similar to: Is this true for Asterisk as SBC?

Hi, I have an NFS share on a NAS device which is mounted on my Samba server. I then share this using Samba. Client who connect are not able to obtain read-write locks. This setup works fine when the Samba version is 3.0.24 (Debian version 3.0.24-6etch9) but anything later and it fails. All systems are running Linux 2.6.18 (or later) in case anyone is wondering. Has anyone else seen the same

Hello, Currently running Kernel 2.4.31 with iptables 1.3.3 i-----------i i R1 i--------i i-----------i i i-----------i i-- i S1 i i-----------i i i-----------i i R2 i---------i

can we write multiple background in one feature file....? can we write multiple feature in one feature file.....? -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to rubyonrails-talk-/JYPxA39Uh5TLH3MbocFF+G/Ez6ZCGd0@public.gmane.org To unsubscribe from this group, send email to

Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM

Hi, This is a bit OT. If I have a phone behind NAT and the phone registers via an SBC can I set NAT=NO and canreinvite=yes in asterisk or will I still have issues ? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080313/41d639ee/attachment.htm

This is a really strange one, IMO. I just bought an SBC 900Mhz two-line portable phone. My intent was to use it with one of my TDM20 lines with asterisk, and have the other line be a PSTN line. But it appears as if for some reason I can't intermingle the two types of lines. If I hook *both* lines from the TDM20 to the unit it works just fine. If I hook *both* of my PSTN lines to the

Anyone out there have Verizon or SBC as the local exchange company in your area? I'm in the process of negotiations for additional interconnect agreements with these carriers (we're a wireless carrier). I'm looking for Asterisk folks who want a T1 (or several) for a PSTN gateway and would like to "network" their Asterisk boxes. This can take place at little or no cost

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress on we are having a few disconnects while calls are in session. I have talked both to some local phone contractors and SBC directly and

SBC installed a T1 ADTSe (Digital Trunking Service Enhanced) e&m wink start with 24 1 way trunks. The CO says they dial pulse DP the seven digit dnis number. The channels work now but take long time to answer and get these messages repeating until I guess the CO stops Pulse dialing the number. Nov 23 19:08:58 WARNING[1827865]: chan_zap.c:4718 ss_thread: getdtmf on channel 8: Operation now

Hi, I am new to the world of Asterisk PBX. I have been given the task of coming up with a solution to our office phone situation. From what I have been reading, Asterisk sounds like it could be ideal. However, most of the information I am finding is focused on the VoIP aspects, which is something we want to use for our remote employees, but the first hurdle is integrating Asterisk with the

Hi, I'm trying to get my * server connected to a softswitch through an SBC. I get the following error when * trys to register. Got SIP response 423 "Interval Too Brief" back from xxx.xxx.xxx.xxx Jan 20 12:43:54 NOTICE[2138]: chan_sip.c:5473 sip_reg_timeout: -- Registration for '1234561234@xxx.xxx.xxx.xxx ' timed out, trying again (Attempt #9) Is there something I can

Hi David, If cost is an issue then, yes you can do all that (the speex at least) with the eZDSP USB stick. Only limitation is a slow emulator that is integrated with USB, and so there is no JTAG output to use with other emulators. If you don't have an emulator anyway then this is an advantage. Cheers, M -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have been seeking streaming audio playing capability in a single board computer environment. I have not yet found any player hardware that exists for this environment. ie. a player card in a PMC format or other hardware that can be used in a VME bus application. Is anyone aware of any such hardware? Thanks. Jim <p>--------------------------------- Do you Yahoo!? Free Pop-Up Blocker -

I have what my SBC/ATT representative calls a "Supertrunk," but he can't tell me the specifics I need to know to configure Asterisk to work with it. By fiddling about, I've come up with the "almost working" configuration below. This works except that it takes about 4 seconds from when the console says the line is answered until it plays my prompt. Watching the

Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am looking for a hosted / virtual IPBX *PLATFORM* for service provider.Such hosted IPBX platform is aimed to be as a service, so that final customers don't have to install, maintain, or upgrade any PBX hardware or software. It should have a control panel for end users to create / edit extension, conference rooms , IVR menus, etc Could you please indicate companies that offer such

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to

Hi, I think that the web-driven SIP Phone (free) doddle (beta version) can be useful with your Asterisk applications. You can pre-fill it with your sip settings (Asterisk host name or IP /?realm / sip user), you just need to setup the HTML link as that: (Attached is the HTML page example) ? /**************************/ simple HTML code example: /*************************/ <html> <head>

Since I passed from version 1.0 to the 1.2.3. I have Pb with the callerid. If somebody call with presentation of the number all is well. If somebody make call in masked number, i couldn't send a callerid to the phone. It is in a call center and i use the callerid to present the name of the number called to the operator. Before that went. To identify the sda, I use the assignment of the