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True agree, problem is somehow the people purchased.... am supporting to overcome that. Trying level best... around 20 phones has been purchased.... ?? On Mon, 5 Dec 2016, 8:55 p.m. Victor Villarreal, <mefhigoseth at gmail.com> wrote: > With all the money you plan to invest in firmware, licenses, etc., you > have bought a Grandstream IP phone or Yealink... > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided...

...es ldap suffix = dc=mastercg,dc=com ldap ssl = no ldap user suffix = ou=Users idmap uid = 500-50000 idmap gid = 500-50000 printing = cups print command = lpq command = %p lprm command = Thanks for your time. -- Ing. Fernando Villarreal Proyectos Especiales Direcci?n Operaciones S?LICE S.A. Tranquilidad Tecnol?gica San Mart?n 1035 Of.49 Cdad. Mendoza - Argentina (M5500AAK) Telefax: +54 261 4294141 (PBX) Ext. 37 http://www.silice.biz Telefon?a IP: voip.silice.biz ----------------- AVISO LEGAL ----------------- Este mensaje...

...nd+Asterisk *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com http://www.ictbroadcast.com Leveraging open source in ICT On Tue, Feb 21, 2017 at 12:28 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: > On Mon, Feb 20, 2017 at 11:36:24AM -0300, Victor Villarreal wrote: > > Hi, Oliver. > > > > Maybe something like this (add this script to your crontab): > > > > ------------------------8<-------------------------- > > > > #!/bin/bash > > # > > # File: asterisk-watchdog.sh > > # Date:...

Hi Thufir, The analysis of a SIP Debug depends on what the problem to be solved. If you experience problems with inbound calls from a SIP trunk or provider, you can type in Asterisk cli 'core set debug 3' and then 'sip set debug ip xxx.xxx.xxx.xxx' where xxx is the IP of your SIP provider or from where it is supposed to come call. Then you make a test call, and look in full log

I think you need to increase verbose output and search in /var/log/asterisk/full for any error message related to IAX2 registration or simil. 2017-06-05 17:12 GMT-03:00 <thelma at sys-concept.com>: > No, I don't think it is IP table issue, I've not upgraded dd-wrt for a > while and it was zoiper was working OK with my previous version of > asterisk. > > After upgrade

..." <1006> disallow=all host=dynamic allow=all nat=yes Is NAT value set to yes OK? Servers is on public IP, client is on private network. Thanks, Motty From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Victor Villarreal Sent: Thursday, October 13, 2016 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit Hi Motty, Please, set Verbose to 3 and Debug to 3 At Asterisk CLI. Then "sip set debug on"....

Hey; What happens is that a script logs into the AMI and originates a call. When the call is answered, it jumps to a context in the dial plan. Thanks Much; John V. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Victor Villarreal Sent: Monday, May 15, 2017 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Automatically dial a number, then an extension Hi John, I think we need to known how you play the audio to the customers, before we can help you. Are you us...

Hi, Oliver. Maybe something like this (add this script to your crontab): ------------------------8<-------------------------- #!/bin/bash # # File: asterisk-watchdog.sh # Date: 2015.05.26 # Build: v1.0 # Brief: Secuencia para monitorizar procesos. # # ${PATH}: Variable de entorno con las rutas a los ejecutables. PATH=/bin:/sbin:/usr/bin:/usr/sbin # ${DAEMON}:

Ok, Please, check your manager.conf and logger.conf for any clue about debugging options, into the Asterisk configuration directory. El 26 mar. 2017 14:52, "Telium Technical Support" <support at telium.ca> escribi?: > I tried that but it had no effect. Still see things like: > > > > [2017-03-26 13:49:39] DEBUG[2088]: manager.c:5693 match_filter: Examining >

On 2017-04-18 08:31 PM, Victor Villarreal wrote: > Maybe excecuting the following command at Asterisk console, will help you: > > asterisk> voicemail show users > > And you will get a list of all mailbox configured in your system. Search > for the user with problems. VoiceMail stocktrans2 Angelica Douglas...

Hi, I''ve a vsat internet link and I want to know if I can make VOIP calls under it. ( Ping in the best case reach 900/ 100 ms by sattelite effect ) Can with SIP protocol make voip calls ? Thanks andres

...other nice sip packet is sngrep Shows realtime the sip flows But i think you have to chk the asterisk answer in the dialplan logic to chk what context its hitting etc. ?????? 6 ????? 2016 10:05 PM,? "Steve Edwards" <asterisk.org at sedwards.com> ???: > On Wed, 6 Jul 2016, Victor Villarreal wrote: > > If you experience problems with inbound calls from a SIP trunk or >> provider, you can type in Asterisk cli 'core set debug 3' and then 'sip set >> debug ip xxx.xxx.xxx.xxx' where xxx is the IP of your SIP provider or from >> where it is suppose...

AFAIK g729 patent is expiring sometime in 2019-2020. Mitul Limbani On Feb 8, 2017 5:02 AM, "Victor Villarreal" <mefhigoseth at gmail.com> wrote: > Hi Steve, > > I understand your question and your point, but I use the g729 codec from > the link that Carlos share, for almost 6 years from Asterisk 1.4 to v13 > without a single problem. > > So, sory but I don't share yo...

Actually now I have the phones with SIP firmware. I will try with 3pcc firmware along with XML files. Or any idea if we have CUCM application can we change the firmware. am ready to buy the developer edition. Regards . On Mon, 5 Dec 2016, 3:34 p.m. Steve Davies, <davies147 at gmail.com> wrote: > I tried... repeatedly... I failed. I bought some 3PCC phones, and they > just worked.

> On Tue, Feb 7, 2017 at 4:47 PM, Steve Edwards <asterisk.org at sedwards.com> wrote: > Now that the g729 patents have expired, how do we use g729 in > Asterisk? > > Will Digium be releasing a g729 codec for 'free' use or do we > download the 'free' codec off the Internet now that we can use it > without moral or legal

Hello, fresh install of Asterisk 13.11.2, client unable to register. For now I have IPtables disabled, also selinux is disabled [1006] type=friend username=1006 secret=mysecret context=sip-phone call-limit=1 callerid="iuser" <1006> disallow=all host=dynamic allow=all any ideas? Thanks, Motty -------------- next part -------------- An HTML attachment was

Does asterisk listen on port 4569 by default? I'm running version Asterisk 11.25.1 and have a problem registering Zoiper (IAX) to Asterisk. I'm getting an error: Registration refused -- Thelma

All; I have an application that dials a list of numbers and then plays a recorded message. My customer uses it to dial a list of customers to confirm their appointment for the next day. No biggie, maybe 25 - 30 calls per day for customers who want the confirmation call. What they need now is a way to dial an extension after the number is dialed and answered. I've seen that before, but I

...innovations.com >> http://www.ictbroadcast.com >> Leveraging open source in ICT >> >> >> >> On Tue, Feb 21, 2017 at 12:28 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com >> > wrote: >> >>> On Mon, Feb 20, 2017 at 11:36:24AM -0300, Victor Villarreal wrote: >>> > Hi, Oliver. >>> > >>> > Maybe something like this (add this script to your crontab): >>> > >>> > ------------------------8<-------------------------- >>> > >>> > #!/bin/bash >>> > # &g...

sorry... typo.... the problematic phone has the 192.168.0.13 the asterisk has 192.168.1.211 when i connect a snom phone on the cable that was in the soundstation 6000 before and configure the phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... it would be helpful if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am