similar to: Inbound routes

Hello all. I have installed AsteriskNow 1.7.1 with all updates. I'm able to make outbound calls without any problem (the external calls are made via an analog line, and the receiver see the CID). However I'm unable to forward incoming calls to the destination I want. What happens is when I make an internal call I ear a "bye". Bellow is the log of the internal call: --

Hello all. I have a asterisk implementation (asterisknow 1.7.1), with a card with 2 FXO interfaces. When I call (or receive a call) from the pstn, I ear echo. This happens if I use a softphone or IP phone, and does not happens if the call is internal. Can you help me with this issue? Best regards, Vitor Flausino

Hello. I don't know if this is a problem, but I was expecting a different behavior. Users, have to dial "0" to get an external line, and afterwords the number they want to dial (exe 12345). The thing is: 1-If user dial "012345" there is an error and the call isn't made and the error is "handle_request_invite: Call from 'XXX' to extension '012345'

I was previously using an old computer running Asterisk 1.2 with zaptel. Once the CPU fried I switch to a new computer and I chose AsteriskNow 1.5 running in 64bits to simplify the installation process. I manage to find my way with configuring dahdi instead of zaptel and to switch all my previous config to the new computer. Now everything is fine except that even if I use the md2 echo

Hello all. Can you help me find where the CDR's are being stored? The result of "cdr show status" is: Call Detail Record (CDR) settings ---------------------------------- Logging: Enabled Mode: Simple Log unanswered calls: No * Registered Backends ------------------- (none) Best regards, -vcf

Hi All, Sorry if this has been around a millions times.. I have been off this list for a few months now.. I have installed the latest asterisknow (upgraded asterisk to 1.6 as well) and I am having a hard time getting my X100P and TDM400P working.. Its all new to me with dahdi because my old server is still running Asterisk 1.0.2 so there have been lost of changes.. Can someone point me in

hello liste i have creat i trunk sip and inboun route for inbound calls the issue whe i use the DID in inboud route i have a error No DID or CID Match. but when i leave this DID field blank i can route the call without any issue how can ido in order to use DID in route inboud "i use elastix" Executing [s at from-trunk:1] NoOp("SIP/358-106-000000c0", "No DID or CID

My AsteriskNow box was updated to Centos 5.6 (2.6.18-238.5.1.el5) and DAHDI doesn't want to load. I've tried building it from the sources, but get this error message: CC [M] /root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/card_bri.o In file included from /root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/xpd.h:31, from

I updated an AsteriskNow system to 2.6.18-194.26.1.el5 with yum update. Upon reboot dahdi modules cannot be found. In yum.log I see that kmod-mISDN, kmod-dahdi-linux and kmod-dahdi-linux-fwload-vpmadt032 were all deleted during the update. I reinstalled the deleted packages but the dahdi modules can still not be found. They do seem to be under /lib/modules/2.6.18-194.17.4.el5, but no for the

Hi, I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow.

Hello, Trying to upgrade (from asterisk18-1.8.6.0-1) to the latest RPM version from Asterisk repo I found that asterisknow-version is needed by package asterisk18-core-1.8.7.0-2 How could this be explained? Best regards, Ioan ######### [root at localhost ~]# yum update asterisk18* -x asterisknow-version Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * base:

I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax machine. Both are connected to a DAHDI board. I'd like to route incoming PSTN fax calls to the extension of the fax machine and process non-fax calls through different dialplan.logic. What's the best way to go about doing this? I've looked into Fax for Asterisk, bit I'm not sure that I want it or NVFax

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

Yes I used. The problem can be the version of Asterisk? I use Asterisk 13 instead of 11. 2016-02-12 14:31 GMT-02:00, Shabbir abbasi <shabbirabbasi92 at gmail.com>: > have changed this > [dongle0] > audio=/dev/ttyUSB1 > data=/dev/ttyUSB2 > > To > > [dongle0] > imei=123456789012345 > > and imei exact same as on your device ? > > On Fri, Feb 12, 2016

Yes, I used IMEI. But in CLI appearing nothing and it not register. 2016-02-12 14:27 GMT-02:00, Shabbir abbasi <shabbirabbasi92 at gmail.com>: > have you tried imei discovery > imei=123456789012345 > > > write imei number instaed of 12345... > > On Fri, Feb 12, 2016 at 8:51 PM, Vitor Mazuco <vitor.mazuco at gmail.com> > wrote: > >> Hi! >>

I have a Asterisk server (PBX in a Flash) with Aastra 57i phones. When there is an incoming call the phone will display two buttons "answer" and "ignore". If you press "ignore" the call is dropped instead of sent to voice mail. The following is the log: -- Called 111 -- SIP/111-00001c14 is ringing -- Got SIP response 486 "Busy Here" back from

Hi ! I want to install a Telephone IP Avaya 4610SW in my Asterisk 11, but I cant install. It asks a TFTP/HTTP Server, but is necessary I install it in mu Asterisk Server for works my Telephone? The manual is here https://downloads.avaya.com/css/P8/documents/003880182 Thanks in advanced.

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

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

I think the way Digium has structured the binary packages could use a major change. I rely on the binary packages rather than compiling by source because my systems are managed by an automated CM tool (I use Ansible but those using Chef or Puppet will face the same problems) and for security reasons. I use the CentOS packages. Some issues with the current package structure are: - It installs