Re: [Asterisk-Users] Re: TDM04B problems
Andrew Sayman wrote: Noah Miller wrote: Depending on your BIOS and motherboard, you may be able to use another IRQ if you move the card to a different PCI slot. - Noah This is a computer meant to be rack-mounted that I'm trying to install this on. I certainly don't see any space for another PCI slot, so I don't think that solution is going to work. ___ 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 Don't know if this will help you any further, but. After some trouble with IRQ sharing mayhem we solved our little problem by tinkering the linux kernel. I forgot the names of the actual modules, but after disabling modules for APIC support and something about IRQ sharing or APIC-IO or such, we effectively disables the APIC from handling IRQ's. I'm not so sure that disabling the APIC only from the BIOS setup will do it (it did not in our 'MSI'-case). We had to disble the APIC from within the BIOS setup also, otherwise our system crashed at boot. After doing so our /proc/interrupt didn't show any 'IO-APIC-level' and 'IO-APIC-edge' containing lines but only 'XT-PIC' containing lines. After that, our TDM04B allways got it's own IRQ and the mayhem never returned. If you're in real nead of those module names, let me know. I've got some notes somewhere at the bottom of the 3 feet tall pile besides my desk that says 'To be examned further someday' ___ 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] Re: Asterisk-Users Digest, Vol 12, Issue 25
Hi, Updating zaptel gives me this during the make. Any ideas, the searches and Wiki gives me no hints. In file included from /usr/src/linux-2.4/include/linux/fs.h:19, from /usr/src/linux-2.4/include/linux/capability.h:17, from /usr/src/linux-2.4/include/linux/binfmts.h:5, from /usr/src/linux-2.4/include/linux/sched.h:9, from /usr/src/linux-2.4/include/linux/mm.h:4, from /usr/src/linux-2.4/include/linux/slab.h:14, from /usr/src/linux-2.4/include/asm/pci.h:32, from /usr/src/linux-2.4/include/linux/pci.h:617, from tor2.c:33: /usr/src/linux-2.4/include/linux/dcache.h: In function `dget': /usr/src/linux-2.4/include/linux/dcache.h:249: warning: implicit declaration of function `__out_of_line_bug_R8b0fd3c5' In file included from /usr/src/linux-2.4/include/asm/io.h:47, from /usr/src/linux-2.4/include/asm/pci.h:35, from /usr/src/linux-2.4/include/linux/pci.h:617, from tor2.c:33: /usr/src/linux-2.4/include/linux/vmalloc.h: In function `vmalloc': /usr/src/linux-2.4/include/linux/vmalloc.h:35: `boot_cpu_data_R0657d037' undeclared (first use in this function) /usr/src/linux-2.4/include/linux/vmalloc.h:35: (Each undeclared identifier is reported only once /usr/src/linux-2.4/include/linux/vmalloc.h:35: for each function it appears in.) /usr/src/linux-2.4/include/linux/vmalloc.h: In function `vmalloc_dma': /usr/src/linux-2.4/include/linux/vmalloc.h:44: `boot_cpu_data_R0657d037' undeclared (first use in this function) /usr/src/linux-2.4/include/linux/vmalloc.h: In function `vmalloc_32': /usr/src/linux-2.4/include/linux/vmalloc.h:53: `boot_cpu_data_R0657d037' undeclared (first use in this function) tor2.c: In function `tor2_spanconfig': tor2.c:206: warning: implicit declaration of function `printk_R1b7d4074' tor2.c: In function `init_spans': tor2.c:274: warning: implicit declaration of function `sprintf_R1d26aa98' make: *** [tor2.o] Error 1 Regards Arun --- [EMAIL PROTECTED] wrote: Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: TDM01B card configuration (Dave Cotton) 2. Re: [SPAM:* SpamScore] RE: [Asterisk-Users] Call Transfer using SIP clients (Frank Schoep) 3. RE: presence and IM again, want to develop a workinghack (Florian Overkamp) 4. calling shell scripts from within * (Terry Wade) 5. Re: Sometimes yes - sometimes no (dialplan) (Ronald_Wiplinger) 6. Dialogic D/300 E1 (Fredrik Lith?n) 7. Transfer and CDR's (Sebastian Zaprzalski) 8. RE: Provider Survey (Mohamed Farid) 9. oh323 problem with cisco 2600 (craz sead) 10. Re: [SPAM:* SpamScore] Re: [Asterisk-Users] Call Transfer using SIP clients (Frank Schoep) 11. About AgentMonitorOutgoing (Gary Li) 12. Re: Call Transfer using SIP clients (Brian Capouch) 13. Re: Linux Distribution for Asterisk server use (Tzafrir Cohen) 14. Re: TDM01B card configuration (Tzafrir Cohen) 15. Re: wi-fi phone advice (Wolfgang Lumpp) 16. Re: [SPAM: SpamScore] [Asterisk-Users] Call Transfer using SIP clients (Frank Schoep) 17. Re: calling shell scripts from within * (Giorgio Incantalupo) 18. Problems installing AMP (Ian Bert Tusil) 19. ENUM (Ronald_Wiplinger) 20. RE: wi-fi phone advice (Florian Overkamp) 21. Re: calling shell scripts from within * (Tzafrir Cohen) 22. Re: Getting FOP working with ICD? (Axel Pache) 23. voicexml (trixter http://www.0xdecafbad.com) 24. Re: [SPAM:* SpamScore] Re: [SPAM: SpamScore] [Asterisk-Users] Call Transfer usingSIP clients (Frank Schoep) 25. Re: MOH - request to schdule in the past (Andrew Furey) -- Message: 1 Date: Tue, 05 Jul 2005 08:37:08 +0200 From: Dave Cotton [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] TDM01B card configuration To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain On Mon, 2005-07-04 at 10:38 -0700, Mike Wissa wrote: When you try to start asterisk. the following errors appear Jul 4 10:37:59 NOTICE[4015]: res_odbc.c:518 load_module: res_odbc loaded. .Jul 4 10:37:59 ERROR[4015]: chan_zap.c:6584 mkintf: Signalling requested on channel 4 is FXO Loopstart but line is in FXS Loopstart signalling
Re: [Asterisk-Users] cisco 7940 + sccp issue
stevanus ha scritto:
Does anyone know how to make this thing (7940) work with asterisk
(chan_sccp module) ?
I've set the configuration according to the wiki and now the phone
just keep asking for CTLSEPxxx.tlv from my tftp server.
Update the skinny firmware.
The phone has to look for SEPMAC.cnf.xml from the tftp server
The SEPMAC.cnf.xml looks like it. LoadInformation tag is useful to
update the phone firmware. Just put the name of the new firmware there
with no extension
device
devicePool
callManagerGroup
members
member priority=0
callManager
ports
ethernetPhonePort2000/ethernetPhonePort
/ports
processNodeNameYOUR_ASTERISK_BOX_IP/processNodeName
/callManager
/member
/members
/callManagerGroup
/devicePool
versionStamp{Jan 01 2002 00:00:00}/versionStamp
loadInformation/loadInformation
userLocale
name/name
langCode/langCode
/userLocale
networkLocale/networkLocale
idleTimeout0/idleTimeout
authenticationURL/authenticationURL
directoryURL/directoryURL
idleURL/idleURL
informationURL/informationURL
messagesURL/messagesURL
proxyServerURL/proxyServerURL
servicesURL/servicesURL
/device
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[Asterisk-Users] SIP/2.0 403 Forbidden
Hi all, I have been worriyng and googling a lot but I can't find my mistake. I am trying to regiter an X-Lite Softphone to Asterisk, but I am getting a SIP/2.0 403 Forbidden response: SEND TIME: 10157385 SEND 10.100.249.12:5060 REGISTER sip:10.100.249.12 SIP/2.0 Via: SIP/2.0/UDP 10.100.249.86:5060;rport;branch=z9hG4bKFAC1B6F2B5414EE9855696A09A83FB22 From: Tester sip:[EMAIL PROTECTED];tag=3354744682 To: Tester sip:[EMAIL PROTECTED] Contact: Tester sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 4806 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 RECEIVE TIME: 10157385 RECEIVE 10.100.249.12:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.100.249.86:5060;branch=z9hG4bKFAC1B6F2B5414EE9855696A09A83FB22 From: Tester sip:[EMAIL PROTECTED];tag=3354744682 To: Tester sip:[EMAIL PROTECTED];tag=as7ae925e2 Call-ID: [EMAIL PROTECTED] CSeq: 4806 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 -- MY config is: /etc/asterisk/sip.conf : ; Note: If your SIP devices are behind a NAT and your Asterisk ; server isn't, try adding nat=1 to each peer definition to ; solve translation problems. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf [1000] username=1000 secret=abc123 context=mytest host=dynamic - /etc/asterisk/extensions.conf : [general] static=yes writeprotect=yes ;Suport phones ;SUPPORTPHONES=SIP/2205SIP/2206SIP/2207SIP/2208SIP/2209 [globals] XLITE=SIP/1000 [mytest] exten = 1367,1Dial(SIP/1000) exten = 2890,1,Wait(2) exten = 2890,2,Answer exten = 2890,3,Playback(demo-echotest) exten = 2890,4,Echo() exten = 2468,1,Dial($XLITE) - Some startup messages from Asterisk: [pbx_config.so] = (Text Extension Configuration) == Parsing '/etc/asterisk/extensions.conf': Found == Setting global variable 'XLITE' to 'SIP/1000' -- Registered extension context 'mytest' -- Added extension '1367' priority 1 to mytest -- Added extension '2890' priority 1 to mytest -- Added extension '2890' priority 2 to mytest -- Added extension '2890' priority 3 to mytest -- Added extension '2890' priority 4 to mytest -- Added extension '2468' priority 1 to mytest Someone has an idea what is causing the SIP/2.0 403 Forbidden? Please help. -- Weitersagen: GMX DSL-Flatrates mit Tempo-Garantie! Ab 4,99 Euro/Monat: http://www.gmx.net/de/go/dsl ___ 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] Cmd MusicOnHold works, but no sound when a call gets holded
exten = 555,1,MusicOnHold(default) i can hear the music, so far so good. But when i hold an incoming call by pressing the HOLD-key on my snom telephone - nothing happens. No output at CLI that the MOH gets played. When debugging SIP on asterisk, in the moment i press the HOLD-key i can see some SIP-INVITE messages from the phones that holds the call to the other one. What * version ? I had the same problem upgrading from .5 to .6 and then it resolved upgrading to .7 (search for my previous post, cisco phones) ___ 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] Users handbook
This is somewhat unique to the site installation. For example, I don't have *69 programmed at my site because frankly there's no need for it with the Cisco 7960's. I do however have an automatic conference booking utility and a speaking clock. Not often found in smaller sites. I think you are on your own here. Chris Mason wrote: At the most recent project I completed I have to post a intranet web page with instructions on using the system and phones. Asterisk is 1.07 stable and the phones are Polycom IP300, IP500, and IP600. Has anyone done an Astersik users guide? Something non-technical but covering most of the features an office worker would use. If nothing exists, should we develop this as a documentation project? After all, the greatest software is little use if the users never hear about the features. Chris ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 addons install problem
On Wednesday 06 Jul 2005 06:07, wei li wrote: Hi there: I have successfully installed the Asterisk 1.0.9 on my Freebsd 5.4 box. When I tend to install the addon for mysql CDR billing, It always return me the following errors: SIP# gmake clean rm -f *.so *.o .depend gmake -C format_mp3 clean gmake[1]: Entering directory `/usr/home/wilson/asterisk/asterisk-addons-1.0.9/format_mp3' rm -f *.o *.so *~ gmake[1]: Leaving directory `/usr/home/wilson/asterisk/asterisk-addons-1.0.9/format_mp3' SIP# gmake ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/include/mysql `ls *.c` app_addon_sql_mysql.c:15:27: asterisk/file.h: No such file or directory app_addon_sql_mysql.c:16:29: asterisk/logger.h: No such file or Perhaps it's now time for the addons package to include the same release of Asterisk directory, i.e., insert -I../asterisk-release num. B ___ 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] g.729 codec -- open source?
Hello, is there an open-source implementation of G.729 codec for use outside of US? I know it's a patented codec, but since there are usually no software patents outside of the US, I don't care about the patent license. I could use open-source implementation of the codec, if there was some. Any ideas? Sincerely, Juraj Bednar. ___ 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] cisco 7940 + sccp issue
Stefan Gofferje wrote: "If the phone just requests CTLSEPxxx.tlv and nothing else, it either have been used on a CallManager with authentication / encryption enabled and is now security locked because the asterisk does not provide the proper tlv-file or the firmware is corrupted. Try to reset to factory settings. if this does not help, try to reflash the firmware. " Hi, I've unlocked the phone by pressing **# and set it back to factory setting. But the problem still exists. Do I really need to reflash the phone? Sorry just wanna assure myself that the action is necessary in order to make my 7940 talk with asterisk using sccp. I had bad experiences in flashing devices therefore I want to avoid this as much as possible :). Best regards, Stevanus Stefan Gofferje wrote: Hi, On 9:20:51 July 06, 2005 stevanus [EMAIL PROTECTED] wrote: I've set the configuration according to the wiki and now the phone just keep asking for CTLSEPxxx.tlv from my tftp server. If this does not help - well shit happens... Just kidding... :-). If you have a legal license for the phone software, you could send the phone to Cisco if nothing else helps. Regards, Stefan ___ 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 on Linksys WRT54G
Check that both sides use the same codec. I
had the same problem before L
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Tuesday, July 05, 2005 4:36
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Asterisk on Linksys WRT54G
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Walid Azab
Sent: Tuesday, July 05, 2005 4:23
AM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk
on Linksys WRT54G
Hi all,
Any one tried installing Asterisk on Linksys WRT54G? We have
but facing problems with SIP to SIP calls. The phones ring and calls are
established but we cannot hear any voice at all. I tried allow=all in the
general section but did not work. So I forced ulaw. Can any one please check it
out and let me know what is wrong?
Here are the conf files:
Asterisk Version: Asterisk
CVS-HEAD-01/17/05-00:35:58 built by [EMAIL PROTECTED]
on a i686 running Linux
==SIP.CONF
[general]
port =
5060 ; Port to bind
to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on
machine)
disallow=all
; Allow all codecs
allow=ulaw
context = bogon-calls ; Send SIP callers that we don't know about here
[2000]
type=friend
; This device takes and makes calls
username=2000 ; Username on
device
secret=1234 ;
Password for device
host=dynamic ; This host
is not on the same IP addr every time
context=from-sip ; Inbound calls from this host
go here
mailbox=100 ;
Activate the message waiting light if this
; voicemailbox has messages in it
[2001]
; Duplicate of 2000, except with different auth data
type=friend
username=2001
secret=1234
host=dynamic
context=from-sip
mailbox=101
==Extensions.conf
[general]
static=yes
writeprotect=yes
[bogon-calls]
exten = _.,1,Congestion
[from-sip]
exten = 2000,1,Dial(SIP/2000,20)
exten = 2000,2,Voicemail(u2000)
exten = 2000,102,Voicemail(b2000)
exten = 2000,103,Hangup
exten = 2001,1,Dial(SIP/2001,20)
exten = 2001,2,Voicemail(u2001)
exten = 2001,102,Voicemail(b2001)
exten = 2001,103,Hangup
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
How are the routers connected to the IP
network? Any nat before them on either end?
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[Asterisk-Users] can we use asterisk as a SIP Redirect Server?
can we use asterisk as a SIP Redirect Server? Thanks Erdem HAKI [EMAIL PROTECTED] ___ 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 addons install problem
On Wed, Jul 06, 2005 at 03:07:16PM +1000, wei li wrote: Hi there: I have successfully installed the Asterisk 1.0.9 on my Freebsd 5.4 box. When I tend to install the addon for mysql CDR billing, It always return me the following errors: My asterisk-addons deb builds independently of asterisk (with the package asterisk-dev installed) The current patch we have to allow that: --- asterisk-addons-1.0.9.orig/Makefile +++ asterisk-addons-1.0.9/Makefile @@ -16,7 +16,8 @@ MODS=format_mp3/format_mp3.so CFLAGS+=-fPIC -CFLAGS+=-I../asterisk +CFLAGS+=-Wall -g +CFLAGS+=-O2 CFLAGS+=-D_GNU_SOURCE INSTALL=install --- asterisk-addons-1.0.9.orig/cdr_addon_mysql.c +++ asterisk-addons-1.0.9/cdr_addon_mysql.c @@ -21,7 +21,6 @@ #include asterisk/module.h #include asterisk/logger.h #include asterisk/cli.h -#include asterisk.h #include stdio.h #include string.h SIP# gmake clean rm -f *.so *.o .depend gmake -C format_mp3 clean gmake[1]: Entering directory `/usr/home/wilson/asterisk/asterisk-addons-1.0.9/format_mp3' rm -f *.o *.so *~ gmake[1]: Leaving directory `/usr/home/wilson/asterisk/asterisk-addons-1.0.9/format_mp3' SIP# gmake ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/include/mysql `ls *.c` app_addon_sql_mysql.c:15:27: asterisk/file.h: No such file or directory app_addon_sql_mysql.c:16:29: asterisk/logger.h: No such file or directory app_addon_sql_mysql.c:17:30: asterisk/channel.h: No such file or directory app_addon_sql_mysql.c:18:26: asterisk/pbx.h: No such file or directory app_addon_sql_mysql.c:19:29: asterisk/module.h: No such file or directory app_addon_sql_mysql.c:20:34: asterisk/linkedlists.h: No such file or directory app_addon_sql_mysql.c:21:31: asterisk/chanvars.h: No such file or directory app_addon_sql_mysql.c:22:27: asterisk/lock.h: No such file or directory cdr_addon_mysql.c:17:29: asterisk/config.h: No such file or directory cdr_addon_mysql.c:18:30: asterisk/options.h: No such file or directory cdr_addon_mysql.c:19:30: asterisk/channel.h: No such file or directory cdr_addon_mysql.c:20:26: asterisk/cdr.h: No such file or directory cdr_addon_mysql.c:21:29: asterisk/module.h: No such file or directory cdr_addon_mysql.c:22:29: asterisk/logger.h: No such file or directory cdr_addon_mysql.c:23:26: asterisk/cli.h: No such file or directory cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory gmake -C format_mp3 all gmake[1]: Entering directory `/usr/home/wilson/asterisk/asterisk-addons-1.0.9/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -I/usr/local/include -D_REENTRANT -D_GNU_SOURCE -D_THREAD_SAFE -O6 -c -o common.o common.c common.c:1:29: asterisk/logger.h: No such file or directory common.c: In function `decode_header': common.c:93: warning: implicit declaration of function `ast_log' common.c:93: error: `LOG_WARNING' undeclared (first use in this function) common.c:93: error: (Each undeclared identifier is reported only once common.c:93: error: for each function it appears in.) gmake[1]: *** [common.o] Error 1 gmake[1]: Leaving directory `/usr/home/wilson/asterisk/asterisk-addons-1.0.9/format_mp3' gmake: *** [format_mp3/format_mp3.so] Error 2 Can anybody give me a help? Thanks a lot. Wilson ___ 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 -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Users handbook
--- Ursprüngliche Nachricht --- Von: Mark Phillips [EMAIL PROTECTED] An: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Betreff: Re: [Asterisk-Users] Users handbook Datum: Wed, 06 Jul 2005 04:50:07 -0400 This is somewhat unique to the site installation. For example, I don't have *69 programmed at my site because frankly there's no need for it with the Cisco 7960's. I do however have an automatic conference booking utility and a speaking clock. Not often found in smaller sites. I think you are on your own here. Chris Mason wrote: At the most recent project I completed I have to post a intranet web page with instructions on using the system and phones. Asterisk is 1.07 stable and the phones are Polycom IP300, IP500, and IP600. Has anyone done an Astersik users guide? Something non-technical but covering most of the features an office worker would use. If nothing exists, should we develop this as a documentation project? After all, the greatest software is little use if the users never hear about the features. Chris ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 -- 5 GB Mailbox, 50 FreeSMS http://www.gmx.net/de/go/promail +++ GMX - die erste Adresse für Mail, Message, More +++ ___ 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] g.729 codec -- open source?
Check out http://www.readytechnology.co.uk/open/g729/ Regards, Sahil Gupta VoiceValley On Wed, 6 Jul 2005, Juraj Bednar wrote: Hello, is there an open-source implementation of G.729 codec for use outside of US? I know it's a patented codec, but since there are usually no software patents outside of the US, I don't care about the patent license. I could use open-source implementation of the codec, if there was some. Any ideas? Sincerely, Juraj Bednar. ___ 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 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] app_rxfax does not receive
Hi all, I try to use app_rxfax. Aplication app_rxfax start O.K., fax trying to send, but it will stop at the beginning of page and after few seconds it stop with error 400. Does anybody has any suggestions? Thanks, Bob. ___ 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] How does Vonage support fax machines?
Before you give up, I have had good results with a Sipura 2002 ATA and using Teliax for faxing, I tried other termination accounts with the same setup and it didn't work. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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] Users handbook
Mark Phillips wrote: This is somewhat unique to the site installation. For example, I don't have *69 programmed at my site because frankly there's no need for it with the Cisco 7960's. I do however have an automatic conference booking utility and a speaking clock. Not often found in smaller sites. I think you are on your own here. If one is implementing an Asterisk solution in an office scenario, it has to have fairly similar features to another Asterisk installation. It's easy enough to edit and remove the parts that are different. What I am suggesting is a comprehensive Here's everything Asterisk can do out of the box document, change or remove what doesn't apply. Let me know if any of you want to pool the work we have already done, I will compile to a complete document and post on the wiki. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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] Users handbook
I'm planning on implementing an Asterisk system at a couple of small offices, and a couple of homes, in the near future... and I don't have any documentation yet. What you're suggesting sounds wonderful, to me. I would contribute, if I had anything... but making it an inclusive manual would be a good idea, I think... you can always edit/remove sections. :) Andrew M Stemen [EMAIL PROTECTED] http://www.andrewmstemen.com Chris Mason (Lists) wrote: Mark Phillips wrote: This is somewhat unique to the site installation. For example, I don't have *69 programmed at my site because frankly there's no need for it with the Cisco 7960's. I do however have an automatic conference booking utility and a speaking clock. Not often found in smaller sites. I think you are on your own here. If one is implementing an Asterisk solution in an office scenario, it has to have fairly similar features to another Asterisk installation. It's easy enough to edit and remove the parts that are different. What I am suggesting is a comprehensive Here's everything Asterisk can do out of the box document, change or remove what doesn't apply. Let me know if any of you want to pool the work we have already done, I will compile to a complete document and post on the wiki. ___ 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] newbie asterisk-addons installation
good day all, my asterisk is working greatly.. and i want to put a billing.. but i have this error when i try 'make' [EMAIL PROTECTED] asterisk-addons-1.0.7]# make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -I/root/asterisk/include/ -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:165:77: macro AST_LIST_REMOVE passed 4 arguments, but takes just 3 app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:165: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:165: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:165: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 please help.. -- shahdan __ 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] Users handbook
On Wed, Jul 06, 2005 at 06:19:53AM -0400, Chris Mason (Lists) wrote: If one is implementing an Asterisk solution in an office scenario, it has to have fairly similar features to another Asterisk installation. It's easy enough to edit and remove the parts that are different. What I am suggesting is a comprehensive Here's everything Asterisk can do out of the box document, change or remove what doesn't apply. Let me know if any of you want to pool the work we have already done, I will compile to a complete document and post on the wiki. Hey, compile is what computers do, not humans. :-) Is there any existing program that could either from /etc/asterisk or using the manager interface figure out enough about the asterisk configuration to generate such a manual (using some templating engine)? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Users handbook
I just wonder what can i do with asterisk and its limits. For example i really don't know yet is asterisk used as redirect server? Thanks for your reply, Erdem HAKI - [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew M Stemen Sent: Wednesday, July 06, 2005 1:29 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Users handbook I'm planning on implementing an Asterisk system at a couple of small offices, and a couple of homes, in the near future... and I don't have any documentation yet. What you're suggesting sounds wonderful, to me. I would contribute, if I had anything... but making it an inclusive manual would be a good idea, I think... you can always edit/remove sections. :) Andrew M Stemen [EMAIL PROTECTED] http://www.andrewmstemen.com Chris Mason (Lists) wrote: Mark Phillips wrote: This is somewhat unique to the site installation. For example, I don't have *69 programmed at my site because frankly there's no need for it with the Cisco 7960's. I do however have an automatic conference booking utility and a speaking clock. Not often found in smaller sites. I think you are on your own here. If one is implementing an Asterisk solution in an office scenario, it has to have fairly similar features to another Asterisk installation. It's easy enough to edit and remove the parts that are different. What I am suggesting is a comprehensive Here's everything Asterisk can do out of the box document, change or remove what doesn't apply. Let me know if any of you want to pool the work we have already done, I will compile to a complete document and post on the wiki. ___ 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 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] chan_capi 0.5.3 asterisk HEAD 2005/07/04 undefined symbol error
Hi all, I have a bit of a problem with chan_capi. Details: chan_capi-cm-0.5.3 from sourceforge, zaptel, libpri and asterisk cvs HEAD from July 4, 2005. Everything works fine except... [app_capiHOLD.so]Jul 4 22:56:58 WARNING[1013]: loader.c:313 __load_resource: /usr/lib/asterisk/modules/app_capiHOLD.so: undefined symbol: get_ast_capi_MessageNumber Jul 4 22:56:58 WARNING[1013]: loader.c:523 load_modules: Loading module app_capiHOLD.so failed! Ouch ... error while writing audio data: : Broken pipe I appreciate any suggestion how I can fix this. Thanks and regards, Patrick ___ 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] Dialplan configuration with Realtime
Hello! Following the instructions on voip-ip.org I have implemented Realtime with MySQL for my Asterisk server. The individual extension configuration is managed in a table called extensions. Still I have to keep some data in the extensions.conf, namely the switch and the include statements. Is there a way to minimize that or completely get rid of them? === extensions.conf === [local] switch = Realtime/[EMAIL PROTECTED] [from-sip] include = local === extconfig.conf === rt_ext = mysql,realtime,extensions Regards, Gunde ___ 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] chan_capi 0.5.3 asterisk HEAD 2005/07/04 undefined symbol error
Patrick ha scritto: Hi all, I have a bit of a problem with chan_capi. Details: chan_capi-cm-0.5.3 from sourceforge, zaptel, libpri and asterisk cvs HEAD from July 4, 2005. Everything works fine except... stop asterisk, rm /usr/lib/asterisk/modules/* rm /usr/include/asterisk/* cd asterisk make clean make upgrage cd chan_capi-cm-0.5.3 make clean make install now run asterisk again Sergio ___ 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] Polycom distributor in the UK ?
John Daragon wrote: Hi; I'm looking for a Polycom distributor in the UK who can supply a small number (around 20) IP301 / IP501 handsets. Can anyone recommend someone ? jd I have been buying from Zycko - very efficient and on the ball. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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 on Linksys WRT54G
I stand corrected... It was late :) PAP2-NA = Useful PAP2 = Useless/Vonage -Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 06, 2005 1:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk on Linksys WRT54G Jay Milk wrote: You're confusing the WRT54G/GS with the PAP2. The PAP2 is the Sipura SPA-2000 in a pretty LinkSys case, the PAP2-NA is the same item rendered useless by locking it to Vonage service. The WRT54G/GS are freely available wireless routers for which LinkSys made available all sources (had to, it runs GPL'd linux). With the available of source, several folks compile their own firmware versions, most importantly www.hyperwrt.org and www.openwrt.org. Hmm. I think *you* may have the models confused. I'm pretty sure the -NA suffixed models are the ones that can be programmed in the field, and the ones without the -NA are locked to a provider. ?? B. ___ 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] Polycom distributor in the UK ?
Chris Mason (Lists) wrote: John Daragon wrote: Hi; I'm looking for a Polycom distributor in the UK who can supply a small number (around 20) IP301 / IP501 handsets. Can anyone recommend someone ? jd I have been buying from Zycko - very efficient and on the ball. Ta. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ 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] MWI from SIP provide
Hi, I am using a sip provider that offers voicemail. They send me a sip notify that there are voicemails, and I would like this notify to be sent to one of the extensions on asterisk (a sipura 2100 or cisco 7960), to light a lamp/give stutter dial tone. The provider is running * too and is flexible about their configuration. My * is running on openwrt, so I cant run vmail on *, besides the provider vmail only kicks in if my system goes offline, so is useful. Any help appreciated... tks Andrew ___ 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] phone comparison matrix
Hi Is there a phone comparison matrix I could consult I have a series of features that I would like to evaluate on the most common phones on the market example: dual-ethernet POE / direct power / both number of lines speed dials programmable buttons BLF LEDS Headset plug conference call built in hands free operation display size codecs communication protocol (SIP, h.323) price availability reliability know bugs / limitations asterisk compatibility If someone has done this recently that would save me some time Patrick ___ 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] UK asterisk
Good day all Im looking for someone in the UK that knows asterisk and thats willing to do a quick job for us,its in at tele city -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ 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] app_conference and AGI
Hi,
i was successful in compiling app_conference and setting up an
conference was quite easy. :-)
Does anyone knows if it is possible to have an IVR accessable from
inside the conference. So, if i dialed into an conference i want to be
able to press '*' and then the actual discussion is muted for me and i
and menu is read to me. Something like the ${MEETME_AGI_BACKGROUND} in
MeetMe.
Thx in advance :)
Tobias Wolf
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[Asterisk-Users] getting Incoming but unable to dial out using oh323
I m using oh323 and i m receiving incoming calls at windows NetMeeing and at SJPhone from SIP IAX softphones but what should i do to be able to call from NetMeeting or any H323 softphone .when i dial any extension... it starts OH323/R4096 and then fails and plays demo-congrates from default context...i think it is some registration problem plz send me a sample oh323.conf .. so that i can call from my h323 phones Sell on Yahoo! Auctions - No fees. Bid on great items.___ 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] DECT VoIP Gateway
Hi all, Just want to share with all of you a new hot DECT VoIP gateway available from www.broad-tel.com/index_en.php. The DECT VoIP gateway is capable of handling both SIP and the H.323 calls. Up to 4 registrations to the SIP proxy or H.323 Gatekeeper. To bring the users most flexibility, the add-on RJ-11 interface for PSTN connection, users not only can make the daily PSTN communication, but also enjoy the convenience brought by VoIP communications. With built-in DECT GAP Compatible base, up to 5 DECT handsets can be registered on the gateway. Cheers, IM ___ 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] how to set language in capi
I am trying to use language=it in asterisk I downloaded the sound package and installed it I added country=it in indications.conf language=it in sip.conf language=it in iax2.conf everything ok in call from sip and from iax The problem arises in outside call, coming trom CAPI Trunk I try language=it in capi.conf: no result: always language=en I found a german forum, and it seems to be a common problem ( I don't undertand a lot of german, anyway) ( cfr: http://www.ip-phone-forum.de/forum/viewtopic.php?t=17639) I also try to put exten = .,1,SetLanguage(it) in some places, but no result I am running asterisk 1-0-9 any help will be greatly appreciated Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ 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] Re: DECT VoIP Gateway
you can find it under VoIP Products/Wireless IP phone. On 7/6/05, IM.Nobody [EMAIL PROTECTED] wrote: Hi all, Just want to share with all of you a new hot DECT VoIP gateway available from www.broad-tel.com/index_en.php. The DECT VoIP gateway is capable of handling both SIP and the H.323 calls. Up to 4 registrations to the SIP proxy or H.323 Gatekeeper. To bring the users most flexibility, the add-on RJ-11 interface for PSTN connection, users not only can make the daily PSTN communication, but also enjoy the convenience brought by VoIP communications. With built-in DECT GAP Compatible base, up to 5 DECT handsets can be registered on the gateway. Cheers, IM ___ 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] DECT VoIP Gateway
Would it be a good replacement of expensive WiFi phones? How much is it?? On 7/6/05, IM.Nobody [EMAIL PROTECTED] wrote: Hi all, Just want to share with all of you a new hot DECT VoIP gateway available from www.broad-tel.com/index_en.php. The DECT VoIP gateway is capable of handling both SIP and the H.323 calls. Up to 4 registrations to the SIP proxy or H.323 Gatekeeper. To bring the users most flexibility, the add-on RJ-11 interface for PSTN connection, users not only can make the daily PSTN communication, but also enjoy the convenience brought by VoIP communications. With built-in DECT GAP Compatible base, up to 5 DECT handsets can be registered on the gateway. Cheers, IM ___ 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 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] ETSI or QSIG
Hello to all, Does Asterisk support QSIG? and the configuration is in capi.conf? and if it supports it, do you have samples of the configuration? I had my Asterisk connecting to a Siemens PBX with ETSI and it was working fine, but peolpe said to me that QSIG could implement more features and turn the calls between the two systems transparent for the users. And I read that QSIG could take the caller name and doesnt need to have a dialtone when is doing the system crossing. But does Asterisk supports QSIG? What are people using to connect Asterisk with the PBXs? QSIG, ETSI or something else? Thanks João ___ 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] Users handbook
Get the handbooks pdf's files, open and modify it regarding your configuration, and put that on a web page accessible for everyone on the office. You can look for the files on Voip-info.org Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew M Stemen Sent: Wednesday, July 06, 2005 6:29 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Users handbook I'm planning on implementing an Asterisk system at a couple of small offices, and a couple of homes, in the near future... and I don't have any documentation yet. What you're suggesting sounds wonderful, to me. I would contribute, if I had anything... but making it an inclusive manual would be a good idea, I think... you can always edit/remove sections. :) Andrew M Stemen [EMAIL PROTECTED] http://www.andrewmstemen.com Chris Mason (Lists) wrote: Mark Phillips wrote: This is somewhat unique to the site installation. For example, I don't have *69 programmed at my site because frankly there's no need for it with the Cisco 7960's. I do however have an automatic conference booking utility and a speaking clock. Not often found in smaller sites. I think you are on your own here. If one is implementing an Asterisk solution in an office scenario, it has to have fairly similar features to another Asterisk installation. It's easy enough to edit and remove the parts that are different. What I am suggesting is a comprehensive Here's everything Asterisk can do out of the box document, change or remove what doesn't apply. Let me know if any of you want to pool the work we have already done, I will compile to a complete document and post on the wiki. ___ 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 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] how to set language in capi
On Wed, 2005-07-06 at 15:51 +0200, [EMAIL PROTECTED] wrote: I found a german forum, and it seems to be a common problem ( I don't undertand a lot of german, anyway) ( cfr: http://www.ip-phone-forum.de/forum/viewtopic.php?t=17639) I also try to put exten = .,1,SetLanguage(it) in some places, but no result According to that forum it should work with SetLanguage in a context that covers incoming calls from the ISDN card: [incomingisdntrunk] exten = _.,1,SetLanguage(it) exten = _.,2,something Notice ^^^ (the _ in front of the .) Make sure you have your italian voice prompts installed correctly. Regards, Patrick ___ 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: FW: [Asterisk-Users] ETSI or QSIG
But doesnt Asterisk supports QSIG already? I just whant to know how to configure it. João George Lin wrote: Joao, We have developed some QSIG stack over asterisk. It will be a paid system. would you be interested in ? Regards George -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Joao Pereira Sent: Wednesday, July 06, 2005 7:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ETSI or QSIG Hello to all, Does Asterisk support QSIG? and the configuration is in capi.conf? and if it supports it, do you have samples of the configuration? I had my Asterisk connecting to a Siemens PBX with ETSI and it was working fine, but peolpe said to me that QSIG could implement more features and turn the calls between the two systems transparent for the users. And I read that QSIG could take the caller name and doesnt need to have a dialtone when is doing the system crossing. But does Asterisk supports QSIG? What are people using to connect Asterisk with the PBXs? QSIG, ETSI or something else? Thanks João ___ 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 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] Re: how to set language in capi
Ok , I solved by myself just put [from-pstn-custom] exten = s,1,SetLanguage(it) in extensions_custom.conf Andrea -- I am trying to use language=it in asterisk I downloaded the sound package and installed it I added country=it in indications.conf language=it in sip.conf language=it in iax2.conf everything ok in call from sip and from iax The problem arises in outside call, coming trom CAPI Trunk I try language=it in capi.conf: no result: always language=en I found a german forum, and it seems to be a common problem ( I don't undertand a lot of german, anyway) ( cfr: http://www.ip-phone-forum.de/forum/viewtopic.php?t=17639) I also try to put exten = .,1,SetLanguage(it) in some places, but no result I am running asterisk 1-0-9 any help will be greatly appreciated Andrea ___ 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] ETSI or QSIG
On Wed, 2005-07-06 at 15:05 +0100, Joao Pereira wrote: Hello to all, Does Asterisk support QSIG? and the configuration is in capi.conf? and if it supports it, do you have samples of the configuration? QSIG is not an option in capi.conf. It is an option in the configuration of my Eicon Diva Server BRI card which is used by chan_capi asterisk. So I guess you could use it to connect to the Siemens. I don't have a sample config. But does Asterisk supports QSIG? Yes. Obviously you could have googled for this info yourself. You may want to do that first next time you have a question... Set signalling to qsig in zapata.conf: http://lists.digium.com/pipermail/asterisk-users/2005-February/091109.html Other info: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf http://www.voip-info.org/tiki-index.php?page=Asterisk+legacy+integration Regards, Patrick ___ 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] I was mistaken about Areski: he does relply to mails and help people.
Cher Areski, Je m'excuse si je me suis trompé sur vos intentions. Cependant j'aimerais souligner que je ne suis pas le seul a avoir trouvé l'installation de votre application un peu trop compliquéeet les instructions du "Idiots Guide" imprécises.Il faut dire que Linux n'est dejà pas facile pour un ancien de Microsoft comme moi. Par conséquent lorsque les instructions ne sont incorectes et que malgré 2 jours d'effort je n'y arrive pas, je suis découragé. Je vous demande des excuses car il me semble que vous prenez vraiment le temps d'aider les gens. Je vais vais reessayer d'installer AreskiCC caril me semble que s'est la meilleure application avec toutes les fonctions qu'il faut (entre-temps j'avais re-installé ma machine et j'utilisais AstCC) . Je vais regarder Register Globals sous php. Je vais copier cette lettre d'excuses sur la liste Digium et le wiki. UNE FOIS DE PLUS JE 'M'EXCUSE. Si je parviens a installer, je partagerais mon experience sous RedHat9 avec tous les details. Sincerement Votre JM Khubeka Sell on Yahoo! Auctions - No fees. Bid on great items.___ 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] how to set language in capi
thank you ! I solved by myself in a different (slightly different) way Now I understand why my first solution didn't work: I didn't put the _ in front of the . thank you again, Andrea Patrick [EMAIL PROTECTED] .xs4all.nlTo Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 06/07/2005 16.21 Re: [Asterisk-Users] how to set language in capi Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com On Wed, 2005-07-06 at 15:51 +0200, [EMAIL PROTECTED] wrote: I found a german forum, and it seems to be a common problem ( I don't undertand a lot of german, anyway) ( cfr: http://www.ip-phone-forum.de/forum/viewtopic.php?t=17639) I also try to put exten = .,1,SetLanguage(it) in some places, but no result According to that forum it should work with SetLanguage in a context that covers incoming calls from the ISDN card: [incomingisdntrunk] exten = _.,1,SetLanguage(it) exten = _.,2,something Notice ^^^ (the _ in front of the .) Make sure you have your italian voice prompts installed correctly. Regards, Patrick ___ 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 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] ETSI or QSIG
Thanks for the help. I also have a Eicon Diva Server BRI and I know it can be used with chan_capi and asterisk, but the QSIG configuration is not direct. Of course I googled before asking to the list and I didnt found any direct explanation if QSIG is supported. Voip-info.org sais that zapata.conf is for configuration of Digium cards I also searched the list for previous statments about QSIG and I read that it isnt fully supported. If you re using an Eicon Diva Server BRI, what are you using to connect? ETSI, QSIG or someting else? Thanks João Patrick wrote: On Wed, 2005-07-06 at 15:05 +0100, Joao Pereira wrote: Hello to all, Does Asterisk support QSIG? and the configuration is in capi.conf? and if it supports it, do you have samples of the configuration? QSIG is not an option in capi.conf. It is an option in the configuration of my Eicon Diva Server BRI card which is used by chan_capi asterisk. So I guess you could use it to connect to the Siemens. I don't have a sample config. But does Asterisk supports QSIG? Yes. Obviously you could have googled for this info yourself. You may want to do that first next time you have a question... Set signalling to qsig in zapata.conf: http://lists.digium.com/pipermail/asterisk-users/2005-February/091109.html Other info: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf http://www.voip-info.org/tiki-index.php?page=Asterisk+legacy+integration Regards, Patrick ___ 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 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] Did the Broadvoice patch break asterisk ? {Scanned}
I have never have the /ext work for me. register=1234:[EMAIL PROTECTED]/ext It has never worked for me. David On Wed, 2005-07-06 at 10:27 +0200, Christian Peter wrote: I dislike the statement in the bug reports you can easily add /ext to your register statement as a workaround because it simply does not work when having provider who redirects with sip 302 responses (eg. nikotel). Also can one tell me the reason for /sipgateid when registering at sipgate? It's not the extension but it does work. Greetings Christian Peter ___ 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 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] Epia C3 Linux
Yep, along with 6 other distros. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JD Austin Sent: Tuesday, July 05, 2005 5:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Epia C3 Linux Tried knoppix? Wiley Siler wrote: OK. Something is truly rotten in Denmark. I took the 2.5 inch drive out altogether and setup a regular 3.5 IDE drive with a CDROM. BIOS recognizes both. Try to install Redhat 9, it dies. Fedora Core 3 dies, kernel panic. How in Zeus Red Ripe Ass did you guys get this to install? Am I going to have to make a custom kernel? To recap This is a Via Mini-ITX board 800MHz Samuel 2 Processor AKA E-Series C3 (not Eden, this one has a fan) Thanks to all, Wiley PS. AstLinux bombed too From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Wiley Siler Sent: Tuesday, July 05, 2005 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have attempted FC3, RedHat 9, Mandriva 10, Gentoo 2005.0, Gentoo 2004.3, Debian Mini, Debian woody, and Windows XP. Nothing will install. All see the HDD. All attempt partitioning (XPO seemingly completes), none will install the OS. BIOS posts the correct HDD and all the installers see the HDD. All bomb out immediately after attempting to partition with the exception of Gentoo. The LIVECD will allow me to set a partition table but it dies when I attempt to apply filesystem ext3 to the root partition. I am officially stumped. Thanks for all the input everyone! Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux It installed directly from the FC3 dvd, no changes...no external drivers required From: Wiley Siler [mailto:[EMAIL PROTECTED]] Sent: Friday, July 01, 2005 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED]] Sent: Friday, July 01, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD AustinTwin Geckos Technology Services LLCemail: [EMAIL PROTECTED]http://www.twingeckos.comphone/fax: 480.288.8195 ___ 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] Dialplan configuration with Realtime
On 7/6/05, Gundemarie Scholz [EMAIL PROTECTED] wrote: Hello! Following the instructions on voip-ip.org I have implemented Realtime with MySQL for my Asterisk server. The individual extension configuration is managed in a table called extensions. Still I have to keep some data in the extensions.conf, namely the switch and the include statements. Is there a way to minimize that or completely get rid of them? No, but you can put extensions.conf into mysql via realtime static while using realtime extensions at the same time. If your goal is to keep everything in the database that will work. Chris ___ 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] Asterisk voicemail
Hi guys,
I'm new to Asterisk, so I'm hoping someone can guide me :-)
Currently, I am having the configuration as follows :
PSTN - Cisco router - Sip Express Router - Asterisk Voicemail
I'm able to get the part from PSTN to Sip Express Router working, but
I can't integrate Asterisk with Sip Express Router (SER).
Basically, SER does all the registering and forwarding of calls. I
need to implement the voicemail in Asterisk, whereby a user calls a
certain IP Phone, and if the user does not pick up the call in time,
the call is diverted to Asterisk's voicemail.
However, I am unable to get Asterisk to activate the voicemail upon
missed calls. Please kindly advise.
Regards,
YY
My current settings are as follows :
-
SER
-
1. ser.cfg (SER's config file)
-
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#
# --- global configuration parameters
# Uncomment these lines to enter debugging mode
debug=3
fork=yes
listen=202.122.25.106
log_stderror=yes
check_via=no# (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
fifo=/tmp/ser_fifo
# -- module loading --
loadmodule /usr/local/lib/ser/modules/sl.so
loadmodule /usr/local/lib/ser/modules/tm.so
loadmodule /usr/local/lib/ser/modules/rr.so
loadmodule /usr/local/lib/ser/modules/maxfwd.so
loadmodule /usr/local/lib/ser/modules/usrloc.so
loadmodule /usr/local/lib/ser/modules/registrar.so
loadmodule /usr/local/lib/ser/modules/exec.so
# - setting module-specific parameters ---
# -- usrloc params --
# store user location in memory, not using database
modparam(usrloc, db_mode, 0)
modparam(rr, enable_full_lr, 1)
# -- tm params --
# set time for which ser will be waiting for a final response;
# fr_inv_timer sets value for INVITE transactions,
# fr_timer for all others
modparam(tm,fr_inv_timer,15)
# - request routing logic ---
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
break;
};
if ( msg:len max_len ) {
sl_send_reply(513, Message too big);
break;
};
setflag(1);
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if(method!=REGISTER){
record_route();
};
# loose-route processing
if (loose_route()) {
route(1);
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if(uri != myself){
route(1);
break;
};
if (uri==myself) {
if (method==REGISTER) {
route(2);
break;
};
setflag(4);
# attempt handoff to PSTN
if (uri=~^sip:[EMAIL PROTECTED]) {## This assumes that
the caller is
log(1, Forwarding to PSTN); ##
registered in our realm
forward(10.10.10.3, 5060); ## Our
Cisco router
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup(location)) {
sl_send_reply(404, Not Found);
#acc_rad_request(404);
break;
};
# timeout occurred ... now to forward to Asterisk's voicemail
service
if(method == INVITE isflagset(4)) {
t_on_failure(1);
};
};
route(1);
}
# ---
# Route Processing
# ---
route[1]{
if(!t_relay()){
sl_reply_error();
};
}
route[2]{
if(!save(location)){
sl_reply_error();
}
}
# voicemail activation!!
#
failure_route[1] {
log(1,Activating voicemail!!\n);
forward(202.122.25.106, 5061);
}
---
ASTERISK
voicemail.conf
---
[default]
1012 =
RE: [Asterisk-Users] Epia C3 Linux
Rob, How in the world did you know that I just ran the memtest86 and it is nothing but error after error. Switched out the ram and I am getting no errors on memtest86 now. I am back in the saddle. Fedora Core 3 is installing as we speak Thank you! Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Thomas Sent: Tuesday, July 05, 2005 6:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux Sounds to me like bad RAM. Try running memtest (your Fedora CD has it, just type memtest at the cd boot prompt) --Rob From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Wednesday, 6 July 2005 10:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux OK. Something is truly rotten in Denmark. I took the 2.5 inch drive out altogether and setup a regular 3.5 IDE drive with a CDROM. BIOS recognizes both. Try to install Redhat 9, it dies. Fedora Core 3 dies, kernel panic. How in Zeus Red Ripe Ass did you guys get this to install? Am I going to have to make a custom kernel? To recap This is a Via Mini-ITX board 800MHz Samuel 2 Processor AKA E-Series C3 (not Eden, this one has a fan) Thanks to all, Wiley PS. AstLinux bombed too From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Tuesday, July 05, 2005 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have attempted FC3, RedHat 9, Mandriva 10, Gentoo 2005.0, Gentoo 2004.3, Debian Mini, Debian woody, and Windows XP. Nothing will install. All see the HDD. All attempt partitioning (XPO seemingly completes), none will install the OS. BIOS posts the correct HDD and all the installers see the HDD. All bomb out immediately after attempting to partition with the exception of Gentoo. The LIVECD will allow me to set a partition table but it dies when I attempt to apply filesystem ext3 to the root partition. I am officially stumped. Thanks for all the input everyone! Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux It installed directly from the FC3 dvd, no changes...no external drivers required From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ 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] Previously: Queue + optional URL
On the snom phones you could use the Action URL's to start some process when the phone receives a call. Nils On Tuesday 05 July 2005 23:52, [EMAIL PROTECTED] wrote: Does anybody know if there is an app that will cause similar to occur on users PC? I have a scenario where users will have snom phones on their desks. Ideally when their phone receives a call I need to popup a web browser with a specific url. Any ideas appreciated. Neil on 5/7/05 10:52 PM, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: voip technocrat a écrit : Hello list, Can any body say what Exactly optinal URL will be used in Queue. It states like this The optional URL will be sent to the called party if the channel supports it but when we will send it to the called user ? When the called user answers the call. and if we send is there any specific use ?. If the called user uses a softphone that supports this functionnality, the URL is loaded in a browser window. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ 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 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 -- snom technology AGGradestr. 46D-12347 Berlin Nils Ohlmeier mailto:[EMAIL PROTECTED] http://www.snom.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] Epia C3 Linux
It's a common (and commonly overlooked) problem and whenever there appears to be no logic behind irrational behavior, the RAM is the first place I look. Because the RAM is effectively changing the running program's code at the bit level, any and all actions are unpredictable, along with their results.-BryceOn Jul 6, 2005, at 08:25, Wiley Siler wrote: Rob, How in the world did you know that… I just ran the memtest86 and it is nothing but error after error….Switched out the ram and I am getting no errors on memtest86 now. I am back in the saddle. Fedora Core 3 is installing as we speak… Thank you! WileyFrom: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Rob Thomas Sent: Tuesday, July 05, 2005 6:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux Sounds to me like bad RAM. Try running memtest (your Fedora CD has it, just type ‘memtest’ at the cd boot prompt) --Rob From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Wiley Siler Sent: Wednesday, 6 July 2005 10:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux OK. Something is truly rotten in Denmark. I took the 2.5 inch drive out altogether and setup a regular 3.5 IDE drive with a CDROM. BIOS recognizes both. Try to install Redhat 9, it dies. Fedora Core 3 dies, kernel panic. How in Zeus’ Red Ripe Ass did you guys get this to install? Am I going to have to make a custom kernel? To recap… This is a Via Mini-ITX board 800MHz Samuel 2 Processor AKA E-Series C3 (not Eden, this one has a fan) Thanks to all, Wiley PS. AstLinux bombed too… From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Wiley Siler Sent: Tuesday, July 05, 2005 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have attempted FC3, RedHat 9, Mandriva 10, Gentoo 2005.0, Gentoo 2004.3, Debian Mini, Debian woody, and Windows XP. Nothing will install. All see the HDD. All attempt partitioning (XPO seemingly completes), none will install the OS. BIOS posts the correct HDD and all the installers see the HDD. All bomb out immediately after attempting to partition with the exception of Gentoo. The LIVECD will allow me to set a partition table but it dies when I attempt to apply filesystem ext3 to the root partition. I am officially stumped. Thanks for all the input everyone! Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux It installed directly from the FC3 dvd, no changes...no external drivers required From: Wiley Siler [mailto:[EMAIL PROTECTED]] Sent: Friday, July 01, 2005 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 LinuxDid it require any special work or did you just download the ISO for FC3 and install? Thanks,Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED]] Sent: Friday, July 01, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Epia C3 LinuxAnyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? ThanksWiley ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Crash without make valgrind
I'm having a little problem. I have a dial-plan with a lot of SetVar's and loops, and under certain circumstances (reproducible) it makes asterisk crash. Wanting to debug this, I compiled using make valgrind. But doing so, I eliminated the crashes and the dial-plan works perfectly. Now from what I understand, valgrind removes compiler optimisation to ease debugging. What kind of optimisation does it remove? Anybody know what could be happening to have a crashwith a standard make and not have it with valgrind? My original setup was a asterisk-1.0.7 emerged on gentoo. Tried updating to the 1.0.8 ebuild, and then tried the tarball for 1.0.9 on a 2.6.11 kernel. Thanks for your help -- Benjamin ___ 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] Send Variables over IAX
Does anyone know if it is possible to send variables over an IAX trunk? Is there a setting in iax.conf that allows this or is there another hack to allow this? Thanks in advance. Jeremiah ___ 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] Asterisk 1.1
How adventurous would a person have to be to try to use the 1.1 from cvs? I want to implement our phone system with the database connections built in, which as I understand is being made very easy in the 1.1 code that is under development. thanks, ___ 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] re: help debugging dialplan
hello all,
another desperate request for help debugging my dialplan...
from a certain extension i do the following:
DBput(CFIM/${CALLERIDNUM}=${CALLERIDNUM})
a NoOp to the console says
DBput: family=CFIM, key=2122022001, value=2122022001
and database show says
/CFIM/2122022001 : 2122022001
so far, so good.
but in a macro, when i try to get the data,
exten = s,1,DBGet(${DB(CFIM/${ARG1})
(ARG1 is 2122022001)
first, i get the following:
Jul 6 18:50:14 NOTICE[587]: pbx.c:1114
pbx_substitute_variables_helper: Error in extension logic (missing
'}')
and the CFIM variable is empty.
so, the following questions:
1. where does the } go? i know i'm missing one, but i don't know what to enclose
2. why isn't CFIM getting the variable from the DB?
anyone who can help me, i very much appreciate it.
thanks,
yair
p.s. when are DBGet and DBSet being deprecated?
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Re: [Asterisk-Users] ETSI or QSIG
On Wed, 2005-07-06 at 16:02 +0100, Joao Pereira wrote: Voip-info.org sais that zapata.conf is for configuration of Digium cards Yup you need a T1/E1 card for qsig stuff in zapata.conf. Don't know how it is done with an Eicon Diva Server card, if possible at all. I also searched the list for previous statments about QSIG and I read that it isnt fully supported. Can't really tell as I don't have any experience with hooking up a PBX through qsig. If you re using an Eicon Diva Server BRI, what are you using to connect? ETSI, QSIG or someting else? I use my Eicon Diva Server card to hook up my ISDN/BRI line through ETSI and capi.conf to asterisk. Regards, Patrick ___ 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] Polycom distributor in the UK ?
according to Polycom the IP301,IP501 are not going to be released in the UK (EMEA) until Q4 this year... try calling hardware.com if they have them available. __ 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] Simpletelecom dead?
Bruce, I too am interested in the telephone number for SimpleTelecom, as my company had put quite a large prepayment to them. You said you posted the number on this list; I searched for all post by you and did not find the posting which contained a phone number. Would you be so kind as to please re post the phone number or give me a better clue as to how to find the number you called. Very much appreciated, S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Ferrell Sent: Tuesday, July 05, 2005 3:47 PM To: C F Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Simpletelecom dead? Oh puhlease! gimme a break... Go have a look at the archives... I kinda stick out all over the place. While you're away from the computer, get your tin-foil hat adjusted... maybe adjust your meds too C F wrote: Well so for all I know you work for sipmpletelcom.com and are just trying to cover up. On 7/5/05, Bruce Ferrell [EMAIL PROTECTED] wrote: tell ya what, when everyone posts all the private backdoor numbers they have, I'll post that one C F wrote: You did send it to the list, but I'm asking you to post the phone number you used to call get a hold of someone. On 7/5/05, Bruce Ferrell [EMAIL PROTECTED] wrote: I thought I sent it out to the list when I sent it to you... I guess it didn't go C F wrote: Can you please share this with everybody? who did you speak to? on which number did you get ahold of them? ___ 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] Stale nonce received?
Ian White wrote: The use of the nonce looks right to me. Can somebody point out what is going wrong here? Yes, I agree, it looks correct. However, what version of Asterisk are you testing against? Current CVS HEAD adds 'stale=true' to the 401 response, and I don't see that in your trace. If you are not testing against the most recent CVS HEAD, please do so. ___ 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] Re: Asterisk 1.1
In article [EMAIL PROTECTED], Chris Gamble [EMAIL PROTECTED] wrote: How adventurous would a person have to be to try to use the 1.1 from cvs? I want to implement our phone system with the database connections built in, which as I understand is being made very easy in the 1.1 code that is under development. I really wish 1.2 Stable would be released. There are so many new features in HEAD, and several modules that are incompatible with 1.0, (e.g. some of the H.323 stacks), that I would love to start using the current codebase without worrying about it still being a moving target and the possibility of stuff being added that breaks things. Anyone here in the know about when HEAD will be branched to 1.2? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Snom phones - any advice
Hi We are about to buy several Snom phones. Does anyone have warnings or advices against these phones ? Our finalists were Cisco, Polycom and Snom. We will be using only the SIP protocol. Thanks Patrick ___ 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] Snom phones - any advice
Greetings, We are just finishing a roll-out of 25 of the SNOM 190s with a SNOM 220 w/sidecar. The only gotcha that I found is that the SNOM 190s use rfc2833 for a default dtfm mode and not inband which is the default for the asterisk server. I haven't ironed out the Mass deployment functionality yet, but will do so. So with a tftp server running you should be fine. Generally speaking, of course. RandyW Patrick Fortin wrote: Hi We are about to buy several Snom phones. Does anyone have warnings or advices against these phones ? Our finalists were Cisco, Polycom and Snom. We will be using only the SIP protocol. Thanks Patrick ___ 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 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] Users handbook
Tzafrir Cohen wrote: Hey, compile is what computers do, not humans. :-) Is there any existing program that could either from /etc/asterisk or using the manager interface figure out enough about the asterisk configuration to generate such a manual (using some templating engine)? I've often thought it would be nice to have a program that takes all the extensions asks a question about which zap are extensions and which are lines, whihc are your trunks etc and then compiles a 1 page extension list. Even just for myself this would be usefull as I have totally run out of numbering space on one of my dev machines and often move around/rename extensions. The idea would be to do a show dialplan via the manager and parse it for various applications/dial lines. Obviously the applications would be easy to do, but the dial lines would require user interaction. I guess you would also need some way to set the perspective from which it is viewed. I.E. Looking from the internal context you would have available X. I don't think people would need a list of the IVR numbers etc from outside, although if you could select a start context you could do this too. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] URGENT: hardware spesifications needed
Lars Boegild Thomsen wrote: On Wednesday 06 July 2005 15:09, Erdem HAKİ wrote: I need to set up Asterisk to serve and register for 1000 users(not simultaneus). What kind of specifications do my server need. Well - the interesting number is the number of simultaneous users really and to some degree the type of calls (sip-sip or for example sip-iax or sip-capi). Ah and transcoding maybe? What codec will the calls come in and go out as, what will the users be doing, will the users be reinviting? In large scenarios with SIP I usually find it is better to set Asterisk up as an Application Server, with SER reinviting calls between users and RTPProxy taking care of the media stream for difficult NATs. You might want to check out some of the work that Zoa has done over at http://www.asteriskguru.com/ with regard to dimensioning and also the wiki page on asterisk dimensioning. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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-CVS-HEAD locks up on 'reload' from CLI (sometimes)
rm -rf /usr/include/asterisk do a fresh checkout and try again. /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jul 5, 2005, at 12:02 PM, Chris Coulthurst wrote: Lately when I issue a 'reload' from the CLI, I find that it will sometimes hang forever, completely locked up. I can press enter and see the CLI prompt move, but no commands are taken. top shows asterisk eating everything up: PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME CPU COMMAND 20669 root 25 0 10068 9.8M 5392 R88.4 1.9 1:02 0 asterisk 20877 root 15 0 1124 1124 896 R 0.3 0.2 0:00 0 top 1 root 15 0 448 448 396 S 0.0 0.0 0:04 0 init 2 root 15 0 00 0 SW0.0 0.0 0:01 0 keventd 3 root 15 0 00 0 SW0.0 0.0 0:00 0 kapmd 4 root 34 19 00 0 SWN 0.0 0.0 0:00 0 ksoftirqd/0 Most recent add-ons have been Speex and h323. I just installed h323 today and this has been going on for about a week, so I know its not that, but I can't remember if this was happening before Speex or not. Anyone have any similar happenings? Chris Coulthurst [EMAIL PROTECTED] ___ 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 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] Previously: Queue + optional URL
[EMAIL PROTECTED] wrote: Does anybody know if there is an app that will cause similar to occur on users PC? Maybe have a look at the Flash Operator Panel. It has the capability for web pops, and can even be shrunk so you can't see it if neccesary. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] problem with iax2 and 2 peers behind nat
Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan, itried with 2 firefly and the error its the same, it could be because the 2 peers are going to the internet with the same ip addres(both behind nat)? if i conect both peers in the same lan there is no problem so i think it cpuld be a problem with nat, i dont konw if i had to change some configuration in iax.conf. Thanks. Juan Lopez. [EMAIL PROTECTED] Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es ___ 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] OT: Congrats, Europe!
http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150tid=147tid=136 http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/ begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ 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
Re: [Asterisk-Users] chan_capi 0.5.3 asterisk HEAD 2005/07/04 undefined symbol error
On Wed, 2005-07-06 at 13:34 +0200, Sergio Chersovani wrote: [snip] stop asterisk, rm /usr/lib/asterisk/modules/* rm /usr/include/asterisk/* cd asterisk make clean make upgrage cd chan_capi-cm-0.5.3 make clean make install now run asterisk again Thanks Sergio. I removed everything, updated to latest HEAD and all is well again. Regards, Patrick ___ 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] Re: Asterisk 1.1
Tony Mountifield wrote: Anyone here in the know about when HEAD will be branched to 1.2? Very soon. We are actively trying to clean up the open bugs and issues so we can prepare a release candidate. ___ 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] Re: app_conference and AGI
The README in the source code states: app_conference doesn't have DTMF-activated features or anything like that. I'm curious how you got audio working on your compliation. I am running CVS HEAD + app_conference in a Xen virtual machine. I can connect to the channel but there is no audio. Here are my configs and Asterisk's output: http://lee.97montrose.org/hacking/app_conference.txt -- Lee Azzarello Network Engineer Progressive Solutions +1 212 937 8939 ___ 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] problem with iax2 and 2 peers behind nat
Juan, That is not going to work. Asterisk shouldnt be behind a NAT to get registration of boxes behind NAT. Put the asterisk on DMZ zone of their router to make that happen. Carlos Alperin [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRID Sent: Wednesday, July 06, 2005 12:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] problem with iax2 and 2 peers behind nat Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan, itried with 2 firefly and the error its the same, it could be because the 2 peers are going to the internet with the same ip addres(both behind nat)? if i conect both peers in the same lan there is no problem so i think it cpuld be a problem with nat, i dont konw if i had to change some configuration in iax.conf. Thanks. Juan Lopez. [EMAIL PROTECTED] Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es ___ 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] Epia C3 Linux
On Tue, Jul 05, 2005 at 08:39:15PM -0400, Michael Stahl wrote: Take a look at the via arena web site. Your processor may look like a 586 to the installer but may not support all of the instructions (causing a crash). The via arena site gives instructions on how to compile and get it installed on your processor! (I have the C3 Nehemiah processor so I didn't need to recompile) You'd expect it to blow up with Illegal instruction then and not with a segfault. If you fear this may be a 386 issue, get the Debian Sarge netinst. It has only i386 kernel. Or try current Rapid, which will also give you an Asterisk installation. But my suspect here is the memory: have you tried memtest? a number of of installers and live-cds now come with it as a boot option. Also note that most installers have a shell available on an alternative terminal (usually console no. 2). It used to be very limited, but the one on current debian (sarge) installer is actually quite usable and even has tab completion for path names (thanks busybox). -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] ISDN PRI No Audio
Our setup: We have a DS3 from Global Crossing terminating into a Adtran MX2800 M13 mux. From there, groups of 4 T1's run into T410P digium cards to 7 individual servers. Each trunk is configured as ISDN PRI, B8ZS/ESF, D-channel being chan 96 with B-channels of 1-95 (we're using NFAS). The D channel is up and there are no alarms. We see the connection on the console from the incoming user as seen below, if the user hangs up it disconnects properly, we can also do a zap destroy and disconnect the user. It seems that bi-directional communication is alright. We were running 1.0.7 but upgraded to CVSHEAD to see if a fix existed. Console Output: -- Accepting call from '414944' to '80094042XX' on channel 2/24, span 4 -- Executing Wait(Zap/48-1, 3) in new stack -- Executing Answer(Zap/48-1, ) in new stack -- Executing Playback(Zap/48-1, tt-monkeys) in new stack -- Playing 'tt-monkeys' (language 'en') -- Executing Read(Zap/48-1, TEST||2) in new stack -- Accepting a maximum of 2 digits. -- User entered nothing. -- Executing SayDigits(Zap/48-1, ) in new stack -- Playing 'digits/1' (language 'en') -- Playing 'digits/1' (language 'en') -- Playing 'digits/1' (language 'en') -- Playing 'digits/1' (language 'en') -- Executing Hangup(Zap/48-1, ) in new stack == Spawn extension (default, 80094042XX, 6) exited non-zero on 'Zap/48-1' -- Hungup 'Zap/48-1' The Problem: No audio and no DTMF tones are passed. We cannot hear the test audio, we cannot send digits back and we cannot hear the digits being said. GBLX setup a tap and said we were not sending them any audio at all so we're fairly certain this problem is on our end and not theirs. We're at a loss here and can't really figure out what's wrong. Can someone provide some insight into this problem? Thanks -- ~Andy Brezinsky ___ 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] Epia C3 Linux
This did wind up being a matter of memory... Thanks, Wiley W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Wednesday, July 06, 2005 10:14 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Epia C3 Linux On Tue, Jul 05, 2005 at 08:39:15PM -0400, Michael Stahl wrote: Take a look at the via arena web site. Your processor may look like a 586 to the installer but may not support all of the instructions (causing a crash). The via arena site gives instructions on how to compile and get it installed on your processor! (I have the C3 Nehemiah processor so I didn't need to recompile) You'd expect it to blow up with Illegal instruction then and not with a segfault. If you fear this may be a 386 issue, get the Debian Sarge netinst. It has only i386 kernel. Or try current Rapid, which will also give you an Asterisk installation. But my suspect here is the memory: have you tried memtest? a number of of installers and live-cds now come with it as a boot option. Also note that most installers have a shell available on an alternative terminal (usually console no. 2). It used to be very limited, but the one on current debian (sarge) installer is actually quite usable and even has tab completion for path names (thanks busybox). -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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 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-CVS-HEAD locks up on 'reload' from CLI (sometimes)
On Wed, Jul 06, 2005 at 11:48:27AM -0500, Brian West wrote: /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jul 5, 2005, at 12:02 PM, Chris Coulthurst wrote: Lately when I issue a 'reload' from the CLI, I find that it will sometimes hang forever, completely locked up. I can press enter and see the CLI prompt move, but no commands are taken. top shows asterisk eating everything up: What exactly is it doing? attach to it with strace (strace -p) or with ltrace to get some clues. [snip] rm -rf /usr/include/asterisk do a fresh checkout and try again. Mind giving some details, apart from a reinstall kind of advice? To allow prevension? Or for those who have some extra files in /usr/include/asterisk Thanks -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] URGENT: hardware spesifications needed
Why not do your research instead of asking the list to do it for you lazy ass! /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jul 6, 2005, at 2:09 AM, Erdem HAKİ wrote: Hello; I need to set up Asterisk to serve and register for 1000 users(not simultaneus). What kind of specifications do my server need. For example: Xenon processor 1 GB RAM 120 GB HDD etc... Thanks for your help.. Erdem HAKI – [EMAIL PROTECTED] ___ 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 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] zaptel missing /dev/zap after FC3 update
I have a P4 HT system running * on a 2.6.9-1.667smp kernel (Fedora Core 3). Everythng was testing out and the configuration was working. After running YUM update, kernel 2.6.11-1.35_FC3smp was installed. Now Zaptel cannot find /dev/zap. Waiting for zap to come online...Error: missing /dev/zap! I have already recompiled zaptel, libpri, and asterisk after changing the /usr/src/linux-2.6 symbolic link (linux-2.6 - /lib/modules/2.6.11-1.35_FC3smp/build/). There is only a TDM22b installed I reverted to the older kernel, recompiled and have the same issue. Any thoughts? ___ 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] Re: app_conference and AGI
On Wed, 2005-07-06 at 13:00 -0400, Lee Azzarello wrote: [snip] I can connect to the channel but there is no audio. Here are my configs and Asterisk's output: http://lee.97montrose.org/hacking/app_conference.txt Maybe it is a codec problem: translate.c:134 ast_translator_build_path: No translator path from unknown to unknown translate.c:134 ast_translator_build_path: No translator path from unknown to alaw Did you try using a codec that is supported by asterisk and each phone. alaw or ulaw would be a good start. Regards, Patrick ___ 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] Some problems setting outgoing PRI Origination Number
Hello,
Quick Diagram:
Telco-PRI - Asterisk - Norstar PRI - Norstar PBX
(DMS100) (TE405P) (DMS100)
|
|
V
Cisco 7960G
(SIP)
I'm trying to change the Origination Number on my outgoing PRI, and running
into a weird
problem. If I make a call from a SIP extension off asterisk using the
following context:
[from-sip]
exten = 800,1,Answer
exten = 800,2,SetCallerID(6132718)
exten = 800,3,Dial(Dial(${TRUNK-TELCO}/5551234)
I am able to change the Origination Number!
Protocol Discriminator: Q.931 (8) len=39
Call Ref: len= 2 (reference 175/0xAF) (Originator)
Message type: SETUP (5)
[04 03 80 90 a2]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability:
Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: u-Law (34)
[18 03 a9 83 95]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Ext: 1 Channel: 21 ]
[6c 0c 21 83 36 31 33 32 37 31 38 38 35 33]
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan
(E.164/E.163) (1)
Presentation: Presentation allowed of network
provided number (3)
'613271' ]
[70 08 a1 32 35 35 30 30 34 38]
Called Number (len=10) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan
(E.164/E.163) (1) '5551234' ]
However if I call from behind the channel bank, I am unable to change the
number.
[from-norstar]
exten = 800,1,Answer
exten = 800,2,SetCallerID(6132718)
exten = 800,3,Dial(Dial(${TRUNK-TELCO}/5551234)
Protocol Discriminator: Q.931 (8) len=39
Call Ref: len= 2 (reference 176/0xB0) (Originator)
Message type: SETUP (5)
[04 03 80 90 a2]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability:
Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: u-Law (34)
[18 03 a9 83 95]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Ext: 1 Channel: 21 ]
[6c 0c 21 ff 36 31 33 32 37 31 38 38 35 33]
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan
(E.164/E.163) (1)
Presentation: Unknown (127) '613271' ]
[70 08 a1 32 35 35 30 30 34 38]
Called Number (len=10) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan
(E.164/E.163) (1) '5551234' ]
Any ideas why I can set it using SIP extension only?
Slackware 10.0
Asterisk 1.0.9
TE405P
PB
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Re: [Asterisk-Users] OT: Congrats, Europe!
Vahan Yerkanian wrote: http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150tid=147tid=136 http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/ Hear, hear! I'm not in general much of a praying kind of person, but do pray that someday the US will wake up to the damage that the software patent madness is doing to innovation, and do something about it before it's too late. B. ___ 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] converting windows .wav to .gsm
HI ALL; I have problem converting a windows .wav file to .gsm format by Sox. Could anyone help. Cheers, Mohammad ___ 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] ETSI or QSIG
you re using an Eicon Diva Server BRI, what are you using to connect? ETSI, QSIG or someting else? I use my Eicon Diva Server card to hook up my ISDN/BRI line through ETSI and capi.conf to asterisk. I had that configuration too, but isnt QSIG better? because QSIG can send the caller name and provide more services. The calls passing with QSIG will be transparent, and dont have dialtones e in the middle of the number dialing. I dont know If I should continue in the hard task of configuring QSIG or I just give it up for ETSI Does someone knows if the QSIG task is reachable and if it is worth the time? João Pereira Regards, Patrick ___ 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 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] Maximum Number of Mailboxes in Asterisk
Valued Colleagues, Can anyone tell me whether the Maximum Number of Mailboxes in Asterisk is hardcoded or configurable?! I suppose the maximum number of allowed voice messages per mailbox is hardcoded as #define MAXMSG 100 in ~asterisk/apps/app_voicemail.c Thanks ramin ___ 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] Crash without make valgrind
Well you could get a backtrace of the core to give us a little bit of clue why its crashing! /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jul 6, 2005, at 10:44 AM, Benjamin Lawetz wrote: I'm having a little problem. I have a dial-plan with a lot of SetVar's and loops, and under certain circumstances (reproducible) it makes asterisk crash. Wanting to debug this, I compiled using make valgrind. But doing so, I eliminated the crashes and the dial-plan works perfectly. Now from what I understand, valgrind removes compiler optimisation to ease debugging. What kind of optimisation does it remove? Anybody know what could be happening to have a crashwith a standard make and not have it with valgrind? My original setup was a asterisk-1.0.7 emerged on gentoo. Tried updating to the 1.0.8 ebuild, and then tried the tarball for 1.0.9 on a 2.6.11 kernel. Thanks for your help -- Benjamin ___ 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 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] chan_sccp new realease
http://chan-sccp.berlios.de/ 20050705 ftp://ftp.berlios.de/pub/chan-sccp/chan_sccp-20050705.tar.gz - Added support for distinctive rings on stable: SetVar(ALERT_INFO=inside) or outside or feature on head: SetVar(_ALERT_INFO=inside) or outside or feature - Added support for native transfer incoming call-answer- hit transfer (incoming call is now on hold and marked as a transfer) - dial a new number - hit transfer (you can wait to talk to the user (consultative transfer) or just hit transfer (blind transfer) - fixed a segmentation fault when dialing a not configured line (Thanks Mark for the report) - fixed switching lines softkey state, hold/resume issues (thanks to Stefan and Joseph) - fixed segmentation faults on hangups For testers: http://www.voip-info.org/wiki-Asterisk+debugging run asterisk -vvvcg I need the bt full log Sergio Chersovani ___ 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] Retrieving number of messages in a mailbox by an application
Valued Colleagues, Can anyone tell me how the asterisk keeps track of the number of existing old (read) and new (unread) messages in a mailbox? Is there a database table or somewhere else from which this data can be retrieved by an application? Thanks ramin ___ 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] converting windows .wav to .gsm
Can you be a bit more specific as to what the problems is? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mohammad Sent: July 6, 2005 2:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] converting windows .wav to .gsm HI ALL; I have problem converting a windows .wav file to .gsm format by Sox. Could anyone help. Cheers, Mohammad ___ 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] FW: Maximum Number of Mailboxes in Asterisk
Valued Colleagues, Can anyone tell me whether the Maximum Number of Mailboxes in Asterisk is hardcoded or configurable?! I suppose the maximum number of allowed voice messages per mailbox is hardcoded as #define MAXMSG 100 in ~asterisk/apps/app_voicemail.c Thanks ramin ___ 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 1.1
Chris Gamble wrote: How adventurous would a person have to be to try to use the 1.1 from cvs? I want to implement our phone system with the database connections built in, which as I understand is being made very easy in the 1.1 code that is under development. thanks, Not adventurous at all. We use -HEAD in a production environment with about 80 phones on it. No crashes in 4 days. 4 days ago I restared to update libpri/zaptel. -Matthew ___ 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] Incoming 800-number over IAX - first few words are cut-off
I have an incoming 800-number over IAX from Teliax and I'm experiencing the large packet loss on connection. When a call comes in there is no ring tone and the first few words of the welcome message are cut off, regardless of the delay I set. Standard call (not 800-number) coming over IAX with the same provider works just fine only the tall free number. So it seems there are some packet loss only at the beginning, as the call quality sounds just fine, even when I compile something and CPU is at 99% use, there is no packet drop during conversation only on connection of tall free number. -- #Joseph ___ 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] OT: Congrats, Europe!
Juhu, Jippi Jippi Yeah! I am going to dance all night. Cheers S. At 13:04 06.07.2005 -0500, Brian Capouch wrote: Vahan Yerkanian wrote: http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150tid=147tid=136 http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/ Hear, hear! I'm not in general much of a praying kind of person, but do pray that someday the US will wake up to the damage that the software patent madness is doing to innovation, and do something about it before it's too late. B. ___ 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 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] Some problems setting outgoing PRI Origination Number
Paul Belanger wrote: [6c 0c 21 ff 36 31 33 32 37 31 38 38 35 33] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Unknown (127) '613271' ] It _is_ being changed, but the presentation/restriction settings are set to an odd value. Look at the SetCallerPres() app (or CALLERPRES variable in CVS HEAD) to set the outgoing presentation value to something that will work. ___ 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] quadBRI form junghanns.net
Hello, Is anybody there using quadBRI form Junghanns.net with Asterisk ? I would like to order that card but first would like to hear some opinions. Thank you in advance Bartosz ___ 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] converting windows .wav to .gsm
I use wavepad all the time on my windows box. I've never had a problem using it to convert and edit the files. Darren Wiebe [EMAIL PROTECTED] mohammad wrote: HI ALL; I have problem converting a windows .wav file to .gsm format by Sox. Could anyone help. Cheers, Mohammad ___ 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 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] zaptel missing /dev/zap after FC3 update
On Wed, 2005-07-06 at 10:56 -0700, Howard Ratzlaff wrote: I have a P4 HT system running * on a 2.6.9-1.667smp kernel (Fedora Core 3). Everythng was testing out and the configuration was working. After running YUM update, kernel 2.6.11-1.35_FC3smp was installed. Now Zaptel cannot find /dev/zap. Waiting for zap to come online...Error: missing /dev/zap! I have already recompiled zaptel, libpri, and asterisk after changing the /usr/src/linux-2.6 symbolic link (linux-2.6 - /lib/modules/2.6.11-1.35_FC3smp/build/). There is only a TDM22b installed I reverted to the older kernel, recompiled and have the same issue. Any thoughts? README.udev -- Dave Cotton [EMAIL PROTECTED] ___ 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] Re: TDM04B problems
Elwin Andriol wrote: Don't know if this will help you any further, but. After some trouble with IRQ sharing mayhem we solved our little problem by tinkering the linux kernel. I forgot the names of the actual modules, but after disabling modules for APIC support and something about IRQ sharing or APIC-IO or such, we effectively disables the APIC from handling IRQ's. I'm not so sure that disabling the APIC only from the BIOS setup will do it (it did not in our 'MSI'-case). We had to disble the APIC from within the BIOS setup also, otherwise our system crashed at boot. After doing so our /proc/interrupt didn't show any 'IO-APIC-level' and 'IO-APIC-edge' containing lines but only 'XT-PIC' containing lines. After that, our TDM04B allways got it's own IRQ and the mayhem never returned. If you're in real nead of those module names, let me know. I've got some notes somewhere at the bottom of the 3 feet tall pile besides my desk that says 'To be examned further someday' The /proc/interrupts interface is already showing everything as being assigned by XT-PIC. I'll see what I can do with the kernel config looking around at what you've suggested. Hopefully I won't need to ask you to dig through your pile. ___ 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] zaptel missing /dev/zap after FC3 update
Did you try tailing the /var/log/dmesg to see what happened when you loaded zaptel and wctdm with modprobe? Check that /etc/modprobe.conf still contains the correct module entries. Does /lib/modules/2.6.11-1.35_FC3smp/misc still contain and correct wctdm.ko files? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Howard Ratzlaff Sent: July 6, 2005 1:57 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] zaptel missing /dev/zap after FC3 update I have a P4 HT system running * on a 2.6.9-1.667smp kernel (Fedora Core 3). Everythng was testing out and the configuration was working. After running YUM update, kernel 2.6.11-1.35_FC3smp was installed. Now Zaptel cannot find /dev/zap. Waiting for zap to come online...Error: missing /dev/zap! I have already recompiled zaptel, libpri, and asterisk after changing the /usr/src/linux-2.6 symbolic link (linux-2.6 - /lib/modules/2.6.11-1.35_FC3smp/build/). There is only a TDM22b installed I reverted to the older kernel, recompiled and have the same issue. Any thoughts? ___ 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 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] Bounced mail apologies
My apologies for any bounced mail from me today. My mail server was having a bit of a fit. MARK. ___ 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
