Re: [asterisk-users] Bridge configuration in Asterisk 13 [Spam score:8%] [Spam score:8%]

Thank you once again Richard. I think that covers all my confusion. Regards, Patrick. From: Richard Mudgett rmudg...@digium.commailto:rmudg...@digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Date:

Re: [asterisk-users] T.38 not working - help needed with log interpretation

Hi, On Tue, Dec 2, 2014 at 9:24 AM, Recursive li...@binarus.de wrote: - Packets 14313, 14314: The provider re-invites asterisk for T.38 (confirmed by viewing the packet's details), asterisk answers Trying ... to the provider - Packets 14315, 14321, 14322: Asterisk re-invites the local

[asterisk-users] UPDATE instead of RE-INVITE

HI, It is possible to disable/remove INVITE method in 200 OK responses? I want to receive from another SIP/PBX the the media path redirection in a UPDATE message rather than an INVITE, after calls are transfered. My asterisk is version 11. e.g: SIP/2.0

[asterisk-users] PJSIP configuration question

I'm working with a SIP provider to try and transition our sip connection with them to PJSIP. I thought I had transitioned the settings correctly, but whenever I attempt an Originate it never even tries to send any PJSIP messages. I'm currently running Asterisk 13.0.0. Anyone have any

Re: [asterisk-users] PJSIP configuration question

Kia ora, Dan Cropp wrote: I’m working with a SIP provider to try and transition our sip connection with them to PJSIP. I thought I had transitioned the settings correctly, but whenever I attempt an Originate it never even tries to send any PJSIP messages. What dial string are you providing to

Re: [asterisk-users] PJSIP configuration question

Thank you for the speedy reply. My originate string is something like the following where x is really the sip provider's supplied IP address 1234567890 is really the phone number I am dialing PJSIP/outbound.vitelity.net/1234567890 In the chan_sip based solution, it's...

Re: [asterisk-users] PJSIP configuration question

I should mention, I am actually sending this via AMI in both the chan_sip and the pjsip case. Pjsip originate... Action: Originate ActionID: S8 Channel: PJSIP/outbound.vitelity.net/1234567890 Exten: createcall Context: TestApp Priority: 1 Timeout: 6 CallerID: Dan Cropp1234 Variable:

[asterisk-users] Failed to authenticate device - who?

I have a bunch of these in my logs: [Dec 9 08:21:21] NOTICE[-1][C-0285] chan_sip.c: Failed to authenticate device einsteinsip:einstein@98.158.139.74;tag=65696e737465696e0131323530333532333739 The problem is that I already know my own IP address. How do I determine the address of the host

[asterisk-users] Asterisk 11.6-cert9, 11.14.2, 12.7.2, 13.0.2 Now Available (Security Release)

The Asterisk Development Team has announced security releases for Certified Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2. These releases are available for immediate download at

[asterisk-users] AST-2014-019: Remote Crash Vulnerability in WebSocket Server

Asterisk Project Security Advisory - AST-2014-019 ProductAsterisk SummaryRemote Crash Vulnerability in WebSocket Server Nature of Advisory Denial of Service

[asterisk-users] PJSIP configuration question

Not sure why, but Vitelity changed the settings to IP based authentication on me. Here's the new sip.conf settings they sent me. type=friend dtmfmode=auto host=64.2.142.93 allow=all nat=yes canreinvite=no trustrpid=yes sendrpid=yes When I use these settings to originate calls using the

Re: [asterisk-users] PJSIP configuration question

On Wed, Dec 10, 2014 at 1:27 PM, Dan Cropp d...@amtelco.com wrote: Not sure why, but Vitelity changed the settings to IP based authentication on me. Here's the new sip.conf settings they sent me. type=friend dtmfmode=auto host=64.2.142.93 allow=all nat=yes canreinvite=no trustrpid=yes

Re: [asterisk-users] PJSIP configuration question

Thanks George. That was the ip address I was given. Unfortunately, my contact at Vitelity is gone for the day so I can’t verify it with him. I added the qualify_frequency as you suggested and it does appear that I have something configured incorrectly…. --- Transmitting SIP request (483

[asterisk-users] OT: Openfire - new version of B9 plugin released

Hey people. I just released B9 version 0.3. This version contains new commands (create conference, invite conference), but the major feature is socket connection that can be configured in a console admin page. In the page you can enable socket connections, change ip and port for binding and

Re: [asterisk-users] PJSIP configuration question

On Wed, Dec 10, 2014 at 2:03 PM, Dan Cropp d...@amtelco.com wrote: Thanks George. That was the ip address I was given. Unfortunately, my contact at Vitelity is gone for the day so I can’t verify it with him. I added the qualify_frequency as you suggested and it does appear that I have

Re: [asterisk-users] PJSIP configuration question

snip I translated those settings to the following for pjsip.conf... [transport1] type = transport bind = 0.0.0.0 protocol = udp [outbound.vitelity.net] type = aor remove_existing = yes contact = sip:64.2.142.93@5060 This is incorrect. The contact should be: contact = sip:64.2.142.93 It