similar to: PJSIP client - Incoming doesn't work after IP change

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://123.231.72.210:33878> ---> > INVITE sip:0 at 54.206.59.252

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, did setting from_domain work? > > Best

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 header not the public > address. I have include

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

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 >>

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, 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 > >

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 Madushan Maybe 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

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

Le 28/02/2020 à 23:43, hw a écrit : > On Thursday, February 27, 2020 3:03:47 PM CET hw wrote: >> Hi, >> >> sometimes 'pjsip show registrations' shows registrations to the VOIP >> provider as Rejected. I have already added >> >> >> max_retries = 0 >> auth_rejection_permanent = no >> >> >> in pjsip_wizard.conf and still

I have a pjsip install that is not pulling it's realtime registrations from an ODBC database. When I have them directly pull from a MySQL database everything seems to work. I am having trouble finding a requirements document for the pjsip realtime registrations. Is there some kind of special config that has to be used to trigger the connection for realtime registrations over ODBC?

Ah ok, thanks. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp Sent: Monday, October 05, 2015 8:20 AM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-05 09:16 AM, Ryan, Travis wrote: [snip] > > > So

Hi, sometimes 'pjsip show registrations' shows registrations to the VOIP provider as Rejected. I have already added max_retries = 0 auth_rejection_permanent = no in pjsip_wizard.conf and still asterisk does not recover. I need asterisk to keep trying to register and to renew the registration without requiring manual intervention. How can I make asterisk do that?

Hello, I am slightly confused by the difference between chan_sip and pjsip. Especially the new (to me) objects aor and contact. I am having trouble mapping them to the typical SIP configuration settings on a phone. Suppose I have a phone with two line buttons, for two extension numbers. Now, I think that means two 'endpoints' in pjsip right? But what exactly is the difference between

hi, i want use asterisk+pjsip as voip client with multiple registrations (perf testing) i'm using this example configuration for one account [308] type=registration outbound_auth=308 server_uri=sip:308 at example.com:5060 client_uri=sip:308 at example.com:5060 [308](auth-userpass) username=308 password=pass [308](aor-single-reg) contact=sip:example.com:5060 [308](endpoint-basic)

---------------------------------------- From: "Joshua Colp" <jcolp at digium.com> Sent: Sunday, October 4, 2015 12:12 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-04 01:09 PM, Bryant Zimmerman wrote: > -- > Joshua > Thanks for your reply. It thought the same thing, but when I

Hello, I have some user that had have a hardwarephone and an softphone. I use pjsip driver and set "Max Contacts = 2" to have register both at the same time. But Only the softphone is ring. the hardware phone is mute. How can i fix this?

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp Sent: Sunday, October 04, 2015 12:44 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-04 01:42 PM, Bryant Zimmerman wrote: >

I use realtime for my Asterisk configuration and are now making the transition to Asterisk 13 and PJSIP. I used alchemy to set up my databases and I can now configure my endpoints. While trying to configure a trunk I can see that there is a database table called ps_registrations that should be used to make the registration to the provider but there is no corresponding entry in the