similar to: valid users option

Hie, I can't connect to my samba 3.022 pdc (I can join the domain and access share) and no error messages. I find in the log : the dump_data message musn't be clearest (domain,netboisname...) ? An idea ? [2008/04/30 15:26:50, 5] lib/util.c:show_msg(464) size=627 smb_com=0x2e smb_rcls=0 smb_reh=0 smb_err=0 smb_flg=136 smb_flg2=51201 smb_tid=1 smb_pid=65279

Hi all: We are looking for someone with experience in Alcatel PBX - PRI - Asterisk integration Please get in touch off list.. We're wanting to hire a professional subcontractor, developer or company to get around some issues like these: Asterisk shows PRI to Alcatel is up, but when trying to dial from Alcatel to Asterisk results in a disc tone (Asterisk do send calls properly into

Hello i've samba 3.022 with a ldap 2.2.26. I've no pb to join domain with my win2000, but when I reboot I'm reject (bad username ...). I find in debug that the first param sent by the client was the login and I think it must be the machines name. Any idee ? -- Denis Rohou Service Informatique ville de Lannion 22113 lannion 02-96-46-64-22

Hi all, Did anyone had success with Q.Sig on * 1.2, especially with Alcatel 4400 (which seems to only support Q.Sig) ? I am thinking about interconnecting 15 sites together with asterisk (probably using IAX or SIP). I have a very heterogeneous environement using both PRI and BRI, but my pilot will start with the Alcatel at the central site. Any help/example is most welcome. BR, - Patrick -

Hi guys, I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P. Input calls VOIP Proider ---> Asterisk ---> Alcatel Output Calls VOIP Proider <--- Asterisk <--- Alcatel In alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems: 1. When users dial 2 on phone (alcatel) they don't received a dial tone,

Hello,all I can't add samba to the domain with the command net join, the error is like this: net join -U usrname -S ***** Enter usrname's password: [2009/03/26 11:03:55, 0] utils/net_rpc_join.c:net_rpc_join_newstyle(393) Error in domain join verification (credential setup failed): NT_STATUS_INVALID_COMPUTER_NAME Unable to join domain **. I can ping the domain server and

Hi, Does anyone have any experience connecting asterisk to alcatel 4200 series pbx with bri cards? Does it should work with asterisk bri in NT mode, and alcatel bri with TE mode? Cheers, Igor Neves.

Hi, Is Asterisk "fully QSIG-compliant"? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. I am using switchtype=euroisdn and all works fine. However, it seems that Alcatel's latest firmware has dropped support for euroisdn which is really despicable. So now I need to see if I can migrate to QSIG which is supported by

Hi, I have a PRI euroisdn link between an Alcatel PBX and Asterisk. I'm having some trouble with overlap dialing. Suppose I dial '874053' from an Alcatel extension ('7034') where '87' is an Alcatel prefix of type "ARS Prof.Trg Grp Seiz.with overlap". I'm expecting Asterisk to receive '1004053' (where '100' is a prefix which always shows

Hello, I have connected asterisk box with legacy PBX Alcatel OmniPCX 4400 (and also another * box connected to A4200). These PBXes have function to assign name to extensions and display it on phone. Asterisk box is connected via PRI with euroISDN signalling (also I have tried QSIG). Is it possible to set callerid with name and display it on alcatel digital phones? With command SetCALLERID

Hi, during gradual migration to Asterisk, I put Asterisk in front of a legacy Alcatel PBX: PRI PSTN <--> Asterisk <--> E1 cable <--> Alcatel PBX After successful deployment of VoIP phones, it's time to drop Alcatel PBX! I'd like to keep some of analog lines to support modem, fax and some older stuff. What's the best choice? A channel bank or a TDM2400P card? Can I

Does anyone have any input into integrating asterisk with a alcatel 4400 PBX. Acording to what i've found is that Alcatel uses R2 for E1 -- regards Vikram (http://www.vicramresearch.com)

Hello, I have 2 analogue ports on an Alcatel PBX patched to 2 FXO ports on my <mailto:*@home> *@home 2.8 running on top of CentOS. Both FXO Ports are on ONE Digium card. What I would like is: If someone calls extn 281 on my Alcatel PBX it is routed through to Extn 233 on my * thruogh FXO port/module 4 If someone calls extn 282 on my Alcatel PBX it is routed through to Extn 234 on my *

Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. For alcatel users use asterisk lines, should dial 0 to take tone from asterisk. In default configuration in alcatel, if user dial 0 this error occour: !! Unexpected Channel selection 3 -- Extension '' in context 'default' from '' does not exist. Rejecting call on channel 0/31, span 1 In alcatel

An Asterisk extension calls an Alcatel extension via a PRI link which rings 4 times for about 10-15 seconds and then drops. So if the Alcatel user doesn't answer within 10-15 seconds the call is aborted. (A timeout is *not* specified in the Asterisk Dial command.) It seems however that either Asterisk or Alcatel drop the call prematurely (it's more likely to be on the Asterisk side). What

Hi guys, Recentlly i did a asterisk gateway and use it with an alcatel pabx. All is working, i have only two problems. 1. When call incomming to asterisk, it forward to digium card to PABX Alcatel. The user that start the call can't hear the control tone of ring ring ring. Tha calls stay without sound until the called part answer the call. At this point, conversation follow normaly. 2. When

Hello, For a prospective customer, I need to evaluate the cost of an Alcatel OXE IP trunking licence price. The setup would be: PSTN -----<E1>---- Alcatel ----<SIP>----Asterisk ---------- SIP Phones At the moment, only a couple of simultaneous calls are needed between Alcatel PBX and Asterisk but later, this number will grow.

Hi, I'm seeing something I'd like suggestions on: I have a queue with agents that log in using agentcallbacklogin. The extension that is logged in with is a Local channel. Now, if a call comes in to the queue and is handled by an agent (in our case using Cisco 7960 SIP phones) and transferred (attended) to another extension, the agent remains unavailable during the remains of the call.

Hello everybody, I just buy a X101p clone and i'm new in asterisk. Here is my configuration : ISDN line ---- Alcatel ----PSTN ext 68-----Asterisk with X101p clone ------sip phone ext 200 - 203 ||| ISDN phones ext 60-67 >From sip phone to ext 60-67 it works. 9+extnumber >From sip phone to Land lines it works. 9+0+phone number >From ext

Hello, Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the latest 13.16.0 release), we have a problem with attended transfers to an alcatel pbx in which the call being transferred still has music on hold even after the transfer has completed. Is this a known issue? Any new flags that need setting, etc? Thanks Jason -------------- next part -------------- An HTML attachment