Re: [asterisk-users] Recent UnixODBC Issues

Happy Taco Tuesday All, The change I previously mentioned[1] has been merged into all applicable branches and will be in the next release. From our own testing it has shown to solve all the problems mentioned and should return UnixODBC to a stable state. I hope that this won't be the case but

Re: [asterisk-users] Delay after Answer

Well, I thought I had the problem solved. Ported everything over to PJSip and build RDNS records for the phones and the server, but I am still experiencing the problem on incoming calls. ** On 6/7/2016 1:00 PM, Faheem Muhammad wrote: I've faced the same issue. The issue was related to DNS,

[asterisk-users] Unable to create channel DAHDI

In trying to troubleshoot the Delay after Answer problem I had before (which seems to be fixed), I have somehow created a new problem: Outgoing calls are now failing with the following message: [Jun 7 13:28:09] WARNING[9247][C-]: app_dial.c:2429 dial_exec_full: Unable to create

Re: [asterisk-users] Delay after Answer

I've faced the same issue. The issue was related to DNS, the reverse lookup query failure caused the delay around(7-9 seconds). The purpose of reverse lookup is to block IP Spoofing attacks. Regards, Faheem On Tue, Jun 7, 2016 at 7:48 PM, Brent Davidson wrote: > I

[asterisk-users] Asterisk dumping core. How to read fax debug

All; I have a customer running Asterisk 11.6-cert13. The server is solely used for faxing. The problem he has is that Asterisk is dumping core on a very regular basis, maybe half a dozen times a day at least. From the logs, I see that the last commands executed before the dump were fax

Re: [asterisk-users] Delay after Answer

I've seen this sort of thing where a DNS server is programmed in resolv.conf but is not accessible over the network. Threads get blocked, and you have to wait for the DNS query to timeout. On 16-06-07 10:48 AM, Brent Davidson wrote: I am having an issue with a couple of phones where they

Re: [asterisk-users] PJSIP subscribe

Hello, thank you for the answer... how can I see the correct status? any configuration on asterisk or softphone side? Regards El 07/06/2016 a las 16:36, George Joseph escribió: I can confirm that Bria shows offline but if the client is using the tuple status instead of the person status then

[asterisk-users] Delay after Answer

I am having an issue with a couple of phones where they ring, but there is a long delay after the phone is picked up before the audio starts. My setup: * Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC * Server is CentOS 7 * Quad core CPU with 16GB Ram * 2 Snom 300 phones. * NO NAT.

Re: [asterisk-users] PJSIP subscribe

On Mon, Jun 6, 2016 at 11:13 AM, Annus Fictus wrote: > Hello, > > I'm trying to use presence with PJSIP and I have a "issue". > > I created correctly hint priorities like: > > exten => 1000,hint,PJSIP/1000 > exten => 1001,hint,PJSIP/1001 > > Extension 1000 can subscribe

Re: [asterisk-users] Want to detect sound

Try MixMonitor. Land the call to a local channel and answer it. This code will record the silence as well. exten => _X.,1,MixMonitor() exten => _X.,n,Dial(Local/100@context1) [context1] exten => _X.,1,Answer() exten => _X.,n,Dial(SIP/${EXTEN} On Tue, Jun 7, 2016 at 2:16 PM, Mamadou NGOM

Re: [asterisk-users] Recent UnixODBC Issues

Marek Červenka wrote: Dne 6.6.2016 v 17:42 Joshua Colp napsal(a): Happy Monday all, Since I sent my previous email a lot has been learnt about our UnixODBC problem and a path has emerged ensuring both better performance while making sure people are not required to upgrade their UnixODBC unless

Re: [asterisk-users] Recent UnixODBC Issues

Dne 6.6.2016 v 17:42 Joshua Colp napsal(a): Happy Monday all, Since I sent my previous email a lot has been learnt about our UnixODBC problem and a path has emerged ensuring both better performance while making sure people are not required to upgrade their UnixODBC unless they want to. So

[asterisk-users] Want to detect sound

Hello everybody,I manage not to detect one silence with record () when I make as follows:Exten = > 0178900271, n, Record ($ ${ link_recorded_pseudos_clients } pseudo_ Client_Id} wav, 5,5) exten = > 0178900271, n, GotoIf ($ [" $ {STAT (e, RECORDED_FILE} " = "0"]? Erreur_enregistrement_PPX17_1)When