search for: 3ddynamic

...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/20040114/9a15582...

I've got my ATA's setup with DHCP, host=3Ddynamic, and turned the registration on in the ATA All is well as long as my sip.conf is configured using the phone number like so... [1231231234] type=friend host=dynamic context=main canreinvite=no username=1231231234 secret=worknow disallow=all allow=ulaw allow=alaw It appears to me that the Cisco AT...

...t 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 command. Asterisk version 1.4.13, Debian GNU/Linux Sid (up to date); this phone = has successfully registered wit...

...ng the above, the dial string passed to the person on the other box is > SIP/${NUMBER}@a.b.c.d > > > > How can you use authentication, along with SIP, along with specifying > extension? > > > > My sip.conf has a friend defined: > > > > [priv] > > host=3Ddynamic > > secret=3Dpriv > > disallow=3Dall > > allow=3Dulaw > > canreinvite=3Dno > > nat=3Dno > > context=3Dfrom-internal\ > > type=3Dfriend > > > > I need to specify the sip channel to use the priv peer, priv secret, and > pass the extension. I...

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

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

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

...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 port range I need to open? Shou...