similar to: Pri doesn't accept Zap/g2 to call

Hi Folks, After going to the paifull steps of installing AreskiCC and finally being able to access the webinterface, connecting to *, importing rates and setting up accounts I am not being able to make a CALL: No matter what number i try to dial I get the same response: The number you have dialed is currently unavailabel. Please enter thenumber you want to dial starting with 1 for local and

Hi all, I am quite disappointed at the application AreskiCC. I have installed everything following the instructions but the thing doesn't want to work. First of all, when I start the index.php page, any name/password logs in. After the login it takes me to a page with a single option "LOGOUT" We are monitoring the database and it seems like the application doesn't

Hi all, I am using a Sangoma with two PRI's. As far as land phones, the calls are fine but it refuses all cellphone calls: My configuration in Zaptel is span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,0,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-61 and on Zapata.conf: [channels] language=it context=default switchtype=national

Hi all, After finally making the web interface for AreskiCC work I am now running into new issues. 1 - In Asterisk the manager doesn't seem to connect 2 - When I try to create the file additional_areskicc_sip.conf it says "Could not open buddy file '/etc/asterisk/additional_areskicc_sip.conf' Any clues? -------------- next part -------------- An HTML

Hi List; What the following mean: CONSOLE=Phone/phone0 CONSOLE=Console/dsp TRUNK=Zap/g2 I know SIP/John and Zap/1 but I do not know above (I do not know also the difference between Zap/2 and Zap/g2)? Any advise? Regards Bilal ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids.

I need your help I have a sangoma A104D on my dell server; I got card status ok with no alarm If I dialed the extension 6210006, it shows the output as stated below, but there is no ringing from the pstn number nor the iax softphone am using on my pc. I will be glad if someone can give me a working config? What I want to achieve is to send all my call to the pstn on A104D? The pstn am

Hi, I have installed AreskiCC on Slackware 10.1 with Asterisk latest CVS and Postgres 7.4. First of all the instructions are very confusing and hard to follow if you are not an expert. But, I managed to install it and.obviously t doesn't work. The other instructions I found on wiki are a great effort but incomplete. Basically the first thing that happens is that when I load

Hi, thanks for responding. Well, that's exactly wher I am stuck. My defines.php is with username=root and password=. but it still says the password is wrong. Now, let me be more specific: Postgres is running as user postgres at the Login: postgres, ok. The one thing I am not 100% sure is if php-pgsql is installed because I installed PHP from the Slackware 10.1 disk. Other then that, I

I have made bri-stuff.0.0.2rc19 to work (I think) but I can not achieve any in-dialing nor I can dial out; this is what I have from "pri intense debug span 1" command ---------- *CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 -- Executing Playback("SIP/201-a862", "tt-weasels") in new stack -- Playing 'tt-weasels' (language

Hello, I cannot make any call with my new quadBRI card from Junghanns.net in my asterisk box. My asterisk box is built on Fedora Core 3 linux system. After compiling drivers from junghanns.net the card driver was loaded correctly with - modprobe zaptel - insmod qozap.ko ( as FC3 is running a 2.6 linux kernel) - ztcfg -vvv ( shows 12 channels correctly configured and the 4 leds on the card were

Hi, I have installed a quadBri card, with Asterisk-1.0.10 and the bristuff-0.2.0-RC8s (* 1.0.10) When calling 0207654321 the following happens: -- Executing Goto("Zap/1-1 ", " salsa-helpdesk-day|s|1 ") in new stack -- Goto (salsa-helpdesk-day,s,1) -- Executing Dial ("Zap/1-1 ", "Zap/g1/0201234567|30 ") in new stack -- Requested transfer capability:

Before you jump ahead, yes I do have chan_zap.so loaded.. Call Flow: Asterisk 1 --IAX2--> Asterisk 2 ---> PRI -- Accepting AUTHENTICATED call from 22.22.22.22: > requested format = ulaw, > requested prefs = (), > actual format = ulaw, > host prefs = (ulaw|alaw|gsm), > priority = mine -- Executing

We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller --> ( zap -> iax ) ---> ( iax -> whatever ) --> called client server often the called can't hear the caller (both machines on public ip) 'iax2 show netstats" on client machine shows more and more dropped packets on the

Hi I have this escenario: |SIP or H323 phone|---->|Cisco2600|----E1-pri---->|Asterisk|------>IVR, A2Billing, etc... The problem is that I can not hear any audio when call from 'sip or H323 phone' and configure something like: exten => _01XXXXXXX,1,Playback(thank-you-for-calling|noanswer) ... It works if I remove the 'noanswer' parameter but in this case it connects

I have a strange problem: if I put on hold an incoming call from my Digium TE110P, I can't resume it and the person at the phone continues to hear MOH until the line falls. My TE110P is connected with an italian E1 NT. If I put on hold a call on a SIP channel I can resume it without any problems. Is there someone that can help me? These are my configurations: zaptel.conf

Hello, I'm fairly new to asterisk and Trixbox, I'm setting up a Trixbox based email to fax gateway. At this time, I have a ZAP PRI link between the eFax server and my VoIPSwitch. The ZAP channels are configured, the B and D channels are up, and I have green link lights on either end of my cabling, but when I dial the number I have assigned to my eFax server, the call never seems to route

I'm having problems getting an Xorcom USB Bri 4 dialling out in Australia. I can receive calls into the system without an issue, but I can not for the life of me dial out of the system. Below are my configs, I'm hoping its something simple that I just can't see as I've been looking at it for to long. Can any one point me in the right direction. P.S. Yes it is meant to be in TE

Nik, Looks like you're making some progress. When I first started using A@H I had trouble getting the outbound dialing to work. I wasn't sure where to start, so what I did was skip the macros in the dial plan. I wanted to play around with exactly what digits the telco wanted to see. So I put a specific extension in my [default] context like this: exten =>

Problem: Zap Channels Disappear @ random intervals. (Channels have disappeared on both gateways twice this week). My Setup: I have 2 Dell 1850 Power Edge Servers - Configured as. P4 2.8Ghz 512 ECC Memory SCSI Array (2 Drives Mirror) Configuration is really simple the boxes are setup as PSTN termination gateways: (SIP/ZAP) - We are running a SER/Asterisk Hybrid. My asterisk

Hi, I'm trying to build a SIP / ISDN BRI gateway. I'm using asterisk and zaptel 1.0.6 with the bristuff patches. I have a Billion HFC card connected to a BRI ISDN line. Unfortunately each time asterisk tries to make a call on that channel, I get NOTICE[6236]: app_dial.c:759 dial_exec: Unable to create channel of type 'ZAP' == Everyone is busy/congested at this time When I try