similar to: Routing inbound call to correct sip trunk

O.K., so AsteriskNow 1.4.26.2, FreePBX 2.6.0RC2.1 We have a Digium TE205P connected to a single span if ISDN PRI. The Telco has assigned us two local numbers to test incoming calls. I created an inbound route for one of those DID's and assigned it to one of our extensions. Sounds simple enough. Too simple, apparently, when I dial the number the caller gets a recording that it's a

Hi, I have a problem with dialing status. I'm using Asterisk 1.6 and a patton 4554 gateway for ISDN calls. When I call fixed telephone (not mobile phone) after few ringing the status change to "answer" but the phone is still ringing, so if I hangup before someone really answer, the call is reported as answered but it isn't. This gives me problem for call charge. Some I idea

I am currently using AsteriskNow v2. What I am trying to accomplish is having all calls from an area code go directly to the person responsible for that area. While searching for a solution for this I did come across a post that had a few examples. So Josh at extension 1902 would receive all calls from the 808 area code. exten => s,1,GotoIf($${CALLERIDNUM:0:3}" = "808?1902|1)

Hi Can anybody help me on this ? I am using Asterisknow 1.5.0-Beta(Freepbx) I am having a problem getting the sip trunks to register. It makes no different which provider one is using. Trunk name: callcentric Peer Details: context=from-pstn fromdomain=callcentric.com fromuser=1777xxxxxxx host=callcentric.com insecure=very secret=pasword type=peer username=1777xxxxxxx Register String:

Hi, we have and Asterisk server connected to a Patton Smartnode 4638 with 4 BRI. We configured 4 SIP account on Patton (1001, 1002, 1003, 1004). The system is fully functional, but we have a problem to recognize incoming calls from Asterisk: when a call come from SIP/1001 (BRI 1 on Patton) or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call coming from SIP/1004. I have contacted Patton

Hello all. I have installed AsteriskNow 1.7.1-64bits with freePBX. The system has 1 DAHDi card with 2 analog FXO ports (to pstn) and 1 FXS port connected to a FAX machine. I want the every call received on port FXO-2 to be redirected to the FAX machine. So, what I configured was that every call with DID 12345 (let's suppose that 12345 is the DID connected to the FXO-2)to be redirected to the

Sending this to asterisk-users, in case anyone has AsteriskNOW experience they can share. Joe ---------- Forwarded message ---------- From: Joe Wood <schmoe at gmail.com> Date: Wed, Aug 18, 2010 at 9:22 AM Subject: AsteriskNow REGISTER'ing s@ extension for all inbound trunks To: asterisknow at lists.digium.com Hello. Can someone tell me why AsteriskNow is reverting to registering

Hi, I currently have some systems on AsteriskNOW 1.7 & have been happy with its clean simplicity & reliability. Are many people here using AsteriskNOW 2.0.x? How do you feel about it? Did Digium stick with their previous philosophy of keeping everything very vanilla & making it clean & simple for someone who understands how to manage CentOS, FreePBX, tftp, ntpd, etc. but

Hello, Is it possible to set things such as parts of config files are edited though AsteriskNOW GUI while other parts remain "hand editable" ? AsteriskNOW website include screenshots but not much information (such as user manual) beside that. This thing was the one that kept us from using freepbx (let me say I don't mean it's not possible with freepbx : I mean we couldn't

I have a patton 1400 setup to handle the bri interface. As a trixbox user, I wanted a sip trunk rather than having to re-compile bri support into trixbix. Anyway, I have it working now so that asterisk can make calls and they are passed properly to the telephone network. Incoming calls however are another matter. I have (after turning on cli debug in the 1400) determined that its getting stuck

Hi, In my lab, I'm testing BRI spans in NT/ptmp mode. My setup is: asterisk 11.2.1 libpri 1.4.14 dahdi 2.6.1 wctdm24xxp (HA8 hybrid with B400M) SIP phone <----> Asterisk with HA8 <----> Patton SN4638 <----> Asterisk <----> SIP phone The single BRI line I'm testing remains down: CLI> pri show spans PRI span 1/0: In Alarm, Up, Active I'm quite certain this

connecting legacy PBX to Asterisk (for the very same reason, those PBX use TE-PTMP). If others could join this thread and say if they agree or not with NT-PTMP being the 2nd most needed mode, would be great. Please, do not hesitate to comment. > > > Right now, I would not preclude the possibility that NT-PTMP support > might be added, but I could not give you a concrete time at which

Hi all We use pci/pci-e BRI cards in our installations. Due to echo problems (that was before Oslec and others), we quickly switched to cards with hardware-based EC. So we use exclusively Digium B410p cards that provide 64ms tail EC. For several reasons we'd like to switch to external BRI gateways like the Patton smartnodes (the price is getting really close to a B410p). I'm

Hello all. As part of a project I'm working on to migrate from Asterisk 1.0.11.1 to the latest LTS version, I'm looking into providing a HA/failover solution for the new Asterisk installation I'll be deploying. It would appear my best bet would be to use the R850 Digium appliance. Does anyone have any experience running one of these devices along with AsteriskNOW? I ask because

Hello all. I was hoping someone out there might have some advice or suggestions regarding an upgrade from an archaic Asterisk version. I've been given the daunting task of upgrading a very old Asterisk-1.0.x install to a recent LTS version. I'll also need the install to have high-availability and failover support. From my research, it would appear that AsteriskNOW-3.0 might be my

Hi, I'm trying to configure a 4638 to pass inbound and outbound to and from ISDN and SIP interfaces. I'm using web interface at the moment. Setup is: ISDN -- <BRI> -- Patton 4638 -- <SIP> Asterisk -- <SIP> -- <IP Phone> I can call from IP phone but can't receive any incoming call : I can't see any SIP message coming in when a call comes in. Previously,

Hi, Changing the line bellow helped to get incoming calls but I add to remove secret= option in sip.conf (otherwise, Patton wouldn't respond to 407 Auth required challenges). If someone could enable secret and still get incoming calls (in any SmartWare 5.X), please, do not hesitate to share here ... interface sip IF-ASTERISK bind context sip-gateway ASTERISK route call dest-table

Hi, 1. Has someone met any success at all, connecting a Digium B410P to a Patton Smartnode 4638 (with latest 5.3 firmware) ? 2. If positive, then, which signalling was used on both sides ? My project's goal is use a Patton Smartnode 4638 to act as telco BRI lines, from a B410P-enabled asterisk box. In my testings, I can see channel is respectively up (with patton web server status page)

Hi, My setup is: IPPhone1 --- Asterisk1 with B410P ---- Patton 4638 --- Asterisk2 --- IPPhone2 I want to evaluate Asterisk1 in TE/PtmP mode. So, Patton box is configured in NT/PtmP (with 3 BRI links between both systems). Anyway, asterisk -rx "pri show spans" keeps replying : PRI span 1/0: Provisioned, Down, Active PRI span 2/0: Provisioned, Down, Active PRI span 3/0: Provisioned,