similar to: Re:SIP Problem

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 am not registered my SIP Phone with Asterisk i spend almost one day but find no luck my configs are. sip.conf [general] port=5060 bindaddr=192.168.10.195 disallow=all allow=alaw allow=ulaw [101] username=101 type=friend secret=1234 host=192.168.10.195 context=sip callerid="101"<101> defaultip=192.168.10.176 extensions.conf [globals] [incoming] exten =>

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

Hello list, this is my first post on this list. I have a server with Asterisk and a Linksys SPA3102 with 3 SIP phones. I have configured the SPA PSTN line as trunk to receive and send calls. I can call outside from SIP phone throw the PSTN line and all is OK, the problem is when I receive a call from the PSTN, on the out caller phone there is a demo playback. I want to redirect the call to a

Hello All, I have a problem that is alien to me and obvious for some of you :). I have asterisk setup with few sip clients(using linphonec). In a proper context, I have mentioned extensions 107 as simputer@X.X.X.X (x.x.x.x=asterisk server ip) Asterisk Sever-------------------------simputer(sip ua) I can make calls from sipua to asterisk but not reverse way. I get the following display on

Hello All, I have a problem that is alien to me and obvious for some of you :). I have asterisk setup with few sip clients. In a proper context, I have mentioned extensions 107 as simputer@bogus.com Asterisk Server-------------------------simputer(sip ua) I can make calls from sipua to asterisk but not reverse way. I get the following display on asterisk terminal ---------------------

Hi all, I have Asterisk running on FreeBSD 4.x and I have made configurations to sip.conf, extensions.conf and voicemail.conf. I have also setup SIP phones on two different PCs. My problem is that when one of the SIP phones logins in, the other won't. My sip.conf has: [101] type=friend host=dynamic username=101 secret=test dtmfmode=rfc2833 context=from-sip mailbox=201

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 =>

Hi all, I have Asterisk running on FreeBSD 4.x and I have made configurations to sip.conf, extensions.conf and voicemail.conf. I have also setup SIP phones on two different PCs. My problem is that when one of the SIP phones logins in, the other won't. My sip.conf has: [101] type=friend host=dynamic username=101 secret=test dtmfmode=rfc2833 context=from-sip mailbox=201

I have: RedHat 9.0 TDM40B asterisk-0.9.0 compiled from sources zaptel-0.9.1 likewise /etc/zaptel.conf contains fxoks=1-4 loadzone = us defaultzone=us loaded modules zaptel and wcfxs /etc/askterisk/zapata.conf contains [channels] language = en signalling = fxo_ks context = phones channel => 1-4 /etc/askterisk/extensions.conf contains [general]

Anyone out there using NetLogic DIDs? And have inbound working? I got outbound working, but no joy so far with inbound. Here are the relevant parts from my conf files: iax.conf [general] tos=lowdelay jitterbuffer=no register => username:secret@zoot.netlogic.net [netlogic] type=friend host=dynamic context=sourcekit-main auth=plaintext username= secret= disallow=all allow=ulaw allow=all

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

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.

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 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 =>

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 =>

I've been beating myself up over this script but clearly I'm missing something. If I enter an extension like 101 it rings through fine, but if I pick 2 for sales it hangs up with this message: == Spawn extension (sales, s, 1) exited non-zero on `Zap/1-1' Since I'm not sure what that exacly means I cannot take appropriate action. Any help would be appreciated. [default]

Hello. You can't have two phones login with the same extension. You need to assign one phone to 101, and the other to 102. Set the user to 101 on one and 102 on the other. -Brian On Feb 11, 8:07am, "Juki" wrote: } Subject: [Asterisk-bsd] Asterisk not accepting multiple SIP phone logins } Hi all, } } I have Asterisk running on FreeBSD 4.x and I have made configurations to }

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

I would like to create a simple menu that would allow a caller to decide whether they want to leave a message or be forwarded to another number (i.e cell phone). Thanks in advance for any insight. Here's my current extension.conf [general] static=yes writeprotect=yes [globals] [default] exten => 101,1,Dial(SIP/101,20) exten => 101,n,Voicemail(101 at default) ;This automatically