Re: [Asterisk-Users] asterisk with Dialogic BRI /2VFD
Hi Tom, thx for the answer... --- Tom [EMAIL PROTECTED] wrote: richard Coco wrote: Hi all, i have an Asterisk box with an Eicon 4BRI with chan_capi-cm and every thing works fine. We now plan to install a new Asterisk using a Dialogic BRI/2VFD. Is the Dialogic card supported and can i use chan_capi-cm? Has anyone managed to install this card? Unfortunately i was unable to find documentation about Asterisk with Dialogic? thx in advance for your input!!! Richard, i think the only dialogic cards that with work are the jct models. then i think you need to buy drivers from digium hope this helps Tom -- This message has been scanned for viruses and dangerous content and is believed to be clean. Thank You For Choosing Cache Communications ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk connected with CAPI
Hi Patrick,
i've absolutly no idea what these magic lines do but
it WORKS!!! ;-)))
so thx for you input...
--- Patrick [EMAIL PROTECTED] wrote:
On Fri, 2005-11-04 at 03:55 -0800, richard Coco
wrote:
Hi all,
i'm trying to install a EICON DIVA 4BRI (on CentOS
4.1
2.6.9-22.0.1.EL) using latest package from
sourceforge (chan_capi-cm-0.6.tar.gz).
I have installed divactrl_2.1.tar.gz and untared
protocols_all.tar.bz2 in /usr/share/eicon.
---
lsmod gives me the following...
Module Size Used by
divacapi 157937 0
capi 18177 0
capifs 5961 2 capi
kernelcapi 44641 2 divacapi,capi
md5 4033 1
ipv6 234881 12
lp 12077 0
autofs423237 0
i2c_dev11329 0
i2c_core 22081 1 i2c_dev
sunrpc159269 1
microcode 6881 0
button 6481 0
battery 8901 0
ac 4805 0
uhci_hcd 31065 0
parport_pc 24577 0
parport37129 2 lp,parport_pc
divas 76345 0
divadidd 13081 2 divacapi,divas
e100 41793 0
mii 4673 1 e100
floppy 58481 0
dm_snapshot16901 0
dm_zero 2369 0
dm_mirror 27825 0
ext3 116809 2
jbd71385 1 ext3
dm_mod 56661 6
dm_snapshot,dm_zero,dm_mirror
---
Starting divactrl
---
[EMAIL PROTECTED] asterisk]# divactrl load -c 1 -f
ETSI
Start adapter Nr:1 - 'Diva Server 4BRI-8M 2.0
PCI',
SN: 7113 ... OK
[EMAIL PROTECTED] asterisk]#
but the /var/log/asterisk/messages gives me
following
errors when i try to start asterisk:
Nov 4 12:25:45 WARNING[2658]: CAPI not installed,
CAPI disabled!
Nov 4 12:25:45 WARNING[2658]: chan_capi.so:
load_module failed, returning -1
Nov 4 12:25:45 WARNING[2658]: Loading module
chan_capi.so failed!
Is CAPI really not installed or have i forgotten
something? Here my capi.conf and modules.conf
;
; CAPI config
;
;
; general section
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[EICON]
controller=1,2,3,4
isdnmode=msn
incomingmsn=*
softdtmf=on
relaxdtmf=on
accountcode=
context=incoming
echocancel=yes
devices=2
group=1
;
; Asterisk configuration file
;
; Module Loader configuration file
;
[modules]
autoload=yes
noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
noload = app_intercom.so
load = chan_modem.so
load = res_musiconhold.so
load = chan_capi.so
noload = chan_alsa.so
[global]
chan_modem.so=yes
chan_capi.so=yes
thx in advance
Iirc CentOS 4.1 uses udev so you have to add the
proper udev rules so
the capi devices are properly created Stick the
following lines in a
file called e.g. 10-capi.rules and add it to
/etc/udev/rules.d:
SYSFS{dev}=68:0,NAME=capi20
SYSFS{dev}=191:[0-9]*,NAME=capi/%n
Use tabs between the , and NAME!
Once you have done this as root do udevstart. Unload
all the capi
modules and load them again. With capiinfo you can
check if it all went
well (it should give output). If it doesn't work
then reboot the box.
Regards,
Patrick
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Re: [Asterisk-Users] IP Phone with Standard Power Ethernet
Hi, OptiPoint 4x0/600 supports IEEE 802.3af. --- chris gamble [EMAIL PROTECTED] wrote: I am looking at phones for my asterisk system and seem to have a problem. The only Power over Ethernet phones I can find that support the IEEE standard are 3com. Cisco uses its own proprietary ( and is expensive to boot ), snom has a different but equally non-IEEE method, and i'm havent found another phone that I'm confident can do the job for our office. Whats a good high quality ip phone that uses IEEE power over ethernet -- or is there a problem with IEEE power over ethernet?? ___ 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 __ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html ___ 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] Help on installing h323
Hi, install oh323... see link for installation http://lists.digium.com/pipermail/asterisk-users/2005-April/100061.html --- craz sead [EMAIL PROTECTED] wrote: Hi all could somebody help me how to install and setup H323 i would like to connect asterisk box with huawei/cisco, but i still dont understand about installing h323 on asterisk thaks __ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail ___ 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 Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com ___ 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] 4 port BRI options ?
Hi, Eicon Diva 4BRI Card and chan_capi. --- Brett, Gary [EMAIL PROTECTED] wrote: Hi there I am in the UK.. and am running latest asterisk on FC1 (2.4 kernel). I would like to know what the best option is for a 4 port BRI card. I notice Digium don't provide one.. I have heard the Junghanns do one...but are there others ?? Is the Junghanns card reliable/stable with good sound quality ?? I notice it is very expensive in a per port comparison with the Digium cards hence why I am also looking for alternative cards Your experiences would be greatly appreciated Gary ___ 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] 4 port BRI options ?
--- Brett, Gary [EMAIL PROTECTED] wrote: Is the Eicon that much better ? sorry, i have only experience with Eicon... maybe someone else is able to give a feedback... __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] SIP Soft Video phone for Asterisk usage
--- Ronald Wiplinger [EMAIL PROTECTED] wrote: Nardis Dome wrote: in your sip.conf: [general] videosupport=yes ; That helped a lot in your eyeBeam settings- try to enable all the h.263 codec. hope it helps.. However, I am still not there. I have installed eyeBeam on 612 and 617. While 612 gets the video of 617, 617 sees 612 as a picture, like a big spreadsheet with dots in each cell. Absolutely no picture to recognize. 612 sees itself clear. What could be still wrong? I have enabled all codecs on both Xten. In asterisk it has the same settings (realtime shows the same record - of course user and password is different) Sometimes i have the same probleme. I have to restart eyeBeam or reboot my PC... __ Discover Yahoo! Use Yahoo! to plan a weekend, have fun online and more. Check it out! http://discover.yahoo.com/ ___ 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] SIP Soft Video phone for Asterisk usage
Hi, did you enable the right video-codecs in eyeBeam? settings-media-video-Advanced-Codecs --- Ronald Wiplinger [EMAIL PROTECTED] wrote: Nardis Dome wrote: try eyeBeam, it works fine for me... [] type=friend secret= auth=md5 callerid=myCallerId canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex allow=gsm allow=h261 allow=h263 Thanks, I bought eyeBeam for two computers on the LAN for testing, but I get with above settings on both screens: Remote party does not support video What do I miss? bye Ronald ___ 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] SIP Soft Video phone for Asterisk usage
in your sip.conf: [general] videosupport=yes ; in your eyeBeam settings- try to enable all the h.263 codec. hope it helps... --- Ronald Wiplinger [EMAIL PROTECTED] wrote: Nardis Dome wrote: Hi, did you enable the right video-codecs in eyeBeam? settings-media-video-Advanced-Codecs I have here 1. H.263++QCIF 128 2. H.263+ 3. Basic H.263 and in asterisk allow = 'ulaw;alaw;speex;gsm;h263;h263p' --- Ronald Wiplinger [EMAIL PROTECTED] wrote: Nardis Dome wrote: try eyeBeam, it works fine for me... [] type=friend secret= auth=md5 callerid=myCallerId canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex allow=gsm allow=h261 allow=h263 Thanks, I bought eyeBeam for two computers on the LAN for testing, but I get with above settings on both screens: Remote party does not support video What do I miss? bye Ronald ___ 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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 -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] SIP Soft Video phone for Asterisk usage
Ronald Wiplinger wrote: I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. try eyeBeam, it works fine for me... [] type=friend secret= auth=md5 callerid=myCallerId canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex allow=gsm allow=h261 allow=h263 __ Do you Yahoo!? Yahoo! Small Business - Try our new Resources site http://smallbusiness.yahoo.com/resources/ ___ 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] VIDEO ON 1.0.7 stable
--- listas iPfone [EMAIL PROTECTED] wrote: Hi all I need to know if the video support for h.263 is active in version stable 1.0.7 to use with eyeBeam in asterisk it works for me... [] type=friend secret= auth=md5 callerid=myCallerId canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex allow=gsm allow=h261 allow=h263 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] forum www.asterisk-italia.it
--- Paolo Losi [EMAIL PROTECTED] wrote: For all italian speaking users please visit and contribute to www.asterisk-italia.it! hi, www.asterisk-italia.it could not be found. I only found www.asteriskpbx.it... domé... __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] OH323 incoming audio stutter
The effect that I am seeing is that a call starts off fine, but suddenly after a few minutes the audio coming into Asterisk via OH323 gets very broken up to the point of being about 90% silence with occasional brief snippets of audio getting through. hi, any errors or warnings in Asterisk console? more info please... __ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo ___ 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] oh-323 compilation error !
Hi, try this... http://www.mail-archive.com/asterisk-users@lists.digium.com/msg86875.html domé --- urban viking [EMAIL PROTECTED] wrote: What are the gcc and lib requirements to compile oh323 channel (version 0.6.5) ? your 2 cents are welcome . Many thanks, I am unable to compile it ! gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/include/asterisk -I../wrapper -g -c -o chan_oh323.o chan_oh323.c In file included from /usr/include/string.h:33, from chan_oh323.c:34: /usr/lib/gcc-lib/i386-redhat-linux/3.2/include/stddef.h:201: syntax error before typedef In file included from chan_oh323.c:34: /usr/include/string.h:38: syntax error before extern /usr/include/string.h:39: parse error before __THROW /usr/include/string.h:43: parse error before __THROW /usr/include/string.h:56: parse error before __BEGIN_NAMESPACE_STD /usr/include/string.h:58: syntax error before extern ... _ MSN Hotmail : antivirus et antispam intégrés http://www.msn.fr/newhotmail/Default.asp?Ath=f ___ 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 __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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] oh323 compilation
Hi, attached an installation tip from Joao posted the 7th of jan. Get oh323 from http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz Get pwlib from http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz Get asterisk-oh323 from http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.6.5.tar.gz Untar the files #tar zxvf openh323-Janus_patch4-src-tar.gz #tar zxvf pwlib-Janus_patch4-src-tar.gz #tar zxvf asterisk-oh323-0.6.5.tar.gz #tar zxvf asterisk-1.0.3.tar.gz Install Pwlib #cd pwlib #./configure make clean make opt make install ldconfig Patch and Install OpenH323 #cd openh323 #patch -p1 ../asterisk-oh323-0.6.5/openh323_1.13.5-make.patch #./configure make clean make opt make install ldconfig Asterisk #cd asterisk-1.0.3 #make make install make samples Asterisk-oh323 #cd asterisk-oh323-0.6.5 Edit the Makefile #make make install ldconfig Don't forget to apply the chan-oh323 patch to openh323 before compiling. --- Gabriel Millerd [EMAIL PROTECTED] wrote: I have been struggling with oh323 compilation for some time now. I am trying to use the voip-info suggested walk through that points to here ... http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en ... which asks for versions OpenH323 (v1.13.5) PWlib (v1.6.6). Anyone know how to get these? The website http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries used openh323_1.12.2.tar.gz pwlib_1.5.2.tar.gz and sourceforge doesnt have direct downloads for the EXACT versions needed. Thanks very much. ___ 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 __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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] SIP Phone binary
X-lite for Linux. http://www.xten.com/apps/xprolinuxbeta/ --- Klaus Peras [EMAIL PROTECTED] wrote: Hey there, does anybody know a SIP-Client that I only have to unpack and can run it on Linux just like SJPhone, except SJPhone?? I need a Softphone for a Levigo Thin-Client, wich is not having a compiler. regards Klaus Peras begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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
[Asterisk-Users] MOH and OptiPoint400 std SIP
Hi all, I have some serious issues with the moh. Has anyone ever managed to get music on hold working on Siemens OptiPoint400 std SIP (with softphones and others hardphones moh works fine). thx in advance... domé __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] MOH and OptiPoint400 std SIP
Hi Chris, thx for your feedback, but trying the version v2.3.14 the file transfer failed with *incompatible image type*. For the moment i have the firmware sip_v2_45_8_P1.app and everything works fine... excepted MOH. maybe another tip? I'm using OptiPoint 400 phones here and MOH works fine. The firmware version is v2.3.14. -- Chris Hills IT Services North East Worcestershire College ___ 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] Asterisk -- PABX
--- [EMAIL PROTECTED] wrote: At the moment all I know is that they have Siemens PBX system. They will give me more details soon. since HiPath4x00 V1.0 you can use oh323 and HG3550 (STMI board in the HiPath) for interconnection between Siemens HiPath4x00 PBX and Asterisk. domé __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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
[Asterisk-Users] one way audio with X-lite for Linux/Suse 9.2
Hi all, i have installed X-Lite (xlite-linux-22)Suse 9.2 (2.6.8-24) and i have one way audio. The calling number can hear me but i don't hear the called number. Calling my mailbox works fine, i am able to hear my messages. I use a usb handset from Tedas AG. Another strange thing is that the pc, where the X-Lite is installed, start a http connection with brands.xten.net and try to get a file called settings_1103f_9.ini The response is HTTP/1.1 404 Not found. any suggestions... thx in advance __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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] Review: Asterisk at CeBIT 2005 / Asterisk at Linux-Tag 2005
Hi, we have nearly the same setup, excepted that the connection between the HiPath (HiPath4300) and Asterisk (RX100 Fujitsu/Siemens, CentOS 3.3) is an H323-trunk (oh323). We have different Hard- and Softclient connected to Astersik (Xlite, Opticlient SIP and OptiPoint400/410 SIP). It works very well. Unfortunately we have only some issues with MOH and OptiPoint400 SIP (with Xlite and Opticlient it works). Except these issues the interoperability between Asterisk and theseems to work well... ;-))) domé --- Thilo Rößler [EMAIL PROTECTED] wrote: I try to be a little bit more precise :-) We had an asterisk-server (HP-Server, Debian Sarge), a Siemens HiPath (cannot tell about the exact version, have to look it up tomorrow). They were connected via an ISDN-Card (Eicon 4BRI). 2 Siemens System-phones (old ones left over from HiCom-times) were connected to the HiPath. 4 IP-Phones (2 SNOM, 2 Cisco) were connected to our ethernet. Furthermore, we had an analog phone connected via a Fritz!Box ATA. As a sub-PBX, an old Agfeo was connected to our asterisk. Just for fun, we connected on old phone with a dial (don't know the proper englich term) to it. When somebody tried to dial a number outside, the call was routed via the internet-connection to another asterisk located in Darmstadt which then passed the call to the PSTN. Don't ask me about the names of the people who were impressed ... they were to much ;-) We did not try chan_cornet up to now. Maybe we'll do so in the future! -- Thilo Rößler Linup Front Robert-Koch-Strasse 9 64331 Weiterstadt Tel: 06151/9067-0 Fax: 06151/9067-299 Mobil: 0151/18242584 http://www.linupfront.de E-Mail: [EMAIL PROTECTED] __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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] Review: Asterisk at CeBIT 2005 / Asterisk at Linux-Tag 2005
--- Thilo Rößler [EMAIL PROTECTED] wrote: We had with us a demo-installation including different IP-phones, digital and analog phones as well as a Siemens HiPATH PBX to which our Asterisk-server served as a VoIP-gateway, and many people were impressed by the features as well as the quality of calls. Hi Thilo, it sounds very good. Can you please tell more about your demo-installation. Witch HiPath do you use for the interconnection with Asterisk (H4K or H3K). Do you interconnect it with oh323 or over an ISDN card? You also make reference about people who were impressed by features (did you heard about chan_cornet?). thx for your feedback. domé. __ Do you Yahoo!? Make Yahoo! your home page http://www.yahoo.com/r/hs ___ 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] Asterisk H323 support
Date: Mon, 21 Feb 2005 00:20:39 -0800 (PST) [zone:-], [EMAIL PROTECTED] mentioned in msg: Re: [Asterisk-Users] Asterisk H323 support that ... with Openh323 - v1.12.2 and pwlib - v1.5.2 I use asterisk-oh323 v.0.6.3b and it works fine What version of Asterisk are you running? And on what os and distribution? -- Kuniyoshi Murata.iChat/AIM:macwebcaster English-Japanese Interpreter mailto:[EMAIL PROTECTED] Macintosh Webcast Specialist http://www.macwebcaster.com Hi, Asterisk 1.0.1 CentOS 3.3 24.21-20.EL.c0 pwlib 1.5.2 openh323 1.12.2 asterisk-oh323 0.6.3b __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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] Asterisk H323 support
Hi, with Openh323 - v1.12.2 and pwlib - v1.5.2 I use asterisk-oh323 v.0.6.3b and it works fine hope it helps cu... --- kolo sos [EMAIL PROTECTED] wrote: Hi, anybody knows what's missing or problem why i cant compile asterisk-oh323 in my machine? i got this compiled successfully ...Openh323 - v1.12.2 ...pwlib - v1.5.2 except ...asterisk-oh323 - v0.6.5 here's the output as i run make... [EMAIL PROTECTED]:~/voip/asterisk-oh323-0.6.5$ make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/home/mkoy/voip/asterisk-oh323-0.6.5/wrapper' ./check_ver /home/mkoy/pwlib pwlib ./check_ver /home/mkoy/openh323 openh323 g++ -DP_LINUX=2.4.26 -ffunction-sections -fdata-sections -D_REENTRANT -Wall -fPIC -DP_USE_PRAGMA -DPHAS_TEMPLATES -I/home/mkoy/pwlib/include/ptlib/unix -I/usr/include/pwlib -I/home/mkoy/pwlib/include -DPTRACING -I/home/mkoy/openh323/include -DHAS_IXJ -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.5.2\ -DOPENH323VERSION=\1.12.2\ -I/home/mkoy/pwlib/include/ptlib/unix -I/home/mkoy/pwlib/include -I/home/mkoy/openh323/include -I/home/mkoy/openh323/include/openh323 -I../asterisk-driver -c asteriskaudio.cxx -o asteriskaudio.o asteriskaudio.cxx: In destructor `virtual PAsteriskSoundChannel::~PAsteriskSoundChannel()': asteriskaudio.cxx:167: error: `baseChannel' undeclared (first use this function) asteriskaudio.cxx:167: error: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: *** [asteriskaudio.o] Error 1 make[1]: Leaving directory `/home/mkoy/voip/asterisk-oh323-0.6.5/wrapper' make: *** [subdirs_build] Error 1 [EMAIL PROTECTED]:~/voip/asterisk-oh323-0.6.5$ Kolosos Philippines __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com ___ 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 __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250 ___ 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
[Asterisk-Users] Incomming calls on h323
Hi all, What has to be configured in Asterisk extensions.conf using OH323 for incomming calls? Curently outgoing calls can be setup, but an incoming call fail. My extensions.conf has the following entry for an outgoing call: exten = _61,1,Dial,OH323/h323:[EMAIL PROTECTED],tr What about incoming calls? thx for the feedback __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! 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
[Asterisk-Users] H.323 trunking
Hi,
Could someone help me on configuring a H.323 trunk.
I am trying to set up the following scenario:
[SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)]
I am using the following versions:
Linux CentOS 3.3/2.4.21-.EL.co
asterisk 1.0.1
pwlib_1.5.2
openh323_1.12.2
asterisk-oh323-0.6.3b
Calling from Asterisk (2004) to the H.323phone
(61-8004) gives me the following error
-- Executing Dial(SIP/2004-8350,
H323/192.168.204.130) in new stack
Dec 7 13:45:19 WARNING[1032209]: channel.c:1901
ast_request: No channel type registered for 'H323'
Dec 7 13:45:19 NOTICE[1032209]: app_dial.c:742
dial_exec: Unable to create channel of type 'H323'
== Everyone is busy/congested at this time
Dec 7 13:45:29 WARNING[1032209]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
[general]
static=yes
writeprotect=no
;Trunk=Modem/g1
[default]
exten = 2004,1,NoOp( call for ${EXTEN})
exten = 2004,2,Dial(SIP/${EXTEN},10,tr)
exten = 2004,3,Congestion
exten = 2005,1,NoOp( call for ${EXTEN})
exten = 2005,2,Dial(SIP/${EXTEN},10,tr)
exten = 2005,3,Congestion
exten = _61,1,Dial,H323/192.168.204.130
ps: 61 is a prefix. All the extensions 61xxx should be
routed to the H.323 trunk.
thx for your feedback
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Re: [Asterisk-Users] H.323 trunking
--- Michael Manousos [EMAIL PROTECTED]
wrote:
See below.
Nardis Dome wrote:
Hi,
Could someone help me on configuring a H.323
trunk.
I am trying to set up the following scenario:
[SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)]
I am using the following versions:
Linux CentOS 3.3/2.4.21-.EL.co
asterisk 1.0.1
pwlib_1.5.2
openh323_1.12.2
asterisk-oh323-0.6.3b
Calling from Asterisk (2004) to the H.323phone
(61-8004) gives me the following error
-- Executing Dial(SIP/2004-8350,
H323/192.168.204.130) in new stack
Dec 7 13:45:19 WARNING[1032209]: channel.c:1901
ast_request: No channel type registered for 'H323'
Dec 7 13:45:19 NOTICE[1032209]: app_dial.c:742
dial_exec: Unable to create channel of type 'H323'
== Everyone is busy/congested at this time
Dec 7 13:45:29 WARNING[1032209]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
[general]
static=yes
writeprotect=no
;Trunk=Modem/g1
[default]
exten = 2004,1,NoOp( call for ${EXTEN})
exten = 2004,2,Dial(SIP/${EXTEN},10,tr)
exten = 2004,3,Congestion
exten = 2005,1,NoOp( call for ${EXTEN})
exten = 2005,2,Dial(SIP/${EXTEN},10,tr)
exten = 2005,3,Congestion
exten = _61,1,Dial,H323/192.168.204.130
Change this into:
exten = _61,1,Dial,OH323/192.168.204.130
hi michael,
thx for the answer, but now i have the following
error:
Executing Dial(SIP/2004-b1cf,
OH323/192.168.204.130) in new stack
-- H.323 call to 192.168.204.130 with codec ALAW
-- Called 192.168.204.130
-- H.323 call 'ip$localhost/11490' cleared, reason
24 (Call ended with Q.931 cause)
-- Hungup 'OH323/L11490'
== No one is available to answer at this time
Dec 7 16:48:25 WARNING[1687569]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
what is the meaning of *reason 24*. Is there a problem
with my codec?
thx in advance...
ps: 61 is a prefix. All the extensions 61xxx
should be
routed to the H.323 trunk.
thx for your feedback
Michael.
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Re: [Asterisk-Users] H.323 trunking
hi michael,
thx for the answer, but now i have the following
error:
Executing Dial(SIP/2004-b1cf,
OH323/192.168.204.130) in new stack
-- H.323 call to 192.168.204.130 with codec ALAW
-- Called 192.168.204.130
-- H.323 call 'ip$localhost/11490' cleared, reason
24 (Call ended with Q.931 cause)
-- Hungup 'OH323/L11490'
== No one is available to answer at this time
Dec 7 16:48:25 WARNING[1687569]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
what is the meaning of *reason 24*. Is there a problem
with my codec?
thx in advance...
--- Michael Manousos [EMAIL PROTECTED]
wrote:
See below.
Nardis Dome wrote:
Hi,
Could someone help me on configuring a H.323
trunk.
I am trying to set up the following scenario:
[SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)]
I am using the following versions:
Linux CentOS 3.3/2.4.21-.EL.co
asterisk 1.0.1
pwlib_1.5.2
openh323_1.12.2
asterisk-oh323-0.6.3b
Calling from Asterisk (2004) to the H.323phone
(61-8004) gives me the following error
-- Executing Dial(SIP/2004-8350,
H323/192.168.204.130) in new stack
Dec 7 13:45:19 WARNING[1032209]: channel.c:1901
ast_request: No channel type registered for 'H323'
Dec 7 13:45:19 NOTICE[1032209]: app_dial.c:742
dial_exec: Unable to create channel of type 'H323'
== Everyone is busy/congested at this time
Dec 7 13:45:29 WARNING[1032209]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
[general]
static=yes
writeprotect=no
;Trunk=Modem/g1
[default]
exten = 2004,1,NoOp( call for ${EXTEN})
exten = 2004,2,Dial(SIP/${EXTEN},10,tr)
exten = 2004,3,Congestion
exten = 2005,1,NoOp( call for ${EXTEN})
exten = 2005,2,Dial(SIP/${EXTEN},10,tr)
exten = 2005,3,Congestion
exten = _61,1,Dial,H323/192.168.204.130
Change this into:
exten = _61,1,Dial,OH323/192.168.204.130
ps: 61 is a prefix. All the extensions 61xxx
should be
routed to the H.323 trunk.
thx for your feedback
Michael.
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[Asterisk-Users] asterisk-oh323-0.6.3b and logical Channel
Hi all, There are three methods to establish a connection between H.323 endpoints - Fast Connect (Fast Start) - H.245 Tunnelling - H.245 Logical Channel Wich one is used by Asterisk? How to configure a H.245 Logical Channel using asterisk-oh323? Does anyone have an idea? thx in advance... __ 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
RE: [Asterisk-Users] Spawn extension
--- Colin Anderson [EMAIL PROTECTED] wrote: Unfortunately, there doesn't seem to be any kind of granularity there so you can't branch based on what went wrong, just that something went wrong. hth. ok, but how can i fix this probleme. I don't know where to start with my trouble shooting. Unfortunately the SIP debug don't give more informations (only Service Unavailable). any suggestions?? Thx __ Do you Yahoo!? All your favorites on one personal page Try My Yahoo! 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
[Asterisk-Users] Spawn extension
hi, calling from Asterisk to PBX via Eicon Diva 4BRI gives me the following error. -- Executing NoOp(SIP/2004-41dc, call for 998004) in new stack -- Executing Dial(SIP/2004-41dc, CAPI/99:8004|20|r) in new stack == Everyone is busy/congested at this time -- Executing Congestion(SIP/2004-41dc, ) in new stack == Spawn extension (default, 998004, 3) exited non-zero on 'SIP/2004-41dc' What is the meaning of the exited non-zero line? thx for your feedback __ 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
[Asterisk-Users] integrating Asterisk to existing TDM-based PBX
Hello, i'm looking for informations in integrating Asterisk to existing TDM-based PBX (particularly Siemens HiPath4000/Hicom300E) similar to the document you can find on www.pham.org/asterisk/asterisk-meridian-a1.pdf for Nortel. Unfortunately the page http://www.voip-info.org/wiki-Siemens+Hicom is not up to date. would be grateful for any pointers. thx. __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ 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
