Re: [asterisk-users] Missed call

On some SIP phones it is possible to turn off the missed call notifications, but I am not aware of a way to do the same on any cell phones. On 5 Jun 2015 07:29, Mehmet Avcioglu meh...@activecom.net wrote: There is no signal that is sent to display a missed call. Your cell phone does that. If

[asterisk-users] Missed call

Hi list! I configured Asterisk to forward the incoming call for a number to both phones. I wrote that: exten = _0049351222,n,Dial(SIP/0049351222SIP/004935,,R) of course it works... Now the problem is, that when a phone get the call, on the other phone I get 1 missed

[asterisk-users] Accessing an account from more than one phone

Hi again! I'm thinking about using my mobile phone to receive (and send) calls when I'm not at home (for example in holiday). I can make my Asterisk reachable from Internet, of course, or I can use a VPN, that's not the problem... My question is: can I log in to the same account from more

Re: [asterisk-users] תשובה: Missed call

Zitat von Israel Gottlieb isr...@gmail.com: If you the c option in the dial command it will send answered else where sip message to the phone and most ip phones understand that The cell will always display a missed call  I'm very sorry, but I can't understand what you mean... Could you

Re: [asterisk-users] תשובה: תשובה: Accessing an account from more than one phone

Zitat von Israel Gottlieb isr...@gmail.com: Shalom to you too So that's the way to go Toda! :) I'll try this weekend... Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] sslv3 alert unexpected message

Dne 3.6.2015 v 17:57 Marek Cervenka napsal(a): hello, my webrtc calls ends after ~60seconds with res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0xb6c02a94' due to reason 'sslv3 alert unexpected message', terminating. any ideas where can be problem? or howto debug this problem?

Re: [asterisk-users] Missed call

There is no signal that is sent to display a missed call. Your cell phone does that. If it rings and is not answered it counts that as a miss. The only way to avoid it is to not ring it. So instead of simultaneous ringing you can do sequential. -- Mehmet Avcioglu meh...@activecom.net On

[asterisk-users] תשובה: תשובה: Accessing an account from more than one phone

Shalom to you too So that's the way to go   הודעה מקורית   מאת: Luca Bertoncello נשלח: יום שישי, 5 ביוני 2015 09:51 אל: Asterisk Users Mailing List - Non-Commercial Discussion השב ל: Asterisk Users Mailing List - Non-Commercial Discussion נושא: Re: [asterisk-users] תשובה: Accessing an account

Re: [asterisk-users] תשובה: Missed call

At the end of the Command you could use options one of them is the c (not apital) which sends a cancel event to the phone http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial On Fri, Jun 5, 2015 at 9:53 AM, Luca Bertoncello lucab...@lucabert.de wrote: Zitat von Israel Gottlieb isr...@gmail.com:

Re: [asterisk-users] תשובה: Accessing an account from more than one phone

Zitat von Israel Gottlieb isr...@gmail.com: Shalom, Israel! Using chan_sip you need to create another ‎user aand then dial both Using pjsip you can connect 2 devices Thank you. Unfortunately it seems that I don't have pjsip available as package on the OpenWRT where I installed Asterisk...

[asterisk-users] תשובה: Missed call

If you the c option in the dial command it will send answered else where sip message to the phone and most ip phones understand thatThe cell will always display a missed call

[asterisk-users] תשובה: Accessing an account from more than one phone

Using chan_sip you need to create another ‎user aand then dial both Using pjsip you can connect 2 devices   הודעה מקורית   מאת: Luca Bertoncello נשלח: יום שישי, 5 ביוני 2015 09:24 אל: ML, Asterisk users השב ל: Asterisk Users Mailing List - Non-Commercial Discussion נושא: [asterisk-users]

Re: [asterisk-users] תשובה: Missed call

Zitat von Israel Gottlieb isr...@gmail.com: At the end of the Command you could use options one of them is the c (not apital) which sends a cancel event to the phone http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Ach! Thank you! I'll try this evening Luca Bertoncello

[asterisk-users] Logging in local time

Hi again! I just noticed, that my Asterisk (running on an OpenWRT-Switch) writes the logs using GMT... On the Switch the time is right configured and a date says me the current LOCAL time. I didn't found in logger.conf or other file an option to set the timezone. Can someone help me?

[asterisk-users] Problem with SIP-TLS

Hi list! I'm trying to configure my Asterisk to accept SIP-TLS connections, too. I followed this HowTo: http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/ But as soon I try to connect to my Asterisk using SIP-TLS I get on Asterisk-CLI: == Problem setting up ssl

Re: [asterisk-users] Logging in local time

Am 5. Juni 2015 16:29:21 MESZ, schrieb Luca Bertoncello lucab...@lucabert.de: Hi again! I just noticed, that my Asterisk (running on an OpenWRT-Switch) writes the logs using GMT... On the Switch the time is right configured and a date says me the current LOCAL time. I didn't found in

Re: [asterisk-users] תשובה: Missed call

Israel Gottlieb isr...@gmail.com schrieb: At the end of the Command you could use options one of them is the c (not apital) which sends a cancel event to the phone http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Shalom Israel, unfortunately it does not work as expected... I wrote:

Re: [asterisk-users] Problem with SIP-TLS

ricky gutierrez xserverli...@gmail.com schrieb: compilation problems with the module srtp , check the module module show like srtp Now available on OpenWRT... :( Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ --

Re: [asterisk-users] Problem with SIP-TLS

2015-06-05 14:29 GMT-06:00 Luca Bertoncello lucab...@lucabert.de: I think it is a problem on Asterisk for OpenWRT... :( Regards Luca Bertoncello (lucab...@lucabert.de) compilation problems with the module srtp , check the module module show like srtp -- rickygm

Re: [asterisk-users] Problem with SIP-TLS

ricky gutierrez xserverli...@gmail.com schrieb: Hi lucas , dou you try this: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial Tested right now. Same problem... I think it is a problem on Asterisk for OpenWRT... :( Regards Luca Bertoncello (lucab...@lucabert.de) --

Re: [asterisk-users] Problem with SIP-TLS

2015-06-05 12:21 GMT-06:00 Luca Bertoncello lucab...@lucabert.de: Hi list! I'm trying to configure my Asterisk to accept SIP-TLS connections, too. I followed this HowTo: http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/ But as soon I try to connect to my Asterisk

[asterisk-users] 11.17.1 : ReceiveFax then signal 11 ??

dialplan [FaxIncoming] exten=s,1,NoOp(Incoming fax on 46-va) same=n,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)}) same=n,Answer() same=n,ReceiveFAX(${FAXFILE}.tif,df) same=n,Hangup() exten=h,1,NoOp(FAXSTATUS: ${FAXSTATUS} FAXERROR:

[asterisk-users] Free Fax for Asterisk : registered, but license not effective

I've been having problems segfaulting when receiving faxes. So I tried FFA. I used the register utility and my license is in /var/lib/asterisk : ls /var/lib/asterisk/licenses/ -l total 4 -rw-r--r--. 1 root root 325 Jun 5 22:41 FFA-DNCXXX.lic but: Executing [17772822954@FaxIncoming:4]