Re: [Asterisk-Users] VoicpulseConnect problems?
I was having problems with voicepulse about a week or two ago... Incoming calls would fail, and one incoming call, would block all outgoing calls. Then one day, the DTMF tones stopped working. I could call into Asterisk, but I could not navigate because my tones were being ignored. I ran debug, and Asterisk was not seeing the tones. I emailed the voicepulse folks, and they fixed it the next day. It was like they knew the bug, and reset whatever needed fixing. Their reply was terse, but the fix was good. Aside from this event, voicepulse has been very good for me! Try to email their support Folks, I'm having trouble with my voicepulse numbers. Over the past week, incoming calls have been very slow to be answered, but they seem fine while the call is in progress. When the caller hangs up, asterisk takes a while (over 2 minutes in some cases). This system does not make outgoing calls. Today, after rebooting my machine and rotating the log files, I have absolutely NO incoming calls being received. My cell phone dials the number, tells me it's connected, and then happily hangs up 10 to 12 seconds later, while asterisk (and the logs) show no indication at all of any incoming calls. Looking at my syslog and asterisk messages, the only thing I'm seeing over the past week that did not use to happen is this message in the asterisk logs: Apr 28 10:06:45 WARNING[4282]: Host 'gwiax-in-01.voicepulse.com' not found at line 72 But that's been happening for about the same time as the slow-down issue, and still calls _were_ being answered, albeit slowly. I'm HoSed. :) Has anyone else run into this? Got any ideas on what's up at VPConnect? Do I need to placate the rain-god or something? Any help would be appreciated! Thanks, Maya __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 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] 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
Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage
I made the Vonage mistake too. Cost me a hundred bucks! They will hit you with a Disconnect Fee, if you don't return their equipment in the original box. Sux, if you ask me. Vonage does offer an attractive flat rate for 500 minutes. You could, purchase a digium card, and plug the Vonage POTS line into the digium card, just like you would a POTS line. Kinda a workaround, but I tried it and it works. Or you can buy the softfone addon, and pay more per month to Vonage and settle for SIP... Or better yet, try: http://connect.voicepulse.com You get free incoming calls. A low monthly rate, and a cheap per minute rate for outgoing. Best part of all, is you get trunking.. 4 simultaneous calls in and out. Which is what I think you said you were looking for. Check the list archive. This thread just happened a couple days ago. A couple that I remember as supporting IAX: voipjet.com nufone.net You also might try opbx.com JD Austin wrote: Im a newbie to this list (joined today). Other than Broadvoice, what voip providers work well with Asterisk? I'd like a service that will allow trunking so that I can have more than one outbound/inbound call if possible. JD Kerry Garrison wrote: You arent going to make this happen as you describe. Vonage is not a good service to use with Asterisk. To quote from the Wiki: Vonage service is locked to the ATA they send you. It is not possible to connect Asterisk (or any other SIP UA) directly to your main Vonage service. http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage If you want to use Asterisk, you will need a different provider. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J W Sent: Tuesday, March 22, 2005 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage I just installed Asterisk on my server and I have Vonage softphone. I need my Asterisk server to receive calls through the Vonage Softphone DID and make outgoing calls through the Vonage ATA using an X100p to connect to it. Can someone help me out on configuring this? I really need this for my business and would greatly appreciate the help. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ 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 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 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] How to number extensions - Which way is best?
Combing thru the Wiki, I did find that Asterisk does have some secret sauce with respect to sorting out what the caller is dialing... It is covered in the wiki page Asterisk Extension Mapping http://www.voip-info.org/tiki-index.php?page=Asterisk+Extension+Matching This page does an excellent job of explaining the intuitive logic that Asterisk uses to determine when you are dialing an extension or dialing an outside number. To followup on the excellent suggestions thus far, I have concluded... 1) For an office environment, where dialing 9 for an outside line is not uncommon, the dial 9 option may make sense for a cleaner implementation. 2) For a home (or small office) environment, where nobody expects to have to dial a 9, it is better to rely on the Asterisk Extension Mapping logic. PLUS... Numbering your extensions in the 100 to 119 range (or for larger environments 1000 to 1199) will provide the cleanest interface. This is because a leading 1 indicates a long distance call, and the number following a leading 1 cannot be a 0 or a 1 for long distance. Therefore, asterisk can determine with the second digit dialed that you are dialing an extension, and not a long distance number. Anyone have any comments on the above suggestions? If this topic has been discussed already, please point me to it. I have looked, and I don't see any discussion in the past couple of months or in the wiki. When setting up your inside extensions, it can be helpful to choose the numbering carefully. Ideally, you would like it to not conflict with the dialing of an outside number. For example, the extension... 1212 Is very similar to dialing a new york long distance number: 1-212-555.1212 The phone company intentionally avoids some number combinations. Local numbers never start with a 1. This way, the Phone company switch knows that any number that starts with a 1 needs long distance routing. -- Some switch systems avoid this issue by requiring that extensions dial a 9 to get an outside number. This allows the use of any extension numbers internally, as long as they do not begin with a 9! I can setup asterisk to work this way. -- What would be a wise choice for your extension numbering if you were just setting up a new system? Is there a better to use set of extensions, that avoids confusion with dialing of external numbers. -- Or perhaps, I am way off base. Does asterisk have some magic sauce that makes this a moot issue? Thanks in advance. ___ 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] Group=????
I had a similar dilema just a day ago.
I wanted to know which number a user had dialed, when they entered my
system. NOT their number, the number they dialed.
It appears that Asterisk can easily know the number that was dialed, but
does not record it into any variables, at least not reliably.
The trick is to setup a pattern that matches the inbound number.
If the number dialed is: 212-555-1212
Then a pattern of:
exten = 2125551212,1,Macro(housenum)
Will ID that the call came in on that number.
Then it is up to you to set a variable in the macro housenum
that will tell you later, that this call belongs to group 1.
I hope this helps...
Good Day list,
I have worn out my google toolbar today looking for a way to
determine which group an incoming call belong to, but have not been very
fruitfull in my endeavors.
I am trying to figure a way to determine which group the
incoming call that I just answered is part of so I can do some
branchinbg scripts in my dialplan. I know I can control my dialplan
logic by creating separate context's for each channel type, however, I
have some things that would be easier if I could just determine which
group= setting the incoming call is on.
Example Zapata.conf:
context=from-pstn
signalling=fxs_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=0
channel=3
context=from-pstn
signalling=fxs_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=1
channel=4
I would like to know if there is a variable or method to have
set (For example $GROUP) such that
Exten s,1,NoOp(${GROUP})
Would display 0 if the call came in on channel 3 of my
card
And would display 1 if the call came in on channel 4
of my card.
Thanks for your time in this issue.
Ron
___
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] How to number extensions - Which way is best?
If this topic has been discussed already, please point me to it. I have looked, and I don't see any discussion in the past couple of months or in the wiki. When setting up your inside extensions, it can be helpful to choose the numbering carefully. Ideally, you would like it to not conflict with the dialing of an outside number. For example, the extension... 1212 Is very similar to dialing a new york long distance number: 1-212-555.1212 The phone company intentionally avoids some number combinations. Local numbers never start with a 1. This way, the Phone company switch knows that any number that starts with a 1 needs long distance routing. -- Some switch systems avoid this issue by requiring that extensions dial a 9 to get an outside number. This allows the use of any extension numbers internally, as long as they do not begin with a 9! I can setup asterisk to work this way. -- What would be a wise choice for your extension numbering if you were just setting up a new system? Is there a better to use set of extensions, that avoids confusion with dialing of external numbers. -- Or perhaps, I am way off base. Does asterisk have some magic sauce that makes this a moot issue? Thanks in advance. ___ 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] Voicepulse DNID is blank - Any other options?
I just signed up for a second voicepulse number. I assumed that I would be able to differentiate which number the caller dialed. But DNID is empty and DIALEDPEERNUMBER is empty and CHANNEL contains the same info (almost, with the exception of a randomly assigned suffix) for both numbers. Does anyone know how I might determine which number was called? Note, this is not CALLERID. I need the number that the caller CALLED. As a last resort, I guess I could use a different provider for the second number. Can anyone shed any light? Thanks in advance! ___ 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] Voicepulse DNID is blank - Any other options?
Ah.. the obvious. I don't know why I missed it. I am just a newbie at this PBX stuff. Thanks for the pointer. It worked. First off. Hopefully, someday soon, I will contribute more than silly questions to this list! Thanks again! Maybe I am missing your exact point, but what about handling this in your extensions.conf [voicepulse-incoming] exten = 2124007999,1,Goto(nyc,s,1) exten = 2124007998,1,Goto(nyc2,s,1) That will put calls to 2124007999 into context nyc and calls to 2124007998 into context nyc2. I guess the real questions is what is your ultimate goal? On Sun, 6 Feb 2005 12:52:21 -0800 (PST), Tim Burt [EMAIL PROTECTED] wrote: I just signed up for a second voicepulse number. I assumed that I would be able to differentiate which number the caller dialed. But DNID is empty and DIALEDPEERNUMBER is empty and CHANNEL contains the same info (almost, with the exception of a randomly assigned suffix) for both numbers. Does anyone know how I might determine which number was called? Note, this is not CALLERID. I need the number that the caller CALLED. As a last resort, I guess I could use a different provider for the second number. Can anyone shed any light? Thanks in advance! ___ 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 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
