Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032
On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote: Lex Thanks a lot. These morning i call Digium Support. One issue that i miss in my before e-mail is that i have my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my MFC/R2. Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall. They told me they can help me because they dont have UNICALL support. So... I need to investigate more or wait for a new zaptel or anything else. Generally you can always use a newer zaptel. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone have a method of keeping an incrementaltally of calls?
Hi JR !
You could use dbget/dbput to have something like that
i.e
Set(foo=${DB(counter/counter_val)})
Set(foo=${MATH(${foo}+1)}) ;
Set(DB(counter/counter_val)=${foo})
Stelios S. Koroneos
Digital OPSiS - Embedded Intelligence
http://www.digital-opsis.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
JR Richardson
Sent: Tuesday, April 08, 2008 5:02 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Anyone have a method of keeping an
incrementaltally of calls?
Hi All,
I thought I read a post a while back of a system call or
something in the dialplan whereby a call count can be
incremented and spit out to a text file.
Not like a group count of active channels.
What I would like to accomplish is have an incremental count
of a specific dialplan routine that gets called, so after a
week or month, I can see how many times a specific dilaplan
action has been used.
Thanks for any advice.
JR
--
JR Richardson
Engineering for the Masses
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Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032
Ruben, I am also in Monterrey and have used digium hardware on R2 and PRI. MFC/R2 is not supported by digium but the zaptel driver requirement is the same.. what changes is using libpri vs unicall. Just go ahead and ask them for the firmware update or as Tzafir says use a newer zaptel that should include the updated firmware. If in trouble add me to gtalk I'll try to help out any way possible, Lex On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote: Lex Thanks a lot. These morning i call Digium Support. One issue that i miss in my before e-mail is that i have my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my MFC/R2. Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall. They told me they can help me because they dont have UNICALL support. So... I need to investigate more or wait for a new zaptel or anything else. Generally you can always use a newer zaptel. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: Where isSoundPointIPWelcome.wav used?
If you are busy doing something else and you hear this soft, pleasant and unobtrusive sound, you suddenly realize Oh sh..., a Polycom just rebooted! :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?
Steve Totaro wrote: On Mon, Apr 7, 2008 at 6:36 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Apr 07, 2008 at 10:37:42AM +0100, James Williamson wrote: Tzafrir Cohen wrote: On Mon, Apr 07, 2008 at 06:11:02AM +0100, James Williamson wrote: Snap, Well, after trying to buying a TDM400P and then getting persuaded to buy a TDM410P because they no longer sell the 400 model I'd say I'm not impressed. It took three 2.6 kernel builds (zaptel 1.4 won't even build with the latest kernel release) What version have you tried? Of Zaptel and of the kernel? AFAIK 1.4.9.2 builds with latest kernel (or maybe there's actually a small warning with our drivers, fixed in SVN) Please provide an error log. Yes, zaptel 1.4.9.2 does compile against a 2.6.24.4 source tree, although the latest release on the website (something like a week ago) was 1.4.8 which doesn't: [EMAIL PROTECTED] zaptel-1.4.8]# make make[1]: Entering directory `/usr/local/src/zaptel-1.4.8' make -C /lib/modules/2.6.24/build SUBDIRS=/usr/local/src/zaptel-1.4.8 HOTPLUG_FIRMWARE=yes modules make[2]: Entering directory `/usr/src/linux-2.6.24' WARNING: Symbol version dump /usr/src/linux-2.6.24/Module.symvers is missing; modules will have no dependencies and modversions. scripts/Makefile.build:46: *** CFLAGS was changed in /usr/local/src/zaptel-1.4.8/Makefile. Fix it to use EXTRA_CFLAGS. Stop. make[2]: *** [_module_/usr/local/src/zaptel-1.4.8] Error 2 make[2]: Leaving directory `/usr/src/linux-2.6.24' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/local/src/zaptel-1.4.8' make: *** [all] Error 2 Yeah, fixed long ago in 1.4.9 (even though there's a very simple workaround for it) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir Another vote for 1.2.X. Openvox is the same as Digium TDM400P, it is the reference design and the cards are made very well. I suggest trying a Sangoma card. I've installed the 1.2.x zaptel drivers, this still doesn't work. Is there anyone in the UK who's successfully got a TDM410P to support caller id or am I just wasting my time? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?
On Tue, Apr 08, 2008 at 09:42:48AM +0100, James Williamson wrote: I've installed the 1.2.x zaptel drivers, this still doesn't work. Is there anyone in the UK who's successfully got a TDM410P to support caller id or am I just wasting my time? Before going further in wasting time, what do you have in zapata.conf ? (And for the record: I don't think 1.2 should be any better than 1.4 in picking up caller ID) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?
Hi James, I've installed the 1.2.x zaptel drivers, this still doesn't work. Is there anyone in the UK who's successfully got a TDM410P to support caller id or am I just wasting my time? I can (after many hours of testing) confirm that Zaptel 1.4.5.1 with the following patch to wctd.c does appear to give the best results. http://bugs.digium.com/view.php?id=9264 (However fails to patch against any newer version of Zaptel) Distinctive ring is broken :( it reports 0,0,0 for both ring types. So to conclude, Zaptel 1.4.5.1 + patch and Asterisk 1.4.19 - does appear to work. I am just confused why the patch above has failed to make it into the main Zaptel branch as this would resolve the CID problem for a lot of people. Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: Where isSoundPointIPWelcome.wav used?
That would at least be long enough to cover the entire boot process. ;) Lee, John (Sydney) wrote: It's played at the completion of the boot process. It's always been very quiet on the models I've worked with. Thanks Erik. I can probably replace it with my beloved Mozart Symphony no 40 :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?
On Tue, 8 Apr 2008, James Williamson wrote: I've installed the 1.2.x zaptel drivers, this still doesn't work. Is there anyone in the UK who's successfully got a TDM410P to support caller id or am I just wasting my time? I have it working on TDM400Ps under 1.2.x Applying the patch mentioned earlier was pretty crucial... # lsmod Module Size Used by zttranscode 6408 0 wctdm 32544 1 zaptel194360 6 zttranscode,wctdm oslec 7640 1 zaptel Extract from 'dmesg': Open Source Line Echo Canceller Installed Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.23 Zaptap registered 'sample' char driver on major 33 ACPI: PCI Interrupt Link [LNKB] enabled at IRQ 12 PCI: setting IRQ 12 as level-triggered ACPI: PCI Interrupt :00:14.0[A] - Link [LNKB] - GSI 12 (level, low) - IRQ 12 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Not installed Module 2: Not installed Module 3: Installed -- AUTO FXO (UK mode) Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) Registered tone zone 4 (United Kingdom) This is in /etc/moprobe.d/dsx: options wctdm opermode=UK and this is in /etc/modprobe.d/zaptel # automatically generated file; do not edit install wctdm /sbin/modprobe --ignore-install wctdm $CMDLINE_OPTS /sbin/ztcfg install ztdummy /sbin/modprobe --ignore-install ztdummy $CMDLINE_OPTS /sbin/ztcfg alias wcfxs wctdm alias wct2xxp wct4xxp # cat /etc/zaptel.conf fxoks=1 fxsks=4 loadzone=uk defaultzone=uk dsx:/etc# cat /etc/asterisk/zapata.conf ; zapata.conf: [trunkgroups] [channels] ; Default settings applicable to all channels usecallerid=yes cidsignalling=v23 cidstart=polarity hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes ;echotraining=yes ;echocancelwhenbridged=yes immediate=no faxdetect=no ; Channel 4: PSTN line context=incoming group=1 usecallerid=yes faxdetect=none signalling=fxs_ks rxgain=8 txgain=8 callerid=asreceived channel = 4 And a call: # rasterisk Asterisk 1.2.26.1, Copyright (C) 1999 - 2007 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = Connected to Asterisk 1.2.26.1 currently running on dsx (pid = 18853) dsx*CLI set verbose 9 Verbosity was 0 and is now 9 == Starting post polarity CID detection on channel 4 -- Starting simple switch on 'Zap/4-1' -- Executing NoOp(Zap/4-1, INCOMING CALL - From: 07712191046) in new stack Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium B410P, bristuff and BRI support in 1.6
2008/4/8, Jean-Denis Girard [EMAIL PROTECTED]: I agree with Tzafir, I'm not aware of zaptel support for HFC-USB; I checked bristuff, it doesn't support it. That's what I thought but it's better to ask as I don't feel so easy to read source code at the moment. Would be interesting to see one. +1 +1 makes 3 Regards, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?
Hi Tzafrir, As I can see from that report, a modified version of that patch was applied to 1.4: http://bugs.digium.com/9264#80824 It was reverted right before the release of 1.4.8, as it required more testing, and re-applied immediately after it. Hence 1.4.9[.2] includes it. I did download and compile 1.4.9.2 - however CID still gets missed and I get the dreaded... [Apr 5 16:21:13] NOTICE[12685]: chan_zap.c:6191 ss_thread: Got event 2 (Ring/Answered)... [Apr 5 16:21:15] WARNING[12685]: chan_zap.c:6254 ss_thread: CID timed out waiting for ring. Exiting simple switch Some calls seem to capture the CID, but a majority get missed. So hence I reverted back to 1.4.5.1 + patch, which appears to work fine (no missed CID so far) Distinctive ring is broken :( it reports 0,0,0 for both ring types. Is it reported anywhere? I am digging for that now, there does appear to be quite a few bugs about distinctive ring. I was hoping by moving to Zaptel 1.4.9.2 - to resolve both issues or at least try to work out the problem from here and eliminate trying to fix something in an earlier release, in addition I also downloaded the latest svn but to no avail . 1.4.9.2 also introduced a BATTERTY / NO BATTERY issue for me which caused the calls to hang up on polarity reversal which is needed for the CID part i.e cidstart = polarity. I did play with the battthreash and debounce settings - but looking at other posts this was bad to change the values as it then caused other issues. I have been working on this now for a while and tried all sorts of suggestions, posts, patches - to little avail. I am not a defeatist and would like to help the project resolve these issues as this does seem to me to be a problem for many UK people with TDM4XX cards. I will keep going until I find a solution using the latest codebase. Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] testing please ignore
If I see this, then messages are getting through. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?
On Tue, Apr 08, 2008 at 10:44:39AM +0100, Matt Brown wrote: Hi James, I've installed the 1.2.x zaptel drivers, this still doesn't work. Is there anyone in the UK who's successfully got a TDM410P to support caller id or am I just wasting my time? I can (after many hours of testing) confirm that Zaptel 1.4.5.1 with the following patch to wctd.c does appear to give the best results. http://bugs.digium.com/view.php?id=9264 (However fails to patch against any newer version of Zaptel) As I can see from that report, a modified version of that patch was applied to 1.4: http://bugs.digium.com/9264#80824 It was reverted right before the release of 1.4.8, as it required more testing, and re-applied immediately after it. Hence 1.4.9[.2] includes it. Distinctive ring is broken :( it reports 0,0,0 for both ring types. Is it reported anywhere? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?
Hi Tzafrir, As I can see from that report, a modified version of that patch was applied to 1.4: http://bugs.digium.com/9264#80824 It was reverted right before the release of 1.4.8, as it required more testing, and re-applied immediately after it. Hence 1.4.9[.2] includes it. I should have read the bug report http://bugs.digium.com/9264#8082 more carefully. I have now added the line: fwringdetect=1 to the /etc/asterisk/zapata.conf so the patched part of the code is then utilized. This is appears to be working and hopefully will make it into 1.4.10. Now to investigate the distinctive ring part :-) Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?
Hi Tzafrir, Distinctive ring is broken :( it reports 0,0,0 for both ring types. Is it reported anywhere? http://bugs.digium.com/view.php?id=6296 Would appear to be the best match, however due to lack of feedback, this bug was suspended. Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?
Tzafrir Cohen wrote:
On Tue, Apr 08, 2008 at 09:42:48AM +0100, James Williamson wrote:
I've installed the 1.2.x zaptel drivers, this still doesn't work. Is
there anyone
in the UK who's successfully got a TDM410P to support caller id or am I just
wasting my time?
Before going further in wasting time, what do you have in zapata.conf ?
(And for the record: I don't think 1.2 should be any better than 1.4 in
picking up caller ID)
I'm running a 2.6.22 kernel, zaptel 1.4.9.2 and asterisk 1.4.14. I've
ensured the driver is loaded in opermode=UK,
dmesg output:
wctdm24xxp: reg is a04c0004
Resetting the modules...
During Resetting the modules...
After resetting the modules...
Port 1: Installed -- AUTO FXS/DPO
Port 2: Installed -- AUTO FXO (UK mode)
Port 3: Not installed
Port 4: Not installed
VPM100: Not Present
Found a Wildcard TDM: Wildcard TDM410P (4 modules)
My zapata.conf looks like this:
[trunkgroups]
[channels]
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
;echotraining=yes
;echocancelwhenbridged=yes
immediate=no
faxdetect=no
fwringdetect=1
context=incoming
group=1
usecallerid=yes
faxdetect=none
signalling=fxs_ks
rxgain=8
txgain=8
callerid=asreceived
channel = 2
I've added a debug entry into my extension.conf:
[incoming]
exten = s,1,Verbose(Callerid = ${CALLERID} - ${CALLERIDNUM})
When I make an incoming call (I've got two landlines), I see this on my
terminal:
Asterisk Ready.
== Starting post polarity CID detection on channel 2
-- Starting simple switch on 'Zap/2-1'
[Apr 8 13:29:11] NOTICE[6815]: chan_zap.c:6171 ss_thread: Got event 2
(Ring/Answered)...
-- Executing [EMAIL PROTECTED]:1] Verbose(Zap/2-1, Callerid = - )
in new stack
Callerid = -
== Auto fallthrough, channel 'Zap/2-1' status is 'UNKNOWN'
-- Hungup 'Zap/2-1'
[Apr 8 13:29:13] NOTICE[6815]: cdr.c:434 ast_cdr_free: CDR on channel
'Zap/2-1' not posted
-- Starting simple switch on 'Zap/2-1'
-- Executing [EMAIL PROTECTED]:1] Verbose(Zap/2-1, Callerid = - )
in new stack
Callerid = -
== Auto fallthrough, channel 'Zap/2-1' status is 'UNKNOWN'
-- Hungup 'Zap/2-1'
[Apr 8 13:29:16] NOTICE[6816]: cdr.c:434 ast_cdr_free: CDR on channel
'Zap/2-1' not posted
== Starting post polarity CID detection on channel 2
-- Starting simple switch on 'Zap/2-1'
[Apr 8 13:29:28] WARNING[6817]: chan_zap.c:6234 ss_thread: CID timed
out waiting for ring. Exiting simple switch
-- Hungup 'Zap/2-1'
[Apr 8 13:29:28] NOTICE[6817]: cdr.c:434 ast_cdr_free: CDR on channel
'Zap/2-1' not posted
Executing last minute cleanups
The penultimate line appears after I hang up.
James
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Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?
As I can see from that report, a modified version of that patch was applied to 1.4: http://bugs.digium.com/9264#80824 It was reverted right before the release of 1.4.8, as it required more testing, and re-applied immediately after it. Hence 1.4.9[.2] includes it. I should have read the bug report http://bugs.digium.com/9264#8082 more carefully. I have now added the line: fwringdetect=1 to the /etc/asterisk/zapata.conf so the patched part of the code is then utilized. This is appears to be working and hopefully will make it into 1.4.10. Now to investigate the distinctive ring part :-) Hmm scratch that ! (apologies for any line wraps) It would appear its still broken .. and introduces a new bug ! [Apr 8 13:42:54] NOTICE[5846]: chan_zap.c:6234 ss_thread: CallerID number: 07875-xx, name: (null), flags=4 -- Executing [EMAIL PROTECTED]:1] Verbose(Zap/4-1, Incoming call from BT line CallerID= 07875xx) in new stack Incoming call from BT line CallerID= 07875xx -- Executing [EMAIL PROTECTED]:2] GotoIf(Zap/4-1, 0?3:8) in new stack -- Goto (incoming,s,8) -- Executing [EMAIL PROTECTED]:8] Set(Zap/4-1, CALLERID(all)=Incoming Call: 07875xx) in new stack -- Executing [EMAIL PROTECTED]:9] MixMonitor(Zap/4-1, 1207658573.0.gsm) in new stack -- Executing [EMAIL PROTECTED]:10] Dial(Zap/4-1, ZAP/1r1||tr) in new stack -- Called 1r1 == Begin MixMonitor Recording Zap/4-1 -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Hungup 'Zap/1-1' == Spawn extension (incoming, s, 10) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' == End MixMonitor Recording Zap/4-1 Second try == Starting post polarity CID detection on channel 4 -- Starting simple switch on 'Zap/4-1' [Apr 8 13:43:09] NOTICE[5848]: chan_zap.c:6191 ss_thread: Got event 2 (Ring/Answered)... [Apr 8 13:43:11] WARNING[5848]: chan_zap.c:6254 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/4-1' Third try ... (bug appears) == Starting post polarity CID detection on channel 4 -- Starting simple switch on 'Zap/4-1' [Apr 8 13:43:12] NOTICE[5849]: chan_zap.c:6191 ss_thread: Got event 4 (Alarm)... [Apr 8 13:43:12] NOTICE[5849]: chan_zap.c:4185 zt_handle_event: Alarm cleared on channel 4 [Apr 8 13:43:14] WARNING[5849]: chan_zap.c:6254 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/4-1' Some calls do show CID (as above) others do not - mainly not .. then as above if I call a second or third time the following happens ... (never on first call !) The call is answered and I get a high pitch tone for around 5 secs then the line goes quiet, However the Zap channel then remains open for upto 1min or more then resets The kernel/syslog shows this BATTERY NO BATTERY issue Apr 8 13:42:59 max kernel: [17434415.32] NO RING on 1/4! Apr 8 13:42:59 max kernel: [17434415.58] Setting FXS hook state to 0 (00) Apr 8 13:43:09 max kernel: [17434425.22] RING on 1/4! Apr 8 13:43:09 max kernel: [17434425.22] 63639009 Polarity reversed (-1 - 1) Apr 8 13:43:09 max kernel: [17434425.476000] NO RING on 1/4! Apr 8 13:43:10 max kernel: [17434426.964000] RING on 1/4! Apr 8 13:43:12 max kernel: [17434428.148000] NO RING on 1/4! Apr 8 13:43:12 max kernel: [17434428.388000] 63639801 Polarity reversed (1 - -1) Apr 8 13:43:12 max kernel: [17434428.804000] NO BATTERY on 1/4! Apr 8 13:43:12 max kernel: [17434428.90] BATTERY on 1/4 (-)! This is new, and have not seen this in 1.4.5.1 + patch ... I think I will have to stay on this version until I can find out what is going on ... Is is possible to get very verbose granular debug info out of Zaptel ? I have debug=1 and tried debug=9 in the modprobe section . but this is the most I get from Zaptel. Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wait for dialtone feature on FXO device
On 03/04/2008, Steve Davies [EMAIL PROTECTED] wrote: Anyone interested in this feature? I have a version 0.1 patch, which is currently against 1.2.25-bristuffed, but which should port trivially to almost any version. I am away until Tuesday 8th April, but if there is enough interest, I will open a new-feature ticket and upload the patch to the bugtracker so that more capable programmers can laugh at it ;-) It should work reasonably on North-American and UK systems, which seem to use the same dialtone frequencies. http://bugs.digium.com/view.php?id=12382 Patch has been attached. Currently only for asterisk 1.2.25, but if no-one else provides a 1.4.x patch soon, I will probably need to do that for myself anyway. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF between Asterisk servers.
I find it hard to believe no one knows, so is it just plain no helping? J If someone would like to atleast point me in the right direction that will deal specifically with what I'm asking, that would be appreciated too. Much thanks. From: Mark Hamilton [mailto:[EMAIL PROTECTED] Sent: April 7, 2008 11:48 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: DTMF between Asterisk servers. Hello, I'm a little confused on DTMF. A sip peer is registered on two Asterisk servers. No dtmfmode is set for them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both register on each other. A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the call is transferred to Asterisk 2: RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/[EMAIL PROTECTED],,t T,) Where 12351 accepts the call on Asterisk 2, and in some cases, that call is transferred out to a PSTN number, or wherever, but not within Asterisk anymore via provider2, dtmf=rfc2833. When the call comes in, I'd like it to relay DTMF just dandy. How can I do so? There is no NAT between the Asterisk servers or in front of them. However, Asterisk2 has iptables which allows all UDP traffic to/fro Asterisk1. When Asterisk2 transfers the call to external endpoints, there might be a LAN, but relative ports are open on those LANs. Please help. Thanks in advance, Mark. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF between Asterisk servers.
I believe that what you described should just work with the caveat that dtmf=inband is rarely the right thing to do over SIP, and is prone to all sorts of DTMF detection and debounce issues. I assume you've tried calling a POTS endpoint and listening to see if you get DTMF passed through? 1) You did not give a great deal of information about what the current situation was, or what investigations you've already tried, which is probably why no-one felt they could reply. 2) It may also have been because less than 23 hours had elapsed... Regards, Steve On 08/04/2008, Mark Hamilton [EMAIL PROTECTED] wrote: I find it hard to believe no one knows, so is it just plain no helping? J If someone would like to atleast point me in the right direction that will deal specifically with what I'm asking, that would be appreciated too. Much thanks. From: Mark Hamilton [mailto:[EMAIL PROTECTED] Sent: April 7, 2008 11:48 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: DTMF between Asterisk servers. Hello, I'm a little confused on DTMF. A sip peer is registered on two Asterisk servers. No dtmfmode is set for them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both register on each other. A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the call is transferred to Asterisk 2: RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/[EMAIL PROTECTED],,tT,) Where 12351 accepts the call on Asterisk 2, and in some cases, that call is transferred out to a PSTN number, or wherever, but not within Asterisk anymore via provider2, dtmf=rfc2833. When the call comes in, I'd like it to relay DTMF just dandy. How can I do so? There is no NAT between the Asterisk servers or in front of them. However, Asterisk2 has iptables which allows all UDP traffic to/fro Asterisk1. When Asterisk2 transfers the call to external endpoints, there might be a LAN, but relative ports are open on those LANs. Please help. Thanks in advance, Mark. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium HPEC license counting
Not that I;m complaining But I just got my 2 HPEC license keys from digium... for TDM800P and TDM400P asterisk asterisk # zaphpec_enable Digium High-Performance Echo Canceller Enabler Copyright (C) 2006, Digium, Inc. Version 1.0.2 Use the '-l' option to see license information for software included in this program. Found key 'HPEC-KEY1' for 8 channels. Found key 'HPEC-KEY2' for 4 channels. Found valid HPEC licenses for 13 channels. Since when does 8+4 = 13 ??? maybe I should ask thinkgeek.com to make another t-shirt like this one: http://www.thinkgeek.com/tshirts/itdepartment/60f5/ ? When they first issued my TDM800P key they incorrectly set it up as a single channel license instead of 8 channel... but after going back and forth with them a couple times they got it fixed... when I had a 4+1 license it correctly showed 5 channels... is it possible that somehow my old license for KEY1 is giving me an extra license and not showing it? They didn't actually issue me a new key... just fixed it on their end and had me re-register it. After I re-registered I unloaded the zaptel, wctdm, and wctdm24xxp modules and re-loaded them all... so I;m not really sure how that original single channel license might still be lingering... but that's all I can think of. -- Matt Disclaimer Statement: This e-mail is confidential and is intended for the above-named recipient(s) only. If you are not the intended recipient and/or have received this e-mail in error, please notify us by telephone and delete this e-mail from your system without retaining a copy in any form. Any unauthorized use or disclosure of this e-mail is prohibited. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: Where is SoundPointIPWelcome.wav used?
On Tue, Apr 08, 2008 at 12:25:09AM -0500, Erik Anderson wrote: On Tue, Apr 8, 2008 at 12:06 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: When I downloaded the sip and bootrom from Polycom website, I noticed a file called SoundPointIPWelcome.wav. However, I have no idea where and when it was used. I played the wav file but I have never heard the phone using this wav file before. Does anyone know what it is used for? It's played at the completion of the boot process. It's always been very quiet on the models I've worked with. Oh. Is *that* how Telovations is making phones say Hi to me after boot. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032
Lex Thanks, I all ready download the last svn branches from zaptel And i am going to test these afternoon. My phone number es 81-83481611. Thanks Ruben Lex Lethol escribió: Ruben, I am also in Monterrey and have used digium hardware on R2 and PRI. MFC/R2 is not supported by digium but the zaptel driver requirement is the same.. what changes is using libpri vs unicall. Just go ahead and ask them for the firmware update or as Tzafir says use a newer zaptel that should include the updated firmware. If in trouble add me to gtalk I'll try to help out any way possible, Lex On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote: Lex Thanks a lot. These morning i call Digium Support. One issue that i miss in my before e-mail is that i have my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my MFC/R2. Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall. They told me they can help me because they dont have UNICALL support. So... I need to investigate more or wait for a new zaptel or anything else. Generally you can always use a newer zaptel. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: Where isSoundPointIPWelcome.wav used?
And would chew up the entire internal memory so no...not appropriate. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis Sent: Tuesday, 8 April 2008 6:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie Polycom: Where isSoundPointIPWelcome.wav used? That would at least be long enough to cover the entire boot process. ;) Lee, John (Sydney) wrote: It's played at the completion of the boot process. It's always been very quiet on the models I've worked with. Thanks Erik. I can probably replace it with my beloved Mozart Symphony no 40 :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help with Cisco 7960
That's what he's doing, he's asking someone with better sight to help him out and tell him what buttons to press! :) I've dialed in the dark enough times to know you don't need braille on the buttons to find the 3x4 array and use it properly without eyes. Sorry, Steve, but I had a twinge of 'what if *I* was blind' when you said that. Moj Steve Totaro wrote: In that case, I guess I would ask somone with better sight to help me out, uless they have braille on the buttons. Thanks, Steve Totaro On Sun, Apr 6, 2008 at 5:09 PM, Christian [EMAIL PROTECTED] wrote: Hello, I know how to unlock the phone and what the password is. I am asking this kind of question because i am visually impaired and cannot see the screen. many thanks, Christian On 2008-04-06 at 17:05 Steve Totaro wrote: You probably have to unlock it first. Google or voip-info.org is your friend. On Sun, Apr 6, 2008 at 5:02 PM, Christian [EMAIL PROTECTED] wrote: Hello all, I need some help with my Cisco 7960 enabling TFTP. Does anyone know what numbers to press in the menu? Or can I enable this through telnet? Many thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Mojo Wentworth* HORAN COMPANY, LLC 403 Lincoln Street, Suite 210 Sitka, AK 99835 (907) 747- (907) 747-7417 - Fax [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help with Cisco 7960
Mojo thanks for the perspective check, hope this is of help: Reset the 7940 and 7960 IP Phones to the Factory Default In order to perform a factory reset of a phone if the password is set, complete these steps: * Unplug the power cable from the phone, and then plug in the cable again. * The phone begins its power up cycle. * Immediately press and hold # while the Headset, Mute, and Speaker buttons flash in sequence. * Release # after the Speaker button is no longer lit. * The Headset, Mute, and Speaker buttons flash in sequence in order to indicate that the phone waits for you to enter the key sequence for the reset. * Press 123456789*0# within 60 seconds after the Headset, Mute, and Speaker buttons begin to flash. * If you repeat a key within the sequence, for example, if you press 1223456789*0#, the sequence is still accepted and the phone resets. * If you do not complete this key sequence or do not press any keys, after 60 seconds, the Headset, Mute, and Speaker buttons no longer flash, and the phone continues with its normal startup process. The phone does not reset. * If you enter an invalid key sequence, the buttons no longer flash, and the phone continues with its normal startup process. The phone does not reset. * If you enter this key sequence correctly, the phone displays this prompt: * Keep network cfg? 1 = yes 2 = no * In order to maintain the current network configuration settings for the phone when the phone resets, press 1. In order to reset the network configuration settings when the phone resets, press 2. * If you press another key or do not respond to this prompt within 60 seconds, the phone continues with its normal startup process and does not reset. Otherwise, the phone goes through the factory reset process. Thanks, Steve Totaro On Tue, Apr 8, 2008 at 3:37 PM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: That's what he's doing, he's asking someone with better sight to help him out and tell him what buttons to press! :) I've dialed in the dark enough times to know you don't need braille on the buttons to find the 3x4 array and use it properly without eyes. Sorry, Steve, but I had a twinge of 'what if *I* was blind' when you said that. Moj Steve Totaro wrote: In that case, I guess I would ask somone with better sight to help me out, uless they have braille on the buttons. Thanks, Steve Totaro On Sun, Apr 6, 2008 at 5:09 PM, Christian [EMAIL PROTECTED] wrote: Hello, I know how to unlock the phone and what the password is. I am asking this kind of question because i am visually impaired and cannot see the screen. many thanks, Christian On 2008-04-06 at 17:05 Steve Totaro wrote: You probably have to unlock it first. Google or voip-info.org is your friend. On Sun, Apr 6, 2008 at 5:02 PM, Christian [EMAIL PROTECTED] wrote: Hello all, I need some help with my Cisco 7960 enabling TFTP. Does anyone know what numbers to press in the menu? Or can I enable this through telnet? Many thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Mojo Wentworth* HORAN COMPANY, LLC 403 Lincoln Street, Suite 210 Sitka, AK 99835 (907) 747- (907) 747-7417 - Fax [EMAIL PROTECTED] Mojo thanks for the perspective check, hope this is of help: Reset the 7940 and 7960 IP Phones to the Factory Default In order to perform a factory reset of a phone if the password is set, complete these steps: * Unplug the power cable from the phone, and then plug in the cable again. * The phone begins its power up cycle. * Immediately press and hold # while the Headset, Mute, and Speaker buttons flash in sequence. * Release # after the Speaker button is no longer lit. * The Headset, Mute, and Speaker buttons flash in sequence in order to indicate that the phone waits for you to enter the key sequence for the reset. * Press 123456789*0# within 60 seconds after the Headset, Mute, and Speaker buttons begin to flash. * If you repeat a key within the sequence, for example, if you press 1223456789*0#, the sequence is still accepted and the phone resets. * If you do not complete this key sequence or do not press any keys, after 60 seconds, the Headset, Mute, and Speaker buttons no longer flash, and the phone continues with its normal startup process. The phone does not reset. * If you enter an invalid key sequence, the buttons no longer flash, and the phone continues with its normal startup process. The phone does not reset. * If you enter this key sequence correctly, the phone displays this prompt: * Keep network cfg? 1 = yes 2 = no * In order to maintain the current network configuration settings for the phone when the phone resets,
[asterisk-users] Zaptel 1.2.25 and 1.4.10 released
The Asterisk.org development team has announced the release of Zaptel versions 1.2.25 and 1.4.10. These releases contain many bug fixes as well as performance enhancements. A couple of the more major changes include: modifications to the wctdm24xxp and wcte12xp drivers to increase interrupt latency resilience, numerous bug fixes and updates to the xpp drivers, as well as some Makefile updates. For further details and a more complete list see the respective Changelog files. Both releases are available as a tarball as well as a patch against the previous release. They are available for download from downloads.digium.com. Thank you for your support! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTCP not being sent when on hold
Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0' In sip.conf I have rtpkeepalive=15 but that does not seem to help. Does anyone know what I can do to fix this, other than increase the timeout on Bria? Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] testing please ignore
On 04/08/08 07:23, John covici wrote: If I see this, then messages are getting through. You are lucky :-/ I sent two messages to the list and they never arrived. -- #Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advice on best operator phone (with attendant console)
To be fair to the engineers at grandstream - an update to the latest 1.1.6.16 firmware seems to make the phones very very stable. I now have a couple GXP2000s running at high call volume for the past 3 days without any issue (usually it would happen within an hour). Problem is that Polycom/Aastra seem to not be interested in sales outside US and Europe. Their channel management seems quite weak and their sales people simply seem uninterested in third world country sales. After being on the phone with Polycom US for 30 minutes I still could not get hold of a person responsible for APAC/EMEA sales (my call got transferred 6 times). Maybe its just my bad luck but it has happened twice now :) Grandstream on the other hand is extremely helpful in negotiating good deals, giving heavy discount, arranging for shipping from nearby warehouses etc.. I think the problem may be that they release their firmwares WAY too quickly, earning them a bad reputation. On Sun, 2008-04-06 at 09:41 +0500, faraz wrote: Guys thanks a lot. I should be going with a Polycom 650 for all such jobs. If grandstream receives such bad reviews- how are they selling anything? Phones hanging or voice cut-outs are simply unacceptable!! On Sun, 2008-04-06 at 14:12 +1000, Rob Hillis wrote: I'd find that very strange considering that the 57i itself has facility for at least 20 BLF buttons and each attendant console has facility for another 60! Matt Watson wrote: We are using 57i + 560M combination as well... though we are not using the 57i ct... but the idea of giving them a cordless is a good idea. The only downside to the Aastra 57i + 560M is that it can only subscribe to 50 extensions for BLF... i haven;t run into this cap yet myself, but I have heard others talk about it... I think it was a cap introduced in one of the newer versions of firmware... not sure though, and not sure why. I'm running the latest 2.2 firmware on it... the addition of one-touch transfers in the last firmware was very nice so operator can transfer very fast, instead of having to do xfer-BLF key-xfer (for attended transfer), now they can just hit the BLF key for a blind transfer. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Sigma Networks [EMAIL PROTECTED] Sent: Saturday, April 05, 2008 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Advice on best operator phone (with attendant console) We have been marketing ipPBX systems based on asterisk for 3+ years. For the last year we've been placing Aastra 57iCT with 560M sidecars. Our attendants like the idea of a cordless handset so the attendant can go to the copy room, etc. The LCD based sidecar means you can keep it up to date without marking up paper strips. We deploy Thirdlane PBX Manager which allows us to setup the BLF (busy lamp field) via a web interface. Aastra 57iCT: http://neobits.com/aastra_-_a1758-0131-10-05_-_57i_ct_p11471.html Aastra 560m: http://neobits.com/aastra_-_a1760--10-55_-_560m_p11472.html Thirdlane PBX Manager: http://www.thirdlane.com/products/pbxmanager Feel free to contact me off list if I can be of any assistance. Regards, Jim ph: 408-701-9929 Faraz R. Khan wrote: One of our clients is using a Grandstream GXP2000 with an attendant console. We have used the same phone with past clients successfully however this particular operator processes around 200 calls a hours and the GXP2000 for sure does not like the quick line shuffling and call volume. We get the following problems randomly: 1. menu stops working 2. transfer key stops working 3. Line 1 LED gets stuck 4. Voice 'gaps' (blackouts) for 4-5 seconds 5. The phone also completely locks up regularly 6. ping response goes from 8ms to 3000ms (after which the phone locks up) Wondering which operator phone would work best. I have the following choices: 1. Linksys SPA 932/962 with attendant console 2. Polycom 601/650 with attendant console I cant confirm online whether the BLF functionality will work with Asterisk 1.2.26. Is somebody using either of these phones in a high volume environment successfully? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] MWI for voicemail - H323
Hi , How does the Asterisk provide Voicemail Message waiting indication to an h323 endpoint configured with Asterisk. Please provide the required Setup / comfiguration details or redirect to appropriate to resource. Awaiting an earliest positive response. Thanks in advance, Anisha ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Message waiting indication(MWI) for voicemail - to H323 endpoints
Hi , How does the Asterisk provide Voicemail Message waiting indication to an h323 endpoint configured with Asterisk. Please provide the required Setup / comfiguration details or redirect to appropriate to resource. Awaiting an earliest positive response. Thanks in advance, Anisha ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call forwarding
hi im starting off with asterisk and i need to know how to do call forarding... in out old telephony system we used to press *21*number# and all the calls would be forwarded to that number and we used #21# to undivert is that possible in asterisk and how do i do it i have attached my extensions.conf file and would appreciate it if you could help me out with some code if its possible to make such a diversion work urgent thank you in advance - You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
