similar to: Any idea how making Asterisk "transparent"?

Hello! I would like to setup an NTP server for my Windows network using CentOS 6.3 with firewall turned on. As I learned the NTP protocol uses port 123 UDP. I have two NIC cards. One for internal network and one for access internet. Both cards in private address range. The problem is when I am using firewall described below the client cannot access the server. No idea why. Without firewall

Hello! I would like to store ISDNCAUSE on automatic call-out campaign (possibly gives more detail on failed call). How is it possible? I have tried 'failed' and 'h' extension. No luck. Extension 'failed' does not know anything about ISDNCAUSE and 'h' extension is not called at all. Any idea? I am using Asterisk 1.2.14 on FC4 if it counts. Cheers, a

Hi all! I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers and tools but my calls aren't logged. I'd enabled mysql log and noticed that asterisk send a 'DESC cdr' so connection is working between asterisk and mysql and I am able to call other phones so Asterisk is working as well. No error messages on startup though. Any idea why is it happen? As I realized

Hi all! On outgoing call sometimes Asterisk use/give back the caller id sent back by called number instead of number called by me. This is annoying and misleading statistics if other side use some exotic number. For example I have called number 12345678 and CDR include the number 333 as callerid which was sent back by called number/set/switch/whatever. Normally it cannot be an issue but I have

Hi all! Is anyone using the $subject setup? What I would like to do the following setup: 1. OXE is setup for receiving calls, handling Agents 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI The incoming calling route: 1. OXE handles incoming calls, answer 2. Transfer to extension 9xxx 3. Asterisk answer (using one channel) 4. IVR is handling calls 5. If needed IVR

Hi all! I have the following setup: Phone lines -> traditional PBX -> Welltech 3802 -> VPN -> Asterisk -> Linksys PAP2/Welltech ATA-151 -> phone There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2) PBX extensions. Asterisk is a proxy here. Each device successfully register itself. I tried the setup above with Linksys and Welltech devices as well. I setup

Hi, after a few of upgrades, I noticed these messages in full debug log: Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring prilocaldialplan Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring overlapdial Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring

Hi all! I would like to make a service with recording sounds and playing back to caller. I had wrote the script but it failed at Read statement with file not found error. I have put some file test into script and this is what happen on verbose level 9. -- Executing [8298 at default:8] Record("DAHDI/27-1",

Hello! My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on Fedora Core 4. I am making automatic call-out campaign with this setup on 4 PRI. The scripts for this: ==================================================================== caller php script write this to outgoung folder: fwrite($outfile,"Channel: Zap/g1/$phonenumber\n"); fwrite($outfile,"MaxRetries:

Hello List! We're having trouble making call deflection on ISDN PRI. We would like to transfer a call to an external extension but keeping the callerid of the caller so it can be presented to the receiver of the transferred call. At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware TE420B. We've ordered the service (CD) from the phone company. The

Hello out there, I'm running an Asterisk 1.8.15-cert1 with DAHDI. Today I noticed that Asterisk is signalling to the calling party the current internal CallerID whenever I put a call to another internal phone. Example: Customer calls 020212345-555 -> IVR answers and puts caller to the chosen queue -> Someone picks up the phone (Internal ext. 321) -> CallerID shown on customers

Does anybody have any experience using a Patton SmartNode as a SIP/Telco gateway with Asterisk? They seem really inexpensive and appear to support all of the necessary features, but I don't have any experience with their products, so I don't know if they are any good. We are currently using a Cisco 2600 w/ PRI card and it works fine, but I was looking for someone else as a possible

Hello Does anyone have any experience with configuring BT (British Telecom) ISDN2e lines to work with Patton SmartNodes? I have a Patton SmartNode 4638, which is now connected to 3 x ISDN2e lines - and in turn connected to our internal LAN. I'm having huge issues configuring the SmartNode to successfully "see" the ISDN channels - and to be honest, I'm lost as to how to then

My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. The legacy PBX hangs up, but asterisk thinks that it is still ringing. I have added

Hello Does anyone have any experience with configuring BT (British Telecom) ISDN2e lines to work with Patton SmartNodes - and then Asterisk? I have a Patton SmartNode 4638, which is now connected to 3 x ISDN2e lines - and in turn connected to our internal LAN. I'm having huge issues configuring the SmartNode to successfully "see" the ISDN channels - and to be honest, I'm

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't

Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local

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)

Hello to all, I have a problem with an asterisk qsig. I have three machines: Nortel CS1000 --- Card Sangoma PRI ---> Asterisk QSIG ---SIP Trunk---> Asterisk I use Snom phones on Asterisk. If I call from Asterisk to Nortel, Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone, but if I call from Nortel to Asterisk, the QSIG does not send

Dear Customer Support, i connected the asterisk to a e1 interface of our hipath4000. outgoing calls from a sip peer of my asterisk to an up0 telephone which iss connected to the hipath4000 are working. If you want to dial from an up0 device to the e1 interface where asterisk is connected to, you have to use the prefix 83. But when you enter the 3rd cipher this error appears at the cli