similar to: smbldap-useradd not creating machine accounts in correct fashion

Hello - If I create multiple pieces of output in grid, and use grid.remove() to try to remove the output from the graphics device, I cannot seem to remove the final piece of output from the device until I 'refresh' the graphics device by giving it focus. The function grid.remove() does appear to remove the grob from the display list, however. I noticed this when the example on page

Hi I'm trying to upgrade from Samba 3.0.23c on FC4 to 3.0.25b from Samba sources. Everything appears to function correctly until an MS word document is opened from a share - the file opens but the smbd process in question rockets to 99%+ CPU, stays there & needs kill -9'ing to stop it. This happens reliably. I have an appropriate strace & a level 7 log but can't see anything

Yes, I think the dial does get executed (sonny calling outbound 202-555-1212): core set verbose 3 Console verbose was OFF and is now 3. -- Executing [912025551212 at from-internal:1] Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from "" <sonny> to 12025551212 through fromgw") in new stack [Mar 15 19:27:06] NOTICE[16648][C-00000022]: Ext. 912025551212:1 @

That was the issue, thanks. I now am able to get the caller ringing on an outbound call, but an external phone number (E164) I am dialing does not ring. On Sun, Mar 15, 2015 at 12:19 PM, George Joseph <george.joseph at fairview5.com > wrote: > > > On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> I have setup my

On Sun, Mar 15, 2015 at 5:56 PM, Sonny Rajagopalan <sonny.rajagopalan at gmail.com> wrote: > George, > > I have the detailed log below. (Resending after trimming the email to 40KB.) > > The sequence below just repeats ad-nauseam. Is this a SIP trunk issue? > > Thanks! > I don't see anything obvious. The registration works though, right? You might want to compare

I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic configuration works, and I am connected to a SIP trunk using SIP.US, and have set up my inbound calling which works correctly (when I call my PBX DID, the call does come into my PBX network). The issue is that I am not able to make outbound calls, because the call fails with the error:

Thanks again. How do you create that message context in extensions.conf? On Mon, Nov 16, 2015 at 9:44 PM, Thyda ENG <engthyda at gmail.com> wrote: > According to what I have done , I add the message_context to the > pjsip.endpoint_custom.conf in /etc/asterisk and then I create that > message_context in the extension.conf, and it works. > > On Tue, Nov 17, 2015 at 9:34 AM,

I'm trying to understand why my group mapping doesn't work in a consistent fashion. I've studied "Important Samba-3.0.23 Change Notes" & chapter 13 of TOSHARG but am still struggling. I'm on 3.0.23a-1.fc4.1 (Fedora Core 4) as a PDC, tdbsam backend. 'net groupmap list' gives this: Domain Power Users (S-1-5-21-1365060548-1276164359-2333037906-31037) ->

On Wed, Feb 17, 2016 at 8:56 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > I made some progress. The first thing I have realized is that it is my > Twilio configuration in pjsip_wizard.conf that was killing me. I have since > removed that entire file from /etc/asterisk and I am able to make > "from-internal" context calls (i.e., calls that do not

So, the only thing that is needed in the endpoint definition in pjsip.conf (there is no such file pjsip.endpoint_custom.conf) is *message_context=astsms* Is that correct? Anything I need to do in extensions.conf? I see that the messages are received at Asterisk (when I turn on pjsip set logger on) but they are not delivered to the other endpoint. What gives? Any help appreciated. Thanks! On

I have a working Asterisk 13.1.0 running, and I am trying to configure a SIP trunk for outbound and inbound calling, and a DID for the Asterisk server, which is used for incoming calls from PSTN. I configured my SIP.US trunks (showing one gateway, gw1, here for brevity, have two: gw1 & gw2, which are both configured on my end): [sonnyGW1] type=registration transport=transport-udp

Hi, I am using Asterisk 13.6.0 and was wondering if I can programmatically add users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk server using API of some sort. Please do let me know. Thanks, Sonny. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Yes, it is enabled on port 5060. I do receive a TCP ACK back from the server, so I know the TCP segment is received at the server hosting the Asterisk build. On Wed, Feb 17, 2016 at 8:36 AM, A J Stiles <asterisk_list at earthshod.co.uk> wrote: > On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote: > > OK. Let me ask this. Is anything else necessary, except choosing TCP as > the

Thanks George, for your mighty quick response. I made the changes (re: server_uri_pattern etc.) and still, no luck--it fails for the same error. BTW, there is nothing for transport (but this is the same config from my SIP/UDP + Twilio days, which worked): *CLI> pjsip show transport twilio-siptrunk Unable to find object twilio-siptrunk. *CLI> pjsip show identifies No objects found. I did

Hello, I am looking for documentation support for enabling instant messaging between endpoints using Asterisk 13.1.0 and vanilla VoIP clients such as Zoiper. Where do I enable this support on the server side and does it need anything on the client side? I see plenty of online help for chan_sip, but nothing for chan_pjsip. I imagine there is both pjsip.conf configuration and extensions.conf

OK. Let me ask this. Is anything else necessary, except choosing TCP as the preferred protocol on the client, to make TCP w Asterisk work? At the moment, I have only changed one line in pjsip.conf from my working UDP setup: [transport-tcp] type=transport protocol=tcp ; <--------------- only this line was changed. On Wed, Feb 17, 2016 at 8:28 AM, Sonny Rajagopalan < sonny.rajagopalan at

Oh, wow! Changed it and now I am getting calls into my context (fromgw). Unfortunately, the actual caller ID (6175551212) is not getting passed (but I know Asterisk is getting this). How do I "reap" this actual caller ID in my dialplan? On Fri, Mar 13, 2015 at 4:55 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > > <snip> > >

Hello, I have an Asterisk 13.6.0 PBX using PJSIP connected to the Twilio gateway. I am able to make calls outbound through the gateway, but I am not able to make calls into the PBX from external PSTN. Specifically, an incoming call is _received_ by Asterisk, but it is not able to route the call internally owing to the following error: [Feb 18 21:08:47] NOTICE[4606]:

Nope, there are no contacts to show that pertain to these endpoints (only my SIP trunks show up). On Mon, Feb 15, 2016 at 5:31 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > >> Does this help: >> > > Yes, the transport parameter is in the Contact header so it's interesting > it didn't work. If you use pjsip show contacts what

Sonny Rajagopalan wrote: > Sorry, I was not being very clear, Joshua, and thanks for your patience > with this issue. > > I had set pjsip set logger on and core set debug 99. See absolutely > zilch on asterisk CLI. Or in /var/log/asterisk/messages. If the messages > are not reaching Asterisk, what could be the issue? I am a little > perplexed as to why Asterisk wouldn't