Hello Treesa,
You should take a look at libss7 of course for the ss7 messaging but i
am mostly sure the BSSAP Part isnt included there.
But if i remember it right you only need BSSAP between the MSC and the
bases (BSS) not behind the MSC.
And i guess you want to use asterisk to act as an SS7 -
And i guess you want to use asterisk to act as an SS7 - SIP translator
towards the PSTN so there shoudlnt be a BSSAP involved only plain ISUP which
could be handled from libss7.
Assuming the above I can hint you to this
On 2014-02-26 16:17, Frederic Van Espen wrote:
And i guess you want to use asterisk to act as an SS7 - SIP
translator
towards the PSTN so there shoudlnt be a BSSAP involved only plain ISUP
which
could be handled from libss7.
Assuming the above I can hint you to this
Thank you very much for the offer, but unless these would become the
official packages, I don't think that would serve the purpose I'm trying to
raise in this thread, which is to simplify the official community install
method to a single repository. For my cases I can work around this
complexity,
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3263/
---
(Updated Feb. 26, 2014, 10:04 a.m.)
Status
--
This change has been
On 2014-02-26 16:38, Kaloyan Kovachev wrote:
On 2014-02-26 16:17, Frederic Van Espen wrote:
And i guess you want to use asterisk to act as an SS7 - SIP
translator
towards the PSTN so there shoudlnt be a BSSAP involved only plain ISUP
which
could be handled from libss7.
Assuming the above I
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3264/
---
(Updated Feb. 26, 2014, 10:29 a.m.)
Status
--
This change has been
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3267/#review10958
---
/branches/12/channels/chan_pjsip.c
On Feb. 24, 2014, 10:51 p.m., Joshua Colp wrote:
/branches/11/res/res_rtp_asterisk.c, lines 554-560
https://reviewboard.asterisk.org/r/3256/diff/1/?file=54416#file54416line554
Agreed, and pjnath does not provide a mechanism to do just that without
destroying/re-creating as Matt
On Wed, Feb 26, 2014 at 3:38 PM, Kaloyan Kovachev kkovac...@varna.net wrote:
I would like to use this opportunity to ask for review on
https://reviewboard.asterisk.org/r/2150/ and
https://reviewboard.asterisk.org/r/2170/
Their inclusion in Asterisk will close not only SS7-27, but almost all
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3240/#review10961
---
Ship it!
Ship It!
- Joshua Colp
On Feb. 21, 2014, 5 p.m.,
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3237/
---
(Updated Feb. 26, 2014, 4:42 p.m.)
Status
--
This change has been
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2904/
---
(Updated Feb. 26, 2014, 5:04 p.m.)
Status
--
This change has been
On 02/26/2014 03:38 PM, Kaloyan Kovachev wrote:
On 2014-02-26 16:17, Frederic Van Espen wrote:
And i guess you want to use asterisk to act as an SS7 - SIP translator
towards the PSTN so there shoudlnt be a BSSAP involved only plain
ISUP which
could be handled from libss7.
Assuming the above
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3193/
---
(Updated Feb. 26, 2014, 11:22 a.m.)
Status
--
This change has been
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1073/
---
(Updated Feb. 26, 2014, 5:43 p.m.)
Status
--
This change has been
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1194/
---
(Updated Feb. 26, 2014, 5:46 p.m.)
Status
--
This change has been
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3243/
---
(Updated Feb. 26, 2014, 11:51 a.m.)
Review request for Asterisk
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3191/
---
(Updated Feb. 26, 2014, 12:10 p.m.)
Review request for Asterisk
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3267/
---
(Updated Feb. 26, 2014, 12:41 p.m.)
Review request for Asterisk
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3267/#review10962
---
/branches/12/channels/chan_pjsip.c
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3267/#review10963
---
Ship it!
Ship It!
- Joshua Colp
On Feb. 26, 2014, 6:52
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3267/
---
(Updated Feb. 26, 2014, 1:01 p.m.)
Status
--
This change has been
Hi,
Really digging in the crates for this one! As of yesterday
FreeSWITCH supports AES-GCM mode with SRTP:
http://jira.freeswitch.org/browse/FS-5937
We have a patch for pjsip, which leaves Asterisk as one of the few
open source RTP/SRTP implementations I care about without support for
On Wed, Feb 26, 2014 at 08:48:49AM -0500, Jared Smith wrote:
On Tue, Feb 25, 2014 at 3:23 PM, Ben Langfeld b...@langfeld.me wrote:
Unfortunately those packages are of Asterisk 1.8:
http://dl.fedoraproject.org/pub/epel/6/x86_64/repoview/asterisk.html.
That's a total no-go for me, I'm
On Feb. 23, 2014, 9:04 p.m., Corey Farrell wrote:
/trunk/channels/sip/reqresp_parser.c, lines 103-118
https://reviewboard.asterisk.org/r/3250/diff/1/?file=54390#file54390line103
I feel this section should only apply when scheme is 'tel:'. I'm
concerned with changes to how sip
Hi!
I've been working on some signaling-related problems (missing or extra
answer signal etc.) and I must admit that I still don't understand some
details of Asterisk signaling.
For example, I don't understand the following part of log, created by
simultaneously enabling IAX2 debug on a
27 matches
Mail list logo
