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On Fri, Sep 16, 2016 at 5:55 AM, Madushan Geethanga <mgliyanage.rc at gmail.com > wrote: > Hi, > > Tried with both softphone (Ekiga) and snom IP phone, contact header > contains the public IP. but from header contains the private IP. after > OPTIONS method sent by the server. client sends an Register with expires 0. > Ok,...

On Thu, Sep 15, 2016 at 8:38 AM, Madushan Geethanga <mgliyanage.rc at gmail.com > wrote: > Hi, > > Thanks for the reply. > > Yes my PABX is on the cloud when I try to register to the server, the > server sends registration OK with public address but OPTION method > includes the private address of the server in from hea...

Le 09/09/2016 ? 18:32, Madushan Geethanga a ?crit : > Hi, If you're not using RTP encryption did you uncheck the option in your RTP TAB from identity ? > > This is the log. ex dialling 0 from snom phone > > > <--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878 > <http://123.231.72.210:3387...

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-asterisk at tinsleyviaduct.com> wrote: > On Thu, 3 Mar 2016 08:21:14 +0530 > Madushan Geethanga <mgliyanage.rc at gmail.com> wrote: > > > 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 &g...

On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad <faheem2084 at gmail.com> wrote: > > > On Wednesday, 14 September 2016, Madushan Geethanga < > mgliyanage.rc at gmail.com> wrote: > >> 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 aster...

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 -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 disallow=all allow=ulaw allow=alaw transport=system-udp auth=2001

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. Best Regards, Madushan -------------- next part -------------- An HTML attachment was scrubbed... URL:

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? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Guys Which module on Asterisk is the one in charge of playing the VoiceMail Server Audio to the end customer? I have work with MRFP but is it a module included in the SW? Need and external source? BR Joaquin This email is confidential and may be subject to privilege. If you are not the intended recipient, please do not copy or disclose its content but contact the sender immediately upon

...aybe I was not clear ?. After SIP negotiation and SDP set up on the VoiceMail Server ?. Is there a file to specify a MGw (the machine that deliver RTP packages to end user)? From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Madushan Geethanga Sent: 15 July 2016 13:00 To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] VoiceMail Audio playing Hi, VoiceMailMain is used to retrieve voice mails http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMailMain Bes...