similar to: Dahdi, PRI and all circuits are busy now

When using a PRI on my asterisk system, and when calling a busy number I often get an "All Circuits Are Busy" message. I've trouble shot it and made absolutely sure all circuits are not busy... but the number being called is in fact the issue. How do I get Asterisk to give me a busy signal instead of "All Circuits Are Busy" when calling a busy number? Is this something

Hello, I'm trying to figure out what was the return code of SIP for a call. The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to retrieve the peer name using ${CHANNEL(peername)}, I have an error message that CHANNEL does not have peername or it is not available to be used. I tried to print it with NOOP on a live channel, and also after hangup, both with the same error

Hi, We have quite a large setup working just fine most of the time. We have 60 outgoing lines on PRI and we never use all of these lines. But sometimes we get the "all circuits busy now" message, seemingly random. Sometimes we get it before the call even goes through to PSTN. Sometimes after 5 or 6 rings etc. It seems that the carrier is signalling something and asterisk always

Hello, I have *@Home 1.5 installed and all is working fine for incoming calls and sometimes outgoing calls. Installed in the box is a digium TDM04B (4xFXO Ports) setup as ZAP1 to ZAP4. I have incoming calls coming in on lines 1-4 in that order and outgoing calls prefering ZAP4 then ZAP3 then ZAP2. When i try to dial out to the PSTN from a SIP phone it sometimes works (normally after a reboot)

Hi, I've set up FreeWorldDialup on my asterisk server but when I dial the service numbers, I get message '486 Busy Here '. When I dial any other number, it says 'All Circuits are busy now'. What is the problem with my settings. I've followed all the instructions step by step. Zeeshan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everybody After setting up trunk with FWD, all I get on my Asterisk box is message saying that all circuits are busy now, try your call later. Even 612 (time) says the same thing. Why is it that and how can I fix it. Zeeshan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am not sure why I am getting this message, I have an outbound route that goes to asterisk gateway1 then asterisk gateway2 When all lines on asterisk gateway1 are full, I get the message " all our circuits are busy now" then few second later, the phone rings, going to the second route! And the call can be established, how can I get rid of this message?? thanks --------------

I have downloaded and installed Asterisk@home and I have installed X-Lite on my Windows machine and I am able to connect it to the Asterisk server. I went ahead an created an account on Broadvoice today and followed the directions on http://voip-info.org/wiki-Asterisk+settings+Broadvoice and http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but when ever I try and make a call from

hello liste i have creat i trunk sip and inboun route for inbound calls the issue whe i use the DID in inboud route i have a error No DID or CID Match. but when i leave this DID field blank i can route the call without any issue how can ido in order to use DID in route inboud "i use elastix" Executing [s at from-trunk:1] NoOp("SIP/358-106-000000c0", "No DID or CID

Dear community, Please help. I've been looking around the internet (and in this great forum) for help with DUNDi setup between servers (I'm using Elastix) and while I can get my servers to lookup extensions on each other very well, I have not been able to successfully make calls between servers. For my test environment, I have 3 servers setup for now, and these are the steps I've

Hi, I'm re-posting this; the first post was filtered because I wasn't a member of the mailing list... I have a small diff for Vorbis which replaces some loops with memcpy. This allows us to take advantage of memcpy's optimisations when copying the floating point data. Does this look OK? - Michael Index: smallft.c ===================================================================

I own a TDM2400 board, with three FXO modules and one FXS. I'am having trouble with analog sip phones, from two different equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), sometimes when I am calling someone, then I press flash, and then call someone else, both calls stay connected after I hang up. [Sep 29 07:18:06] VERBOSE[3218] logger.c: -- Called g2/16 [Sep 29

Hi. My configuration is Elastix 1.5.2-2 (asterisk 1.4.24, libpri-1.4.3-5, dahdi-2.1.0.4-7 ) and OpenVox d210e connected to telco provider (Euro ISDN). Here is my /etc/dahdi/system.conf: # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS ClockSource span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 # Span 2: TE2/0/2 "T2XXP

Hi, I have installed Elastix 1.5.2 (Barranquilla, Colombia (TELCO: METROTEL)) with a TE220P (2xE1) and TDM2400P (16FXS), openr2 is included in 1.5.2 version. The outcoming calls are ok, but with incoming call i have an error: ERROR*[*14972*]* chan_dahdi.c: Chan 2 - Protocol error. Reason = Multi Frequency Cycle Timeout, R2 State = Seize ACK Transmitted, MF state = Category Request Transmitted,

I have two servers, each connected to the PTSN via PRI. When I call from site A (951-999-9999) to site B (555-1212) and the phone at site B is on the phone, I hear the normal ring tone for about 20 seconds, then the message "all circuits are busy now. please try your call again latter" followed by the congestion tone. Instead, I want this to busy ring and then hang up without any

Here is the current dial plan section: [custom-michael] exten => _900,1,Playback(custom/extn-xfer) exten => _900,2,SayDigits(${EXTEN}) exten => _900,3,MixMonitor........... exten => _900,4,Dial(SIP/${EXTEN}|${DEFRT}) exten => _900,5,Playback(custom/extn-xfer2) exten => _900,6,Goto(custom-michael,901,4) exten => _901,1,Playback(custom/extn-xfer) exten =>

https://bugzilla.samba.org/show_bug.cgi?id=5199 Summary: Exclusion of source arg ancestor short-circuits recursion Product: rsync Version: 3.0.0 Platform: Other OS/Version: Linux Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org

Hello, I have an issue with Digium TDM 400 card series. When I try to make outgoing call (PSTN call) for example, the Zap channel could not be created and busy channel message appeared. Below is the full log : [Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Executing [s at macro- dialout-trunk:20] Dial("SIP/213-09648720", "ZAP/g1/08170709XXX|300|") in new stack [Feb

Hello, I have MeetMe rooms generated dynamically and it always have two people inside that are entered by dialplan. I wish to make in some way a timeout mechanism that after X amount of time, it will disconnect the users and kick them out of the conference. How can I do such thing ? Thanks, Ido -------------- next part -------------- An HTML attachment was scrubbed... URL: