[Asterisk-Users] Cisco ata-186 port died
This might be the problem. I remember that i turned of the ringer (its an older style telephone with a switch on the back to switch to pulse, and turn the ringer off) so maybe the ATA had to much resistance and blew something. Anyone have expirience with this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ata-186 port died
I use both ports on my cisco ata-186. I run them using ulaw. Today I made numerous calls using my analog phone on port 2. I picked it up about an hour after the last call I made and the line was dead. There is no power at all over the phoneline to the phone, and the red light doesnt light up. The configuration is verified as unchanged. Has anyone seen this problem before. I was unsucessful in finding anything on google and wiki about it. jacob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] making * more like a normal pbx
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
Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
ya mine worked.
--
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
On Jun 14, 2004, at 7:31 AM, Charlie Hedlin wrote:
Wouldn't this keep the 9 since you didn't use {EXTEN}:1 to include the StripMSD portion?
I found this thread interesting, as it apears simpler than the dialplan I used:
exten => _9NXX,1,StripMSD,1
exten => _NXX,2,Prefix,1512
exten => _1512NXX,3,Dial(${TRUNK1}/${EXTEN})
exten => _1512NXX,4,Dial(${TRUNK2}/${EXTEN})
exten => _1512NXX,5,Congestion
usedcanon wrote:
looks fine to me
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter
Sent: 12 June 2004 12:48
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area
code for 7 digit dialing
Does this look right
exten => _9NXX,1,SetCallerID(831-XXX-)
exten => _9NXX,2,Dial(IAX2/[EMAIL PROTECTED]/1831${EXTEN})
exten => _9NXX,3,Congestion
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Re: [Asterisk-Users] making * more like a normal pbx
I searched for these things, however I don't know the proper terminology, so I come on here, people give me ideas, then i look on wiki. -- 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 On Jun 14, 2004, at 8:59 AM, Jay Milk wrote: You really need to start making friends with google and the wiki. This same question was asked just a few days before you discovered this mailing list. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob Hunter Sent: Monday, June 14, 2004 4:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] making * more like a normal pbx 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
Re: [Asterisk-Users] making * more like a normal pbx
no, i have no affiliation with them. I just think they have great service. j hunter [EMAIL PROTECTED] On Jun 14, 2004, at 10:48 AM, Steve Totaro wrote: I think he just wants to promote gafachi.com x-tad-bigger- Original Message -/x-tad-bigger x-tad-bigger /x-tad-biggerx-tad-biggerFrom:/x-tad-biggerx-tad-bigger /x-tad-biggerx-tad-biggerJay Milk/x-tad-biggerx-tad-bigger /x-tad-bigger x-tad-biggerTo:/x-tad-biggerx-tad-bigger /x-tad-biggerx-tad-bigger[EMAIL PROTECTED]/x-tad-biggerx-tad-bigger /x-tad-bigger x-tad-biggerSent:/x-tad-biggerx-tad-bigger Monday, June 14, 2004 11:59 AM/x-tad-bigger x-tad-biggerSubject:/x-tad-biggerx-tad-bigger RE: [Asterisk-Users] making * more like a normal pbx/x-tad-bigger You really need to start making friends with google and the wiki. This same question was asked just a few days before you discovered this mailing list. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob Hunter Sent: Monday, June 14, 2004 4:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] making * more like a normal pbx 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
Re: [Asterisk-Users] making * more like a normal pbx (cisco ata-186)
got it.. so one more question, this is on a cisco ata-186, (SIP) so it probably wont work. I have gone through the entire dialplan portion of the manual and can't find any function to make it function in this way. I am running firmware 2.16. j hunter [EMAIL PROTECTED] On Jun 14, 2004, at 12:26 PM, Andrew Thompson wrote: Jacob Hunter wrote: 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 Google asterisk ignorepat - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] telephones to use with asterix
People like the Cisco 7960. I personally used my cisco-ata with analog phones that have a screen. In my dream world I would have a cisco 7920 wifi phone. j hunter [EMAIL PROTECTED] On Jun 14, 2004, at 1:38 PM, Erick Perez wrote: Sirs, i just joined the mailing list and i have a question: What kind of phones can be used with asterix (phones with screen). Basically to see whos calling, display the time,etc...Just like normal phones with display screen do. Thanks, Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is nufone web site down?
i always like to use DNS servers from 2 ISP's different ISP's, thats helped me reduce problems like this. j hunter [EMAIL PROTECTED] On Jun 13, 2004, at 12:51 AM, Matt Darnell wrote: Must be my ISP. Thanks. -Matt - Original Message - From: Shaun Ewing [EMAIL PROTECTED]> To: [EMAIL PROTECTED]> Sent: Saturday, June 12, 2004 9:42 PM Subject: RE: [Asterisk-Users] Is nufone web site down? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Darnell Sent: Sunday, 13 June 2004 5:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Is nufone web site down? Can anyone get to www.nufone.net? Is their VoIP down? -Matt I don't know about their VoIP, but their site works for me. -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Local calls to x100p all else to iax term
Affirm! a way to externalize it (even if it looks just like extensions.conf) include file => prefixes.conf j hunter [EMAIL PROTECTED] On Jun 13, 2004, at 8:11 AM, Randy Bush wrote: I have a list of all my local prefixes(free) on my POTS. Is there a way to integrate that so * decides if it is going to use iax or POTS? There is about 60 prefixes.. 1831-XXX ; your first prefix will be 555 exten => _91831555,1,Dial(pstn...) ; your second prefix will be 678 exten => _91831678,1,Dial(pstn...) ; ok, no match from any of the previous 60 prefixes, so IAX exten => _91NX.,1,Dial(IAX...) Remember that you can combine prefixes using the exten regular expression: ; your first prefix will be 555, 556, 557, 558 or 559 exten => _9183155[5-9],1,Dial(pstn...) i think the point may be that there can be massive prefix lists and folk don't want the extensions.conf from hell. it would be a nice hack to have an external utility to load one's named [0] prefix lists into something that can be used for matching analogously to DBGet(). randy --- [0] - named so one can have more than one. i have out-dials in four locations each with a non-trivial set of prefixes. so i kinda wanna do exten => biwa.,1,Dial(SIP/[EMAIL PROTECTED],60,Ttr) exten => hawi.,2,Dial(SIP/[EMAIL PROTECTED],60,Ttr) exten => marais.,3,Dial(SIP/[EMAIL PROTECTED],60,Ttr) exten => tokyo.,4,Dial(SIP/[EMAIL PROTECTED],60,Ttr) exten => _.,5,Dial(SIP/[EMAIL PROTECTED],60,Ttr) and let some external process/cron/gui/... keep the four databases up to date. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is nufone web site down?
check out dslreports.com j hunter [EMAIL PROTECTED] On Jun 13, 2004, at 12:36 AM, Matt Darnell wrote: Can anyone get to www.nufone.net? Is their VoIP down? -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 831/408 iax termination
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
Re: [Asterisk-Users] 831/408 iax termination
i dont know ive emailed them asking j hunter [EMAIL PROTECTED] On Jun 13, 2004, at 4:20 PM, Aaron J. Angel wrote: Jacob Hunter wrote: anyone know a company that will terminate did 831/408 area codes in california. FYI i already checked voicepulse, negative. Why don't they terminate calls to those NPAs? This bothers me... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 831/408 iax termination
actually even better, can everyone list your iax termination (besides the obvious nufone) so that we can get to know other iax termination providers? -- 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 On Jun 13, 2004, at 4:52 PM, Jacob Hunter wrote: i dont know ive emailed them asking j hunter [EMAIL PROTECTED] On Jun 13, 2004, at 4:20 PM, Aaron J. Angel wrote: Jacob Hunter wrote: anyone know a company that will terminate did 831/408 area codes in california. FYI i already checked voicepulse, negative. Why don't they terminate calls to those NPAs? This bothers me... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange voicemail things
When I call an extension (say my extension 1000)and it goes directly to voicemail the first time, it does exactly 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
Re: [Asterisk-Users] Strange voicemail things
These is the diag -- Executing Dial(SIP/200-23ef, SIP/1000|20|Ttr) in new stack Jun 13 17:34:28 NOTICE[1209214400]: app_dial.c:674 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy at this time -- Executing VoiceMail(SIP/200-23ef, b83) in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/83/INBOX/msg format: wav, 0x810fdb0 -- User hung up j hunter [EMAIL PROTECTED] On Jun 13, 2004, at 5:30 PM, Jacob Hunter wrote: When I call an extension (say my extension 1000)and it goes directly to voicemail the first time, it does exactly 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
[Asterisk-Users] errors on startup
How can I fix these/or will they fix them selves when i install a x100p FYI: currently I am not running any digium cards (soon ill have a x100p). I get these errors. Jun 13 17:26:08 WARNING[1074399968]: res_musiconhold.c:523 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. Jun 13 17:26:08 WARNING[1074399968]: chan_iax2.c:6854 load_module: Unable to open IAX timing interface: No such device Jun 13 17:26:09 WARNING[1074399968]: chan_skinny.c:2566 reload_config: Unable to get our IP address, Skinny disabled -- 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
Re: [Asterisk-Users] Strange voicemail things
by the way it only fails when the softphone SIP/1000 is not turned on. j hunter [EMAIL PROTECTED] On Jun 13, 2004, at 5:36 PM, Jacob Hunter wrote: These is the diag -- Executing Dial(SIP/200-23ef, SIP/1000|20|Ttr) in new stack Jun 13 17:34:28 NOTICE[1209214400]: app_dial.c:674 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy at this time -- Executing VoiceMail(SIP/200-23ef, b83) in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/83/INBOX/msg format: wav, 0x810fdb0 -- User hung up j hunter [EMAIL PROTECTED] On Jun 13, 2004, at 5:30 PM, Jacob Hunter wrote: When I call an extension (say my extension 1000)and it goes directly to voicemail the first time, it does exactly 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
Re: [Asterisk-Users] Comfort Noise
make sure you have it set to transmit silence -- 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 On Jun 13, 2004, at 5:53 PM, Matt wrote: 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 Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comfort Noise
I'm not an expert at all, im a total newbie. I think what this is, is a feature to cut down on bandwidth use by not transmitting the silence. My cisco ata-186 has this feature, I turn it off. From what I have read they encourage you to because * can mis interpret it as a disconnect. -- 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 On Jun 13, 2004, at 5:53 PM, Matt wrote: 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 Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
How do you prepend. I want to be able to dial 7 digits instead of of 11 for local calls. Can someone post there extensions.conf part that is relavent? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 NuFone lines- which one to dial out on
I am setting up 2 nufone lines. I want to make them both availiable for dial-out. How do you syntax it in extensions.conf so that it figures out which one is avaliable and dials out on it. Also how do you setup the name part of callerid for the outgoing lines? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
im in US On Jun 12, 2004, at 1:14 AM, Jacob Hunter wrote: How do you prepend. I want to be able to dial 7 digits instead of of 11 for local calls. Can someone post there extensions.conf part that is relavent? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
Does this look right
exten = _9NXX,1,SetCallerID(831-XXX-)
exten = _9NXX,2,Dial(IAX2/[EMAIL PROTECTED]/1831${EXTEN})
exten = _9NXX,3,Congestion
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[Asterisk-Users] Cisco ATA-186 Firmware upgrade
I am currently running 2.16. Is there good reason to get the update to 3.1? Anything significant? Otherwise I am happy how it is, i just don't want to miss out on anything. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI on Cisco ATA-186 (SIP)
I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki [EMAIL PROTECTED] ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a mailbox=mailboxnumber. What do I need to do to get this to work? And which context am I putting there, is it the same context as the sip device, or is the context from the voicemail.conf. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MWI on Cisco ATA-186 (SIP)
the mail=1234 seems to have worked... is it necessary to do the [EMAIL PROTECTED] I dont think i set a contect (as there is only 2 mailboxes) so would it be default.. On Jun 12, 2004, at 6:40 AM, usedcanon wrote: It should go in sip.conf the context is whatever context you specified in voicemail.conf Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:31 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MWI on Cisco ATA-186 (SIP) I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki [EMAIL PROTECTED] ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a mailbox=mailboxnumber. What do I need to do to get this to work? And which context am I putting there, is it the same context as the sip device, or is the context from the voicemail.conf. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA-186 Firmware upgrade
im interested if there are any codec adds or major things like that... On Jun 12, 2004, at 6:44 AM, usedcanon wrote: There probably are a number of fixes. I have not used the ATA's for some time, however as the saying goes .. If it ain't broke don't fix it. So if it is working for you don't bother. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:12 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco ATA-186 Firmware upgrade I am currently running 2.16. Is there good reason to get the update to 3.1? Anything significant? Otherwise I am happy how it is, i just don't want to miss out on anything. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on
gafachi.com too.. they only have new york for did, BUT they have 2c a minute anywhere in the us... I have nufone inbound and gafachi out On Jun 12, 2004, at 12:15 PM, Reid A. Forrest wrote: FWIW, if anyone is interested the same goes for Voicepulse. I've been using it for multiple inbound and outbound calls for about a month. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, June 12, 2004 12:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on I have tried it with 4 simultaneous calls and it worked like a charm. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 12, 2004 8:41 AM Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on On Sat, 12 Jun 2004, Simon Dorfman wrote: Steve, Do you know if the same is true with inbound calls? Let's say I have an 800 number with nufone and I have 10 Snom phones hooked up to *. If 10 people call in 10 seconds, assuming I have * configured correctly, can all 10 Snom phone users pick up and answer the 10 calls? Thanks, Simon in New Orleans I believe that the answer is yes - though I haven't tried it personally. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on
If anyone signed up for gafachi because of this please email me On Jun 12, 2004, at 3:03 PM, Jacob Hunter wrote: gafachi.com too.. they only have new york for did, BUT they have 2c a minute anywhere in the us... I have nufone inbound and gafachi out On Jun 12, 2004, at 12:15 PM, Reid A. Forrest wrote: FWIW, if anyone is interested the same goes for Voicepulse. I've been using it for multiple inbound and outbound calls for about a month. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, June 12, 2004 12:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on I have tried it with 4 simultaneous calls and it worked like a charm. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 12, 2004 8:41 AM Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on On Sat, 12 Jun 2004, Simon Dorfman wrote: Steve, Do you know if the same is true with inbound calls? Let's say I have an 800 number with nufone and I have 10 Snom phones hooked up to *. If 10 people call in 10 seconds, assuming I have * configured correctly, can all 10 Snom phone users pick up and answer the 10 calls? Thanks, Simon in New Orleans I believe that the answer is yes - though I haven't tried it personally. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on
In the Asterisk chat on IRC... its very helpful... luck and chance! theyre great! On Jun 12, 2004, at 7:12 PM, Steve Totaro wrote: How did you find them? There is nothing in google except that they used to be a budget webhosting company and some sort of media business prior to that. http://web.archive.org/web/*/http://www.gafachi.com - Original Message - From: Jacob Hunter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 12, 2004 8:56 PM Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on Just so you all know gafachi.com has a very good referral program.. you get a good amount of minutes (10% of whatever your referrals purchased). So make sure to tell people u refer to put your username as their referral code. They are a great service provider, from what i see... great support in chat and AIM... so id support them and pass on the word. jacob On Jun 12, 2004, at 4:56 PM, Jacob Hunter wrote: If anyone signed up for gafachi because of this please email me On Jun 12, 2004, at 3:03 PM, Jacob Hunter wrote: gafachi.com too.. they only have new york for did, BUT they have 2c a minute anywhere in the us... I have nufone inbound and gafachi out On Jun 12, 2004, at 12:15 PM, Reid A. Forrest wrote: FWIW, if anyone is interested the same goes for Voicepulse. I've been using it for multiple inbound and outbound calls for about a month. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, June 12, 2004 12:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on I have tried it with 4 simultaneous calls and it worked like a charm. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 12, 2004 8:41 AM Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on On Sat, 12 Jun 2004, Simon Dorfman wrote: Steve, Do you know if the same is true with inbound calls? Let's say I have an 800 number with nufone and I have 10 Snom phones hooked up to *. If 10 people call in 10 seconds, assuming I have * configured correctly, can all 10 Snom phone users pick up and answer the 10 calls? Thanks, Simon in New Orleans I believe that the answer is yes - though I haven't tried it personally. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
I have a list of all my local (free) on my POTS prefixes. Is there a way to integrate that so * decides if it is going to use iax or POTS? There is about 60 prefixes.. 1831-XXX extension.conf clipping 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
[Asterisk-Users] Local calls to x100p all else to iax term
Sorry I screwed up. I wanted to write back with the proper subject. I have a list of all my local prefixes(free) on my POTS. Is there a way to integrate that so * decides if it is going to use iax or POTS? There is about 60 prefixes.. 1831-XXX extension.conf clipping 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
