search for: gafachi

...I would classify myself as "novice," and there is probably really nothing so trivial that I couldn't possibly have screwed it up. :-) I'm trying to use the AMD command to detect answering machines, and have tested it with no luck. This is what I get: > Channel SIP/gafachi-081c81a8 was answered. -- Executing [15155515509 at robocop2:1] Set("SIP/gafachi-081c81a8", "CALLERID(number)=6666666666") in new stack -- Executing [1515555509 at robocop2:2] Set("SIP/gafachi-081c81a8", "CALLERID(name)=Robocop") in new stack --...

Hi, folks. I'm having a heck of a time trying to get outgoing T38 faxing (I don't need inbound right now) working with FFA and Gafachi. G711 faxing works (as well as can be expected over the internet), but I want the higher reliability of T38. I'm running Asterisk 10-beta1. When I drop my callfile in to make the call, I get this: -- Attempting call on SIP/18884732963 at gafachi1a for application SendFAX(/srv/httpd/...

Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere <http://www.iconnecthere.com> Vonage <http://www.vonage.com> Teliax

...s doing something similar or sees a blatant problem with it. We're currently rolling out SPA-942 phones for the standard desk phone with vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls for satellite offices are handled by VoIP providers (for voice Vitelity inbound, Gafachi outbound). These satellite offices are using a T.38 fax DID from Gafachi, passed through the Asterisk server to a Linksys 3102 ATA and then to a POTS fax machine. This all works well thus far. Our HQ has a full voice PRI, terminated on the Asterisk server with a TE120P. Additionally, right now the...

...ven when I grab the calls. I experience this with Asterisk 1.6.0.1 and Asterisk 1.4.22. Here is information on how I do the call: ----------------------------------------------------------------- .call file contents: ----------------------------------------------------------------- Channel: SIP/GAFACHI/18183455555 CallerID: 18183455512 MaxRetries: 0 RetryTime: 60 WaitTime: 30 Context: outboundmessage1 Extension: s Priority: 1 Set: PassedInfo=18183453041-m1d ----------------------------------------------------------------- extensions.conf for outboundmessage1 context: ----------------------------...

...in is: [ext-fax-voicenation] exten => s,1,Noop(Receiving Fax for: ${FROM_DID} From: ${CALLERID(all)}) exten => s,n(receivefax),StopPlaytones exten => s,n,Set(FAX_FILE_NAME=/var/www/html/vncake/fax_temp/${FROM_DID}-${CALLERID(number)}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID}.tif) ; Gafachi is known to have a broken ecm implementation - disable on receive - also send with 'z' option exten => s,n,Set(trunk_name=${CUT(CHANNEL,-,1)}) exten => s,n,Noop(trunk name is ${trunk_name:4}) exten => s,n,ExecIf($[ "${trunk_name:4:7}" = "gafachi"]?Set(FAXOPT(ecm...

...tly what it should do (plays announcement and then records, the second time when i call back (within about a minute), it goes directly to a beep (for recording), no announcement. Another thing, during this time when I call 0 (my voicemail access number) it gives me a fast busy. any help. -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 583 bytes Desc: not availabl...

once u press 9 is there a way to make it so it restores dial tone, like most pbx's do? so dial tone , 9, dialtone, then ur local num -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 314 bytes Desc: not availabl...

Hi, We are using MixMonitor to record the call. When the call is bridged, the latency is significant. We tried to increase the internet speed and the server RAM and processor speed and still we are having that issue. We use VoiceTrading and Gafachi's Termination minutes to make calls. As we are in US and VoiceTrading in Europe, somebody suggested to move the termination minute provider to within USA. So, we bought the minutes from Gafachi. Still we are having the call latency issues. $ConversationFile = $ConversationPath."conv_&...

Hi everyone, I've got my * system up and running and I'm really pleased. I've gone with G.711 (alaw) and I've stumbled across a problem; when people place calls internally some people think they have been cut off if the line is quiet for a few seconds. Is there a way of getting comfort noise on the call? I'm using the STABLE release and cisco 7960 phones under FC-1 Cheers

Hi All, I have been reading through the list quite a bit, and I am going to post this more as a poll than anything else. I am working on setting up a very small business with something that resembles a professional voice system. My idea is to use Asterisk with a SIP provider and SIP clients. I currently have a Vonage account already. So adding the 9.99 a month Soft Phone would be easy.

I just signed up with livevoip for 800 DID and have very choppy audio. From PSTN to my asterisk is ok but asterisk to PSTN is terrible. I am using IAX and was assigned to server iax01.nyc.*. I do not believe it is a bandwidth problem on my end and I have no problems using iax with gafachi. I opened a ticket with livevoip but no response yet. Would I be better off using sip with them? Is there a server with better response/bandwidth? I admit that I am running a cvs head may 2004 prior to 1.x.x release. Could that be the problem? Regards, John

anyone know a company that will terminate did 831/408 area codes in california. FYI i already checked voicepulse, negative. -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 306 bytes Desc: not availabl...

Folks, Randomly, when the phone is taken off-hook, the the Iaxy produces a irritating banshee scream as opposed to a dial-tone. Cycling the power fixes the issue, & sometimes it magically goes away by itself. Has anyone experienced this issue & potentially fixed it? I'm using asterisk CVS head as of jun 17 2004. Thanks, Glen

Hey there, I don't know who else has suffered broadvoices terrible service, but I am about to my end with them. The lack of a LBR codec, the outages, the changing of servers without notifying subscribers haspushed me to my end. Now most incoming calls are abbruptly cut off within a minute of the call starting. Anyone know of any other * friendly providers that have DID, besides Voicepulse,

...u. Do I set it in the sip.conf file? I have also ordered 2 licenses from Digium. My understanding is that because this Cisco phone can handle the encoding, * just passes it thru. Is this correct? Also, I am using LiveVoip for my call termination via IAX (very happy with them, very unhappy with Gafachi). I cannot find any information from LiveVoip that indicates whether they accept G729. Is it likely or is it just dependant upon the provider? My interest is to improve voice quality over DSL and/or Cable Modem connections. I have QoS working (Sveasoft), and it has improved the situation, but t...

Hello list, I read on voip-info.org that Asterisk 1.4 support T38 passthrough. So I guess this means that I can have a Grandstream HT503 with T38 support and an analogue faxmachine on the other side of my Asterisk and a T38-account with a ITSP on the other side of my Asterisk machine, right ?! The fax coming from the faxmachine passes the HT503 to my Asterisk and my Asterisk sends the fax to

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

Hi - I am looking for suggestions for ITSPs for faxing with asterisk 1.8. We are based in the US, so would need an ITSP with US DIDs. #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to receive faxes via T.38. Sending faxes is not a requirement. Does anyone have a working asterisk 1.8.4 configuration and ITSP provider that they can recommend? We have been trying T.38

Asterisk People, I'm currently using Asterisk and with a SIP voip provider and I'm having problems where DTMF input in my IVR app is getting corrupted intermittently. For example, if someone enters 1025, it may come though correctly as 1025, or it may come trough as 10025, or 100255. DTMF digits will just double up. This doesn't happen all the time. Asterisk will just pick times