similar to: Problem w/ PC port on Polycom 335

Hi all. When a user transfers a call by pressing the "transfer" soft button on their phone, I'd like it to "beep" at them when the transfer is complete. I've got it turned on in features.conf: xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr ; to indicate a failed transfer However, it seems that

Hi all, I've trying to fix a problem. If I'm in a call and I press the * key, the call goes silent but doesn't hang up. I need to be able to send the * key for various IVR's that I interact with. Since I thought this was related to the features.conf file, you can view it at: http://www.diehlnet.com/features.conf Any ideas are welcome. TIA, -- Mike Diehl

Hi all. I just installed bunch of IP450's and everything went well and my customer is happy.... except that they are unable to transfer calls to other extenstions. They can dial them directly just fine. However, when the user is in a call and presses the transfer soft key, they get dial tone, and start typing the extension, say 1008. But by the time they get 100 typed in, the phone tries

Hi all. I'm in the process of configuring a phone system for my family and friends. I'm wondering if I should try to implement the "Vertical Services" (http://www.nanpa.com/number_resource_info/vsc_assign) in the Asterisk dialplan, or if I should delegate those functions to the various ATA's. For example, the Sipura SPA 2002 can handle*69 internally. On the other

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'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, Is it possible to send a SIP header to a PAP2T or SPAxxxx and cause the device to automatically answer? I can do this with my Polycom phones and would like to do it with my ATA's. Any ideas? -- Take care and have fun, Mike Diehl.

Hi all, I'm trying to get moh working on * version 1.4.4. I've setup a test extension that answers the call and runs the musiconhold command with the appropriate class name. All I get on the phone is silence. The console tells me that moh started and immediately stopped, but it complains that there is "No class: moh0" *CLI> [Feb 22 12:17:36] WARNING[31142]:

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

Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone

Hi all, While trying to track down a T.38 issue, I came across a series of log entries like this: ============================================================================ [Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr: Unable to allocate socket: Too many open files [Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor: Cannot create socket

Hi all, I'm trying to tighten things up a bit and I seem be be running into something that doesn't make sense to me. I've got 2 contexts, one for customers, and one for guests, that I include into [customers] and [default], in extensions.conf, as below: ============================================================= [default] include = dial_GUEST [customers] include = parkedcalls

Hi all, I'm looking into being able to send/receive SMS messages with my asterisk box in the US. I've seen the SMS command as well as the Kannel program. I'd prefer to do it from Asterisk. I've tried something like: exten => 999,n,sms(15551234567,s,"This is a test") in my dialplan, but when this runs, it dials the phone number and then nothing. What am I

Hi all, Lately, I've been seeing more and more instances where I get a flood of warning messages like this: [Apr 26 14:09:50] WARNING[21054] udptl.c: No UDPTL ports remaining The next thing I know, my server is dropping calls and starting to misbehave. I use fax via T.38, so I can't just turn udptl off. I could expand the port range, but I suspect that will just mask the situation.

Hi all, I'm trying to use a wrt54gl router running dd-wrt as a provisioning server for a remote installation. I've got dhcp working and I have provisioning files ready to go. I understand that I need to set bootp option 66 to point to the tftp/ftp/http server. In fact, I have this working completely with the ISC dhcp server The problem is that I don't know how to get

Hello List, i am trying to boot a 2.6.28 vanilla kernel as a DomU. This is some of my config: ---------------------------------------------- root = ''/dev/xvda2 ro'' disk = [ ''drbd:pgone,xvda2,w'', ''phy:/dev/vg/pgone-swap,xvda1,w'',

Hi all, I've got a problem where many times, there is silence at the first 1-2 seconds of a call. Then it clears up and it's crystal clear. I've not put a sniffer on it, yet, but I suspect that the media channel is still being set up. The server shouldn't be too overloaded. Can anyone give me some advise on how to solve/mitigate this problem? Mike.

Hi all, I've cross-posted this to the -users and -biz groups. Hope that's OK. I have a customer who REALLY needs to be able to send/receive faxes reliably. I could probably get hylafax configured, but I'm not sure how reliable it is. If it is considered reliable, would someone let me know? Otherwise, is there a product/service they can buy that will allow them to fax to/from

Hi all, I have 2 servers running 1.6.2.9 and I'm about to build a third server. This suggests the possibility of doing a rolling upgrade of all of my servers. This brings up the question of what version to install and upgrade to. I don't have many upgrade opportunities, so I'd like to get as much bang for my buck. Since I've applied some custom patches to my 1.6, I'd

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