search for: sipauth

...istration message ( with Autherization section, and that get accepted by the destination host). Why is this ? Isnt there a way to tell * to send with Autothorization message the first attempt ? Asterisk sends this first 9 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: REGISTER sip:sipauth.deltathree.com SIP/2.0 Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK6b37ba4a From: <sip:1510xxxxxxx@sipauth.deltathree.com>;tag=as3e96887d To: <sip:1510xxxxxxx@sipauth.deltathree.com> Call-ID: 1dfb33d406292f5327646eae0683e11b@66.121.81.25 CSeq: 106 REGISTER User-Agent: Asterisk PB...

...sip.conf file recently. Here is what I have: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls allow=gsm allow=ulaw allow=alaw tos=lowdelay tos=185 ;register= xxxxxxxx:xxxx@sipauth.deltathree.com ;register= xxxxxxxx:xxxx@sipauth.deltathree.com [iconnect] type=friend username=xxxxxxxx password=xxxx host=sipauth.deltathree.com ;host=213.137.73.178 canreinvite=no Has anybody experienced this before?

...the sip debug on, I can see that something is happening everytime a call is received from iconnecthere, but I get an invalid tone on the caller side. The call never rings anywhere on the asterisk. Would appreciate any help on this. Thanks Below is my sip file register=442087926805:somepassword@sipauth.deltathree.com:5060 [iconnecthere] type=friend secret=somepassword username=11232634 host=sipauth.deltathree.com canreinvite=no ;nat=yes context=default ;dtmfmode=inband disallow=all ;allow=all allow=gsm allow=ulaw allow=alaw allow=g726 allow=g723 This is the sip debug info when a call comes in f...

...response 480 "Temporarily not available" back from 213.137.73.176.. i had a look on google and aparantly if you change the codec it fixed it or something.. i tried all of them nothing seemed to work. heres a copy of my sip.conf [iconnect] type=friend username=xxxxxxxx secret=xxxx host=sipauth.deltathree.com qualify=1000 callerid="1111178197026" allow=g721.1 and for the xlite [2001] type=friend callerid="JUSTIN XLITE" <2001> username=2001 secret=2001 context=-phones mailbox=2001 host=dynamic nat=no qualify=yes allow=9721.1 ive been playing around with the na...

...is my sip.conf [general] ;calls arrive from sip lands here port=5060 context=default-in disallow=all allow=ulaw allow=alaw allow=g729 maxexpirey=180 defaultexpirey=160 ;Connect to Free World Dialup (no NAT) register=61358:xxxxxxx@fwd.pulver.com/61358 ;Connect to iconnect register=15108688610:xxxx@sipauth.deltathree.com/15108688610 canreinvite=no [iconnect] ;incoming does not land here, why ? outgoing is fine type=friend secret=xxxx username=yyyyyyyy host=sipauth.deltathree.com dtmfmode=inband ; required by iconnect context=iconnect-in canreinvite=no allow=alaw allow=ulaw allow=g729 [fwd] ;inco...

...Peer User/ANR Call ID Seq (Tx/Rx) Format 213.137.73.140 133357xx 1e7ff5217c3 00104/00000 UNKN (d) 1 active SIP channel(s) ast1000*CLI> sip show history 1e7ff5217c3dbd3d737b8ddc6ceaf087@192.168.254.250 ast1000*CLI> * SIP Call 1. TxReqRel REGISTER sip:sipauth.deltathree.com SIP/2.0 / 104 REGISTER 2. Rx SIP/2.0 100 Trying / 104 REGISTER However, I seem to be successfully registered: ast1000*CLI> sip show registry Host Username Refresh State...

...I am trying to register a SIP service(s) with FWD and ICONNECTHERE. Here is my sip.conf: ------------------------------------------------ [general] port =5060 bindaddr =0.0.0.0 context =default calledid=No CallID register => 400703:PASSWORD@fwd.pulver.com/400703 register => USERNAME:PASSWORD@sipauth.deltathree.com/400703 [fwd] type=friend username=USERNAME secret=PASSWORD host=fwd.pulver.com dtmfmode=inband [iconnect] type=friend username=USERNAME secret=PASSWORD host=sipauth.deltathree.com dtmfmode=inband ---------------------------------------------- Here is error output i get: File chan_...

...Configuration files: (commented lines represent options I tried). n= my Iconnect phone number p=Iconnect password u=Iconnect User id sip.conf [general] port = 5060 ;bindaddr = 0.0.0.0 ;bindaddr = 212.199.7.106 context = from-sip ;callerid=No CallID ;register=nnnnnnnnnnn:pppp@sipauth.deltathree.com/ nnnnnnnnnnn ;register= uuuuuuuu:pppp @natrelay.deltathree.com/phone ;register=uuuuuuuu:pppp@sipauth.deltathree.com/phone [iconnect] type=friend secret=pppp username=uuuuuuuu host=sipauth.deltathree.com ;host=natrelay.deltathree.com ;fromdomain=xxx.xxx.xxx.xxx ;dtmfmode=...

...IP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to ;bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls allow=gsm allow=ulaw allow=alaw [iconnect] type=friend username=******** password=**** host=sipauth.deltathree.com ;host=213.137.73.178 All I have been able to find about this topic is that 19 is supposed to be comfort noise (whatever that is) Any help is appreciated

...9;s the 'host' line that stops inbound from working. no matter what I do I just cant seem to get it work. the minute I add that line to the configuration everything stops working. any clues would be greatfully welcomed! ------sip.conf----------------- register = 1917NNNNNNN:SSSS:AAAAAAAA@sipauth.deltathree.com/1917NNNNNNN [iconnect-out] type=friend secret=SSSS username=AAAAAAAA host=sipauth.deltathree.com dtmfmode=inband --------------------------------

...nymized: [general] disallow=gsm allow=ulaw port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = from-sip ; Default for incoming calls register=65000:blahblahblah at inoc-dba.pch.net/65000 register=14155551212:blahblahblah at sipauth.deltathree.com/14155551212 register=18438:blahblahblah at fwd.pulver.com/18438 register=jtodd:blahblahblah at iptel.org/90764 [iconnect] type=friend secret=[blahblahblah] username=[blahblahblah] host=sipauth.deltathree.com [fwd] type=friend secret=[blahblahblah] username=18438 host=fwd.pulver.com...

Hi, I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :(( I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :( This is my configurations files: - sip.conf - [general] port = 5060

Hi ;) I'm a french newbie and i installed asterisk 1 day ago. I've got an ATA186 and a computer with Sjphone installed. If i want to call the sjphone from the ata or call the ata from de sjphone everything is ok. My problem is ,that i can't call the voicemail or any other phone number ..as 600 for exemple from the ata or the jphone. I don't know why but i looked after a long

...ile dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames WARNING[150546]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames Appropriate sections of the configuration files are below: sip.conf [iconnecthere] type=friend username=xxxxxxxx secret=xxxx host=sipauth.deltathree.com extensions.conf exten => _82.,2,Dial,SIP/${EXTEN-2}@iconnecthere The same config works fine on Free World Dialup - using 81 as the prefix, so I'm quite perplexed as to why this happens on iConnectHere. Any help or insight would be greatly appreciated. Regards, Shaun Ewin...

Hi All, I can only seem to get iconnecthere working with incoming calls intermittently. One minute it seems to work, and the next it doesn't. I am not aware of anything being changed in the config files. Outgoing calls work ok all the time. The Asterisk box is behind NAT so that does complicate things slightly. However, the Iconnecthere PCPhone client software works perfectly for

...le but for some reason my incoming is still not working. At this point I wish to use iconnecthere merely for inbound calls. Also my asterisk server is behind nat. The following is my current sip.conf file or at least the part that I see as pertinent here. [general] register=15555555555:1111@sipauth.deltathree.com/15555555555 context = sip-incoming The pertinent part of my current extensions.conf is this [sip-incoming] exten => _.,1,Goto(home,s,1) Please note that 15555555555 above has been substituted for my 11 digit iconnecthere number and 1111 above has been substituted for my secre...

Hello, Recently I am seeing this message on my asterisk console received from Iconnect. Apr 30 11:37:21 NOTICE[1125329600]: chan_sip.c:5648 handle_request: Unknown SIP command 'NOTIFY' from '213.137.73.41' It is prety annoying as it appears once every four seconds. I've seen similar posts in the archives which points me to NAT keep alives being send by the remote end. I am

Just wondering whether we have a resolution to iconnect incoming problem, which started few days ago. Cheers SW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040808/ecc99c4a/attachment.htm

I am going to have to find a fix for this problem or I'm going to have to quit using iconnect. About one call in 10 or so, iconnect's gateway gives me an error (console output appended below). So upon receiving the error, which as a 4XX error means, "Fatal," asterisk gives up and drops the call. But not iconnect!! The phone at the other end starts ringing, and rings

...ccount via SIP -------------------------- These are my peers in sip.conf: ----------------------------------------------------------- [outgoing_sip_iconnect] ; for routing calls outbound to the PSTN numbers via iconnecthere ; type=peer username=!!!My Iconnect Number!!! secret=!!!MyPassword!!! host=sipauth.deltathree.com canreinvite=no qualify=no disallow=all allow=gsm allow=ulaw allow=alaw allow=G726 [outgoing_sip_fwd] type=peer username=!!!My FWD Number!!!! secret=!!!MyPassword!!!! host=fwd.pulver.com disallow=all allow=ulaw allow=G726 ------------------------------------- And this is what I get...