search for: autocreatepeer

Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of course) does this adequately protec...

Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure because anyone can bypass the SER and register themselves as a peer with the asterisk. assuming i block incoming requests on the port asterisk is r...

Good day all How do I get my asterisk and sip to use the password.I'm using x-lite.If I use just the username and no password it still logs on? Here is my sip.conf entry? [101] type=friend callerid="Test User" <101> context = test_1 ; Default context for incoming calls username=101 secret=123456 host=dynamic dtmfmode=inband ; Choices are inband, rfc2833, or info

...uthorized people have been able to access the server (bots) and they have been able to make calls (in today's case to Cuba). Here's a copy (slightly modified) of my sip.conf: [general] context=default ; Default context for incoming calls videosupport=yes rtcachefriends=yes autocreatepeer=no t38pt_udptl=yes allowoverlap=no udpbindaddr=0.0.0.0 srvlookup=yes ;pedantic=yes disallow=all allow=alaw allow=ulaw allow=speex [1001] type=friend username=1001 secret=blah subscribecontext=default regexten=1001 callerid="blah" <XXXXXXXXXX> hos...

Hello, is it possible with Asterisk to force SIP authentication? Right now, it seesm that just any SIP client can at least connect to my PBX, which I don't want. I want users to authenticate with username and password and otherwise deny them access. Thanks Florian

....0.0.0 :2000 LISTEN asterisk . . . 0.0.0.0 :2727 asterisk 0.0.0.0:4520 asterisk 0.0.00:5060 asterisk x.x.x.x:5060 ser 127.0.0.1:5060 ser My config is like follows ;sip.conf [general] context=default port=5062 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no autocreatepeer=yes [2092] type=friend username=2092 canreinvite=no context=default mailbox=2092 host=dynamic nat=no dtmfmode=info disallow=all allow=ulaw allow=alaw ;extensions.conf ;leave voice messages exten => 2092, 1, Voicemail(u2092) exten => 2092, 2, Hangup ;play voice messages exten => 999...

Hi, Is there a "easy" way of adding 100 extensions? I mean, I don't want to create 100 section in the sip.conf. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040930/d8163cb5/attachment.htm

hello all, i have a problem, and i'm tearing my hair out...any assistance is appreciated. I am trying to redirect from SER to Asterisk, both on the same machine. In 1.09 I didnt need to set up a peer for SER, just autocreatepeer=yes, and rewritehostport from SER as below, and asterisk accepted the requests without a problem. When I updated to 1.23 requests from SER to asterisk die quietly, no matter how verbose my asterisk is. It's as if the requests dont exist at all. My setup is as follows: asterisk and SER on the...

...Mask Port Status 2001/2001 (Unspecified) (D) 255.255.255.255 0 UNKNOWN 2000/2000 (Unspecified) (D) 255.255.255.255 0 UNKNOWN I am pasting sip.conf & extension.conf sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = INVALID ;Autocreatepeer= yes [2000] type=friend username=2000 secret=2000 host=dynamic context=from-sip mailbox=100 canreinvite=no qualify=300 nat=no [2001] type=friend username=2001 secret=2001 host=dynamic context=from-sip mailbox=100 canreinvite=no qualify=300 nat=no extensions.conf [globals] CONSOLE=Console/dsp...

hi all, I keep asking the question and getting no replies, so i'll keep asking :-) In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. if in ser i was sending to 151@myServer, it will make it 151@myIP:5070, and asterisk will match i...

...Phone 202 shows an INVITE from 103 to 202. 103 happens to be the last listed in sip.conf and the first listed in 'sip show peers' (I have confirmed that this is dependent on the order in the conf file, not numeric order) sip.conf :- [general] port = 5060 bindaddr = 0.0.0.0 pedantic = no autocreatepeer = no context = sip registertimeout=20 localnet = 10.10.10.0/255.255.255.0 srvlookup = yes tos=0xb8 rtptimeout=300 rtpholdtimeout=1800 maxexpirey=3600 defaultexpirey=1200 [sip-101] ; Aastra 480i phones for general office type=peer insecure=very disallow=all allow=ulaw allow=alaw host=dynamic dtmfmo...

...d the asterisk servers are running 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to asterisk but not the other way. The call from asterisk to pri to tnt is good, the TNT is passing SIP invite to the SIP Asterisk server. I have tried many variations of using sip options insecure, autocreatepeer, permit/deny, host, user, etc.... but can't seem to get asterisk to accept an unauthenticated call from the TNT using SIP. I keep getting SIP/2.0 407 Proxy Authentication Required. I know others have done this, but with older Asterisk versions, I'm wondering what versions of Asterisk are...

Dear Asterisk Gurus: Our county is finally ready to begin implementing IP telephony. We intend to use a Cisco router as our PSTN gateway and Asterisk as our soft switch. The plan is to use SIP between the Cisco router and Asterisk. We will have a single PRI T1 connected to the Cisco router for PSTN access. My question is this: Are Cisco routers able to pass caller ID information (from PRI

...le clueless! Thanks for all of your help. Asterisk 1.4.31 built by root @ some_server.foo.net on a x86_64 running Linux on 2010-06-10 14:32:34 UTC Sip Settings: Global Settings: ---------------- SIP Port: 5060 Bindaddress: 0.0.0.0 Videosupport: No AutoCreatePeer: No Allow unknown access: Yes Allow subscriptions: Yes Allow overlap dialing: Yes Promsic. redir: No SIP domain support: No Call to non-local dom.: Yes URI user is phone no: No Our auth realm asterisk Realm. auth: No Alwa...

...d in SIP.conf. This, in addition to the SIPDOMAIN variable, makes the SIP channel even more useful. Thank you, Mark, for your additions! Now, ENUM/E.164 will propably work even better. I'll give it a try. Now, to be the documentation-pain-in-the-*** I would like to get an explanation of the autocreatepeer SIP.conf setting and functionality? It's not in sip.conf.sample yet. /Olle

I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO I can call out to the PSTN just fine, but inbound calls all appear in my [bogon-calls] context. Can anyone help me locate why? (Config files are on the Wiki) I have done a packet sniff & decoded using Ethereal-0.10.0, but this doesn't tell me a great deal - I just see

...esn't tell me a great deal - I just see the rejection message: y.y.y.y x.x.x.x INVITE sip:1234@x.x.x.x:5060 x.x.x.x y.y.y.y Status: 100 Trying x.x.x.x y.y.y.y Status: 503 Service Unavailable y.y.y.y x.x.x.x Request: ACK sip:1234@x.x.x.x:5060 (resent as retested with 0.7.1 & the addition of autocreatepeer=yes) Thanks a lot, Fran.

Hi, We're routing SIP calls through Asterisk and we want to be able to reinvite calls without Asterisk performing codec conversion. We've performed the following test: Asterisk has license for G.729 installed sip.conf [general] context=default autocreatepeer=yes disallow=all allow=alaw allow=g729 canreinvite=yes nat=no We have configured two endpoints: EP1, preferred codec order aLaw, G.729 EP2, preferred codec order G.729 EP1 places call to EP2, we see two call legs: EP1 to * is aLaw * to EP2 is G.729 Is there a sip.conf parameter to disable codec...

...rting to think that it might be the phone's fault and not mine... ;-> Still: Anyone out there has this working with Asterisk? The symptoms are that the G100 won't register, i.e. it doesn't respond to the SIP nonce that Asterisk sends out. A static IP won't help, and if I use autocreatepeer=yes then ringing the phone works, but upon picking up the handset the phone keeps ringing... However, since providers like Nikotel are selling the G100 there must be a way to get it working; my current guess is that with SER or some other SIP proxy it might come to life. Comments? Cheers &...

...; Port to bind to ;bindaddr = 10.0.0.229 ; Address to bind SIP channel to bindaddr = 0.0.0.0 context = 82.184.xx.xx ; Default context for incoming calls srvlookup = no ; Enable DNS SRV lookups on outbound calls ;;;;;;; tried with or without following lines, still mute :-( autocreatepeer=yes externip=82.184.xx.xx register => asterisk:xxxxx@10.0.0.229/100 ;asterisk actually registers on ser! realm=82.184.xx.xx ;;;;;;; tried also with public ip host, nat=no, canreinvite=yes, type=peer [asterisk] type=friend secret=xxxxx username=asterisk host=10.0.0.229 nat=yes canreinvite=no ;d...