similar to: ztdummy with kernel 2.6

Has anyone received this message while attempting to do a MAKE INSTALL on ZAPTEL depmod: *** Unresolved symbols in /lib/modules/2.4.26-gentoo-r9/misc/ztdummy.o I am not running DEVFS so that's not the issue. Zaptel drivers/modules all load, but ztdummy won't /lib/modules/2.4.26-gentoo-r9/misc/ztdummy.o: /lib/modules/2.4.26-gentoo-r9/misc/ztdummy.o: unresolved

I wanted to solve this myself, but it is time to admit I know nothing about the kernel Internals and modules. I'm starting in the users list, hoping someone has seen this and knows of a fix, as the developers are clearly busy and this is not critical for me. I cannot get ztdummy to load, as it complains about unknown symbols. I search The kernel headers and all of the zap source and these

Hi all, Trying to build ztdummy on an old redhat 7.3 box running kernel 2.4.20-43.7.legacysmp. Yes, I have the kernel sources installed. Yes, I set them up with make oldconfig; make dep. The build error is: make ztdummy gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/

Hi there, anyone has a suggestion about what do I need to do to solve this? Any comments appreciated. Philipp [root@asterisk]# modprobe ztdummy /lib/modules/2.4.22/misc/zaptel.o: unresolved symbol ppp_output_wakeup_Rd2dbbfe0 /lib/modules/2.4.22/misc/zaptel.o: unresolved symbol ppp_input_error_Rd2ca1e76 /lib/modules/2.4.22/misc/zaptel.o: unresolved symbol ppp_unregister_channel_Rb8cf9a74

I have searching the forums here on how to get Music On Hold working and I have been able to get * to accept a command for MusicOnHold and for Meetme after loading the ztdummy module. I used the default config for /etc/zaptel.conf since I saw no guidance on this. My problem now is that when I activate MusicOnHold, the sample music file sounds very slow and distorted. My best guess is that it is

I followed an asterisk howto at http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html and everything was setup perfectly. I logged into the voicemail, changed my greeting, but it still played the old one. I checked the proper folder in /var/spool/asterisk/voicemail/ and I found 3 gsm files that had the default messages, and the 3 wav's that I recorded. I have format=wav set in the

I have a TDM400P card all installed, and I can call out from a sip phone, but when I call from the PSTN into the asterisk machine, as soon as the Answer() gets called, the dial tone changes and is sounds like there is a lot of static on the line. Below is the part of the dial plan for answering the call. exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,Dial(Sip/pfriedel,20,tT)

Hi all, I've updated Asterisk trunk LDAP schema [0] [1] to include queues and other attributes needed for a working LDAP backend (I'll open a bug to include these changes on svn). SIP users and dialplan are perfectly working, but when I call a queue the whole Asterisk (1.6.2.0) crashes: on extconfig: [settings] sipusers => ldap,"dc=nodomain",sip sippeers =>

The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension number&password could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there

Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? So, that leaves me with the modem route, which seems more and more unlikely,

So I just saw this VoIP-centric article at slashdot (http://slashdot.org/article.pl?sid=04/05/22/1840220) which mentions e164.org. It's a "non-profit public DNS root designed to map phone numbers to Internet protocols." Is anyone on this list actually using this? They have asterisk config instructions: http://www.e164.org/config.php I wonder if someone can help me understand

Hi, I have access to two providers. On one of them the authuser is the same as the username, so outgoing works. On the other one I can only get incoming - what ever combination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as:

Hi, I've been looking at the problem of the default caller ID. When a call comes in with no CID or witheld it's always set to 'asterisk' which is what the phone displays. I've been looking for an option to change that. The only place I can find is DEFAULT_CALLERID in chan_sip.c. This is set by the 'callerid' option in the sip.conf. However the documentation

why can't we compile the asterisk coading in windows, it's done in c++ so it should work in windows as well

Dear list, We are encountering troubles with dovecot using LDAP userdbs on Debian stretch (but if I?m reading valgrind correctly, we can reproduce this with vanilla dovecot master). Minimal reproducer below. While testing an upgrade to Debian stretch (dovecot-core=1:2.2.27-3+deb9u1), auth-worker has stopped working. We are using two LDAP user databases; one which is iterable, and one which

Hi all, i've read a thread about this UPS from Peter van Valderen. He tried to develop a driver for this specific type of UPS. I've downloaded nut-2.2.2 and tried to apply the patches (lakeview.patch & lakeview.v2.patch) but both resulted in rejected chunks. I've tried to ./configure with type lakeview and then a make, but the make command fails. Does anybody has any

I have been using Sixtel from the beginning of the year and service was getting worse and worse. Yesterday I tried to access the website to get the CDR and I got an error saying that the domain no longer exists. I checked the whois and it says that the domain is on hold. Have they finally folded? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel:

Hello, I am relatively new to Asterisk and I need to compile the G.723.1 codec for Asterisk. I downloaded the ITU source code, placed it in the codecs directory, but apparently Asterisk needs a rather different library than the one provided from ITU. As I've seen in the mailing list archives, there are quite a few users who were able to compile G.723.1 in *, so, could someone kindly share it

Hi, We have two of these 4 port FXO cards. However, we are having some problems with incoming/outgoing calls. The latest version of Asterisk/zaptel from CVS is being used. Voicemail, internal SIP <-> SIP calls between Pingtel xpressa hard phones work terrific, echotest is fine, and so on. The zaptel and wcfxs modules load fine, and show all 8 FXO interfaces in dmesg:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to