Re: [asterisk-users] VOIP Provider
2008/10/2 Gregory Malsack [EMAIL PROTECTED] Hi All, Can anyone recommend a good VOIP provider in the Milwaukee/Chicago area? We need flat rate billing per line/trunk, trunking, did's, and iax or G.729 compatibility. Thanks, Greg No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.7.5/1703 - Release Date: 10/2/2008 7:46 AM Bandwidth.com is good, has flat rate, trunk, not sure about their stock of DIDs. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP Provider
Hi. We recommend Fonet Global, they work with Asterisk many years ago and provide sip termination, DIDs, etc. At 03:39 p.m. 02/10/2008, Steve Totaro wrote: 2008/10/2 Gregory Malsack mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Hi All, Can anyone recommend a good VOIP provider in the Milwaukee/Chicago area? We need flat rate billing per line/trunk, trunking, did's, and iax or G.729 compatibility. Thanks, Greg No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.7.5/1703 - Release Date: 10/2/2008 7:46 AM Bandwidth.com is good, has flat rate, trunk, not sure about their stock of DIDs. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP Provider wooes
Gregory Malsack wrote on 1/4/08 4:48 PM: Does anyone know of a good VOIP dialtone provider in the northern Chicago area. My client has tried Broadvoice and Mix and is having problems with latency in the middle of the traceroute between him and the provider. I use Broadvoice and haven't had any problems with them, and I'm in Southwest Michigan... oddly enough, I use broadvoice's New York node rather than their Chicago node, despite being a closer traceroute to Chicago from here (I go through Chicago on the way to anywhere on the net from here), because I had problems with their Chicago node dropping out. But I get pretty good connections with minimal latency from New York, despite sending the packets right past the Chicago one. :) Strange, but it works for me. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Provider for business
Good luck with that one. Most unlimited providers have limits. (even if they say unlimited) /b On Sep 19, 2007, at 12:32 AM, Jim Boykin wrote: Can someone suggests a good and resonable cost voip provider with business unlimited plan in USA and allows simultaneous outgoing calling. Thanks ~Jim ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Provider for business
Am Mittwoch, den 19.09.2007, 11:02 +0530 schrieb Jim Boykin: Can someone suggests a good and resonable cost voip provider with business unlimited plan in USA and allows simultaneous outgoing calling. My experience with business unlimited is that they very well know which customer uses more than his share of minutes. Providers that buy minutes in millions probably get good prizes, but they calculate for an average call volume. If you are far above profitability - and you seem to exactly plan that - you will not stay their customer for long. IMO you would better find a VoIP provider with good minute rates - if you can afford it, service level agreements, and good customer management. This might not even be more expensive in the long run, as cheap stays cheap: Problems with a cheapo provider will cost YOUR money. YMMV, of course, and quality can not be always expressed in numbers as easy as (call minute/price) quantity. BR Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Provider for business
How would we be able to determine the reasonable cost for an unlimited plan for an unspecified business? If the business was General Electric, I would bet they would consider $1M/month very reasonable for unlimited service. A plan for a corner shop might be reasonable at $19.95/month, typical for this type of service. Why don't you take a pay as you go plan and pay for what you use? -- Chris Mason -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip provider settings problem, please help
- Original Message - From: Jody Gugelhupf [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, August 27, 2007 3:55 PM Subject: [asterisk-users] voip provider settings problem, please help hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when picked up, but after a while it stops working, incoming calls for this provider are not shown anymore in the CLI, but from other providers it always works, but the phone is ringingn nevertheless when calling my skypho account...when i then turn off the pbx and restart after sumthing like 2 hours my skypho account is working fine again, the incmiong calls are shown in the asterisk CLI, but after, i don't know let's say an hour or so it again stops working, incoming calls for my skypho account can not be seen in the asterisk CLI, then if i turn off the pbx for an hour or so it works again, so i thought it must be a setting issue, maybe something with the register? althought it always shows it registered when i use 'sip show registry' someone has an idea what i have to set or do to have it working permanently? what could be the problem here? i got no clue whatsoever and i have been using asterisk only since half a year, please help me, i'm totaly desperate, thx in advance jody :) Smell's like a NAT issue. Are you behind NAT ? As some one else mentioned try to set qualify=yes as well as register often. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip provider settings problem, please help
hi Anselm :)
thx for your tip, though i have qualified turned on, anyhow here are my
complete sip.conf and
extensions.conf, thx for any help :)
sip.conf
[general]
allowoverlap = yes
realm = mydomain.tld
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
tos = lowdelay
disallow = all
allow = alaw,ulaw,gsm,ilbc,g729
trustrpid = no
dtmfmode = auto
externip = XXX.XXX.XXX.XXX
localnet = 10.0.0.0/255.255.0.0
nat = yes
canreinvite = yes
rtcachefriends = yes
fromdomain = sshn.net
qualify = yes
register = user:[EMAIL PROTECTED]
register = user:[EMAIL PROTECTED]
register = user:[EMAIL PROTECTED]
register = user:[EMAIL PROTECTED]
register = user:[EMAIL PROTECTED]
register = user:[EMAIL PROTECTED]
register = user:[EMAIL PROTECTED]
extension.conf:
[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
priorityjumping = no
[globals]
TRUNKOPTIONS =
EMERGENCY = 0
EMERGENCY_TRUNK =
TRANSFERS_CTX = DefaultOutgoingRule
CALLBACK_CTX = DefaultOutgoingRule
DISA_CTX = DefaultOutgoingRule
DISA_PASSWD =
DYNAMIC_FEATURES = automon
TRUNK =
SIP/3124XSIP/9083XXXSIP/069929SIP/webcalldirectDESIP/webcalldirectNLSIP/iXcallSIP/messagenet
OUTGOING_PREFIX =
[_all_]
include = _all-extensions_
include = _all-resources_
include = _all-applications_
include = _catch-all_
[_catch-all_]
exten = _X.,1,AGI(dial.php|entity=group=5extension=${EXTEN})
exten = _X,1,AGI(dial.php|entity=group=5extension=${EXTEN})
[app-AgentCallbackLogin_92]
exten = *100,1,AGI(dial.php|entity=1group=6extension=*100)
[_all-applications_]
include = app-AgentCallbackLogin_92
include = app-AgentCallbackLogout_94
include = app-audiorecorder
include = app-callermailbox
include = app-cancel-CFB-calling-extension
include = app-cancel-CFNR-calling-extension
include = app-cancel-CFU-calling-extension
include = app-CFB-calling-extension
include = app-CFNR-calling-extension
include = app-CFU-calling-extension
include = app-dnd-off
include = app-dnd-on
include = app-mailbox
[app-AgentCallbackLogout_94]
exten = *101,1,AGI(dial.php|entity=2group=6extension=*101)
[app-audiorecorder]
exten = *99,1,AGI(dial.php|entity=3group=6extension=*99)
[app-callermailbox]
exten = *98,1,AGI(dial.php|entity=4group=6extension=*98)
[app-cancel-CFB-calling-extension]
exten = *91,1,AGI(dial.php|entity=5group=6extension=*91)
[app-cancel-CFNR-calling-extension]
exten = *93,1,AGI(dial.php|entity=6group=6extension=*93)
[app-cancel-CFU-calling-extension]
exten = *73,1,AGI(dial.php|entity=7group=6extension=*73)
[app-CFB-calling-extension]
exten = *90,1,AGI(dial.php|entity=8group=6extension=*90)
[app-CFNR-calling-extension]
exten = *92,1,AGI(dial.php|entity=9group=6extension=*92)
[app-CFU-calling-extension]
exten = *72,1,AGI(dial.php|entity=10group=6extension=*72)
[app-dnd-off]
exten = *79,1,AGI(dial.php|entity=11group=6extension=*79)
[app-dnd-on]
exten = *78,1,AGI(dial.php|entity=12group=6extension=*78)
[app-mailbox]
exten = _*98.,1,AGI(dial.php|entity=13group=6extension=_*98.)
[macro-agentcallbacklogin]
exten = s,1,AgentCallbackLogin(${CALLERID(num)}||${CALLERID(num)[EMAIL
PROTECTED])
[macro-agentcallbacklogout]
exten = s,1,Answer
exten = s,n,System(asterisk -rx agent logoff Agent/${CALLERID(num)})
exten = s,n,Playback(agent-loggedoff)
exten = s,n,Playback(vm-goodbye)
[macro-audiorecorder]
exten = s,1,AGI(record.php)
[macro-reroute]
exten = s,1,Goto(${ARG2},${ARG1},1)
[macro-callback]
exten = s,1,Wait(2)
exten = s,n,AGI(agi-callback.agi,${CALLERID(num)},${ARG1},${ARG2},${ARG3})
[macro-cancel-CFB-calling-extension]
exten = s,1,DBdel(CFB/${CALLERID(num)})
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,Playback(call-fwd-on-busy)
exten = s,n,Playback(de-activated)
[macro-cancel-CFNR-calling-extension]
exten = s,1,DBdel(CFNR/${CALLERID(num)})
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,Playback(call-fwd-no-ans)
exten = s,n,Playback(de-activated)
[macro-cancel-CFU-calling-extension]
exten = s,1,DBdel(CFU/${CALLERID(num)})
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,Playback(call-fwd-cancelled)
[macro-CFB-calling-extension]
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,BackGround(ent-target-attendant)
exten = s,n,Read(toext,then-press-pound)
exten = s,n,Wait(1)
exten = s,n,Set(DB(CFB/${CALLERID(num)})=${toext})
exten = s,n,Playback(call-fwd-on-busy)
exten = s,n,Playback(for)
exten = s,n,Playback(extension)
exten = s,n,SayDigits(${CALLERID(num)})
exten = s,n,Playback(is-set-to)
exten = s,n,SayDigits(${toext})
[macro-CFNR-calling-extension]
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,BackGround(ent-target-attendant)
exten = s,n,Read(toext,then-press-pound)
exten = s,n,Wait(1)
exten = s,n,Set(DB(CFNR/${CALLERID(num)})=${toext})
exten = s,n,Playback(call-fwd-no-ans)
exten = s,n,Playback(for)
exten = s,n,Playback(extension)
exten = s,n,SayDigits(${CALLERID(num)})
exten = s,n,Playback(is-set-to)
exten = s,n,SayDigits(${toext})
[macro-CFU-calling-extension]
exten = s,1,Answer
exten = s,n,Wait(1)
Re: [asterisk-users] voip provider settings problem, please help
Am Montag, den 27.08.2007, 08:55 -0400 schrieb Jody Gugelhupf: hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when picked up, but after a while it stops working, incoming calls for this provider are not shown anymore in the CLI, but from other providers it always works, but the phone is ringingn nevertheless when calling my skypho account...when i then turn off the pbx and restart after sumthing like 2 hours my skypho account is working fine again, the incmiong calls are shown in the asterisk CLI, but after, i don't know let's say an hour or so it again stops working, incoming calls for my skypho account can not be seen in the asterisk CLI, then if i turn off the pbx for an hour or so it works again, so i thought it must be a setting issue, maybe something with the register? althought it always shows it registered when i use 'sip show registry' someone has an idea what i have to set or do to have it working permanently? what could be the problem here? i got no clue whatsoever and i have been using asterisk only since half a year, please help me, i'm totaly desperate, thx in advance jody :) Jody, you could post the relevant parts of your sip.conf here. For me (with a similar problem) introducing qualify=yes to the provider context in sip.conf solved the problem about 99.9% of the time; about three times a week I am off for less than 5 minutes at one particular providers - others work fine (I have a cronjob checking asterisk -rx sip show registry | grep 022396whatever which reports if status is NOT Registered - it does not do anything if the peer is not registered except sending me a notifier mail, so I have some kind of tracking). I am not familiar with italian voiceone though. Best, Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP provider for Turkey from India with Asterisk
Plainvoip has a very good A-Z and I have found they are fairly inexpensive. They also offer TollFree orig and some local dids. www.plainvoip.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy Sent: Friday, May 26, 2006 9:21 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoIP provider for Turkey from India with Asterisk Hi Friends, At present, I am using VoIPJET.COM provider for make calls to USA. I have two doubts. 1) I am unable to make call to UK Mobile phone. Why? 2) I want to make calls to Turkey country from India. With VoIPJET, I am unable to make call to Turkey and unable to find VoIP provider for Turkey. Please tell me VoIP Provider for Turkey from India. Looking forward for your response. ThanksRegards, Chandramouli Sneak preview the all-new Yahoo.com. It's not radically different. Just radically better. This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --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] VOIP provider
Thanks for the information, I will surely look into it! Nitin On 5/10/06, Kerry Garrison [EMAIL PROTECTED] wrote: Have you looked at CBeyond? I like their T1 SIPConnect product. From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Nitin GuptaSent: Wednesday, May 10, 2006 7:04 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] VOIP provider Hi, I am looking for voip providers in bay area, any suggestions? My requirements are: handling around 2000 calls a day (incoming) and around 1000 calls a day outgoing. I have a Asterisk PBX server to take care of routing calls to appropriate deparment. So I am looking mainly for IAX2 or SIP protocol support from VOIP provider. Also a dedicated t1 line in case provider can provide this too. Thanks, Nitin ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] VOIP provider
Have you looked at CBeyond? I like their T1 SIPConnect product. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nitin GuptaSent: Wednesday, May 10, 2006 7:04 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] VOIP provider Hi, I am looking for voip providers in bay area, any suggestions? My requirements are: handling around 2000 calls a day (incoming) and around 1000 calls a day outgoing. I have a Asterisk PBX server to take care of routing calls to appropriate deparment. So I am looking mainly for IAX2 or SIP protocol support from VOIP provider. Also a dedicated t1 line in case provider can provide this too. Thanks, Nitin ___ --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] Voip Provider
Title: Message Hi, feel free to contact me off-list, we can have a test if you want. [EMAIL PROTECTED] -Message d'origine-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mark AdamsEnvoyé: samedi 28 janvier 2006 15:50À: asterisk-users@lists.digium.comObjet: [Asterisk-Users] Voip Provider Hi Everyone, I know this may be off subject but I am not sure who to ask. I am currently looking for voip termination that is closest to replicating U.S. pots service. I run I.V.R. systems and I want to point Sipura 2100s to a voip terminator and have the DTMF tones properly detected. All that I need is outbound service and the problem I run into now is that when the called party presses a key on the phone it does not play it back properly to my system. I have tried to dial through voxee and plain voip and they both have the same problem. Im not sure if this is an asterisk issue or what. When I dial through packet 8, aptella or vonage everything works fine. I think my problems are because I am going through their asterisk servers. If anyone can help I would appreciate it, there is a potential for me using thousands of minutes per day if I could only find compatible service. I use the generic term U.S. Pots service because my dialers work perfectly on normal analog phone lines. Ive been looking for service for 2 months and I havent had any luck. P.S. I do not need any special services, just proper DTMF tone handling. Mark AdamsInfinity Marketing 1-800-430-1478 Main 530-579-8856 Fax ___ --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] Voip Provider
On Jan 28, 2006, at 6:50 AM, Mark Adams wrote: x-tad-smallerHi Everyone,/x-tad-smallerx-tad-smallerI know this may be off subject but I am not sure who to ask. I am currently looking for voip termination that is closest to replicating U.S. pots service. I run I.V.R. systems and I want to point Sipura 2100’s to a voip terminator and have the DTMF tones properly detected. All that I need is outbound service and the problem I run into now is that when the called party presses a key on the phone it does not play it back properly to my system. I have tried to dial through voxee and plain voip and they both have the same problem. Im not sure if this is an asterisk issue or what. When I dial through packet 8, aptella or vonage everything works fine. I think my problems are because I am going through their asterisk servers. If anyone can help I would appreciate it, there is a potential for me using thousands of minutes per day if I could only find compatible service./x-tad-smallerx-tad-smallerI use the generic term U.S. Pots service because my dialers work perfectly on normal analog phone lines. I’ve been looking for service for 2 months and I haven’t had any luck./x-tad-smallerx-tad-smallerP.S. I do not need any special services, just proper DTMF tone handling./x-tad-smallerThis might be a codec negotiation issue with the termination service. I am using Teliax with my asterisk server to terminate my SIP and IAX calls from several ATAs and softphones. All of that works fine with DTMF. I am using the G729 codec exclusively for my Teliax calls. You also need to be sure that the extensions for each ATA/phone have the DTMF configured righteously. HTH, Marty ___ --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] voip provider in a box
Dear trixter Our software AstBill is now in use/beeing implemented by many smaal service providers and a few very large. It is Open Source. I love to work with you on this and if any features are missing we be happy to implement it. Are Casilla -- http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk and Drupal Consultants http://astbill.com - Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com On 10/22/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: I am tasked with evaluating ready made solutions for a voip provider.Does anyone have any recommendations for software, specifically theenvironment will be a chargable voip provider (ie broadvoice, vonage,etc).They wanted me to see what was there and write something if nothing they like exists.Thanks--Trixter http://www.0xdecafbad.com Bret McDanelUK +44 870 340 4605 Germany +49 801 777 555 3402US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.1 (GNU/Linux)iD8DBQBDWjJh+1olxlzQw5cRAiaVAJ47j+iPhoQ1bBIpHdX4L+w/3gvfpACfUcfqme9ecSPfEqNVSfqlvNMsFZc==UATX-END PGP SIGNATURE- ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] voip provider in a box
On Tue, 2005-10-25 at 08:21 +0100, Are wrote: Dear trixter Our software AstBill is now in use/beeing implemented by many smaal service providers and a few very large. It is Open Source. I love to work with you on this and if any features are missing we be happy to implement it. I didnt initially want to use it because of the mysql 5.x requirement, however since I originally posted that mysql 5.x became 'stable' and I might consider it. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --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] voip provider in a box
We don't have a complete package quite yet. I think we have most of what you will need but we do not have support at present yet to accept customers payments. We can do that easily via 3rd party sofware but we can't do it ourselves yet. Anyway, www.aleph-com.net/astpp is the link. Darren Wiebe [EMAIL PROTECTED] trixter aka Bret McDanel wrote: I am tasked with evaluating ready made solutions for a voip provider. Does anyone have any recommendations for software, specifically the environment will be a chargable voip provider (ie broadvoice, vonage, etc). They wanted me to see what was there and write something if nothing they like exists. Thanks ___ --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 ___ --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] voip provider in a box
Bret, See my recent post: http://lists.digium.com/pipermail/asterisk-users/2005-October/130542.html I'll send you an email off list with the features and future roadmap. Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ trixter aka Bret McDanel wrote: I am tasked with evaluating ready made solutions for a voip provider. Does anyone have any recommendations for software, specifically the environment will be a chargable voip provider (ie broadvoice, vonage, etc). They wanted me to see what was there and write something if nothing they like exists. Thanks ___ --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 ___ --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] voip provider in a box
Hey ho, We have something like that (tailored for huge installations), contact me off list for more info. zoa. trixter aka Bret McDanel wrote: I am tasked with evaluating ready made solutions for a voip provider. Does anyone have any recommendations for software, specifically the environment will be a chargable voip provider (ie broadvoice, vonage, etc). They wanted me to see what was there and write something if nothing they like exists. Thanks ___ --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 signature.asc Description: OpenPGP digital signature ___ --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] voip provider in a box
We have a turn-key solution available that does exactly what you are asking for. You can reach someone for more information at 415.442.4010. TKS Paul [EMAIL PROTECTED] trixter aka Bret McDanel wrote: I am tasked with evaluating ready made solutions for a voip provider. Does anyone have any recommendations for software, specifically the environment will be a chargable voip provider (ie broadvoice, vonage, etc). They wanted me to see what was there and write something if nothing they like exists. Thanks ___ --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] voip provider request
Doesn't www.sipgate.co.uk do that? After all, they provide free NCFA numbers to the asking. -Original Message- From: trixter http://www.0xdecafbad.com [mailto:[EMAIL PROTECTED] Sent: Thursday, June 02, 2005 9:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voip provider request I am looking for a voip provider that provides good rates (below 5 cents/min or unlimited) to UK NCFA numbers. Braodvoice advertises they do unlimited to NCFA but does not have the ability to actually termiate those calls as per the CTO Nathan Stratton, and last he said they dont even have contracts in place to get service provisioned for that. As such I am looking for another provider to take over my broadvoice service. If anyone knows one I would greatly appreciate hearing about it. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ 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] voip provider request
On Fri, 2005-06-03 at 01:17 -0500, Jay Milk wrote: Doesn't www.sipgate.co.uk do that? After all, they provide free NCFA numbers to the asking. You misunderstand I am asking for termination *to* NCFA. I want to be able to call them, as my signature indicates I already have a NCFA for inbound (which is sipgate). -Original Message- From: trixter http://www.0xdecafbad.com [mailto:[EMAIL PROTECTED] Sent: Thursday, June 02, 2005 9:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voip provider request I am looking for a voip provider that provides good rates (below 5 cents/min or unlimited) to UK NCFA numbers. Braodvoice advertises they do unlimited to NCFA but does not have the ability to actually termiate those calls as per the CTO Nathan Stratton, and last he said they dont even have contracts in place to get service provisioned for that. As such I am looking for another provider to take over my broadvoice service. If anyone knows one I would greatly appreciate hearing about it. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ 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 -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] voip provider request
You misunderstand I am asking for termination *to* NCFA. Can you also terminate through Sipgate? They say: United Kingdom, 1.19 p/min I figure if they can provide origination for NCFA numbers, they can also terminate to them... your +44 870 number is a NCFA one, no? --Luki ___ 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] voip provider request
Naw, I understood you full well. You'd think if they provide origination to NCFA numbers, they'd provide termination to them as well, wouldn't you? As far as their website is concerned, there are only two UK rates, and no disclaimers that NCFA would be excluded. -Original Message- From: trixter http://www.0xdecafbad.com [mailto:[EMAIL PROTECTED] Sent: Friday, June 03, 2005 1:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voip provider request On Fri, 2005-06-03 at 01:17 -0500, Jay Milk wrote: Doesn't www.sipgate.co.uk do that? After all, they provide free NCFA numbers to the asking. You misunderstand I am asking for termination *to* NCFA. I want to be able to call them, as my signature indicates I already have a NCFA for inbound (which is sipgate). -Original Message- From: trixter http://www.0xdecafbad.com [mailto:[EMAIL PROTECTED] Sent: Thursday, June 02, 2005 9:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voip provider request I am looking for a voip provider that provides good rates (below 5 cents/min or unlimited) to UK NCFA numbers. Braodvoice advertises they do unlimited to NCFA but does not have the ability to actually termiate those calls as per the CTO Nathan Stratton, and last he said they dont even have contracts in place to get service provisioned for that. As such I am looking for another provider to take over my broadvoice service. If anyone knows one I would greatly appreciate hearing about it. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ 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 -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ 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] voip provider request
On Fri, 2005-06-03 at 00:10 -0700, Luki wrote: You misunderstand I am asking for termination *to* NCFA. Can you also terminate through Sipgate? They say: United Kingdom, 1.19 p/min I figure if they can provide origination for NCFA numbers, they can also terminate to them... your +44 870 number is a NCFA one, no? Ahh yes they do however they are 7.51ppm 24/7 for NCFA and 3 ppm for NCFA. 7.51 ppm is over $0.13/min using current exchange rates and I was looking for sub $0.05/min or even unlimited like broadvoice advertises but does not actually provide. Thanks though Anyone else know of any providers that allows you to call a UK NCFA (+44 870) for $0.05 USD or less per minute and is BYOD? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] voip provider request
On Fri, 2005-06-03 at 03:33 -0500, Jay Milk wrote: Naw, I understood you full well. You'd think if they provide origination to NCFA numbers, they'd provide termination to them as well, wouldn't you? As far as their website is concerned, there are only two UK rates, and no disclaimers that NCFA would be excluded. They hide it: http://www.sipgate.co.uk/faq/index.php?aktion=anzeigentype=rubrik=110#num6 -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] voip provider request
Ahhh... Sneaky. Because of the special billing agreements on NCFA numbers, there's bound to be a lower limit to how these calls are priced. I doubt BT gives sipgate (or any other VOIP provider) a signigicant discount on these calls. If you can reasonably expect that there are a lot of other sipgate/uk users being called, I'd go with them, as those calls are free after all. -Original Message- From: trixter http://www.0xdecafbad.com [mailto:[EMAIL PROTECTED] Sent: Friday, June 03, 2005 4:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voip provider request On Fri, 2005-06-03 at 03:33 -0500, Jay Milk wrote: Naw, I understood you full well. You'd think if they provide origination to NCFA numbers, they'd provide termination to them as well, wouldn't you? As far as their website is concerned, there are only two UK rates, and no disclaimers that NCFA would be excluded. They hide it: http://www.sipgate.co.uk/faq/index.php?aktion=anzeigentype=r ubrik=110#num6 -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ 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] voip provider request
On Fri, 2005-06-03 at 10:14 -0500, Jay Milk wrote: Ahhh... Sneaky. Because of the special billing agreements on NCFA numbers, there's bound to be a lower limit to how these calls are priced. I doubt BT gives sipgate (or any other VOIP provider) a signigicant discount on these calls. If you can reasonably expect that there are a lot of other sipgate/uk users being called, I'd go with them, as those calls are free after all. There isnt, so I cant. I found one company that accidentally told me that they pay $0.001/min to NCFA and they also offer unlimited but they are $250/mo (although that does allow resale of the service), but they dont wanna talk to me about slaes I bet its even worse once htey have my money, and $250 is a bit steep since I dont plan on resale. broadvoice gave unlimited as well, and at that point they were reselling GBLX ($35k/mo minimum spending limit on resale packages of that caliber). There is a carrier in the UK (not voip afaik) that gives unlimited to NCFA with a 1 pence connect fee. So there are carriers out there I just havent been able to find any why I asked here. Of course I need to be able to use asterisk with it, I do not want their hardware, although I do have a card I could use to connect to that at the cost of dropping my analogue service to my asterisk box. maxvoip.com.br does as well, but they arent local, language issues and afaik they dont sell to the US. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] voip provider request
On Fri, Jun 03, 2005 at 10:14:58AM -0500, Jay Milk wrote: Ahhh... Sneaky. Because of the special billing agreements on NCFA numbers, there's bound to be a lower limit to how these calls are priced. I doubt BT gives sipgate (or any other VOIP provider) a signigicant discount on these calls. If you can reasonably expect that there are a lot of other sipgate/uk users being called, I'd go with them, as those calls are free after all. The rates BT can charge are set by Ofcom (the UK telecoms/media regulator). There are 5600 Digital Local Exchanges (DLEs) in the UK, and to get single or dual tandem access you need to connect to around 770 of them. BT are regulated (as they have SMP [significan market power]) on how much they can charge for the single or dual tandem hop. Non-geographic numbers are usually mapped to geographic numbers (or virtually via an IN platform). 0845 numbers generally do not generate revenue (on ingress), 0870 numbers do (0870 are max 10p per minute, but not classified as premium rate). There are a few altnets who have connected to around the 770 DLEs and these include THUS, CW, probably Energis and maybe Carphone Warehouse (TalkTalk brand). Most others are either using these, or directly connecting to BT and paying too much money ;) Steve -- NetTek Ltd Fax +44-(0)20 7483 2455 Skype / In stevekennedyuk / UK +442088167166 / US +13106518226 Vonage UK +442079932612 / US +13108577715 / UK mob 07775 755503 Personal Blog http://stevekennedy.blogspot.com Euro Tech News Blog http://eurotechnews.blogspot.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] voip provider request
On Fri, 3 Jun 2005, trixter http://www.0xdecafbad.com wrote: Anyone else know of any providers that allows you to call a UK NCFA (+44 870) for $0.05 USD or less per minute and is BYOD? The closest I've seen is http://www.iax.cc/ (sixTel) who charges just over 5c/min. Direct from their web site: United Kingdom -Premium 448 5.301¢ /min (FWIW, I haven't had any problems with them for outbound, but they have been *VERY* slow to provision DIDs lately.) -SC -- Stanley Cline // Telco Boi // sc1 at roamer1 dot org // www.roamer1.org it seems like all you ever buy is Abercrombie and cell phones --a friend ___ 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] Voip Provider in Brazil
Asterisk wrote: Hi all, Is there a VOIP provider that can deliver local Rio de Janeiro numbers? I am looking for a normal Rio number for my Asterisk box. I'm using a RJ DID, right now http://www.libretel.com with IAX DID (they offer SP also). Have not tried much on it, noticed DTMF can be a little picky, butdidn't try anything on troubleshooting. ___ 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] Voip Provider in Brazil
Uhmm it is great that I can get it but it is a little but expansive. 16.95 US dollars a month. I was hoping for a cheaper on or local one here in Rio de Janeiro. Thanks for this one Johannes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julio Arruda Sent: Sunday, May 15, 2005 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voip Provider in Brazil Asterisk wrote: Hi all, Is there a VOIP provider that can deliver local Rio de Janeiro numbers? I am looking for a normal Rio number for my Asterisk box. I'm using a RJ DID, right now http://www.libretel.com with IAX DID (they offer SP also). Have not tried much on it, noticed DTMF can be a little picky, butdidn't try anything on troubleshooting. ___ 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] Voip Provider in Brazil
I have no idea on rates, services or quality, I just happened to read their name and store it away in my head. maxvoip.com.br Unfortunately I do not think they are BYOD, but they might be. On Sun, 2005-05-15 at 10:55 -0300, Asterisk wrote: Uhmm it is great that I can get it but it is a little but expansive. 16.95 US dollars a month. I was hoping for a cheaper on or local one here in Rio de Janeiro. Thanks for this one Johannes -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] VoIP Provider problems
No, I'm not ignorant of how this works. You'll notice I put it appears bad when I posted my results. Yes, it's not a perfect way to show problems -- but taken with a grain of salt it's not half bad. Especially when sampled over a longer period of time, and if the original poster can correlate the PingPlotter results to the quality of his calls. Now if he shows 30% loss during good and bad calls, that's another story. I posted my results to help the original poster. If he's trying to troubleshoot an apparent bad connection with Sprint, he needs all the help he can get. If they can proove the connection works even the littlest bit, they'll say it's fine and blame Broadvoice. If everyone gets similar levels of loss at those points, one could conclude its a side effect of the routers having better things to do. But if he's the only one showing them, then it would be a starting point to conclude something is wrong with his connection or something along Sprint's backbone. I'm not the original poster either, but for those following this thread keep in mind that a fair number of isp's use an upper-layer device to throttle data flows to some predeteremined rate. For example, I know some cable broadband companies that throttle their users to 128k up and some other value down. Don't have a clue whether their throttling box drops packets, delays them, or what; however, considering they would want to handle both udp and tcp, I'd have to bet some amount they drop udp packets to throttle udp data flows. On the other hand, I know of several dsl broadband companies that don't pay any attention to their uplink congestion, letting their uplink routers drop packets, etc. Since they can't afford to chase uplink utilizations by augmenting bandwidth, dropped packets happen frequently. Nature of the beast for some. ___ 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] VoIP Provider problems
On Friday 01 April 2005 04:28, Joseph Gutowski wrote: Ok, since I guess no one else wanted to bite -- I will. I installed PingPlotter, switched to UDP just to be the same as you, and ran it against sip.broadvoice.com. Absolutley no problems, no packet loss at all. Ran it with all of the published proxy addresses, again no problems. I then used the 63.251.209.126 that you posted, and it was awful (at least it appears awful). I have reliable 20% packet loss at each of two Verio hops (nothing lost at the far end). Don't take this the wrong way, but you are showing a bit of ignorance about how TCP/IP works. The apparent packet loss you are seeing may be just fine tuning of the routers in question. The routers may be set up not to send ICMP host/network unreachables back to the originating system if they are required to send more than one in a configured time period. Routers have better things to do than continually tell you that a host is unreachable. B ___ 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] VoIP Provider problems
Bob Goddard wrote: The apparent packet loss you are seeing may be just fine tuning of the routers in question. This is the conclusion I came to as well; however, with the way PingPlotter works the router is not sending ICMP unreachables but rather ICMP TTL expired responses. In any case, the routers in question may either be: 1) ...intentionally discarding the received UDP ping packets (these are not ICMP pings, but rather UDP packets with TTL down to zero when they get to the router), because the router has better things to do. 2) ...throttling the ICMP TTL expired responses to a certain rate per period of time, as you suggest. This would appear as packet loss. 3) ...actually congested, with the received UDP pings (and other types of packets) getting discarded on the input side at the rate shown in the data. I wish there was a way to measure 3) without being affected by 1) and 2). I agree then, that PingPlotter is not a highly accurate way to measure path quality. Still, though, looking over the data for a couple days now it is easy to see cyclical patterns that go from 1% to 30% (PingPlotter measured) loss, and an easily seen correlation with the voice quality of my outbound Broadvoice calls. Interestingly enough, switching from a Firefly soft phone on my workstation, using IAX2/ulaw, to an analog phone-TDM400 FXS port right at the Asterisk server has made a big difference. So some of the perceived crappiness was in the soft phone-Asterisk path and was probably being exacerbated by the network loss on the net or at Broadvoice's router. -Johnathan ___ 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] VoIP Provider problems
The apparent packet loss you are seeing may be just fine tuning of the routers in question. This is the conclusion I came to as well; however, with the way PingPlotter works the router is not sending ICMP unreachables but rather ICMP TTL expired responses. In any case, the routers in question may either be: 1) ...intentionally discarding the received UDP ping packets (these are not ICMP pings, but rather UDP packets with TTL down to zero when they get to the router), because the router has better things to do. 2) ...throttling the ICMP TTL expired responses to a certain rate per period of time, as you suggest. This would appear as packet loss. 3) ...actually congested, with the received UDP pings (and other types of packets) getting discarded on the input side at the rate shown in the data. I wish there was a way to measure 3) without being affected by 1) and 2). The deceptive part of doing the above is that once you see congestion (lack of an icmp response), you still have absolutely no idea what device was at fault. In other words, as the ttl value is increased and additional icmps are sent, you might see what you believe is congestion, but you still don't have any clue as to whether hop #2, #5, or #10 actually was involved with that congestion. ___ 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] VoIP Provider problems
Rich Adamson wrote: In other words, as the ttl value is increased and additional icmps are sent, you might see what you believe is congestion, but you still don't have any clue as to whether hop #2, #5, or #10 actually was involved with that congestion. Sure. But there is a way around this. The traceroute-style statistics gathering technique that PingPlotter uses tries all the hops at the same time and plots the return rate for each one. So a 10 hop path has 10 packets go out, with individual packet's TTL set to expire at each hop. Done over and over again and averaged over many probes, you get a very clear picture. Packet loss at one node affects all the probes to that node and further ones, resulting in an increasing loss rate as you go down the path. For example: Hop Loss 1 0% 2 1% 3 1% 4 5% 5 5% 6 6% 7 15% 8 15% 9 16% 10 16% It's easy to see there is a big problem between hops 6 and 7 and a smaller problem between hops 3 and 4. With the broadvoice router I was seeing (at first) a jump from 0% to 9% at my local ISP, then small increments over the next 10 hops until it was at about 14%, then a big jump to 29% at the last hop. The data has varied cyclically between as high as the above and as low as 1% all the way across. Right this very moment, it is 2% within my ISP, still 2% all the way to PNAP, then a jump to 14% at the broadvoice ingress router at PNAP. Again, temper the above with the fact that the packet loss may be intentional, and these statistics not representative of real RTP traffic, as per my previous message. But I can predict with high accuracy what the caller on the other end of my broadvoice call will say about my voice quality based on the last number I see for the broadvoice ingress router. -Johnathan ___ 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] VoIP Provider problems
No, I'm not ignorant of how this works. You'll notice I put it appears bad when I posted my results. Yes, it's not a perfect way to show problems -- but taken with a grain of salt it's not half bad. Especially when sampled over a longer period of time, and if the original poster can correlate the PingPlotter results to the quality of his calls. Now if he shows 30% loss during good and bad calls, that's another story. I posted my results to help the original poster. If he's trying to troubleshoot an apparent bad connection with Sprint, he needs all the help he can get. If they can proove the connection works even the littlest bit, they'll say it's fine and blame Broadvoice. If everyone gets similar levels of loss at those points, one could conclude its a side effect of the routers having better things to do. But if he's the only one showing them, then it would be a starting point to conclude something is wrong with his connection or something along Sprint's backbone. ___ 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] VoIP Provider problems
Johnathan Corgan wrote: First off, I have Sprint Broadband Direct internet service, a fixed wireless setup with a 2-5 Mbps downlink and a terrible 128 kbps uplink. So I know I'm in for trouble anyway. The broadvoice edge router (63.251.209.126, their lax site) is another 11 hops away. One hop before that, the packet loss rate has gone up to 13%, so the Internet adds another 4% to my sucky ISP connection. Round trip time to this point is 200ms, so-so but livable. Here's the kicker: Reported packet loss from broadvoice, one additional hop, is a whopping 29%. So between the last Internet router (bbnet2.lax.pnap.net) and broadvoice's edge router, there is an additional 16% loss. Just an update after about 12 hours of data--the data above was worst case. During off-peak hours in the middle of the night the packet loss at my ISP was effectively zero, and only 3% along the way to broadvoice, with a 75ms round-trip time. Broadvoice edge-router still reports 28% packet loss though, and an additional 30ms RTT increase for this last hop. So I even more strongly suspect (or just really hope) they are preferentially discarding non-RTP traffic in favor of voice traffic. I did discover that the multi-second outages are at my local ISP, not at Broadvoice--for some reason Sprint BBD can take up to 4 seconds to respond to a ping, so something is really wrong there--but is there a way to do this type of testing in a more rigorous and controlled fashion? -Johnathan ___ 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] VoIP Provider problems
Ping runs as a low priority service so it is not realistic to measure response time using ping. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johnathan Corgan Sent: Thursday, March 31, 2005 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoIP Provider problems Johnathan Corgan wrote: First off, I have Sprint Broadband Direct internet service, a fixed wireless setup with a 2-5 Mbps downlink and a terrible 128 kbps uplink. So I know I'm in for trouble anyway. The broadvoice edge router (63.251.209.126, their lax site) is another 11 hops away. One hop before that, the packet loss rate has gone up to 13%, so the Internet adds another 4% to my sucky ISP connection. Round trip time to this point is 200ms, so-so but livable. Here's the kicker: Reported packet loss from broadvoice, one additional hop, is a whopping 29%. So between the last Internet router (bbnet2.lax.pnap.net) and broadvoice's edge router, there is an additional 16% loss. Just an update after about 12 hours of data--the data above was worst case. During off-peak hours in the middle of the night the packet loss at my ISP was effectively zero, and only 3% along the way to broadvoice, with a 75ms round-trip time. Broadvoice edge-router still reports 28% packet loss though, and an additional 30ms RTT increase for this last hop. So I even more strongly suspect (or just really hope) they are preferentially discarding non-RTP traffic in favor of voice traffic. I did discover that the multi-second outages are at my local ISP, not at Broadvoice--for some reason Sprint BBD can take up to 4 seconds to respond to a ping, so something is really wrong there--but is there a way to do this type of testing in a more rigorous and controlled fashion? -Johnathan ___ 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] VoIP Provider problems
On Thursday 31 March 2005 15:59, Kellner, Peter wrote: Ping runs as a low priority service so it is not realistic to measure response time using ping. Try tracepath. It's not using port 7 and can be used by normal users. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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] VoIP Provider problems
Ok, since I guess no one else wanted to bite -- I will. I installed PingPlotter, switched to UDP just to be the same as you, and ran it against sip.broadvoice.com. Absolutley no problems, no packet loss at all. Ran it with all of the published proxy addresses, again no problems. I then used the 63.251.209.126 that you posted, and it was awful (at least it appears awful). I have reliable 20% packet loss at each of two Verio hops (nothing lost at the far end). I did traceroutes on all of the Broadvoice proxies, and I didn't get pushed through PNAP. I wonder why your packets seem to reliably following that path when it's so bad. I mean the whole point of routing through PNAP is to increase quality, no? And from my understanding they're supposed to have a magic fuzzy logic to dynamically reroute around problems. Your results suggest a more widespread problem than one customer can't have nice VoIP calls -- you'd think Sprint wouldn't be routing through PNAP. Am I going to slap myself on the forehead in two minutes when I realize I missed something obvious and I'm completely off base here due to lack of sleep? I think you need to have a nice chat with Sprint, because it looks like your connection is pretty icky for anything, nevermind VoIP. I hope it's cheap. And I also hope your VoIP connection is wired if you're getting 9-10% loss on the wireless before you even leave the LAN. If you're starting off with a loss, it's just going to make the natural losses on the net have an even worse effect. ___ 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] VoIP Provider problems
Joseph Gutowski wrote: I installed PingPlotter, switched to UDP just to be the same as you, and ran it against sip.broadvoice.com. Absolutley no problems, no packet loss at all. Well, that's good to hear. I then used the 63.251.209.126 that you posted, and it was awful (at least it appears awful). I have reliable 20% packet loss at each of two Verio hops (nothing lost at the far end). Okay, happy to see independent confirmation of this. I did traceroutes on all of the Broadvoice proxies, and I didn't get pushed through PNAP. I wonder why your packets seem to reliably following that path when it's so bad. I mean the whole point of routing through PNAP is to increase quality, no? And from my understanding they're supposed to have a magic fuzzy logic to dynamically reroute around problems. Your results suggest a more widespread problem than one customer can't have nice VoIP calls -- you'd think Sprint wouldn't be routing through PNAP. Well, you're right, Sprint is going through sprintlink.net - PNAP - BV, no route changes during the day since I started. Not a lot I can do about that, unfortunately. And I also hope your VoIP connection is wired if you're getting 9-10% loss on the wireless before you even leave the LAN. If you're starting off with a loss, it's just going to make the natural losses on the net have an even worse effect. It appears I happened to pick the most congested time to measure, and got 8-9% packet loss on my Sprint uplink. That's the wireless, as in a fixed wireless MMDS rooftop dish link to a mountain top about 15 miles away. It turns out that off peak there is zero loss over this link and typically it is only about 2-3% loss. So it's not as bad as it first seemed. On the premises it is all wired and first router is always zero packet loss. As I write this the trailing 10 minutes of data shows an aggregate 9% loss to BV with 3% of that on the Sprint BBD uplink side. This is much better than my first tests, and my SIP calls through broadvoice show the difference too. Anyway, I haven't tried the other broadvoice proxies yet, I'm really hoping at least one doesn't have PNAP on its path. (At least I can be thankful I haven't run into any of the weird NAT or authentication issues that have been discussed--worked great first time.) At the time I got this wireless link (which with a 4Mbps downlink, is pretty sweet for typical traffic patterns), there was no DSL in my area. Now SBC has service, but I've yet to really look into it. (As an aside, got my TDM400 card today, installed, and have the FXS port working with an analog phone. Woohoo. FXO next!) -Johnathan ___ 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] VoIP Provider problems
We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line about 2 Mb. ADSL has slower upload speeds than download speeds (your 2Mbps is download). so you may have problems with your outgoing packets of sound. g.711 codec (the default codec for most voip providers because there is virtually no sound quality loss) uses about 84Kbps per channel or simultaneous connection. For example if you have an Upload speed of 128Kbps. and you try to have 2 phone conversations you would need 168kbps transfer speed. That is 40kbps more than your upload speed. This is a major problem with ADSL the upload and download speeds are not equal. Another potential problem is that your provider is over subscribed for the available bandwidth. What this means is that when allot of people are using their connection to your provider. The provider may not be able to handle all those users at once and packets get dropped or delayed. Dropping or delaying packets is very bad for VoIP especially if they do not do QoS or ToS routing which most providers do not. What is your upload speed? Some other possibilities are to use some compression codecs which will cause some sound quality loss like gsm or iLibc and g.729 to pack more calls in the limited bandwidth limitations. Another option is to use SDSL where the speeds of both the upload and download are the same. We feel our quality decrease when in US are about 9:00 or 10:00 in the morning. This time is when businesses in the us are opening and starting to do business In the united states. Both for phones and Data. We do not know if this is it correct or all the people using VoIp provider feel the same quality? This may mostly be in relation to you Internet provider and how many hops you have to take to get to the VoIP provider and if they oversubscribe their bandwidth capacity. One provider may be good for one person with one person in a different ISP than an ISP you have. And you are even right next door to each other. This is as a result of how the internet is connected and may not nessessarly be geographic. For example you may be connecting to a server in your own city lets say Chicago but you are actually routed to San Francisco then back to Chicago. But it will not always take the same path the next time you may be routed through New York. This is a simplification of how it works. The closer you are to a Tier 1 provider(they own the major trunks interconnects) the less time it will take to get to your target. Anyone knows any provider without this kind of problems? I have seen many Providers have both Good and bad connection links. It is best to have a provider that routes with QoS and/or ToS within their routers and have only one or two hops between your provider and a tear 1 provider. Witch provider do you use to get the best sounds quality? It is not that simple. But you can begin by doing a traceroute to the many providers at different times of the day. This will see the route changes and time delays between hops to get to VoIP Providers gateways. Hope this helps in understanding the problems involved with choosing a provider. Thanks, Max ___ 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] VoIP Provider problems
It is not that simple. But you can begin by doing a traceroute to the many providers at different times of the day. This will see the route changes and time delays between hops to get to VoIP Providers gateways. The best tool I've found for monitoring connections, routes, congestion, is called PingPlotter. http://pingplotter.com/ It's a shareware visual traceroute. It continually graphs the traceroute style responses. There is a scrollable timeline to view how things change. You can get raw data out of it as well. It records changes in routes. Their web site also has some tutorials on how to use pingplotter to track down problems. Unfortunately it's windows only. It will run under vmware though. I have no affiliation with them. I've just found it very useful. --Rob ___ 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] VoIP Provider problems
Give me a try! www.shelltel.com And don't use G711 for your calls. invest in the G729 codec. you'll find your calls will start working better. I'm a G729 shop. Thanks Michael D. Schelin 626-814-2454 Max W Blackmer Jr wrote: We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line about 2 Mb. ADSL has slower upload speeds than download speeds (your 2Mbps is download). so you may have problems with your outgoing packets of sound. g.711 codec (the default codec for most voip providers because there is virtually no sound quality loss) uses about 84Kbps per channel or simultaneous connection. For example if you have an Upload speed of 128Kbps. and you try to have 2 phone conversations you would need 168kbps transfer speed. That is 40kbps more than your upload speed. This is a major problem with ADSL the upload and download speeds are not equal. Another potential problem is that your provider is over subscribed for the available bandwidth. What this means is that when allot of people are using their connection to your provider. The provider may not be able to handle all those users at once and packets get dropped or delayed. Dropping or delaying packets is very bad for VoIP especially if they do not do QoS or ToS routing which most providers do not. What is your upload speed? Some other possibilities are to use some compression codecs which will cause some sound quality loss like gsm or iLibc and g.729 to pack more calls in the limited bandwidth limitations. Another option is to use SDSL where the speeds of both the upload and download are the same. We feel our quality decrease when in US are about 9:00 or 10:00 in the morning. This time is when businesses in the us are opening and starting to do business In the united states. Both for phones and Data. We do not know if this is it correct or all the people using VoIp provider feel the same quality? This may mostly be in relation to you Internet provider and how many hops you have to take to get to the VoIP provider and if they oversubscribe their bandwidth capacity. One provider may be good for one person with one person in a different ISP than an ISP you have. And you are even right next door to each other. This is as a result of how the internet is connected and may not nessessarly be geographic. For example you may be connecting to a server in your own city lets say Chicago but you are actually routed to San Francisco then back to Chicago. But it will not always take the same path the next time you may be routed through New York. This is a simplification of how it works. The closer you are to a Tier 1 provider(they own the major trunks interconnects) the less time it will take to get to your target. Anyone knows any provider without this kind of problems? I have seen many Providers have both Good and bad connection links. It is best to have a provider that routes with QoS and/or ToS within their routers and have only one or two hops between your provider and a tear 1 provider. Witch provider do you use to get the best sounds quality? It is not that simple. But you can begin by doing a traceroute to the many providers at different times of the day. This will see the route changes and time delays between hops to get to VoIP Providers gateways. Hope this helps in understanding the problems involved with choosing a provider. Thanks, Max ___ 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] VoIP Provider problems
Robert Terzi wrote: The best tool I've found for monitoring connections, routes, congestion, is called PingPlotter. http://pingplotter.com/ It's a shareware visual traceroute. It continually graphs the traceroute style responses. There is a scrollable timeline to view how things change. You can get raw data out of it as well. It records changes in routes. Thanks for the excellent link. I've had Asterisk on a home network and Broadvoice for a couple weeks now. IAX2 calls between Firefly soft-phone on my desk and other soft phones directly on the net have worked fairly well, but reported voice quality when going out over broadvoice to the PSTN has really stunk, making it only marginally useful. So I've downloaded this utility and am now tracing out sip.broadvoice.com, using UDP (as my ISP filters icmp.) Actually, the trace doesn't get past broadvoice's edge router, so I replaced the final IP address with that of the edge router itself so I could see the data instead of destination unreachable. Anyway, with only 20 minutes I've data I'm seeing some rather disappointing results. First off, I have Sprint Broadband Direct internet service, a fixed wireless setup with a 2-5 Mbps downlink and a terrible 128 kbps uplink. So I know I'm in for trouble anyway. First hop off my home lan over the wireless starts at about 9% packet loss. Sucks but for normal TCP based stuff (email, web, ssh, etc.) life goes on but just a little slower. The broadvoice edge router (63.251.209.126, their lax site) is another 11 hops away. One hop before that, the packet loss rate has gone up to 13%, so the Internet adds another 4% to my sucky ISP connection. Round trip time to this point is 200ms, so-so but livable. Here's the kicker: Reported packet loss from broadvoice, one additional hop, is a whopping 29%. So between the last Internet router (bbnet2.lax.pnap.net) and broadvoice's edge router, there is an additional 16% loss. No wonder my outgoing voice to the PSTN is choppy, filled with several second gaps, and makes people laugh at me for spending $20 a month on VOIP. I admit I can help things a bit by getting an ADSL or SDSL link with a better provisioned uplink, but even if I had 0% loss to broadvoice, their own net connection seems seriously under-provisioned. One thing might be affecting this and make these numbers suspect--broadvoice might have QoS on the edge router such that non-RTP packets get lower-class status, so my UDP pings are artificially dropped in favor of real RTP traffic (actually, I'd be doing this if I were them.) Anyone care to comment on how realistic a test this is? I'll do these tests for a few hours and hit the different broadvoice proxy networks and see if there is a difference, and compare to loss rates for other sites over my ISP uplink. Anyway, my Digium 11b card comes in tomorrow, so I'll be off to more fun setting up the IVR and voicemail, etc., for my home line off the PSTN...love this Asterisk thing. -Johnathan ___ 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] VoIP Provider problems
On Tue, 2005-03-29 at 12:36 +0200, Ismael Gil wrote: Hello all, We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond) If you find the same problem with multiple ITSP's, then it may not be them that is at fault. We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line about 2 Mb. Sounds like it should be quite adequate... how many simultaneous calls are you doing? We feel our quality decrease when in US are about 9:00 or 10:00 in the morning. What time is that for your local time? Is there something that might be happening at/around that time for you? eg, here, around 3 - 6pm is quite busy as school kids get home and go on the internet, same for people getting home from work. In fact, my vague recollection is that things just get busier until around 11pm, before they really slow down. While this doesn't have any relation to *your* adsl connection, think about what this might be doing to your ISP's internet connection We do not know if this is it correct or all the people using VoIp provider feel the same quality? Not that I would know, but I get the feeling that most people get extremely good quality calls over a decent internet connection. Anyone knows any provider without this kind of problems? Witch provider do you use to get the best sounds quality? I've not used any, so can't comment on this. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] VoIP Provider SIP Call Flow
James Rothenberger wrote: I am testing a call flow in which an inbound SIP call (to the Asterisk from a PSTN connection from a SIP VoIP provider) is not answered (nobody there and no voicemail) and the call is terminated on the PSTN side. After the SIP CANCEL is sent to the Asterisk from the PSTN, The SIP phone sends a 487 response back to the Asterisk (Request Terminated) as it should. What is NOT occurring is that the 487 is NOT propagated back to the provider. The asterisk simply sends an OK back in acknowledgment of the initial CANCEL. How do I force the Asterisk to send the 487? I also have the same signaling problem with 486, 481, and 408 SIP responses. I am using asterisk v1.0.0. I know what you mean. We saw the same issue. Try version 1.0.3 or better. It should work as you expect it. Thank you! -- Andres Network Admin http://www.telesip.net ___ 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] VoIP Provider
Hi, * Erik Lagerway wrote/schrieb: There is a provider in the US - www.AddaLine.com, who just launched a SIP service with some great rates for North America I have been using their service for months and I am extremely happy with the service. looks like Germany is again laggin behind all others in the communication field. Or I asked at the wrong place. There might not be to many people from Germany in this list. Anyway, thanks for the answer. CU MartinD: -- +++ GMX - Mail, Messaging more http://www.gmx.net +++ Bitte lächeln! Fotogalerie online mit GMX ohne eigene Homepage! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Provider
Iconnecthere seems to have better rates... -Original Message- From: Martin Dommermuth [EMAIL PROTECTED] Date: Thu, 12 Jun 2003 19:48:43 +0200 (MEST) To: [EMAIL PROTECTED] [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoIP Provider Hi, * Erik Lagerway wrote/schrieb: There is a provider in the US - www.AddaLine.com, who just launched a SIP service with some great rates for North America I have been using their service for months and I am extremely happy with the service. looks like Germany is again laggin behind all others in the communication field. Or I asked at the wrong place. There might not be to many people from Germany in this list. Anyway, thanks for the answer. CU MartinD: -- +++ GMX - Mail, Messaging more http://www.gmx.net +++ Bitte lächeln! Fotogalerie online mit GMX ohne eigene Homepage! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Provider
Martin == Martin Dommermuth [EMAIL PROTECTED] writes: Martin looks like Germany is again laggin behind all others in the Martin communication field. Or I asked at the wrong place. There Martin might not be to many people from Germany in this list. One possibility is Pulver's LibrTel at http://www.libretel.com. Their ratesheet says usd 0.04 for most of the country, USD 0.03 for Frankfurt and USD 0.23 for mobiles. They are still in beta, and I expect will be offering just SIP. I don't have the international rates for http://nufone.net ; they may be competitive with librtel and support IAX and H.323 as well. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Provider
On 12 Jun 2003, James H. Cloos Jr. wrote: One possibility is Pulver's Libr=C3=A9Tel at http://www.libretel.com. Whenever I try any of their access numbers (at least the ones around me, in the DC area), I get a recording The number you have reached is not in service. This does not inspire great confidence. miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Provider
(BHi (B (BWe can found a couple of ITSP at Jasomi networks's PR. (B (Bhttp://www.jasomi.com/pr_deployment.html (B (BDoes anyone try it? (B (Bmack (B (BOn Thu, 12 Jun 2003 19:48:43 +0200 (MEST) (BMartin Dommermuth [EMAIL PROTECTED] wrote: (B (BHi, (B (B* Erik Lagerway wrote/schrieb: (B (B (B There is a provider in the US - www.AddaLine.com, who just launched a (B SIP service with some great rates for North America (B (B I have been using their service for months and I am extremely happy with (Bthe (B service. (B (Blooks like Germany is again laggin behind all others in the (Bcommunication field. (BOr I asked at the wrong place. There might not be to many people from (BGermany in this list. (B (BAnyway, thanks for the answer. (B (BCU (BMartinD: (B (B (B-- (B+++ GMX - Mail, Messaging more http://www.gmx.net +++ (BBitte l$BgD(Bheln! Fotogalerie online mit GMX ohne eigene Homepage! (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP Provider
On Mon, 9 Jun 2003, Gary wrote: Just a quick look at their rates show they just might be into rip off's.. Australia0.06 (0-61-0) Australia-Cellular 0.31 (0-61-7) Australia-Cellular 0.31 (0-61-8) Australia-Cellular 0.31 (0-61-1) Australia-Cellular 0.31 (0-61-4) Australia-Cellular 0.31 (0-61-5) Australia-Cellular 0.31 (0-61-71) Australia-Cellular 0.31 (0-61-78) Australia-Cellular 0.31 (0-61-79) Australia-Melbourne 0.06 (0-61-3) Australia-Sydney 0.06 (0-61-2) What they label as Australia-Cellular is bullshit !! Only one being 61-4 is actually cellular. Also their rates are quite high for VoIP. 0.06 for minutes within the USA? That's twice as much as other VoIP providers. Their international rates to places I call are also similarly high. miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
