Displaying 20 results from an estimated 1000 matches similar to: "Looking for a reporter for SQLite3 with Lighttpd and PHP"
2013 Feb 06
1
Problem using ast_tls_cert script
Hi List,
I'm trying to set my Asterisk 1.8.20.1 with TLS on CentOS 5.9, it was easy
and straightforward with Debian 6.0.6, but when I introduce this command on
CentOS:
#./ast_tls_cert -C 10.200.108.17 -O "MyCompany" -d /etc/asterisk/keys/
I got this error message:
hostname: Unknown host
Same result happens when using server's hostname:
#./ast_tls_cert -C ast-centos -O
2013 Jun 19
6
Mailing a fax with mutt does not succeed
Hello everyone,
I'm trying to send a received fax with mutt, when I try it from the Linux
shel it works, but when trying with Asterisk's System command it doesn't.
Successful Linux command:
echo | mutt -s "New fax" earohuanca at gmail.com -a /tmp/faxes/201306191111.tif
Unsuccessful Asterisk Command:
same => n,System(mutt -s "New fax" elder.arohuanca at
2013 May 15
3
Cut offs on outgoing SIP calls
Hello everyone,
I've suffering cut offs after 6 or 7 seconds a call is answered, incoming
calls are working fine, but outgoing ones show the gollowing messages when
are being dropped:
[2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt:
Retransmission timeout reached on transmission
ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical
Response) -- See
2013 Apr 12
1
Polycom Soundpoint IP 330 provisioning
Hello all,
I need the bootrom.ld file to set up some Polycoms I have
Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A
I've publiched on my FTP files downloaded from
http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html
(3.2.3 combined and split zips) but my phones are still showing the
message: "error, application is
2012 Nov 29
3
Need qualifications of SIP trunk providers
Hello List,
Since I'm looking for a new VoIP provider for US origination/termination, I
will very appreciate if you can chare your experience with Flowroute,
Vitelity and Voip.ms
Thanks in advance!
Elder D. Arohuanca
dCAP 1497
Lima - Peru
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2013 Nov 25
1
Asterisk 11.6.0 not starting up
Hello Friends:
I've just installed Asterisk 11 on my Linux (debian) server but it is not
starting up when trying with "asterisk -vvvvvvvvvvc" and "service asterisk
start". Starting process just stop and shows: "Illegal instruction" as
final output.
Looking at logs I fouind at /var/log/asterisk/messages :
[Nov 25 11:09:26] Asterisk 11.6.0 built by root @
2010 Jul 08
1
Incoming call doesn't finish when internal phone hangs up
Hello guys,
I have this problem when a call is received in my PBX:
(Caller) --> (Redirecting Service) --> (E1 PRI) --> (Asterisk PBX) -->
(Internal Phone)
Reception works fine, but when conversation finishes and the agent at
internal phone hangs up, the call at caller's side is still alive for
many seconds until it hangs up.
The problem is that Telephone Company is billing me
2015 Aug 14
2
chan_sip.c: Retransmission timeout reached on transmission
Hello friends:
I am facing cutoffs randomly when negotiating calls.
The PBX dials the destination, the provider (softswitch) receives the
request *[1]* and sudenly the PBX hangs up the call* [2]* while the
provider is still dialing it, as a consequence the remote peer receives a
ghost call. Along the atempt I could see six times a messages regarding NAT
isuues *[3]*
I hope anyone can give me an
2008 Sep 27
3
test call generator
Hello everyone
I am trying to look for a free test call generator that will get me some
stats like PDD, ASR and call quality etc on each route. As well as do test
at every interval too
If you know something like this please enlighten me.
Sam
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2009 Feb 22
1
I can`t send DTMFs through FXO lines - dahdi
Hi,
I've just installed DAHDI at two PBXs as follows:
*PBX-1 PBX-2*
FXO ------------- FXS
When I try to send calls from PBX-1 to PBX-2 I just receive the message:
"Starting simple switch on 'DAHDI/1-1"
It seems like if PBX-1 couldn't send DTMFs, but when I set immediate=yes at
chan_dahdi.conf at PBX-2 dialplan is executed at PBX-2 but nothing is heard
at
2008 Oct 14
1
SIP channels seem not to close after call is finished
Hello everyone,
I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my
queue interfaces, despite the fact it is free at that time, can you give
help?
1. I see many sip channels from that extension:
[root at mysweetpbx]# asterisk -rx "*sip show channels*" |grep 648
Peer User/ANR Call ID Seq (Tx/Rx)
Format Hold
2014 May 28
1
Asterisk crashes suddenly
Hello friends,
I have been experienced suddenly stops for my Asterisk server, I do not why
is it happening. Asterisk's debug messages only tell me I have lacked g729
codec for translation to one peer minutes before the crashes occur
[2014-05-27 09:48:30] WARNING[15384][C-0000017c] channel.c: Unable to find
a codec translation path from (ulaw) to (g729)
[2014-05-27 09:48:30]
2011 Feb 07
1
About maxlen parameter in queues
Dear list,
I want to avoid sending calls to a queue when it is full. From the fact that
'maxlen' must be at least 1 (I wish it could be zero but it isn't) I'd like
to know if there's a way to do it. Setting the Queue() timeout to a little
value is not the most suitable option.
I'm using asterisk 1.4.21 but I don't know if there are some options
available on release 1.8
2008 Oct 02
1
Asterisk Queue question
When the asterisk a queue reset their counters?
I 'm talking about this kind of info in asterisk console.
>show queue 600
600 has 0 calls (max unlimited) in 'ringall' strategy (4s
holdtime), W:0, C:14, A:8, SL:0.0% within 0s
I just say that because I have a queue with strategy "Fewest Calls"
working for a couple of mouths, and a new agent has been added this
2013 Dec 23
0
How to recognize the Telco provider on outgoing calls only by sounds?
Dear list:
When I call an specific number on the PSTN, the provider who holds the
destination number give back an specified sound just after admitting their
incoming calls. Is there a way to allow Asterisk to compare sounds received
to decide what is the Telco answering the call?
I'm planning to do it to select the right provider to route further calls
at least cost.
In my country there are
2009 Apr 28
2
How to get PBX's clock with AMI?
Dear all,
I wanna know what can I do to get the PBX's clock from
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090428/3218b8b0/attachment.htm
2005 Apr 21
2
do not understand what to do to correct this error
----------------------------------------------------------------------------
----
FULL RSYNC LOG
----------------------------------------------------------------------------
----
/usr/bin/rsync -av --force --delete-excluded
--exclude-from=/usr/local/etc/snapback/snapback.exclude -e ssh --delete
peru.cbm.mercyships.org:/ /home/backup/peru/hourly.0/
<bunch of lines deleted>
wrote 873039
2010 Mar 19
0
Setting Caller ID for attended transfer
Hello list,
I'm sending calls to a queue in the attended way, that is, *1.* the original
call is put on hold, *2.* a second line is open to call the queue,
*3.*after an agent is connected the original call is transfered to its
final
destination.
1. Zap/1-1 <--> SIP/agentA-tag1
2. SIP/agentA-tag2 <-->
SIP/agentB-tag
3.
2009 Nov 05
4
collumn error when exporting to Excel
Dear all,
I am attempting to export my results (data.frame) created with the help of a number of you to Excel.
In the procedure my column structure is however lost and all results are placed together into the first Excel column.
I have tried: write(), write.table(), write.matrix(), export() and have the same results. I Have checked the import/export FAQ and did a Google search to no avail.
Any
2008 Jun 13
1
dependency on /usr/lib/nx
I do a:
sudo yum update
and I get:
yada, yada, yada,...
---> Package freenx-server.i386 0:0.7.2-8.el5 set to be updated
--> Running transaction check
--> Processing Dependency: /usr/lib/nx for package: freenx-server
Importing additional filelist information
--> Finished Dependency Resolution
Error: Missing Dependency: /usr/lib/nx is needed by package freenx-server
Not only