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...my sip.conf allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes videosupport=yes disallow=all allow=ilbc allow=gsm I also had 2 extensions (me at 250 and 998 is my SPA-3102) and my users.conf goes below: [general] fullname=New User userbase=6000 hasvoicemail=yes vmsecret=1234 hassip=yes hasiax=yes hasmanager=no callwaiting=yes threewaycalling=yes callwaitingcallerid=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes callgroup=1 pickupgroup=1 host=dynamic localextenlength=0 allow_aliasextns=no allow_an_extns=no hasagent=no hasdirectory=no [250] callwaiting=yes cid_number= con...

...Advice on any more correct approach would be appreciated, but is not the focus of this post: Users.conf ;several handsets setup like this... [6001] callwaiting = yes context = numberplan-custom-1 email = jsmall at visinet.com.au fullname = Joshua Small hasagent = yes hasdirectory = yes hasiax = no hasmanager = no hassip = yes hasvoicemail = no host = dynamic mailbox = 6001 secret = XXXXX threewaycalling = yes registeriax = no registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 vmsecret = 1234 ;some PSTNS [trunk_2] callerid = asreceived context = DID_trunk_...

...Advice on any more correct approach would be appreciated, but is not the focus of this post: Users.conf ;several handsets setup like this... [6001] callwaiting = yes context = numberplan-custom-1 email = jsmall at visinet.com.au fullname = Joshua Small hasagent = yes hasdirectory = yes hasiax = no hasmanager = no hassip = yes hasvoicemail = no host = dynamic mailbox = 6001 secret = XXXXX threewaycalling = yes registeriax = no registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 vmsecret = 1234 ;some PSTNS [trunk_2] callerid = asreceived context = DID_trunk_...

...dial the numbers set in the "default" context. Please, could anyone help me resolve this. Thanks in advance. This is a part of users.conf [951XXXXXX] callwaiting = yes cid_number = 951XXXXXX context = numberplan-custom-1 email = fullname = New User group = hasagent = no hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 951XXXXXX secret = 000000 threewaycalling = yes vmsecret = 1234 zapchan = registeriax = no registersip = yes

...onds to the SIP INVITE challenge from the Sip Provider. Her eis my users.conf entry for Asterisk registration to the Sip Provider. (I know I don't have T38 as allowed codecs, not sure what to add for T38) [trunk_66] ;register allow = ulaw dialformat = ${EXTEN:1} canreinvite = no hasexten = no hasiax = no hassip = yes host = provider.com insecure = very port = 5060 registeriax = no registersip = yes trunkname = abc username = abc disallow = gsm,g726,alaw contact = abc secret = abc Any ideas appreciated. Thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http:/...

...xfax/${EXTEN},30,r) exten => _X.,n,Hangup exten => _X.,n(dialanika),Verbose(2,Outgoing using pbxanika) exten => _X.,n(dialanika),Dial(SIP/pbxanika/${EXTEN},30,r) exten => _X.,n,Hangup And here my users.conf: [00493511111111] fullname = luca secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = myproxy host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup= pickupgroup= dial=SIP/0049351111...

...configuration: The asterisk version is 1.4.10 ----------------------------------------------------------------------------- In users.conf I defined a user 6002: [6002] fullname = Test Agent email = test.agent at agent.com secret = 1234 zapchan = 1 hasvoicemail = yes vmsecret = 1234 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = international host=dynamic ----------------------------------------------------------------------------- In agents.conf I added the agent agent => 6002,1234,Test Agent ------------------------------------------------------------------...

...an receive incoming call fine, Here is a copy of relevant parts of the configs and other info Trunk Info [trunk_1] disallow = allow = all callerid = 028012xxxx contact = context = DID_trunk_1 dialformat = ${EXTEN:1} fromdomain = iinetphone.iinet.net.au fromuser = 028012xxxx group = hasexten = no hasiax = no hassip = yes host = sip.nsw.iinet.net.au insecure = very port = 5060 provider = registeriax = no registersip = yes secret = xxxxxxxx trunkname = Custom - iinet trunkstyle = customvoip username = 028012xxxx The dialplan, Just dial 0, then number, then strip the first 0 and dial [numberplan-c...

...gt; _X.,n(dialanika),Verbose(2,Outgoing using pbxanika) > exten => _X.,n(dialanika),Dial(SIP/pbxanika/${EXTEN},30,r) > exten => _X.,n,Hangup > > And here my users.conf: > > [00493511111111] > fullname = luca > secret = MYSECRET > dahdichan = 1 > hassip = yes > hasiax = no > hash323 = no > hasmanager = no > callwaiting = no > context = myproxy > host = dynamic > dtmfmode=rfc2833 > canreinvite=no > sendrpid=pai > type=friend > nat=force_rport,comedia > qualify=yes > qualifyfreq=60 > transport=Auto > avpf=no > force_avp...

...= DialPlan1 include = default include = parkedcalls [timebasedrules] *******part of extensions.conf that was added by asterisk-gui (svn)******* *******part of users.conf that was added by asterisk-gui (svn)******* [trunk_1] allow = all context = DID_trunk_1 dialformat = ${EXTEN:1} hasexten = no hasiax = yes hassip = no host = iax2.fwdnet.net port = 4569 registeriax = yes registersip = no secret = rycort4e trunkname = Custom - fwd trunkstyle = customvoip username = 788694 [6000] callwaiting = yes cid_number = 6000 fullname = proton hasagent = yes hasdirectory = no hasiax = no hasmanager = no has...

...[May 29 19:42:13] NOTICE[2526]: chan_sip.c:20163 handle_request_invite: Call from '00493511111111' to extension '00493513333333' rejected because extension not found. users.conf on Ubuntu-PBX: [00493511111111] fullname = 00493511111111 secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = default host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup= pickupgroup= dial=SIP/0049351111...

...tional signalling = pri_cpe channel => 1-23 context = default group = 63 --- /etc/asterisk/users.conf (asterisk 1.4.22 w/ SVN-branch-2.0-r4657 GUI) --- [span_1] group = 1 hasexten = no switchtype = national signalling = pri_cpe trunkname = Span 1 trunkstyle = digital ; GUI metadata hassip = no hasiax = no context = DID_span_1 zapchan = 1-23 --- /etc/asterisk/users.conf (asterisk 1.4.24[0-1] w/ SVN-branch-2.0-r4661 GUI) --- group = 1 hasexten = no switchtype = national signalling = pri_cpe trunkname = Span 1 trunkstyle = digital ; GUI metadata hassip = no hasiax = no context = DID_span_1 zapch...

...hdi: Ignoring any changes to 'vmsecret' (on reload) at line 31. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hassip' (on reload) at line 35. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hasiax' (on reload) at line 39. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hasmanager' (on reload) at line 47. [Mar 23 14:01:10] WARNING[4315]: sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel a...

...*1 ; One Touch Record a.k.a. Touch Monitor atxfer => *2 ; Attended transfer parkcall => #72 ; Park call (one step parking) --users.conf-- [6003] callwaiting = no context = from-internal fullname = IT Support hasagent = no hasdirectory = no hasiax = yes hasmanager = no hassip = yes hasvoicemail = no host = dynamic mailbox = 6003 threewaycalling = no vmsecret = 1234 registeriax = yes registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 disallow = all allow = all [dtrr] ;bi-directional trunk to 2nd asterisk system. allow = ulaw,alaw...

...ith same host = provider.sip.com value) and when as INVITE is challenged, the Asterisk does match the correct trunk and seems to send out correct Auth credentials...but not the one below.. [trunk_1] ;register to SP allow = ulaw ;context = test dialformat = ${EXTEN:1} canreinvite = no hasexten = no hasiax = no hassip = yes host = provider.sip.com insecure = very port = 5060 registeriax = no registersip = yes trunkname = test trunkstyle = customvoip username = 3035551122 disallow = gsm,g726,alaw contact = 3035551122 secret = xxxxx -------------- next part -------------- An HTML attachment was scrubb...

...t disallow errors, now i dont receive any errors but when i turn on sip debugging it just say unauthorized. below is my users.conf file [6058] allow = all callwaiting = yes cid_number = 6058 context = default email = user@mydomain.com fullname = Bob smith group = hasagent = yes hasdirectory = yes hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host=dynamic mailbox = 6058 secret = myPassword threewaycalling = yes vmsecret = 1337 zapchan = registeriax = no registersip = yes Any idea why this doesn't work but registering normal SIP configured accounts does? -------------- next part...

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

...t;</o:p></p> <p class=3DMsoNormal>fullname =3D Joshua Small<o:p></o:p></p> <p class=3DMsoNormal>hasagent =3D yes<o:p></o:p></p> <p class=3DMsoNormal>hasdirectory =3D yes<o:p></o:p></p> <p class=3DMsoNormal>hasiax =3D no<o:p></o:p></p> <p class=3DMsoNormal>hasmanager =3D no<o:p></o:p></p> <p class=3DMsoNormal>hassip =3D yes<o:p></o:p></p> <p class=3DMsoNormal>hasvoicemail =3D no<o:p></o:p></p> <p class=3DMsoNorma...

...group 'asterisk' == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf == Found Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) # tail /var/log/asterisk/messages [Jan 27 08:17:40] WARNING[23451] chan_dahdi.c: Ignoring hasiax at line 39. [Jan 27 08:17:40] WARNING[23451] chan_dahdi.c: Ignoring hasmanager at line 47. [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: G.729 transcoding module version 34, Copyright (C) 1999-2007 Digium, Inc. [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: This module is supplied under a commerci...

.... It breaks when a call goes out on a Queue, because it seems to add each phone to the group, which breaks my GotoIf() statement. Here's some relevant information: Users.conf (added by Asterisk-GUI) [trunk_2] provider = Bandwidth (SIP) ; GUI metadata context = DID_trunk_2 hasexten = no hasiax = no hassip = yes host = 216.82.224.202 registeriax = no registersip = no usecallerid = yes nat = no ;Testing trunkname = Bandwidth.com (Sip) ; GUI metadata username = secret = disallow = all allow = ulaw,alaw,g726 sip.conf [general] context = frombandwidth ;other variables, etc....