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Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look at your install and they said we are having a NAT problem but didn'ttell me if it was with the asterisk conf or the Cisco ASA. _________________________________________________________________ Hotmail: Trusted email with Microsoft?s powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/

how do i troubleshoot no ring tone. It was working and all i added was the lines below now it doesn't ring. Edit sip_nat.conf for proper NAT: localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here) externrefresh=10 fromdomain=DOMAIN.com (Set your external domain name here) nat=yes qualify=yes canreinvite=no Add extra codecs to

I did a clean install to freepbx 2.6.1 and now when i do *76 i get a 1 second flash on the screen then the phone hangs up. the FOP says it is on DND but some ext are still getting calls. once i do a *76 FOP still says I am on dnd. I am running asterisk 1.6.0.21. before i was getting a message like dnd activated and dnd deactivated. i posted this on the freepbx site and here is what i got

what is the general view about the versions of the packages that are used with asterisk. lame asterisk asterisk-addons dahdi libpri i like to say on a version and not upgrade due to my experience with Linux and upgrading screwing up things. When it comes to Asterisk i have only one server build under my belt and I have had issue along the way. What do most people do with the software

i have posted this question couple of times and never really got any hits i wasn't able to provide any debug info Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309) Verbosity is at least 4 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 == Extension Changed 117[ext-local] new

here is the debug from the CLI. I think I know where the problem is I just can figure out how to fix it. The IP in the From and To i think is where the problem is. When I make an outbound call. i get the message "the call cannot be completed as dialed". if i call another ext it works. I posted the debug for both calls. ==============outbound call=========================== <---

I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? _________________________________________________________________ With Windows Live, you can

running freepbx 2.6.1 and asterisk 1.6.0.21 i did a clean install with the versions listed above. I was on freepbx 2.5.0 and asterisk 1.6.0 ( i think) I have aastra 57i phones. with the old versions i could hit the dnd(*76) button and i would hear dnd activated and the the light at the top of the phone would light up. Now i hit dnd (*79) and it hangs up after 1/2 second. the light doesn't

what is the general view about the versions of the packages that are used with asterisk. lame asterisk asterisk-addons dahdi libpri i like to say on a version and not upgrade due to my experience with Linux and upgrading screwing up things. When it comes to Asterisk i have only one server build under my belt and I have had issue along the way. What do most people do with the software

i have this card installed Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) following the steps below found on freepbx site cd /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0 make make install make config /sbin/ztcfg echo "/sbin/ztcfg" >> /etc/rc.d/rc.local cd /usr/src/libpri-1.4.10.2 make clean make make install when i run make config i do not get

here is the issue phones freepbx-2.7.0]# ./start_asterisk start STARTING ASTERISK Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. mpg123: no process killed ----------------------------------------------------- Asterisk could not start! Use 'tail

Can someone tell me how to setup a Aastra 75i phone? I have been trying to set it up and have pointed it to our asterisk server and selected http for download. What is the path? I have created two extension in asterisk for testing. I can't even get the phones to call each other. _________________________________________________________________ Lauren found her dream laptop. Find the PC that?s

I have question thats not really about astrisk but I figure you guys are doing this sort of thing. We use Aastra 6757i phones. there is some support for XML. the question is how would i go about learning to customize these phones? _________________________________________________________________ Bing brings you maps, menus, and reviews organized in one place.

I would like to have my ip phone list a menu with different status on it. for example, i could have button on the phone named status. when the button is pressed a list would display on the screen like 1:dnd, 2:break, 3:lunch, etc. we use aastra 57i phones. It looks like i could use xml some way to customize the phone. Also how can i dynamiclly pull the extension so I don't have to cusomize

i have posted this before but was unable to resolve it. i have some new info so i figured i would try again. the trace from bandwidth.com are below. they are telling me that the ip that is bold should be our ip not bandwidth.com. i have changed every setting that i can see and nothing fixes this. Where would i change this at? they cannot tell me. INVITE sip:+185993133333 at 216.82.224.202

Is there a way to tell if an extension is in use? We run a call center and it would be helpful for the people taking calls to see if we are on the phone or DND. Is that setup in Asterisk or on the phone? the phone as busy lamp field but i will just turn on after a while even if the extension is not i use. the FOP in FreePBX doesn't appear to be that helpful. i am not sure what it is supposed

After i rebuilt my server i did default install of Asterisk using the steps off freepbx site. i used these steps before without any issues. this time i have to start Asterisk manually every time the server reboots. if i start it by using ./start_asterisk script in the freepbx directory i get this from grep root 3840 0.0 0.0 4480 544 pts/1 S 12:13 0:00 /bin/sh

I have setup my asterisk box using freepbx. I can call extension and make outbound calls. the outbound calls drop between 10-30sec. we are using bandwidth.com and they have logged our call. below is your bad followed by what they say is a good call. I can't figure out where the problem is on your end. I know we are missing some stuff at the bottom but I don't know where to start.

I am sure many of you have seen my post asking question that I cannot seem to resolve. While the responses i have been getting have been helpful i still cannot seem to get this working 100%. So I have waving the white flag here. I give up. I need someone to come to my office and help me get this working. If anyone is interested the office is in Lexington KY. If someone is interested we can