[Asterisk-Users] Best VoIP provider for Asterisk
Well, is very, very sad to see that every time, we start saying who is the best and more reliable, that company automatically start going down hill.. I used to love T**iax. But lately..they are not the same as last yearI cant call them reliable as they used to be. I used to say that they were excellent!...and they still pretty good.but sometimes that is not good enough Good luck in your search Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy Sent: Tuesday, May 23, 2006 8:56 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Best VoIP provider for Asterisk Hi Friends, Can you please tell me who is the best VoIP Service Provider using Asterisk (With trail version for sometime) . Waiting for your quick response. Thank you. Regards, Chandra. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Now that Nufone is dead...
I never liked Jeremy, having that out of the way, :) What happen to him can happen to ANYONE! It happened to Broadvoice big time Also Vonage!!.. but they are more prepared to deal with the root cause!! They have more resources!! And more MONEY!!! It has nothing to do with reputation!!! Don't spit out without facts!!! Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, May 23, 2006 8:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Now that Nufone is dead... I think the point everyone is making is that no reputable company would have had this happen. Can you see Vonage losing all their DIDs? No! NuFone clearly did something that screwed their contract with their CLEC... On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Here's one for all the naysayers: I only sent an email to NuFone accounting to inquire about that $2.50/month fee and they're falling over themselves to not only get all my questions answered but to also helping me getting my account set up in the most economical way for me after their upstream provider problems. Proves me right for sticking with them. jens On 23 May 2006, at 21:41, Jens Vagelpohl wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I see it now on the FAQ, but this must be a new thing. I paid $50 in December 2004 and still have over $39 (yes, I don't use it often). If I remember correctly the 800 DIDs were advertised as free of monthly fees, call fees only. jens On 23 May 2006, at 20:13, Tom Vile wrote: $2.50 p/month for 800 DID. On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 They bill you for having the 800 number? I thought they only did that for Michigan DIDs. They only bill my actual call time. jens On 23 May 2006, at 16:54, Tom Vile wrote: Then you are a luck one aren't you. Haven't had my 800 number for over a month now but they still bill you for having the number. Interesting. On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 23 May 2006, at 15:48, Carlos Chavez wrote: Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? Nufone is not dead, works perfectly fine for me. jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 through Shorewall rpoblem
Real Funny, you stated in one of your previous post that you do this ALL the time -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Wednesday, February 22, 2006 10:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAX2 through Shorewall rpoblem I am trying to put a Shorewall firewall in front of my PBX, all the other port forwards work fine but forwarding port 4569 to the PBX is not working, it is being logged as rejected even though there is a DNAT rule in shorewall. Anyone seen this and have a solution? -- Chris Mason ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Firewall/Embeded System/CF/etc
I am trying to build an silent non moving parts (fans,HD.etc) embedded system...Firewall/Asterisk/FXo/FXs/CF/etc Looking for anyone running asterisk with Coyote, IPcop, m0n0wal, Shorewall, etc in the same system/box!!! Offlist please... Thanks in advance!! Manny ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone doing NAT through m0n0Wall?
Mark, we work on a few of NAT to NAT issues and resolved them by using the new version 1.2.1 and externhost= No sure how you got externip= to do FQN because we were not able to get it to work... Please..Can you let me know, how you got it to work? that way I can avoid upgrading couple of my clients in a production environments... TIA,,, Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Thursday, December 22, 2005 7:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Anyone doing NAT through m0n0Wall? Hi Folks, I've just built myself a m0n0Wall based around a WRAP board and whilst it work really well for everything else I'm having some issues with Asterisk's NAT abilities. Here's my setup, A bunch of hardphones (various types) littered around the house. SPA-3000 handles the house POTS line which forwards to extention 2005. X-Ten Pro on my laptop for when I'm out and about. Grandstream BT-101 at my dad's house via our cable modems. Until replacing the Linksys with the m0n0Wall everything was working fine and dandy. I have externip=g7ltt.dyndns.org set in my sip.conf file. Without it I could not make my dad's phone work. With the m0n0Wall in place and the externip setting set I can make no calls internally but all the external phones work just fine. The reverse is true when I remove the externip setting; the internal phones work but the external ones don't. I've done some tracing with both firewalls and have noted the following; Linksys: externip set all SIP and IAX2 frames from * have my public address as the reply-to regardless of the NAT requirement of the phone in use. In other words it offers up the external address for internal calls. All data flows through the Linksys when addressed to the public IP address and is then forwarded back to the * server. m0n0Wall: externip set as above and the firewall drops the packets. externip not set and the * NAT doesn't work. I know that the m0n0Wall requires a rule to be added to make it work as before but what I don't understand is why is Asterisk forcing all calls to use its public IP address when externip is set? Surely this doubles network traffic; one packet goes to the router. another goes from the router to the internal host. Why doesn't go directly over the LAN for internal stuff? I had assumed that the addition of a nat=yes statement in the relevant phone stanza would turn on or off the NAT reqirement for that phone device but this doesn't seem to be the case. Any ideas would be greatly appreciated. Mark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone doing NAT through m0n0Wall?
I was told that externhost= only apply to 1.2.1 what version are you using? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francis Ballares (VoIPware.ca) Sent: Thursday, December 22, 2005 10:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone doing NAT through m0n0Wall? Hi Colin, You should use externhost=yourhost.somethingddns.com and you should put the local network parameter in your sip.conf. This will identify that your local lan doesn't need to use the externhost parameter when you try to connect internally- and asterisk should just work fine. regards, Francis ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk, Small Business, and Teliax
If you need any free advice, let me know.. ;) Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Berman Sent: Friday, December 09, 2005 7:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk, Small Business, and Teliax I'm a beginner here and am interested in Teliax. I own a small business and was wondering if you guys could help me out here. I'm basically looking for 6-8 telephone lines, but I notice that Teliax supports 4 simultaneous calls on their Corporate plan. So could I get two Corporate plans and set Asterisk to use both of them and then have, in essence, 8 people talking at the same time? If someone tries to call, would the phone ring busy or would it still go through? I plan on having a T1. Thanks for any help, Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
Well, as the user stated on the original message, the asterisk server is behind a NAT and the client is also behind a NAT. if you make it work just by opening ports, let me know..I have never been able to get it to work, thats why I dont use sip, just plain iax2 for everything J Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bharath Sent: Wednesday, November 23, 2005 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain Thanks Michael, I think thats is the problem, I have opened only ports 5060-5082, I need to open 1-2 as well. I will try that and post the result when i get back home. Thanks On 11/23/05, Michael West [EMAIL PROTECTED] wrote: I'm pasting something from another user on this list from 14/11/05 I would recommend that you do a little research on google, voip- info.org, and the list archives. To connect to an Asterisk box that sits behind NAT, you need to forward ports 5060 and 1-2 too the asterisk box, and you need to configure the externip, localnet, and nat variables in sip.conf. audio problems are almost always due to the RTP stream (ports 1-2) not being forwarded properly, either due to the port forwarding setup or the sip.conf settings. Tom -- Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Bharath Khambadkone Sent: Wednesday, November 23, 2005 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain By default AMP had NAT=yes in sip.conf, I read in some posts to change it to one, i was just trying my luck if that works. I have tried NAT=yes, The Phone gets registered, I can also make recieve calls but as soon as the call is picked I dont hear anything at both ends. Does this have anything to do with codecs? Thanks On 11/22/05, C F [EMAIL PROTECTED] wrote: On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote: Hello All, I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution. My Asterisk is on a public domain, there is no NAT or firewall in front of If no nat then why do you have nat=1 in sip.conf? the asteris box. I have sucessfully connected iax2 softphones was able to recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2)with [EMAIL PROTECTED] installed. my Sip.conf : [2008] ;(Sipura2002) username=2008 type=friend secret=2008 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2008 [2009] ;X-Lite Soft Phone username=2009 type=friend secret=2009 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2009 Thanks in advance.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Asterisk and Broadvoice, no incoming voice
Price is about the only good thing... quality? Jajajajaj reliable? Jajajajja -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Schafer Sent: Monday, September 26, 2005 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice I'm relatively new to the whole VOIP game, here's what I want to do. I am using VOIPJet for all of the outbound calls on our AAH box. I have one landline that I would like to busy forward to an inbound VOIP number. Broadvoice was recommended to me for price and quality. Can anyone make a suggestion for a good VOIP Provider for my inbound requirement? The bulk of my inbound calls will come in on the land line, but I would also like the leverage the group/conference feature in AAH (8+ext) and an inbound SIP seems to be a good answer for having a couple of different people call in at once (three people call the SIP number). Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] qualify=yes
I was just wondering if I can leave qualify=yes set on all my trunks??? Originally I had it only setup in one. Is nice to see the ms reported back.. Is a bad idea? Is that hammering the other servers? Or mine? Thanks Manny A. Wise EL OBELISCO, Inc. www.calltheus.com TollFree: (800)230-0106 Tampa: (813)283-0265 Miami: (786)347-5725 NewYork State: (631)492-3212 Washington Sate: (360)469-0317 FAX: 1.512.597.1779 FWD: 68346 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sudenly unable to get incoming from Broadvoice
Well, some smarty pants lady at broadvoice, claim that the problem is in our end, well, I have taking my box out of the picture, I went to bv control panel and have forwarded the calls to my home phone number, she STILL insist that the problem is my asterisk box, the one I deleted the Broadvoice trunk.. ;) Maybe I should just leave the trunk deleted and don't fight it anymore... :( The real funny part is the if I call from teliax to my 10 digit number the call get forwarded to my home, NO problem.. Is only when the number is called from a real PSTN number that the person get fast busy, well fast busy today, yesterday was this number has been disconnected... Maybe next she is going to say bv will not work with asterisk box out of the picture...jajajajaja Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Friday, August 19, 2005 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sudenly unable to get incoming from Broadvoice Yes, I've restarted asterisk and even rebooted my machine. sip show registry shows pbx*CLI sip show registry HostUsername Refresh State sip.varphonex.com:5060 8281625105 Registered sip.broadvoice.com:5060 [EMAIL PROTECTED] 3495 Registered pbx*CLI I did the same on my friends machine and it show the same thing. Why is the refresh period so large and what can I do to shorten it? I've ruled out any ISP issues. I can receive calls on my other VoIP services just fine. Mark Tom Rymes wrote: Have you restarted Asterisk to see if that helps? What does 'sip show registry' show? Tom On Aug 19, 2005, at 10:42 AM, Mark Phillips wrote: So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; [EMAIL PROTECTED]:PASSWORD-GOES-HERE: [EMAIL PROTECTED]/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very host=sip.broadvoice.com fromuser=9738281625 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband disallow=all context=ext-local canreinvite=no authname=9738281625 allow=ulaw allow=g726 allow=g729 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Sudenly unable to get incoming from Broadvoice
Well, mine started to work for a while and down again..I give up ;( But the good news is.. I just got a DID from teliax... ;) The part that really bother me was the recording when the number was called two days ago the number you have call has been disconnected.. The fast busy, not going though, etc.. I can live with that, but saying this number has been disconnected and they tell me that is my box, really piss me off. oopsss I should not be saying that in public, sorry.. ;) Manny PS.. I guess I will have to change my signature now to poin to the teliax phone number, thanks god it was not business cards...jajajejejejjijijjiji -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Friday, August 19, 2005 4:17 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sudenly unable to get incoming from Broadvoice OK, now I know of 5 peeps that suddenly are having this problem. It has to be them right? Mark (in the rainy end of NNJ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Small Form Factory Machine
Well, I was using a dual intel cpu mid ATX tower with Rioworks motherboard, I was real happy with performance, but was bother by the size of the unit and the amount of heat that was putting out, I when with a small factor unit (shuttle)with 1gig ram and a 3200+ AMD cpu, and had all kind of troubles!, I fought that system for weeks, then I did my home work and since I don't like to re-invent the wheel, I when with a microATX aluminum case with a Intel cpu and I am real happy, I look after a very successful company that sell the asterisk system as a turnkey system and I look at what motherboard they were using, and bingo, it worked real well I am sure they spend lots of money trying to find out a good motherboard to put in the mentioned system, since they need to support it all over the world. Contact me off list if you want more details... Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Gamble Sent: Friday, August 12, 2005 3:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Small Form Factory Machine Chris Gamble wrote: I have 2 TDM04b cards currently running in an asterisk at home box that I am ready to replace with the CVS version of asterisk. What I am looking for is thoughts / recommendations. I want to move this to a small form factor ( shuttle ) machine and was wandering what expeience / advice there was for this? I have seen the incompatible motherboard list at digium ( and in fact I think my current machine is on the list ! ), but wanted to know what others are doing for small form factor tdm setups? Thanks, I'm curious, what shuttle model has 2 pci slots? I have a TE110P running on a shuttle SB61G2 2.8GHz P4 w/ 512 RAM and * 1.0.8. I haven't put it through its paces yet, but I will as soon as our remote office gets their server. I guess the usual pointers apply (i.e. don't share interrupts, etc.) and the wiki was informative when I first set mine up. Didn't even have to ask then and it has been functioning well since. I did forget, one of those ports is an AGP slot! You are a life saver! In the mean time, is there a good small form factor machine i can use for this? Something to put away on a shelf and forget it exists? Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] TDM400P FXO channel hookstate always Offhook outbound digits sent before provider dialtone
That's a good advice, BUT Remember we talking [EMAIL PROTECTED] here, it will get overwritten every time you do a configuration reload... That is what I did for my cellsocket, but guess what, I had to fix it every time I mess around with the system even adding an extension, it was s annoying, that now I have two systems running, one with [EMAIL PROTECTED] and another plain *... ;) Heck at least I learn to deal with both now.. jejejejeje Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes Sent: Friday, August 12, 2005 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM400P FXO channel hookstate always Offhook outbound digits sent before provider dialtone Open the [EMAIL PROTECTED] AMP interface, click on trunks, and click on the entry for your ZAP trunk. Then, put 'ww' in the Outbound Dial Prefix Tom On Aug 12, 2005, at 1:29 PM, Stephen Joyce wrote: I have an [EMAIL PROTECTED] 1.3 server (Asterisk 1.0.9) and recently TDM400P with (1) FXO card on port 4. Inbound calls are always but outbound calls fail 75% of the time with intercept messages dial tone provider that include we're sorry, your call did not go through, and we're sorry, when placing a local call it is now necessary to dial an area code and the 7-digit number. I have connected a test set in monitor mode to the phone line to to what's being sent out the line by the Zap channel and 10 digits are sent but the first digit is usually sent only as I hear the dial tone being drawn from the line, so it appears that it's sent before the provider is ready to receive it. I can't find any sort of setting that would allow me to manually configure a dialing delay on the line, suspect this would provide a band-aid. When looking at the Asterisk CLI, I see that the correct number is dialed by my dial plan. I am calling from SIP extension 1100 and 770-555-1234, which is a local 10-digit phone number. -- Executing Dial(SIP/1100-9adc, ZAP/g0/7705551234) in new stack -- Called g0/7705551234 -- Zap/4-1 answered SIP/1100-9adc The status of the channel is Offhook regardless of whether or not phone line is actually Offhook or completely idle. I'm assuming that when the line seems idle, it should show as Onhook. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Asterisk@home backup/restore question
@home do that for you everyday...;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili Sent: Sunday, May 15, 2005 2:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] [EMAIL PROTECTED] backup/restore question Hello, How do I routinely backup all necessary configuration files on [EMAIL PROTECTED] Is there any procedure/tool/script for it? And if I need to move * with existing configuration on a new hardware, what is the best way to do it? Thanks I.N. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] voipjet anyone?
I am not an expert yet ;)...but VoipJet is very picky.. in your exten =
when I tried with the default it didn't work, when tried as they ask in the
FAQ's it workedyou must keep the exact format with the account
number...
[Voipjet]
exten = _1NXXNXX,1,SetCallerID(4153574000); Set your CallerID as a ten
;it number like this. See our FAQ
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} ; VoipJet.com
NANPA
exten = _011.,1,SetCallerID(4153574000); Set your CallerID as a ten digit
number like this. See our FAQ.
exten = _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} ; VoipJet.com WORLD
;Do not change IAX2/1234 in the above two lines!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JD
Sent: Friday, May 13, 2005 1:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] voipjet anyone?
Is it me... or is it voipjet?
This week I've been trying various providers, just can't seem to get
voipjet to work.
I signed up with voipjet but so far can't get it to work inbound or out
bound.
I always get 'all circuits busy'.
May 12 22:27:05 VERBOSE[2442]: -- Executing
[1;36;40mDial[0;37;40m([1;35;40mSIP/101-ad89[0;37;40m,
[1;35;40mIAX2/voipjet/4803442640[0;37;40m) in new stack
May 12 22:27:05 VERBOSE[2442]: -- Called voipjet/4803442640
May 12 22:27:05 WARNING[2442]: Call rejected by 66.246.220.19: No such
context/extension
May 12 22:27:05 DEBUG[2442]: Immediately destroying 6, having received
reject
Outbound settings:
notransfer=yes
auth=md5
context=from-pstn
host= 66.246.220.19
secret= md5hashstring
type=friend ; also tried peer and user
username=1234
Im using [EMAIL PROTECTED], but that shouldnt matter; people have this
working or is it me?
JD
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[Asterisk-Users] cellsocket problem
This is what I getafter Zap/4-1 answer I can press # and
the call go thru just fine..I just can find a way to force the # go in
automaticly @ end... :-( any ideas?
===Connected to
Asterisk 1.0.7 currently running on pbx (pid = 1089)
Verbosity is at least 3
-- Remote UNIX connection
-- Executing Macro(SIP/2007-a956,
dialout-trunk|4|2831234) in new stack
-- Executing GotoIf(SIP/2007-a956, 0?4) in new stack
-- Executing SetCallerID(SIP/2007-a956, 2007) in new
stack
-- Executing Goto(SIP/2007-a956, 6) in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing SetGroup(SIP/2007-a956, OUT_4) in new
stack
-- Executing CheckGroup(SIP/2007-a956, 1) in new
stack
-- Executing SetVar(SIP/2007-a956,
DIAL_NUMBER=2831234) in new stack
-- Executing SetVar(SIP/2007-a956, DIAL_TRUNK=4) in
new stack
-- Executing AGI(SIP/2007-a956, fixlocalprefix) in
new stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse
/etc/asterisk/localprefixes.conf
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Dial(SIP/2007-a956, ZAP/4/2831234) in
new stack
-- Called 4/2831234
-- Zap/4-1 answered SIP/2007-a956
-- Hungup 'Zap/4-1'
== Spawn extension (macro-dialout-trunk, s, 11) exited
non-zero on 'SIP/2007-a956' in macro 'dialout-trunk'
== Spawn extension (from-internal, 2831234, 1) exited
non-zero on 'SIP/2007-a956'
-- Executing Macro(SIP/2007-a956, hangupcall) in new
stack
-- Executing ResetCDR(SIP/2007-a956, w) in new stack
-- Executing NoCDR(SIP/2007-a956, ) in new stack
-- Executing Wait(SIP/2007-a956, 5) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited
non-zero on 'SIP/2007-a956' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero
on 'SIP/2007-a956'
pbx*CLI
[cellsocket-custom]
exten = _NXX,1,Dial(Zap/4/${EXTEN}#)
exten = _NXX,2,Macro(outisbusy) ; No available
circuits
also tried
[cellsocket-custom]
exten = _NXX,1,Dial(Zap/4/w${EXTEN}#)
exten = _NXX,2,Macro(outisbusy) ; No available
circuits
also tried
[cellsocket-custom]
SHARP=#
exten = _NXX,1,Dial(Zap/4/w${EXTEN}${SHARP})
exten = _NXX,2,Macro(outisbusy) ; No available
circuits
attachment: winmail.dat___
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[Asterisk-Users] Cellsocket with @home
I am posting this in case someone need
help.
=
YOU THA
MAN!
No sure how I will repay you, but
anything you need, just let me know!
Thank you, thank you, thank you
-- Executing GotoIf(SIP/2007-12c7,
0?4) in new stack
-- Executing SetCallerID(SIP/2007-12c7,
2007) in new stack
-- Executing Goto(SIP/2007-12c7,
6) in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing SetGroup(SIP/2007-12c7,
OUT_4) in new stack
-- Executing CheckGroup(SIP/2007-12c7,
1) in new stack
-- Executing SetVar(SIP/2007-12c7,
DIAL_NUMBER=2831234#) in new stack
-- Executing SetVar(SIP/2007-12c7,
DIAL_TRUNK=4) in new stack
-- Executing AGI(SIP/2007-12c7,
fixlocalprefix) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
-- AGI Script fixlocalprefix completed,
returning 0
-- Executing Dial(SIP/2007-12c7,
ZAP/4/2831234#) in new stack
-- Called 4/2831234#
-- Zap/4-1 answered SIP/2007-12c7
-- Hungup 'Zap/4-1'
== Spawn extension (macro-dialout-trunk, s, 11)
exited non-zero on 'SIP/2007-12c7' in macro 'dialout-trunk'
== Spawn extension (from-internal, 2831234, 1)
exited non-zero on 'SIP/2007-12c7'
-- Executing Macro(SIP/2007-12c7,
hangupcall) in new stack
-- Executing ResetCDR(SIP/2007-12c7,
w) in new stack
-- Executing NoCDR(SIP/2007-12c7,
) in new stack
-- Executing Wait(SIP/2007-12c7,
5) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited
non-zero on 'SIP/2007-12c7' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited
non-zero on 'SIP/2007-12c7'
pbx*CLI
-Original Message-
From: Tha Man!!
Sent: Thursday, May
12, 2005 8:34 AM
To: Manny Wise
Subject: Re: [Asterisk-Users] Cellsocket
help needed
There is no real easy way (that I
know of) to add the # at the end without getting your hands a bit
dirty.
Go into AMP and click on
Maintenance.
Then click on phpMyAdmin.
That will open a new window with the
phpMyAdmin mysql interface.
On the left, click on the dropdown
box and select asterisk
Under the box it will show the
structure of the DB.
Click on extensions
Now on the right near the top, click
on Browse
This is where AMP keeps all your
extension info.
This will be the hardest part
because you are going to have to do the identification.
Typically, the dial commands are
kept near the bottom and start with outrt
You're going to want to find the
name of the outbound route that will be using the cellsocket.
Look for the entries for that route
that contain dialout-trunk, those will be the ones you want to
edit.
Click on the little pencil icon for
that line.
Go to the dialout-trunk
line and add a # on to the end
example:
dialout-trunk,1,${EXTEN:}#
Then click on go at the bottom
to say the changes.
This will save the change and take
you back to the previous list.
Make sure you have all the dialout-trunk's
in the list for that outrt modified.
Once done, close the phpMyAdmin
window.
Now go back to AMP. We need to force
it to regen its configs.
Go under setup.
Go under extensions.
Click on any extension on the right.
Don't change anything and just click
on Submit Changes
This will pop up the red bar at the
top.
Click on it to apply changes.
After all this, you should be good to
go.
Good luck!!
- Original Message -
From: Manny Wise
To: The Best!!!
Sent: Wednesday, May
11, 2005 6:29 PM
Subject: Re:
[Asterisk-Users] Cellsocket help needed
I have made some progress, but still doesn't workI
created the trunk, * is dialing the ttrunk... I see from telenet asterisk -r
that the trunk pass the call to the cellsocketI see that it says Zap/4-1
answered.but to make it work I have to press the # in the cisco phone, then
I see the cellphone dialing, I followed the instruction from the other post
and have tried several convinations and still don't put the # at end, Do you
have some samples?? thanks again!!
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[Asterisk-Users] dial * get dial tone, push pin dial out......
I need some help...using @home v1.0 with TDM11B and 4 trunks..everything working fine internally I need to be able to dial into * , get dial tone and pust a code (PIN) or whatever 4 digits and be able to call out, what I need?? Samples exten.conf please? THANKS Manny ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 7960 firmware
I will appreciate if you can share it with me too, I was ready to buy the cd from ebay...and BTW is not legal...but only according to cisco ;) Tae care.. Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Laroff Sent: Monday, May 09, 2005 3:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] cisco 7960 firmware Don't know if it is legal, but if you email me off the list I have them. -Josh On 5/9/05 1:43 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cellsocket help needed
I need help from someone who has a working cellsocket, I have received couple email of people who wanted to help, but they just think they know how it supposed to work, but they dont have a working units, and they confused more..I need someone with a working solution to get my cellsocket going. Thanks!!! Write offlits @ mawise (AT) mail.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Cellsocket help needed
NO it doesn't cost more to call cell, but with my unlimited minutes plan, I can have an extra channel .and days like today that broadvoice and Voipjet were to max capacity and busy most of the time, I could have put it to a good use -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, May 08, 2005 1:43 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cellsocket help needed I don't see what the use for that is in the US. Since it doesn't really cost more money to call cell phones. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cellsocket NEED HELP
I just got a cellsocket for my * box...I need help, will give you a channel on my box in exchange for your time to help me out, if you have experience to configure this things, please contact me out list... Manny Mawise(at)hotmaildotcom Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channels ???
Can I send an receive call on the same channel (line to the wall) Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@Home bug
After installation of [EMAIL PROTECTED] v1, I have an annoying message in the screen, anyone know how to fix it INIT: Id s0 respawning too fast: disable for 5 minutes Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialing out...
I am very new to Asterisk, running [EMAIL PROTECTED] .06..I have a problem, everything on my box seems to be working ok, except: I have a local telco line hookup to my FXO port.. receive calls just finebut when I try to dial out, it ring the phone connected to the FXS port of the same card.not sure where to start looking to fix the problem, thanks in advance for your time Manny = Connected to Asterisk CVS-v1-0-04/23/05-12:26:39 currently running on asterisk1 (pid = 1078) Verbosity is at least 3 -- Remote UNIX connection -- Registered '1234' (AUTHENTICATED) at 192.168.1.10:4569 -- Executing Macro(SIP/-5a82, dialout|1|9317) in new stack -- Executing GotoIf(SIP/-5a82, 0?4) in new stack -- Executing SetCallerID(SIP/-5a82, 8132310123) in new stack -- Executing Goto(SIP/-5a82, 6) in new stack -- Goto (macro-dialout,s,6) -- Executing SetVar(SIP/-5a82, length=1) in new stack -- Executing Dial(SIP/-5a82, ZAP/g0/317) in new stack -- Called g0/317 -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Hungup 'Zap/1-1' == Spawn extension (macro-dialout, s, 7) exited non-zero on 'SIP/-5a82' in macro 'dialout' == Spawn extension (from-internal, 9317, 1) exited non-zero on 'SIP/-5a82' -- Executing Macro(SIP/-5a82, hangupcall) in new stack -- Executing ResetCDR(SIP/-5a82, w) in new stack -- Executing NoCDR(SIP/-5a82, ) in new stack -- Executing Wait(SIP/-5a82, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/-5a82' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/-5a82' asterisk1*CLI CONFIDENCIALIDAD. El contenido de esta comunicación, así como el de toda la documentación anexa, es confidencial y va dirigido únicamente al destinatario del mismo. En el supuesto de que usted no fuera el destinatario, le solicitamos que nos lo indique y no comunique su contenido a terceros, procediendo a su destrucción. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 7940 issue.
Anyone with experience on the Cisco 7940G, got the unit on ebay for a great price, but can figure out the password, tried the factory reset, can figura that either, any help will be appreciated, I trying to enter the network config to make it work with *... Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
