Re: [Asterisk-Users] Grandstream 100 pricing question
On Wed, Jun 22, 2005 at 08:35:58AM +0200, Francesco Peeters wrote: The shop I saw these also sells - pretty cheap - little devices (forgot the name, they look like a translucent blue ice-hockey puck) that do SIP conversion for analog telephones or PBX extensions. (I am thinking migration period here: first connect one of those to each of the two PBXs as an extension, so you can use it to 'dial' in to the * server. Is this a webshop in Europe? Care to share the URL? Regards Ming-Wei ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Solaris 10
Wang Xiangzhou wrote: Sun claims that Linux apps can run on Solaris 10 natively. Is there anyone to run Asterisk on Solaris 10 and what the results are. Thanks, William why not just compile asterisk on sol10? Ming-Wei ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple asterisk process
Hong Kim wrote: I'm running * on Redhat9 with E100P and ISDN PRI. When I executed asterisk, I could see about 25 asterisk processes. Did someone experienced this? Regards, Hong __ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I only see one :) $ ps -ef |grep asterisk root 12536 1 0 Nov22 ?00:00:00 /opt/asterisk/sbin/asterisk xming 7486 7481 0 19:44 pts/000:00:00 grep asterisk $ let me guess, you are using 2.4.x kernel? In 2.4 kernel, all threads are listed ad processes Ming-Wei ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sparc hardware, Linux and X100P
Jeff Owen wrote: Hi, I'm thinking about buying a Sun Blade 100 from Ebay. I see that it has PCI slots. I want to run Gentoo Linux on it and install my X100P card. My Question is... Will the X100P card work happily with Linux on a Sparc processor? Has anyone every tried this or the TDM400 series? Thanks in Advance Chris Did you got gsm codec working? Ming-Wei ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gentoo
Stefan de Konink wrote: Ed Brady wrote: The latest portage tree has the latest release of *. However if you plan on keeping up to date with CVS head, I suggest you for-go using the portgage install, and use the source instead. Or make a portage_overlay with an asterisk_cvs ebuild :) running * (self compiled, installed in /opt) on sparc64/Gentoo ;) xming ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP in mysql and reinvite
Hi, I have have the following problem, I have configure sip fiends in mysql with MYSQL_SIP_FRIENDS, but I cannot find a way to force * to allow reinvite. canreinvite=yes in sip.conf works apparently only in subsections of friends and peers and not as a global option, Anyone has any idea how can I force reinvite? Without reinvite I don't get sound with 2 UA behind NAT en * on the Internet TIA Ming-Wei ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and rtp proxy
Hi, Has anyone got any experience with * and rtp proxy (or something else that can rewite the SDP msg)? The nat=yes in the sip.conf only concerns SIP, and apparently I have problems with the rtp streams. TIA Ming-Wei ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Sparc64
Ok, I may have spoken to early, I have * compiled and running on Sparc64/Linux, tried to configure sip softphones etc., everything works till here. Yesterday I tried to place a call to the demo but right after the call is bridged with the demo sounds it receives a SIGBUS and terminates with Bus error Any ideas how can I track this down? Ming-Wei ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Sparc64
Sunrise Ltd wrote: Ming-Wei Shih wrote: I have a login on the wiki but IMHO this does not belong to the wiki, it should be in the src. It belongs on the Wiki for as long as it takes to get it into the CVS, because that's where people will be looking for help. Once the modifications are in the CVS, the page will be obsolete and we'll remove it then. But until then it will be useful. We have already got an Asterisk Build Notes for Solaris Wiki page, referenced from the Asterisk Solaris Wiki and it has received 60 hits although it has only been up for about 48 hours. http://www.voip-info.org/tiki-index.php?page=Asterisk%20Build%20Notes%20for%20Solaris rgds benjk To all UltraLinux users, you can find my patch here http://www.voip-info.org/tiki-index.php?page=UltraLinux+Sparc64 this is a less ugly patch than the one I posted yesterday Ming-Wei ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux for Asterisk
Steve Totaro wrote:
please post the makefile hackings. i have a sparc64 gathering dust.
---
this is against CVS-NHEAD-07/28/04-15:58:08
and includes my install path in /opt
Ming-Wei
diff --recursive -u asterisk/Makefile asterisk.orig/Makefile
--- asterisk/Makefile 2004-07-28 16:03:20.0 +0200
+++ asterisk.orig/Makefile 2004-07-18 19:58:05.0 +0200
@@ -16,8 +16,7 @@
# Create OPTIONS variable
OPTIONS=
-#OSARCH=$(shell uname -s)
-OSARCH=v9
+OSARCH=$(shell uname -s)
ifeq (${OSARCH},Linux)
PROC=$(shell uname -m)
@@ -51,10 +50,10 @@
#K6OPT = -DK6OPT
#Tell gcc to optimize the asterisk's code
-OPTIMIZE=-mcpu=v9 -mtune=v9 -O2 -fomit-frame-pointer -pipe
+OPTIMIZE=-O6
#Include debug symbols in the executables (-g) and profiling info (-pg)
-DEBUG= #-pg
+DEBUG=-g #-pg
# If you are running a radio application, define RADIO_RELAX so that
the DTMF
# will be received more reliably
@@ -79,7 +78,7 @@
# Where to install asterisk after compiling
# Default - leave empty
-INSTALL_PREFIX=/opt/asterisk
+INSTALL_PREFIX=
# Staging directory
# Files are copied here temporarily during the install process
@@ -99,17 +98,17 @@
# Don't use together with -DBUSYDETECT_TONEONLY
BUSYDETECT+= #-DBUSYDETECT_COMPARE_TONE_AND_SILENCE
-ASTLIBDIR=$(INSTALL_PREFIX)/lib/
-ASTVARLIBDIR=$(INSTALL_PREFIX)/var/lib/
-ASTETCDIR=$(INSTALL_PREFIX)/etc/
-ASTSPOOLDIR=$(INSTALL_PREFIX)/var/spool/
-ASTLOGDIR=$(INSTALL_PREFIX)/var/log/
-ASTHEADERDIR=$(INSTALL_PREFIX)/include/
+ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk
+ASTVARLIBDIR=$(INSTALL_PREFIX)/var/lib/asterisk
+ASTETCDIR=$(INSTALL_PREFIX)/etc/asterisk
+ASTSPOOLDIR=$(INSTALL_PREFIX)/var/spool/asterisk
+ASTLOGDIR=$(INSTALL_PREFIX)/var/log/asterisk
+ASTHEADERDIR=$(INSTALL_PREFIX)/usr/include/asterisk
ASTCONFPATH=$(ASTETCDIR)/asterisk.conf
-ASTBINDIR=$(INSTALL_PREFIX)/bin
-ASTSBINDIR=$(INSTALL_PREFIX)/sbin
+ASTBINDIR=$(INSTALL_PREFIX)/usr/bin
+ASTSBINDIR=$(INSTALL_PREFIX)/usr/sbin
ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run
-ASTMANDIR=$(INSTALL_PREFIX)/share/man
+ASTMANDIR=$(INSTALL_PREFIX)/usr/share/man
MODULES_DIR=$(ASTLIBDIR)/modules
AGI_DIR=$(ASTVARLIBDIR)/agi-bin
@@ -117,7 +116,7 @@
INCLUDE=-Iinclude -I../include
CFLAGS=-pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations $(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU
_SOURCE #-DMAKE_VALGRIND_HAPPY
CFLAGS+=$(OPTIMIZE)
-CFLAGS+=$(shell if $(CC) -S -o /dev/null -xc /dev/null /dev/null 21;
then echo ; fi)
+CFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null
/dev/null 21; then echo -march=$(PROC); fi)
CFLAGS+=$(shell if uname -m | grep -q ppc; then echo -fsigned-char; fi)
CFLAGS+=$(shell if [ -f /usr/include/osp/osp.h ]; then echo
-DOSP_SUPPORT -I/usr/include/osp ; fi)
diff --recursive -u asterisk/codecs/gsm/Makefile
asterisk.orig/codecs/gsm/Makefile
--- asterisk/codecs/gsm/Makefile2004-07-28 16:04:54.0 +0200
+++ asterisk.orig/codecs/gsm/Makefile 2004-06-22 19:42:13.0 +0200
@@ -41,7 +41,7 @@
ifneq (${PROC},x86_64)
ifneq ($(shell uname -m),ppc)
ifneq ($(shell uname -m),alpha)
-OPTIMIZE+=-mcpu=v9 -mtune=v9 -O2 -fomit-frame-pointer -pipe
+OPTIMIZE+=-march=$(PROC)
endif
endif
endif
diff --recursive -u asterisk/codecs/ilbc/Makefile
asterisk.orig/codecs/ilbc/Makefile
--- asterisk/codecs/ilbc/Makefile 2004-07-28 16:05:53.0 +0200
+++ asterisk.orig/codecs/ilbc/Makefile 2004-06-28 22:10:28.0 +0200
@@ -2,7 +2,7 @@
ifeq (${OSARCH},Darwin)
CFLAGS+=-Wall -Werror -fPIC -O3 -funroll-loops -fomit-frame-pointer
else
-CFLAGS+=-Wall -Werror -fPIC -O3 -mcpu=v9 -mtune=v9 -O2
-fomit-frame-pointer -pipe
+CFLAGS+=-Wall -Werror -fPIC -O3 -march=$(ARCH) -funroll-loops
-fomit-frame-pointer
endif
LIB=libilbc.a
diff --recursive -u asterisk/codecs/lpc10/Makefile
asterisk.orig/codecs/lpc10/Makefile
--- asterisk/codecs/lpc10/Makefile 2004-07-28 16:06:39.0 +0200
+++ asterisk.orig/codecs/lpc10/Makefile 2004-06-22 19:42:14.0 +0200
@@ -31,7 +31,7 @@
ifneq ($(PROC),ppc)
ifneq ($(PROC),x86_64)
ifneq ($(PROC),alpha)
- CFLAGS+= -mcpu=v9 -mtune=v9 -O2 -fomit-frame-pointer -pipe
+ CFLAGS+= -march=$(PROC)
endif
endif
endif
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Sparc64
[EMAIL PROTECTED] wrote: I'd have thought this was on Sparc64 (i.e. UltraSparc III) if it's a Sun Ultra60, nought to do with Opteron. Steve Ultrasparc is the CPU/arch of SUN and Futjisu, it has nothing to do with AMD Opteron, except maybe on Linux they all have the 64-bit kernel 32-bit userland quirks. I have a login on the wiki but IMHO this does not belong to the wiki, it should be in the src. this is against CVS-NHEAD-07/28/04-15:58:08 and includes my install path in /opt please find the patch in the Best Linux for * thread Ming-Wei ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux for Asterisk
I am running * CVS head on Gentoo/i586 and Gentoo/Sparc64 (US60 2x450/1GB RAM), they are running great. On sparc64 * does not compile out-of-the-box, some hackings in the Makefiles are needed, Ming-Wei ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] question regarding Asterisk. X-Lite, and firewall
Michael Wang wrote: Hello, I have a one-way audio problem. If any one can give me a clue on how to solve it, I'd highly appreciate. My configuration is: Both Asterisk server and a SIP phone run within a LAN. Asterisk: CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp 14262. The Linux box that running Asterisk server is RedHat 2.4.18-14. Asterisk server runs on IP: 192.168.1.102. X-Lite (phone A) is on Win2K, with IP 192.168.1.100. They are both behind a router with dynamic IP address. Assume its public IP is aaa.bbb.ccc.ddd. I have another X_Lite SIP phone (phone B) that is NOT in the LAN I mentioned above. Rather, it has its own public IP address, say eee.fff.ggg.hhh. I have configured the router to forward all traffic to its port 5161 to Asterisk server's 5060 port, and configured SIP phone A to use 192.168.1.102:5060 and phone B aaa.bbb.ccc.ddd:5161 as proxy server respectively. Both phones registered successfully. Now, I used phone B to call phone A. The entire SIP hand-shake went through successfully. However, I can only get voice from phone A to phone B, not the other direction. I found that RTP traffic went from phone A - Asterisk - phone B. However, on the other direction, phone B tried to use 192.168.1.102 as destination of Asterisk to send voice too. Obviously, the IP is a private IP, hence, is not reachable. try this in your sip.conf disallow=all allow=ulaw allow=alaw nat=yes or use a STUN server Ming-Wei ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
