similar to: bug? asterisk -rx "show dialplan default"

Could someone explain to a non-US dummy the following phrases I have seen on the list. "I can provide you with tier 1 termination 6/6. I can blend or NPANXX breakout." "We provide US48 termination, blended rate for 1 MOU and above is .008 with 6/6." What is 6/6? What is US48? What is blended? What is MOU? What is NPANXX breakout? -------------- next part --------------

One of my coworkers needs to get a softphone set up to my Asterisk system; he's a Linux user, so it looks like about the only IAX2 option is iaxComm. For ease of use (he'll be using this a fair bit), I'm recommending that he get a USB handset; I'm just having trouble finding any US retailers for them. :) Could someone recommend a USB handset that's compatible with iaxComm

I've a call center using IAX softphones provided by a third party. We've observed problems where the IAX phones seem unable to use our PRI trunks. A sample anonymized call is provided below with the PRI debug calls embedded. Any thoughts, comments or suggestions would be welcome. In anonymizing it, I preseved the format and number of digits sent. -- Accepting AUTHENTICATED

Hello, Is there any possibility of integrating plans of vonage with asterisk. Regards Vijay Gandhi

I have ASTCC installed, and compare it with NuFone, however, I find that the billing of NuFone is always a few secondes more (6 to 24 seconds) Does anybody has an explanation / solution for it? bye Ronald

hello list, I subscribe to Sixtel.net for a DID just to see how it worked. they say DID active inmediatly , but after 4 days , I have no DID , I tried to call tech support office and it seems they all ways are in hollidays, a support ticket no good luck, I call back service also not good luck. so anyone here has experience with them? are they a SCAM? fortunally only put there $10 bucks but

Hi, guys. May be someone could give me advise? I am trying to find good VOIP provider ONLY for OUTGOING calls with low per channel cost and cheap rates on Eastern Europe, Turky and xUSSR. Should support g729 or g723 codecs, SIP or IAX connectivity. -- ========================================================================= = Best regards, Nikolay Pavlov.

Chaps, I recently added an incoming VOIP account to my asterisk box. When the PSTN number associated with this account is dialled, the call rings once and then asterisk starts playing music on hold, even though all the extensions continue to ring. Variations of answer() and ringing() don't seem to help. I'm sure I'm missing something spectacularly obvious, but the wiki and googling the

Hello, I'm trying to use MySQL for Dialplans and have followed the Asterisk RealTime Extensions setup. The MySQL table is called "extensions" and I have entered two records.. ext 1000 and 2000. I also added switch => Realtime/default@extentions in extensions.conf and extensions => mysql,asterisk,extensions in extconfig.conf I do a *CLI> dialplan reload but when I show

Hi all, I'm using asterisk in Realtime mode and all my dialplans are defined in extensions table...My problem is the following... If I have for a specific context an extension with value _X. and another entry for the same context with extension 123456, if an incoming call from the same context comes to asterisk to extension 123456 the _X. is matched first...is there a way to force asterisk to

Are there any standards for setting up pbx dialplans for businesses/offices? What I mean is that, which numbers are reserved for a specific use ex. 0 for operator ? Putting Zero for operator in the dialplan seems to be the common practice of businesses. If there is such a standard, * and # are used for what ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've built the latest CVS of asterisk -- not the zaptel or libpri directories, just the asterisk directory. asterisk installs successfully, but there are severe problems. I built this system in the past and ran it, but now building it again fails. This is the CVS as of this morning, 2003-06-13, but I had problems on 06-11/12 as well. After make; make install; make samples; make config, I

I've got soft phone that allows me to dial SIP URI's. I'd like to route these calls through a provider to be completed, because I'm beind a NAT box and doing it directly doesn't work. Right now I've got an extension defined like this: Dial(IAX2/${FWDUSERID}:${FWDPASSWD}@${FWDSERVER}/**356<username>) This will connect a call to FWD and call a user at FWD. It works

After John Todd's talk at Astricon about the ISN project, I spent much of the weekend playing around with it. I have discovered that the default dialplans on my Sipura gear, as well as my Grandstream phones, intercept the "*" key that is a required part of ISN numbers and interpret it as a "metacharacter." Googling for a while has turned up evidence that this can be

Hello I'm learning more about dialplans and have a couple of questions: 1. Am I right in understanding that the actions that can be performed in extensions.conf can be of two types only: - internal commands (Dial, Wait, etc.) - calls to external scripts throught AGI? 2. I'd rather write scripts in Python instead of Perl or PHP. Does someone have a skeleton that I could use to build one,

Due to changes at the office, I'm finally getting around to setting up an AA to deal with incoming calls. One of the big changes is that we're dropping the old alphanumeric pager and will just send pages to our phones. I've got the outbound greeting message working in a test context no problem right now, but I'm kind of stuck on how to capture a DTMF sequence from a user and

I went to put in the new sound files over the weekend, but forgot to backup the custom folder and lost my custom digital receptionist files. I then had to copy the old files back from a duplicate machine. The problem is now though that voicemail just hangs up when I dial it. Other apps work - *70 for example gives me 'call waiting...activated' so I know it's accessing the files

On 06.11.2015 14:06, Rusty Newton wrote: > > It should be able to handle typical regular expression. I don't see > anything wrong with what you are doing. Please file a bug at > issues.asterisk.org/jira. Do include a debug log on the issue > captured when Asterisk attempts to execute these extensions. > >

I have a new Asterisk system going into production next week and I'm a bit stumped as to the best way to handle the Dialplans for it. The Asterisk system is replacing 4 separate PSTN lines with both SIP & PSTN inputs. The setting up of the dial plan is giving me some design headaches, which probably means I'm missing something obvious and doing this the hard way. I have separate

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, First of all, hello again! Been a while since I've posted to the list, but I've been here lurking and watching ;-) Anyway, I wanted to pose a general question to the list to see if it turns up new suggestions for everyone/me. What is your preferred editor when coding in the dialplan? This is mainly aimed at those of you who write