctl 1

Has anyone ever gotten * to work on commercial unixes such as HP-UX, Solaris, AIX? What about other architectures than x86? -- . -----BEGIN GEEK CODE BLOCK----- Version: 3.1 GCS/GIT> d-@ s+:+ a? C+++ BLHIS$ U+++ P+> L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+> h---- r+++ y++++ ------END GEEK CODE BLOCK------ . -------------- next part

are there any open source sip softphone (Window OS version )?

Anyone seen or tried this yet? http://www.voip-weblog.com/50226711/uplink_connects_sip_skype.php Michael

Hello everybody, Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc. Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that.

ive had enough of cisco unity and microsoft exchange and im looking for alternatives to our voip system. right now, we have 3 cisco callmanagers, 1 cisco ip icd system, and 1 cisco unity voicemail system. all phones are cisco 7940/7960's and some ata186/188's. voice gateways are cisco vg200's with pri cards (5 total). im running h323 on the gateways and phones are of course

A customer of mine wants an IVR where the first 3 choices are 1 English 2 Spanish 3 French I can build the IVR but how do I get the system prompts to then speak the selected langauge. For example, a caller has selected Spanish and so is routed to the Spanish part of the IVR. At some point he breaks out of the IVR to leave a VM. How does the system know to continue offering him Spanish?

ive got asterisk running with chan_sccp and three cisco phones (2 7910's and 1 7960). lots of bugs. when i press the speed dial button on either 7910, asterisk dies. also, if i dial from the 7910 to 7910, everything works fine. i can dial from or to the 7960 once, and then one of the 10's and the 60 die and try to reregister. if i take the 7960 out of the mix and remove its

Anyone know how many simultaneous calls you can receive on a NuFone DID? -Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050310/e20d2eb8/attachment.htm

-----Forwarded Message----- From: IEG <dennis.andring@gmail.com> To: derek@kfuq.net Subject: Re: [Fwd: Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM] Date: Tue, 13 Sep 2005 03:04:42 -0700 The answer is a multiplexed terminal node controller (TNC) This was the very thought behind "trunked" communications around 800mhz. Gee ... there are a bunch of cell phone

Hi, I am very new to Asterisk. I wanted to know how to retrive the Voicemails. I could see some voicemails assosiated with some extensions. Any ideas?? --------------------------------- How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Guys, I have three Samba file servers running different versions of CentOS and Samba. They are all joined to a proper Windows 2003 domain. I'd like to migrate the WIndows domain to 2016 and all the Linux servers to CentOS 7. I can do Windows 2003->2012->2016 without too much trouble. I'm concerned about the Samba side. The servers are all using extended file system

Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP router using MGCP IP protocol, instead of controlling it through an E1. Have anyone tried this configuration? How does MGCP works? I've tried to search for it on Google, but I only find the protocol specification for it. Is Asterisk fully capable of this? I can't find any documentatin covering the use of MGCP in Asterisk.

I have been searching for some documentation that would indicate if Asterisk supports H.248 and everything I have come across seems to indicate I should use MGCP which I would agree is a better choice but unfortunately the equipment I am trying to integrate only does H.248. Could anyone point me to something related to this. -- Chad Whitten Metro Network Solutions (601) 366-6630 Phone (601)

I am putting together my first *. I had it running with two other pc's running xlite and setup voicemail and a couple of menus and submenus and had that running well. I had order a couple of oem x100p cards from digitnetworks. I installed them as they said with their voicepet2.2.zip drivers and did the modprobe on zaptel and wcfxo and then ran ztcfg -vv and got this: Zaptel Configuration

Giorgio, I'll answer in reverse order: I've not had reports of "noise" from my users. However, when I went down to get the s/w version from the phone that seems to be acting up the most, the user reported that earlier they were actually on a call that was ok then spontaneously dropped the audio. Per my instructions (based on another similar report I read on Digium's site),

I've got Asterisk up and running nicely using a couple of different softphones. Audio quality is suffering a bit due to the hardware that I am working with. So I tried to use a Polycom hardphone but the politics is enough to give you a headache. Polycom seems to support SIP only if you buy it thought their vendors. So I'm looking at a Cisco phone. Has anyone successfully implemented

My setup: X100P and Quicknet PhoneJack. I can't seem to properly set up a Zap channel for my X100P. Here are some of my configurations: [zaptel.conf] fxsks=1 #X100P fxoks=2 #Quicknet PhoneJack defaultzone=us loadzone=us [zapata.conf] [channels] context=local signalling=fxs_ks channel->1 ;X100P [extensions.conf] ... [local] exten=>_NXXNXXXXXX,1,Dial,Zap/1 ;I'm pretty sure the