Re: [asterisk-users] Asterisk Behind Firewall

Hi, I have used a sonicwall Firewall, it has a sip transformation feature. It is necessary to use a firewall to protect your server Best Regards, Madushan On Mon, Jan 4, 2016 at 11:45 PM, IPN Comm wrote: > I was wondering if anyone can give me any pointers or insights of

Re: [asterisk-users] Ignoring audio media offer because port number is zero

and this is working without any issue in elastix. I need to setup this in pure asterisk installation. Best Regards, Madushan On Thu, Feb 11, 2016 at 2:26 PM, Madushan Geethanga <mgliyanage...@gmail.com > wrote: > Hi, > > I'm trying to setup a sip trunk. The asterisk vers

[asterisk-users] Ignoring audio media offer because port number is zero

Hi, I'm trying to setup a sip trunk. The asterisk version is Asterisk certified/11.6-cert12. I had to change the CallerID to a pilot number of the trunk as provider only accept it. but when i dial I'm getting below error. Cannot hear anything [Feb 11 14:19:45] WARNING[3920][C-005d]:

[asterisk-users] Asterisk Call Forwarding

Hi I have to setup call forwarding. How do we setup Call forwarding in asterisk?. Eg. user dials a number and insert some mobile number for forwarding and dial another number to cancel the forwarding. thanks a lot. Best Regards, Madushan --

Re: [asterisk-users] Asterisk Call Forwarding

Hi, Thanks Phil, I will implement this and get back to you. Best Regards, Madushan On Thu, Mar 3, 2016 at 4:12 PM, Phil Reynolds < phil-aster...@tinsleyviaduct.com> wrote: > On Thu, 3 Mar 2016 08:21:14 +0530 > Madushan Geethanga <mgliyanage...@gmail.com> wrote: > > >

Re: [asterisk-users] Asterisk Call Forwarding

Hi What is redacted means? same => n,GotoIf($["${CALLERID(num)}"="**"]?divert:void) Thanks Best Regards, Madushan On Thu, Mar 3, 2016 at 10:58 PM, Madushan Geethanga <mgliyanage...@gmail.com > wrote: > > Hi, > > Thanks Phil, I will implement t

Re: [asterisk-users] VoiceMail Audio playing

Hi, VoiceMailMain is used to retrieve voice mails http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMailMain Best Regards, Madushan On Fri, Jul 15, 2016 at 3:07 PM, Joaquin Alzola wrote: > Hi Guys > > > > Which module on Asterisk is the one in charge of playing

[asterisk-users] (no subject)

Hi, Im trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is working fine but i cannot dial out. i don't hear anything on the phone and asterisk CLI also does not show anything. my config is. please advice. [2001] type=endpoint context=out-local

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

yes I have unchecked it. On Fri, Sep 9, 2016 at 10:27 PM, Administrator TOOTAI <ad...@tootai.net> wrote: > Le 09/09/2016 à 18:32, Madushan Geethanga a écrit : > >> Hi, >> > > If you're not using RTP encryption did you uncheck the option in your RTP > TAB from

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

thanks for the reply. if i config the extension in softphone it works fine. but with snom its not working Bet Regards, Madushan On Fri, Sep 9, 2016 at 10:31 PM, Madushan Geethanga <mgliyanage...@gmail.com > wrote: > yes I have unchecked it. > > On Fri, Sep 9, 2016 at 10:27 PM

[asterisk-users] Asterisk 13 PJSIP with Snom 710

Hi, I'm trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is working fine but i cannot dial out. i don't hear anything on the phone and asterisk CLI also does not show anything. my config is. please advice. [2001] type=endpoint context=out-local

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

<sip:0@54.206.59.252;user=phone>;tag=z9hG4bK-bskkkx1t5bas Call-ID: 313437333433383639323238313539-ahn3begiq66q CSeq: 1 ACK Max-Forwards: 70 User-Agent: snom710/8.7.5.35 Contact: <sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835>;reg-id=1 Content-Length: 0 Best Regards, Madushan O

[asterisk-users] Asterisk 13 externip

Hi, What is the equal option for externip in asterisk 13 with pjsip. I have tried external_media_address=XX.XX.XX.XX external_signaling_address=XX.XX.XX.XX but asterisk 13 writes local ip to the from header. any suggestions? Best Regards, Madushan --

Re: [asterisk-users] Asterisk 13 externip

gmail.com> > wrote: > >> >> >> On Wednesday, 14 September 2016, Madushan Geethanga < >> mgliyanage...@gmail.com> wrote: >> >>> Hi, >>> >>> What is the equal option for externip in asterisk 13 with pjsip. I have >&

Re: [asterisk-users] Asterisk 13 externip

Sep 16, 2016, at 18:51, George Joseph <gjos...@digium.com> wrote: >> >> >> >> On Fri, Sep 16, 2016 at 5:55 AM, Madushan Geethanga < >> mgliyanage...@gmail.com> wrote: >> >>> Hi, >>> >>> Tried with both softphone (Ekiga) an

[asterisk-users] DTMF not detecting from Mobile phones

Hi, I have a Asterisk running in Amazon AWS with an IVR configured. The problem I'm having is I'm not getting DTMF from mobile phones. The Landlines works without any issues. I have configure DTMF to rfc 2833. I checked with dtmf debug but I'm not receiving dtmf from mobile devices. please let me

Re: [asterisk-users] Asterisk 13 externip

jos...@digium.com> wrote: > > > On Thu, Sep 15, 2016 at 8:38 AM, Madushan Geethanga < > mgliyanage...@gmail.com> wrote: > >> Hi, >> >> Thanks for the reply. >> >> Yes my PABX is on the cloud when I try to register to the server, the >>

[asterisk-users] Asterisk 13 + PJSIP softphone issues

Hi, Do all sip phones supports Asterisk 13 + PJSIP?. I'm having issues of connecting ekiga softphone and Snom 710 to Asterisk. zoiper phones works fine. Asterisk is behind NAT. The phone sends register with expires 0. Best Regards, Madushan --

Re: [asterisk-users] PJSIP client - Incoming doesn't work after IP change

Hi, My snom phone doesn't work until i restart the phone after the ip change. Best Regards, Madushan On Thu, Mar 16, 2017 at 10:08 AM, Madushan Geethanga < mgliyanage...@gmail.com> wrote: > Hi, > > Thank you very much for the reply, is there a setting for it in snom > phones

Re: [asterisk-users] PJSIP client - Incoming doesn't work after IP change

me. > How often is the phone set to register? It can be that the IP is changing > in between registrations. > > > On Thu, Mar 16, 2017 at 12:12 AM, Madushan Geethanga < > mgliyanage...@gmail.com> wrote: > >> Hi All, >> >> I have a Asterisk 13 with PJSIP running on c

[asterisk-users] PJSIP client - Incoming doesn't work after IP change

Hi All, I have a Asterisk 13 with PJSIP running on cloud, and snom phones at client site, but when my clients home internet public ip changes incoming does not work. the PJSIP Contact seems to be not updated. is there a solution for this. I have allowed multiple registrations to the extension.