Re: [Asterisk-Users] Port for Asterisk
I set an Asterisk server, what ports would I need to open for my firewall? I'm using IAX and SIP if that helps. Thanks. Read the Wiki below: http://www.voip-info.org/wiki-Asterisk+firewall+rules -Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ex-girlfriend-logic
Here is the logic that I wanted b/c my bro-n-law is in AU and uses my phone number in the US as his now for his family to contact him as an extension off my * server.
The need was defined to use something like the ex-g/f logic to route calls to his extension instead of me getting all the calls then forwarding them to him. Since his address book is much larger than I had expected or wanted to write up in extensions.conf here is how I achieved it:
I created a *db named 'route'. Basically in there I would input a CID# and an extension to route to.
database put route phone_num extension
In the extensions.conf I put in the following:
[incoming]exten = s,1,NoOp("Incoming:" ${CALLERID})exten = s,2,LookupCIDNameexten = s,3,LookupBlackListexten = s,4,DBget(exten=route/${CALLERIDNUM})exten = s,5,NoOp("Transfering to extension: " ${exten})exten = s,6,Goto(default,${exten},1)exten = s,104,Goto(2200)exten = s,105,GotoIfTime(06:00-22:30|*|*|*?default,2200,1)exten = s,106,Goto(2200)exten = s,2200,Background(press1tospeaktome)exten = s,2201,Wait(3)exten = s,2202,Voicemail(u2200)exten = s,2203,Hangup
Then the next problem was how to deal with my family and what to do with them. Since my parents are in the US and my sister is in AU, I created a menu context and send them to an extension that sends them to a menu context so they can decide to press 1 for me and 2 for my sister bro-n-law.
We have been very happy with this solution and the only draw back is that if there is no caller id number presented, I get those calls but have handled them various ways and plan to re-impliment shortly.
Also, I used the WIKI setup for the PHP and CID name lookup (LookupCIDName)to allow my bro-n-law to input all his phone numbers and point them to his extension so there wasn't any resources of me to input his address book into the *database of route and he could update on demand or his leisure.
Enjoy.
-Jeff
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Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I have it working finehere are my configs: On my NAT Firewall I have port 5060 UDP and 1-2 UDP open to my * box. In SIP.conf I have the following: [general]context=incomingport=5060bindaddr=192.168.234.111maxexpirey=180defaultexpirey=160tos=0x08nat=nosrvlookup=yesvideosupport=nodtmfmode=inbanddisallow=all ; Disallow all codecsallow=ulawlanguage=enexternip=no-ip.com_hostnamelocalnet=192.168.234.0/255.255.255.0 register=phone_number:password@sip.broadvoice.com/broadvoice [broadvoice]type=friendusername=phonenumberfromuser=phonenumbersecret=passwordhost=sip.broadvoice.commaxexpirey=15fromdomain=sip.broadvoice.comnat=yescanreinvite=noinsecure=veryqualify=yesdtmfmode=inbanddisallow=all ; Disallow all codecsallow=ulaw In the EXTENSIONS.conf I have an extension defined to forward the "broadvoice" extension to my Sipura SPA-2000 Line 1 extension. It works fine for me thru NAT. Hope this helps. -Jeff - I'm having the same issue too. Just signed up, never had a problem before with a sipura box at a friends house configured as an extension off my Asterisk box. I'm behind NAT, and my firewall has the asterisk box configured as a DMZ, also I forwarded the sip and RTP ports just to be sure. Anyone solve this one yet? I sent an email off to broadvoice but they were less than helpful. Tim Jackson [EMAIL PROTECTED] [2004-10-24 00:29:02 -0500]: I'm having the same issue, and I'm not behind NAT. Maybe this is a BV issue? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Terry Evans Sent: Saturday, October 23, 2004 4:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice I just signed up for the BroadVoice service a few hours ago, but for the life of me I can't get any incoming voice. The incoming connection is fine as it rings my extension from outside, but I can't hear anyone talking. Outgoing voice is working fine though. I've been looking through the archives, but I haven't found a solution to the problem yet. I even tried another router since someone had a problem with that, but still no dice. I've had my Asterisk server running fine for a few months, but this is the first time I've tried a VOIP service with it. I just downloaded and installed the lastest CVS and the problem is still there also. Here's some of my configuration information: sip.conf (I've tried with nat=no and it didn't help) [general] context=from-sip ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) maxexpirey=3600 defaultexpirey=120 callerid=No CallID tos=lowdelay; 0x18 ; reliabile before dtmfmode=inband srvlookup=yes ;progressinband=no nat=yes notifymimetype=text/plain [broadvoice] type=friend username=801527 (hid real number) fromuser=801527 (hid real number) secret= (hid real password) fromdomain=sip.broadvoice.com host=sip.broadvoice.com canreinvite=no dtmfmode=inband context=broadvoice-inbound nat=yes (tried nat=never also) disallow=all allow=ulaw insecure=very I have the following ports forwarded to my linux server (it's behind a NAT router): 5060, 2-21000 (from my rdp.conf file), 4445, and 4569. All of those have both TCP and UDP forwarded for now. I've tried several different combinations from different posts, including splitting the broadvoice section up into parts for incoming and outgoing, but it still didn't work. Anyone have any ideas? Let me know if traces, etc. will help and I'll capture and post some. Thanks, Terry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-3k *
I have my brother-n-law in Australia who just purchased a SPA-3k. He is wanting to connect to my * server. For the * entry I have the following: sip.conf: [2203] ; Dustintype=friendhost=dynamiccontext=defaultsecret=supersecretpasscodemaxexpirey=1800defaultexpirey=1600callerid="Dustin-Debbie" 2203mailbox=2203dtmfmode=rfc2833canreinvite=nonat=always I also have the ports opened on my firwall for 5060 TCP/UDP and 1-2 UDP pointing to my NAT'd * server. He is also NAT on his side and has the SPA-3k with Firmware 2.0.9 with settings all default except for the following: Line 1: Proxy: DynDnsAddrofMy*Host Use OutBound Proxy: No Register: Yes UserID: 2203 AuthID: supersecretpasscode And all that * is giving is errors saying the following: ct 19 07:05:55 NOTICE[6150]: Registration from 'Dustin Debbie sip:[EMAIL PROTECTED]' failed for 'HisIPAddressinAU'Oct 19 07:06:00 NOTICE[6150]: Peer '2203' is now REACHABLE!Oct 19 07:07:04 NOTICE[6150]: Peer '2203' is now UNREACHABLE! Any suggestions on where to research to get working? Thanks, Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CID troubles...
I have noticed that the data/time on the caller-id is incorrect on my Motorola 5ghz soho phone system. I go thru and reset the time on the handsets b/c there doesn't seem to be an option on the base for time. Then when I recieve a new call the caller-id is passed thru and the time gets screwed up again on *all* handsets. The configuration is incoming (PSTN/broadvoice) - * - SPA-3k. Does anyone know how the caller id parts work? Does it transfer the time with it? The time on the box is something like 20 mins off but the time that the handsets get reset to are 3-4hrs off when a caller-id is presented. Any pointers on where to go research would be great. Thanks, Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: CID troubles...
Figured it out...the SPA does a NTP call to the server running ntpd. As long as the timezone offset is up to date the caller-id will display the appropriate time and update all the handsets with the correct time. -Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
