[Asterisk-Users] Digits being dropping when dialing from certain analog phones
FC2, Asterisk 1.0.0, Zaptel 1.0.0
TDM400P Port 1 FXS Port 4 FXO
Standard analogue handset plugged in with pstn line.
Problem:
I have 2 analog phones that I use, when plugged directly into pstn line
both phones work perfectly, dialing no issues. When I plug the handsets
into the TDM400P, one works perfectly the other drops random numbers.
Its like the tone is slightly different on the second handset and its
not picking up some numbers (12356 it seems). Is there a way to adjust
the tone detection, make it more sensitive?
Keys dialed from handset were
9 0418800185
I tried hitting the keys slowly as well as at my normal speed, all tones
are heard in the handset for all numbers.
Error in asterisk -vvvgc
-- Starting simple switch on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, Zap/g2/088008) in new stack
-- Called g2/088008
-- Zap/4-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/4-1
-- Hungup 'Zap/4-1'
== Spawn extension (internal, 9088008, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, Zap/g2/0488008) in new stack
-- Called g2/0488008
-- Zap/4-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/4-1
-- Hungup 'Zap/4-1'
== Spawn extension (internal, 90488008, 1) exited non-zero on
'Zap/1-1'
-- Hungup 'Zap/1-1'
/etc/zaptel.conf
fxols=1
fxsls=4
Loadzone=au
/etc/zapata.conf
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
signalling=fxo_ls
callgroup=1
pickupgroup=1
immediate=no
context=internal
busydetect=yes
callerid=James Bean690 ;assuming extension 690
mailbox=690 ;stutter tone for voicemail - you can
use an optional context here
transfer=yes
channel=1
group=2
signalling=fxs_ls
context=pstn
channel=4
/etc/asterisk/extensions.conf
[pstn]
exten = s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten = s,2,Dial(Zap/g1,45,t) ;Dial the group=1 zap card mod above
exten = s,3,Hangup
[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone
exten = _9X.,1,Dial(Zap/g2/${EXTEN:1})
exten = _9X.,2,Congestion()
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Re: [Asterisk-Users] Whoa.... I'm owned but found ??
I get this warning on zap channels: Sep 26 07:17:21 WARNING[1102232496]: chan_zap.c:5903 do_monitor: Whoa I'm owned but found (23)... Sep 26 07:45:53 WARNING[1102232496]: chan_zap.c:5903 do_monitor: Whoa I'm owned but found (23)... Sep 26 07:59:59 WARNING[1102232496]: chan_zap.c:5903 do_monitor: Whoa I'm owned but found (27)... Sep 26 08:18:35 WARNING[1102232496]: chan_zap.c:5903 do_monitor: Whoa I'm owned but found (28)... Sep 26 08:22:07 WARNING[1102232496]: chan_zap.c:5903 do_monitor: Whoa I'm owned but found (31)... Sep 26 08:26:27 WARNING[1102232496]: chan_zap.c:5903 do_monitor: Whoa I'm owned but found (31)... Sep 26 08:27:20 WARNING[1102232496]: chan_zap.c:5903 do_monitor: Whoa I'm owned but found (30)... Sep 26 08:44:13 WARNING[1102232496]: chan_zap.c:5903 do_monitor: Whoa I'm owned but found (24)... Sep 26 09:52:18 WARNING[1102232496]: chan_zap.c:5903 do_monitor: Whoa I'm owned but found (26)... Sep 26 10:14:26 WARNING[1102232496]: chan_zap.c:5903 do_monitor: Whoa I'm owned but found (24)... Don't cross messages between lists. Anyway, be more specific. - Original Message - From: shabanip To: [EMAIL PROTECTED] Sent: Saturday, September 25, 2004 12:02 PM Subject: [Asterisk-Users] Whoa I'm owned but found ?? I get this message at CLI. what does it mean? - shabanip -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - WellGate 3502a : ulaw/alaw only?
Greetings, I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. So far everything is working as expected, except the fact only ulaw and alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both sides after the call is established. Am I missing something here? WellTech claims to support even g729, and I see those in the embedded website's codec priority page. Is there something more to be done to enable g723.1/gsm codecs? P.S. 3502A is also affected by the registration bug, if you connect a phone to TEL2 jack and call someone, everything goes by the TEL1's account/password... regards, Vahan Yerkanian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK callerid Patch for the latest CVS
Hello! Is it possible for somebody to email me the patch for the UK callerid (for the X100P cards). I know that the TDM100P patches are included but .. I still use the X100Ps Many thanks Vassilis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * works, but after a few seconds audio always stops.
--On Sunday, September 26, 2004 02:29 +0200 Philipp von Klitzing [EMAIL PROTECTED] wrote: FAQ: Turn of silence suppression in X-Lite by setting Transmit silence to YES (in AUDIO settings). AHA! That's it! Works now. Not sure what silence suppression has to do with all but it works now. Tnx much! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Got SIP response 400 Bad Request ; Cisco 7940 inbound station/station call problem.
Hello Everyone, Ive been struggling with this issue for about two days; hopefully its something trivial that has been overlooked. Basically, I have a Cisco 7940 handset running the SIP 7.1 firmware, which can place outbound calls to any destination, however It can not receive calls from hard/soft phones on the network. Inbound calls through the IVR (Zap channel) are correctly routed, phone rings, no problems there. All phones on the network are using the G711u codec, are on RFC1918 address space, and do not traverse a firewall. For the life of me, I cant seem to find a logical answer for this one. Everything appears to be properly configured. Ive even tried with canreinvite=yes to see if the phones can talk directly to each other with the same results. Actual debugging output: -- Executing Dial(SIP/4002-1c9e, SIP/2000|20) in new stack -- Called 2000 -- Got SIP response 400 Bad Request back from 192.168.1.100 == No one is available to answer at this time Again, the problem is isolated to one specific handset, everything else operates perfectly. Any help in this matter would be greatly appreciated, thank you! Regards, Mark Brad Global ID Systems ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - WellGate 3502a : ulaw/alaw only?
On 26/09/2004 15:41 Vahan Yerkanian said the following: I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. welltech still hasn't responded to my complaint to them regarding their SIP registration process. in the interim however, i've patched asterisk 1.0 to handle this wellgate special case. i'll be emailling the patch over to you in a separate email. i run asterisk (0.9.0 and 1.0) on freebsd 4.10 too. alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both sides after the call is established. i've seen the same. the wellgate's do not support the GSM codec and stock asterisk does not do g723.1. you'd need to only use ulaw or alaw for them. i've got this in my wellgate sip.conf entries: disallow=all allow=ulaw allow=alaw -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] German Termination and DIDs
Alfred Nurnberger schrieb: Try sipgate.de. They have free DIDs in many german citys and their rate into Germany is very affordable (aprx. $0.02 / min.) ... Hi, I'm sure, they won't anymore. For it were Sipgate and Nikotel, who got those letters from RegTP, forbidding to give DIDs to non-local subscribers. Have a look at http://voipliste.de/privat.html for some voip providers! Most of them do offer local DID (to local) users. I assume, that some of them will soon offer local DIDs even from smaller german cities - I don't think that Sipgate or Nikotel will be the first offering smaller cities. Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Asterisk - WellGate 3502a : ulaw/alaw only?
Hello Dinesh, my welltech does support g729 g723.1 and g711u and a. It works like a charm here. BUT; the registration problem still exists (even after using your rc) -- Best regards, Dannymailto:[EMAIL PROTECTED] belGOnet.com a Euro-pictures division - internet solutions place princesse elisabeth 9/11 - 1030 Brussels - Belgium Tel : +32-(0)2-215.67.65 - Fax : +32-(0)2-215.66.65 domains - hosting - hardware - VoiP - consultancy - backuping CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL No legal consequences can be derived from the contents of the email neither is belGOnet.com committed to them. The content of this email is exclusively intended for adressee(s) and information purposes. belGOnet.com accepts no liability for any damage resulting from the use and/or acceptation of the content of this email. Sunday, September 26, 2004, 10:58:37 AM, you wrote: DN On 26/09/2004 15:41 Vahan Yerkanian said the following: I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. DN welltech still hasn't responded to my complaint to them regarding their SIP DN registration process. in the interim however, i've patched asterisk 1.0 to DN handle this wellgate special case. i'll be emailling the patch over to you DN in a separate email. i run asterisk (0.9.0 and 1.0) on freebsd 4.10 too. alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both sides after the call is established. DN i've seen the same. the wellgate's do not support the GSM codec and stock DN asterisk does not do g723.1. you'd need to only use ulaw or alaw for them. DN i've got this in my wellgate sip.conf entries: DN disallow=all DN allow=ulaw DN allow=alaw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Move Over Asterisk - Ondo is Here. - Email from Brekeke Announcing their RTP Proxy
On Sat, 25 Sep 2004, SeshKanuri wrote: Dear Valued OnDO users, Huh?? I'm not an OnDO user!? This is the ASTERISK list. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp
have posted a while ago on issues of receiving faxes by an Asterisk host using an x100p fxo interface attached to BT pstn the asterisk installation is the cvs download as of 23/09/04 is anyone able to confirm that the rxfax / txfax application that seems to be 'bundled' in thecvs download is the latest as per the www.opencall.org site which i think is at 0.0.1k ?? TIA GT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Asterisk - WellGate 3502a : ulaw/alaw only?
Hi, Have some welltech devices myself, but after about one year of trials I decided to throw them out ... Even when you succeed in making it register it freezes after a random amount of time, no matter what SW u use on it. Regards, Kiss Karoly On Sun, 26 Sep 2004, Danny Zak wrote: Date: Sun, 26 Sep 2004 11:22:43 +0200 From: Danny Zak [EMAIL PROTECTED] To: Dinesh Nair [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Subject: Re[2]: [Asterisk-Users] Asterisk - WellGate 3502a : ulaw/alaw only? Hello Dinesh, my welltech does support g729 g723.1 and g711u and a. It works like a charm here. BUT; the registration problem still exists (even after using your rc) -- Best regards, Dannymailto:[EMAIL PROTECTED] belGOnet.com a Euro-pictures division - internet solutions place princesse elisabeth 9/11 - 1030 Brussels - Belgium Tel : +32-(0)2-215.67.65 - Fax : +32-(0)2-215.66.65 domains - hosting - hardware - VoiP - consultancy - backuping CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL No legal consequences can be derived from the contents of the email neither is belGOnet.com committed to them. The content of this email is exclusively intended for adressee(s) and information purposes. belGOnet.com accepts no liability for any damage resulting from the use and/or acceptation of the content of this email. Sunday, September 26, 2004, 10:58:37 AM, you wrote: DN On 26/09/2004 15:41 Vahan Yerkanian said the following: I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. DN welltech still hasn't responded to my complaint to them regarding their SIP DN registration process. in the interim however, i've patched asterisk 1.0 to DN handle this wellgate special case. i'll be emailling the patch over to you DN in a separate email. i run asterisk (0.9.0 and 1.0) on freebsd 4.10 too. alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both sides after the call is established. DN i've seen the same. the wellgate's do not support the GSM codec and stock DN asterisk does not do g723.1. you'd need to only use ulaw or alaw for them. DN i've got this in my wellgate sip.conf entries: DN disallow=all DN allow=ulaw DN allow=alaw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pthread problem
Hi All, I have successful compiled asterisk from CVS. But when I start it up, it show an messages like this: [chan_sip.so]Sep 26 13:29:35 WARNING[-1084308832]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: __use_ast_pthread_create_instead__ Sep 26 13:29:35 WARNING[-1084308832]: loader.c:429 load_modules: Loading module chan_sip.so failed! I have search the archive but did not found the right answer. Please help me... -- Muhammad Rully Sumbayak Nusanet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp
Graham Turner a écrit : have posted a while ago on issues of receiving faxes by an Asterisk host using an x100p fxo interface attached to BT pstn the asterisk installation is the cvs download as of 23/09/04 is anyone able to confirm that the rxfax / txfax application that seems to be 'bundled' in thecvs download is the latest as per the www.opencall.org site which i think is at 0.0.1k ?? Yes -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What about a higher level configuration language
Hi all. I've been reading through Wi-Ki and at the extensions.conf file description (http://www.voip-info.org/wiki-Asterisk+config+extensions.conf) The author says this: One day, someone is going to write a proper scripting language for Asterisk that can understand a simpler, easier (and more traditional) scripting syntax. All it would need to do is translate the high level scripting code into the low level stuff that is required here. But until then, we have to live with what we've got. I just haven't use asterisk enough yet, so I wont be able to make the specifications, but I'm sure with the help of the community and if there are enough people needing/wating this, we can make a good language specification. Personally I can and will make the compiler from the higher level to current language, and even maybe vice-versa. Just need the specification of both languages and support of the comunity (if no one needs this it is not worthy). Please, send oppinions and ideas about this. RODOLFO --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 0439-2, 24/09/2004 Tested on: 26/09/2004 15:04:00 avast! - copyright (c) 2000-2004 ALWIL Software. http://www.avast.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail /w asterisk - voicemail() problems
I've setup the voicemail that auths against the mysql db. Now, everything works ok, except voicemail() calls fail with Sep 26 18:09:34 WARNING[157070336]: app_voicemail.c:1517 leave_voicemail: No entry in voicemail config file for '' all my users are in 'sip' voicemail context, but adding context to it: voicemail(@sip) doesn't help.. while if I put a vmbox # to it, it works that way: voicemail([EMAIL PROTECTED]). here is my voicemail.conf: -- [general] format=wav|gsm serveremail=asterisk attach=yes maxmessage=180 maxgreet=60 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 dbname = asteriskcdrdb ; Name of database in your Mysql server dbhost = localhost ; Hostname of server dbuser = astintvm ; Username in MySQL dbpass = x ; Password for user in MySQL and part of my extensions.conf: --- exten = 450,1,Answer exten = 450,2,Wait(1) exten = 450,3,VoicemailMain(@sip) exten = 450,4,Wait(1) exten = 450,5,Hangup exten = 451,1,Answer exten = 451,2,Wait(1) exten = 451,3,VoiceMailMain([EMAIL PROTECTED]) exten = 451,4,Wait(1) exten = 451,5,Hangup exten = 452,1,Answer exten = 452,2,Zapateller exten = 452,3,Voicemail() exten = 452,4,Wait(1) exten = 452,5,Hangup The last extension, 452, is the one that doesn't work. regards, Vahan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
On Sat, 25 Sep 2004, Florin Andrei wrote: On Fri, 2004-09-24 at 05:47, Greg Boehnlein wrote: Anyone else having the problems that Gary is reporting? Um, well, not really. I'm rebuilding your package on Fedora 2 (kernel 2.6) and i had to add a linux 26 at the end of the make line, otherwise all kinds of weird things happened. I think I'll add a kernel version check to it and either force a make or a make linux26 based on what is running. Also, in /etc/init.d/zaptel, insmod doesn't work properly. It has to be replaced with modprobe. I have no idea why. There are some other changes i've made to the initialization scripts, to bring them closer to Red Hat best practices. I'll probably email you privately when i'm closer to a stable state. On the init.d front, Mark committed my much improved redhat init scripts to CVS last month, but I haven't updated the RPMS to use them. Anyway, the RPMs are way cool! :-) Thanks, You are welcome. Keep in mind that any changes that you make should take into account the original design goal, which is to have a universal .spec file for all versions of RedHat from 7.3 onward. I.E. please try to keep distro specific changes defined ONLY for that distro and not globally applied to everything. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[3]: [Asterisk-Users] Asterisk - WellGate 3502a : ulaw/alaw only?
Hello Kiss, well we didn't have this kind of errors .. YET :) -- Best regards, Dannymailto:[EMAIL PROTECTED] belGOnet.com a Euro-pictures division - internet solutions place princesse elisabeth 9/11 - 1030 Brussels - Belgium Tel : +32-(0)2-215.67.65 - Fax : +32-(0)2-215.66.65 domains - hosting - hardware - VoiP - consultancy - backuping CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL No legal consequences can be derived from the contents of the email neither is belGOnet.com committed to them. The content of this email is exclusively intended for adressee(s) and information purposes. belGOnet.com accepts no liability for any damage resulting from the use and/or acceptation of the content of this email. Sunday, September 26, 2004, 3:56:39 PM, you wrote: KK Hi, KK Have some welltech devices myself, but after about one year of trials I KK decided to throw them out ... KK Even when you succeed in making it register it freezes after a random KK amount of time, no matter what SW u use on it. KK Regards, KK Kiss Karoly KK On Sun, 26 Sep 2004, Danny Zak wrote: Date: Sun, 26 Sep 2004 11:22:43 +0200 From: Danny Zak [EMAIL PROTECTED] To: Dinesh Nair [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Subject: Re[2]: [Asterisk-Users] Asterisk - WellGate 3502a : ulaw/alaw only? Hello Dinesh, my welltech does support g729 g723.1 and g711u and a. It works like a charm here. BUT; the registration problem still exists (even after using your rc) -- Best regards, Dannymailto:[EMAIL PROTECTED] belGOnet.com a Euro-pictures division - internet solutions place princesse elisabeth 9/11 - 1030 Brussels - Belgium Tel : +32-(0)2-215.67.65 - Fax : +32-(0)2-215.66.65 domains - hosting - hardware - VoiP - consultancy - backuping CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL No legal consequences can be derived from the contents of the email neither is belGOnet.com committed to them. The content of this email is exclusively intended for adressee(s) and information purposes. belGOnet.com accepts no liability for any damage resulting from the use and/or acceptation of the content of this email. Sunday, September 26, 2004, 10:58:37 AM, you wrote: DN On 26/09/2004 15:41 Vahan Yerkanian said the following: I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. DN welltech still hasn't responded to my complaint to them regarding their SIP DN registration process. in the interim however, i've patched asterisk 1.0 to DN handle this wellgate special case. i'll be emailling the patch over to you DN in a separate email. i run asterisk (0.9.0 and 1.0) on freebsd 4.10 too. alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both sides after the call is established. DN i've seen the same. the wellgate's do not support the GSM codec and stock DN asterisk does not do g723.1. you'd need to only use ulaw or alaw for them. DN i've got this in my wellgate sip.conf entries: DN disallow=all DN allow=ulaw DN allow=alaw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users KK ___ KK Asterisk-Users mailing list KK [EMAIL PROTECTED] KK http://lists.digium.com/mailman/listinfo/asterisk-users KK To UNSUBSCRIBE or update options visit: KKhttp://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP with IAXy provisioning
Hi,
I am trying to configure a basic Asterisk setup with Asterisk running
on Linux and a GS 102 phone and an analogue phone connected (through
IAXY adaptor) to Asterisk.
I am stuck at the step of provisioning the IAXy adaptor. I will
quickly give an overview of my setup.
I installed Redhat 9, got the latest version of Asterisk (not 1.0)
from the CVS and installed it successfully. I did not install the DHCP
server on Linux earlier, but from an updated version of the IAXY
installation guide, I read that:
Q: My IAXy never gets an IP from the DHCP server?
A: Make sure your /etc/dhcpd.conf is similar to the following:
authoritative; # Declare authority
server-identifier mydhcp; # An identifier
option domain-name mydomain.net; # My domain
option domain-name-servers 192.168.0.1; # The IP of my Master DNS
192.168.0.2; # The IP of my Slave DNS
ddns-update-style ad-hoc;
subnet 192.168.0.128 netmask 255.255.255.192 { # Subnet/Netmask Served
range dynamic-bootp 192.168.0.139 192.168.0.189; # A range of IPs
option routers 192.168.0.129; # The IP of your router
}
So I am assuming that my Linux installation needs the DHCP server
properly configured too.
2. Can anyone suggest IP, subnet mask, gateway settings for the linux
machine. As my machine is stand alone (not on any network), I think I
can keep whatever settings are working for someone else. Also if
someone can provide a working copy of /etc/dhcpd.conf and
iaxy.conf.sample files.
PS: I downloaded the latest iaxyprov tool using the CVS. I noticed
when I ran the make command I got ./iaxyprov executable rather than
./provision as mentioned in the IAXy installation guide. Is this an
updated version? and does it work fine?
any HELP in this regard would be highly appreciated!
Thanks,
T
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[Asterisk-Users] spandsp patch help
I've installed spandsp-0.0.1k on a RHv9 box with CVS-HEAD-09/19/04 and compiled the libraries just fine. Having a problem with patching the asterisk/apps Makefile however. The patch attempt results in: [EMAIL PROTECTED] apps]# patch Makefile.patch patching file Makefile Hunk #1 FAILED at 35. Hunk #2 FAILED at 68. 2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej [EMAIL PROTECTED] apps]# Am I doing something wrong here? (The Makefile.patch is dated 3/16/04 from the www.opencall.org site.) Patching the Makefile by hand followed by 'make clean' and 'make install' results in: make[1]: Entering directory `/usr/src/asterisk/apps' Makefile:75: *** missing separator. Stop. make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [depend] Error 1 Where have I gone wrong with this? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * works, but after a few seconds audio always stops.
Transmit silence = no tells the client not to send anything until it has some audio to send. The other end is dropping the connection because it thinks the client has lost it's connection. Lyle - Original Message - From: Michael Loftis [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 26, 2004 3:28 AM Subject: Re: [Asterisk-Users] * works, but after a few seconds audio always stops. --On Sunday, September 26, 2004 02:29 +0200 Philipp von Klitzing [EMAIL PROTECTED] wrote: FAQ: Turn of silence suppression in X-Lite by setting Transmit silence to YES (in AUDIO settings). AHA! That's it! Works now. Not sure what silence suppression has to do with all but it works now. Tnx much! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What about a higher level configuration language
On Sun, 26 Sep 2004 15:03:58 +0200, Rodolfo Grave [EMAIL PROTECTED] wrote: but I'm sure with the help of the community and if there are enough people needing/wating this, we can make a good language specification. Personally I can and will make the compiler from the higher level to current language, and even maybe vice-versa. Just need the specification of both languages and support of the comunity (if no one needs this it is not worthy). If you want to build a strong house, you need a solid fundament. I don't think the current configuration language is a good fundament to build upon. We would be well advised to first build that solid fundament before thinking about higher level languages. So how about a discussion to give Asterisk an XML based configuration as a base. For backwards compatibility, the existing config language could then sit on top of the XML as one alternative amongst others. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Proper Syntax
I set up the pilot number to voicemail to be 777. When a user calls 777 the voicemail answers and asks for mailbox, then password. Is there a way for the Voicemail to read what extension they are calling from and just ask for the password? I have a person complaining because they have to enter their mailbox number every time they check their voicemail and the old pbx didnt ask for it. I thought I saw this on a post a while ago, but of course now that I need it I cant find it. Thank you Henry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue and Agent functionality
Hello, On Sun, 26 Sep 2004 00:49:35 -0400, Robert Jackson [EMAIL PROTECTED] wrote: [snip] 4. If a caller empties a handled queue (active agents) with no callers, the caller will still hear messages (you are first in queue, etc.). This should not occur. Someone posted a 2-line patch on -dev list recently to fix this issue. The patch works for us. I am assuming this will end up in CVS soon. If there is really an interest for this tiny patch I can post it to bugs.digium.com today. I don't know if Mark's or the author of the queues position announcements are ok with it.. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Proper Syntax
exten = 777,1,VoicemailMain([EMAIL PROTECTED]) Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sun, 26 Sep 2004, Henry Devito wrote: I set up the pilot number to voicemail to be 777. When a user calls 777 the voicemail answers and asks for mailbox, then password. Is there a way for the Voicemail to read what extension they are calling from and just ask for the password? I have a person complaining because they have to enter their mailbox number every time they check their voicemail and the old pbx didn't ask for it. I thought I saw this on a post a while ago, but of course now that I need it I can't find it. Thank you Henry This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-09-26%5Cca2dfc4a03aa424992d9a0dca8957323C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail /w asterisk - voicemail() problems
That is because it is a required argument. http://voip-info.org/wiki-Asterisk+cmd+VoiceMail And you can see the difference from voicemailmain(): http://voip-info.org/wiki-Asterisk+cmd+VoicemailMain quote who=Vahan Yerkanian all my users are in 'sip' voicemail context, but adding context to it: voicemail(@sip) doesn't help.. while if I put a vmbox # to it, it works that way: voicemail([EMAIL PROTECTED]). -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What about a higher level configuration language
On Sun, 2004-09-26 at 17:18, Benjamin on Asterisk Mailing Lists wrote: [snip] So how about a discussion to give Asterisk an XML based configuration as a base. For backwards compatibility, the existing config language could then sit on top of the XML as one alternative amongst others. I think this was briefly discussed at Astricon and Mark(?) mentioned that XML was good for a lot things but not for *'s config. Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transferring Calls
I have the same problem and suppose that by some works on a new application similar to parkandannounce app. it should be done. - shabanip - Original Message - From: Alex Forrow [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 17, 2004 5:43 PM Subject: [Asterisk-Users] Transferring Calls We have set up an IP telephoney system hosted by Asterisk and its working pretty well. We primarily use SIP and hardware IP phones. We have the ability to transfer calls to another SIP phone using either the Transfer button on the phone (these phones are Grandstream BudgeTone 100s) or using the # key (the T/t flag must be set in the Dial command in asterisk for this way to work). Both methods seem similar; you enter the number and it transfers. The problems arise when the phone that it is transfered to is Busy or there is no answer: Asterisk just hangs up. Instead of this behaviour, we would like it to return the call to the person that transfered it. Alternativley, it could just do a 3 way call or something until the original person hangs up? I can't believe there is no way to achieve this. I have looked all over the internet but I can't find anything about this. From an Asterisk context point of view, the transfered call looks like a new call, and as far as i can see there is no way to differentiate between a new call and a transferred one. I know asterisk can tell the difference because the phone sends a REFER datagram to initate the transfer. Any help would be really appreciated Thanks, Alex Forrow Seek-it -- Using Opera's revolutionary e-mail client: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for a commercial version of an IAX2 Softphone
Hello All, I have been looking for a commercial version of an IAX2 Softphone for Windows but the ones I have came across (i.e. Iaxcomm, Iaxphone, Diax) do not seem to have an updated version since April 2004 in some cases. We looked at Firefly but we sent emails to Virbiage/Freshtel with questions and could never get a response from them. Has anyone got any recommendations for commercial version an IAX2 softphone that we could test? Thanks in Advance Errol ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for a commercial version of an IAX2 Softphone
Depending on your needs I don't know if you will find 1 that used IAX2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for a commercial version of an IAX2 Softphone
Sorry about that cut off . Like I was saying I'm not sure if you will find once advanced enough using IAX2 currently. Firefly was the most evolved when I too was looking but their oem terms weren't exactly what I wanted to spend given the fact that I probably would be going hardphones eventually. Depending on your need IAXPhone isn't bad for windows. Iaxcomm is my preference for cross platform. Perhaps it will take modifying an open source client and adding new features for this area to progress. -- William ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What type of PRI setup is best
NI-2 gives you the best set of available features (ie.e CNAM callerid) and is my preferred choice for our PRI setups. Alfred. Christopher Jacob wrote: I am having my colo set up 2 PRI's for my new asterisk implementation. They asked the following... ##SNIP## What type (NI2, NTI, 4ESS, or 5ESS) and whether they want to be USER or NETWORK. If the equipment is flexible, NI2, with us as NETWORK is preferred. ##SNIP## We are using a digium quad span T1 card. What is the recommended setup? Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What about a higher level configuration language
XML isn't the magic answer to all questions. The configuration
database for Asterisk is relatively flat while XML supports
hierarchical data much better. Asterisk's current config file are
better suited to the task than XML inherently could be.
If the original poster is still interested writing a preprocessor for
asterisks config, I'd like to see a syntax for extensions.conf that's
easier to maintain, such as:
[context]
exten = _1xxx
:DigitTimeOut(10)
:ResponseTimeOut(20)
BackHere:Answer
:Read(callto,pls-entr-num-uwish2-call,10)
:Read(callfrom,enter-phone-number10,10)
:SetCIDNum(${callfrom})
:Dial(IAX2/${IAXFREE}/1${callto},40):Failed
:Hangup
Failed:Play(try-again)
Goto(BackHere)
Eliminating the need to specify (and keep track of) priorities would
make changes to extensions.conf much easier to implement.
-Original Message-
From: Benjamin on Asterisk Mailing Lists
[mailto:[EMAIL PROTECTED]
Sent: Sunday, September 26, 2004 10:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] What about a higher level
configuration language
On Sun, 26 Sep 2004 15:03:58 +0200, Rodolfo Grave
[EMAIL PROTECTED] wrote:
but I'm sure with the help of the community and if there
are enough people needing/wating this, we can make a good language
specification. Personally I can and will make the compiler from the
higher level to current language, and even maybe
vice-versa. Just need
the specification of both languages and support of the
comunity (if no
one needs this it is not worthy).
If you want to build a strong house, you need a solid
fundament. I don't think the current configuration language
is a good fundament to build upon.
We would be well advised to first build that solid fundament
before thinking about higher level languages.
So how about a discussion to give Asterisk an XML based
configuration as a base.
For backwards compatibility, the existing config language
could then sit on top of the XML as one alternative amongst others.
rgds
benjk
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RE: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
Greg let me know where the new RPM is so I can update my system as it's still down... Thankfully it wasn't my production machine.. Thanks again.. John B. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein Sent: Sunday, September 26, 2004 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9 On Sat, 25 Sep 2004, Florin Andrei wrote: On Fri, 2004-09-24 at 05:47, Greg Boehnlein wrote: Anyone else having the problems that Gary is reporting? Um, well, not really. I'm rebuilding your package on Fedora 2 (kernel 2.6) and i had to add a linux 26 at the end of the make line, otherwise all kinds of weird things happened. I think I'll add a kernel version check to it and either force a make or a make linux26 based on what is running. Also, in /etc/init.d/zaptel, insmod doesn't work properly. It has to be replaced with modprobe. I have no idea why. There are some other changes i've made to the initialization scripts, to bring them closer to Red Hat best practices. I'll probably email you privately when i'm closer to a stable state. On the init.d front, Mark committed my much improved redhat init scripts to CVS last month, but I haven't updated the RPMS to use them. Anyway, the RPMs are way cool! :-) Thanks, You are welcome. Keep in mind that any changes that you make should take into account the original design goal, which is to have a universal .spec file for all versions of RedHat from 7.3 onward. I.E. please try to keep distro specific changes defined ONLY for that distro and not globally applied to everything. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Proper Syntax
Henry,
exten = 451,1,Answer
exten = 451,2,Wait(1)
exten = 451,3,VoiceMailMain([EMAIL PROTECTED])
exten = 451,4,Wait(1)
exten = 451,5,Hangup
is what you're looking for.
VoiceMailMain(${CALLERIDNUM}) might be enough if you're not storing your
voicemailboxes in mysql
regards,
Vahan
Henry Devito wrote:
I set up the pilot number to voicemail to be 777. When a user calls 777
the voicemail answers and asks for mailbox, then password. Is there a
way for the Voicemail to read what extension they are calling from and
just ask for the password? I have a person complaining because they
have to enter their mailbox number every time they check their voicemail
and the old pbx didnt ask for it.
I thought I saw this on a post a while ago, but of course now that I
need it I cant find it.
Thank you
Henry
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RE: [Asterisk-Users] Looking for a commercial version of an IAX2Softphone
Hi William, Most my users will be Windows based. My problem is that I am not a developer so I am unable to modify an open source client. Iaxcomm and Iaxphone only seem to support GSM but it would be nice to have support for iLBC as well. All I really need is to have the color scheme of the skin changed and a few buttons working etc but I would want the most recent version. If anyone wants to contact me off-list to discuss further please feel free. Regards, Errol -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Suffill Sent: 26 September 2004 17:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Looking for a commercial version of an IAX2Softphone Sorry about that cut off . Like I was saying I'm not sure if you will find once advanced enough using IAX2 currently. Firefly was the most evolved when I too was looking but their oem terms weren't exactly what I wanted to spend given the fact that I probably would be going hardphones eventually. Depending on your need IAXPhone isn't bad for windows. Iaxcomm is my preference for cross platform. Perhaps it will take modifying an open source client and adding new features for this area to progress. -- William ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for a commercial version of an IAX2Softphone
Yes We have one... Its going to be released in October... We are in middle of Moving so once we move and get things settled we will be going online Unlike the diax and iaxcom and iaxphone ours is using DirectSound and has many more abilities. We are releasing software for our services but we plan on releasing a limited version that just does IAX2 for Asterisk users -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of E Samuels Sent: Sunday, September 26, 2004 12:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Looking for a commercial version of an IAX2Softphone Hello All, I have been looking for a commercial version of an IAX2 Softphone for Windows but the ones I have came across (i.e. Iaxcomm, Iaxphone, Diax) do not seem to have an updated version since April 2004 in some cases. We looked at Firefly but we sent emails to Virbiage/Freshtel with questions and could never get a response from them. Has anyone got any recommendations for commercial version an IAX2 softphone that we could test? Thanks in Advance Errol ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for a commercial version of an IAX2Softphone
At 19:19 26.09.2004, you wrote: Yes We have one... Its going to be released in October... We are in middle of Moving so once we move and get things settled we will be going online Unlike the diax and iaxcom and iaxphone ours is using DirectSound and has many more abilities. We are releasing software for our services but we plan on releasing a limited version that just does IAX2 for Asterisk users will the client be under an opensource licence? alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for a commercial version of anIAX2Softphone
Only the Asterisk IAX2 Part -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atuc Sent: Sunday, September 26, 2004 1:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Looking for a commercial version of anIAX2Softphone At 19:19 26.09.2004, you wrote: Yes We have one... Its going to be released in October... We are in middle of Moving so once we move and get things settled we will be going online Unlike the diax and iaxcom and iaxphone ours is using DirectSound and has many more abilities. We are releasing software for our services but we plan on releasing a limited version that just does IAX2 for Asterisk users will the client be under an opensource licence? alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IP Phones ?
Hi guys, I know this isn't strictly about Asterisk, but it is related... I am looking to buy a few IP phones, but I don't have a huge budged (hence why I love Asterisk, its amazing and free !), so I was wondering if anyone knew where I could get some cheap IP Phones ? Ideally they should be no more then about £50 ($90). Thanks, Paul. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phones ?
On Sun, 26 Sep 2004 19:04:38 +0100, Paul Tyreman [EMAIL PROTECTED] wrote: Hi guys, I know this isn't strictly about Asterisk, but it is related... I am looking to buy a few IP phones, but I don't have a huge budged (hence why I love Asterisk, its amazing and free !), so I was wondering if anyone knew where I could get some cheap IP Phones ? Ideally they should be no more then about £50 ($90). Thanks, Paul. The cheapest I found was the grandstream budgetone. USD 65. I'm also interested in cheap IP phones, so any news would be appreciated. Marconi. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How Do I configure zaptel for PRI in
I have a strange question. I am new to * and have it up and running for our office phones. We run a small dialer to call clients and remind them of ordership dates etc. I would like to have * take the calls from it and send them through a voip connection. We have a digium quad port t1 card. Are voip is from a 100t eth0. The dialer is transmiting ans sf. directly to the t1. Is it possible to plug the dialers t1 connection into the digium card and have it translate for voip? _ FREE pop-up blocking with the new MSN Toolbar get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium and mailing lists
I was somewhat concerned reading Mark's posting earlier today. Obviously, things are very bad in the US at the moment. Their Government even deported Cat Stevens the other day (check http://news.bbc.co.uk/1/hi/england/london/3686992.stm ). Clearly, given the fact that Digium contributes so much to Asterisk, they shouldn't be forced to risk their company's future by hosting these mailing lists in such an unstable environment where they could get sued for any ridiculous reason. Even an unjustified, ambit claim could generate huge defence costs on Digium's part, and cripple their ability to contribute to Asterisk. Therefore, it seems to be in the best interests of Asterisk's `security' to have the mailing lists hosted by someone other than Digium and maybe in a country that doesn't prohibit freedom of expression. I would certainly be willing to organise hosting through another company that wouldn't be at risk from vexatious legal claims. This would allow genuinely open discussion on the lists and would mean that no messages would need to be censored from the archives. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium and mailing lists
On Sun, 26 Sep 2004, Daniel Pocock wrote: Therefore, it seems to be in the best interests of Asterisk's `security' to have the mailing lists hosted by someone other than Digium and maybe in a country that doesn't prohibit freedom of expression. Amusing bit of stirring there. But, PLEASE, lets pass on the transatlantic flamewar!? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp
Steve, ? Daniel thanks for reply posts the location i download from is as per technote on * installation; export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot prior to the last download i had to manually install the rxfax / txfax applications from opencall.org after latest download rxFAX / txfax are loaded ?? assuming this is latest version of spandsp applications do you have any views on how to proceed with the debug of the failed fax receipt. ?? thanks for your help GT - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 26, 2004 3:22 PM Subject: Re: [Asterisk-Users] spandsp Graham Turner wrote: have posted a while ago on issues of receiving faxes by an Asterisk host using an x100p fxo interface attached to BT pstn the asterisk installation is the cvs download as of 23/09/04 is anyone able to confirm that the rxfax / txfax application that seems to be 'bundled' in thecvs download is the latest as per the www.opencall.org site which i think is at 0.0.1k ?? TIA GT Which CVS download are you refering to? rxfax and txfax aren't in Digium's CVS as far as I know. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phones ?
Not the cheapest ($75-80) but they look interesting. http://ipphone.eezeephone.com/ Jonathan At 03:10 PM 9/26/2004 -0300, you wrote: On Sun, 26 Sep 2004 19:04:38 +0100, Paul Tyreman [EMAIL PROTECTED] wrote: Hi guys, I know this isn't strictly about Asterisk, but it is related... I am looking to buy a few IP phones, but I don't have a huge budged (hence why I love Asterisk, its amazing and free !), so I was wondering if anyone knew where I could get some cheap IP Phones ? Ideally they should be no more then about £50 ($90). Thanks, Paul. The cheapest I found was the grandstream budgetone. USD 65. I'm also interested in cheap IP phones, so any news would be appreciated. Marconi. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium and mailing lists
Another solution would be to keep the discussions on topic and open up a separate mailing list for people interested in open discussions. Jonathan At 07:17 PM 9/26/2004 +0100, you wrote: I was somewhat concerned reading Mark's posting earlier today. Obviously, things are very bad in the US at the moment. Their Government even deported Cat Stevens the other day (check http://news.bbc.co.uk/1/hi/england/london/3686992.stm ). Clearly, given the fact that Digium contributes so much to Asterisk, they shouldn't be forced to risk their company's future by hosting these mailing lists in such an unstable environment where they could get sued for any ridiculous reason. Even an unjustified, ambit claim could generate huge defence costs on Digium's part, and cripple their ability to contribute to Asterisk. Therefore, it seems to be in the best interests of Asterisk's `security' to have the mailing lists hosted by someone other than Digium and maybe in a country that doesn't prohibit freedom of expression. I would certainly be willing to organise hosting through another company that wouldn't be at risk from vexatious legal claims. This would allow genuinely open discussion on the lists and would mean that no messages would need to be censored from the archives. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp
Dear; regarding the details that i found on the site about spandsp; is it correct to assume the following ? + spandsp will only work with a card that is located in the * box? therefore will my welltech or any other voip fxo adapter support FAX ( i know it does t.38) -- My current situation is that i got a faxserver running in my old pbx situation; i need to find a way to replace the whole pbx with a * box; and need to fwd a certain MSN to my fax server, OR just have the * box handle the incomming fax. -- Best regards, Danny mailto:[EMAIL PROTECTED] belGOnet.com a Euro-pictures division - internet solutions place princesse elisabeth 9/11 - 1030 Brussels - Belgium Tel : +32-(0)2-215.67.65 - Fax : +32-(0)2-215.66.65 domains - hosting - hardware - VoiP - consultancy - backuping CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL No legal consequences can be derived from the contents of the email neither is belGOnet.com committed to them. The content of this email is exclusively intended for adressee(s) and information purposes. belGOnet.com accepts no liability for any damage resulting from the use and/or acceptation of the content of this email. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iLBC modes
I understand that the iLBC codec supports a variety of operative modes incluing a wideband mode. This could be useful in improving call quality over other codecs, say GSM, but retaining the iLBC strength in packet loss recovery. How do I control the codec to establish the compression settings on my * server? Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium and mailing lists
I agree with the mailing list part, but things arent very bad in the USA. Yusuf Islam was denied entry, not deported. I am sure there is more to this story than is being told or possibly ever will be told. He is not a US citizen and can be denied entry for any reason or suspicion. Please do not spill your feelings about the USA onto this mailing list while trying to mask them behind protecting Digium. Thanks, Steve - Original Message - From: Daniel Pocock [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Sunday, September 26, 2004 2:17 PM Subject: [Asterisk-Users] Digium and mailing lists I was somewhat concerned reading Mark's posting earlier today. Obviously, things are very bad in the US at the moment. Their Government even deported Cat Stevens the other day (check http://news.bbc.co.uk/1/hi/england/london/3686992.stm ). Clearly, given the fact that Digium contributes so much to Asterisk, they shouldn't be forced to risk their company's future by hosting these mailing lists in such an unstable environment where they could get sued for any ridiculous reason. Even an unjustified, ambit claim could generate huge defence costs on Digium's part, and cripple their ability to contribute to Asterisk. Therefore, it seems to be in the best interests of Asterisk's `security' to have the mailing lists hosted by someone other than Digium and maybe in a country that doesn't prohibit freedom of expression. I would certainly be willing to organise hosting through another company that wouldn't be at risk from vexatious legal claims. This would allow genuinely open discussion on the lists and would mean that no messages would need to be censored from the archives. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Simple Manager Proxy
On Sat, 25 Sep 2004, David Troy wrote: [deleted[ I had a need for a much simpler proxy than his op_server.pl; to meet my need I re-worked and simplified his code. See below for this simplified proxy: http://www.popvox.com/simpleproxy.pl Hehehehe.. I mentioned this in the Developer's Session at Astricon.. that op_server.pl might form the basis for a middleware to interface many clients w/ Asterisk for all sorts of things.. presence, management etc.. See how great minds think alike? ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAL tone
Hey group! Could someone could help me configure a DIal plan in order that when i dial 9 at the beginning i receive DIAL TONE? Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone.___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What about a higher level configuration language
On 27/09/2004 00:50 Jay Milk said the following: Eliminating the need to specify (and keep track of) priorities would make changes to extensions.conf much easier to implement. or perhaps allow non-consecutive priorities. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp
Graham Turner a écrit : Steve, ? Daniel thanks for reply posts the location i download from is as per technote on * installation; export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot This only for *, nothing todo with spandsp prior to the last download i had to manually install the rxfax / txfax applications from opencall.org after latest download rxFAX / txfax are loaded ?? latest is 0.0.1k. So if you have it downloaded it's still the same. assuming this is latest version of spandsp applications do you have any views on how to proceed with the debug of the failed fax receipt. ?? If you tell what is your problem perhaps we can help you ;-) You should perhaps check the faq on opencall site to see if your problem is explained there -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What about a higher level configuration language
Dinesh Nair wrote: On 27/09/2004 00:50 Jay Milk said the following: Eliminating the need to specify (and keep track of) priorities would make changes to extensions.conf much easier to implement. or perhaps allow non-consecutive priorities. After this topic was discussed a bit at the developer's confab, I got to thinking about what a great feature that would be. Renumbering priorities is a sadly common task for me in my somewhat chaotic config environment, and having a way to sneak in actions in between existing ones would be a major win. Of course, the problem of the hard-coded priority + 101 situation is problematical. I say we think through what the perfect world would look like in this respect and then see how hard it would be to implement. . . B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp
Danny Zak a écrit : Dear; regarding the details that i found on the site about spandsp; is it correct to assume the following ? + spandsp will only work with a card that is located in the * box? should also work with ztdummy if you don't have a card. -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What about a higher level configuration language
I pray for an end to the priorities as well. The +101 could be easily solved by a default label, or an option to the dial for example: exten = _7XX,1,Dial(yada,10) exten = _7XX,2,Voicemail(unavail) exten = _7XX,3,Hangup exten = _7XX,102,Voicemail(Busy) could be: exten = Dial:_7XX,Dial(yada,10) exten = Hangup:_7XX,Hangup exten = VMUnavail:_7XX,Voicemail(unavail) exten = VMBusy_7XX,Voicemail(Busy) in other words, the dial automatically looks for VMUnavail if not answered, or VMBusy if the line is busy or exten = StartPlan:Dial:_7XX,Dial(yada,10,BeforeAnswer=AA,AfterAnswer=ZZ,Busy=XX,NoAnswer=YY) exten = ZZ:_7XX,Hangup exten = XX:_7XX,Voicemail(unavail) exten = YY:_7XX,Voicemail(Busy) There must be fat better ways of expressing my thoughts, but it's late on Sunday :) Julian - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 26, 2004 8:41 PM Subject: Re: [Asterisk-Users] What about a higher level configuration language Dinesh Nair wrote: On 27/09/2004 00:50 Jay Milk said the following: Eliminating the need to specify (and keep track of) priorities would make changes to extensions.conf much easier to implement. or perhaps allow non-consecutive priorities. After this topic was discussed a bit at the developer's confab, I got to thinking about what a great feature that would be. Renumbering priorities is a sadly common task for me in my somewhat chaotic config environment, and having a way to sneak in actions in between existing ones would be a major win. Of course, the problem of the hard-coded priority + 101 situation is problematical. I say we think through what the perfect world would look like in this respect and then see how hard it would be to implement. . . B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What about a higher level configuration language
Why VMxyz, does every line end up at the VM when it's busy or unavailable or unregistered btw we could then also add a rule for the case the user agent has registered with * (bristuff addon n+201) Best Regards, Marc Asterisk wrote: I pray for an end to the priorities as well. The +101 could be easily solved by a default label, or an option to the dial for example: exten = _7XX,1,Dial(yada,10) exten = _7XX,2,Voicemail(unavail) exten = _7XX,3,Hangup exten = _7XX,102,Voicemail(Busy) could be: exten = Dial:_7XX,Dial(yada,10) exten = Hangup:_7XX,Hangup exten = VMUnavail:_7XX,Voicemail(unavail) exten = VMBusy_7XX,Voicemail(Busy) in other words, the dial automatically looks for VMUnavail if not answered, or VMBusy if the line is busy or exten = StartPlan:Dial:_7XX,Dial(yada,10,BeforeAnswer=AA,AfterAnswer=ZZ,Busy=XX,NoAnswer=YY) exten = ZZ:_7XX,Hangup exten = XX:_7XX,Voicemail(unavail) exten = YY:_7XX,Voicemail(Busy) There must be fat better ways of expressing my thoughts, but it's late on Sunday :) Julian - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 26, 2004 8:41 PM Subject: Re: [Asterisk-Users] What about a higher level configuration language Dinesh Nair wrote: On 27/09/2004 00:50 Jay Milk said the following: Eliminating the need to specify (and keep track of) priorities would make changes to extensions.conf much easier to implement. or perhaps allow non-consecutive priorities. After this topic was discussed a bit at the developer's confab, I got to thinking about what a great feature that would be. Renumbering priorities is a sadly common task for me in my somewhat chaotic config environment, and having a way to sneak in actions in between existing ones would be a major win. Of course, the problem of the hard-coded priority + 101 situation is problematical. I say we think through what the perfect world would look like in this respect and then see how hard it would be to implement. . . B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zqaptel and hdlc
I have hdlc networking and voice channels between two * boxes using a T-1 P2P circuit. I have Digium T-1 cards on both systems. I've loaded zaptel/libpri/asterisk 1.0 on one of the boxes. When I start zaptel and run ztcfg I get Zaptel networking not supported by this build. Has anyone else seen this? Thanks, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What about a higher level configuration language
Brian Capouch wrote: Of course, the problem of the hard-coded priority + 101 situation is problematical. I say we think through what the perfect world would look like in this respect and then see how hard it would be to implement. . . XML will probably able to store much, probably more, of the flat text in a marked up as 'Asterisk XML', created with a nice XSL-template there would even for +101 situations not be any problems. But my major consern is the validness of goto statements, but thats probably also the issue with the current 'list' versions. It is allready possible to start developing an 'Asterisk XML' since with an XSLT preprocessor you are able to generate a valid extension.conf. Stefan de Konink ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What about a higher level configuration language
I like my first suggestion the best... And that is of course fully
subjective:
[context]
exten = _1xxx
:DigitTimeOut(10)
:ResponseTimeOut(20)
BackHere:Answer
:Read(callto,pls-entr-num-uwish2-call,10)
:Read(callfrom,enter-phone-number10,10)
:SetCIDNum(${callfrom})
:Dial(IAX2/${IAXFREE}/1${callto},40):Failed
:Hangup
Failed:Play(try-again)
Goto(BackHere)
[label]:command[:label1,..] as the basic syntax
If the command has the conditional branch, then supply the +101, +201,
etc, labels. In the example above, when the dial-command fails, the
supplied priority is Failed, which will point to the priority marked
Failed -- the preprocessor would of course need to make sure that
Play{try-again} lands on a priority that +101 from the Dial command.
This seems fairly trivial -- if I find some time this week, I'll hack it
out in PHP or PERL.
-Original Message-
From: Asterisk [mailto:[EMAIL PROTECTED]
Sent: Sunday, September 26, 2004 3:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] What about a higher level
configuration language
I pray for an end to the priorities as well. The +101 could
be easily solved
by a default label, or an option to the dial
for example:
exten = _7XX,1,Dial(yada,10)
exten = _7XX,2,Voicemail(unavail)
exten = _7XX,3,Hangup
exten = _7XX,102,Voicemail(Busy)
could be:
exten = Dial:_7XX,Dial(yada,10)
exten = Hangup:_7XX,Hangup
exten = VMUnavail:_7XX,Voicemail(unavail)
exten = VMBusy_7XX,Voicemail(Busy)
in other words, the dial automatically looks for VMUnavail if
not answered,
or VMBusy if the line is busy
or
exten =
StartPlan:Dial:_7XX,Dial(yada,10,BeforeAnswer=AA,AfterAnswer=Z
Z,Busy=XX,NoAnswer=YY)
exten = ZZ:_7XX,Hangup
exten = XX:_7XX,Voicemail(unavail)
exten = YY:_7XX,Voicemail(Busy)
There must be fat better ways of expressing my thoughts, but
it's late on
Sunday :)
Julian
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, September 26, 2004 8:41 PM
Subject: Re: [Asterisk-Users] What about a higher level configuration
language
Dinesh Nair wrote:
On 27/09/2004 00:50 Jay Milk said the following:
Eliminating the need to specify (and keep track of)
priorities would
make changes to extensions.conf much easier to implement.
or perhaps allow non-consecutive priorities.
After this topic was discussed a bit at the developer's
confab, I got
to
thinking about what a great feature that would be.
Renumbering priorities is a sadly common task for me in my somewhat
chaotic config environment, and having a way to sneak in
actions in
between existing ones would be a major win.
Of course, the problem of the hard-coded priority + 101
situation is
problematical. I say we think through what the perfect
world would look
like in this respect and then see how hard it would be to
implement. . .
B.
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[Asterisk-Users] Dialplan question
Hello Asterisk, is it possible to make an extensions that write a call file (like a call back to the callerid) in the outgoing directory WITHOUT using a perl AGI ? -- Best regards, Danny mailto:[EMAIL PROTECTED] belGOnet.com a Euro-pictures division - internet solutions place princesse elisabeth 9/11 - 1030 Brussels - Belgium Tel : +32-(0)2-215.67.65 - Fax : +32-(0)2-215.66.65 domains - hosting - hardware - VoiP - consultancy - backuping CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL No legal consequences can be derived from the contents of the email neither is belGOnet.com committed to them. The content of this email is exclusively intended for adressee(s) and information purposes. belGOnet.com accepts no liability for any damage resulting from the use and/or acceptation of the content of this email. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium and mailing lists
Things aren't bad in the US at the moment. In fact, I think they're pretty good, because people actually seem concerned about PATENT and COPYRIGHT LAWs which your initial post attempted to circumvent. There was no issue with freedom of expression; there was an issue with legality of posts based on content. You're free to express your discontent about G.729 licensing issues, but you're not allowed to advertise a way to *steal* the software. In other terms, you are allowed to loudly and eloquently disagree with the price of goods, but your disapproval does not give you the right to steal it -- or explain to others how to steal and get away with it. FWIW, your little blurp, while obviously politically motivated, contained several inaccuracies: 1) There is no person by the name of Cat Stevens. That former singer changed his name legally to Yusuf Islam decades ago. 2) He was not deported, but rather denied entry. It's the right of any country to turn non-citizens back for any reason, and it happened to several friends of mine attempting to enter the US, Canada and on two occasions, Germany. 3) Yusuf Islam is not a US Citizen and as such has no legal *right* to enter the US as a visitor or for any reason; therefore his proposed legal action will likely not be legal in the sense that any law granting Mr. Islam rights, was indeed violated. -Original Message- From: Daniel Pocock [mailto:[EMAIL PROTECTED] Sent: Sunday, September 26, 2004 1:17 PM To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [Asterisk-Users] Digium and mailing lists I was somewhat concerned reading Mark's posting earlier today. Obviously, things are very bad in the US at the moment. Their Government even deported Cat Stevens the other day (check http://news.bbc.co.uk/1/hi/england/london/3686992.stm ). Clearly, given the fact that Digium contributes so much to Asterisk, they shouldn't be forced to risk their company's future by hosting these mailing lists in such an unstable environment where they could get sued for any ridiculous reason. Even an unjustified, ambit claim could generate huge defence costs on Digium's part, and cripple their ability to contribute to Asterisk. Therefore, it seems to be in the best interests of Asterisk's `security' to have the mailing lists hosted by someone other than Digium and maybe in a country that doesn't prohibit freedom of expression. I would certainly be willing to organise hosting through another company that wouldn't be at risk from vexatious legal claims. This would allow genuinely open discussion on the lists and would mean that no messages would need to be censored from the archives. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Registration Timeout, No FW
Hi people, My asterisk wont register with any sip providers, I have tried three different but they all end up with: Sep 26 17:36:36 NOTICE[114696]: chan_sip.c:4035 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again There is no firewall and my server has a public IP. Could this be a Asterisk problem? -Fredrik vK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Transferring Calls
-Original Message- From: Alex Forrow [mailto:[EMAIL PROTECTED] Sent: Friday, September 17, 2004 9:13 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Transferring Calls Both methods seem similar; you enter the number and it transfers. The problems arise when the phone that it is transfered to is Busy or there is no answer: Asterisk just hangs up. Instead of this behaviour, we would like it to return the call to the person that transfered it. I am not sure that you can do exactly that, but using a stdexten macro you can give the caller the option to wait until the extension is available or go to voicemail. Alternativley, it could just do a 3 way call or something until the original person hangs up? This is called an attended transfer and is very frequently a feature of the actual sip phone you are using. We are using this very Effectively with Cisco 7960's. I can't believe there is no way to achieve this. I have looked all over the internet but I can't find anything about this. Here are some pages that helped us work around this issue: * http://www.voip-info.org/wiki-Asterisk+tips+campon - Basically sets up an IVR menu that allows the caller to hit 1 to leave a message or hold on the line to get answered. * http://www.junghanns.net/asterisk/page6.html - Once set up it enables Follow me type of use. Then in your normal context execute the macro instead of Dial. Integrating these two features together has allowed us to accomplish the same goal. This even had an unexpected side effect for us over Our previous system which preformed like you wanted: it kept our Receptionists from dealing with the same call nearly doubling their effectiveness. I hope this helps, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DIAL tone
This is in the notes in the default extensions.conf - ignorepat = 9 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Gonzalo Gasca Meza Sent: Sunday, September 26, 2004 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] DIAL tone Hey group! Could someone could help me configure a DIal plan in order that when i dial 9 at the beginning i receive DIAL TONE? Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp with TDM fxo card?
Has anyone made spandsp to work with a digium tdm fxo card? I finally got the rxfax and txfax modules to compile, the spandsp lib installed (and in the libpath), and now receive: -- Starting simple switch on 'Zap/1-1' -- Executing RxFAX(Zap/1-1, /var/fax.tif) in new stack -- Hungup 'Zap/1-1' I've tried to adjust rxgain/txgain in zapta.conf, but never get to a point of receiving anything more then the above at the cli. The fax machine is an older Brother unit. Any thoughts? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialplan question
-Original Message- From: Danny Zak [mailto:[EMAIL PROTECTED] Sent: Sunday, September 26, 2004 5:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Dialplan question Hello Asterisk, is it possible to make an extensions that write a call file (like a call back to the callerid) in the outgoing directory WITHOUT using a perl AGI ? The only way that I can think of would be to execute a system Shell script or something to that effect. To my knowledge there is no way to write a file directly from within the dialplan. Hope this helps, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialplan question
I forgot to add a link to the system command: http://www.voip-info.org/wiki-Asterisk+cmd+System -Original Message- From: Robert Jackson Sent: Sunday, September 26, 2004 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Dialplan question -Original Message- From: Danny Zak [mailto:[EMAIL PROTECTED] Sent: Sunday, September 26, 2004 5:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Dialplan question Hello Asterisk, is it possible to make an extensions that write a call file (like a call back to the callerid) in the outgoing directory WITHOUT using a perl AGI ? The only way that I can think of would be to execute a system Shell script or something to that effect. To my knowledge there is no way to write a file directly from within the dialplan. Hope this helps, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/as terisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium and mailing lists
- Original Message - From: Jay Milk [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Sunday, September 26, 2004 4:35 PM Subject: RE: [Asterisk-Users] Digium and mailing lists You're free to express your discontent about G.729 licensing issues, but you're not allowed to advertise a way to *steal* the software. In other terms, you are allowed to loudly and eloquently disagree with the price of goods, but your disapproval does not give you the right to steal it -- or explain to others how to steal and get away with it. I may agree with you from a moralistic point of view, but I'd like to understand how instructions makes the author of the instructions liable for any illegal activity committed by someone who used the instructions. Should Microsoft be liable because someone wrote a virus after reading a Visual Studio Macro How-To? Should a screwdriver manufacturer be liable for my house being robbed because their instructions tell you how much torque their screwdriver can sustain and the robber got the idea to jimmy my window open? If I give you a knife and the instructions on how to slaughter livestock in a Kosher manner then you go out and slaughter some humans, am I responsible for their murder? Is Intel just as at fault in this situation in your opinion? If someone explains how to use development code and someone chooses to commit an illegal act with it, why should the author be punished? As for the mode of transmission, was Microsoft responsible because the 9/11 terrorists communicated via hotmail? J.Christian Hoffmeyer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
