similar to: no audio on end-point when call is connected/bridged via PBX

I have a samba server configured at a clients office and sometimes when he is drilling down into directories in his "File Explorer" it stalls. I tail'ed the log files and I ran tcpdump, but I can't see anything that stands out. What should I look at? I am running Samba Version 2.2.3a-12.3 for Debian. brian -- Brian Lavender http://www.brie.com/brian/

Hi, is it possible to realize the following scenario? And if yes: how? ;) The current setup is as follows: We have a Samba 3 server on a linux machine as PDC and an OpenLDAP server as passdb backend (on the same host). All users and groups were inserted via the SMBLDAP tools by IDEALX. So far, so good. Everything runs fine. Now our plan is it to use another, external LDAP server for pure

Any reason why I don't get audio on the channel after it rings and the end user picks up. Here are my files. CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus [default] include => stdexten exten => s,1,Answer() exten => s,n,Wait(1) exten => s,n,Dial(SIP/callwithus/1111444444,120,A,(demo-thanks)) exten => s,n,Wait(2)

Dear Manager, We are using UPS from multiple vendor in distributed remote sites for different network devices, including Workstation which is running on MS Windows 10 Operation system. We are looking for a software that will be running on the local Workstation and able to shut the same machine depending the power condition of the UPS. Please note that UPS will be communicating to the software

Hello, When I run a "df -i" in my clients I get 95% indes used or 5% inodes free: Filesystem Inodes IUsed IFree IUse% Mounted on lustre-mds-01:lustre-mds-02:/cetafs 22200087 20949839 1250248 95% /mnt/data But if I run lfs df -i i get: UUID Inodes IUsed IFree I

Can anybody help me on configuring utstarcom F1000 on asterisk? Is there a way to do it or is it impossible? ___________________________________________________________________________ Este mensaje se dirige exclusivamente a su destinatario y puede contener información privilegiada o confidencial. Si no es vd. el destinatario indicado, queda notificado de que la utilización, divulgación y/o

Hi folks! Based upon your experience on the field what wifi sip phone would you reccomend ? A customer asked for a wireless * install and I'm looking for advice, tnx Alessio Focardi [[*] - Interconnessioni Italy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060628/05b2fb30/attachment.htm

Hello everybody, I was installed a lot of PDC under Linux Debian, here in my country, but i have a requirement from a client who wish to install a Primary Domain Controller with at least 30 Windows NT Backup domain controllers in all regions of his organization. I'll like to know if it is possible and where i find more information about a possible solution for this. Best Regards to all in

I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is

the legs separately as if they were not related to the same call. So the ingress leg negotiates ulaw, and despite it knowing that the peer also supports g729 fails the call since it's already decided on ulaw and the egress leg only accepts g729. If this is design intent I'm wondering if there is demand enough to justify a feature request? Any advice on how I can work around this issue?

I have just started working with Asterisk and have run into a road block concerning IAX and an inbound DID from callwithus.com. I am getting nowhere and I don't really know how to isolate the problem. The asterisk version is 1.2.7 on ubuntu, sits behind a firewall with iptables. I can connect and make a call to other internal extensions using zoiper and iax. When I try and use the number,

JID Pri S Owner Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host

I am trying to use a simple tool in the Dial plan so that if the first number does not connect the logic will go to the second and/or third. Basically, I want the call to ring and connect to the first number Then, if it is not answered I want another number to try to get connected Then, if second number does not answer I want the third to be tried i only list the scenario for the first two

Hi, What I need to know is how to change revert:true to revert:false from a draggable after I drop it on a droppable so it doesn''t return to its original place. Something like: [CODE] <!-- Draggable image --> <img alt="Product" id="item" src="icon.png"> <script type="text/javascript"> new

I am using Polycom 501s with asterisk 1.2.4. When transfering to call parking wih "#1" -> 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on. It seems like the polycom is hanging up before asterisk is able to

Running build 1.8.5.0 (compiled from source) I seem to be having an issue with codec negotiation. I have a Grandstream HT503 FXO port connected to a pstn line, a Polycom SP501, and a SIP trunk with callwithus. What I'm essentially looking to accomplish is for ulaw or g729 (preferably ulaw) to be used to the Grandstream FXO or any other internal endpoint, and for g729 only to be used outbound

Hello everyone, When looking for information about clustering of spatial data in R I was directed towards DBSCAN. I've read some docs about it and theb new questions have arisen. DBSCAN requires some parameters, one of them is "distance". As my data are three dimensional, longitude, latitude and temperature, which "distance" should I use? which dimension is related to

please help!!! I am dialing into our asterisk server(TDM400p) from the psnt. I hear our voicemail message and I press the extention 1000. The Polycom ip phone in the office rings. I pickup but neither side can hear one another. What have I done wrong? thanks sip.conf: [general] context=local-access ; Default context for incoming calls bindport=5060

Hi, I am setting up an asterisk server for testing purposes and cannot get voicemail to work at all. My host OS is Linux From Scratch 6.3 and the asterisk software versions I built are zaptel-1.4.5.1 and asterisk-1.4.12. I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk server and client phone are on different computers but are on the same LAN, i.e. no NAT. I have an

Dear User, I got the following error running a regression quantile: > rq1<-rq(dep ~ ., model=TRUE, data=exo, tau=0.5 ); > summary(rq1) Erro em rq.fit.fnb(x, y, tau = tau + h) : Error info = 75 in stepy: singular design Any hint about the problem? Thanks a lot, ________________________________________ Ricardo Gon?alves Silva, M. Sc. Apoio aos Processos de Modelagem Matem?tica