[Asterisk-Users] Re: Voicepulse connect - unable to dial out, asterisk says 9696
Thanks, I had added more funds a couple weeks ago, however I just found this on the voicepulse connect KB:- Why do I hear error message '9696' when dialing with Asterisk? Question Why do I hear error message '9696' when dialing with Asterisk? Answer When making outgoing calls using the VoicePulse Connect! service, if you hear a message stating 9696, your account has been suspended for negative balance or misuse. Please contact technical support through the Account Center to resolve the issue. So it looks like even though I added $20 credit, I'm still suspended. Anyway I contacted them, lets see what happens. Mike On 7/17/05, S. William Schulz [EMAIL PROTECTED] wrote: Mike Dent wrote: -- Called NBhX:[EMAIL PROTECTED]/12124565900 -- Call accepted by 66.234.228.160 (format ulaw) -- Format for call is ulaw -- Hungup 'IAX2/66.234.228.160:4569/1' -- Executing HasNewVoicemail(SIP/2008-cf55, 2002) in new stack rt*CLI and Asterisk speaks back to me 96 96 I seem to recall a similar error mentioned and the problem turned out to be a need to add more funds to the account. I can't say that this is the case, but it might be worth checking. S ___ 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] Re: Voicepulse connect has doubled their rates
Two other things appear to have changed if you make auto payments. On the auto-recharge section of their account center website, the minimum threshold has been raised from $1.00 to $10.00, and the recharge value from $10.00 to $25.00. It looks like the LD rate has dropped from 2.95 cents to 2.4 cents per minute... - Original Message - From: Tim Burt [EMAIL PROTECTED] Newsgroups: gmane.comp.telephony.pbx.asterisk.user Sent: Wednesday, March 30, 2005 1:39 PM Subject: Voicepulse connect has doubled their rates Today I received an email informing me that effective April 1, my per number charge for VOIP will almost double. This is the downside of VOIP. It is unregulated. I have published and distributed my new VOIP phone number, and now, with no warning, my monthly charge has doubled. Ouch.. Beware of which provider you choose! There is nothing to prevent them from doubling my rates again on May 1st! ___ 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
[Asterisk-Users] Re: Voicepulse Open Access Asterisk Problems
I got this working if anyone out there is looking to do the same. See: http://www.dslreports.com/forum/remark,12899866~mode=flat#12899866 After some more experimenting, I discovered that you MUST use the long register statement ala Broadvoice. Unlike Broadvoice the service has been ROCK SOLID. Too bad you must have a regular account first :( On Thu, 17 Feb 2005 19:03:39 -0500, Brian Dingman [EMAIL PROTECTED] wrote: I can't seem to dial out with Voicepulse Open Access service using *. Incoming works fine. Another user posted a few weeks back that they were having problems and there are some threads at dslreports.com about this as well. Maybe someone here can figure out what the issue is from the sip debug info below. I am at a loss. The audible error message from Allison is 0984 (from VP server) Here is all the pertinent info: [sip.conf] [general] port = 5060 bindaddr = 0.0.0.0 srvlookup=yes tos=lowdelay maxexpirey=3600 disallow=all allow=ulaw musicclass=default language=en relaxdtmf=yes ;useragent=Asterisk PBX ;nat=yes register = s00**:[EMAIL PROTECTED] externip=asterisk.briandingman.com localnet=192.168.1.0/255.255.0.0 [voicepulse] type=friend context=voicepulse-incoming username=s00** secret= host=access1.voicepulse.com dtmf=inband nat=yes qualify=yes canreinvite=no insecure=very [1000] type=friend host=dynamic ;callerid=Brian 1000 dtmfmode=rfc2833 mailbox=1000 context=Home ;nat=no ;qualify=yes secret= Error message from CLI: -- Executing Macro(SIP/1000-fbdb, vp-dial|16109951010) in new stack -- Executing Dial(SIP/1000-fbdb, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/voicepulse-e009 is making progress passing it to SIP/1000-fbdb Feb 17 17:08:42 WARNING[8523]: chan_sip.c:6811 handle_response: Forbidden - wrong password on authentication for INVITE to '1000 sip:[EMAIL PROTECTED];tag=as3e632d2a' -- SIP/voicepulse-e009 is circuit-busy == Everyone is busy/congested at this time -- Executing Hangup(SIP/1000-fbdb, ) in new stack == Spawn extension (macro-vp-dial, s, 2) exited non-zero on 'SIP/1000-fbdb' in macro 'vp-dial' == Spawn extension (Home, 16109951010, 1) exited non-zero on 'SIP/1000-fbdb' -- Got SIP response 481 Call Leg Does Not Exist back from 66.234.228.159 (Sorry for the length) SIP Debug info: -- Executing Macro(SIP/1000-cd47, vp-dial|16109951010) in new stack -- Executing Dial(SIP/1000-cd47, SIP/[EMAIL PROTECTED]) in new stack We're at 68.163.52.50 port 15640 Answering/Requesting with root capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport From: 1000 sip:[EMAIL PROTECTED];tag=as74c56bff To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 17 Feb 2005 22:10:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 214 v=0 o=root 8523 8523 IN IP4 68.163.52.50 s=session c=IN IP4 68.163.52.50 t=0 0 m=audio 15640 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 66.234.228.159:5060 -- Called [EMAIL PROTECTED] asterisk*CLI Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;received=68.163.52.50;rport=50210 From: 1000 sip:[EMAIL PROTECTED];tag=as74c56bff To: sip:[EMAIL PROTECTED];tag=as1ecc3219 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: VoicePulse SW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=uasw001.voicepulse.com, nonce=5d626333 Content-Length: 0 11 headers, 0 lines Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK600a4321;rport From: 1000 sip:[EMAIL PROTECTED];tag=as74c56bff To: sip:[EMAIL PROTECTED];tag=as1ecc3219 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 66.234.228.159:5060 We're at 68.163.52.50 port 15640 Answering/Requesting with root capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 68.163.52.50:5060;branch=z9hG4bK709030d1;rport From: 16109951010 sip:[EMAIL PROTECTED];tag=as74c56bff To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username=s00**, realm=uasw001.voicepulse.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=5d626333, response=HASH***, opaque= Date: Thu, 17 Feb 2005 22:10:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS,
[Asterisk-Users] Re: voicepulse silence during conversations
Sean Kennedy [EMAIL PROTECTED] wrote: Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. I have noticed this too, especially when speaking to someone who is using a cell phone. I assume that VP is using silence suppression. -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.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] Re: VoicePulse OpenAccess
Thanks but I already tried their knowledgebasethey dont have any configs for OpenAccess only VoicePulse Connect. Also, tried calling their support and they were unable to assist in getting * working with OpenAccess only VoicePulse Connect. Does anyone have * working using SIP and VoicePulse Open Access? If so could you share your configs? From: Matthew Marlowe matthew.marlowe at gmail.com Subject: Re: VoicePulse OpenAccess Newsgroups: gmane.comp.telephony.pbx.asterisk.user Date: Mon, 20 Dec 2004 08:49:39 -0500 They have an entire knowledge base with example scripts, etc on there web site. You can also call them and reach someone in tech support during the day ___ 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] Re: VoicePulse OpenAccess
OpenAccess, according to the tech support rep I just got off the phone with who answered after only one ring said the setup would bethe exact same as VoicePulse Connect. - Original Message - From: Keith O'Brien To: asterisk-users@lists.digium.com Sent: Tuesday, December 21, 2004 12:44 PM Subject: [Asterisk-Users] Re: VoicePulse OpenAccess Thanks but I already tried their knowledgebase they dont have any configs for OpenAccess only VoicePulse Connect. Also, tried calling their support and they were unable to assist in getting * working with OpenAccess only VoicePulse Connect. Does anyone have * working using SIP and VoicePulse Open Access? If so could you share your configs? From: Matthew Marlowe matthew.marlowe at gmail.comSubject: Re: VoicePulse OpenAccessNewsgroups: gmane.comp.telephony.pbx.asterisk.userDate: Mon, 20 Dec 2004 08:49:39 -0500 They have an entire knowledge base with example scripts, etc on there web site. You can also call them and reach someone in tech support during the day ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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
[Asterisk-Users] Re: VoicePulse OpenAccess
I did finally get through to them. However, after troubleshooting for about ½ hour they were at a loss as to why it wasnt working. I sent them my sip.conf file and extensions.conf file and they didnt notice any problems. What you are saying doesnt make sense. VoicePulse Open Access uses SIP, VoicePulse Connect Uses IAX. How can the setup be the same if they are using entirely different protocols? I have working IAX configsI am looking for a working SIP config with VoicePulse. Their tech asked around and none of their engineers had an example sip.conf that works with their service. They also dont have a sip.conf file in their knowledgebase. If anyone has this working, can they share their sip.conf file?? Message: 14 Date: Tue, 21 Dec 2004 16:23:00 -0500 From: Matthew Marlowe [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: VoicePulse OpenAccess To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 OpenAccess, according to the tech support rep I just got off the phone with who answered after only one ring said the setup would be the exact same as VoicePulse Connect. ___ 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] Re: Voicepulse down for anyone else?
Message: 8 Date: Mon, 18 Oct 2004 14:32:07 -0400 From: Deon Rodden [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voicepulse down for anyone else? To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Can you give me more info on general issues across the net ? Yeah, VoicePulse seems to be having issues, it's usual though. I wish they weren't the only place I knew to get flat rate incoming DID's Nationally in the U.S from. In this case, it wasn't VoicePulse's problem. Level 3 is a huge provider of internet services, and as of this morning/last night, they experienced routing problems that affected sites all over america and europe. It was a mess. After a couple of hours, everything was running again, and should have taken care of all issues. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: VoicePulse changes
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Randy Bush Sent: Thursday, July 15, 2004 3:35 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: VoicePulse changes the message arrived here some hours after calls through them stopped working. not very professional. there should have been considerable, like multiple days, of overlap. think about the customers who are out of reach of configuring their * server but still rely on the service. The message says AUGUST 15th. This is JULY. AUGUST *follows* JULY concurrently/serially. Yeah...that's believable. Mike, wake upthey made changes, the broke things. While they obviously tried to give everyone a month's notice, it just didn't work out that way. While the old config still works (now), I find it difficult to believe that the two events were not related. Especially when it automagically fixed itself. Also like that I call in and get the automated status report, which reports everythig fine, yet still wait on hold for over 45 minutes, and find out that they actually already DO know that there is a problem. None of it struck me as particularly professional. If there is someone from Voicepulse here, feel free to stand up for youself and tell us your side of things. From here it's not looking too good. Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: VoicePulse changes
On Jul 16, 2004, at 7:03 AM, [EMAIL PROTECTED] wrote: Yeah...that's believable. Mike, wake upthey made changes, the broke things. While they obviously tried to give everyone a month's notice, it just didn't work out that way. While the old config still works (now), I find it difficult to believe that the two events were not related. Especially when it automagically fixed itself. For the record, it hasn't automagically fixed itself for everybody. My service is still broken, regardless of whether I use the new or old configuration. Multiple E-mails to [EMAIL PROTECTED] have, as of this moment, gone unanswered. As I'm on the road, I do not have the option of sitting on hold for 45 minutes. Very unprofessional. And now they're making me look bad. -fedl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: VoicePulse changes
Remco Barende wrote: On Thu, 15 Jul 2004, Chris Glover wrote: On Thu, 15 Jul 2004 [EMAIL PROTECTED] wrote: Note The previous method for terminating IAX2 calls using Connect! will cease to be available at midnight (GMT) on August 15th, 2004. The message I got was at 1:51 AM EST. That means I was given negative 5 hours and 51 minutes to make this change. How did you get -5 hours? I make it 31 days! :-) Different time zone ?? :) Maybe if you circle the globe enough times, crossing the international date line each time, of course, it would be possible to get to August 15th yesterday ;-) SRB ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: VoicePulse changes
+3, Funny -Original Message- Maybe if you circle the globe enough times, crossing the international date line each time, of course, it would be possible to get to August 15th yesterday ;-) SRB ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: VoicePulse changes
the message arrived here some hours after calls through them stopped working. not very professional. there should have been considerable, like multiple days, of overlap. think about the customers who are out of reach of configuring their * server but still rely on the service. randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: VoicePulse changes
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Randy Bush Sent: Thursday, July 15, 2004 3:35 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: VoicePulse changes the message arrived here some hours after calls through them stopped working. not very professional. there should have been considerable, like multiple days, of overlap. think about the customers who are out of reach of configuring their * server but still rely on the service. The message says AUGUST 15th. This is JULY. AUGUST *follows* JULY concurrently/serially. Your current outage is *unrelated* to the email you got. Mike :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicepulse Connection
That's what I'm trying to get at. *normally* you expect to dial 00 but
when you're using voicepulse, Asterisk needs to start all international
number with 011. Think of it this way, in VoicePulse's mind, you're
always dialing from the US. Of course the user will try dialing 00
because that's the 'normal' way to do it.
So what you have to do is change your dial plan to intercept a 00
prefix and reformat it using a 011 prefix, something like this:
exten = _00[1-9].,2,Dial,IAX2/[EMAIL PROTECTED]/011${EXTEN:2}
Message: 9
Date: Mon, 23 Feb 2004 07:49:07 -0700
From: Michael Welter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2879 - 10
msgs
Reply-To: [EMAIL PROTECTED]
In the UK it's 00 then the country code. So a call from the UK to my
phone would be 0013036742575.
Miie
Matt Lawson wrote:
I think international number dialed through voicepulse have to start
with 011... (even if you're located in another countery). I asked them
about that once, and that's what works for me (We've been dialing Spain
and Germany recently, but never Japan)
HTH,
Matt
-- __--__--
Message: 4
Date: Sat, 21 Feb 2004 09:04:43 -0300
From: Daniel Bichara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Voicepulse Connection
Reply-To: [EMAIL PROTECTED]
Hi,
I have two * connecteds and I wish a phone connected to * #1 calls
PSTN via Voicepulse connected to * #2, as follows:
telephone --- Asterisk #1 Asterisk #2 Voicepulse
When I dial 81-90... (japan), * #1 will route call to Voicepulse at
*#2. Everything works between #1 and #2 but when #2 calls Voicepulse I
get an error message:
-- Called user:[EMAIL PROTECTED]/[EMAIL PROTECTED]
Feb 21 10:01:34 WARNING[98311]: chan_iax2.c:4445 socket_read: Call
rejected by 66.234.228.132: No such context/extension
I am clueless!!! What could it be? Follow my confs...
Exten.conf - *#1
exten = _81.,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
# Exten.conf - *#2
[outvoicepulse]
exten = _.,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = _.,2,Congestion
# Iax.conf - *#2
[voicepulse]
context=VPWS
secret=password
auth=md5
type=friend
host=66.234.228.132
disallow=all
allow=speex
allow=gsm
jitterbuffer=no
Daniel
-- __--__--
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Re: [Asterisk-Users] Re: Voicepulse
Atually with the root servers dropping their domain name announcement nothing would have helped. Well, except for hard codeing the IP rather than using fqdn in the config. Or making a static entry in the local hosts file ( both have it's issues) I prefer to use IP rather than fqdns when possible. But that can introduce other problems if the host system decides to move you to another host machine by just changing the DNS name. Using fqdns in mission critical applications is not a good idea IMHO, it just adds another layer of something that can go wrong. Just my $.02 worth ;) -b Quoting Chris Albertson [EMAIL PROTECTED]: --- Steve Sobol [EMAIL PROTECTED] wrote: Matt Lawson wrote: I was just about to write the same thing. It says busy. Is is REALLY busy or is something else wrong? This on the heels of switch-1.nufone.net being missing out of DNS. We have customers that expect their VOIP to work. Is there anybody that's reliable? I've been doing some testing and so far I'm not 100% impressed by the VOIP services I've seen. They provide a good service but my local phone company and ATT longdistance service is more reliable. But this is not to say _you_ can't built a reliable VOIP based system. Get _two_ providers and set up your dial plan in extensions.conf to fail over if one service fails to connect to dial via the next one and finally if both fail use pstn. your users will see a system the just works. About Nufone's problem. I bet they'll start thinking about getting a backup DNS service and maybe geographic deversity. A company should be able to even stay on the air if there is a server room fire using techniques like round robin DNS and West cost and East coast servers run by different, unrelated hosting companies. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voicepulse
On Fri, 16 Jan 2004, Bill Hamel wrote: Using fqdns in mission critical applications is not a good idea IMHO, it just adds another layer of something that can go wrong. Or, it just changes the type of things that can go wrong. In Australia, the recently-introduced number portability infrastructure for mobile phones means that every call to a mobile number must be routed through a 'central' directory[1] instead of being sent straight to the right network based on the number prefix. Sure, this is one more thing that can go wrong, but one of the benefits is that you can now switch between networks without having to change your phone number. I mention this because it's the same (well, similar) tradeoff between using DNS and IP addressing. Will your communications partners *guarantee* that they will *never* change the IP addresses of their servers? If they do change, can you guarantee that you can make all the configuration changes in all of your servers that communicate with theirs in a timely-enough fashion to meet service levels? I'm not suggesting that the DNS response to this is entirely flawless, but it does reduce administration overhead for adds/moves/changes -- especially between different organisations. Also, using IP address *may* cut you out of load-balancing or disaster recovery processes that your partners may make to ensure that you get good service from them. I'm not trying to say that IP addressing is a bad idea. You just need to assess the risks of using either method. Cheers, Vic Cross [1] This is a drastic oversipmlification -- it's actually much worse than one single directory as I understand it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voicepulse
On Tue, 2004-01-13 at 19:42, Chris Albertson wrote: --- Steve Sobol [EMAIL PROTECTED] wrote: Matt Lawson wrote: I was just about to write the same thing. It says busy. Is is REALLY busy or is something else wrong? This on the heels of switch-1.nufone.net being missing out of DNS. We have customers that expect their VOIP to work. Is there anybody that's reliable? I've been doing some testing and so far I'm not 100% impressed by the VOIP services I've seen. They provide a good service but my local phone company and ATT longdistance service is more reliable. That would be a big DUH! Now the question comes down to choice and price. But this is not to say _you_ can't built a reliable VOIP based system. Get _two_ providers and set up your dial plan in extensions.conf to fail over if one service fails to connect to dial via the next one and finally if both fail use pstn. your users will see a system the just works. About Nufone's problem. I bet they'll start thinking about getting a backup DNS service and maybe geographic deversity. A company should be able to even stay on the air if there is a server room fire using techniques like round robin DNS and West cost and East coast servers run by different, unrelated hosting companies. Of course had you paid attention to the problem you would have been able to understand that no DNS arrangement would fix having the root servers modified by a registrar who screwed up. DNS servers don't work if your whois doesn't point to the proper places. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicepulse
I was just about to write the same thing. It says busy. Is is REALLY busy or is something else wrong? This on the heels of switch-1.nufone.net being missing out of DNS. We have customers that expect their VOIP to work. Is there anybody that's reliable? I am having probelms connecting to voicepulse this morning. Is anybody else having issues.. burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voicepulse
Matt Lawson wrote: I was just about to write the same thing. It says busy. Is is REALLY busy or is something else wrong? This on the heels of switch-1.nufone.net being missing out of DNS. We have customers that expect their VOIP to work. Is there anybody that's reliable? If you'd read the other messages on the topic, you would see that the Nufone failure *wasn't* Nufone's fault, it was a payment that was not applied correctly by the registrar. It could have happened to anyone. The fact that someone over whom Nufone has no control screwed up their DNS doesn't mean they're not reliable. -- JustThe.net Internet New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Geek In Charge * 888.480.4NET (4638) * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voicepulse
--- Steve Sobol [EMAIL PROTECTED] wrote: Matt Lawson wrote: I was just about to write the same thing. It says busy. Is is REALLY busy or is something else wrong? This on the heels of switch-1.nufone.net being missing out of DNS. We have customers that expect their VOIP to work. Is there anybody that's reliable? I've been doing some testing and so far I'm not 100% impressed by the VOIP services I've seen. They provide a good service but my local phone company and ATT longdistance service is more reliable. But this is not to say _you_ can't built a reliable VOIP based system. Get _two_ providers and set up your dial plan in extensions.conf to fail over if one service fails to connect to dial via the next one and finally if both fail use pstn. your users will see a system the just works. About Nufone's problem. I bet they'll start thinking about getting a backup DNS service and maybe geographic deversity. A company should be able to even stay on the air if there is a server room fire using techniques like round robin DNS and West cost and East coast servers run by different, unrelated hosting companies. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
