similar to: OT - Packet 8 firmware

Does anyone know of an ATA which has a modifieable (open source) firmware AND that supports faxing. Granted most of them don't do faxing well but that is more of a problem with the network side. Michael Blood -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050831/8c13db6b/attachment.htm

Hi to all in a scenario where: - the bandwith is shared with other traffic (HTTP,VPN,ecc) - the PBX is on a remote VPN peer - due to many reasons Qos is not usable There is a IAX trunk between 2 Asterisk 1.4 i've tried different codecs (ulaw,alaw,gsm) but the main problem still remain the same: too many voice packet get lost. The main problem is surely on the network, but the strange thing

Each time I send a message to the ML I receive this message... (thi mislead me to double-post some days ago). Could someone please unsubscribe the "blocked" address? But I guess that's not possible, as anyone else shuold have noticed this, too... =( -- Lapo 'Raist' Luchini lapo@lapo.it (PGP & X.509 keys available) http://www.lapo.it (ICQ UIN: 529796) --------------

On Thu, Feb 2, 2017 at 7:13 PM, TE Dukes <tdukes at palmettoshopper.com> wrote: > Lately I have been getting slow and partial page loads, server not found, > server timed out, etc.. Get knocked off ssh when accessing my home server > from work, etc. Its not the work connection because I don't have problems > accessing other sites, just here at home and my home server. >

On 02/02/2017 10:12 PM, TE Dukes wrote: > >> -----Original Message----- >> From: CentOS [mailto:centos-bounces at centos.org] On Behalf Of Matt >> Garman >> Sent: Thursday, February 2, 2017 8:52 PM >> To: CentOS mailing list >> Subject: Re: [CentOS] Spotty internet connection >> >> On Thu, Feb 2, 2017 at 7:13 PM, TE Dukes <tdukes at

> -----Original Message----- > From: CentOS [mailto:centos-bounces at centos.org] On Behalf Of Matt > Garman > Sent: Thursday, February 2, 2017 8:52 PM > To: CentOS mailing list > Subject: Re: [CentOS] Spotty internet connection > > On Thu, Feb 2, 2017 at 7:13 PM, TE Dukes <tdukes at palmettoshopper.com> > wrote: > > Lately I have been getting slow and

> -----Original Message----- > From: CentOS [mailto:centos-bounces at centos.org] On Behalf Of Steve Clark > Sent: Friday, February 3, 2017 6:36 AM > To: CentOS mailing list > Subject: Re: [CentOS] Spotty internet connection > > On 02/02/2017 10:12 PM, TE Dukes wrote: > > > >> -----Original Message----- > >> From: CentOS [mailto:centos-bounces at

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 all! i have a dsl connection with 1280 kbps for downstream and 256 kbps for upstream, and i want to manage the bandwidth to give high priority to voip traffic and low priority to p2p traffic. i found the script jdg-qos. i readed on this forum (i have a dsl-g604t router with MCMCC firmware) http://www.dslreports.com/forum/remark,16250220 that the two parameters of the jdg-qos script (DWIFLIMIT

Hi, I'm following, http://www.voip-info.org/wiki-Asterisk+phone+sjphone. However, I cannot find the SIP tab. Would someone please give me a few pointers? The screen capture can be seen at URL below http://www.dslreports.com/forum/remark,12022987~mode=flat Regards, Norman Zhang

Has anyone tried getting Speakeasy VOIP to work with Asterisk? I just got Speakeasy DSL and am thinking of trying out their VOIP [1] with the hope that the quality/stability will be better than broadvoice. I searched in the usual places (voip-info.org, the asterisk users mailing list archives [2], dslreports.com) and couldn't find anything. Any other places I should look? Thanks, Simon

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

I found some old messages regarding a possible pkt8 DTA "bypass". Anyone is using Packet8 with Asterisk? ========== http://www.dslreports.com/forum/remark,7292324~root=voip~mode=flat Got Softphone Working with Packet8 Friendly name: {Anything you'd like.} SIP domain: packet8.net SIP proxy: packet8.net Leave everything else at the default. When you login, it will ask for a username

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

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

Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip