search for: yves030

Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* *** -------------- next

...e 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 21.12.2016 um 15:13 schrieb Mauricio Tavares: > On Wed, Dec 21, 2016 at 7:50 AM, Yves <yves030 at gmx.de> wrote: >> Hi Mark, >> >> yes, you are right... these are different VLANs >> I configured the other phone to use the same IP (192.168.1.13)... and it >> worked flawlessly... on the SAME Networkcable in the same plug... >> so it must have something to...

hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . ?service zaptel restart? or there is any other command Thanks and regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the

hi, I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... calls go out and call come in... no 32 seconds limit. but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. as far as I know, there is no firewall in

Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y eth0 of the server is configured to 10.26.192.107 The Problem: SIP registration works, phone rings in- and outbound, but there is no audio, nor the caller neither the callee can hear

Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34

...IP with 3 handsets: all are ringing > on incoming calls (Administrator TOOTAI) > 3. Re: Asterisk 1.8, Siemens C610IP with 3 handsets: all are > ringing on incoming calls (Chris Bagnall) > > > ---------- Forwarded message ---------- > From: "Yves A." <yves030 at gmx.de> > To: Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users at lists.digium.com> > Cc: > Date: Sun, 17 Feb 2013 15:07:43 +0100 > Subject: Re: [asterisk-users] ODBC and SQLIte3 > looks like a mistake in your extconfig.conf... > do you wa...

On 22.11.2014, at 13:40, Yves A. <yves030 at gmx.de> wrote: > I have a really strange problem which is driving me crazy for days now. > > If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, > everything works... calls go out and call come in... no 32 seconds limit. > > but as so...

Hi, I migrated from asterisk 1.6 to 11.3. The Server has a Sangoma A104 quadPri card installed. OS is a fresh installed Ubuntu 12.04 64bit libpri, dahdi etc. all latest releases.. Sangoma says... driver is compatible with ANY asterisk version... I tried driver 3.5.8... Setup ended with error. I tried (latest) driver 7.0.1 Setup went through, Asterisk is showing dahdi channels... all fine....

.... am I wrong?=20 nevertheless... any more hints for my original feature-request? thank you all, yves Am 16.01.2013 19:03, schrieb Bharat Lalcheta: Please study meetme application's options. You will get almost all = feature you ask for in it On Jan 16, 2013 5:37 AM, "Yves A." <yves030 at gmx.de> wrote: Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to m...

I sent this last night but it never showed up in the thread list so I'm trying again. Please pardon me if it duplicates. So I've been banging my head against the rack on this one and am now turning to the group for help. I'm in the process of bringing five Asterisk servers (all originally built from source code by myself) from various versions (1.6.2.x,11.6-cert13, and 13.1-cert2) up

...-users at lists.digium.com> > Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but > only when.... > Message-ID: <CF4F37ED-8DDF-43DC-9E9C-79A292E86FAE at vtl.ee> > Content-Type: text/plain; charset=windows-1252 > > On 22.11.2014, at 13:40, Yves A. <yves030 at gmx.de> wrote: >> I have a really strange problem which is driving me crazy for days now. >> >> If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, >> everything works... calls go out and call come in... no 32 seconds limit. >...

Hi list, I have tried to set the value for "timerb" in sip.conf, general section and in user-context... tried on asterisk 1.4 up to version 13... no success. The value for timerb remains unchanged. (reload, restart, reboot.... all does not help...) sip show settings always show 32000ms for timerB. How can I configure the timerb value? thx, yves --- Diese E-Mail wurde von Avast

Hi, I have written an click2dial application that rings an agent soft phone and connects the agent with a customer. very often I can see, that the agent softphones signal a busy back to server, although the phone is definitely hung up and the previous calls where handled normally. I testet 3cx Version 6 and X-Lite V1 up to V3... all show the same misbehaviour. I did a SIP Trace and can see,

Hi, in my company we use an asterisk installation with around 50 soft- and hardphones of all kind. From time 2 time the users (almost only Softphone users) report some voice qualities... mostly echoes. These problems do not occur on all PCs at the same time and since setup of our PBX almost any PC user has gotten these issues. When I come there to check, everything is fine again... and I

Hi, I am using asterisk 11.8 in combination with spandsp to send and receive T38 Faxes. All works fine, but I do not know how to get the remoteheader from the fax I receive. When I send a fax, there are Faxopts to set the localstationid and the headerinfo, but for receiving, there seems to only exist the Faxopts remotestationid but for sure on any fax I receive there is a remoteheaderinfo

Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: >> but as soon as I configure another sip registration on another server, >> outgoing >> calls drop after 32 seconds. > Are both your servers behind the same NAT router? > thanks for taking part... I don?t know... one is siptrunk.ovh.net and the other one is sip.ovh.fr how can i determine and how could that affect... I

Hi, thanks for your answer. Unfortunately this is, what I already know. I was wondering, why it is possible to set ID and Header for an outgoing fax (which will then in turn be inserted via asterisk on top of the transferred "image") , while it seems to not be possible to get the Header from a received fax (only the id), although it is present in the faxdocument. The ID is also