search for: fromdomain

...from 147.135.8.128 ==================================================== Here is my sip.conf: ======================================================= register => XXXXXXXX@sip.broadvoice.com:PPPPPPPP:XXXXXXXX@sip.broadvoice.com/broadvoice [XXXXXXXX] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=XXXXXXXX secret=PPPPPPPP username=XXXXXXXX insecure=very context=default authname=XXXXXXXX dtmfmode=inband dtmf=inband canreinvite=no ================================================ -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel...

...0493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYVERYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=luca_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493511111111 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [pbxfax] type=peer defaultuser=00493512222222 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=fax_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493512222222 fromdomain=172.16.34.132 u...

Is there a way to override the fromdomain specified in the sip.conf and instead set the value from the dialplan? If we use: Set(CALLERID(num)=user at domain.com The SIP From header turns into: user at domain.com@10.10.10.10 We want user at domain.com, and we can't have an entry in sip.conf for every provider. -- Eric Chambe...

https://issues.asterisk.org/jira/browse/ASTERISK-20841 The patch was already posted by someone but then was deleted because of guide lines. Is it really that hard to fix? Since 1.8 there is this problem but nobody seems to care about. Asterisk isnt only used with itsp who dont care about fromdomain. Or are the developers saying, we dont care about people who are using Asterisk in smaller installations?

...com/25461001 register=>25461002:xxxxx@sip.provider.com/25461002 register=> 25461003:xxxxx@sip.provider.com/25461003 . . . register=>25461099:xxxxx@sip.provider.com/25461099 [provider-25461000] type=friend context=default secret=xxxx username=25461000 host=sip.provider.com fromuser=25461000 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes [provider-25461001] type=friend context=default secret=xxxx username=25461001 host=sip.provider.com fromuser=25461001 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes [provider-25461002] type=friend contex...

How can I force my users to be obliged to give a 'fromuser' and 'fromdomain' -parameter in their SIP-configuration ?? Is this set in the [general] -section of sip. conf ?? Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091109/43dbedb0/attachment.htm

...ber register => user:pass at sip.voiparound.com/number register => user:pass at sip.webcalldirect.com/ register => user:pass at ixcall.net/number register => user:pass at freedigits.net/number register => user:pass at sip.messagenet.it:5061/number context=inbound bind=0.0.0.0 nat=yes fromdomain=sshn.net localnet=10.0.0.0/255.255.255.0 externip=195.xxx.xxx.xxx srvlookup=yes [authentication] [eutelia-out] ;maxexpirey=360000 ;defaultexpirey=180000 type=friend allow=alaw context=inbound username=xxxx secret=xxxxx fromuser=number fromdomain=voip.eutelia.it host=voip.eutelia.it dtmfmode=inba...

...d secret=passwd ; password host=dynamic ; how to find a client qualify=yes ; qualify peer is no more than 2000 ms away nat=yes ; this phone is not natted host=dynamic ; this device registers with us fromuser=user1 ; calling from user fromdomain=domain.tld ; calling from domain realm canreinvite=no ; allow RTP voice traffic to bypass Asterisk context=internal ; the internal context controls what we can do fromdomain=domain.tld ; from domain -the domain that the username is from mailbox = 5001 at default ; avoid-Receive...

...ually reads the incoming call from BV but then issues a busy signal sending the call to BV's voicemail. I also modified extensions.conf as follows: [from-sip-external] include => from-pstn I have set up my sip trunk in AMP as follows: Trunk Name: Broadvoice Peer Details: dtmfmode=inband fromdomain=sip.broadvoice.com fromuser=21xxxxxxxx host=sip.broadvoice.com qualify=yes secret=password type=peer username=21xxxxxxxx My Incoming Settings are: User Context: sip.broadvoice.com User Details: context=from-pstn dtmfmode=inband fromdomain=sip.broadvoice.com host=sip.broadvoice.com nat=yes secret=p...

...1.0/255.255.255.0 disallow=all allow=ulaw register => num1:pass@sip.broadvoice.com register => num2:passsip.broadvoice.com tos=0x18 srvlookup=yes nat=never insecure=yes [sip.broadvoice.com] type=peer username=NUM1 fromuser=NUM1 authuser=NUM1 secret=SECRET host=sip.broadvoice.com context=sip fromdomain=sip.broadvoice.com canreinvite=no nat=never dtmfmode=inband [sip.broadvoice.com.home] type=peer username=NUM2 fromuser=NUM2 authuser=NUM2 secret=SECRET host=sip.broadvoice.com context=sip fromdomain=sip.broadvoice.com canreinvite=no nat=never dtmfmode=inband [broadvoice-incoming] type=peer host=1...

...; is still in memory (due to caching or other reasons), the ; information will not be removed from realtime storage [8157582715] type=friend accountcode=2 context=ottos secret=XXXX username=2715 fromuser=8157582715 insecure=very host=63.175.151.3 ;voip.essex1.com fromdomain=63.175.151.3 ;voip.essex1.com canreinvite=no disallow=all allow=ulaw allow=alaw ;qualify=yes [8159092441] type=friend accountcode=12 context=rwest secret=XXXX username=2441 fromuser=8159092441 insecure=very host=63.175.151.3 ;voip.essex1.com fromdomain=63.175.151.3 ;voip.essex1.com canreinvite=no...

...neral] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes tos=0x18 checkmwi=10 videosupport=yes allow=all relaxdtmf=yes rtptimeout=60 rtpholdtimeout=300 register => XXXXX:pass@sipgate.de/XXXXX ;XXXXX == sipgateid [XXXXX] type=friend insecure=very nat=yes username=XXXXX fromuser=XXXXX fromdomain=sipgate.de secret=pass host=sipgate.de qualify=yes -- Michiel van Baak michiel@vanbaak.info http://michiel.vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called users?"

...0 sip device) i can dial an asterisk extension of targetiptelusername prefixed with 3 to call the sip user registered with qu. does the target iptel username need the iptel.org domain appended to it or is it somehow 'implied' by the above ?? another wiki example suggests the use of a 'fromdomain' statement in an [iptel] section of sip.conf ?? TIA GT

Hi people, My asterisk wont register with any sip providers, I have tried three different but they all end up with: Sep 26 17:36:36 NOTICE[114696]: chan_sip.c:4035 sip_reg_timeout: Registration for 'whatever@provider.tld' timed out, trying again There is no firewall and my server has a public IP. Could this be a Asterisk problem? -Fredrik vK

...istry" from asterisk, it still says it's registered to broadvoice fine. ? However, call doesn't come through. Would you have any idea why? ?Any help will be much appreciated. Thanks Woojin Here's the excerpt from sip.conf [broadvoice] type=friend nat=no host=sip.broadvoice.com fromdomain=sip.broadvoice.com username=5083021402 fromuser=5083021402 secret=password-for-1st-BV-account dtmfmode=inband context=sip canreinvite=no insecure=very [broadvoice2] type=friend nat=no host=sip.broadvoice.com fromdomain=sip.broadvoice.com username=5083021425 fromuser=5083021425 secret=pa...

..." <103> host=dynamic nat=no allow=gsm allow=ulaw allow=alaw dtmfmode=inband quality=yes [kphone1] type=friend username=kphone1 secret=<password> callerid="Diablo" <102> host=dynamic allow=gsm qualify=yes [sip.broadvoice.com] type=peer host=proxy.dca.broadvoice.com fromdomain=sip.broadvoice.com fromuser=2129999999 secret=<password> context=incoming canreinvite=no [broadvoice-out] type=peer dtmfmode=inband host=147.135.0.128 user=2129999999 username=2129999999 authuser=2129999999 fromuser=2129999999 fromdomain=sip.broadvoice.com md5secret=<password> qualify=...

...voice.com:MyPassword2:305xxxxx66@sip.broadvoice.com [301] type=friend regexten=301 username=301 secret=xxxxx callerid="Agent #1" <301> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=305xxxxx22 secret=MyPassword1 username=305xxxxx22 insecure=very context=from-broadvoice authname=305xxxxx22 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw [sip.broadvoice.com1] type=p...

.... My question is that does 1.8.29.0 release require any more changes to be done to the sip.conf and extensions.conf to make the below work ? The sip.conf contains following enteries ================================== [100] type=friend username=100 secret=100 host=dynamic port=5060 dtmfmode=rfc2833 fromdomain=dynamic nat=no canreinvite=false context=exten-100 [101] type=friend username=101 secret=101 host=dynamic port=5060 dtmfmode=rfc2833 fromdomain=dynamic nat=no canreinvite=false context=exten-101 The extensions.conf contains ======================== [exten-100] exten => 101,1,Dial(SIP/101) ;ex...

....0 nat=yes My /etc/asterisk/sip_additional.conf contains this relevant portion: register=> nnnnnnnnnn@sip.broadvoice.com:pppppppppp:nnnnnnnnnn@sip.broadvoice.com/2 00 [from-broadvoice] username=nnnnnnnnnn user=nnnnnnnnnn type=user secret=pppppppppp nat=yes insecure=very host=sip.broadvoice.com fromdomain=sip.broadvoice.com dtmfmode=inband context=from-pstn [sip.broadvoice.com] username=nnnnnnnnnn user=nnnnnnnnnn type=peer secret=pppppppppp nat=yes insecure=very host=sip.broadvoice.com fromuser=nnnnnnnnnn fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband context=from-pstn canreinvite=no aut...

Hi Nitesh, check out my config that I have for the Faktortel config in the asterisk@home sourceforge forum, you'll probably be able to work out how to set it up from there. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Wednesday, February 23, 2005 4:12 PM To: