similar to: CALL SETUP TIME

> On Thu, May 28, 2009 at 02:00:15PM -0500, research at businesstz.com wrote: >> Hello >> >> May i please know if asterisk is now supporting sip call encryption. It >> has been a requirement from one of my client to ensure that all >> conversation is well secured from any potential sniffers or inside >> hackers >> >> I have reviewed and shall

Hello May i please know if asterisk is now supporting sip call encryption. It has been a requirement from one of my client to ensure that all conversation is well secured from any potential sniffers or inside hackers Please help or suggest any solution that you feel may help Kind regards Sam

I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has

Hai.. After a new setup i was confronted again with the unable to login with RDP. so here is some extra info for the debugging this. I used RDP to connect a Windows 7 64 bit, connected in rdp with ipadres of the pc. and again unable to login. since im trying to setup a smb.conf with minimal changes, i only added : auth methods = sam, winbind restarted samba on both DC's and yes..

I see with in Achim response, " Also I read the manual . . ." What manual? How do I a copy of "the manual?" Just asking. --- ------------------------- Bob Wooden of Donelson Trophy 615.885.2846 (main) www.donelsontrophy.com [4] "Everyone deserves an award!!" On 2015-04-30 08:51, Achim Gottinger wrote: > Hi Louis, > > Am 30.04.2015 um 15:31

Hai Achim, i have tested the following : auth methods = winbind result RDP login works, ADUC does not work. test with : auth methods = winbind, sam RDP and ADUC works, DNS tools also works. logged in as DOMAIN\administrator Greetz, Louis >-----Oorspronkelijk bericht----- >Van: achim at ag-web.biz [mailto:samba-bounces at lists.samba.org] >Namens Achim Gottinger

I have a scenario whereby two servers are acting in active-standby mode. In case the active server fail, the shared IP is activated on standby server for continuity. However, SIP phones (all are Polycom) takes quite a long time to register to the Standby Server (up to 1-10min). While Polycom allow double registration, we would like to make it simple by provision only one registration server at a

Hi Louis, Am 30.04.2015 um 15:31 schrieb L.P.H. van Belle: > Hai.. > > After a new setup i was confronted again with the unable to login with RDP. > so here is some extra info for the debugging this. > > I used RDP to connect a Windows 7 64 bit, connected in rdp with ipadres of the pc. > and again unable to login. > > since im trying to setup a smb.conf with minimal

Greetings!! We're getting some strange results using call files.. no matter the technology, DAHDI, SIP, etc., we get a "Call failed to go through, reason (3) Remote end Ringing" message when attempting to originate a call from a call file. Numbers changed to protect the innocent.... using call file.... //------------CALL FILE------------// Channel: DAHDI/g1/918005551212

Hai Rowland / Andrey, that.. was a stupid one to miss.. that was because it was checking against defaults of samba, forgot to put that one back.. and yes, tested it also with, and im noticing the same. (different id's ) so.. back to winbind... and now id's are same again.. thanks. . and andrey, im using my sysvol scripts to set it up. have a look here,

On Sat, Mar 14, 2020 at 2:02 PM John Roman <john at dev1ce.com> wrote: > greetings asterisk users :) > ive just deployed version 17 and migrated as best I can to pjsip. I can > receive calls, and get to my mailbox prompt, however placing calls seems > impossible with the following error on dial: > > Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel

Hai Rowland, Ok, cool, thanks for that. Thats good to have that confirmed, the search show the same here. Enabled that one, and yes, i can see the msNPAllowDailin but only in attribut editor, Dail-in tab still errors. Reappy-ing the file : MS-AD_Schema_2K8_R2_Attributes.txt Is that possible, that "should" fix the missing parts. I suspect a failure in the structure of the AD. (

Greetings, upgrading from 1.4.17 to 1.4.19 some asterisk gateway of ours (used for gatewaying ISDN-PRI and SIP), I noticed an annoying thing: when the PSTN party answers, for a few seconds (4/5 sec typical) some SIP client could not hear anything (the ringing was heard well!), then the audio comes back again with no problem. Looking for any differences between the behaviour of version 1.4.17 and

You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN

When I execute a pickup on a ringing phone I get CALL FAILED REASON CODE 603. I am dialing **212 with the following config. Anyone have a suggestion? EXTENSIONS.CONF -snip- [BLF_Group_Pickup] ; Defines how the extension to pick up a ringing phone in your BLF group exten => _**XXX,1,Pickup(${EXTEN:2}) exten => _**XXX,n,Hangup() [BLF] ; Defines a BLF Hint for phones exten =>

Hello, all. I've got a problem where we set up call pickup for a customer. If the Bob's extension rings and Bob is in Jim's office, Bob can press the button on his Snom 320 that says "Bob" and pick up his line. It works great for calls coming in from the outside but does not work for internal calls. Internal calls generate a app_directed_pickup.c:204 pickup_exec: No

Hi, I've looked through the list archives, bug tracker and cvs changelogs and can't see anything that refers to the particular problem i've seen recently. I'm running CVS-D2004.06.09.14.00.00-06/24/04-00:43:55 which I realise is not exactly recent, but I wanted to find out more before I updated. We've had asterisk running for our office using 7960's and a TE410P for a

Hi Shahnawaz Have you considered how you are going to address location issue for Mobile users calling 911. You should think of SS7 MAP/TCAP to atleast know their Cell ID Regards Sam > Thanks very much everybody who contributed their thoughts. I would try > to get some DID's so that each physical location can be identified by > 911 call Center. > > Regards > > Shahnawaz

greetings asterisk users :) ive just deployed version 17 and migrated as best I can to pjsip. I can receive calls, and get to my mailbox prompt, however placing calls seems impossible with the following error on dial: Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 517890) dunkel*CLI> dunkel*CLI> == Setting global variable 'SIPDOMAIN' to

Hi everybody, I am trying to make up call flow diagrams for for a setup which include ser as a sip proxy/registrar and asteriks as a voicemail server. Is my sequence correct?: UA 1 send an invite to SER. SER forwards this invite to UA2. UA2 sends back a sends back a 100 trying and 180 ringing message. SER forwards these. However UA2 doesnt answer the phone,so what happens then?...is there a