similar to: Grnvoip

Hi, Having some issues here with getting asterisk realtime for the dialplan (extensions.conf) setup: mysql> desc extensions_table; +----------+--------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +----------+--------------+------+-----+---------+----------------+ | id | int(11) | NO | MUL | NULL |

Hi all, I have a customer that tried to use the Texas One-Call number (a toll-free call) to have the utility company come out and mark buried pipes and cables. That call resulted in a recording telling her to dial 811, instead. So, as a service provider, how do I terminate a call to 811? In NM, I send it to NM's One call local number. I wasn't able to find such a number for TX. Is

Hi all, I've got a new server up, but it's not staying up.... After a day or so, it segfaults with: [Mar 22 23:17:49] WARNING[12177]: res_odbc.c:1406 _ast_odbc_request_obj2: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.2(a) Driver]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use

Hi all, I've upgraded to Asterisk 13.14.0 and now I'm seeing that Asterisk is using 100% CPU. I have one AMI agent connected that is acting rationally. I've got a hand full of SIP (RT) registrations. There is no other call activity. I've tried to unload various modules; nothing resolved the issue. Any suggestions? -- Mike Diehl

Hi all, I've got a user who wants to receive voicemail notifications at two different email addresses. I could probably setup an alias in /etc/aliases, but then I'd have to manage that across multiple servers, which I don't want to do. Is there a way I can tell Asterisk to send to multiple addresses? Mike -------------- next part -------------- An HTML attachment was scrubbed...

Hi all, I'm trying to configure a few conference bridges. I've started with the very basic: [general] [default_bridge] type=bridge [default_user] type=user [default_bridge] type=bridge [5340] type=bridge However: confbridge list Conference Bridge Name Users Marked Locked? ================================ ====== ====== ======== *CLI> It doesn't seem to be

Hi all, I'm installing Hylafax on my Asterisk system. From what I've read, I can either use IAXModem or T38Modem to provide the virtual fax device. So at the risk of starting a religious war, which one should I use? I don't mind running IAX if I have to. I want as much flexibility and stability as I can get. So, what are your recommendations? Mike. -------------- next part

Hi all, I have a user who would like to stream their favorite radio station from iHeart radio for their music on hold. It this TECHNICALLY possible? If so, any pointers would be appreciated. Is this LEGAL in the US? Thanks in advance, Mike. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I've got a device that seems to become unreachable for about 2 minutes, every hour. From what I can tell, it isn't due to network or server issues. Any ideas? TIA. -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701

I just had a user complain about a call getting dropped and another one failing to go through. I'm trying to interpret the log entries for each call and would like to confirm my understanding. The first entry is from an outbound call to a 1-866 number. The 5058971234 at 216.31.xx.x refers to a user account on my server. Voip_one refers to my provider. This sequence of log entries

Hi all. I'm looking to change from a standard telephone line to a VoIP phone line at home. I'm looking for recommendations for VoIP providers that I can use with Asterisk. One of the catches is that I often telecommute and sometimes I do some side business; these practices violate many provider's acceptable use policies. So, I need a provider who doesn't care how I use the

Hi all, I'm contemplating an upgrade from 10.2.4 to 11.4.x. However, the 1.8.x to 10.4.x upgrade was painful; some of the modules had been renamed, if I recall correctly. So, is there a list of MAJOR changes and GOTCHA's between 10.x and 11.x? I'm hoping for something a little less granular than the release notes from 10.2.x to 11.4.x. I don't mind reading, but that is almost

Hi all, I've been seeing some strangeness lately on my 10.2.1 server. It's gotten to the point that a few times each day, I see masses of SIP clients becoming unreachable. They're not all on the same network, and we don't see any calls drop. In a few seconds, they all come back. I don't think it's a connectivity issue because we don't drop calls, and the endpoints

Hi all, I'm trying to build a fax relay mechanism where faxes come in and get relayed out to their final destination. I'm using the h extension to store various results from both legs. This data is being saved correctly for the first (receiving) leg. The second leg isn't calling the h extension when it's finished. The second leg is being initiated by a .call file like:

Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet link is solid, but the device becomes unreachable within a day or so of being rebooted. Then the customer goes to reboot the device, they report that all 4 lights are lit. The ISP reports that the device does respond to ping, so it's not completely dead. I've had the same symptoms with SPA303's

Hi all, I have a user who is reporting dropped calls at his site. We don't have any other users complaining of this. So far, this is what we know: 1. The manager bought all new Polycom phones. (POE) 2. They replaced the network switch with a POE version. 3. It's not just one or two of the phones that have problems. 4. It doesn't matter if they use the headset or the cordless

Hi all, I've got a customer who's reporting "ghost calls." Essentially, the phone rings, they pick up, and there's no body there. It is NOT one-way audio, and it doesn't happen all the time. We use voipmonitor to watch calls, and this is what we saw for the call in question: | calldate | caller | called | duration | whohanged |

Hi all, Lately, I've seen an increase in the number of attacks against my system from the so-called "Friendly Scanner." When one of these script kiddies targets my server, all I see for symptoms is a few of my trunks become lagged due to server load and a stream of messages on the console that resemble this: [Aug 2 20:27:50] == Using SIP VIDEO CoS mark 6 [Aug 2 20:27:50] ==

Thank you for your time. I've put my replies to your questions in-line, below. On Wednesday, June 07, 2017 10:19:41 AM Antony Stone wrote: > On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote: > > > Hi all, > > > > I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've > > discovered that my server crashes as soon as I leave a

Hi all, I've got a user with a couple of Cisco SPA303's. When I dial their phones with a dial string like: dial(sip/phone-a,300,rwkxttT) The phone rings, as expected. However after exactly 60 seconds, I get: [Aug 21 02:09:56] -- Got SIP response 480 "Temporarily not available" back from a.b.c.d:5062 [Aug 21 02:09:56] -- SIP/phone-a-00006a9d is circuit-busy [Aug 21