[asterisk-users] Help - Asterisk SIP Messages Parameter Modification

Dear All, I have a query. I want to know if there is any possiblity to modify SIP Messages Parameters using the asterisk CLI mode. I want to change the parameters for e.g in INVITE message. How it can be done in asterisk. Kindly assist me. Regards, *Waleed A. Khan* --

Re: [asterisk-users] Modify Contact in PJsip

Hi Joshua If i put the default_user option per endpoint would it work?  So what exactly does the contact_user option do? I know that in freeswitch there is the option extension-in-contact.We   basically need to achieve the same functionality  Thanks Original message From: Joshua

Re: [asterisk-users] Modify Contact in PJsip

Do you know if this can be achieved with the standard sip stack in asterisk? Kind Regards Andrew Colin Converged Telecoms (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Switchboard: +27 (0)10 591 4600 Email:  and...@convergedgroup.net Web:  

Re: [asterisk-users] Modify Contact in PJsip

Ok thanks Joshua Do you know what this error means when I dial out in pjsip and the call fails Unable to create request with auth.No auth credent als for any realms in challenge Kind Regards Andrew Colin Converged Telecoms (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09)

Re: [asterisk-users] pjsip show xxxx like endpoint?

George, and Matthew I can open an issue later today, but if you want to do it that would be awesome as well. Please post the issue number back to this thread so I can follow it. Ideally the Like would work with all pjsip show commands so we can reduce the list and drill down just like

Re: [asterisk-users] Modify Contact in PJsip

On 15-10-19 09:12 AM, Andrew Colin wrote: Do you know if this can be achieved with the standard sip stack in asterisk? If you are referring to chan_sip I don't believe so but it is possible there is some obscure option or method to do it that I am aware of. -- Joshua Colp Digium, Inc. |

Re: [asterisk-users] Modify Contact in PJsip

On 15-10-19 07:41 AM, Andrew Colin wrote: Hi Guys We are using the wizard to configure our pjsip trunk(see below) How do we get this setting to work contact_user=username We want to change the contact field in the sip invite to display the username of the trunk The Contact header can not

Re: [asterisk-users] Modify Contact in PJsip

On 15-10-19 08:17 AM, Andrew Colin wrote: Hi Joshua If i put the default_user option per endpoint would it work? No, it's a global only option. So what exactly does the contact_user option do? It sets the Contact user in an outbound registration so that the URI dialed by the remote SIP

Re: [asterisk-users] pjsip show xxxx like endpoint?

George and Mat Here is the link to the Jar Issue. https://issues.asterisk.org/jira/browse/ASTERISK-25477 Thanks Bryant From: "George Joseph" Sent: Sunday, October 18, 2015 10:17 PM To: "Asterisk Users Mailing

Re: [asterisk-users] Why I get repeat messages many times

No, It directly goes the context astsms when we send the message. but it still repeats the message sometimes. On Mon, Oct 19, 2015 at 3:25 PM, jg wrote: > > I am using the asterisk 13 and I config my dialplan for the SIP messaging > as the following : > >

[asterisk-users] Modify Contact in PJsip

Hi Guys We are using the wizard to configure our pjsip trunk(see below) How do we get this setting to work contact_user=username We want to change the contact field in the sip invite to display the username of the trunk [trunk_defaults](!) type = wizard transport = transport-udp

Re: [asterisk-users] Why I get repeat messages many times

I am using the asterisk 13 and I config my dialplan for the SIP messaging as the following : http://highsecurity.blogspot.com/2012/03/asterisk-10-110-sms-messaging-or-sip.html [astsms] exten => _.,1,NoOp(SMS receiving dialplan invoked) exten => _.,n,NoOp(To ${MESSAGE(to)}) exten =>

Re: [asterisk-users] Asterisk HA with heartbeat and systemd

If you’re still in the planning stage, there’s a lot more to think about. Your Asterisk failure detection will be very simplistic (is the process dead). Synchronization of data – without risking synchronization of corrupt data to a peer. Prevent a deteriorating/failing peer corruption from

[asterisk-users] Asterisk HA with heartbeat and systemd

Hello, I'm setting up an active-passive Asterisk solution on Debian Jessie platforms. I'm using heartbeat package. As I'm not yet familiar with either systemd or heartbeat, I've got a couple of questions: 1. At the moment, I'm using /usr/share/heartbeat/hb_standby or

[asterisk-users] Asterisk / PJSIP unexpectedly opens additional UDP port

Dear all, I have drastically stripped down my Asterisk installation to about 70 modules since I wanted it to be as slim as possible. Nevertheless, Asterisk still opens an unwanted port: root@spock:/install/asterisk-13.6.0# netstat -apnv|grep asterisk ... udp0 0 192.168.20.48:5060

[asterisk-users] ARI all subscribe

I wasn't able to make it back to the devcon after lunch or to as many of the talks as I'd have liked (the excessive a/c exacerbated by symptoms enough to be painful), so I probably missed something relevant to this... What is the syntax of an ALL subscription websocket url in ari? I'd like to