Posts Tagged isdn
libpri 1.4.12 Now Available
The Asterisk Development Team announces the release of libpri version 1.4.12. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/
The following are some of the issues resolved in this release:
- Add call transfer exchange of subaddresses support and fix PTMP call transfer signaling.
- Invalid PTMP redirecting signaling as TE towards NT.
- Add Q931_IE_TIME_DATE to CONNECT message when in network mode.
(issue #18047 (JIRA PRI-114). Reported by: wuwu. Patched by rmudgett) - Swap of master/slave in pri_enslave() incorrect.
(issue #18769 (JIRA PRI-120). Reported by: jcollie. Patched by jcollie) - Fix I-frame retransmission quirks.
- Crash if NFAS swaps D channels on a call with an active timer.
- DMS-100 not receiving caller name anymore.
(issue #18822 (JIRA PRI-121). Reported by: cmorford. Patched by rmudgett) - B channel lost by incoming call in BRI NT PTMP mode.
- Implement the mandatory T312 timer for NT PTMP broadcast SETUP calls.
This release contains several new features, among them:
- ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support
- ETSI Advice Of Charge (AOC) support
- ETSI Explicit Call Transfer (ECT) support
- ETSI Call Waiting support for ISDN phones
- ETSI Malicious Call ID support
- Add Display IE text handling options.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1….
Thank you for your continued support of Asterisk!
Asterisk 1.8.0-Beta3 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta3.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.
Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.
http://www.asterisk.org/asterisk-versions
This release contains fixes since the last beta release as reported by the community. A sampling of the changes in this release include:
- Fix a regression where HTTP would always be enabled regardless of setting.
(Closes issue #17708. Reported, patched by pabelanger) - ACL errors displayed on screen when using dynamic_exclude_static in sip.conf
(Closes issue #17717. Reported by Dennis DeDonatis. Patched by mmichelson) - Support “channels” in addition to “channel” in chan_dahdi.conf.
(https://reviewboard.asterisk.org/r/804) - Fix parsing error in sip_sipredirect(). The code was written in a way that did a bad job of parsing the port out of a URI. Specifically, it would do badly when dealing with an IPv6 address.
(Closes issue #17661. Reported by oej. Patched by mmichelson) - Fix inband DTMF detection on outgoing ISDN calls.
(Patched by russellb and rmudgett) - Fixes issue with translator frame not getting freed. This issue prevented g729 licenses from being freed up.
(Closes issue #17630. Reported by manvirr. Patched by dvossel) - Fixed IPv6-related SIP parsing bugs and updated documention.
(Reported by oej. Patched by sperreault) - Add new, self-contained feature FIELDNUM(). Returns a 1-based index into a list of a specified item. Matches up with FIELDQTY() and CUT().
(Closes #17713. Reported, patched by gareth. Tested by tilghman)
Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:
- Secure RTP
- IPv6 Support in the SIP Channel
- Connected Party Identification Support
- Calendaring Integration
- A new call logging system, Channel Event Logging (CEL)
- Distributed Device State using Jabber/XMPP PubSub
- Call Completion Supplementary Services support
- Advice of Charge support
- Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta3
Thank you for your continued support of Asterisk!
Libpri 1.4.12-Beta1 Now Available
The Asterisk Development Team has announced the release of libpri version 1.4.12-beta1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/
This beta release contains some fixes and several new features, among them:
- ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support
- ETSI Advice Of Charge (AOC) support
- ETSI Explicit Call Transfer (ECT) support
- ETSI Call Waiting support for ISDN phones
- ETSI Malicious Call ID support
For a full list of changes in the current release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.12-beta1
Thank you for your continued support of Asterisk!
Using the Patton SN4554 for ISDN with Elastix
For any business migrating to SIP, the Patton SN4554 is a brilliant way to bring two ISDN lines (4-channels) in to their new PBX system, especially considering you don’t have to break open your PBX Server to install a PCI card.
This basic How-To shows you how to set it up to work with your Elastix system:
First off, you’ll want the following config which is a bit of a mash-up from some other documentation on making it work with an Aastra 800 (Include the header):
#----------------------------------------------------------------#
# #
# Template for SN4554/2BIS/UI to use with Elastix #
# Use with firmware R5.1 or higher #
# #
# This template uses dhcp to retrieve an IP address. Comments in #
# the file indicate what to change (Start with '#') #
# #
# See the SmartWare Software Configuration guide for command #
# details (http://www.patton.com/manuals/SCG-r52.pdf) #
# Modified by Josiah Spackman #
# http://chillingsilence.wordpress.com #
#----------------------------------------------------------------#
cli version 3.20
webserver port 80 language en
system
ic voice 0
low-bitrate-codec g711alaw64k
system
clock-source 1 bri 0 0
clock-source 2 bri 0 1
profile ppp default
profile tone-set default
profile voip default
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20
fax transmission 1 relay t38-udp
profile pstn default
profile sip default
profile aaa default
method 1 local
method 2 none
context ip router
interface IF_IP_WAN
# replace 'dhcp' with your fix IP if needed, e.g. "ipaddress 172.16.1.20 255.255.0.0"
ipaddress dhcp
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu
context ip router
# uncomment the following line if you need to add routing table entries
# route 0.0.0.0 0.0.0.0 172.16.1.1
context cs switch
national-prefix 0
international-prefix 00
routing-table called-e164 RT_ISDN_TO_SIP
# 'T' in the following lines specifies "digit collection". The time for collection can be adjusted replacing 'T' wih 'T2' for 2 seconds.
route T dest-interface IF_SIP
routing-table calling-e164 RT_SIP_TO_ISDN
route default dest-service SV_HUNT_PSTN MP_Unknown-Subscriber
# This mapping table sets the ISDN type of number for calls towards ISDN to 'subscriber'
mapping-table calling-e164 to calling-type-of-number MP_Unknown-Subscriber
map default to subscriber
interface isdn IF_ISDN_0
route call dest-table RT_ISDN_TO_SIP
interface isdn IF_ISDN_1
route call dest-table RT_ISDN_TO_SIP
interface sip IF_SIP
bind context sip-gateway GW_SIP
route call dest-table RT_SIP_TO_ISDN
# This is the IP of your Asterisk. Replace with domain name if DNS server is available
remote 192.168.0.250
service hunt-group SV_HUNT_PSTN
drop-cause normal-unspecified
drop-cause no-circuit-channel-available
drop-cause network-out-of-order
drop-cause temporary-failure
drop-cause switching-equipment-congestion
drop-cause access-info-discarded
drop-cause circuit-channel-not-available
drop-cause resources-unavailable
route call 1 dest-interface IF_ISDN_0
route call 2 dest-interface IF_ISDN_1
context cs switch
no shutdown
# The parameters realm, username and password have to match your Asterisk configuration
authentication-service AUTH_AASTRA800
realm 1 smartnode-gw
username patton password 6953789
# The parameters domain, username, identity (=username) below have to match your Aastra800 configuration
location-service LS_AASTRA800
domain 1 smartnode-gw
identity-group default
authentication inbound
authenticate 1 authentication-service AUTH_AASTRA800 patton 6953789
identity 6953789
authentication inbound
authenticate 1 authentication-service AUTH_AASTRA800 patton 6953789
registration inbound
context sip-gateway GW_SIP
interface WAN
bind interface IF_IP_WAN context router port 5060
context sip-gateway GW_SIP
bind location-service LS_AASTRA800
no shutdown
port ethernet 0 0
medium auto
encapsulation ip
bind interface IF_IP_WAN router
no shutdown
port bri 0 0
clock auto
encapsulation q921
q921
protocol pp
uni-side auto
encapsulation q931
q931
protocol dss1
uni-side user
bchan-number-order ascending
encapsulation cc-isdn
bind interface IF_ISDN_0 switch
port bri 0 0
no shutdown
port bri 0 1
clock auto
encapsulation q921
q921
protocol pp
uni-side auto
encapsulation q931
q931
protocol dss1
uni-side user
bchan-number-order ascending
encapsulation cc-isdn
bind interface IF_ISDN_1 switch
port bri 0 1
no shutdown
Copy this all into a notepad window and save it as “SN4554.cfg”
What you’re going to want to modify is the references to “192.168.0.250″ and point it to your Elastix / Asterisk system.
All the ISDN ports are set to “Point-to-Point”. You can adjust the “pp” settings to “pmp” if you require “Point-to-Multipoint”, though to be honest ISDN isn’t my strong point so I’m not entirely sure how relevant that is.
This will leave the WAN port using DHCP, hopefully you’ve got a semi-intelligent DHCP server that will give out the same IP each time.
Open the WebGUI of your SN4554 up, login with the default Username “administrator” and a blank password.
Click on “Import / Export” on the left-hand side, then hit the Import Configuration tab.
Select the file, import it, then reload the device.
Now, in asterisk you want to add a new SIP trunk. Name it “ISDN” and put the following in the SIP PEER Details:
username=patton type=friend secret=6953789 qualify=1200 insecure=very host=192.168.0.141 dtmfmode=RFC2833 disallow=all context=from-pstn canreinvite=no allow=alaw&g729
There’s no register string or USER details, though it may be worth specifying the max channels as “4″.
You’ll need to adjust the “host” setting to the WAN IP Address of your Patton box, as we’re semi-insecure with such a basic password.
You *could* adjust the password and username in the config etc as applicable, but specifying the host should be enough, especially considering (in my instance) the whole system is LAN-accessible only.
When you’re finished, you should be able to login to your Elastix system via SSH and run:
asterisk -rx ’sip show peers’
And see:
ISDN/patton 192.168.0.141 N 5060 OK (19 ms)
If you do, congratulations, you’re ready to go!
Now you just need to setup inbound and outbound routes as applicable.
If this was useful to you, please leave a comment and say hi.
I’d also like to thank Byron from SnapperNet in New Zealand, he’s been such a great help, and also provided me with the initial configuration samples.
This How-To has also been re-posted here: http://chillingsilence.wordpress.com/2010/05/06/using-the-patton-sn4554-for-isdn-with-elastix/
Asterisk 1.4.26 released
The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.26.
Asterisk 1.4.26 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
This release resolves a large assortment of issues reported by the community.
For a summary of the changes in this release, please see the release summary:
http://svn.asterisk.org/svn/asterisk/tags/1.4.26/asterisk-1.4.26-summary.txt
For a full list of changes in this release, please see the ChangeLog:
http://svn.asterisk.org/svn/asterisk/tags/1.4.26/ChangeLog
The following list of issues were resolved with the participation of the community, and this release would not have been possible without your help!
- Fix handling of the ‘state_interface’ option of the ‘queue add member’ CLI command.
(closes issue #15181. Reported and tested by loloski. Patch by seanbright) - Fix a possible crash in pbx_spool.
(closes issue #15072. Reported, and patched by garlew) - MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport.
(closes issue #14659. Reported, patch, and testing by klaus3000) - Don’t fast forward past the end of a message.
(closes issue #14554. Reported and patched by lacoursj) - Prevent phantom calls to queue members.
(closes issue #14631. Reported and patched by latinsud) - No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller.
(closes issues #15420, #15416, #15389, #15205. Reported by scottbmilne, avinoash, alecdavis, vinsik. Tested by scottbmilne, alecdavis. Patched by alecdavis)
Thank you for your continued support of Asterisk!

