[asterisk-users] Identify port on Khomp card.

environment: - Elastix v2.3 - Khomp KFXO IP - All the lines are attached to the card on the ethernet interface. -- Att.* *** Luis H. Forchesatto -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

[asterisk-users] Asterisk does not persist callgroup and pickgroup configuration.

they are not configured with callgroup/pickgroup, the fields are empty. Manually inserting callgroup/pickgroup on the extensions worked just fine but the next day the configuration just vanished and the extensions was not working. Has someone a clue of whats going on here? -- Att.* *** Luis H. Forchesatto

Re: [asterisk-users] Extension cant pickup calls but can transfer.

and see if it works now. I'll be back with the result soon. 2013/3/8 A J Stiles asterisk_l...@earthshod.co.uk On Thursday 07 March 2013, Luis H. Forchesatto wrote: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls

Re: [asterisk-users] Extension cant pickup calls but can transfer.

Yes, it worked :D Thankyou guys for the help. 2013/3/8 Luis H. Forchesatto luisforchesa...@gmail.com I think I found the problem. Better looking the sip_additional.conf file I noticed that a few extensions didnt had a callgroup and pickgroup configured, even with the interface appointing

[asterisk-users] Extension cant pickup calls but can transfer.

Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the

Re: [asterisk-users] Extension cant pickup calls but can transfer.

Its only ONE phone who doesnt pickup calls. 2013/3/7 Yves A. yves...@gmx.de do you have only ONE phone, that can´t pickup, or is this a general problem? is pickup configured (feature.conf) AND enabled ? regards, yves Am 07.03.2013 19:05, schrieb Luis H. Forchesatto: Greetings. I

Re: [asterisk-users] Extension cant pickup calls but can transfer.

release etc. - check call-group and pickup group... is the non working extension configured there? regards, yves Am 07.03.2013 20:28, schrieb Luis H. Forchesatto: Its only ONE phone who doesnt pickup calls. 2013/3/7 Yves A. yves...@gmx.de do you have only ONE phone, that can´t pickup

Re: [asterisk-users] Redirect incoming call to SIP trunk.

Solved. 2013/3/5 Luis H. Forchesatto luisforchesa...@gmail.com Greetings. I got two asterisk servers, one is connected to another via sip trunk. The incoming calls are routed to the time period an if matches is transfered to the designed extension. If don't, is redirected to a second time

[asterisk-users] Redirect incoming call to SIP trunk.

Greetings. I got two asterisk servers, one is connected to another via sip trunk. The incoming calls are routed to the time period an if matches is transfered to the designed extension. If don't, is redirected to a second time period. Then, if the call matches the second time period it need to be

[asterisk-users] RJ11 x RJ45

manuais na internet mas não entendi ao certo como tem que ser feito. -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

[asterisk-users] Monitor extensions status.

asterisk extensions). But anyway...I'll be open to opinions. My environment: - Asterisk 1.6.2.13 - Server running Elastix 2.0.0 - DAHDI v. 2.3.0.1 -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.com

[asterisk-users] Call hangs when selected queue number 1.

this happens, the queue 1 worked fine for months. -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] Volume issue.

Hi experts. Recently I've insalled a PCI Khomp Pane on my server and inserted 4 chips to make call with it. The calls are good and no issue was noticed but I got reports that when someone call the chips the call volume is uncommonly low for both sides and they deploy some failures on the audio,

Re: [asterisk-users] DTMF Issue.

Up? 2012/8/20 Luis H. Forchesatto luisforchesa...@gmail.com Thanks for your answer. The logs where posted at pastebin, here the links: - Working Phone: http://pastebin.com/q3pHcwna - Not working phone: http://pastebin.com/iiCHPMmn 2012/8/20 Rusty Newton rnew...@digium.com On 8/20/2012

[asterisk-users] DTMF Issue.

Hi I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of ATA on the network who autenticate the phones: Linksys PAP2, Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP server at the same network all with g729 codecs and rfc2833 for the DTMF. Making calls via

Re: [asterisk-users] DTMF Issue.

Thanks for your answer. The logs where posted at pastebin, here the links: - Working Phone: http://pastebin.com/q3pHcwna - Not working phone: http://pastebin.com/iiCHPMmn 2012/8/20 Rusty Newton rnew...@digium.com On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote: Hi I've got a little issue

Re: [asterisk-users] Extensions DTMF

Of *Luis H. Forchesatto *Sent:* Wednesday, August 15, 2012 11:45 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Extensions DTMF ** ** Any clues? 2012/8/15 Luis H. Forchesatto luisforchesa...@gmail.com 2.3.0.1

Re: [asterisk-users] Extensions DTMF

I've swapper the fones and the good phone stopped working. The good device is a Overtek OT-ATA200SP and the bad phone device is a Linksys PAP2. 2012/8/18 Luis H. Forchesatto luisforchesa...@gmail.com Hi Before I swap the phones, I was wondering if asterisk couldn't be lost somewhere. I've

[asterisk-users] Extensions DTMF

Greetings Recently I've noticed some of the extensions on our VoIP server are not beign recognized by the IVR of a few destinys I've tested. I press que IVR number but it simply don't transfer. This is not ocurring to all extensions. I'm using rfc2833 to all extensions and Elastix on CentOS 5.5.

Re: [asterisk-users] Extensions DTMF

of Asterisk did your Elastix install? ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Luis H. Forchesatto *Sent:* Wednesday, August 15, 2012 8:43 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject

Re: [asterisk-users] Extensions DTMF

Any clues? 2012/8/15 Luis H. Forchesatto luisforchesa...@gmail.com 2.3.0.1 2012/8/15 Danny Nicholas da...@debsinc.com DAHDI version? ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Luis H. Forchesatto *Sent

Re: [asterisk-users] Voicemail full.

Yep, maxmsg is set to some value and it reached. To make things work again i've moved que messages to a new directory and voicemail is working now. 2012/8/3 Steve Edwards asterisk@sedwards.com Un-top-posting... On Fri, 3 Aug 2012, Luis H. Forchesatto wrote: I've made a call to our

[asterisk-users] Voicemail full.

Hi I've made a call to our elastix server and the call was redirected to the voicemail, which the user should leave a message. Intead recording the call the service returned a message like Sorry, but the user's mailbox can't accept more messages. I'm a little lost in the configs here, what

[asterisk-users] URA

Hi Recently our asterisk system stopped beign recognized by URA in others telephones exchanges. What's the troubleshoot steps for this issue? -- Att.* *** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com