similar to: Asterisk 1.2.13 make problem

Hi iam installing zaptel 1.2.10 on my FC5 when i make iam getting following error any one suggest me whats wrong, i have installed source also in the same server. grep: /lib/modules/2.6.15-1.2054_FC5/build/include/linux/autoconf.h: No such file or directory ZAPTELVERSION="1.2.10" build_tools/make_version_h > version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \

I'm tring to compile fax applications on Debian system. the spandsp library compiles ok, and when i try to patch the make file in apps directory as is said in the instructions it returns errors. I'm using cvs version of asterisk . -------------------------- voipgw:/usr/src/asterisk/apps# patch < Makefile.patch patching file Makefile Hunk #1 FAILED at 35. Hunk #2 FAILED at 68. 2 out of

Below is my try of installation to the latest CVS. I have not updated 3 months. What do I miss since last time? bye Ronald /usr/local/src/asterisk # make clean; make update; make install build_tools/make_version_h > include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp

Hi all I have installed 1.2.12.1 in FC5 with libpri.1.2.4 when i start iam getting the following error and it quits == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_zap.so]Oct 23 16:16:07 WARNING[11084]: loader.c:325 __load_resource: libpri.so.1.0: cannot open shared object file: No such file or directory Oct 23 16:16:07 WARNING[11084]: loader.c:554

I'm trying to install the SPANDSP software into Asterisk to support incoming (mainly) Fax. I'm following the info in http://www.voip-info.org/wiki-Asterisk+Fax. I downloaded and installed the spandsp software from ftp://ftp.opencall.org/pub/spandsp/ and followed the directions in several documents listed on the on the Tiki page. I get down to patch < Makefile.patch that fails with

Hi Every one and Lerale Erwan I have briefly describe my problem and I have provide the steps as follows: I have intalled redhat properly and from the konsole I checked with mysql. "rpm -qa | grep mysql" and the konsole provide me the message: mysql-3.23.54a-11 mysql-server-3.23.54a-11 Then I have download the asterisk and addons: By the using of : cd /usr/src export

Hi, I'm trying to compile asterisk 1.2.14 on a Debian Sarge amd64 with kernel 2.6.8-12-amd64-k8 make[2]: Entering directory `/usr/src/asterisk-1.2.14/codecs/gsm' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -fomit-frame-pointer -fPIC -c -DNeedFunctionPrototypes=1 -funroll-loops

On 15-04-09 12:06 PM, Chad Wallace wrote: >> but don't know where to put those lines. I have BABY defined as >> >channel variable: >> > >> >BABY = SIP/babytel_out >> > >> >but that seems circular, somehow. > You put them in the context for your clients... From what you show > below, I'd say they go in the "local_200"

We've experienced the same problem twice now in the past month. The asterisk pid stops responding. We aren't able to connect to the CLI and all calls are dropped. The lots are pretty bare as well. This is the message log: Oct 24 09:12:35 WARNING[20711] channel.c: Avoided initial deadlock for '0x8444a70', 10 retries! Oct 24 09:12:35 WARNING[20711] channel.c: Avoided initial

I've got the VICI predictive dialer runnning over IAXs to another asterisk server. It dials fine. I can make phone calls manually with no problem. When VICI dials a new number it rings the other end once and I get the error: Jan 26 13:53:10 NOTICE[10206]: Dropping incompatible voice frame on IAX2/VOIP3/5 of format slin since our native fo rmat has changed to gsm I have set ALLOW=ALL in

On Tue, 24 Jul 2007 wilsonk at cpsc.ucalgary.ca wrote: Hi Kelly, Sorry for the delay, I'm way behind in email :( > I built llvm-2.0 (the svn version) by just typing "./configure;make". I > then downloaded clang from the svn repository (revision 40382) and changed > to it's directory under .../llvm/tools/clang. I did a "make" without > incident for clang,

It looks as if that is more of a question/issue with your router, rather than Asterisk. I have SIP devices working on my LAN, all hardwired, and have no need to open any ports or have the router address SIP in any way My switch is not managed, and the router ports on the LAN side are all unmanaged, just a huge Ethernet "wirenut" You SHOULD be able to communicate between devices on the

Hello everyone, I am running an Enterprise 250 with Dual UltraSparc II's and 2 Gb ram running Aurora Sparc Linux 2.0 (2.6.13 kernel for Aurora is based on Fedora Core 3). GCC version 3.4.2. I built llvm-2.0 (the svn version) by just typing "./configure;make". I then downloaded clang from the svn repository (revision 40382) and changed to it's directory under

Hi, we're running Asterisk 1.2.10 and have been randomly being left blank voicemails with long messages that we can't hear. I've searched and searched but cannot find a solution. This is what happens: Internal Server runs Asterisk 1.2.10 where our mailboxes are Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are bridged between this server and our internal server.

Hi iam not able to start asterisk give me following error any help STARTING ASTERISK /usr/sbin/safe_asterisk: line 42: 4633 Illegal instruction (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} Asterisk ended with exit status 132 Asterisk exited on signal 4. Automatically restarting Asterisk. /usr/sbin/safe_asterisk: line 42: 4637 Illegal

Hi i have 20 people calling agents calling when ever they calling i get this below error May 30 00:46:57 WARNING[2840]: channel.c:785 channel_find_locked: Avoided deadlock for '0x8b2f50', 10 retries! and the voice go choppy, and voice breakages iam using Latest SVN, any suggestion to come over this problem ram -------------- next part -------------- An HTML attachment was scrubbed...

In Asterisk 1.2.13 in app/app_voicemail.c, line 4700 ext_pass_cmd is checked to decide whether to use vm_change_password or vm_change_password_shell to change a user's password for his voicemail account. I wonder, what is the difference between vm_change_password and vm_change_password_shell - what is that shell? The only reference I found on the Internet was the following bug report:

Hi iam keep getting this log in my asterisk log is this harm anything, and how can stop this, any suggestions Jun 4 18:21:47 DEBUG[2093] chan_sip.c: Stopping retransmission on '45629314783bd11604363618632f07b9@201.x.x.x' of Request 102: Match Found Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received

Hi all, I have a server with an Athlon 64 3200 and Fedora Core 2 x86_64. I have compiled and installed Asterisk 1.0.6 without any problems. When I try to make asterisk-addons-1.0.6 it say me: [root@server1 asterisk-addons-1.0.6]# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` make -C format_mp3 all make[1]: Entering directory

On Jul 31, 2008, at 4:52 AMPDT, Richard Pennington wrote: > Any code that I generate for the Sparc fails at assembly time using a > gas assembler built for the Sparc. > > I get code like the following from the code generator: > > save -96, %o6, %o6 > > and get a syntax error on the save instruction. > > I think sparc syntax should be: > > main: >