Re: [asterisk-users] PoE module
Here's a cheap solution for PoE piggybacked over your existing network. http://www.amazon.com/gp/product/B0002R6X9S On Jul 14, 2013, at 3:12 PM, bilal ghayyad wrote: Hello; We have a cisco switches but they are not PoE and we need only to have PoE device so the cables come for it first to provide the power and then goes to the switch (to be like batch panel), is there something like this that can be used for the IP Phones? Regards Bilal -- This message has been scanned for viruses and dangerous content and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Under heavy attack
On Oct 30, 2010, at 2:28 PM, Zeeshan Zakaria wrote: My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria It's been an extremely busy day for the exploiters. I moved my phone system from one circuit that I have (10Mb) to another that is behind a firewall (100Mb) and the fail2ban alerts are all gone. I'm not really concerned that someone will determine the passwords, as I use the phones serial numbers to determine that. But still, very irritating to see so many attempts at exploiting my phone system. fail2ban is nice, but I recommend you put your system behind a firewall and only allow necessary connections. pfsense is doing the trick for me. - Niles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Firmware
On Jul 21, 2010, at 7:05 PM, Apu Islam wrote: Can any good men on this group share me the firmware of a Cisco 7960 Phone? Currently this one has Call Manager Firmware installed, I am trying to convert it into SIP. Much appreciated. Apu Try google keywords: index of P0S3-06-3-00.bin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Ubuntu
On Jun 4, 2010, at 8:40 AM, Danny Dias wrote: Hello Asterisk users, I'm having a little problem with an Asterisk installation on Ubuntu, i had installed many asterisks on CentOS but never in Ubuntu, the problem is that Asterisk and DAHDI does not start at system start...i have to make /etc/init.d/asterisk start and /etc/init.d/dahdi start manually every time i reboot the machine (my laptop for testing) So, what should i do in order to solve this situation? Did you run update-rc.d on your asterisk/dahdi init.d scripts? Do man update-rc.d Niles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] migrate from zaptel to dahdi
On Jul 6, 2009, at 10:00 AM, Jerry Geis wrote: Over the weekend I tried to migrate a system from asterisk 1.4.25, libpri 1.4.1 ZAPTEL 1.4.12.1 to asterisk 1.4.25, libpri 1.4.7 (not 1.4.1) and DAHDI 2.2.0 I removed all old zaptel by: mv /etc/zaptel.conf /tmp mv /etc/asterisk/zapata.conf /tmp rm /etc/init.d/zaptel rm /etc/sysconfig/zaptel rm /etc/modprobe.d/zaptel 2 /dev/null /dev/null rm /etc/udev/rules.d/zaptel.rules rm /etc/rc.d/rc*/*zaptel rm /sbin/zt* rm -rf /usr/share/zaptel rm -rf /usr/include/zaptel Then I just did a CLEAN install of dahdi, libpri and asterisk again. After upgrading incoming calls seemed to work just fine. Outgoing calls gave me an error 99 I have a TE205P installed. I did change the extensions.conf to use DAHDI and not Zap. I had to quickly change back as it is a production system. Any thoughts on what might have happened here? I didnt know if have two libpri versions confused things or what? ANy thoughts for the next time I try are appreciated. Jerry, Check the dahdichanname setting in asterisk.conf. I had the same issue myself - Niles ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Mac OS X
On Apr 23, 2009, at 3:14 PM, Rick Dwyer wrote: Hello list. I posted this over on the Biz section but some of the members thought I might find more people running Asterisk on the Mac over here. Here's my question: I have looked at PHLink and PhoneValet and neither seem to be able to do what I need, so I am looking at Asterisk. What I want to do is allow callers to call a our phone line and unsubscribe their phone number from our call center list. So, basically, when they call in, they would be greeted with a message something like: please enter your 10 digit phone number followed by the pound sign. They would then have the number read back to them to confirm it or reenter it. Once confirmed, it would write the phone number to a text file for importing into MySQL or FileMaker. Is what I am trying to accomplish within the realm of what Asterisk can do on the Mac platform... or any platform... and if so, how difficult of an install is it? I have read varying accounts from it being a breeze to being frustrating. The main distinction between running Asterisk on Linux as opposed to OSX, is that you'll have access to hardware device drivers. If you're going to be using a SIP/IAX Trunk, then you'll be just fine on OSX. What your attempting to do falls closer to the category of breeze. You can install the asterisk-addon package to handle your SQL queries from within the dialplan, or you can use AGI to have a perl or php script do that work. Niles ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax solution for Asterisk
On May 21, 2008, at 9:55 AM, Sanjay Rajdev wrote: We are using Asterisk 1.4.13 on FC6 and have a T1 card with 20 DID's. We are wanting to use one of the DID's for Fax, is this possible or do we have to add some addition Hardware and what is the best way to do this. I know that similar thing would have been asked multiple time already, but I was not able to find anything that could answer my questions. Regards, Sanjay Rajdev I have 3 running installations of Asterisk using IAXmodem and Hylafax. Very very reliable, no additional hardware required. http://www.voip-info.org/wiki/index.php?page=Asterisk+IAXmodem http://www.voip-info.org/wiki/view/Hylafax Niles ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voice mail indicator on phone
Jerry, I'd imagine that you can achieve this through SIP Event Notify, via AGI using sipsak (www.sipsak.org) I'm doing a similar thing with Cisco phones, and it works great. Here's an example of what I pass to the phones. NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 From: sip:asterisk;tag=2427962554 To: sip:cisco Call-ID: [EMAIL PROTECTED] CSeq: 101 NOTIFY Contact: sip:[EMAIL PROTECTED]:5060 User-Agent: sipsak voicebox Event: simple-message-summary Content-Type: application/simple-message-summary Content-Length: 22 Niles On May 7, 2008, at 8:57 AM, Jerry Geis wrote: Is there a method from the dialplan that I can turn on a voicemail indicator on a polycom phone. Like a blinking light or something. Then I would also need to turn it off. Is there a way to do that? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting callerid across servers
On Mar 11, 2008, at 3:25 PM, Jerry Geis wrote: I have a situation when a T1/PRI line comes into box 1 then uses SIP over to box 2 and all my phones are on box 2. if the person is not at their desk on ring no answer I am calling their cell phone which places the call back over SIP to box 1 and out the T1 . How can I setup this configuration so the original caller ID will show up on the cell phone. Thanks, Jerry Jerry, What CID are you expecting to show on the cell phone? Based on what information you have provided, the original call is coming outside of your system, and you will not be able to duplicate their CID when you pass the call to your users cell phone. You can always screen the call though, allowing the recipient to know who is calling them. Niles ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Read function
On Mar 9, 2008, at 1:34 AM, Daniel Suleyman wrote:
Dear all, interesting behaivior of the Read function.
I have SIP phone(XLITE) attached to my Asterisk.
SIP.conf
[7007]
type=friend
qualify=900
host=192.168.85.27
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=alaw
allow=ulaw
extensions.conf
1,1,Answer;
1,2,Read(CNT,,2)
1,3,SayNaumber(${CNT})
Function read do not write anything to CNT or write .
in SayNumber it is always equel to ; even if I previously defins
CNT = 123;
And read function not exit if I pres #.(I think it is exit only on
timeout)
Strange can anybody point on mistake?
You have a spelling error at extension 1, priority 3.
SayNaumber, as opposed to SayNumber
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Re: [Asterisk-Users] Autostart Asterisk on Slackware?
Goran Dj. wrote: Maybe trivial question, but I cannot find an answer: How to autostart Asterisk (daemon) on Slackware 10? I know that I should put something in /etc/rc.d, but what? In my /etc/rc.d/rc.local # Put any local setup commands in here: /sbin/ztcfg /etc/rc.d/rc.hdlc /usr/sbin/asterisk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail Hangup detection issue
Hello, I finished an Asterisk installation this weekend, and I'm experiencing a problem when a user hangs up on a line after leaving a voicemail message. I found two similar issues when reading through the archives, and have not been able to resolve my issue from their answers. http://lists.digium.com/pipermail/asterisk-users/2004-April/042453.html http://www.marko.net/asterisk/archives/0212/.html At first, the voicemail messages would contain 7 minutes of busy signal at the end of the message, and now it contains about 40 seconds after adding busydetect=yes to zapata.conf and installing the latest CVS. Any ideas would be greatly appreciated. Niles Ingalls I'm using a Wildcard T100P, and have 11 incoming lines. zapte.conf span=1,0,0,esf,b8zs loadzone=us defaultzone=us fxsls=1-11 zapata.conf ; Zapata telephony interface ; Configuration file [channels] musiconhold=default language=en context=default switchtype=dms100 signalling=fxs_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes busydetect=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no channel = 1 channel =3-11 group=2 callgroup=2 pickupgroup=2 context=fxo1 channel = 2 ___ 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
