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  • FreePBX: Asterisk in the Cloud (EC2) Audio Problems

    - by neezer
    Please pardon the newbie question, but I can't seem to figure this out. I followed the Voxilla's tut to the tee: http://voxilla.com/2009/10/15/voxill...p-by-step-1457 But in making calls, my softphones connect, yet no audio (in either direction). I know from poking around the forums that this is generally caused by two factors: NAT and audio codecs. I (being new to the arena), however, don't know which. I believe I have Asterisk and the clients restricted to just ulaw, and I also believe I have the correct ports open, and my externip set correctly (I think the Voxilla AMI does this automatically, since it's in the cloud). I'm a bit lost. I'd be happy to post whatever configuration files that might help, provided you tell me where they are on the filesystem. But like I said before, this is effectively a vanilla install of Voxilla's own FreePBX AMI. I'd appreciate any help or guidance here. Thanks!

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  • FreePBX: Asterisk in the Cloud (EC2) Audio Problems

    - by neezer
    Please pardon the newbie question, but I can't seem to figure this out. I followed the Voxilla's tut to the tee: http://voxilla.com/2009/10/15/voxill...p-by-step-1457 But in making calls, my softphones connect, yet no audio (in either direction). I know from poking around the forums that this is generally caused by two factors: NAT and audio codecs. I (being new to the arena), however, don't know which. I believe I have Asterisk and the clients restricted to just ulaw, and I also believe I have the correct ports open, and my externip set correctly (I think the Voxilla AMI does this automatically, since it's in the cloud). I'm a bit lost. I'd be happy to post whatever configuration files that might help, provided you tell me where they are on the filesystem. But like I said before, this is effectively a vanilla install of Voxilla's own FreePBX AMI. I'd appreciate any help or guidance here. Thanks!

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  • Can a FreePBX backup be restored to a different version?

    - by Tim Long
    I run a small PBX based on the FreePBX distro of Asterisk. The installation has been steadily upgraded but for various reasons, we want to start again on a new server with a clean install from the distribution media. Will I be able to take a backup from the old server and restore it to the new server, even though the installs are different versions? How sensitive are FreePBX backups to the build version? Is it possible to get at least a partial restore?

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  • Asterisk/FreePBX: Allow other Google Talk clients to ring when using motif module

    - by larsks
    I've recently installed FreePBX to act as a link between a SIP soft phone and my Google Talk account. It was easy to set up and outbound calls work just fine, but I've run into two problems with inbound calls that I'm not sure how to resolve. I'm using an inbound route to forward all calls from Google to my soft phone. If the soft phone is not currently registered, Asterisk answers and immediately generates a fast-busy signal (reporting CHANUNAVAIL in the logs), and the call is lost. If the soft phone is registered, Asterisk "answers" the call before rining the soft phone, which means that other Google Talk clients never ring (since from their perspective someone has answered the call). For solving (1) seems like I could use the ChanIsAvail() function (or this answer) to prevent Asterisk from answering in the event that the phone isn't registered. However, I'm not sure what to do about (2), because the behavior I want is for Asterisk to not "answer" the call until I answer the call on the soft phone. How do I configure Asterisk (ideally within the FreePBX framework) such that I can continue to receive calls at other Google Talk clients in addition to forwarding them to a SIP phone?

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  • How to display incoming trunk name at called device end with Asterisk?

    - by netmano
    We have an Asterisk with FreePBX, and using Grandstream and Panasonic VoIP phones. Now when an incoming call rings on the phones only displays "Line 1" (as it is configured at account 1 on phone) and the caller number. We would need to see the trunk name where the call comes. Please, suggest how can be achieved. I was wondering about rewriting the callerID to extend by the trunk name (like "LP - +12345").

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  • How do I configure additional phone lines asterisk/trixbox?

    - by Matt
    I have a 4 port Digium card in there, and have 4 lines running smoothly. Now, we added ANOTHER 4 port card and have 4 more analog lines coming into the Trixbox server. It still runs the 4 fine, but what do I need to do to add the additional 4 phone numbers/lines? I want it to act exactly as before, there's nothing special about the new lines. We just need more lines so that when we have 4 out of state customers call, we can have 4 more call and not get the busy signal. Trixbox CE 2.8

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  • Asterisk is hacking itself [duplicate]

    - by Shirker
    This question already has an answer here: How do I deal with a compromised server? 11 answers I've got some strange logs on my asterisk (and there a lot of extensions were tried): chan_sip.c: Failed to authenticate device 6006<sip:[email protected]>;tag=f106f3fe but IP XX.XX.XX.39 is its OWN IP! cat /etc/asterisk/* | grep 6006 returns nothing. asterisk -rv Asterisk 11.4.0 How its possible, that its hacks itself? And how could I trace, where it comes from?

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  • elastix cdr stop working

    - by dreddko
    CDR was working before 19 march. Unfortunately i dont remember what kind of changes i made to configuration, but this exactly not changes to CDR config. elastix 2.4.0 asterisk 11.7.0 mysql 5.0.95 elastix*CLI> cdr show status Call Detail Record (CDR) settings ---------------------------------- Logging: Disabled Mode: Simple /etc/asterisk/cdr.conf [general] enable=yes unanswered = yes /etc/asterisk/cdr_mysql.conf [global] hostname = localhost dbname=asteriskcdrdb password = *MYPASSWROD* user = asteriskcdruser userfield=1 ;port=3306 ;sock=/tmp/mysql.sock loguniqueid=yes mysql> SHOW GRANTS FOR 'asteriskcdruser'@'localhost'; +-----------------------------------------------------------------------------------------------+ | Grants for asteriskcdruser@localhost | +-----------------------------------------------------------------------------------------------+ | GRANT USAGE ON *.* TO 'asteriskcdruser'@'localhost' IDENTIFIED BY PASSWORD 'HASHHERE' | | GRANT ALL PRIVILEGES ON `asteriskcdrdb`.* TO 'asteriskcdruser'@'localhost' | +-----------------------------------------------------------------------------------------------+ 2 rows in set (0.00 sec)

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  • Asterisk Register username with special character like "@"

    - by Najibul Huq
    I am using a SIP provider that has provided me with a username like: [email protected] (Note this is only the username part) And has a numerical password. My Register string looks something like this: [email protected]:[email protected] But this is not working, as asterisk is only sending the first part +112223344 before the first @. My provider is adamant about having the full form of it. This is the first time I am facing this issue that is quite unusual for me. Please help.

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  • Asterisk + FreePBX + GoTalk. Inbound route not working.

    - by user289581
    I'm running asterisk 1.6.2.6 and freepbx-2.7.0 My trunk is configured as follows: Outgoing Settings Trunk name: GoTalk Peer Details: host=sip.gotalk.com username=09xxxxxx secret=YNxxxxxx type=peer fromuser=09xxxxxx fromdomain=sip.gotalk.com canreinvite=no insecure=very Incoming Settings User Context: 09xxxxx User Details: username=09xxxxx fromuser=09xxxxx type=peer secret=YNxxxxx insecure=very host=dynamic fromdomain=sip.gotalk.com context=from-pstn Register String: 09xxxxxx:[email protected]/09xxxxxx I have an inbound route called Incoming with DID 09xxxxxx diverted to local extension 200 When I do a sip trace and dial my telephone number 0741xxxxx I just get failure beeps. I never see any SIP traffic from GoTalk to my asterisk server trying to connect the call. Seems I'm not registering correctly for incoming calls because GoTalk aren't sending them to me. I am correct in using the GoTalk username 09xxxxxx as the DID, aren't I ? I've tried using my phone number but it makes no difference.

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  • Asterisk terminating outbound call when picked up, sends 'BYE' message

    - by vo
    I'm running Asterisk 1.6.1.10 / FreePBX 2.5.2.2 and I've got an outbound trunk setup. Everything use to work fine until recently (perhaps due to upgrade to FC12 or other things I'm not sure). Anyway the setup does not appear to have issues registering and setting up the call, RTP packets go both ways and you can hear the ringing from the other side. However it appears that when the call is picked up or thereabouts, the incoming RTP packets cease. Upon closer inspection with Wireshark, there are these particular packets that seem to be the cause: trunk->asterisk SIP/SD Status: 200 OK, with session description asterisk->trunk SIP Request: ACK sip:<phone>@trunk:6889 asterisk->trunk SIP Request: BYE sip:<phone>@trunk:6889 [..about a dozzen RTP packets in/outbound..] trunk->asterisk SIP Status: 200 OK, CSeq: 104 Bye [..outbound RTP continues, phone is silent..] Then the inbound RTP packets cease, however the asterisk logs dont show any activity at this point. The last entry reads 'SIP/ is answered SIP/'. Then when you hangup the extension, you get asterisk->trunk SIP Request: BYE sip:<phone>@trunk:6889 trunk->asterisk SIP Status: 481 Call Leg/Transaction does not exist My trunk peer settings in FreePBX are: username=<user> fromuser=<user> canreinvite=no type=friend secret=<pass> qualify=no [qualify yes produces 401/forbidden messages] nat=yes insecure=very host=<sip trunk gateway> fromdomain=<sip trunk gateway> disallow=all context=from-pstn allow=ulaw dtmfmode=inband Under sip_general_custom.conf i have stunaddr=stun.xten.com externrefresh=120 localnet=192.168.1.1/255.255.255.0 nat=yes Whats causing Asterisk to prematurely end the call and still think the call is in progress? I have no idea where to look next.

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  • IP telephony open source systems

    - by danke
    I'm trying to pick an IP telephony technology to learn. I heard of Asterisk, trixbox, freePBX, and my head was already spinning being not sure what to learn. Then I came across this article listing some more like Kamailio, Yate, CallWeaver, FreeSWITCH, SipXecs and now my head REALLY is spinning http://www.cio.com.au/article/323016/five_open_source_ip_telephony_projects_watch . Can someone give me a run down of how all these technologies tie together? What is the trend now, because I'd like to start learning. Note: Anyone please re-tag this question if you know better, because I'm new to this field and not sure about the best tags.

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  • Asterisk Connection not working

    - by Tamas Ionut
    I have installed Asterisk on VirtualBox by following the steps from here. Everything went ok until I got to navigate to an IP to configure Asterisk using FreePBX: 10.0.2.15 (Shouldn't be something like 192.168.x.y?? ). However, when I navigated to that url from outside of VirtualBox, that url pointed to nothing. Also I am logged in as root@localhost. Should I be logged in as root@server? I have also validated the installation as described here and everything went well. I am a complete beginner at Asterisk.

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  • asterisk incoming/miss call notification (to skype)

    - by tike
    My environment: Centos 5.6, Asterisk with freepbx , skype i.e.sends message with parameter skype.sh skype_user message. Now i wanted to send skype notification so that my asterisk server notification are sent to skype rather than email (or to both skype and email). I know, there is voicemail.conf, voicemail_general and vm_email.inc, which has these body created. vm_email.inc emailbody=${VM_NAME},\n\nThere is a new voicemail in mailbox ${VM_MAILBOX} But i dont see where is something like "mail" command. What my thought to do is: instead of saying "mailcmd" pass system ( /path/to/script) and it would simply send message as rest is already configured. Any suggestion where i could run script rather than sending email Or Executing script on every incoming call, so that i could send as notification on every call over the Skype. (however, ultimate goal is to achieve miss call notification or voice mail notification over Skype.)

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