VOIP-info.org Comments

/ How to set up for free VOIP [ID: 90903]

Fri, 27 Aug 2010 22:33:56 GMT

I currently have a Linksys SPA-942 IP phone, an IPKall number & account & a Google voice account. How do I set up everything so I can make & get free IP calls? Or is there something else I can use wto do this using my IP phone??
alsat () at 2010-08-27 22:33 GMT

/ Asterisk Server [ID: 90899]

Fri, 27 Aug 2010 16:03:11 GMT

How can I use MP3 files for music on hold. I would like to use a classic rock mp3 song from the eagles for my music on hold. I have tried converting it to wav and also mp3 but it will not play the file. Is there a special license I need to have installed in order for it to play the file the standard sample moh music it plays no problem but I can't get it to play the eagles any suggestions thank you
thinktech () at 2010-08-27 16:03 GMT

/ Re: Astrisk & Cisco 7911 HELP FOR A NEWBIE!! [ID: 90897]

Fri, 27 Aug 2010 14:02:06 GMT

If you are using TFTPD32 for DHCP you should be able to see in the Log Viewer tab what the phone is doing. Could you copy it here for me pls.
d_turburville () at 2010-08-27 14:02 GMT

/ Re: A simple script to autodial [ID: 90895]

Fri, 27 Aug 2010 12:51:58 GMT

Using asterisk, you can easily create a call file using a shell script. Here is a basic example that i have used in the past. When you run the shell script it creates the call file in the asterisk outgoing call directory, asterisk automatically reads the file and dials the number and connects it to an extension. #!/bin/sh echo -e "Channel: SIP/PeerName/+14125551212\nCallerID: Test <4125550000>\nMaxRetries: 1\nWaitTime: 5\nContext: sip\nExtension: 15555551234\nPriority: 1\nArchive: Yes" > /var/spool/asterisk/outgoing/4125551212.call Hope this may help get you going. Kent Pirlo VoIP Innovations www.VoipInnovations.com
voip_innovations () at 2010-08-27 12:51 GMT

/ sip stack [ID: 90894]

Fri, 27 Aug 2010 11:38:53 GMT

hi,,, please give me the clarification about this ,,,, how many types of sip stack is possible available in industries.... thanks and regards sant
santhoshknd () at 2010-08-27 11:38 GMT

Asterisk TAPI / manager v.1.2 [ID: 90892]

Fri, 27 Aug 2010 11:18:08 GMT

make sure you add "originate" permission in manager.conf for AsteriskTAPI to work with latest Trixbox.
norim (miro) at 2010-08-27 11:18 GMT

/ Re: Astrisk & Cisco 7911 HELP FOR A NEWBIE!! [ID: 90890]

Fri, 27 Aug 2010 10:28:02 GMT

Hi, I am beginning to lose hope with all of this. I tried your system tftpd32, the DHCP etc, but no joy. The phone is simply dead. I have tried both with POE, a switch and the phones physical power supply. One of the things I did was to setup an standard DSL router with 4 ethernet ports and enables DHCP. So I had my PC plugged into it as well as the cisco phone. This allowed me to have an isolated network. it appears that the phone is getting an IP from the DHCP side as I am able to ping it. After that did not work, I simply plugged the AC supply into the phone, with an ethernet cable directly to my PC, and ran the DHCP side from TFTP32. I can confirm that I do not need an cross over cable as I am able to ping the handset from my LT. Is it possible that the handset itself has been hardcoded to work only in a set range of IPs, even though I managed to wipe the unit? It just doesnt seem to be working and it is not the handset because I have another doing the exact same thing. The NIC card on the phone is working as I can see the address allocated to it through log files of Tftpd32 and can also ping. So is there anything else that I can look for if you dont mind advising me again? Thanks..
arriben () at 2010-08-27 10:28 GMT

/ Re: Astrisk & Cisco 7911 HELP FOR A NEWBIE!! [ID: 90889]

Fri, 27 Aug 2010 10:26:12 GMT

Unfortunately I dont think so. If you have 2 spare ports on your PoE switch you could put them both in a different vlan to the other ports and use them one for the laptop and one for the phone. That should keep the traffic logically separate from the rest of your network.
d_turburville () at 2010-08-27 10:26 GMT