I had in a same situation and solved by Background 1 sec. silence.
On Wed, Nov 25, 2015 at 5:45 PM, Brian :: wrote:
> add a pause in the dialplan for a second then proceed..
>
>
>
> On Wed, Nov 25, 2015 at 2:27 PM, Tony Mountifield
> wrote:
>
>> In article <20151125133008.6369360.14455.17...@gm
Your call is up on VoiceMail you should check dialstatus before sending
user to VoiceMail.
On Thu, Aug 7, 2014 at 4:12 PM, D'Arcy J.M. Cain wrote:
> This just started after upgrading to 11.11.0. After a call is
> completed (both ends hang up) the call still shows as active.
>
> # asterisk -x "
you can use GROUP and GROUP_COUNT
n,Set(GROUP()=aname)
n,GotoIf($[${GROUP_COUNT(aname)} > 8]?${EXTEN},200)
200,Hangup
On Thu, Jul 10, 2014 at 3:24 PM, Rafael Visser
wrote:
> Hi guys.
>
> Does somebody knows how to get the concurrent calls from the dial plan?
>
> Or.
>
> How can i control not t
if that is the case then check again Perl Asterisk AGI.
On Mon, Apr 28, 2014 at 7:33 PM, Haley,Scott A
wrote:
> One more thing. I have this exact same script working on an Asterisk 1.8
> box. This is a new Asterisk 11.7 box.
>
> Thanks,
> Scott Haley
> 5-2244
>
>
> -Original Message-
> F
file is executable?
can you show ls -l /var/lib/asterisk/agi-bin
On Mon, Apr 28, 2014 at 7:12 PM, Haley,Scott A
wrote:
> It runs but hangs with the output of:
> perl tbsdial.agi 81101
> GET VARIABLE astexten
>
>
> Right now, it is a simple perl script. Here is the entire script.
>
> #!/usr/bin/p
Hello,
Try this
[6004]
type=friend
host=dynamic
disallow=all
allow=ulaw
allow=alaw
callerid=6004
secret=XXX
context=default
port=9060
nat=force_rport,comedia
deny=0.0.0.0
permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, PrvIP/255.255.0.0
On Wed, Apr 16, 2014 at 12:56 P
Hello,
you can check the asterisk binary with.
file /usr/sbin/asterisk
and linked library
ldd /usr/sbin/asterisk
On Wed, Nov 20, 2013 at 2:51 PM, Jonas Kellens wrote:
> On 20-11-13 14:43, A J Stiles wrote:
>
> On Wednesday 20 November 2013, Jonas Kellens wrote:
>
> Hello,
>
> I have installe
if you don't use MOH just don't load module res_musiconhold.so
On Fri, Oct 18, 2013 at 6:24 PM, Alban Elziere wrote:
> Thank you for pointing this thread.
> So, looks like no solution exists to correct this (as I understand)... as
> it is part of the standard. Have you found a trick to avoid tha
some more information's will help sort out the issue.
On Fri, Oct 18, 2013 at 2:30 PM, shiva kumar wrote:
> Dear All,
>
>
> i had an issue when we are going to call back the number from asterisk
> its ringing as the customer mobile is switched off.
> And also it also not saying busy when
We are using Debian 32bit and 64bit on standalone and on VMs without any
issue.
On Thu, Oct 17, 2013 at 10:15 AM, Frederic Van Espen
wrote:
> On 10/17/2013 09:47 AM, Alban Elziere wrote:
>
>> I'm using Ubuntu server (32bit mainly), standalone or VM (esxi) with good
>> stability.
>>
>
> Same her
9 codec. isn't it true?
>
>
> On Sat, Oct 12, 2013 at 1:08 PM, Asghar Mohammad wrote:
>
>> HI,
>> You don't need a g729 installed in pass throw mode. if both ends have
>> codec g729 you can just enable on both peers.
>> and asterisk should pass the codec from 1
hi,
you have not mentioned which cdr backend you are using.
peer ip is saved in variable CHANNEL(peerip).
if you are using mysql for cdr backend you can create a field in cdr table
(field name can b any of your choice)
in dialplan assign the value of CHANNEL(peerip) to you ip field and
asterisk wil
HI,
You don't need a g729 installed in pass throw mode. if both ends have codec
g729 you can just enable on both peers.
and asterisk should pass the codec from 1 end to other.
but make sure you are not doing transcoding of any type answering the call
playing voice prompts etc.
On Sat, Oct 12, 20
Hi,
Bad boys trying to guess a valid username.
in sip.conf uncomment alwaysauthreject=yes and Asterisk always reject 1st
invite.
On Tue, Oct 1, 2013 at 5:26 PM, Gareth Blades <
mailinglist+aster...@dns99.co.uk> wrote:
> On 01/10/13 15:44, gincantalupo wrote:
>
> On Tue, Oct 1, 2013 at 5:07 AM,
Hi Asmaa,
Have you enabled debug to console in logger.conf?
enable debug in logger.conf console => notice,warning,error,debug and
reload Asterisk.
On Sun, Sep 29, 2013 at 4:48 PM, Asmaa Ahmed wrote:
> Hi Asghar,
>
> Thanks a lot for your proposed solution!
> MWI is turned on or off by the prese
HI Asmaa,
I don't know how MWI works in Voicemail but as i understand it just create
a .call file and put in /var/spool/asterisk/outgoing and asterisk execute
that file.
i am using similar method for sending fax to from email.
i show you some examples from my php scripts.
1. in voicemail contex
Hi,
If you post your configuration someone may help you.
On Sat, Sep 28, 2013 at 5:03 PM, Sean Darcy wrote:
> On 09/27/2013 09:08 PM, Sean Darcy wrote:
>
>> We have zoiper connected over iax to asterisk in Sydney. The call is to
>> asterisk in New York. The caller in NZ can hear clearly. Nothin
Hi,
Please Search the List there is already a post and solution.
On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg wrote:
> Hi All,
>
> This is my 1st post so lets go.
>
> What I need to achieve is the following. I have server with both IPv4
> addresses and IPv6 addresses. The problem that
Hello,
paste you extension context.
On Fri, Sep 20, 2013 at 4:21 PM, Asmaa Ahmed wrote:
> Hello,
>
> I have Asterisk 1.8.10.1
> Moving to nat=force_rport,comedia hasn't solved the problem. Still having
> the same error!
>
> I am not sure if this is related to the problem here, but I was trying
Hello,
i think your logic is wrong please explain me what are you trying to do?
[internal]
exten => 7002,1,Answer()
exten => 7002,n,Playback(vm-nobodyavail)
exten => 7002,n,Hangup()
exten => 7001,1,Dial(SIP/7001,60)
exten => 7001,n,Hangup()
try this dial 7002 and you should listen vm-nobodyavail
Hello,
If Asterisk version is > 1.6 use nat=force_rport,comedia
On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed wrote:
> Hello,
>
> I have set the direct media to be off, but still doesn't work. I am not
> sure about NAT configuration!
>
> SIP.conf, [general] section
> context=internal
> allowgues
you have insecure=port,invite in sipgate peer?
On Thu, Sep 19, 2013 at 12:26 PM, wrote:
> On Thu, 19 Sep 2013, Miguel Oyarzo wrote:
>
>
>> Challenge authentication look good.
>>
>> <--- SIP read from UDP:217.10.79.23:5060 --->
>> SIP/2.0 200 OK
>>
>> Are you sure this number format 01179553708
remove content of /var/log/asterisk/messages "> /var/log/asterisk/messages"
run asterisk and post content of /var/log/asterisk/messages to pastebin.
On Thu, Sep 19, 2013 at 9:39 AM, Asmaa Ahmed wrote:
> Hello,
>
> No, another installation haven't solved the problem!
> It looks more like somethi
i think you messed 2 installs of asterisk.
if you compile asterisk from sources it not insert init script.
you can test installing to /opt.
1. cd to asterisk sources folder
2. make distclean
3. ./configure --prefix=/opt/asterisk
4. make
5. sudo make install
6. /opt/asterisk/sbin/asterisk -c
yo
SELinux exists in Ubuntu?
On Wed, Sep 18, 2013 at 2:45 PM, Ishfaq Malik wrote:
> Have you checked your SELinux settings?
>
>
> On 18 September 2013 13:13, Asmaa Ahmed wrote:
>
>> Hello,
>>
>> I have started using Asterisk recently on my Ubuntu server. I installed
>> it first using apt-get an
become root sudo su - or su -l give your password.
if asterisk is already running connect to "asterisk -rvvvc" otherwisw
"asterisk -c".
if you want asterisk run as daemon "asterisk" and then connect to asterisk
"asterisk -rvvvc"
On Wed, Sep 18, 2013 at 2:13 PM, Asmaa Ahmed wrote:
> Hello,
>
re/zoneifo) disable all existing options usegmtime etc.
added new cli option cdr mysql cdrzone. it will show you selected timezone.
patch can be download from
http://www.world-call-trade.com/asterisk/cdr_mysql_cdrzone.patch
please report back here.
BEST REGARDS
Asgha
Hi list,
I am using Asterisk1.6.2 form a long time and upgarding to
Asterisk-1.8.23.1.
I am using mysql backend for cdr.
in asterisk-1.6.2 i have usegmtime=yes and it works as expected insert cdr
date in GMT0.
now i tested Asterisk-1.8.23.1 and asterisk-11.5 with same results no
matter what i confi
i have used a2billing some time ago maybe there is somthing new .
you can try shoot up loglevel to 4 and see the verbose of agi that may give
you some hint.
On Tue, Sep 10, 2013 at 7:34 PM, jg wrote:
>
> Maybe the ringtone from downstream is not
>> reaching asterisk, and thus a2billing is ap
hi,
it seems your vpn connection drop.
is you vpn on WiFi of any other high latency network?
On Tue, Sep 10, 2013 at 1:05 PM, Administrator TOOTAI wrote:
> Hi all,
>
> I face the subject strange behavior: calls arre dropped after 15 minutes
> on an asterisk 1.8.15.0. Only phones (SNOM300) connec
sip set debug on and see trace of upload on pastebin.
On Wed, Aug 21, 2013 at 8:25 PM, jg wrote:
> At first I also thought this might be a phone setting. But then I found
> the same 60s to be true for a variety of SIP phones (Snom, Cisco, ...),
> despite the 300s timeout value in the Dial cmd.
he,
some bad boys trying to guess configured extensions.
in sip config in general set alwaysauthreject = yes .
in cli sip set debug on and watch ip and block in firewall, iptables.
On Mon, Aug 19, 2013 at 7:50 PM, Ira wrote:
> Hello Steve,
>
> Sunday, August 18, 2013, 3:35:54 PM, you wrote:
>
just remove username.
type peer authenticate by ip
On Sun, Aug 18, 2013 at 7:01 PM, Andrew Colin wrote:
> change server two to host = dynamic
>
> then add register = on server 1
>
> On 8/18/2013 6:29 PM, Gopalakrishnan N wrote:
>
> Even I tried the type as friend.. but no use...
>
>
> On M
As my understanding Asterisk always pass-thu g729 if both ends have this
codec.
But if you answer the call or play some audio before dialing to end point
then asterisk stay between both legs.
In case of VM. you should install g729 if your prompts are in g729 format.
As a2billing play voice prompts
make a call and post cli log
On Tue, Jul 2, 2013 at 11:54 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:
> still the peer shows unreachable let me restart and give a try...
>
>
> On Wed, Jul 3, 2013 at 2:49 AM, Asghar Mohammad wrote:
>
>> *1st Locatio
Unable to create channel of type 'SIP' (cause 20 -
> Subscriber absent)
> == Everyone is busy/congested at this time (1:0/0/1)
> -- Executing [2001@Test:2] Hangup("SIP/3081-27d2", "") in new
> stack
> == Spawn extension (Test, 2001, 2) e
Jul 2, 2013 at 10:25 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:
> can't we use without register command both way as peer to peer?
>
>
> On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad wrote:
>
>> 1. you permiting 10.10.10.0 on b but you should permit
onitored
>
>
> and the IP segment is two different segment. where am able to ping each
> other.
>
>
>
> On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad wrote:
>
>> hi,
>> paste server a trunk also, if you want register why you are not using
>> host=dyna
hi,
paste server a trunk also, if you want register why you are not using
host=dynamic?
both servers are on 10.10.10.0 ? if no then check your deny permit seting.
On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:
> Also tried one more scenario, particularly f
hi.
check here for agi
http://forum.voxilla.com/threads/introducing-waits-w-in-dial-destination-number-variable.14628/
On Fri, Jun 7, 2013 at 7:50 PM, Sean Darcy wrote:
> On 06/07/2013 01:17 PM, Yves A. wrote:
>
>> This would be possible with an agi...
>> the agi can wait for silence or 10 seco
hi,
you can add more w (ww1234#) for more delay.
On Fri, Jun 7, 2013 at 7:17 PM, Yves A. wrote:
> This would be possible with an agi...
> the agi can wait for silence or 10 seconds, as u like and then play the
> dtmf tones and bridge the call to your extension afterwards.
>
> yves
>
>
what is host architecture ?
try to install ubuntu x86 not x86_64.
On Thu, Jun 6, 2013 at 5:12 PM, wrote:
> I'm trying to install and run Asterisk 11 on Ubuntu 12.04.2 running over
> Oracle VM VirtualBox (v 4.1.8). So far I have tried it following two
> guides. The first is the one from "Asteris
asterisk trying connect to mysql via socket remove that line from config
files.
1 check if port 3306 is open in iptables on both servers.
2 check permissions on db for user Asterisk.
On Mon, Jun 3, 2013 at 9:18 PM, Olivier CALVANO wrote:
> on this server we don't have mysql.socket because he do
work around was block dtmf.
set wrong type of dtmf in incoming trunk.
On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:
> So any resolution for this?
>
> I suspect it could be related to RE INVITE
>
>
> On Tue, May 28, 2013 at
i had this in past there was an ATA configured to send 9 at the end of
dialing in my case.
On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:
> Hi,
>
> I am receiving DTMF without any reason after call establishment.
>
> The log as follows, and I suspect some
please provide more information.
how you are try to build asterisk, what is output of configure. witch
headers configure script not found etc.
On Tue, May 28, 2013 at 9:29 AM, upendra wrote:
> hi,
>
>
> anyone can help me to debug this ??
>
>
> --
> upendar
>
>
> On Mon, May 27, 2013 at 4:09 P
you don't need register => string here, it only need you want asterisk
register to another sip proxy as client.
just remove that line and you should fine.
for X-lite or any other sip phone the user "AlphaUser" is sufficient.
On Fri, May 24, 2013 at 12:32 PM, luke devon wrote:
>
> Hi all ,
>
>
sip set debug peer 90102 and check in log why call drop or upload log
somewhere. configuration seems ok.
On Wed, May 15, 2013 at 10:09 PM, Daniel - Asterisk wrote:
> Current configuration follows:
>
> [general]
> context=default
> allowguest=no
> alwaysauthreject=yes
> allowoverlap=yes
> allowtr
please show us peer configuration.
On Wed, May 15, 2013 at 9:46 PM, Daniel - Asterisk wrote:
> Users (softphones) are behind a NAT, Asterisk has its own public ip address
>
>
> On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad wrote:
>
>> asterisk is behind nat?
>>
&g
asterisk is behind nat?
On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk wrote:
> 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 12:5
what problem you encounter?
On Tue, May 14, 2013 at 9:42 PM, Lobna Hegazy wrote:
> Dear All,
>
>I'm trying to connect to Asterisk CDR database using PHPMyAdmin but
> unfortunately all my trials and searches failed. So I'd be more than
> grateful if someone helped me with right steps to d
i think DIALSTATUS is not suitable for failover if trunk is down you get
dialstatus after time out in dial string.
it is too late for failover, you can use some script to check if
destination host is up.
if you want to do failover when destination host is up then dialstatus are
good.
On Tue, May
Dial(DAHDI/i0/number, is it not Dial(DAHDI/r0/number or Dial(DAHDI/R0/number
or Dial(DAHDI/g0/number or Dial(DAHDI/G0/number?
On Mon, May 13, 2013 at 12:53 PM, Yves A. wrote:
> mmh... actually supportline is closed...
>
> why proceeds the call to dahdi/pseudo-??
>
> i have never seen this b
solved?
On Sun, May 12, 2013 at 5:39 PM, Joseph wrote:
> On 05/12/13 12:18, Asghar Mohammad wrote:
>
>> you can try to set usegmtime=no in cdr.conf
>>
>
> I commented it out, as "no" is the default setting; but for some reason it
> was enabled on G
you can try to set usegmtime=no in cdr.conf
On Sun, May 12, 2013 at 3:40 AM, Joseph wrote:
> Which file in Asterisk have a setting for time zone?
> When asterisk record incoming call in Master.csv the time is 6hr. ahead.
>
> I'm on: Canada/Mountain zone
> --
> Joseph
>
> --
> __
what you mean by interface?
if you want connect sip phone with asterisk there are 2 file to modify
1. sip.conf
2. extensions.conf.
for creating sip user add following in sip.conf
[ivr_user]
defaultuser=ivruser ;username for sip phone
secret=ivruser ;password for sip phone
context=ivrco
he is using debian. debian have yum?
On Sat, May 11, 2013 at 2:44 PM, Andrew Colin wrote:
> Do a yum install kernel-devel kernel-headers
>
> Reboot and it will work
>
> Sent from my iPhone
>
> On 11 May 2013, at 12:20 PM, Alec Davis wrote:
>
> >
> >
> >> -Original Message-
> >> From: a
installing kernel source on debian use "*apt*-*get insatll* linux-headers-$(
*uname* -r)"
On Sat, May 11, 2013 at 12:20 PM, Alec Davis wrote:
>
>
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> >
install kernel source.
On Sat, May 11, 2013 at 10:15 AM, Harish Mandowara wrote:
> Dear,
>
> I have redhat enterprise linux 6.3.
>
> after "uname -a" i am getting
>
> Linux genesys-dell 2.6.32-279.el6.x86_64 #1 SMP Wed Jun 13 18:24:36 EDT
> 2012 x86_64 x86_64 x86_64 GNU/Linux
>
> now when i am
p.conf? What are the directives
> exactly please?
>
> Thanks in Advance,
>
> Nick.
>
> On 5/10/13, Asghar Mohammad wrote:
> > hi,
> > you can try to change sip user agent and sdp session s , owner in sip
> > config same as your phone,s (modem).
> >
hi,
you can try to change sip user agent and sdp session s , owner in sip
config same as your phone,s (modem).
asterisk by default send user agent = asterisk version , s= asterisk , o=
asterisk.
some providers are not happy if they see "asterisk" word :)
On Sat, May 11, 2013 at 12:27 AM, Sergej
hi,
asterisk insert cdr when call is hangup and last dial statment,
i dont understatnd why you are using 2 dial statment on same extenstion?
if you you want dial to both extensions you can use
506,1,Dial(SIP/223&SIP/276) if you want dial both same time or if you want
to do failover the check Dial s
are you talking about sip to pstn? thats called fxo ATA.
On Tue, Apr 30, 2013 at 8:59 PM, Don Kelly wrote:
> Guys and gals - these are all excellent answers - I am not being clear, I
> think.
>
> ** **
>
> Let me see if I can illustrate it.
>
> ** **
>
> If you cannot see my diagramme,
try
UserByAlias=yes in general and type=user in user context.
On Fri, Apr 26, 2013 at 9:48 AM, s m wrote:
> oh yes, i'm using h323 not openh323
>
>
> On Fri, Apr 26, 2013 at 1:16 AM, Asghar Mohammad wrote:
>
>> nuFone h323 or openh323?
>>
>>
>>
nuFone h323 or openh323?
On Thu, Apr 25, 2013 at 9:33 PM, s m wrote:
> flavor? i do not understand what you mean. please explain more.
> thanks
>
>
> On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad wrote:
>
>> what flavor of h323 you are using?
>>
>>
>
what flavor of h323 you are using?
On Wed, Apr 24, 2013 at 8:50 AM, s m wrote:
> thanks Asghar,
> i do it, but no thing happened:(
> asterisk do not identify host line as ip address of the other end
>
>
> On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad wrote:
>
>&g
address of the other end in h323.conf
> file? i define the address by "host=192.168.0.146" but asterisk can not
> find it? why?
>
>
> On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad wrote:
>
>> please post cli output for both calls.
>>
>>
>> On Mon,
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m wrote:
> hello everybody
>
> i want to have sip connection between two asterisk systems (145 and
> 146). connection from 145 to 146 is ok but i can not call from 146 to
> 145.
> this is h323.conf file in 145:
> [peer14
AGI is your friend. check A2billing.
On Fri, Apr 19, 2013 at 10:43 AM, Lenz Emilitri wrote:
> Not sure if that's what you are looking for, but I would think about
> having the dialplan call a web service (maybe using CURL) and passing
> account and current number. The system would reply with the
wring with my extensions in extensions.conf
> but i don't know how to fix it.
> please let me know if you have any other suggestion.
> thanks
> sam
>
>
> On 4/11/13, Asghar Mohammad wrote:
> > hi,
> > try
> > exten=> _2.,1,Dial(SIP/to-232/2${EXTEN:
699403.18
> AGI Tx >> agi_version: 1.8.15.0
> AGI Tx >> agi_callerid: 2003
> AGI Tx >> agi_calleridname: Carlos Chavez
> AGI Tx >> agi_callingpres: 0
> AGI Tx >> agi_callingani2: 0
> AGI Tx >> agi_callington: 0
> AGI Tx >> agi_callingtns
; On 4/11/13 11:54 AM, Asghar Mohammad wrote:
> > i am using exten =>
> > _XXX.,n,Set(CDR(cli_name)=${CHANNEL(peerip)}) cli_name is field
> > in mysql and it work fine. show me cli output without AGI.
> >
> >
> > On Thu, Apr 11, 2013 at 6:41 PM, Carlos Chavez
i am using
exten => _XXX.,n,Set(CDR(cli_name)=${CHANNEL(peerip)})
cli_name is field in mysql and it work fine.
show me cli output without AGI.
On Thu, Apr 11, 2013 at 6:41 PM, Carlos Chavez wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> On 4/11/13 11:18 AM,
hi,
it is not difficult in php and mysql i have created a simple billing system
for my wholesale postpay clients without any AGI.
it report ACD ASR all calls ANSWERD calls filter by date by callerid etc.
do billing as soon as call end.
for billing i am using mysql trigger.
report live calls.
2 inte
hi,
you have not assign any value to CDR(userfield).
try
code => #111,self,SET(CDR(userfield)=111)
On Thu, Apr 11, 2013 at 12:53 AM, Carlos Chavez wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> I am trying to set the CDR(userfield) to a certain vaule using the
> application
hi,
try
exten=> _2.,1,Dial(SIP/to-232/2${EXTEN:1})
Note space before underscore.
On Thu, Apr 11, 2013 at 2:50 PM, s m wrote:
> this is my [from-trunk] extension:
>
> [from-trunk]
> exten=>_2.,1,Dial(SIP/to-232/2${EXTEN:1})
>
> and this is [to-231] in sip_additional.conf:
>
> [to-232]
> host=1
hi,
open debug only on problematic peer.
sip set debug peer "peer name"
or
sip set debug ip "peer ip"
On Fri, Mar 29, 2013 at 2:02 PM, Marie Fischer wrote:
> Hello everybody,
>
> I am trying to find an intermittent SIP error with one provider and
> thought the best first step would be to have "
Wimax and FH using a diguim
> cards
>
>
> 2013/3/22 Asghar Mohammad
>
>> hi,
>> i think we miss understood you Question?
>> you need round robin on tdm trunk or on 2 internet connections?
>> what are you asking about " burden-sharing between Wi
2 for the second provider in order to use the burden-sharing
> between Wimax and FH
>
>
> thanks and regards
>
> 2013/3/21 Asghar Mohammad
>
>> hi,
>>
>> exten => _0612.,1,Set(CALLERID(number)=520460587)
>> exten => _0612.,n,MixMonitor(zap_g2_$
hi,
exten 000,1.Progress() work in some situation.
On Thu, Mar 21, 2013 at 9:30 PM, Gerard wrote:
> On 03/21/13 14:14, Gerard wrote:
> >> I think a simple tcpdump of the traffic will show the mystery. It can
> >> be your provider doing something nasty. Have you tried using some
> >> other cheap
:)
On Thu, Mar 21, 2013 at 10:27 PM, Jaap Winius wrote:
> On Thu, 21 Mar 2013 12:21:44 -0700, Michael L. Young wrote:
>
> > How are you determining that it is not listening on IPv4?
> >
> > bindaddr=:: should allow you to support dual stack.
>
> That's what I thought would happen. When I set bin
please post sip.conf.
On Thu, Mar 21, 2013 at 8:01 PM, Nick Khamis wrote:
> Hello Everyone,
>
> I have disallow=all and allow=g729 set in sip.conf however, it seems
> that asterisk still thinks it support other codecs:
>
> Capabilities: us - 0x8008000e (gsm|ulaw|alaw|h263|testlaw). How
> can
hi,
${myVar}STATUS is empty you have not assign any value here your var
Set(__${myVar}STATUS=) is empty.
use instead Set(__myVar=${ARG1}STATUS) and remove second line.
On Thu, Mar 21, 2013 at 7:45 PM, Administrator TOOTAI wrote:
> Hello,
>
> I have a variable created like
>
> ... Set(__myVar=${A
hi,
exten => _0612.,1,Set(CALLERID(number)=520460587)
exten => _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =>
_0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten => _0612.,n,Hangup()
Not
please see,
http://lists.digium.com/pipermail/asterisk-users/2013-March/278130.html
On Thu, Mar 21, 2013 at 5:47 PM, Jaap Winius wrote:
> Hi folks,
>
> Following an upgrade to Debian wheezy, I'm now running Asterisk 1.8.13.1.
> As opposed to Asterisk 1.6.2.9 that I ran with squeeze, this versio
hi,
sflphone work fine installed and tested on debian with nat and without nat.
please check setting in preferences my sflphone use alsa device. you should
check with alsamixer maybe sometime mic get muted or you agent mute the mic.
also check out what advice Mitch.
NB. you can test with IAX also.
n a successful call and a failed
> call, and I can see no difference except for things like port numbers and
> call IDs.
>
> It only fails occasionally, not on every call.
>
>
> Mitch
>
>
> On 03/20/2013 01:16 PM, Asghar Mohammad wrote:
>
>>
>>
>>
On Wed, Mar 20, 2013 at 7:11 PM, Asghar Mohammad wrote:
> hi,
> problem seem to client end i am going to install SFLPhone i will let you
> know when finish, have you check firewall on clients pc?
>
>
>
>
--
_
Hi ishfaq,
if you want just loging some info into db you can do in dialplan without
any AGI.
i am doing billing on the fly in dialplan and mysql for every single user
without AGI and enhanced call capacity almost double.
let me know you need some examples.
On Wed, Mar 20, 2013 at 12:56 PM, Ishfaq
hi Bharat,
why you are giving same answer as mine over and over ? please read
posts carefully.
On Wed, Mar 20, 2013 at 6:48 AM, Bharat Lalcheta
wrote:
> Did u changed rtp.conf ?
> port is showing 39408. Asterisk definetly drop rtp packet for this port if
> not updated in rtp.conf
> Regards,
> Bha
rt.
try to change rtpend=3 or if there is option in softphone restrict it
to use same range as in rtp.conf.
let me know if this solve you problem.
On Tue, Mar 19, 2013 at 10:54 PM, Asghar Mohammad wrote:
> hi,
>
> "User Datagram Protocol, Src Port: 39409 (39409), Dst Port: 1342
witch softphone you are using? on client pc installed some kind of
virtualpc like vmware or virtualbox? client pc have more then one network
interfaces?
you can capture sip invites from soft phone by enabling debug on client ip
sip set debug ip "ip of softphon" upload sip trace then somebody can ha
hi,
rtp set debug ip 1.2.3.4
On Tue, Mar 19, 2013 at 2:09 PM, Mitch Claborn wrote:
> Thanks for the suggestions.
>
> 1) directmedia was taking the default of "yes". I set to "no". Will
> watch and see.
>
> 2) NAT is turned off (nat=no). I've never done any RTP debugging. Is
> that "rtp set d
hi,
try srvlookup=yes
On Tue, Mar 19, 2013 at 3:15 AM, Jaap Winius wrote:
> Hi folks,
>
> Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9
> to 1.8.13, my server is no longer able to register a connection to a SIP
> account at my ISP (XS4ALL in the Netherlands). At the sa
hi satish,
try to debug rtp on that ip and look rtp flow you can also test
directmedia=no i encounter this as well i server is on public ip and
clients connect via vpn , vpn server is also same asterisk server calls
come in via public ip and go to call center via vpn i solved this by
directmedia=no
uld help a lot if I could capture fragmented distribution of time per
> call -- time in IVR, Queue, Call etc.
>
> Regards,
> Sans
>
>
>
>
>
>
>
>
>
> On Mon, Mar 18, 2013 at 4:33 PM, Asghar Mohammad wrote:
>
>> hi,
>>
>> 00:00 -- Call Con
; 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec.
>
> In the given schematic what will be the "Answered" time and "billed" time.
>
> Thank you for the help in advance!!
>
>
>
>
>
>
>
>
>
> On Sun, Mar 17, 2013 at 10:
"If you have analog FXO ports then the call is considered answered as soon
as dialing is completed" not always true if FXO configured properly it
should not send back answered as soon as dialed.
On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling wrote:
> If you have analog FXO ports then the call is
hi,
billsec is time in seconds after call has answered, duration is total time
in seconds of call.
as your calls answered imidiatly your billsec and duration is almost same.
On Sun, Mar 17, 2013 at 5:14 PM, RSCL Mumbai wrote:
> Hi,
>
> Attached is a sample CDR.
>
> I need some help to understand
HI bilal,
I don't think DAHDI can send SMS you have 2 options chan_mobile or
chan_datacard "ex chan_dongle" chan_datacard i have not
tested but with some mobile phones you can send sms i have tested also with
some made in china unbranded phone that are capable to send and receive sms
but not good
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