Re: [asterisk-users] voicemail message not deleted

> Am 26.07.2021 um 07:28 schrieb Fourhundred Thecat <400the...@gmx.ch>: > > Hello, > > I have this in my voicemail.conf: > > attach=yes > > delete=yes > > I do get an email when new voicemail is received, and I do get the > voicemail message as attachment. > > However, the original

Re: [asterisk-users] BLF support in Asterisk and early/confirmed/terminated/proceeding NOTIFY states.

> Am 03.10.2020 um 03:26 schrieb Jobst Schmalenbach : > > I have a setup with Yealink phones & Asterisk Server (all latest patches). > > I am using BLF to display the states of other phones. While this works MOST > of the time (busy, being called) it does NOT work when a phone is NOT >

Re: [asterisk-users] Voice broken during calls (again...)

You could also use the 'mtr' command under Linux. > Am 22.06.2020 um 17:41 schrieb Marek Greško : > > Hello, > > try pinging your sip peer ip address following way: > > ping -n -M do -s 1300 -i 0.1 -c 100 ${ipaddress} > > Post several lines and the statistics. > > Were you also thinking

Re: [asterisk-users] Voice "broken" during calls

> Am 14.06.2020 um 16:38 schrieb Luca Bertoncello : > > Am 13.06.2020 um 22:56 schrieb Antony Stone: > > Hi again, > >> 2b. Take your Thomson telephone to some other location with Internet access, >> let it register to your home Asterisk server, and them make a call to the >> same >>

Re: [asterisk-users] Voice "broken" during calls

So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello : > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer " for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177

Re: [asterisk-users] Voice "broken" during calls

> Am 13.06.2020 um 13:36 schrieb Luca Bertoncello : > > Am 13.06.2020 09:30, schrieb Luca Bertoncello: > > Hi again (again) > > I noticed right now another strange detail... > I made a call using my mobile phone (connected to the Asterisk). The quality > was top... > Maybe is the problem in

Re: [asterisk-users] MWI Delayed on Polycom VVX phones

> Am 15.01.2019 um 15:23 schrieb Doug Lytle : > > Hi all, > > When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has > resulted in a MWI clearing delay of around 5 minutes. > > After listening to a voicemail and deleting it, the Polycom VVX 601's MWI > light is left on

Re: [asterisk-users] Best Asterisk Platform

Am 23.12.2015 um 14:31 schrieb er ic : > What is the best asterisk platform to use? What are you guys using? > > I am looking for something to host either in our data center or at the > customer prem where I have the control over the unit and not through a >

Re: [asterisk-users] how can queue agents choose which call to answer?

Am 23.09.2014 um 19:49 schrieb Marie Fischer ma...@vtl.ee: Hi everybody, I'm looking for a solution for the following scenario: • Asterisk queue • At peak hours, there will be more callers then queue members/agents, so some callers will spend some time on hold • Agents should be able

Re: [asterisk-users] How to implement priority queuing within a single queue ?

Am 25.01.2013 um 17:22 schrieb Olivier: Hi, Let say that in a call center, callers are recognized and categorized in 4 priority levels (priority 1 for Very Very Important Personalities, 2 for VIP, and so on) before entering a Queue. How can you make sure a priority 2 caller is answered

Re: [asterisk-users] How to implement priority queuing within a single queue ? [SOLVED]

Am 25.01.2013 um 17:39 schrieb Olivier: 2013/1/25 Michael Keuter li...@mksolutions.info Am 25.01.2013 um 17:22 schrieb Olivier: Hi, Let say that in a call center, callers are recognized and categorized in 4 priority levels (priority 1 for Very Very Important Personalities, 2

Re: [asterisk-users] Is there a need to secure RTP ports?

Am 23.01.2013 um 18:33 schrieb Carlos Alvarez: On Wed, Jan 23, 2013 at 10:20 AM, Sebastian Arcus s...@open-t.co.uk wrote: I have an Asterisk server with one SIP trunk to a SIP provider. As my server registers with the SIP provider, I don't have any SIP ports open at my end to the Internet.

Re: [asterisk-users] Asterisk 11 - Security event logging over syslog

Am 24.11.2012 um 11:25 schrieb Michael Keuter: Am 24.11.2012 um 01:33 schrieb Matthew Jordan: On 11/22/2012 04:00 PM, Michael Keuter wrote: Hi all, I am just testing with Asterisk 11.01. The SIP security event logging works fine for me for console and file logging, but the security

Re: [asterisk-users] Asterisk 11 - Security event logging over syslog

Am 24.11.2012 um 01:33 schrieb Matthew Jordan: On 11/22/2012 04:00 PM, Michael Keuter wrote: Hi all, I am just testing with Asterisk 11.01. The SIP security event logging works fine for me for console and file logging, but the security events are not logged over syslog. logger.conf

[asterisk-users] Asterisk 11 - Security event logging over syslog

Hi all, I am just testing with Asterisk 11.01. The SIP security event logging works fine for me for console and file logging, but the security events are not logged over syslog. logger.conf: ... syslog.local0 = notice,warning,error,security Is this on purpose, a fault on my side, or is this a

Re: [asterisk-users] Intruder

Am 16.11.2012 um 18:08 schrieb Michael L. Young: - Original Message - From: Felix Vazquez felix.vazq...@theboshgroup.com To: asterisk-users@lists.digium.com Sent: Friday, November 16, 2012 11:20:46 AM Subject: [asterisk-users] Intruder I am in the asterisk CLI and can see an

Re: [asterisk-users] Asterisk 10/1.6.1 and Dahdi/Libpri compatilities in BRI /PtmP

Am 21.06.2012 um 18:05 schrieb Richard Mudgett: My previous message was incomplete. On thing to note is I had to forbid hfcmulti in modprobe.d in the second box to comply with a warning from dahdi. Without that, I could see this line in the output of lsmod: mISDN-core hfcmulti 1.

Re: [asterisk-users] Dual- or Quad ISDN cards for PCI-X Slots

I recently bought 2 Beronet BN4S0 cards on eBay, each for under 100 €. they have 4 S0 ports. Sent from my iPad Michael Am 29.05.2012 um 14:18 schrieb Kevin P. Fleming kpflem...@digium.com: On 05/29/2012 01:48 AM, Michelle Konzack wrote: No, it does not fit, since PCI 2.0 is 5V and has only

Re: [asterisk-users] looking for solid state like PC suitable for Asterisk

Am 10.05.2012 um 15:03 schrieb Bart Coninckx: There seems to even be a 1.6 Ghz Intel Atom device. One site I'm looking to use this for has about 40 SIP phones and three BRIs. It's always a guessing game whether devices like this are up for that. If they do have some processing power, I

Re: [asterisk-users] Pickup calls coming from queues

Am 23.01.2012 um 23:25 schrieb Alec Davis: How can I test this solution on a 1.8.8.1 system ? If I'm not mistaken, diff https://reviewboard.asterisk.org/r/1619 do not apply to 1.8.8.1. I've just checked out 1.8.8.1 and download my patch from

Re: [asterisk-users] Pickup calls coming from queues

,,,127) ;Ring the queue [blf] exten = 8501,hint,Queue:itg_queue I'm sure I've over complicated my examples. Alec -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Keuter Sent: Thursday, 26 January

Re: [asterisk-users] Anyone have a reliable T.38 Solution

Am 05.01.2012 um 04:55 schrieb Matt Darnell: On Wed, Jan 4, 2012 at 1:02 AM, David Klaverstyn da...@klaverstyn.com.au wrote: I'm using the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and rx_fax on multiple installations with no problems. David, Are you running 10.0 or