similar to: Ringing() doesn't play sound while phone is ringing

Hey all, Does anyone know what this means? I was running asterisk fine. Installed it on a new pc and I am using the g729b. codec that is optional. I ran the install for the codec it went ok but when I run askterisk via asterisk -vvvgc it gives me this error anyone know? I make sure I entered in the correct reg number. I followed the steps correctly. Too Registration error! Please try

Morning, Have some questions regarding receiving and sending faxes... 1:st example: exten => 101,1,Answer() exten => 101,2,Wait(3) exten => 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten => 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf) exten => 101,5,System(mutt -s 'New FAX for you sir' -a

Hey guys, Im interested in hearing about servers (and thier hardware specs) that successfully run both asterisk and samba for an office of maybe about 12 extensions (SIP) and about 12 workstations. Im hopeing to not only replace a traditional PBXs with Asterisk/Linux but to provide a solution to needs such as a file serving, email serving, etc. Ive read the Success stories form voip-info.org but

HI, I'm new to WINE and Ubuntu! I am using Ubuntu classic 11? I've got WINE installed because my webbulider is WINDOWS based. I also have a problem watching YOU TUBE videos in FULLSCREEN MODE. It worked fine with Windows - is there something I can do to rectify this? I do not understand all the components in a system such as what DLL, kernels etc are, so please describe what I can do if

Hi everybody In advance sorry for my bad english and if my problem was already exposed (I didn't find any tips in the mailing list archive. Bad luck) I have some questions about asterisk 1.6 release : 1) how can I do a n+101 priority jumping if a SIP canal is busy ? I read that the general parameter "priorityjumping" is depreciated in the 1.6 release and I already try the

Hi all, I have a voice menu system ( Outlined below ), and I'd like to give the user some feedback when they dial an extension ( ringing, music, SOMETHING ). As it stands, when a user enters an extension from the menu system, they hear silence while the line rings. I even tried including the Ringing application before calling my macro to dial the phones, with no luck. Any help is

Hi, I have a server with 2 TE110P cards. 1 card is plugged to telco line, another card is plugged with a Hicom PBX. I want to send some call to VoIP phones and all other to my PBX. I don't known how to make my dialplan : ===========Extensions.conf========== [incoming_call] exten => 090200000,1,Goto(callcenter,100,1) exten => 022956353,1,Goto(callcenter,100,1) exten =>

I know I can dial multiple channels in sequence exten => 101,1,Dial(SIP/101,10) exten => 101,2,Dial(SIP/102,10) extne => 101,3,Dial(Zap/1/5551212) What the boss would really like is to be able to ring 2 lines simultaneously. exten => 101,Dial(Sip/101,10) && Dial(Sip/102,10) so that both extensions ring at the same time... mostly so that he can have the remote phone at his

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729

Hi everyone, I just installed asterisk on my system with the purpose of rerouting calls on sip channels. I don't think i need any hardware for that. I am using LIPZ4(zultys) and sjphone as softphones. I tried setting up both of them and to call one from the other on the same machine, however could not. I 1-) I could connect sjphone in isolation to freeworld dialup howver i got no sounds

When i made internal call into my LAN using x-lite sip phone client I retrive in askterisk CLI : ----------- ERROR ---------- Verbosity is at least 6 -- Remote UNIX connection -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack -- Called 201 May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call

Hi everyone, My manager interface seems to be producing wrong CallerIDs when internal extensions call each other. Can anyone see anything wrong in the configuration snippets pasted below? The following instance has extension 101 call 103. The phone does show the right caller ID, but notice that the manager interface has the CallerID as the target number (103). Thanks a lot for your time.

Hello. I'm having trouble with callerid on outgoing calls. The recipient of the call only sees "unknown" rather than the number I'm specifying. If I set callerid info when calling an internal extension then I see the callerid name and number when I call that extension. I did that thusly: exten => 101,1,Set(CALLERID(number)=1112223333) exten =>

hi, I am not registered my SIP Phone with Asterisk i spend almost one day but find no luck.I know very well this is not kind a problem discussed in this group but i try my best and all in vein so finally i am here hoping you ppl helping me out.I discussed this problem in asterisk's-users group and adding feedback from asterisk-users group my configs are sip.conf [general] port=5060

Hi, I can't receive/send calls with Asterisk. Could someone please give me a few pointers on my configuration? Regards, Norman Zhang ; sip.conf [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=x.x.x.x localnet=192.168.22.0 mask=255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register =>

All, I have been beating my head against a wall trying to figure out how I would implement a separate moderator code and participant code for the same conference using meetme, the deal is I dont want the participants to be able to join until the moderator is in the conference. Is it possible to do this using the apps as they are , or is their a way to use an Agi script, is that the only way?

I'm not having any trouble with interrupts, but here's my /proc/interrupts on Fedora Core 2 on a hyper-threading CPU and using the SMP kernel (2.6.5-1.138). I don't think I need to worry about uhci_hcd, nothing is plugged into USB, but libata is the disk driver. How do I get libata and wctdm to use different interrupts? $ cat /proc/interrupts CPU0 CPU1 0:

Hi, I get the following message when I try to run Asterisk. This happened after I changed some of the configuration files: extensions.conf, zaptel.conf and zapata.conf. I have both TDM40B and X100P cards and am trying to use them together. Please let me know if there is something else I should do. Thnx, Gladys [chan_local.so] => (Local Proxy Channel) == Registered channel type

The FXO lines on my TDM400 are connected to PSTN lines in Israel. As far as I know, the lines around here have disconnect supervision (I've seen some other Israelis on this list, anyone know for sure?), because it's worked on Dialogic cards, which reported hangup, not busy detect (while when I connect a Dialogic card to a PBX, I have to measure the busy signal's frequency/cadence or

I am trying to put together a list of 'departments' to request as voice prompts. I have the biggies (sales, accounting, shipping, etc...) but I want to make sure I do not miss any. If anyone anyone has some suggestions (Ha... that is like going to an NRA meeting ans asking if anybody has a gun :-) ) please forward them to me (and / or post here although, with the volume of this