Somewhere in my mail archive I have a permission from Broadsoft or
Cylantro to implement this - before they
published it as a draft. They wanted Asterisk to include this
functionality. After that, they published it openly
as a draft, so there should be no problems.
It's probably also
]
From: Raj Jain [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 02, 2008 9:58 AM
To: [EMAIL PROTECTED]; Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Implementation of Broadsoft Sip Access in
Asterisk to enable SLA for Sipura/Linksys
Gene
On Dec 28, 2007 6:22 AM, Kamanashis Roy Shuva [EMAIL PROTECTED]
wrote:
Hi,
I am expecting someone to discuss this who have at least a little
idea about sip actually.
You're already discussing this w/ Olle J., who is lead developer and a SIP
expert in the Asterisk community :-)
I guess
] On Behalf Of
Alex Balashov
Sent: Friday, November 02, 2007 6:42 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] UAC leg cancel on early media / MoH.
Ah, thank you!
This was in 1.2.24. Do you suppose this behaviour differs in 1.4.x?
On Fri, 2 Nov 2007, Raj Jain
Edwin,
You are running into a RE-INVITE glare scenario. The Asterisk boxes
facing each other are racing to send RE-INVITE to each other to drop
the RTP hairpin. The Asterisk 1.4 does not retransmit a RE-INVITE on
receving a 491 response. It is treating 491 as a permanent failure and
therefore
I've seen that, in main/utils.c, threads are created with a stack size
limited to 240KBytes. Probably this is the reason why my asterisk
application often crashes with a SIGSEGV... (yes, I'm using large
buffers).
So I'm wondering what's the reason behind this limit and if it can be
I think its the missing Contact in MERA's 200 OK to Asterisk's RE-INVITE.
Contact is mandatory in 2XX responses.
Raj
On 4/2/07, Litnitchii Alexandr [EMAIL PROTECTED] wrote:
Hi all.
I have an issue sending/receiving faxes in next environment:
PSTN - Cisco (t.38 origination) - Mera (t.38
