search for: stablish

Displaying 20 results from an estimated 43 matches for "stablish".

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2005 May 11
2
inter domain trust - another network
Hi all!!! I am trying to stablish a interdomain relationship, but i am having serious problem making samba and nt seeing domains in another network segments. Which is the best and easier way to configure samba to stablish relashionships between domains in differents network segments? For example i have domain A in network 192.168....
2006 Sep 15
2
VPND
Hi, I have one FC5 server and one CentOS 4.4 server and I want to stablish a VPN between them, i have read that VPNd could be an option, anybody know where do I find the rpm for CentOS or with repository should i have to install in order to get it? Also if somebody has some other software that recommend me to stablish a VPN -- Guillermo Garron "Linux IS user frie...
2007 Nov 08
0
make h323 native transfer on stablished call
Hi all: I don't know if exist any other mailing more apropiated for this question. If exist, please let me know. I need orientation for this situation: 1. 1.4.13-BRIstuffed with support for h323 with asterisk-h323 module 2. An analog Pbx with support por h323 make asterisk a call, that asnwer and put with MOH 3. At this point I want asterisk to make a native h323 transfer of the current
2010 May 26
2
Interdomain trust with different WINS servers
Hi I am trying to stablish a relation between two different Samba domains. Each domain has two PDCs, all they are WINS servers, and the two domains are in different subnets. They are configured to use LDAP as the user/group/machine database. First, I will create the relation, being DOM1.CORP the trusted domain, and DOM2.CORP...
2003 Aug 13
1
FWD SIP phone format=2, FWD call format=4, why?
Hi! I'm trying an asterisk-FWD connection. I'm using X-Lite OR SIPPS as the IP phone. I configured the X-Lite and SIPPS to use GSM codec. Whe I call FWD, I get this info on the channels when the call has not been stablished yet: sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.246.69.223 613 1770bf3430d 00102/00000 00000ms 0000ms 2 150.187.xxx.yyy ildefonso C72ACD25-1A 00101/11482 00000ms 0000ms 2 2 active SIP channel(s) -- SIP/fwd-161b...
2009 Sep 27
2
Check value interval in a if loop
...ptions: if(pvalue>0.05 or pvalue<0.1) (not possible because R don't recognise OR as operator) if(pvalue>0.05 || pvalue<0.1) (this one is not good enough, cause it fulfills one condition pvalue>0.05 but it doesn't get other contidion pvalue<0.1) does anyone know a way to stablish a inteval as an statement for my if loop. Thanks in advance Lucas _________________________________________________________________ [[alternative HTML version deleted]]
2007 Oct 08
2
inbound call voip providers
Hello: I want to have a local telephone number that, when the people calls this number (via mobile or normal PSTN), the voip provider stablishes a SIP session to my asterisk box. It is possible? If yes... What providers have this service in Europe? It is difficult to configure and get things working ok? Will my asterisk box see the mobile or normal PSTN phone# that is calling the number (callerID)? Who many SIMULTANEOUS sessions (cal...
2006 Dec 19
2
Effect.Pulsate on last scriptaculous
Somone have tested the last scriptaculous version that ships with last prototype? I you make an Effect.Pulsate, the element stays hidden after the effect finish if the element don''t have opacity stablished. This is for the changes on the setStyle method on prototype. The original code is: ---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------...
2010 Sep 14
4
If then else with command for
Hey listers, I am trying to do something simple... Check the program below... I would like to create a variable named COLOR according to the conditions that I stablished... But the problem is that it seems that my variable COLOR is checking just on sample, may be last in the loop... Certainly, I am missing something... Thanks in advance, Marcio x<-c(288,139,196,159,536,134,623,517,96,467,277,155,386,241,422,6263,612,532,250,412,339,55,290,249,164,97,74,144,12...
2003 Oct 14
3
H.323 - SIP gateway
Hi all! Please I need someone that have already done an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia
2002 Apr 22
1
regapi with global registry
...ry to the global configuration (/etc/wine.*reg) If it isn't really possible to write to the global wine registry using standard wine tools this is really a drawback in wine which would be needed for a wine 0.9 or 1.0 release. That would allow multi-user wine environments to be easier stablished. Regards, Raul Dias
2015 Apr 24
4
I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
Hi Guenther, Thanks for ur reply I have concern from long time I'm not able to login through softphone with AWS Cloud.Please let me know is there any document or guide line for the same. Regards Akhilesh On Fri, Apr 24, 2015 at 1:26 PM, Guenther Boelter <gboelter at gmail.com> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 04/24/2015 03:34 PM, akhilesh
2015 Apr 27
1
I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
Hi Helvio, Could you tell me what is process to setup an environment for IAX. Regards Akhilesh On Fri, Apr 24, 2015 at 4:25 PM, Helvio Junior <helvio.listas at gmail.com> wrote: > Hi Akhilesh, > > SIP protocol use port 5060 (default) and many other ports to stablish > calls. You need to check if there is AWS firewall rule that allow your > communication from your client external IP and your AWS host. > > Also, think in use IAX intead of SIP, because SIP protocol has many > trouble when used with NAT, also IAX protocol use only one port (4569) to...
2015 Apr 25
1
I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
...e register message to the server. For me it seems like the softphone is not able to reach the server ! Best Regards, On Fri, Apr 24, 2015 at 10:55 AM, Helvio Junior <helvio.listas at gmail.com> wrote: > Hi Akhilesh, > > SIP protocol use port 5060 (default) and many other ports to stablish > calls. You need to check if there is AWS firewall rule that allow your > communication from your client external IP and your AWS host. > > Also, think in use IAX intead of SIP, because SIP protocol has many > trouble when used with NAT, also IAX protocol use only one port (4569) to...
2006 Jan 05
5
multiple links and nat
Hi, this might be a dumb question, but I''m not finding much information online. I''m trying to setup a 2.6 linux box to run nat across multiple upstream links as a simple way to aggregate bandwidth. I found the instructions in lartc section 4.2 (http://lartc.org/howto/lartc.rpdb.multiple-links.html) fairly clear and straightforward. I implemented those, and a couple of trivial
2015 Apr 24
0
I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
Hi Akhilesh, SIP protocol use port 5060 (default) and many other ports to stablish calls. You need to check if there is AWS firewall rule that allow your communication from your client external IP and your AWS host. Also, think in use IAX intead of SIP, because SIP protocol has many trouble when used with NAT, also IAX protocol use only one port (4569) to everything. When i...
2016 Feb 22
0
Remote Desktop slow startup in Windows 10 / 8
Hi, I'm having a slow startup (30") of Remote Desktop Client (mstsc) in windows 8 and windows 10. I've tried it in Windows 7 and it starts correctly (3-5"). Once connection is stablished it goes smoothly. I've tried connecting Remote Desktop with an IP and I've got same behaviour. When I'm starting Remote Desktop from Windows 10 or 8 I get next message in samba AD 4.1.17 contiously: Warning: 60 extra bytes in incoming RPC request [2016/02/22 18:43:33.373483, 3]...
2006 Nov 13
0
[Bug 1262] ssh disconnect message from master control is confusing
...: trivial Priority: P2 Component: ssh AssignedTo: bitbucket at mindrot.org ReportedBy: alexandre.nunes at gmail.com This may seem trivial, but when I disconnect from any host for which I had a previous connection, and the newer connection is using the previously stablished master control, ssh outputs this at disconnection: Connection to master closed. The thing is that I happen to have a server called master. So that looks quite ambiguous. I suggest changing that to the real server's name (as in the control socket name) or to something like "Connection...
2008 May 23
0
Asterisk chan Skype
Hello! Iam configuring chan Skype on my asterisk box, doing some test calls I saw that asterisk answer the calls but hungs up before the call are stablished. Is this a license problem? Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080523/68ceea02/attachment....
2010 Jun 30
1
Samba 3.3 ldap tools
I am currently trying to setup my Samba server to act as a samba PDC, with ldap as a backend. I have a selfsigned CA, that has signed the certificates to my ldap server. Starting my smbd, i keep getting the message smb_ldap_setup_connection: ldap://ldap.streetgeek.lan/ Failed to issue the StartTLS instruction: Connect error Connection to LDAP server failed for the 1 try! smbldap_open: already