similar to: What would I need to do this?

Hi, We have Asterisk, running on a P4 box running Suse 9.1, making calls using IAX through SimpleTelecom and Nufone. What we are looking for is toll quality voice. The problem is that voice over calls routed through SimpleTelecom and nNufone occassionally breaks. We also have a digium card and the calls over the digium card using the Zaptel Interface have a very good quality. We

The scenario we have is fax send/recieve software that ONLY talks T38 and an asterisk box. We have ITSP providers that do NOT talk T38 but G711 only. Does asterisk have the capability to take the T38 call from an ATA or T38 software then bridge/transcode it and do G711 out to the PSTN providers? If not is there another product PAID or FREE software or hardware that can do this easily and

Às 14:46 de 27/07/2019, Ilia Mirkin escreveu: > On Sat, Jul 27, 2019 at 7:37 AM Ralph Corderoy <ralph at inputplus.co.uk> wrote: >> The video plays, CPU load is less (my aim), but there's ‘tearing’ of the >> picture as if small rectangles that are updates are appearing in the >> wrong location, off by a little. If I step through the frames with >> mpv's

Here is the setup: Phone A (in NYC) on own bandwidth. Phone B (in LA) on own bandwidth. Asterisk box in Houston,TX on own bandwidth. Both phones contact asterisk to register. Not much bandwidth used for this as it is a few packets every hour or so. Phone A calls Phone B. Phone A sends a call request to asterisk and asterisk calls phone B. Both phones are connected and both people are talking.

I stand corrected! However you do get my point ... The bigger the company the worse it is. Having to deal with these guys is a nightmare. The company that brings me out in spots is Rogers Cable (24/7). They have this electronic air-head called "Gertrude" or something, (an android) who can't understand the word "NO" and has trouble with "YES" (actually like my

Hi All I'm trying to set up a machine with a simple samba configuration. I've written the config file with no problems, however any time I attempt to connect to the shares it doesn't work. Firstly, the smb.conf: [global] netbios name = TEMPSRV workgroup = LIPSON map to guest = Bad User [ittechs] comment = Should not see me... path = /var/smb/ittechs read only = no force user =

hi, we've got (and getting more) serious problem with load. I'm just calculate that ther is about 300 mailbox with 20Gb mails. and the whole system is _very_ slow and just getting slower and slower. it's a fast lan every client has 100Mb connection to the server with gigabit crad and gigabit switches. when I push the delete button in a mail I usualy have to wait 10,20 or 30

Hello Michael, you are referring to the following behavior - did I get it correctly?: outbound broken: asterisk offers g722 / g711 to provider (callee), callee answers g711. Asterisk now transcodes between caller and callee (g722 <-> g711). inbound works: call from provider: g711 -> asterisk drops g722 and passes g711 to internal callee -> no transcoding. As far as I know,

hi i want samba(PDC) join client too samba but not use ldap domain join with winbind or net rpc it's usualy success join, but usualy failed log.client_machine is "failed to get machine password for account client_machine$: NT_STATUS_LOGON_FAILURE" same configure then why success why failed? how to samba PDC samba client domain join without ldap?

Just got started and stuck at the same time. Working with the C-Api, i have come to the point where i want to emit a binary file for my module. I queried the X86-Target and called LLVMCreateTargetMachine this way: > LTarget := LLVMGetTargetFromName('x86'); > LMachine := LLVMCreateTargetMachine(LTarget, 'i386-win32-msvc', nil, > nil, LLVMCodeGenLevelDefault,

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

Hello, I am doing some testing with my brother. We both have asterisk running with a Cisco 7960 locally and it works great. Using SIP between the asterisk boxes works great also. If I use IAX to call his remote extension, it fails because the remote asterisk server tries to use GSM to talk to the 7960. I end up going to his voicemail, which works fine. If he calls the same way it has the

Hello list, I have many IP hardphones like Siemens 300 basic ( old ) , cisco ata.. etc I need: G711 from old phones must be convert to G729 via asterisk and send to provider ( G729 from digium ) I have this problems: oh323 (last version): ------------- asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider (

I have a connect to * via H.323/g711 from device A and want to connect to B which want for H.323/g729 h323.conf contains disallow=all allow=alaw allow=g729 but outgoing faststart/TCS contains only g711 (from h323_request(format) i think) and so no codec negotiation and no voice. Howto run up g711/H323 -> * -> g729/H323 PS intel's g729 was used. ast 1.0.3-6 PPS stupid -

Anyone have a customer sending/receiving multi-page faxes over Verizon Business SIP trunk/g711 ? Verizon Business indicates they don't support it, and I have 2 recent customers that it doesn't work for, and 1 current large customer telling me he's going to make it work <grin>. The issues is the latency/jitter on fax/g711 over Verizon Business seems to spit out only 11

We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second

https://bugzilla.netfilter.org/show_bug.cgi?id=1080 Bug ID: 1080 Summary: call ^^18557^O92^847^^@@ Epson printer customer care phone number Product: bugzilla Version: other Hardware: x86_64 OS: All Status: NEW Severity: enhancement Priority: P5 Component: netfilter

https://bugzilla.mindrot.org/show_bug.cgi?id=2596 Bug ID: 2596 Summary: Brother printer (((1~855~709~2847))) Technical Support Phone Number Brother Printer Customer Care Phone Number Product: Portable OpenSSH Version: 4.0p1 Hardware: Other OS: Other Status: NEW

https://bugzilla.mindrot.org/show_bug.cgi?id=2597 Bug ID: 2597 Summary: Epson printer (((1~855~709~2847))) Technical Support Phone Number Epson Printer Customer Care Phone Number Product: Portable OpenSSH Version: 4.0p1 Hardware: Other OS: Other Status: NEW

Hi All, I want to configure fax with Asterisk and I found that we can do this reliably using G711 codec only. Currently my provider is supporting G729 and G711. During the call initiation the call starts with G729 (1'st priority) and somehow if the receiver is unable to receive call then we are providing the Caller to send a fax, but at that point they are using G729 codec. At this point how