search for: peercontext

...he error persists. Greatings server B [general] bindport=4569 bindaddr=0.0.0.0 trunkdfreq=40 amaflags=billing disallow=all allow=gsm allow=g729 ;allow=g723.1 ;allow=ulaw tos=lowdelay jitterbuffer=yes [cornjundiai] type=friend username=corncea context=default secret=spojundiai host=133.1.150.1 peercontext=default qualify=yes trunk=yes insecure=very canreinvite=no incominglimit=30 disallow=all allow=gsm ;allow=g729 ;allow=g723.1 ;allow=ulaw [gatewaycea] type=friend username=sipcea context=default secret=ceaiax host=200.XXX.XXX.XXX peercontext=default qualify=yes trunk=yes insecure=very canreinvite=n...

...some more changes. Here is what I have thus far: Iax.conf on one server [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm mailboxdetail=yes [vvfarm] type=friend username=colo secret=testpassword auth=plaintext host=64.194.211.170 context=iax-incoming peercontext=vvfarm-extensions qualify=yes trunk=yes Extensions.conf on the same server [iax-incoming] exten => _###,1,Dial(SIP/17${EXTEN}-1,20) [remote-extensions] exten => _1XXXX,1,Dial(SIP/17${EXTEN}-1,20) exten => _1XXXX,n,Voicemail(${EXTEN:0:3}|su) exten => _1XXXX,n,Dial(SIP/${EXTEN}-1) e...

...each other as they were internal. I have succeeded in one way. Server B can call Server A without problem, but Server A cannot call Server B. Here's the iax configuration of servers Server A: ================== [ipek] auth=rsa context=from-internal host=XXX.XXX.XXX.XXX inkeys=ipek outkey=odtu peercontext=from-internal type=friend username=odtu Server B: ================== [odtu] auth=rsa context=from-internal host=YYY.YYY.YYY.YYY inkeys=odtu outkey=ipek peercontext=from-internal type=friend username=ipek When I try to call from A to B, I hear a message saying "All circuits are busy now, Plea...

...c:6778 socket_read: Rejected connect attempt from 127.0.0.1, who was trying to reach '123@' What is missing in the iax.conf to make asterisk accept the outbound calls? [iaxmodem] type=peer callerid="iaxmodem" username=iaxmodem secret=password host=dynamic disallow=all allow=alaw peercontext=from-internal context=from-internal TIA, Bruno

...-----+---------+-----+-------------+------+------ ---+-------------+------------+----------------+-----------+--------+--- --------+----------+----------------+------------+---------------+------ ----------+ | id | name | accountcode | amaflags | callgroup | callerid | canreinvite | context | peercontext | defaultip | dtmfmode | fromuser | fromdomain | host | insecure | mailbox | nat | pickupgroup | port | qualify | restrictcid | rtptimeout | rtpholdtimeout | secret | type | username | disallow | allow | regseconds | ipaddr | cancallforward | +----+-----------+-------------+-...

...load" in the CLI, asterisk tries to find the IP of the peer (based on its DNS name) and everything starts working again. This is the section for my trunk in one of my servers: [sometrunk] type=friend username=someusername secret=somesecret auth=plaintext host=host.no-ip.org context=incoming peercontext=incoming qualify=yes trunk=yes Is there any way to tell asterisk to try to find the peer's IP address if that peer is "unreachable" or each 10 minutes? Thanks in advance. Ronaldo.

...l. on Srv1: iax.conf: [general] bindport=4569 bindaddr=0.0.0.0 language=fr bandwidth=low jitterbuffer=no forcejitterbuffer=no autokill=yes calltokenoptional=192.168.0.20 [Srv2] type=peer host=192.168.0.20 qualify=yes trunk=no encryption=aes128 disallow=all allow=alaw allow=g729 context=Incoming peercontext=Incoming extension.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp ; Console interface for demo [Incoming] exten => _X.,1,Playback(demo-thanks) exten => _X.,2,Ha...

...on, but I can't get contexts to work right. My goal is to do a Dial(whatever@somecontext) and vary the somecontext based on different criteria. This is going on over trunked IAX2 links. How do I set up my IAX-accounts to manage this? I have tried to play around with 'context' and 'peercontext' on the server being dialed, but no luck. Is it legal to have multiple 'context' lines in one object? Is what I'm trying to do possible? Any help appreciated. Eivind Trondsen LinuxLabs AS -- Eivind Trondsen Tlf: +47 23 89 71 85 LinuxLabs AS Mob: +47 928 40 009...

...324948,1,Dial(Zap/1/) -- iax.conf -- register => athome:homeat@zoot.netlogic.net [athome] type=friend host=zzz.netlogic.net context=zap auth=plaintext username=zzz secret=xxx disallow=all allow=ulaw allow=all --- on zzz.netlogic.net iax.conf [athome] type=friend host=dynamic context=athome peercontext=zap auth=plaintext username=zzz secret=xxx disallow=all allow=ulaw allow=all -----Original Message----- From: Matt Schulte Sent: Friday, March 18, 2005 7:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Netlogic inbound DID issue Per Mike's iss...

Hello List. I have a question regarding connecting two asterisk servers. I'm trying to learn how asterisk comunicates from server to server. I already have a server running smoothly now, I'm installing another one to test it along side the actual one. I would like to run different scenarios: 1. Have one of the boxes at a different location outside the LAN and have them communicate. 2.

...obile. Default IP's can also optionally be given but ; are not required. Caller*ID can be suggested to the other ; side as well if it is for example a phone instead of another ; PBX. ; ;[dynamichost] ;host=dynamic ;secret=mysecret ;mailbox=1234 ; Notify about mailbox 1234 ;inkeys=key1:key2 ;peercontext=local ; Default context to request for calls to peer ;defaultip=216.207.245.34 ;callerid="Some Host" <(256) 428-6011> ; ; ;[biggateway] ;type=peer ;host=192.168.0.1 ;context=* ;secret=myscret ;trunk=yes ; Use IAX2 trunking with this host ; ; ; Friends are a short cut for creating...

I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been heavily modified. Currently asterisk runs on localhost and only listens on SIP/RTP at 127.0.0.1 . Therefore, all of the SIP traffic appears to