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Hi All, I have the problem setting up TDM22B card. Steps what I have followed are: [1] compiled zaptel-1.0.9.2 & installed the same. [2] modprobe wcfxo /lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg

Hello ppl, Reading all over the net. Learnt quite a lot, but that has left me confused-a-lot as well. Need answers to a few questions. Before that, I have an ISP(fax gateway) which will help me send/recv faxes using the T.38 protocol. I am using Asterisk 1.2.12.1. Now to the few questions I had: 1) Do I need any additional hardware on the Asterisk box?? I did download the spandsp and

Hi All I am not sure what I wish to do it possible but I would like to see if you guys know any better. I have a site who has the extensions: 1231, 1232. 1233, 1234 Each of these users can dial each other on the extension number an also has an external CLI mapped to them. On all internal calls or calls to services such as call forwarding their Caller ID is: Name <XXXX> What

Hello ppl, This post is to do with the variables 'nat' or 'canreinvite' for sip entities. Idealy users, wont be static, they could be roaming all over the globe. So, setting someone as behind NAT, and disabling canreinvite, etc., restricts the roaming capabilities of a user. Is there any way for Asterisk to auto detect, if a user is behind NAT, also, if two users are behind the

Hello ppl, The scenario : I restart asterisk, sip show peers shows nothing. I make a call from 7013 to 7011. I get the following o/p : SIP Seeding peer from astdb: '7013' at 7013@192.168.10.53 for 3600 SIP Seeding peer from astdb: '7011' at 7011@192.168.10.45 for 240 And then the call goes thru. So, does 'Seeding', means * registers both users?? But a subsequent

Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the

Hi All, I have two SIP softphones(Windows Messenger) running on different subnet (Phone-1 on IP XXX.XXX.25.ABC & Phone-2 on IP XXX.XXX.15.XYZ) and my Asterisk Server is running on IP XXX.XXX.25.PQR.Because of some security issues both the subnets are completely isolated ( U cant even PING from one to other) and I want to connect Phone-1 & Phone-2 to the *. How can I proceed? Please