similar to: asterisk-users Digest, Vol 83, Issue 3

Hello, this is Jesse with Webformix. We are managing an Asterisk installation for residential VOIP service, and we are having a problem where all inbound calls to DIDs which are assigned to us by our wholesaler but not yet assigned to a downstream customer get caught in a routing loop. ** For example: we get assigned DID block 1230-1239 and only 1230-1233 are given to our customers, then our

> -----Original Message----- > From: john lawler [mailto:maillist@tgice.com] > Sent: Wednesday, January 07, 2004 1:38 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] yet another question on DID trunks > > Hey Steven, > > Sorry to bother you yet again w/ a question on my seemingly endless > quest to get DID trunks setup for a customer. > >

Okay, first a little background - I've been with Packet8 since a month after they started. I found that we were outgrowing their services and decided to move to an asterisk box in the office. I found a service provider that offered me a reasonable rate. After a fair ammount of testing I decided to stick with their services and port my 3 primary DID's from Packet8 to the new service.

Well I know from personal experience that NuFone is working on a solution for its customers as fast as it can. I know they found an alternate termination provider and are working to have a solution for the TF and Local DID's he currently has on his platform. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

Greetings, I'd like to thank everyone that has responded to my original email. I have received information from several companies, and will be testing several of them. I also would like to update a statement from my original message to clarify it: >My strikelist: nufone, voicepulse, iax/sixtel The strikelist is just a list of carriers that didn't meet the needs a resonable

Hi, This is what I want to do: 1. Asterisk to answer calls via DID's, currently using SIPGATE 2. Provide a menu, and allow users to dial out. 3. According to the country and area they dial, the call should connect via one of up 4 carriers depending on cost. 4. If the carrier is busy it should go to the next one in line and so forth. I have tried to set this up, but it never answers the

I have a block of DID's that I ported to Vitelity about 7 days ago. The problem is if a POTS caller dials into the system, his dtmf is not heard at READ() or Background() while a prompt is played. After the prompt is finished, then dtmf is heard. I've been working with their support, but it still not resolved. SIP callers are not effected. Yesterday, I purchased a DID from

On Tue, Jun 30, 2015 at 3:34 AM, Tony Kasule <timotsmith at gmail.com> wrote: > Hello, > > Anyone to help me with this issue? It has never worked :( > > On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> > wrote: > >> Hello users, >> >> I have a Digium Te235 and asterisk 13 which have worked well with 1 >> carrier but

The fact that your SIP people are hearing their own voice, but the inbound caller is not is the correct behaviour for your 'echoless' digital termination. If you were to tap the channel in the T1 leaving your premises you would find that the echo is coming in from beyond your interfaces. Similarly if you called another party that was also on all-digital facilities (ie. call back into one

Hello users, I have a Digium Te235 and asterisk 13 which have worked well with 1 carrier but we have failed to add a 2nd carrier. The second telco brings their E1 line over finer, terminated in a RAD modem and they give me ethernet to the E1 card. It's the first time i am having install such a solution, which ideally would be not a big problem. However, The physical layer has failed to

I've some DID's that I'm using for in-bound faxing, but I'm having some trouble with getting that working perfectly on my T1. So I'm thinking of pointing them to an analog line. Will the DID's simply come in over the analog, presumably sending the DID digits via DTMF? Or is that not something that'll work? Thanks, -Ken

What response do you get to *CLI> pri show spans ? On 30 June 2015 at 09:34, Tony Kasule <timotsmith at gmail.com> wrote: > Hello, > > Anyone to help me with this issue? It has never worked :( > > On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> > wrote: > >> Hello users, >> >> I have a Digium Te235 and asterisk 13

Hello, Anyone to help me with this issue? It has never worked :( On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> wrote: > Hello users, > > I have a Digium Te235 and asterisk 13 which have worked well with 1 > carrier but we have failed to add a 2nd carrier. The second telco brings > their E1 line over finer, terminated in a RAD modem and they give

> I recently turned up some 1.8.6.0 call servers in productions, SIP trunks in > routing calls to upstream carrier via SIP trunks out.? I spent a lot of time > in the lab testing 1.8 which included heavily testing DTMF with no issues > that came up.? It all just seemed to work fine.? But then again you can?t > reproduce every real work scenario in the lab. > > > > I?m

Is there anyone out there who has VoicePulse Connect working with DTMF? I've been unable to get it to work from the start, and the recent VoicePulse updates did not help. A caller to my DID's hears Asterisk, but pressing DTMF does nothing: On call setup "iax2 debug" shows: ----------------- Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK

Hello everyone, I'm hoping someone can help me with this. I have a business customer in the U.S. (Michigan, AT&T Territory). I need to get 4 trunks into an asterisk Box. My intention is to use an Eicon Diva Server card with 2 BRI Circuits. The reason for this is that the business needs DID's on the trunks (20 of them). A full or fractional PRI is overboard for them, as they will

> Watch the output of 'pri debug span 1' on the Asterisk server while > placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) > might be relevant. Yes, this is exactly what is happening. Thanks a lot. I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about. Currently I have this in my dialplan:

I have the following upcoming install and I'm trying to do it without having to resort to Digium t1 cards. I have a Cisco IAD2431 being installed by our Carrier. That Carrier will be providing 2 IP Trunks via ethernet handoff into the Cisco IAD2431. The CIsco IAD2431 has Two T1 ports installed and we would have to install a digium card to support those two t1's. What I'd like to do

Hi guys, I posted a somewhat similar question about a month ago and got a thoughtful resonse from Steven Critchfield, but I've got a quick follow up question to it. I'm looking to setup a 16 extension / 10-14 phone line Asterisk install for a customer who would like to have DID numbers for the extensions, since they're currently on Centrex and already have the 1-to-1

I have a SIP carrier which accepts only G729 connections from my Asterisk server. If all the server does is Dial() (out) two legs of a call which are natively bridged, with no processing the media (and no DTMF detection, etc), do I need to install a G729 codec of my own? All the media from each leg connected to the other is already encoded into G729 by the SIP carrier from which it's coming