search for: careinvite

I'v a sip client and a sip trunk to FWD: [general] port=5060 context=default tos=reliability disallow=all allow=ulaw careinvite=no [freeworlddialup] context=default type=friend username=MYUSERNAME secret=MYPASSWORD host=fwd.pulver.com [igor] type=friend callerid="Me" host=dynamic dtmfmode=rfc2833 careinvite=no When i try to call a FWD number from SIP client i obtain a lot of build_route: messages from asterisk...

...he new account 444 ??? Below are the conf files and verbose output. Thank you very much for your help :) --------- - iax.conf --------- [general] bindport=4569 delayreject=yes language=fr autokill = yes calltokenoptional = 0.0.0.0/0.0.0.0 minregexpire = 60 maxregexpire = 500 mohsuggest=default careinvite=no rtcachefriends=yes [444] type=friend host=dynamic context=special secret=iop --------- - extconfig.conf: --------- [general] [settings] iaxusers => mysql,asterisk,iaxfriends iaxpeers => mysql,asterisk,iaxfriends voicemail => mysql,asterisk,voicemail --------- - Mysqldump from iax...

...[packet8] type=friend disallow=gsm secret=mypassword allow=g723.1 ;expirey=15 sip_codec=g723.1 username=0403531400 fromuser=0403531400 host=packet8.net [0123456789] type=friend allow=g723.1 defaultip=192.168.1.247 user=0123456789 fromuser=0123456789 context=default host=dynamic secret=mypassword careinvite=no expirey=30 reinvite=no and from the extensions.conf, I dial like this: exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@packet8) exten => _91NXXNXXXXXX,2,Congestion In the dta310, I have the following SIP server settings: Ip address: the asterisk machine's IP Port: 5060 Domain Name: th...

How can I make it more readable? Name/username 601/601 123456789/123456789 voipbuster/abcd 601 = hotline 123456789 = Peter Pan only voipbuster/abcd is easy read/understandable! bye Ronald Wiplinger

...optimumwireless.com localnet=172.16.0.0/16 register => username:secret at my.service_provider.tld language=es ;allow=gsm allow=all [voipprovider] type=friend host=208.78.163.3 username=username fromuser=username secret=password port=5060 dtmfmode=rfc2833 nat=yes insucure=port,invite allow=all careinvite=yes I don't know what else to try. When I try to call I get this at the cli: == Using SIP RTP CoS mark 5 -- Executing [91xxx763xxxx at default:1] Dial("SIP/102-b6a06a40", "SIP/1xxx763xxxx at voipprovider") in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx763xxxx at...

Can anybody please answer this. Both clients are behind different NAT's. One of them starts a SIP call to the other through Asterisk. Asterisk sets up the call. Issues reinvite and connects them together. After this point does the media stream flow through Asterisk or Peer to Peer? Does such a call use any system resources of Asterisk server after connection? Thank you in advance.

...t=blah ;dtmfmode=inband ; Choices are inband, rfc2833, or info ;mailbox=1000 ; Mailbox for message waiting indicator context=sip nat = yes callerid="Me" <2124> [mysjphone] type=friend host=dynamic port = 5060 dtmfmode=inband username=mysjphone secret=mypassword context = sip careinvite = no nat = yes ################### the relevant section in extensions.conf are ##### [sip] exten => 1,1,Dial(SIP/zultys,20,tr) exten => 2,1,Dial(SIP/mysjphone,20,tr) exten => 1000,1,Dial(SIP/zultys&SIP/mysjphone,20,tr) exten => _8.,1,Dial(SIP/${EXTEN-1}@fwd.pulver.com,tr) exten...

...f the call. Any ideas what I did wrong? here is my messenger sip.conf portion; [marko] type=friend reinvite=no username=marko host=dynamic mailbox=1300 here is my cisco 7905 sip.conf portion; [123] type=peer reinvite=no callerid= "Marko Rakar" username=123 secret=1234 dtmfmode=inband careinvite=yes host=dynamic defaultip=192.168.3.52 incominglimit=2 outgoinglimit=2 here is a part of my sip debug file 9 headers, 0 lines Sending to 192.168.3.54 : 14250 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.54:14250 From: "marko" <sip:marko@asterisk>;...