search for: _xxxxx

I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register => 11111@fwd.pulver.com/11111 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=11111 fromuser=11111 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to

The SIP channels have a "call-limit" parameter (which is badly documented and I haven't tested yet) How can I have the same behaviour for IAX channels? I can't see anything related to it. Ah, I'm using Asterisk 1.2.13... maybe there is something in the 1.4 versions... but I can't change to 1.4 right now because of MFC/R2 BarZ

...Asteriskm config: **iax.conf** [general] bindaddr=192.168.0.160 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay authdebug=no [asterisk1] type=peer username=asteriskm auth=plaintext secret=asgard host=192.168.0.161 qualify=yes **extensions.conf** [general] [1ST-T1] exten => _XXXXX,1,AGI(rexx.agi) exten => 12345,1,Dial(IAX2/asterisk1/80483) exten => 12345,n,Hangup() Asterisk1 config: **iax.conf** [general] bindaddr=192.168.0.161 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay authdebug=no [asteriskm] type=user context=incoming-iax auth=plaintext secr...

...rfc2833, or info context=sip callerid="Yehiel" <10082> The same on 201 (the other line on Cisco ATA) the extension.conf file: [sip] exten => 200,1,Dial(SIP/200,20,tr) exten => 201,1,Dial(SIP/201,20,tr) include => fwd [fwd.pulver.com] exten => _XXXXX,1,Dial(SIP/${EXTEN}@fwd.pulver.com) exten => _XXXXX,2,Congestion Now when I type in * "sip show peers" it gives : Name/username Host Mask Port Status 201/201 192.168.0.167 (D) 255.255.255.255 5060 Unmonitored 200/200...

All, Is there a way of rejecting a call using SIP in the Asterisk Dialplan? Essentially, I want to look at the called number and if it matches something I don't like I want to send back a SIP response which will not cause the other end to 'hunt'. The response codes that will achieve this are: 401 Unauthorized 403 Forbidden Is there a way of getting Asterisk to send back such a

I've tried to register * at FWD but * segfaults, so i guess my register-line in sip.conf isn't correct. It looks like: register => fwd#:pwd@192.246.69.223 should it be different? Chris

Hello! How can one select outgoing MSN when dialing out from ttyI-interfaces? I have successfully done this with CAPI e.g... exten => _XXXXX.,2,Dial,CAPI/60:bBYEXTENSION ...in extensions.conf. Currently correponding for my ISDN modem interface is... exten => _XXXXX.,2,Dial(Modem/g1:${EXTEN}) ...but this selects only MSN of outgoing group g1 for dialout MSN number. I also tried to create multiple groups wait different MSNs and sam...

...conf" and "h323.conf" files are: *************************************** extensions.conf >>>> *************************************** [default] ..... ..... [outgoing] exten=>_7XX,1,Goto(voip-h323|${EXTEN}|1) exten=>_*XX,1,Goto(servicios|${EXTEN}|1) exten=>_XXXXXXXXX,1,Dial(Zap/1/${EXTEN}|30) exten=>_XXXXXXXXX,2,Playback(invalid) exten=>_XXXXXXXXX,3,Hungup() exten=>_X,1,Playback(invalid) exten=>_X,2,Hungup exten=>_XX,1,Playback(invalid) exten=>_XX,2,Hungup exten=>_XXXX,1,Playback(invalid) exten=>_XXXX,2,Hungup exten=>_XXXXX,1,Pla...

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

I've buyed SIP traffic from platinumcalling.com.au. in sip.conf i have: [platinum] context=default type=peer username=MYUSERNAME secret=MYPASSWORD host=sip.platinumcalling.com.au in my extension.conf: [default] exten => _XXXXX.,1,Dial,SIP/${EXTEN}@platinum|60|r When i try to call a number i got: Got SIP response 404 "Not Found" back from 203.30.19.164 SIP/platinum-67f2 is circuit-busy What's the problem ??? Thanks, Igor

Is it at all possible to set a Global Variable freely whenever a context gets used without having to enter an extension priority to use SetGlobalVar? This is really limiting the dialplan for me. Heres an example of what I would like to be able to do. [globals] AREACODE= [local] exten=_NXXXXXX,1,Dial(SIP/${AREACODE}${EXTEN}/blah) [anyoldcontext1] AREACODE=313 include=local [anyoldcontext2]

Hello, I try set Ua---SER----Asterisk (voicemail/ARA) | Ua ser stable asterisk cvs head I read http://mail.iptel.org/pipermail/serusers/2005-February/015997.html to forward unavailable or busy sip agents to asterisk voicemail in failure route. How may I configure extensions.conf and ser.cfg ? I have been trying without success! Regards Harry

Hi To dial an outside line i have to dial 0. I want to have that when we dial local numbers, that is we are in the 08 area, I don't want to have to dial 08, how to set this up in asterisk 1.6? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460

How does one do translation for calls that come in from other pbx's where the incoming caller ID is an internal extension number on their pbx? Eg. when I get a call from Free-World-Dial the CID shows up as "429102" which is essentially their internal extension number sans any routing prefix. To dial the number back I need to dial the extension with FWD's routing prefix

In several places online, and in the Asterisk F.O.T. book, there is a warning against using '_.' saying: "[it] should probably never be used". However, the need often arises act on numeric extensions that begin with *'s and #'s, and '+', and of course _X. does not match I have tried exten => _[0-9*#+]. but that seems to be the functional equivalent to _X.

Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk