similar to: how to configure caller id

i belong to india. i am making pbx using sangoma fxo card. i want that when ever call comes to my PSTN line i should see the no from where call is coming. so i have to configures chan_dahdi.conf according to my region. i checked dahdi.conf and in that they have mentioned for india

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm testing a telephone connected to FXS port of a Sangoma A200 card. But I'm observing that callerid is not working. The configuration that I'm using in chan_dahdi.conf is the following one: - --------------------------------------------------------------------- ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit

Greetings folks- I'm experiencing issues with a freshly installed box. When a call comes in via PRI (Sangoma AFT-A104), I see this in my logs: [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 12 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 8 (Invalid argument) [Dec 15 14:26:10]

When I restart or reboot I can not dial out. The dial() incorrectly sees dahdi/1 as busy. I call in once from a cell phone, which is successful then I can call out with out issue. Any ideas would be much appreciated. Sangoma B600de asterisk-1.6.0.9 dahdi-linux-2.1.0.4 linux-2.6.28-gentoo-r5 wanpipe-3.3.16 ###chan_dahdi.conf ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit

Hello all, I've got an intermittent issue with my asterisk set-up, and I'm pulling my hair out! Mostly everything works fine, but I get an error once every few days that sometimes (but not always) causes asterisk to stop accepting new calls and also stop responding to commands on the console (asterisk -r). The usual errors look something like this: [Feb 19 13:09:52] ERROR[18728]

Hi, building current git fails with gcc -std=gnu99 -std=gnu99 -I.. -I/usr/lib/ocaml -I../ocaml -I../src -I../src -g -O2 -fPIC -Wall -c guestfs_c_actions.c In file included from ../src/guestfs.h:84, from guestfs_c_actions.c:35: ../src/guestfs-actions.h:28: error: expected declaration specifiers or '...' before 'va_list' This is fixed by an #include

I am trying to set up a Grandstream GXV-3000 Video phone to Asterisk ver 1.2.21.1. The problem I'm having is that it can call other SIP phones, but not vice versa. Can someone tell me where is the problem? TIA! Here's part of my configurations: ---------- sip.conf ---------- ; 113 is the Grandstream phone [113] type=friend username=113 secret=secret context=default dtmfmode = rfc2833

Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because TDMoE module in asterisk is not so stable and TDMoE is stale. And It seems that RedFone doesn't not support load

Hi Everyone, We are using two Sangoma U100 (USB FXO) units connected to an Acer Aspire Revo (little PC running on Atom). The units work beautifully except for Monday :-) It maybe a conincedence or maybe the fact that Saturday/Sunday is off and something happens where one of these U100 modules goes into sleep and that's when all the 4 Dahdi channels are lost. So, I have been getting Monday

Dear Folks, [Test_Context] exten => _911.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _911.,2,Set(CALLERID(num)=xxxxxxx) exten => _911.,3,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)}) exten => _911.,4,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)}) exten => _911.,5,Set(${CALLERID}=${CALLERID(num)}) exten =>

Dear ALL I have audiocode MP -124 with configure in asterisk Endpoint configuration means every analog phone register in asterisk now thing is that i have one more SIP trunk with mediant 2000 [auodiocode-mp-124]-----[ * ]------[mediant 2000]-----E1 When i call from audiocode MP -124 phone i got this error -- Executing Dial("SIP/20-0889c4d8", "SIP/mediant/1")

Can you post some of your sip configs and your extension configs. Thanks, -gcc -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Ryan Parlee Posted At: Sunday, April 04, 2004 12:10 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Asterisk - Cisco 7960 - NAT Subject: [Asterisk-Users] Asterisk -

Dear all I have asterisk setup now what happend when i dial 4 digit number my asterisk wait for few digit why when i press # key it is dialing fast but without # wait for few number is there any configuration for dialplan I have setup asterisk with avaya system i have 5 avaya system on 5 location i use 16XX,22XX,33XX,44XX,20XX to reach avaya extentions but when

For some reason I can't get the Dial command 'C' flag to work. The calls are recorded in the CDR with the 'C' on. Does anyone have an idea? extensions.conf: exten => 114,1,Dial(SIP/114,,C) It shows in the lastapp: cdr: | 2004-10-05 13:16:02 | "112" | | 114 | intern | SIP/112-3fb6 | SIP/114-0e7a | Dial | SIP/114||c | 6 |

I am trying to setup an interactive menu where a caller hits the main menu and can then dial an extension. As far as I can tell the "Waitexten" app is failing to read 3 digits and just reading the first and then announcing that it is invalid since all extensions are 3 digits. How do I make Waitexten wait for 3 digits? I have setup the extension "100" for users to reach the

Hi list I encountered problem in using Background command. Below is my extensions.conf. [mainmenu] exten => 4,1,Wait(1) exten => 4,2,Background(thank-you-for-calling) exten => 4,3,Goto(n01|s|1) [n01] exten => s,1,NoOp(${CONTEXT}) exten => s,2,Background(thank-you-cooperation|m) exten => s,3,WaitExten() exten => s,4,Playback(digits/pound) exten => 1,1,Playback(digits/1)

Just another bit of info which might help solve this: Looking at the Asterisk log messages - I notice when I start up Asterisk, I see the error: pbx_config.c: Can't use 'next' priority on the first entry! Could I be right that its something got to do with priorities? I changed the incomingpstn context to the following eliminating the 'n' field and still the same errors were

Hi I can't for the life of me work out why this is not working. When in the campon contect if you hit a DTMF key 2 you get moved to the exten => 2 defined in the mainmenu context not the exten => 2 defined in the campon context. What is wrong? The same happens if you hit key 1. [campon] exten => _*1XXX,1,Answer exten => _*1XXX,2,SetCallerID(${CALLERIDNUM}) exten =>

Hi, I am able to dial through asterisk PBX having TE120P card to E1 card running application. Communication was established successfully Now, I want to do the reverse way out. I am using the following configurations 1)zaptel.conf span=1,1,0,ccs,hdb3,crc4 defaultzone=us bchan=1-15,17-31 dchan=16 2)zapata.conf group=1 signalling=pri_net switchtype=euroisdn context=incoming

Hi all, Running bristuffed 1.2.4 system with solely Cisco 7940/7960 phones with SIP. I'm using also op_panel 0.25 (snapshot). I'm using * queues. I want to properly implement DND via *78 and *79. I'm using op_panel's documentation RECIPE 1 solution with astdb and dnd variables and this is fine for FOP. The DND works in normal cases, since I catch it with my Macro dialsip, HOWEVER