Hi all,
Any good TTS (free or commercial) for asterisk?
Rgds,
Ringo
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Thanks. Do it support multi-language?
On Mon, May 24, 2010 at 11:55 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Mon, 24 May 2010, Rilawich Ango wrote:
Any good TTS (free or commercial) for asterisk?
I like Cepstral with the Allison (Smith) font. Allison Smith does the
sounds
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rilawich Ango
Sent: Monday, May 24, 2010 11:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TTS for asterisk
Thanks. Do it support multi-language?
On Mon, May 24, 2010 at 11:55 PM, Steve
Hi all,
I am looking for a voice recognition technology integrated to
asterisk. Any suggestion about it?
Ango
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Hi all,
Refer to http://www.voip-info.org/wiki/view/Asterisk+config+features.conf.
It shows we can use variable BLINDTRANSFER to call back the one who
transfer the call. However, in my tests below. The result is not as
expected.
case 1:
A calls B (dial(sip/B||Tt)
B answers and connects to A
B
Hi,
I have 3 queue set in the table as below.
name,autopause
1000,1
2000,1
3000,1
In queue 1000, the autopause works after member failed to answer call.
However, other queues don't work for the autopause function.
queue 1000:
-- Nobody picked up in 25000 ms
-- Auto-Pausing Queue Member
Thanks. Finally, I find that it was caused by the use of the table wrongly.
On Thu, Oct 22, 2009 at 10:23 PM, Miguel Molina
mmol...@millenium.com.co wrote:
Rilawich Ango escribió:
Hi,
I have 3 queue set in the table as below.
name,autopause
1000,1
2000,1
3000,1
In queue 1000
Hi,
I have a queue and 3 agents in the queue like below
SIP/1001
SIP/1002
SIP/1003
When I dial the queue number, the agent start to ring. How can I get
the instance ringing agent as I want to pause the agent
(pausequeuemember) after the queue timeout? Any application or
variable can use to
Thanks. Is it possible to do the same after Queue command? After
Queue command, hangup will hangup the call and won't go to the next
priority.
On Mon, Aug 17, 2009 at 7:22 PM, Trevor Hammondstre...@concipient.net wrote:
On Monday, August 17, 2009, Rilawich Ango wrote:
Thanks. DIALSTATUS
Hi,
I have a 4 port analog cards with asterisk 1.4.26.1 (centos5.3)
installed. After I dial an outgoing call, it returns error and call
drop as below. Anyone can tell me what the problem is. ango
-- Executing [8...@internal:20] Dial(SIP/601-09425ab8,
dahdi/g0/8200|50|T) in new stack
[Aug
,hangup
exten = s-CONGESTION,1,Congestion
On Mon, Aug 17, 2009 at 8:24 AM, Rilawich Ango maillist...@gmail.com
wrote:
HI,
Actually, I want to do the following.
A (user) talks to B (CS). At the end of the talk, B hangup and A will
goto the survey system. That's why I need to play prompt
:
On Friday, August 14, 2009, Rilawich Ango wrote:
Hi,
Can I play a prompt after hanging up a call? I have tried below but
failed.
...
exten = s,n,Dial(SIP/1234)
...
exten = h,1,Playback(demo-instruct)
-- Executing [...@macro-safedial:2] Playback(SIP/3601-09856bf0,
demo-instruct
Hi,
Can I play a prompt after hanging up a call? I have tried below but failed.
...
exten = s,n,Dial(SIP/1234)
...
exten = h,1,Playback(demo-instruct)
-- Executing [...@macro-safedial:2] Playback(SIP/3601-09856bf0,
demo-instruct) in new stack
[Aug 14 17:24:03] WARNING[2496]:
Hi all,
Any good recommendation of IP phone in term of sound quality and
price (reasonable) using with asterisk?
ango
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Thanks. I wonder do I need to reload it if I am using
realtime/database? I have to change the accountcode during the call
so it is not possible to do it if reload is needed.
On Fri, May 29, 2009 at 9:35 PM, Tarek Sawah tareksa...@hotmail.com wrote:
accountcode is a setting you add to your SIP
Hi,
I use realtime and I found that changing accountcode needed to
restart asterisk to activate that code and shown in CDR. Does it has
a way to update accountcode without restart asterisk?
ango
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I am using 1.4.24 with realtime.
On Fri, May 29, 2009 at 5:21 PM, Rilawich Ango maillist...@gmail.com wrote:
Hi,
I use realtime and I found that changing accountcode needed to
restart asterisk to activate that code and shown in CDR. Does it has
a way to update accountcode without restart
Hi all,
I download asterisk-addon 1.6.1 but the VoIP phone failed to
register to the system with the message below.
[May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317
realtime_mysql: MySQL RealTime: Invalid database specified: asterisk
[May 26 15:45:11] WARNING[29665]:
, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Tuesday 26 May 2009 02:52:18 Rilawich Ango wrote:
Hi all,
I download asterisk-addon 1.6.1 but the VoIP phone failed to
register to the system with the message below.
[May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317
realtime_mysql
HI,
I want to allow user to press 0 to the voicemail if the user don't
want to wait in the queue. Below is what I set but it doesn't work.
Anyone can help? ango
file: features.conf
[applicationmap]
opervm = 0,self/both,Macro,opervm
file: extensions.conf
...
exten =
Thanks all. I figure out to exit the queue by setting context in queue.conf.
On Thu, May 21, 2009 at 11:20 PM, Kevin P. Fleming kpflem...@digium.com wrote:
Mark Michelson wrote:
Not to undermine Kevin's requests to read what is documented, I can say that
what you want actually will not be
Can you try to disable call waiting in your phone?
On Fri, May 15, 2009 at 6:44 AM, sean darcy seandar...@gmail.com wrote:
sean darcy wrote:
I have two internal analogue extensions off a TDM400P. If the first is
busy, I'd like to ring the second. So:
[incoming]
exten =s,1,Answer()
exten
Hi,
After following the messages to install flite, I can find the following files.
/usr/lib/asterisk/modules/app_flite.so
/etc/asterisk/flite.conf
That's mean flite is installed successfully. Then I restart asterisk
but nothing found for that module.
sip*CLI core show application flite
Your
Hi,
I follow the
web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
- mohstream.sh , to configure music on hold to play using mms but
failed. Anyone can play using mms?
ango
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Rilawich Ango wrote:
Hi,
I follow the
web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
- mohstream.sh , to configure music on hold to play using mms but
failed. Anyone can play using mms?
ango
when they answer?
Wouldn't it just be better to play a message after party a answers and then
start ringing party b so that party a knows what's going on?
2009/4/24 Rilawich Ango maillist...@gmail.com
Hi,
Feature originate can be used make call thro' the web. There is a
parameter ,Async
Hi,
Feature originate can be used make call thro' the web. There is a
parameter ,Async, in it. I set it to true but there is no effect.
Actually, I want to do the following.
What I know the function originate is:
originate call --- party A
party A rings
party A answers call
party B rings, party
Normally, there are 10 concurrent calls in peak. You are right that
usage g729 is due to bandwidth consideration.
On Thu, Apr 23, 2009 at 2:42 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Thu, 23 Apr 2009, Rilawich Ango wrote:
Hi all,
I wonder who has the same voice quality
Hi all,
I wonder who has the same voice quality problem as what we have.
Below is our configuration.
Company --- asterisk 1.4.22 (g729) --- CISCO --- T1 --- customer
Sometimes, customers told me that they heard our voice not very clear,
like a call from far far away. I heard the recording is
HI,
Recently, I found that asterisk fail to get the correct context of
the sip phone. Below is the configuration and the log message. In
the log message, asterisk fail to identify the calling party. As a
result, it use a default context instead of int. Anyone know why and
how to fix it?
6, 2009 at 5:00 AM, Rilawich Ango maillist...@gmail.com wrote:
HI,
Recently, I found that asterisk fail to get the correct context of
the sip phone. Below is the configuration and the log message. In
the log message, asterisk fail to identify the calling party. As a
result, it use
The CLI shows zap is necessary for conference recording. Can I enable
conference recording if using ztdummy or dahdi, how? ango
-- Executing [...@owt_meetme:4] MeetMe(SIP/3601-c80b4520,
5599|rcixMP) in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe
My configuration is simple as below.
SIP phone - asterisk - CISCO - T1
Do you mean the hum noise is created by electric-magnetic field?
Asterisk can do nothing to eliminate it?
On Sun, Mar 29, 2009 at 2:47 AM, Steve Edwards
asterisk@sedwards.com wrote:
On Sat, 28 Mar 2009, Rilawich Ango
HI,
We are experiencing the hum noise when the conversion of 2 parties is
established. How can we eliminate that noise? ango
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Hi all,
I found that a new field lastms is used in 1.4.24. What is the
usage of that field and the datatype of it?
ango
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Tilghman,
Thanks. Can you elaborate the usage about it? What is the meaning
of each valid value in this field?
ango
On Mon, Mar 23, 2009 at 11:24 AM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Sunday 22 March 2009 21:40:14 Rilawich Ango wrote:
Hi all,
I found
Hi all,
I enabled recording (mixmonitor) in queue and process started after
queue member pick the call. But recording will stop after picking up
by another extensions of call transfer/parking in the same call. Is
it possible to continue to record the call for call parking/transfer,
how?
Rgds,
Hi all,
Is it possible to install more than 1 asterisk in a single server?
If yes, what do I need to set and take care?
Rgds,
ango
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It seems better to install once with multiple instances. Do we need
to take care the port or IP of each instance?
On Wed, Feb 25, 2009 at 5:36 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
Klaus Darilion wrote:
Rilawich Ango wrote:
Hi all,
Is it possible to install more than 1
to
/usr/local/sbin/safe_asterisk2
Cheers
Geraint
You will also need to look at asterisk.conf in the new installation
directory and as a quickfix to get it running, use a different location for
astrundir
2009/2/24 Rilawich Ango maillist...@gmail.com
- Show quoted text -
Hi all
I also experience that problem. Is it a bug?
On Wed, Feb 4, 2009 at 5:53 AM, Mark Michelson mmichel...@digium.com wrote:
Remco Barendse wrote:
1.4.23.1 is quite badly broken and there are no significant new
features
There are no new features at all, actually. What problems are you
the call.
Hope this might help you.
Regards,
Amit Mehta
Cell: +91 9898340962
On Tue, Jan 6, 2009 at 11:41 AM, Rilawich Ango maillist...@gmail.com
wrote:
Hi all,
I want to build a web page for user to input a phone number. Then,
the number will input to asterisk and it will makes call
Hi,
I wonder how I can relate the CDR records for the case of call
transfer. I can't find their relationship in CDR. Any can advice?
ango
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To
you.
Regards,
Amit Mehta
Cell: +91 9898340962
On Tue, Jan 6, 2009 at 11:41 AM, Rilawich Ango maillist...@gmail.com wrote:
Hi all,
I want to build a web page for user to input a phone number. Then,
the number will input to asterisk and it will makes call. At that
moment, asterisk
Hi all,
I want to build a web page for user to input a phone number. Then,
the number will input to asterisk and it will makes call. At that
moment, asterisk will make another call to a internal ext. Finally
asterisk will bridge 2 calls together for conversion.
Does asterisk can do it?
Yup I just copy and paste to it but it shown not a known channel.
On Thu, Aug 28, 2008 at 6:47 PM, Steven Howes [EMAIL PROTECTED] wrote:
Did you tab complete it to make sure it was right?
On 28 Aug 2008, at 11:39, Rilawich Ango wrote:
I got the message below after I issue the soft
hangup Local.
Andy
On 8/27/08, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi all,
I have the following queue and members. I found that there is a
call stuck in the queue so other call can't enter the queue. I want
to know whether we can remove the call (by CLI) to free the queue.
ango
:
Try CLI soft hangup Local.
On 28 Aug 2008, at 09:01, Rilawich Ango wrote:
Hi ,
Actually, there are 3 queues in the server. Only one queue (2700)
has problem. I want to reset or remove the caller only in 2700
without affecting other queues or calls. Does it work for my case?
On Thu
Hi all,
I have the following queue and members. I found that there is a
call stuck in the queue so other call can't enter the queue. I want
to know whether we can remove the call (by CLI) to free the queue.
ango
2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s
holdtime),
I have a realtime queue and the state of the queue member change as
below. Not-in-use (no call)- Unknown (ringing)- Not-in-use
(answered). The state shown in show queues does not really reflect
the state of the phone. I have searched the net and also the
UPGRADE.TXT by the warning message
Hi all,
I would like to know how can I immunize the background noise in my
case. Anyone can help? I have adjusted txgain rxgain in different
value but the result is the same.
ango
Below is my configuration.
asterisk1.4.21.1
zaptel1.4.11
addon1.4.7
TDM400 (FXOx4)
There is a very large
HI,
I got a one way audio when an ip phone dial to another ip phone in
the same network. What I find is TCP UDP run different legs. Below
is my configuration.
asterisk (192.168.1.10)
ipphone-A (192.168.1.111)
ipphone-B (192.168.1.101)
router (192.168.1.1) external IP (116.48.138.83)
When A
Hi,
I want to send some text to the phone such that the phone can
display the text on its display. I have tried to use SendText but it
doesn't work. Does the phone need to support when asterisk issues the
SendText application?
ango
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Hi all,
There is a setting called autopause in queue.conf to pause a queue
member if they fail to answer a call.
The autopause setting will pause the agent even when they are on the
line. I want to know if it is possible to pause the queue member only
when they don't answer after timeout?
ango
I have a queue with the following setting.
total queue member =30, autofill=1, timeout=25, monitor_format=wav49
asterisk 1.4.18
In busy hour, the loading of CPU reaches over 300%. At that moment,
all members are occupied and many calls are waiting in the queue.
There will be choppy and line cut
Hi,
A makes call to B. B has connection problem with the server (say, the
lan cable is unplugged).
1: A --- server
2: A --- server
3: server B
In 2, server will send the ring to A and it will hear ringing tone.
In 3, server will try to connect B until timeout.
My question is:
A will still
HI,
Does asterisk will ignore the setting in files if realtime is
applied, say asterisk will ignore all the setting in sip.conf if
realtime table sip_buddies is applied?
ango
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.
But why take chances anyway? Move all the relevant
conf files from /etc/asterisk to some other place to
be safe.
cheers
- Ben.
--- Rilawich Ango [EMAIL PROTECTED] wrote:
HI,
Does asterisk will ignore the setting in files if
realtime is
applied, say asterisk will ignore all
Segmentation fault occurs after executing the following cmd.
Dial(SIP/[EMAIL PROTECTED]|35|Ttr)
Is it a bug and how to fix it?
Below is the core dump message converted by gdb.
#0 0x068be1ad in realtime_peer (newpeername=0x1b37844 10.20.0.1, sin=0x0)
at chan_sip.c:2547
#1 0x068becb3 in
Do you mean the problem is solved using asterisk 1.4.18? Are you
using the setting as mine?
Below is my setting. One way audio is a result after A B connected.
PSTN (A)--1200P-- Asterisk -- GXP2000 --blind transfer -- Extension B
You can see that involve many parties in the blind transfer
Hi all,
Recently, I experienced one way audio after call transfer.
incalling call (PSTN) A -- GXP2000 thro' zap --blind transfer-- destination B
Finally A and B reach each others, but there is only one way audio.
Anyone get the same experience before? How to solve the problem?
Asterisk
Hi all,
In my understanding, we can use mssql as a database of asterisk
thro' unixodbc. And we can easy using mysql (realtime) to do the
same. Now, I want to keep 2 connections, one is mysql and one is
mssql. Because both database have information that needed to be read
from asterisk. Can I
) UNSIGNED DEFAULT 1;
For following this issue, see http://bugs.digium.com/view.php?id=12445
Regards,
Atis
On Sat, Apr 12, 2008 at 4:42 AM, Rilawich Ango [EMAIL PROTECTED] wrote:
Do you mean autofill works in 1.4.x? But it doesn't work even I set it.
On Fri, Apr 11, 2008
Anyone can update me about the queue sticking by a caller? Is it
solved in version 1.4.x? How?
On Sat, Apr 12, 2008 at 9:42 AM, Rilawich Ango [EMAIL PROTECTED] wrote:
Do you mean autofill works in 1.4.x? But it doesn't work even I set it.
On Fri, Apr 11, 2008 at 11:07 AM, BJ Weschke
Do you mean autofill works in 1.4.x? But it doesn't work even I set it.
On Fri, Apr 11, 2008 at 11:07 AM, BJ Weschke [EMAIL PROTECTED] wrote:
Rilawich Ango wrote:
Thanks. I have checked that the queue.conf. I keep the default
setting as autofill=yes in my tests. That's mean even
HI all,
I have set up a queue with 2 members (A B). 1st call is waiting
in the queue and a queue member A is ringing but don't take the call.
Member A keeps ringing. Then 2nd call is also get into the queue but
I found that queue member B doesn't ring. That's mean member B is
available to
to no
;to keep backward compatibility with the old behavior.
;
autofill = yes
On Thu, Apr 10, 2008 at 8:57 PM, Don Pobanz
[EMAIL PROTECTED] wrote:
Rilawich Ango Thursday, April 10, 2008 3:28 AM
I have set up a queue with 2 members (A B). 1st call is waiting
in the queue and a queue
]
[mailto:[EMAIL PROTECTED] On Behalf Of Rilawich Ango
Sent: Wednesday, March 12, 2008 6:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk out of service
Hi all,
I got the following message in the log yesterday. After that, no more
Hi all,
I got the following message in the log yesterday. After that, no
more in/out bound call can be made. What is the meaning of the
message? ango
[Mar 12 09:26:17] ERROR[29565] chan_sip.c: We could NOT get the
channel lock for SIP/2367-d8062fb0!
[Mar 12 09:26:17] ERROR[29565] chan_sip.c:
HI all,
How can I modify the from address in sip message? Say, I will a sip
account 1234. I want to change the from address in sip message of
this sip account to 4321.
From: 4321 sip:[EMAIL PROTECTED];tag=as5b42e6
ango
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I have multiple queues in my case. Do you mean multiple queues is one
of the reason to consume memory? How to only reset the queue stats?
You will see asterisk behave its worst with multiple queues and heavy
dialplan logic. I restart my boxes with queues everynight at midnight
just to reset
Hi all,
I found that there will be a memory leak if asterisk running day by
day without restart. Is it good to restart asterisk service daily?
What is the better way to restart it daily like apache?
ango
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asterisk is a way to do it but you have to
maintain a crontab. Is it possible to use logrotate instead? Or
other better way?
On Feb 13, 2008 3:26 PM, Atis Lezdins [EMAIL PROTECTED] wrote:
On 2/13/08, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi all,
I found that there will be a memory leak
yes.
On Feb 1, 2008 12:07 PM, Russell Bryant [EMAIL PROTECTED] wrote:
Rilawich Ango wrote:
Hi,
The server log shows the following message.
[Jan 29 04:59:02] WARNING[1896] config.c: Realtime mapping for
'sippeers' found to engine 'mysql', but the engine is not available
Does
Hi,
The server log shows the following message.
[Jan 29 04:59:02] WARNING[1896] config.c: Realtime mapping for
'sippeers' found to engine 'mysql', but the engine is not available
Does it mean the server failed to file the mysql server? I have
installed mysql and both asterisk and mysql are
Hi all,
I have a TDM400 with all FXO on it. When I make an outgoing call, I
can hear callee but callee claims the volume is too low so that he/she
can't hear very clear. Can I adjust to increase the volume in callee
side? Is it increase the value of txgain can solve the problem?
ango
I have a TDM400 in the server. I want to press **1XX to pickup a
call. It is ok if I pickup a call dialled from 1XX to 1YY (internal
network call). However, it is failed to pick up a call from PSTN
thro' TDM400 card. It seems I can't guess the correct context of it.
How can I know the context
Below is what I got from CLI
[Jan 7 23:02:46] NOTICE[3450]: app_directed_pickup.c:159 pickup_exec:
No target channel found for 111.
On Jan 7, 2008 11:48 PM, Rilawich Ango [EMAIL PROTECTED] wrote:
I have a TDM400 in the server. I want to press **1XX to pickup a
call. It is ok if I pickup
Hi,
Is it possible to let asterisk auto dial out and play the IVR? How?
i.e.
-asterisk auto dial out (use outgoing folder?)
-user pick the call
-play IVR (1-for English, 2-for Chinese)
-Then user can press the number to go through the level of IVR.
On Dec 17, 2007 4:32 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Dec 17, 2007 at 11:34:42AM +0800, Rilawich Ango wrote:
Below is the log I got. It seems related to Polarity Reversal.
--zapata.conf--
;answeronpolarityswitch=yes
hanguponpolarityswitch=yes
--full log--
[Dec
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rilawich
Ango
Sent: 17 December 2007 14:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] dial, answered and then hangup
On Dec 17, 2007 4:32 PM, Tzafrir Cohen
Hi all,
I will a TDM card with FXO modules on it. Below is the dial plan.
When someone can 9123456, CLI will show dialing to 123456 and
answered. After answered, the call hangup. I would like to know what
will cause the case to happen. Anyone can give me some advice to
solve it?
exten =
Below is the log I got. It seems related to Polarity Reversal.
--zapata.conf--
;answeronpolarityswitch=yes
hanguponpolarityswitch=yes
--full log--
[Dec 15 19:35:35] DEBUG[2195] res_config_mysql.c: MySQL RealTime: Retrieve SQL:
SELECT * FROM oi_systemalias WHERE alias = '2272'
[Dec 15
Try to increase the value of busycount. It may help to solve the problem.
On Dec 14, 2007 10:47 PM, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi All;
For me, I am in Kuwait and using the TDM22B and I used
all the below settings and did not resolve my problem,
I do not know if there is any
busydetect = yes
hanguponpolarityswitch = yes
Which of the two?
busydetect will work almost always. But it is suboptimal: it may sotimes
accidentally detect running calls. And it takes a few seconds to detect
a hangup.
Do you mean we need to adjust the value of busycount (larger than
Hi,
How can I merge 2 gsm files into a single file? I have tried to use
soxmix as below but failed.
soxmix 1.gsm 2.gsm 1-2.gsm
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Hi all,
I have a TDM400 with all FXO in it. I can make outgoing out but the
call will be dropped between 20-30mins suddenly. Below is the message
shown in the log in the time the call drop.
[Dec 10 23:23:32] DEBUG[3613] dsp.c: ast_dsp_busydetect detected busy,
avgtone: 200, avgsilence 75
[Dec
HI,
I have tried to add the context but it still doesn't work.
On Dec 9, 2007 11:36 PM, F6HQZ [EMAIL PROTECTED] wrote:
Hi,
Your extension 100 doesn't exist in the context where you have your PickUp
instruction.
You must include the context containing your phones into the context used by
in either of those...
daveC
Rilawich Ango wrote:
HI,
I have tried to add the context but it still doesn't work.
On Dec 9, 2007 11:36 PM, F6HQZ [EMAIL PROTECTED] wrote:
Hi,
Your extension 100 doesn't exist in the context where you have your PickUp
instruction.
You must include
Hi all,
I have a GXP2000 with BLF configured. I follow the configuration
guide to enable the pickup cmd as follow and include it under
corresponding content.
[BLF_group_pickup]
exten = _**1XX,1,Pickup(${EXTEN:2})
exten = _**1XX,n,Hangup
The I press the single key to pickup the call to
I have a TDM400 with all FXO module in it. Only one channel (say
channel 3) is plugged to PSTN. In my understand, a dial command
Dial(zap/g1/12345677) should search an available channel, which is 3,
in group 1 to make a call. However, I found that it will still use
channel 1 to make call even it
on it. Why it still try channel 1 and make call using it?
On Nov 25, 2007 5:00 AM, Gordon Henderson [EMAIL PROTECTED] wrote:
On Sat, 24 Nov 2007, Rilawich Ango wrote:
I have a TDM400 with all FXO module in it. Only one channel (say
channel 3) is plugged to PSTN. In my understand, a dial
HI,
I have 2 TDM400s plugged in a PC. I failed to use same channels to
make a call to PSTN. It shows it can't establish connection after
dial command issued. Below is the log. Actually, the call is
established as I can hear voice from the called party but the
softphone is still showing
Hi,
We are using attended call transfer to transfer the call. In the
direct call, the quality of the voice and dtmf are acceptable. After
transfer, the quality becomes worst. Voice can't be heard clearly and
dtmf wrong detection will occur sometime. I wonder call transfer will
affect he
Does it help to turn on dtmf log in each servers?
On Nov 16, 2007 5:01 PM, 木木 [EMAIL PROTECTED] wrote:
I think you haven't capture the packet from the beginning of the call.
You must capture the SIP packets. And the wireshark will recognise the
packets as RTP.
Hi,
Below is my case.
phoneA (PSTN)
phoneB (SIP)
phoneC (PSTN)
phoneA -- asterisk -- phoneB
phoneA (music on hold), phoneB --attended call transfer-- phoneC
phoneA --connect-- phoneC after phone B hangup
phoneA type some keys in keypad but phoneC always has wrong dtmf detection.
In my case, I
Hi all,
Can I simply the voicemailmain IVR? I just only want some of the
option in voicemailmain, ie read or delete messages. Is it possible
to configure that function?
Ango
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You mean modify the source? Could you give me an example, say I wrong
to remove advance option?
On Nov 14, 2007 1:59 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
vi app_voicemail.c
On Nov 13, 2007 10:34 PM, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi all,
Can I simply
I got the cause of the problem. I set canreinvite=yes and the
mentioned error gone.
On Nov 10, 2007 12:27 AM, Steve Davies [EMAIL PROTECTED] wrote:
On 11/9/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Steve Davies wrote:
I would hazard that it is the port number of '0' that is
Hi all,
I have seen a lot of message talking about asterisk crashed when
using queue and mixmonitor together. I do use both in our system and
also get the crash (segfault) randomly. I don't know it is related to
the reason above as I have no idea about how it happened. I get the
core dump
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