[Asterisk-Users] Help with a fax via Grandstream Handytone 286?
I have an analog Fax machine which I wish to connect to the network and the Asterisk server. It will connect through a GS Handytone 286 converter and then into the LAN. Is there any information out there on what I need to write in *sip.conf* and/or *extensions.conf* to make sure the fax works as a fax? Channel 8 on my T1 is a reserved, dedicated line for the fax number. Do I need to create a special group in *zapata.conf* for the fax, or I can just have the fax extension dial only channel 8? I assume the latter, but I don't know if it will actually work. So far, I've managed to get the system up and running and working with the phones. But my limited knowledge is making it difficult to configure more advanced features. I have re-read the wiki numerous times trying to figure things out...sometimes it helps and sometimes it doesn't. Cheers, -- Shawn Parker Network Administrator Cumulus Broadcasting, LLC. Columbia-Jefferson City, Missouri 1.573.449.4141 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems dialing out with T100P and Adtran
Nevermind. I was a digit off in my zaptel.conf... the span for my adtran settings is 1,1,0,esf,b8zs instead of the one i have listed below. ph...one digit off. cheers, Shawn Parker wrote: I have a T100P card connected to an Adtran and then a T1. I have added the following configurations to Asterisk...but, when I dial 9 and then a local phone number, it bounces between the dial tone and silence and the *error* light on the Adtran blinks. zaptel.conf span=1,0,0,esf,b8zs fxsks=1-8 loadzone=us defaultzone=us zapata.conf [channels] context=from-sip signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=2 txgain=2 group=1 channel => 1-7 extensions.conf ... [from-sip] ignorepat => 9 exten => _9NXXX,1,Dial(Zap/g1/${EXTEN:1}) exten => _91XXXNXXX,1,Dial(Zap/g1/${EXTEN:1}) ; generic phone extension exten => 1001,1,Dial(SIP/1001,20) exten => 1001,2,VoiceMail(u1001) exten => 1001,102,VoiceMail(b1001) exten => 1001,103,Hangu ... sip.conf ... [1001] type=friend username=1001 fromuser=1001 callerid=User Name <1001> host=dynamic nat=yes canreinvite=yes dtmfmode=info [EMAIL PROTECTED] disallow=all allow=ulaw allow=alaw context=from-sip secret=1001 I am using Grandstream BT101 phones, plugged into my LAN. I can dial extension/phone to extension/phone in the office just fine. But, when I dial *9* to get out, nothing happens. I don't get the dial tone back after I dial 9, and if I dial 9 and the number and send the call...the server runs through what looks like a connection to a Zap channel...I don't get any noticable erros...but the call never makes it out. Once, I dialed the number again, while the Adtran was flashing erros and the dial tone was going in and out and it rang my cell phone...but then immediately hung up and closed out the call? I'm new to Asterisk...any help or insight would be much appreciated. Cheers, -- Shawn Parker Network Administrator Cumulus Broadcasting, LLC. Columbia-Jefferson City, Missouri 1.573.449.4141 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems dialing out with T100P and Adtran
I have a T100P card connected to an Adtran and then a T1. I have added the following configurations to Asterisk...but, when I dial 9 and then a local phone number, it bounces between the dial tone and silence and the *error* light on the Adtran blinks. zaptel.conf span=1,0,0,esf,b8zs fxsks=1-8 loadzone=us defaultzone=us zapata.conf [channels] context=from-sip signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=2 txgain=2 group=1 channel => 1-7 extensions.conf ... [from-sip] ignorepat => 9 exten => _9NXXX,1,Dial(Zap/g1/${EXTEN:1}) exten => _91XXXNXXX,1,Dial(Zap/g1/${EXTEN:1}) ; generic phone extension exten => 1001,1,Dial(SIP/1001,20) exten => 1001,2,VoiceMail(u1001) exten => 1001,102,VoiceMail(b1001) exten => 1001,103,Hangu ... sip.conf ... [1001] type=friend username=1001 fromuser=1001 callerid=User Name <1001> host=dynamic nat=yes canreinvite=yes dtmfmode=info [EMAIL PROTECTED] disallow=all allow=ulaw allow=alaw context=from-sip secret=1001 I am using Grandstream BT101 phones, plugged into my LAN. I can dial extension/phone to extension/phone in the office just fine. But, when I dial *9* to get out, nothing happens. I don't get the dial tone back after I dial 9, and if I dial 9 and the number and send the call...the server runs through what looks like a connection to a Zap channel...I don't get any noticable erros...but the call never makes it out. Once, I dialed the number again, while the Adtran was flashing erros and the dial tone was going in and out and it rang my cell phone...but then immediately hung up and closed out the call? I'm new to Asterisk...any help or insight would be much appreciated. Cheers, -- Shawn Parker Network Administrator Cumulus Broadcasting, LLC. Columbia-Jefferson City, Missouri 1.573.449.4141 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linux keeps deleting the ZAP files??
Everytime I reboot my server, the /dev/zap/* files get removed? Why is this happening? I have to recompile Zaptel everytime I reboot!! -- Shawn Parker Network Administrator Cumulus Broadcasting, LLC. Columbia-Jefferson City, Missouri 1.573.449.4141 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] If you run Archlinux...
I remember a thread a while back asking about Linux distro's of choice. My flavor is Archlinux. But, I found a tiny discrepancy with Arch and Asterisk's install. Arch uses *mawk* for it's awk compiler, unfortunately...Asterisk doesn't seem to pick up on it. So, I installed *gawk* as well...and the install went smoothly after that. Just a heads up if you choose Archlinux for your server. :) -- Shawn Parker Network Administrator Cumulus Broadcasting, LLC. Columbia-Jefferson City, Missouri 1.573.449.4141 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Install on Kernel 2.6.x
i know asterisk itself will install on a linux kernel 2.6.x, but i've seen places say that the zaptel drivers wont? is this still true? is it possible to build asterisk/zaptel on a linux 2.6.x kernel? -- Shawn Parker Network Administrator Cumulus Broadcasting, LLC. Columbia-Jefferson City, Missouri 1.573.449.4141 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the best Linux for asterisk
Although I haven't tried it for Asterisk yet, I use Archlinux (http://archlinux.org/) in my production environments. It's similar to Gentoo. It as a minute disk footprint, most popular software packages are available via it's *pacman* package manager, and you can get it in 2.4 or 2.6 kernel flavors. The Server I am building currently for Asterisk will run the latest build of Archlinux. I'll report back any major issues I may have with it...although I don't expect any. If it runs on Gentoo or Debian, then it will [normally] run on Archlinux. I gave up on commercial distro's like Red Hat, SuSe and Mandrake long ago. Cheers, Johnathan Bunn wrote: I would disagree, in any type of server environment you should be able to gain huge boosts from a properly tweaked kernel, I would suggest a lean distro like console-only gentoo setup with a custom tweaked kernel, and if compiling a kernel is hard just find some linux-geek who can ssh to you and build it for you, ( i have built many kernels like that ) On Mon, 16 Aug 2004 11:31:35 -0400, Vlok Stone <[EMAIL PROTECTED]> wrote: When deciding on Linux you decide which kernel to use. Linux IS the kernel part. After that it's what tools you're most comfortable with. That's where distros vary. In a biz environment you won't probably won't use a GUI. At home (less users) you may want it as a dual function server/ end user pc. So for a most reliable system find the most reliable kernel version. Also, the most reliable version of asterisk would be a more appropriate queston. To sum, there is no magic asterisk linux distro. All have the requisite components at their disposal ( well don't use linspire since they run as root for that ease of use/ hack). On Mon, 2004-08-16 at 09:25, Johannes van Hulst wrote: How has experience in Asterisk voip provider? I am trying to setup a reliable Linux system with Asterisk for a voip provider. Therefore I got two more or like identical systems. System 1 AMD Atlhon XP 2200 Asus A7V600-X bios 1002 1Gb memory 333 Mhz Asus 7100 videocard 120GB harddisk System 2 AMD Atlhon XP 2200 Asus A7V600-X bios 1005 1Gb memory 400Mhz Geforce MX 4000 64MB 40 GB Harddisk At both systems I have problems with installing Linux. I tried Redhat 9.0 but there the systems has badblocks all the time on the ext3 partitions and segmentation errors After that I tried Suse 9.1 and there the system is working perfect only when I compile Asterisk I get compile errors all the time with a warning internal error. I tested the partitions and the memory there is no problem. Can somebody help me out how to get a stabile system? Best regards, Han van Hulst ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Shawn Parker Network Administrator Cumulus Broadcasting, LLC. Columbia-Jefferson City, Missouri 1.573.449.4141 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79xx series IP phones
my apologies for using a previous thread. i was in a hurry when i did it and didn't consider that some would be using threaded views. Rich Adamson wrote: One problem a colleague and I discussed moments ago is the system may preform "better" if we're using a Layer 3 switch. Which I would like anyway, but isn't very cost effective for an office with 10 phones and 5 workstations. A L3 switch would be almost half, if not more, the total cost of the project build. I guess it also depends on the amount of traffic and calls made from the office. But, other than the 4 sales people, I don't imagine the office having a high volume of traffic in and out. It is a small publishing house not a call center or support company. The most they use the phone system for, other than selling ads, is talking to each other and calling writers. Layer-3 switching won't add any value unless your network is rather large and you already have a need to segment it into broadcast domains. I do a fair amount of professional network performance work and we've actually been involved with a flat (layer-2 only) switched network that has over 900 active non-voip devices. Adding a couple of voip devices (for testing only) into the mix resulted in excellent quality. But, all of that depends upon how well managed the network is in the first place. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Shawn Parker Network Administrator Cumulus Broadcasting, LLC. Columbia-Jefferson City, Missouri 1.573.449.4141 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79xx series IP phones
Thanks for the responses, everyone. One problem a colleague and I discussed moments ago is the system may preform "better" if we're using a Layer 3 switch. Which I would like anyway, but isn't very cost effective for an office with 10 phones and 5 workstations. A L3 switch would be almost half, if not more, the total cost of the project build. I guess it also depends on the amount of traffic and calls made from the office. But, other than the 4 sales people, I don't imagine the office having a high volume of traffic in and out. It is a small publishing house not a call center or support company. The most they use the phone system for, other than selling ads, is talking to each other and calling writers. Cheers, Joshua M. Thompson wrote: On Fri, 2004-08-13 at 11:31 -0500, Shawn Parker wrote: Does anyone have any knowledge or experience to give me dealing with Cisco 7902G and 7905G IP phones and getting them to work on a lan with Asterisk when *not* using other Cisco hardware? The sales guy is just trying to sell you more Cisco hardware. I have 7910s, 7940s and 7960s at my house all running just fine on a Linksys 24-port fast ethernet switch with zero problems. -- Shawn Parker Network Administrator Cumulus Broadcasting, LLC. Columbia-Jefferson City, Missouri 1.573.449.4141 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 79xx series IP phones
I got a call from our Cisco rep today saying that they couldn't sell just phones to anyone because if "my ethernet isn't to exact spec..." then they won't work at all. I've read over the Wiki documentation and it seems that the 79xx series phones work with Asterisk. They told me that without a Cisco phone system in place or a Cisco router or switch, then the ethernet wouldn't work with the phones. Is this true, or is it someone just trying to sell me a Cisco system? I don't see how my use of a Planet or Netgear switch would alter the spec of my ethernet to cause a IP phone to fail. Seems far fetched to me. I've never had any other problems mixing Cisco equipment with other product lines. Does anyone have any knowledge or experience to give me dealing with Cisco 7902G and 7905G IP phones and getting them to work on a lan with Asterisk when *not* using other Cisco hardware? Cheers, Shawn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New office hardware set up question.
Pardon the newb question and all, but this is my first real experience with phone systems, let alone VoIP and Asterisk. I'm building an office space for a former employer and we are considering Asterisk as the phone system there. But, I've never set up an Asterisk system before so I've got a couple of questions about the required hardware. The network architecture is pretty simple, we have an incoming T1 (into an adtran box). There is a 24-Port switch and a 24-Port patch panel for the network. The office needs 10 phones and they have 5 workstations. I was looking at building a server with a WCT100P card and using Asterisk on it, and using Cisco 7905G IP phones. Now, my question is this, do I need anything else for this set up to work? The IP phones plug into the lan like everything else, each has it's own IP address and appears on the network. If I run a T1 line into the T100P card how does incoming and outgoing calls work? The T1 can be broken up into as many voice/data channels as needed, and the company has existing phone and fax numbers. I'm a network guy, not a phone guy. Is there any other piece of hardware I need between the PBX and the T1? Or anything else I'm leaving out that makes the phones work with Asterisk? -- Shawn Parker Network Administrator Cumulus Broadcasting, LLC. Columbia-Jefferson City, Missouri 1.573.449.4141 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users