similar to: ISDN Error Code 42

Hi, I was trying to use 2 of asterisk servers and interconnected, one of them as a peer to other sever (configured in sip.conf), so all the calls to server 1 will just be passed to server 2 (has PRI Card, TE 412P, only one PRI connected), i was sending calls to server 1 and that would send to server 2 and then dial out using Dahdi, but the problem that i got was the hangup cause codes, i was not

Well, it works for me .. YMMV. Yesterday I had a problem where I had a meridian talking to * via a PRI card, and from * to the pstn via an isdn30 link. The problem was that if the number was bad, or engaged then the meridian line simply dropped, not giving the operator any indication of what occurred. With much help from this list, I managed to construct a dialplan which solved our issues.

Hi Nitesh, Take a look at this http://www.microappliances.com/site/html/index.php?section=Products&page =clienthowto.php I've never implemented it though so I would appreciate some feedback on if it works. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Saturday,

Keywords T1 Q.931 isdn disconnect cause codes itu standard libpri Dont know if anyone wondered what q.931 cause codes are but i wishwe could get these back into the dial plan as a var Standard Q931 Codes Decimal Value Hexadecimal Value Definition 1 01 Unallocated (unassigned) number. This number is not in the routing table or it has no path across the ISDN cloud (network). 1. Check routing

Hello All, Recently I added some Nokia N95 customers and it worked pretty good. Now the customers are complaining about the dialing rules... They are used to dialing +12486543210 and +4479XXXXXX for long distance calls. Is there anyway to create a "+" sign dial plan which will allow them to dial a number with "+" sign. Cheers, Nitesh

Hi Nitesh, check out my config that I have for the Faktortel config in the asterisk@home sourceforge forum, you'll probably be able to work out how to set it up from there. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Wednesday, February 23, 2005 4:12 PM To:

Hi, I am using Asterisk: 1.6.1.20 LibPRI: 1.4.11.4 DAHDI: 2.3.0.1 Echo Canceller: MG2 Wanpipe-Driver: 3.5.15 Sangoma-Firmware: 43 (A104d) I handle some calls with my own PHP-AGI-Script. After a dial-command I use "GET FULL VARIABLE ${answeredtime}" or "GET FULL VARIABLE ${dialstatus}" and get valid information. Sometimes "dialstatus" has the value

Hey everyone, we are using several TE410 cards in a production environment that are connected to several operators PRI's and it works great.. For one of the operators we have seen some strange problems in cdr mismatches however. Our cdr's show phonecalls that are disconnected at a certain timelimit while the operators cdr's show the user has disconnected a lot earlier. I thought

Hi, I've been debuging the call disconnection problem in our architecture: PSTN---E1---OldPBX---E1---Asterisk This is our problem: -SIP user agent "A" calls a pstn phone "B". -"B" hangs up the call. -SIP user agent "A" starts listenning busytones... But the call still on. (and being payed). - Call only ends when it is correctly hanged up in the

Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since

Hello All, I manage to configure Asterisk RealTime and now it loads the SIP users/peers from MySQL DB. The table I am using is of A2Billing DB "cc_sip_buddies". Now the only problem I am facing is incoming calls are failing... The ATA which is assigned this DID number is behind NAT and according to Olle's explanations he said "*there's no support for NAT keep-alives

Hello All, I would like to upgrade my recently installed Asterisk 1.2.21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh

I am wanting to use the ISDN cause code on an Asterisk 1.6 server to determine the status of a call attempt, where the call might not actually connect. Reason is I am checking for valid telephone numbers from a list of numbers, and I would like to know if the call has answered and cleared which I can by writing the hangupcause variable, but where I get an out of order network message, or number

Hello All, I got my TDM04B card installed and configured. Everything works fine I can receive calls and route to appropriate extensions. The only problem I am facing is Slowness. When I dial the PSTN number which is connected to Zap 1-1 after two ring it answers and then run the AGI script. What I did was assign it to a specific extension. So all inbound call on that PSTN number should

Hello, I'm using Asterisk with an ISDN30e PRI line (only 16 channels active). Every now and then I get a CONGESTION error even-though there are only 1 or 2 channels in use out of the 16 at that time. When this happens, the user just needs to re-dial and the call goes through OK. On a SNOM phone when the problem occurs, a "Service Unavailable 907" error is shown. [2008-10-14

Hello All, While looking for solution to solve my Callback DTMF problem, I came across LumenVox Speech Recognition software. Has anyone tried out? Need some feedback before I purchase it... Please help... Cheers, Nitesh

We've got a EuroISDN (32 channels) with a TE405p, running cvs head as of 5 days ago. In the past couple of days, we've hit a scenario where incoming calls to the * pbx from the PSTN are being marked as busy, but outgoing calls work just fine. When we reboot *, the problem goes away. Has anyone else had this ? I've attached a PRI debug below. I've changed the phone numbers (x

Hi all, I have a "Cologne Chip Designs GmbH ISDN network controller" and I want to terminate voip calls via this ISDN card. My question is: How I must to wire the ISDN equipment with my ISDN card? With normal cable or crossover? How I can to check if ISDN card is linked with ISDN equipment? In this moment I have 1:1 cable between ISDN's, the mISDN is installed and "misdn

Hi All, First of all, please no flames about the footer of this message. It's being inserted into all outgoing email by the server here without asking me personally first once I send out email. Since I've been very interested in setting up an Asterisk PBX + IVR + Voicemail system to support a small business and a foundation for quite some time now, in order to replace their current ISDN

> Hi Gareth Blades & Doug, > > Thanks so much for for the feedback. I have searched on lot of documents > but couldn't able to find clear answer regarding it. > > I hope you guys replies are very much help all in aterisk community. > > > Thanks & Regards, > > Vidura Senadeera, > > Network Engineer, > > Debug Solutions > > Sri Lanka .