Hello, I am just curious how many people are hooked up with BroadVoice and have recently been experiencing a lot of dificulty. Joel
I've been using it for 3 months now, it's been up rock solid for about 3
weeks now, the only major complaint I have is the VoiceMail dropping problem
and that's *'s fault, not BroadVoice's
-Chris
I am hooked up with broadvoice and have been having no problems that are major there voice mail system went on the blits for about 30 minutes yesterday but that was about it. what kind of problems you expierencing? ----- Original Message ----- From: "Joel Gathercole" <joel-lists@netrique.biz> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 11, 2004 9:19 PM Subject: [Asterisk-Users] Broadvoice> Hello, > > I am just curious how many people are hooked up with BroadVoice and have > recently been experiencing a lot of dificulty. > > Joel > _______________________________________________ > 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---------------------------------------- My Inbox is protected by SPAMfighter 1117 spam mails have been blocked so far. Download free www.spamfighter.com today!
We just got setup with Broadvoice yesterday for LD. This isn't something I REALLY need (No local numbers avail so we just got a Houston number), but I'm just curious. I can make outbound calls to Broadvoice and they work great, but I can't do inbound. I have bv's voicemail turned off so all I get is a busy signal when I call our bv number. I've tried this with both type=peer and type=friend and I get the same results, any ideas? context=default recordhistory=yes realm=angelinacounty.net port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw dtmfmode=inband tos=reliability register => 7134810061:[password]@sip.broadvoice.com [Broadvoice] type=friend username=7134810061 fromuser=7134810061 secret=[password] host=sip.broadvoice.com context=inbound-pots fromdomain=sip.broadvoice.com nat=yes canreinvite=no dtmfmode=inband -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041023/a22b3283/attachment.htm
Anybody else having Broadvoice registration problems today? -- Gary White admin@netpathway.com Network Administrator Internet Pathway 105 D East Church Street Voice: 601-776-3355 P. O. Box 777 Fax: 601-776-2314 Quitman, MS 39355 Registered Linux User Number 198875 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3182 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041113/a010a4c0/smime.bin
Its working here, some issues tho. All outbound calls have no CID. -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Bruce Komito Sent: Saturday, November 13, 2004 1:16 PM To: Doug Shubert Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BroadVoice Same here... Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sat, 13 Nov 2004, Doug Shubert wrote:> yes.. started around 12:00 noon EST > I get "sip_reg_timeout: Registration for'xxxxxxxxxx@sip.broadvoice.com"> > Does anyone know if this is related to the channels patch? > > Doug > > > Gary White (Network Administrator) wrote: > > > Anybody else having Broadvoice registration problems today? > > > >------------------------------------------------------------------------> > > >_______________________________________________ > >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 > > > > > This message has been categorized as "Legitimate" by BayesianAnalyzer.> If you do not agree, please click on the link below to train theAnalyzer.>http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-1 1-13%5Ccf62fbdc4a664e39b123d2ef9ce2d9a4&C=2> > -- >-----------------------------------------------------------------------> This message has been inspected by DynaComm i:mail >----------------------------------------------------------------------->_______________________________________________ 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
Yes. :-( -jeff -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gary White (Network Administrator) Sent: Saturday, November 13, 2004 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] BroadVoice Anybody else having Broadvoice registration problems today? -- Gary White admin@netpathway.com Network Administrator Internet Pathway 105 D East Church Street Voice: 601-776-3355 P. O. Box 777 Fax: 601-776-2314 Quitman, MS 39355 Registered Linux User Number 198875
Well, back working now. Guess they were having problems again. -- Gary White admin@netpathway.com Network Administrator Internet Pathway 105 D East Church Street Voice: 601-776-3355 P. O. Box 777 Fax: 601-776-2314 Quitman, MS 39355 Registered Linux User Number 198875 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3182 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041113/1747ca48/smime.bin
By the way this was not related to the patch. I installed it Friday and did not start having trouble until today.> Well, back working now. Guess they were having problems again. > >------------------------------------------------------------------------ > >_______________________________________________ >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 >-- Gary White admin@netpathway.com Network Administrator Internet Pathway 105 D East Church Street Voice: 601-776-3355 P. O. Box 777 Fax: 601-776-2314 Quitman, MS 39355 Registered Linux User Number 198875 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3182 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041113/2203e0c6/smime.bin
Anybody else having broadvoice problems?
-- Executing SetAccount("SIP/101-d03b", "LD") in new
stack
-- Executing Dial("SIP/101-d03b",
"SIP/18004321000@Broadvoice") in
new stack
-- Called 18004321000@Broadvoice
-- Got SIP response 408 "Request Timeout" back from 147.135.0.128
== No one is available to answer at this time
Tim Jackson
Network Engineer
Angelina County, Texas
(936)639-4827x101 office
(936)414-6723 mobile
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/SNIP/ My BroadVoice account has been down for over a week with neither an explanation nor a service credit. Our problems may be a little different though because I don't remember what happened when I tried to dial out. I know that I do get a "Request Timeout" error while trying to register though. Anybody else having broadvoice problems? /SNIP/ This is what happens to the VOIP Industry over time - Consolidation, if you want to call it that. A few players will remain at the end and when it is all over. All others will just disappear. Broadvoice has not understood the game it appears. This game is like Heavy Weight Boxing, where the last one standing is the winner. In order to be the last one to be standing, you have to be lean and mean and control your costs and keep acquiring customers, till you reach a critical mass and a value proposition for another investor or a predator. Anyone who has not understood these couple of things will fall by the wayside. Seshu Kanuri 732-213-2422 http://ipphone.eezeephone.com -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited.
> are they really /unlimited/ in the truest sense of the word ? > US$24.95, even if it's only for unlimited calls to Malaysia > (where i am) seems very, very attractive. when something is this > attractive, i start looking for the catch.AFAIK, no one offers truly unlimited service. Companies differ greatly in openness of the contractual details. I subscribe to "unlimited" (POTS domestic US) long distance from SBC. The contract clearly states that monthly usage exceeding 5000 minutes is billed at $0.04 per minute. Not cheap, but it won't break you if go a little over. At the other extreme, there are many horror stories of Vonage customers whose service was terminated, without warning, for excessive usage. Broadvoice appears to be somewhere in between. I am considering their service, and called them to ask about allowed usage. They would not disclose their limits, but when I mentioned that my calls typically run 2500-3000 minutes per month, mostly to the US, they said that this was well below their "alarm" levels. There may be a technical problem with Broadvoice for your application. I suspect that all calls proxy the media stream through their server (in the US). Perhaps a Broadvoice customer can confirm or deny this. If that's the case, the roundtrip delay on your calls to Malaysia will include *four* hops across the Pacific (~400 milliseconds). If there's any echo, it will be very disconcerting. Even if not, you'll have problems when both parties start talking at about the same time. You can use their free trial offer to see if the delay is bothersome. --Stewart
anyone having problems with broadvoice? all proxies down,site down ...
im connected fine, hard code this in your /etc/hosts and try again 147.135.4.128 sip.broadvoice.com sip On Mon, 2004-12-20 at 18:23 -0200, tecepe wrote:> anyone having problems with broadvoice? > all proxies down,site down ... > _______________________________________________ > 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 >-- skamp <skamp@arkayinc.com>
Did somebody connect Asterisk to BroadVoice provider? If so, can you share instruction with me? Thanks.
Broadvoice has instructions on their site on how to configure asterisk with their service, and it works i use broadvoice with asterisk On Tue, 2005-01-11 at 10:43 -0500, Vitalie Apostu wrote:> Did somebody connect Asterisk to BroadVoice provider? If so, can you share > instruction with me? > > Thanks. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- skamp <skamp@arkayinc.com>
Can you give me example of sip.conf and extention.conf which work with broadvoice? I want users who registered with Messenger through sip to be able to make a call thought broadvoice. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of skamp Sent: Tuesday, January 11, 2005 10:55 AM To: vitalie.apostu@compuflex.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BroadVoice Broadvoice has instructions on their site on how to configure asterisk with their service, and it works i use broadvoice with asterisk On Tue, 2005-01-11 at 10:43 -0500, Vitalie Apostu wrote:> Did somebody connect Asterisk to BroadVoice provider? If so, can you > share instruction with me? > > Thanks. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- skamp <skamp@arkayinc.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
> Can you give me example of sip.conf and extention.conf which > work with broadvoice? I want users who registered with > Messenger through sip to be able to make a call thought broadvoice.I posted this just a few days ago: http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm l -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeel<at>jafferali.net
Following links says: HTTP 404 - File not found . Is it a right link http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm ? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nabeel Jafferali Sent: Tuesday, January 11, 2005 11:09 AM To: vitalie.apostu@compuflex.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] BroadVoice> Can you give me example of sip.conf and extention.conf which work with > broadvoice? I want users who registered with Messenger through sip to > be able to make a call thought broadvoice.I posted this just a few days ago: http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm l -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeel<at>jafferali.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
On Tue, 2005-01-11 at 10:59 -0500, Vitalie Apostu wrote:> Can you give me example of sip.conf and extention.conf which work with > broadvoice? I want users who registered with Messenger through sip to be > able to make a call thought broadvoice.* for BV setup guide: http://www.broadvoice.com/support_install_asterisk.html Except use this patch instead of the one they link to: http://edvina.net/broadvoice/broadvoicesip2.txt Cheers, Mike
Hi, I'm currently routing my asterisk server out over broadvoice.. it seems I can do multiple outgoing and incoming calls.... does anyone know if broadvoice actually allows this or not?
I believe Kevin is meaning to buy a T-1 PRI from someone, like a phone company of a CLEC. On Apr 4, 2005 3:40 PM, John Millican <john@millican.us> wrote:> On Monday April 04 2005 3:58 pm, Kevin Kiely wrote: > > T1 PRI > > > > > This brings up the question. What is the best service for concurrent > > calls? > > In the case where I have a small business I might have 10-15 people > > needing > > to call out and they could all be on at the same time. > > -Scott > Even with a T-1 you still need some one to provide termination that will allow > more than one call at a time on that account or multiple accounts with the > same or different providers. > John M > _______________________________________________ > 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 >
On Monday April 04 2005 5:14 pm, Brian McSpadden top posted:> I believe Kevin is meaning to buy a T-1 PRI from someone, like a phone > company of a CLEC. > > On Apr 4, 2005 3:40 PM, John Millican <john@millican.us> wrote: > > On Monday April 04 2005 3:58 pm, Kevin Kiely wrote: > > > T1 PRI > > > > > > > > > This brings up the question. What is the best service for concurrent > > > calls? > > > In the case where I have a small business I might have 10-15 people > > > needing > > > to call out and they could all be on at the same time. > > > -Scott > > > > Even with a T-1 you still need some one to provide termination that will > > allow more than one call at a time on that account or multiple accounts > > with the same or different providers. > > John MWell I canget T-1 from a local provider (since i live in BFE it is much cheaper than att, verizon,...) but they do not provide termination. so just wanted to clarify this for the op. john M
So then in this case voip would not be the best solution? ----- Original Message ----- From: "John Millican" <john@millican.us> To: <asterisk-users@lists.digium.com> Sent: Monday, April 04, 2005 2:41 PM Subject: Re: [Asterisk-Users] Concurrent calls: best provider?> On Monday April 04 2005 5:14 pm, Brian McSpadden top posted: > > I believe Kevin is meaning to buy a T-1 PRI from someone, like a phone > > company of a CLEC. > > > > On Apr 4, 2005 3:40 PM, John Millican <john@millican.us> wrote: > > > On Monday April 04 2005 3:58 pm, Kevin Kiely wrote: > > > > T1 PRI > > > > > > > > > > > > This brings up the question. What is the best service for concurrent > > > > calls? > > > > In the case where I have a small business I might have 10-15 people > > > > needing > > > > to call out and they could all be on at the same time. > > > > -Scott > > > > > > Even with a T-1 you still need some one to provide termination thatwill> > > allow more than one call at a time on that account or multipleaccounts> > > with the same or different providers. > > > John M > > Well I canget T-1 from a local provider (since i live in BFE it is much > cheaper than att, verizon,...) but they do not provide termination. sojust> wanted to clarify this for the op. > john M > _______________________________________________ > 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
Thanks for the feedback Don, It did change the behavior, but not realy fix the problem. Now what happens is that it rings 6 times and then goes into broadvoice voicemail. I do not hear any rings on my softphone, or see anything in debug on the console. I would love to see your configs to figure out what is going on. Or any other suggestions. Thanks much Craig Don Nightingale wrote:>This was a PITA to figure out. Broadvoice sends the phone# as the extension >on incoming calls, modify your extensions.conf: > > >exten => <YOURBVPHONENUM>,1,Dial(SIP/500,60,tr) >exten => <YOURBVPHONENUM>,2,hangup > >It should then do exactly what you want. I have almost the exact setup you >do, but I run the latest CVS head since it's just a toy for me right now. >If it doesn't work, let me know and I'll send you my sip.conf and >extensions.conf for comparison. > > > >
I might be making some progress here. I still am not getting a ring on my test softphone, however I was tailing the log when I was working through some successive config tries and noticed the following log entries Apr 7 02:21:02 DEBUG[2264]: Registration successful Apr 7 02:21:02 DEBUG[2264]: Cancelling timeout 1489 Apr 7 02:21:14 DEBUG[2264]: Stopping retransmission on '108455f96cea220820ae34ef0390720e@192.168.1.11' of Request 102: Found Apr 7 02:21:14 DEBUG[2264]: Stopping retransmission on '11a3e1607215505579a54d3277c62c11@192.168.1.11' of Request 102: Found Apr 7 02:21:18 DEBUG[2264]: Setting NAT on RTP to 4 Apr 7 02:21:18 DEBUG[2264]: Scheduled a registration timeout # 1498 Apr 7 02:21:18 DEBUG[2264]: Stopping retransmission on '43308e245f3efc5c3b486ae0577f7348@sip.broadvoice.com' of Request 106: Found Apr 7 02:21:18 DEBUG[2264]: Registration successful Apr 7 02:21:18 DEBUG[2264]: Cancelling timeout 1498 Apr 7 02:21:32 DEBUG[2264]: Setting NAT on RTP to 4 Apr 7 02:21:32 DEBUG[2264]: Check for res for 9255582025 *Apr 7 02:21:32 DEBUG[2264]: 9251234567 is not a local user Apr 7 02:21:32 DEBUG[2264]: 9251234567 is not a local user* Apr 7 02:21:32 DEBUG[2264]: Stopping retransmission on 'SD50bl801-97575a6b965b7ef4809e5e5a9bb3e199-js1h002' of Response 229181545: Found Apr 7 02:21:34 DEBUG[2264]: Setting NAT on RTP to 4 Apr 7 02:21:34 DEBUG[2264]: Scheduled a registration timeout # 1502 Apr 7 02:21:34 DEBUG[2264]: Stopping retransmission on '43308e245f3efc5c3b486ae0577f7348@sip.broadvoice.com' of Request 107: Found Apr 7 02:21:34 DEBUG[2264]: Registration successful Apr 7 02:21:34 DEBUG[2264]: Cancelling timeout 1502 This number is my VOIP number that I am calling, so asterisk is picking the call up. It just seems to not do anything with it like answer. Any thoughts? Thanks Craig -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050407/67b35bde/attachment.htm