similar to: Carrier needs more call examples

Hi Folks, Is there a way to find out the mobile/landline carrier name based on the phone number? For example, who is the mobile carrier for (415)2345678 I had heard about some query but just don't remember how/what? Thanks in advance. Ritesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

What is the deal with "CSR's"? TWTelecom is telling me that I can't port a number to their service without a Customer Service Record. Apparently this is easy with Verizon, and not so easy with some other companies. Basically I'm at a brick wall with a couple of ports because TWTelecom is telling me I HAVE to get a CSR and certain other providers (Time Warner Cable for

Receiving a fax with 1.6.1: == Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on 'DAHDI/4-1' -- Executing [fax at incoming-pstn-line:1] NoOp("DAHDI/4-1", "Fax Detected") in new stack -- Executing [fax at incoming-pstn-line:2] Goto("DAHDI/4-1", "incoming-fax,s,1") in new stack -- Goto (incoming-fax,s,1) --

Hello, We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP packets with encoded G729 payload. VAD/DTX is enabled. We see that the last

I'm trying to send a fax to my asterisk box, however shortly after connecting the fax machine reports a "communication error" and hangs up. Below is the error I get from RxFax: Fast carrier training failed Nothing is written to the file system as far as the tiff is concerned. Any ideas on how to fix this? Thanks ---- Sending Fax Machine: HP PSC 2000 Series 2210

The side effect of <audio> is also a bandwidth usage? linear to the amount of "fake" listeners? From: Marcin Lewandowski [mailto:marcin at saepia.net] Sent: Tuesday, July 05, 2016 4:31 PM To: Christoph Zimmermann; icecast-dev at xiph.org; Yaniv Sharon Subject: Re: [Icecast-dev] multiple connection (be careful with carrier-grade NAT) As I have written before, I’ve seen that

Hi, Yes I aware to the NAT possibility, But let's assume that this is the issue, there is no reason that 30 listeners >From the same country will connect and disconnect at the same time range... I'm pretty sure that its individual listener/IP. I deleted the Access log files, but in the next time that I will catch similar situation again, I will complete the investigation. Nobody from

All; I have a customer who is looking for a particular DID. (I dialed it and it is not in service). I searched through my preferred upstream provider's list but I came up empty. I wrote them, and this is their reply. "We currently do not have that specific number in stock as this number is owned by another carrier that we do not have a business relationship with." So my

FYI, Follow the Quick Start Guide from Carrier Access to setup the CMG (Customer Media Gateway) Router card. Follow the Asterisk mgcp.conf wiki page setup. The only issue I had was with the CAC CMG card, it defaults to strict policy message exchange and dial-tone will not come across when you go off-hook on a FXS port. To resolve this, get back into the CAC CMG configuration, goto: -profile

Hi, Does anyone know anything about the following? In a hosted environment where several area DIDs are provisioned on a single server, how do most carriers establish the origination DID, number. Asterisk allows us to modify the CallerID, name, number and DNID channel variables before dialling out via SIP. Most carriers will allow us to spoof a callerID when placing a call, and pass it forward.

Hellos, My voip carrier has assigned me a IP based account...where they only give me the IP to call through. I have setup the dial plan exten => _7XXX.,1,Answer() exten => _7XXX.,2,vmauthenticate(${CALLERID(number)}) exten => _7XXX.,3,Dial(SIP/${EXTEN:1}@Y.Y.Y.Y) exten => _7XXX.,4,Hungup() Where Y.Y.Y.Y is the assigned IP. After Dialing I asterisk logs the error SIP/Y.Y.Y.Y-35dc

We've been using RNK Telecom for terminations for our SIP service, but their billing is questionable, they've been in breach of contract multiple times, and when I brought it to their attention, Russ Man, our 'friendly' account manager told me, and I quote "If you are having that much of a problem..please find another carrier." With customer service like that, I've

Hi list, I have search for a clear explanation about this mensage " WARNING T.30 ECM carrier not found", but until now I dont succed on it.Anybody know how can I handle with this problem? I have Asterisk 1.6.2.13 with TDM 410p and the Fax is connected to Dlink FXO dvg 2032s. Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda

I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel bank (12fxs/12fxo). I have the setup partially working thanks to some help from IRC. However I still have the following issues I can't seem to resolve 1. When calling into the system from the PSTN call hangup is not detected. * leaves line in use until it is shutdown. 2. When calling an analog phone connected to

We have a queue monitoring application running so we can see the caller ID of callers in a queue. If we see a VIP in the queue, is there any method to force that call to be first in line? If there's a softphone, or queue managing application already written that does this, I'd love to know.

When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more

All- Recently an Asterisk server we host was hacked and used to route some unauthorized calls. We have since improved our security measures, including installation of fail2ban. The interesting thing is the way in which this was discovered. The unauthorized calls were occurring intermittently last Thurs evening thru Sat morning. On Sat morning, some of our employees were attempting to log-in

Hello, My computer is not connecting to the local network with the following configuration: dmesg.boot contains the following lines: ----------------------------------------- vr0: <VIA VT3043 Rhine I 10/100BaseTX> port 0xec00-0xec7f mem 0xfeafbc00-0xfeafbc7f irq 11 at device 12.0 on pci2 vr0: Ethernet address: 00:40:05:a5:00:04 miibus0: <MII bus> on vr0 amphy0: <Am79C873 10/100

Dear Steve after a lot of compile from spanDSP first version to K and after i have compile * HEAD and Stable i cant understand where im wrong... - Mandrake 9.2 - libtiff3-3.5.7-11mdk I have add Header from your FTP but the result is this error.. Can you give me your opinion? Thanks in advance Dimitri -------- -- Executing RxFAX("Zap/35-1", "/home/user/testfax.tif") in

Hmmm, perhaps I am the only one who doesn't trust their telco (I doubt it) but... I have the rates that I currently pay my telco, and would like to extract my CDR's and add an additional field displaying the actual price paid for the call. I would like to do this based on destination phone number, and outgoing channel. However, I have a few difficulties: 1) I pay a different rate for