[Asterisk-Users] Passing DID to external number?
Hi, We run a small switchboard using Asterisk and Free PBX. We have two main extensions and two ring groups. The first ring group rings the two internal extensions. If the internal extensions do not pick up the call after 15 seconds then the second ring group kicks in which should ring the two internal extensions plus two external numbers. Firstly, how do I pass the DID number of an incoming call to the external number so that the external number sees the incoming number and not the voip dial out number? Secondly, when the second ring group kicks in only one of the external numbers dials when both internal extensions and both external numbers should ring according to the ring group setting. Any ideals what's going wrong? Kind regards Brian. UK ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Passing DID to external number?
Hi, We run a small switchboard using Asterisk and Free PBX. We have two main extensions and two ring groups. The first ring group rings the two internal extensions. If the internal extensions do not pick up the call after 15 seconds then the second ring group kicks in which should ring the two internal extensions plus two external numbers. Firstly, how do I pass the DID number of an incoming call to the external number so that the external number sees the incoming number and not the voip dial out number? Secondly, when the second ring group kicks in only one of the external numbers dials when both internal extensions and both external numbers should ring according to the ring group setting. Any ideals what's going wrong? Kind regards Brian. UK ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with 3com router, Asterisk at Home and sipgate.
Hello, We have an office [EMAIL PROTECTED] system which vwe ran through a Dlink DLS router. The Dlink has developed a fault and we are now trying to use a 3com router. With the Dlink the port forwarding was simple and worked. We routed 5004-5082 and 1-2 to the asterisk box. Things worked well for 6 months until the Dlink packed up. The 3com port forwarding initially appears terrible. We have tried forwarding the same ports and we get no way audio. We have also tried DMZ but without luck. Anyone using a 3com [EMAIL PROTECTED] and Sipgate that can help. Would really appreciate any advise. Kind regards Brian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AAH not getting IP address, likely to be network card?
Hi all, Weuse AAH to run our office telecoms registered with two Sipgate accounts. Today, Sipgate appeared to have had problems with their server with oneway audio on every call. In order to cause the Sipgate message service to pick up in stead of our AAH box, I simply unplugged the network cable. We now have problems where AAH does not seem to access the network. I plugged the network cable back in and rebooted AAH. AAH boots up, I log in as Root and AAH does not give me an IP address. I've used different cables. Everything else can access the network. Network card in the AAH box lights up green! Before I naff around changing the network card, as anyone got any useful thoughts. I think when I pulled out the cable, the card went on the blink..! AAH has been running faultlessly before..! Should have left the bloody cable at start. Regards Brian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ringing extensions in a call group.
Hi all, I've got an Asterisk at home system running the new Free PBX front. It's solved all our small office VOIP phone system which we are using as our only source of telephone communications. Anyway,I have set up a few ring groups. The first rings the internal office extensions. After 15 seconds it switches to the second ring group which rings two cell (mobile) numbers and should ring the same extensions as in the first ring group Ring group 1: extension 100, extension 101 Ring group 2: group 200, group 201, extension 100, extension 101 Ring group 200: cell number A Ring group 201: cell number B All groups are set to ringall When an incoming call arrives, extension 100 rings and after a delay of 5 seconds or so, extension 101 joins in. 10 seconds later both extensions stop and cell number A starts ringing. The problem is that Cell number B or extensions 100 and 101do not ring. I want 200, 201, Extension 100 101 all to ring together while on ring group 2. Why does asterisk not do this? Kind regards Brian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] All extensions now cannot loggin!!!!
Hi all, I was gradually getting to grips with Asterisk at Home and hadgot things workingwith a voipuser account and three extensions (2 X-lites and a Sipura 2100) then suddenly all the extensions went down. Non will login...! Have not got a clue why. Any ideas? Also, as a side issue, I set up the voipuser account by setting up a blank trunk in AMP with the name 'voipuser'. I then set an outgoing route called 'voipuser' linking to the trunk. I then inserted the voipuser blurb in 'sip_additional.conf' and put a bit in 'extensions.conf' whichgot me up and running.The question is, If I follow the usual help stuff they tell me to insert the blurb in the sip.conf and extensions.conf. When I do this I cannot connect as ther is nothing in the AMP panel for trunks and extensions. The way i've done it with the blank trunk and out going route sets up the bits in the AMP panel. Why does the many help stuff never mention about doing anything with the trunk and outgoing route and instead only tells you to edit sip and extension config files Am I missing something?? regards Brian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help needed receiving incoming calls.
Hi All,
I've got Asterisk working and am trying to configure with Sipgate. I can make out going calls. Incoming calls show up on the AMP panel with the trunk showing red. However, the call does not go to the extension.
I initally configured Asterisk by editing the config files. I have followed the various guides and have edited sip.conf and extensions.conf copy as below.
When I then configured x-lite...nothing worked. I then went into AMP setup and used the GUI to set things up. I set up an extension (not called Xlite and trunk and DID etc. I did not delete the bits added to the config files. My trunk settings are as below.
Questions...1. Why did the editing of the config files not work?2. Why did I have to go into the GUI to set it up?3. Why does the trunk show an incoming call that is not being forwarded to the extension.4. When I set up the trunk, I got a second extension showing in the extensions part of the GUI with the Extension title '92 ( sip )' with a user name of '3141217'. The second extension shows the settings I put in the incoming trunk section. Why?
Any help would be gratefully received.
Thanks
Brian.
***
sip.conf
***[general]
port = 5060bindaddr = 0.0.0.0 disallow=allallow=gsmallow=ulawallow=alawcontext = from-sip-externalcallerid = Unknownexternip=***.***.***.***localnet=192.168.0.1localmask=255.255.255.0nat=yesregister = 3141217:[EMAIL PROTECTED]/3141217
#include sip_nat.conf#include sip_custom.conf#include sip_additional.conf
[sipgate]type=friendusername=3141217secret=passwordhost=sipgate.co.ukfromuser=3141217fromdomain=sipgate.co.uknat=yesqualify=yesauthuser=3141217dtmfmode=infocontext=incomingsipgateinsecure=verycanreinvite=nodisallow=allallow=ulawallow=alaw
[xlite1]type=friendusername=xlite1callerid= Brians notebook 201host=dynamicnat=yescanreinvite=nodisallow=allallow=ulawallow=alaw
***
extensions.conf
***
have added the following to inbound context
[incomingsipgate] exten = h,1,Hangup exten = 3141217,1,Dial(SIP/internestelefon,20,tr)
[sipgate] exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) exten = _9.,2,Playback(invalid) exten = _9.,3,Hangup
***
SIP Trunk part in GUI
Outbound caller ID: 3141217Maximum channels:
Outgoing Dial Rules:Outgoing Settings
Trunk Name: brighton outgoingPEER Details:
host=sipgate.co.uksecret=passwordtype=peerusername=3141217
Incoming Settings
User Context: 3141217User Details:
callerid=3141217context=from-pstndtmfmode=infofromdomain=sipgate.co.ukhost=sipgate.co.ukinsecure=verysecret=passwordtype=useruser=3141217username=3141217
Registration String: 3141217:[EMAIL PROTECTED]/3141217
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Re: [Asterisk-Users] Configuring [EMAIL PROTECTED] for Sipgate.
Hi Arik, X-Lite is just set up as an extension for the [EMAIL PROTECTED] I've used X-Lite because it appears to be a bet better at sorting out NAT firewalls for it's self. Once I get that working I can then plugin a Sipura 2001 box. I need the Asterisk to manage two Sipgate and one voipuser lines. If you have Asterisk working with Sipgate then it may be just compairing my config files with yours. My problem is that I can make out going calls through Asterisk but not receive incomming ones. http://www.brianmccarey.com/voip/sip http://www.brianmccarey.com/voip/extensions http://www.brianmccarey.com/voip/trunk I've set up a Sip trunk. Should it have been a IAX? Regards Brian - Original Message - From: Arik Funke [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 08, 2005 1:49 AM Subject: Re: [Asterisk-Users] Configuring [EMAIL PROTECTED] for Sipgate. Hi Brian, I have Sipgate running on Asterisk (not [EMAIL PROTECTED] though). I am not sure what your problem is... You say: I have manage to make out going calls through Sipgate using X-Lite. What does this have to do with Asterisk? What does Asterisk do or not do? I think I might be able to help you out if you give me a bit more info. Best regards, Arik ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuring [EMAIL PROTECTED] for Sipgate.
Hi all, I'm new to the forum. Oh nonewbie question coming, I hear you all cry! I'm playing around with [EMAIL PROTECTED] and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've posted my config files in Adobe pdf format at http://www.brianmccarey.com/voip/sip http://www.brianmccarey.com/voip/extensions http://www.brianmccarey.com/voip/trunk I've spent at least a couple of weeks trying to sort it out and am now seeking your good advice. Asterisk pc is attached to a small network which connects to the internet via a 3COM firewall broadband router. The Asterisk has an IP on the network off DHCP and it's IP is cleared through the firewall by DMZ setting. I'm signed up with sipgate.co.uk Any advice of sorting out incomming calls would be gratefully received. Thanks Brian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Configuring [EMAIL PROTECTED] for Sipgate.
I've re-uploaded the config files in NON pdf Any help welcomed. Regards - Original Message - From: Brian McCarey To: asterisk-users@lists.digium.com Sent: Sunday, August 07, 2005 5:55 PM Subject: [Asterisk-Users] Configuring [EMAIL PROTECTED] for Sipgate. Hi all, I'm new to the forum. Oh nonewbie question coming, I hear you all cry! I'm playing around with [EMAIL PROTECTED] and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've posted my config files in at http://www.brianmccarey.com/voip/sip http://www.brianmccarey.com/voip/extensions http://www.brianmccarey.com/voip/trunk I've spent at least a couple of weeks trying to sort it out and am now seeking your good advice. Asterisk pc is attached to a small network which connects to the internet via a 3COM firewall broadband router. The Asterisk has an IP on the network off DHCP and it's IP is cleared through the firewall by DMZ setting. I'm signed up with sipgate.co.uk Any advice of sorting out incomming calls would be gratefully received. Thanks Brian. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
