similar to: Configuring Sangoma A101D with Asterisk 1.2.18 & zaptel-1.2.17.1

Hi Guys, Anyone might know why this error keeps showing up and inbound/outbound is not working on a Bell PRI with Sangoma A101D? -- Got SABME from network peer. Sending Unnumbered Acknowledgement No calls can be made inbound/outbound. Keeps repeating. No alarms ON and no changes been made to the system. Stopped all a sudden. Asterisk CLI doesn't show anything with full verbose for both

Hi, I have a production asterisk-1.2.8 system with FreePBX & PRI Digium card. I am looking for a paging system to an external speaker. I can page to internal Polycom 501 VoIP. But, what hardware or system do I need to integrate with the asterisk to have this acheived. -- Deepak Linux your Life, Don't Window it [[]] { All for the best }

Hi, I ma using Asterisk 1.2.18 & FreePBX 2.2.1. I have assigned every users in office with Polycom with 2 extensions as below 555 8555 I have configured Follow-me to ring when the users doesn't picks the phone on line 1(555) after 10 seconds & then ring the line 2(8555). But this is not a standard telephony which I have been advised to change like below. If someone calls

Hi, We have a PRI connection & when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 & RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats

Hi, Is there any Royalty one needs to pay when using the inbuilt exisimg asterisk on hold music or when using any other mp3 from a music album. I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. -- Deepak --------------------------------- Yahoo! Answers - Get better answers from someone who

Hello List, We purchased a TE120P card from Digium and it works great. The only problem is that we are still experiencing echo on some calls. I've tried various echo cancellers (right now we are using OSLEC) and still no luck. My question has anyone gone from the TE120P to a Sangoma A101D-X Single Port T1/E1/J1 w/ echo cancellation? Have you noticed a difference? Also I called

Hello All, I have a setup of ABE on rPath linux,Sangoma A101D, and a T-1 line (Not PRI) which is all happily coexisting and all lights are green. The T-1 comes in from the world into a "Shark Box" which splits the T into 384K data and 6 channels voice. The data side is working great. The voice side, not so great. It was originally broken out to 6 pots line and Verizon came back

Hi, I have Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card installed on the box. *Its not detected.* Details are as below :- [root at asterisk ~]# lspci 00:00.0 Host bridge: ATI Technologies Inc RS480 Host Bridge (rev 01) 00:01.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:04.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:05.0 PCI bridge: ATI

Hi, I have A101DE Sangoma Card( http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a101.html ) lspci shows as 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card [root at asterisk ~]# lspci -vvv -s 03:04.0 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card Subsystem: Unknown device

Hi, I have Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card installed on the box. *Its not detected.* Details are as below :- [root at asterisk ~]# lspci 00:00.0 Host bridge: ATI Technologies Inc RS480 Host Bridge (rev 01) 00:01.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:04.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:05.0 PCI bridge: ATI

Hi, I have Asterisk-1.2.18 install with FreePBX & more than 75 extnsion, daily I come accross an issue & try resolving them its either user learning curve or my ignorance. But, I dont know what to say regarding this issue. I have my Dial Plan for internal users to have a 3 Digit Extensions. So instance my Ext is 239 & someone dials the main #, its gets the

Dear, I have an Asterisk PBX with two E1 cards: Digium TE122 and Sangoma A101D. Sangoma card has SNMP support but Digium card has not, and also SNMP does't give me ral time information. Within CLI Asterisk I execute "dahdi show channels" but I don't get information about channels usage. What is the best way to have real time monitoring of E1 channels usage and status ???

Hello Everyone. We have: Asterisk 1.4.21.2 zaptel-1.4.11 libpri-1.4.5 Sangoma A101D Connected to a PRI Cicso 7960G phones (About 30 of them) We have a problem with dropped calls that has going on for a long time. We get up to 5 dropped calls on a bad day. They all seem to be incoming calls. I have a recording of what my users report a dropped call sounds like right before it drops

We will be connecting our Asterisk server to ISDN 30 and intend using the Sangoma A101 card. The install location is in London (UK). Sangoma card at Voipon http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html?gclid=CI32vJz22I0CFQXklAodIgjHaA I would be grateful to hear if this is the right choice of card. Usage reports would be helpful. Regards Rory -- Rory Campbell-Lange

Hi Guys, Running the following with a Sangoma A101D PRI card: *Asterisk 1.4.21.2* *LibPRI version: 1.4.10* No inbound or outbound calls can be made. In fact Asterisk CLI doesn't show any activity. Problem goes away on restart of the system or maybe asterisk. I see post about upgrading Libpri to 1.4.10.2 and then I see posts that even that didn't work. Anyone can weigh in this please?

I am moving my first steps in writing and compiling C code and calling it from R. (I am also new to shared libraries...) My problem is that my C code uses a C function contained in another library (".a", not ".so" - is that the problem?). This is how I compile the file with the functions I want to call from R: cd /export/home/gpetris/ R SHLIB tryit.c cc

Hi, I was wondering if we can check the voicemails remotely from a cell or a landline number. We have SIP 3 Digit Extensions connected to Asterisk server. If users are away from Desk & need to access voicemails can they dial in to Asterisk PBX & check their messages. I know one can check through web link & even have mailed. Aslo I have checked regarding DISA, but I am not kind of

The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain web browser specially VP8 codecs so a friend of mine

Hi, I am in the process of planning a dial plan, In regards to the requirement, I am confused how to go about the dial plan. The scenario is like below. BRANCH - A - (COMPANY) Line 1 -- Extension 239 Line 2 -- Extension 8239 BRANCH - B - (COMPANY) Line 1 -- Extension 239 Line 2 -- Extension 8239 Now what I need is that if a user in Branch - A wants

Hi, I wanted to configure reporting in puppet. I came across the below links which says what needs to be added in config file to enable reports. But I was wondering how I view the visual report/graphs do I need to create my own script. http://reductivelabs.com/trac/puppet/wiki/ReportReference#rrdgraph http://reductivelabs.com/trac/puppet/wiki/ReportsAndReporting -- Deepak