similar to: Dial Plan Problems

Hello, I am new on asterisk and do some tests on freepbx. I have 2 SIP provider: Provider1: In-/Out- Flatrate, only 1 Number Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers On Asterisk site i have 3 phones (branch ??, don't know how its called in asterisk) Is it possible to do something like: Phone 1: Incoming Call: Number1/Provider1 Outgoing Call:

Greetings all, I'd like to start implementing a private DUNDi peering group between one of our asterisk servers hosted at a datacentre and the various asterisk boxes sitting at clients' premises. On most of the clients' boxes the dialplan will have an [in-pstn] section containing the various numbers that should be recognised by that box. Where they're from a VoIP provider they

Greetings list, Over the last few days I've been gearing up to replace a couple of our servers with 1.6 as something of a testbed, but I'm encountering a few problems, and wondering if anyone can help... In extensions.conf, there are a number of contexts defined for each group of users, along the lines of: [groupa] [groupb] etc. In each of those, there's a command include =>

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 asterisk-users-bounces at lists.digium.com wrote on 12/14/2017 09:36:06 AM: > From: "basti" <mailinglist at unix-solution.de> > To: asterisk-users at lists.digium.com > Date: 12/14/2017 09:36 AM > Subject: Re: [asterisk-users] Rewrite Outgoing Number > Sent by: asterisk-users-bounces at

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Hi, I'm trying to mount samba shares onto the file system, but at the moment i only seem to be able to mount them as root. I've tried changing the permissions on the mount point and even mounting it on a subdir of my home directory but mount just keeps telling me that only root can do this. I've put an entry for the mount in /etc/fstab with the user option, what have i missed? I'm

Hi ? I am trying to get 2 accounts from voipblaster to talk to each other. Calls withing voipblaster network is free. If I configure two sip clients?with the two accounts it works fine however with Asterisk I am getting SIP 401 ? In my Sip.conf file I?under general ? register = user:password at sip.voipblaster.com ? then I have a sip peer ? ? [FreeCall](default) type= friend context= incoming

Hello, I'm a little confused on DTMF. A sip peer is registered on two Asterisk servers. No dtmfmode is set for them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both register on each other. A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the call is transferred to Asterisk 2: RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/12351 at

I am using GotoIf all over the place in 1.4.8 but for some reason, the following in my dial plan: ############################################################# exten => _1NXXNXXXXXX,1,GotoIf([${EXTEN} = "15554441212"]?100) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/${EXTEN},60) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/${EXTEN},60) exten => _1NXXNXXXXXX,n,Hangup exten =>

Ok ive been trying to get this to work for about half a year now. Ive searched all over the internet for a solution for my problem. Ive found some solutions, but they only led me to yet more problems. What we want to do is the following: I live in a student complex with 7 other people. Every room has its own internet connection from the same ISP. Ip, gateway, subnet are asigned through dhcp on

Hi, How can I setup Asterisk to place calls if the same dial pattern can be routed through several PRI gateways. I have one way that I tried: exten => _9737XXXX,1,Dial(SIP/${EXTEN:1}@172.17.99.5) exten => _9737XXXX,2,Dial(SIP/${EXTEN:1}@172.17.99.6) exten => _9737XXXX,3,Dial(SIP/${EXTEN:1}@172.17.99.7) exten => _9737XXXX,4,Congestion exten => _9737XXXX,102,Busy

Hi :) I hope someone has a hint concerning Early Media. The situation: My Asterisk is connected to small local carrier who works with several SIP servers. I traced some SIP headers and find something like this: Via: SIP/2.0 UDP sip1.provider1.de In the SDP part I found something like this: o=- 2268929 0 IN IP4 sip2.provider1.de c=IN IP4 sip2.provider1.de If I send

On Wed, 1 Dec 2004 12:37:13 -0800 (PST), Ben Kirkpatrick wrote: > Do you find it difficult to manage four LD providers? > Can you show me part of your LD Macro and how it's used? > > I'm toying with two LD providers now, but don't have failover setup. >Just using each one for what they are best at (least cost). > >Thanks, >--Ben Kirkpatrick > > Not

I am having problems getting incoming caller id to work on a Telstra Onramp 10. I have changed "/DEFAULT_CIDRINGS 2"/ Is there something i'm missing ? My Cisco 7960 just shows "asterisk" Thanks, Nathan [zapata.conf] context=incoming usecallingpres=yes relaxdtmf=no rxgain=0.0 txgain=0.0 busydetect=no pridialplan=local usecallerid=yes callerid=asreceived

Hi all, I've been through the wiki and the archives and I've been unable to find what I'm looking for. Basically I have a phone that I don't want to be able to dial out. I've only got one context at the moment; but from my investigations I think I might need to create another for this specific phone. Can anyone point me in the right direction? Thanks Matt

I'm wondering if anyone out there is running a community PBX for their local Asterisk User Groups or area Linux groups. I've been thinking of setting one up but am stuck as to what services to provide that people would actually find useful. I know that I could setup simple SIP->SIP to allow everyone to call each other, but that's not generally too fun. -- Kyle Sexton

Hi, Has anyone got Gossiptel working with Asterisk? - I am having real problems getting it to register - i'm just getting timeout errors. Thanks --ian

i am running MsOffice 2000 in wine-20031212 without problem. Now i tried to do the same with wine-20040408 but its not working. word stops with showing lot of errors and last with "winedbg>" Any body running office 2000 in this version. Please mail ur config file or any tweak i want to work on it. Thanks SK --

Hi I want to be able to use one provider if I dial 0 before the number and another if I dial 1 before, how can I do that in asterisk 1.6? /ralf Ralf Tr?skman, IT AdLibris AB, Box 3667, 103 59 Stockholm. Bes?ksadress: Sveav?gen 56C, 111 34, Stockholm - Obs ny address! Dir: +46-(0)8-5460 60 91, mob: +46-(0)70-7548074, vxl: +46-(0)8-5460 60 00, fax: +46-(0)8-5460 60 99 ralf at

Is there a way to override the fromdomain specified in the sip.conf and instead set the value from the dialplan? If we use: Set(CALLERID(num)=user at domain.com The SIP From header turns into: user at domain.com@10.10.10.10 We want user at domain.com, and we can't have an entry in sip.conf for every provider. -- Eric Chamberlain