search for: 3dfriend

Hi all, I installed the estara softphone and had no problem registering it with asterisk. I could make calls to other hardware SIP phones (Cisco 7960) from the softphone, but I couldn't call the softphone from the Cisco 7960s. The asterisk console gave me an error message saying "unable to create channel" to my softphone. What could be the problem? I searched the archive with no

[priv] type=3Dfriend dbsecret=3Ddundi/secret context=3Dlongdistance Hope this helps, in your case Dundi will save you a world of work on configuring that many systems, in fact if you structure Dundi like spokes around a small number of master servers, the config gets real easy.Let me know how it goes. On Wed, Apr 16,...

...ip =3D 200.201.202.203 ; Address that we're going to put in = SIP messages if we're behind a NAT tos=3Dlowdelay disallow=3Dall ; Disallow all codecs allow=3Dulaw ; Allow codecs in order of preference dtmfmode=3Dinfo [grandstream1] type=3Dfriend host=3Ddynamic secret=3Dmysecret context=3Doutgoing nat=3Dyes reinvite=3Dno canreinvite=3Dno qualify=3D2000 has anyone done this before? chandra -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/...

Hi there, I have tried very hard to setup the x-lite with asterisk, but until now i didn't get sucess. When i start the asterisk in debug mode, i see the message: sip/2.0 401 unauthorized. I know that this problem with authentication. I put in my sip.conf as below. [2203] type=friend username=2203 auth=md5 secret=1234 reinvite=no canreinvite=no dissallow=all allow=gsm

...88<br />context=3Dinputinterior.se<br />dtmfmode=3Drfc= 2833<br />;h323id=3D"may day"<br />;callerid=3Dmay day<br />disallow=3Dall<= br />allow=3Dalaw:40<br />;allow=3Dalaw<br />tracelevel=3D6<br /><br />[Ava= ya]<br />type=3Dfriend<br />context=3Dinputinterior.se<br />dtmfcodec=3D127= <br />dtmfmode=3Drfc2833<br />ip=3D10.242.14.11<br />port=3D5088</span></p> <blockquote style=3D"border-left: 2px solid #1010ff; padding-left: 5px; mar= gin-left: 5px; width: 100%;"><!...

[general] port =3D 5060 ; Port to bind to bindaddr =3D 0.0.0.0 ; Address to bind to A realm=3D192.168.0.2 context =3D default ;Default for incoming calls [5549] disallow=3Dall = allow=3Dulaw allow=3Dalaw allow=3Dgsm type=3Dfriend ;(inbound and = outbound calls accepted) secret=3Dlocalphone ; obvious password for = testing host=3Ddynamic callerid=3DJason White <5549> dtmfmode=3Dauto mailbox=3D5549 ;(Asterisk VM-system's mailbox #) The output from sip set debug is attached, as captured earlier by the = script comma...

Still getting the same error. Apr 29 11:57:49 WARNING[1125329600]: chan_sip.c:503 retrans_pkt: Maximum retries exceeded on call 6b8b4567327b23c6643c986966334873@211.28.255.135 for seqno 102 (Critical Request) please advise anyone!!!!!someone!!! jai

Hi, I have configured 3 users in my iax.conf, i am using iaxcomm phones. Iaxcomm has excellent voice quality although there is no ringing tones(either ring back or ringing tone),but i can live without right now. I find that for each user i want registered i have to add his name and his ip address.I have been using "host = dynamic".Isnt there any way that i can define a dialmap such as

...<firewall-ip> D N 255.255.255.255 60646 Unmonitored tp1/tp1 <firewall-ip> D N 255.255.255.255 60649 Unmonitored Now, the Cisco phones are set to use nat (nat =3D 1) and in the SIP configuration, the phones are also configured for SIP. [tp1] type=3Dfriend secret=3Dtp1 host=3Ddynamic nat=3Dyes callerid=3D"Test Phone 1" I can make calls out over the phones, but can't get anything back in. If I call voicemail say, then that's fine. But if I try and call another phone behind the firewall, it just sits there :/ IS there a specific por...