search for: remotesecret

Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Before continuing, this is my environment: Asterisk: 1.6.2.16.1 OS: CentOS release 5.5 (Final) 2.6.18-194.32.1.el5 Details: I have this block on sip.conf ----- start ---- ... register => john:j0nhp4ss at 66.128.XX.XXX ... [john-peer] type=peer defaultu...

The Asterisk Development Team has announced the release of Asterisk 1.8.9.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.9.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * ---

...age after the tone". sip.conf: [general] register => NPANXXZZZZ:PASSWORD at SERVICE_PROVIDER_IP registertimeout=29 registerattempts=0 defaultexpiry=60 [DID_NUMBER] type=peer context=default host=SERVICE_PROVIDER_IP authuser=DID_NUMBER fromuser=DID_NUMBER fromdomain=SERVICE_PROVIDER_REALM remotesecret=SERVICE_PROVIDER_PASSWD secret=SERVICE_PROVIDER_PASSWD dtmfmode=rfc2833 disallow=all allow=ulaw qualify=yes I am attempting just to get the starting point where I can direct users through my asterisk box, but it won't direct users to the 's' extention, only to some voicemail box. I'...

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I

...e is no official list with ip addresses of Telekom Germany. But I think all ip addresses comes from the ip range 217.0.0.0/13. I have now the following addition to sip.conf. I think it is the only safe option. Or what would you say? [telekom](!) context=from-trunk type=peer defaultuser= authuser= remotesecret= fromdomain=tel.t-online.de qualify=no dtmfmode=rfc2833 directmedia=no sendrpid=pai trustrpid=no insecure=port,invite disallow=all allow=g722 allow=alaw allow=gsm deny=0.0.0.0/0 permit=217.0.0.0/13 [DTAG-IP_IN18_016](telekom) host=217.0.18.16 [DTAG-IP_IN18_036](telekom) host=217.0.18.36 etc. &...

...t: sip:660 at PU.BL.IC.IP:5060 regserver: useragent: lastms: 0 host: dynamic type: friend context: default deny: 0.0.0.0/0.0.0.0 permit: PU.BL.IC.IP secret: NULL md5secret: NULL remotesecret: NULL transport: NULL dtmfmode: NULL directmedia: NULL nat: force_rport,comedia callgroup: NULL pickupgroup: NULL language: NULL disallow: NULL allow: NULL insecure: NULL trustrpid: NULL...

...office] ; receives calls type=friend transport=tcp dtmfmode=rfc2833 disallow=all allow=ulaw secret=password context=incoming Office: [office-going-to-home] ; places calls type=peer ;; we only call out transport=tcp dtmfmode=rfc2833 disallow=all allow=ulaw fromuser=home-coming-from-office remotesecret=password sip show peer office-going-to-home * Name : office-going-to-home Secret : <Not set> MD5Secret : <Not set> Remote Secret: <Set> ........ Status : OK (28 ms) Useragent : Asterisk PBX 10.0.0 Reg. Contact : sip:home-coming-from-...

Hi guys, I am trying to make a trunk between two asterisk system SIP Trunk on Asterisk 1.6 but I cannt make it work, can any body help me plz? Thank you

I'm using Asterisk 1.8.6.0 on my router talking to my ISP's Taqua 7000 (?) switch. I'm using a config that looks like: [sip_proxy-out] type=peer authuser=208nnnnnnnn remotesecret=xyzzy qualify=100 host=n.n.n.n call-limit=5 nat=no ; sendrpid=yes insecure=no But the Taqua responds to outbound INVITES with 403 Forbidden (oddly, not 401 or 407). Also, from what I can tell, the outbound INVITE doesn't seem to have any fields that would imply authentication (unless console...

...d](my-codecs) > > acl=acl_dtag_inbound > > type=peer > > context=from_dtag > > host=tel.t-online.de > > > > [dtag_outbound](my-codecs) > > acl=acl_dtag_outbound > > type=peer > > defaultuser=USER at t-online.de > > remotesecret=PASS > > host=tel.t-online.de > > fromdomain=tel.t-online.de > > > > The thing is, the provider's SIP server hasn't changed the IP yet. This > > morning it's still the same as yesterday. And yesterday it was the same > > the whole day. > &gt...

...riend host=dynamic context=LocalSets [MyPhoneMacAddress](home-phone) secret=MyPhonePassword [ekiga_inbound](my-codecs) acl=acl_ekiga_inbound type=peer host=ekiga.net context=from-ekiga [ekiga_outbound](my-codecs) acl=acl_ekiga_outbound type=peer host=ekiga.net defaultuser=MyEkigaUser remotesecret=MyEkigaPass fromuser=MyEkigaUser fromdomain=ekiga.net This is my dialplan: [LocalSets] exten => 101,1,Dial(SIP/MyPhoneMacAddress,30) exten => 500,1,Dial(SIP/ekiga_outbound/500,30) exten => 501,1,Set(GROUP(users)=CallsToProvider) same => n,NoOp(There are ${GROUP_COUNT(CallsToPro...

...XX](home-phone) > secret=XXXXXXXXXXXXX > > [dtag_inbound](my-codecs) > acl=acl_dtag_inbound > type=peer > context=from_dtag > host=tel.t-online.de > > [dtag_outbound](my-codecs) > acl=acl_dtag_outbound > type=peer > defaultuser=USER at t-online.de > remotesecret=PASS > host=tel.t-online.de > fromdomain=tel.t-online.de > > The thing is, the provider's SIP server hasn't changed the IP yet. This > morning it's still the same as yesterday. And yesterday it was the same > the whole day. > > Don't know why I didn't...

...vider) >> [USERNAME_PROVIDER] >> context = unauthenticated >> type = peer >> host = proxy.provider.net >> outboundproxy = proxy.provider.net >> defaultuser = USERNAME_PROVIDER >> fromuser = USERNAME_PROVIDER >> fromdomain = proxy.provider.net >> remotesecret = PASSWORD_PROVIDER >> insecure = port,invite >> t38pt_udptl = yes,redundancy,maxdatagram=400 >> t38_rtp = no >> t38_tcp = no >> canreinvite = yes >> > > For asterisk 1.6 & 1.8 you would need to set 'canreinvite=no', it may be > safer to add...

Hello, at first, thanks for helping! In the meantime, I have done a lot of research and trial and error, and I could solve that specific problem. Obviously, the dialplan application "Answer" was playing a key role here. My original dialplan snippet (which produced that problem) was: exten => _00., 1, NoOp() same => n, Set(FAXOPT(gateway)=yes) same => n,

Okay, Scott, I think we are on the wrong path. Maybe I'm wrong though. I will summarize again briefly the problems together: The peer ip address could be another than the ip address of incoming invites After an re-register the REGISTER is send to the new SIP server, answered with OK. But the peer ip address is still the old one (sip show peers). If now is a INVITE, the request is answered

Hi! I'm trying to set up a SIP trunk so that I can test calls, etc., between a new Asterisk box, and an old 1.4 box. --------------------------------------------------------------------------- New box: root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf siptrunk.conf: [box1] ; All box1 extensions; see extensions.conf type=peer context=adhearsion host=172.17.0.17 ; IP