similar to: in which asterisk version is zaptel removed?

Hi, Before I report a bug on http://bugs.digium.com, I would like to know if someone is seeing the same error message. Personally I am not using wctdm24xxp but other modules such as wcte12xp and wctdm. The latter modules load fine and are compiled with pci_register_driver as expected. The only module that seems to require the deprecated function pci_module_init is wctdm24xxp. Is this normal?

Hi, Is Asterisk "fully QSIG-compliant"? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. I am using switchtype=euroisdn and all works fine. However, it seems that Alcatel's latest firmware has dropped support for euroisdn which is really despicable. So now I need to see if I can migrate to QSIG which is supported by

I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating

Hi, I have a PRI euroisdn link between an Alcatel PBX and Asterisk. I'm having some trouble with overlap dialing. Suppose I dial '874053' from an Alcatel extension ('7034') where '87' is an Alcatel prefix of type "ARS Prof.Trg Grp Seiz.with overlap". I'm expecting Asterisk to receive '1004053' (where '100' is a prefix which always shows

Zaptel seems to compile fine until I enter xpp/utils and make there. I get: xpp/utils # make cc -o print_modes -g -Wall print_modes.c print_modes.c: In function `main': print_modes.c:9: error: `fxo_modes' undeclared (first use in this function) print_modes.c:9: error: (Each undeclared identifier is reported only once print_modes.c:9: error: for each function it appears in.)

Hi, Until now I've been using my Digium B410P cards with misdn 1.0.x. I would like to upgrade my systems and am now wondering which is the "best" route to take: - use the latest release of misdn v1 - upgrade to the latest "stable" kernel and use the built-in misdn v2 - use misdn v2 as a seperate package (disable misdn in the kernel) - use dahdi's support for misdn

Hi, I've noticed that a small but meaningful quota of calls from my Alcatel PBX to Asterisk are failing. This does not always happen and it is not easily reproducible but on high traffic I do get a large number of cases. Example: Alcatel PBX extension 7085 calls Asterisk PBX extension 6145 over a PRI E1 link. I see this in the Asterisk log: Dec 14 14:10:31 VERBOSE[11378] logger.c: --

Sorry for top-posting but my webmail forces me to. I added -W to the APPEND line as suggested but I'm still getting the same result: Booting... Altiris, inc. X86PC PreBoot, PXE-2.x Enhanced Build ID=402 PXEPreZero: Invalid PXE Server list format. and the client PC freezes right there. Here's the full content of my dhcp.conf: max-lease-time 86400; ddns-update-style interim;

Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri ____________________________________________________________________________________ Moody friends. Drama queens. Your

Hi, Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to

I realize this may be slightly off-topic but I'm wondering if someone here can lend me a hand. One of my GXW4008 has gone "unconfigurable" via standard HTTP (refuses connection) and I can't use the built-in IVR because I had previously disabled the "keypad update" feature. So I'm stuck with just telnet, the reset button and RS-232. Telnet commands are very limited

Hi, I'm experiencing "frequent" kernel panics on a system with Asterisk 1.4.26.3. There is no core dump, "just" a kernel panic. This is the only data I could copy from the screen: EIP: 0060: [<f8e248b4>] Tainted: P VLI EFLAGS: 00210297 (2.6.23-gentoo-r8 #1) eax: 00000130 ebx: 00000000 ecx: 00220028 edx: 00000978 esi: 346e5802 edi: 00000000 ebp: c3b45500 esp:

Hi, I have a Linux server at ip address 10.215.144.7 running DHCP, TFTP and syslinux. DHCP config contains the following: next-server 10.215.144.7; filename "/pxe/syslinux/pxelinux.0"; and the 'default' pxelinux.cfg contains: LABEL altiris ??? MENU LABEL ^7. Altiris ??? COM32 pxechn.c32 ??? APPEND 10.215.144.60::/BStrap/x86pc/BStrap.0 When a PXE client boots in my network

Hi. Is it possible to override the standard DB function in Asterisk? My dialplan contains a lot of calls to "Set(DB(...))" and "${DB(...)} which of course use astdb to store/read data. I would like to stop using astdb and switch to Clustered MySQL (I don't suppose "clustered astdb" exists?). So instead of rewriting extensions.conf and replacing the DB calls with

I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced

Hi. I am trying to pass a variable from one Asterisk PBX to another. I'm using DUNDi with IAX2. Is there a way to do it? I tried the following but it fails. On peer1: [dundi-outgoing] switch => DUNDI/priv exten => s,1,Set(CDR(userfield)=test) exten => s,2,Set(DUNDIVAR=${ARG1}#TEST) exten => s,3,NoOp(Passing ${DUNDIVAR} to DUNDi peer.) exten => s,4,Goto(${DUNDIVAR},1) On

Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to

Hi all, I am having trouble setting the signalling method for the B410P using DAHDI. Asterisk complains that it has never heard of 'bri_cpe' or 'bri_net' - but it doesn't mind having 'pri_cpe' etc. ERROR[4294]: chan_dahdi.c:11327 process_dahdi: Unknown signalling method 'bri_net' Dahdi - dahdi-linux-complete-2.1.0.4+2.1.0.2 Asterisk - 1.4.23.1 Libpri - 1.4.9

Hi, I'm trying to find a way to implement the following: I have 1 media source (IceS or MPD) and 1 Icecast stream (say, LAN radio). Once in a while I'd like this stream to be interrupted by short announcements (PA system). Input for these announcements can be from another source (IceS, MPD, Asterisk call). Anyway, to make things simple: I'd have dir1 with ogg music files for

Hi, I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall). One trunk is SIP and the other IAX2. Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a "third" noise overlapping with a "scratchy sound" as if it were some kind of