Re: [asterisk-users] Problem with Cisco Phones
Apparently this is a known problem past v8 firmware: http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update- version-9/ I've done some more playing about and what I've noticed is that even when using TCP SIP on the 8.x Firmware conferencing doesn’t work - making it use UDP fixes this. So has anyone managed to get the 9.x firmware working with UDP? Possibly worth a try to see if this resolves the issue? This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Cisco Phones
If I remember correctly, 9.x firmware dropped UDP support altogether. On Thu, Jan 22, 2015 at 4:31 AM, Jordan Cook - Gyron Networks jordan.c...@gyron.net wrote: Apparently this is a known problem past v8 firmware: http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update- version-9/ I've done some more playing about and what I've noticed is that even when using TCP SIP on the 8.x Firmware conferencing doesn’t work - making it use UDP fixes this. So has anyone managed to get the 9.x firmware working with UDP? Possibly worth a try to see if this resolves the issue? This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CALLERID(ani2) inserting
I checked
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
But I cannot find a way to insert CALLERID(ani2), which I can read, but
when I try to set it for a new call, I get a runtime error.
This information, known as isup-oli comes embedded in the From header,like
this
sip:9087311878@64.45.157.104:5060;isup-oli=62;tag=sansay1724414rdb124
and it can be read by using
Set(var=${CALLERID(ani2)}
But how do we add that information to the outbound INVITE? This is
critical in the toll-free industry and call-from-jail industries.
Thanks for your help.
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[asterisk-users] New Feature CALLERID(ani2) read/write
Two years ago we added logic to parse the isup-oli parameters, that arrive
as part of the FROM Sip header. We need to finish the job and allow setting
of this parameter for outbound calling, both in traditional SIP channel and
PJSIP, which I believe will replace all instances of the old SIP channel
soon.
Right now, if we try to set CALLERID(ani2)=$
{var}
, there is a runtime error because this variable is read-only.
The business community around Asterisk needs this feature and there is no
known workaround.
I am also writing about this to the developer list. If somebody wants to
propose a patch, I can contribute to the bounty.
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[asterisk-users] DAHDI-Linux and DAHDI-Tools 2.10.1-rc2 Now Available
The Asterisk Development Team has announced the releases of:
DAHDI-Linux-v2.10.1-rc2
DAHDI-Tools-v2.10.1-rc2
dahdi-linux-complete-2.10.1-rc2+2.10.1-rc2
This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete
* Updates to allow dahdi to compile/run against kernel versions up to 3.19
* xpp firmware and startup scripts improvements
Shortlog of dahdi-linux changes since v2.10.0.1:
Shaun Ruffell (4):
dahdi: smp_mb_{before,after}_clear_bit - smp_mb_{before,after}_atomic.
build_tools/dkms-helper: Use bash to process dkms-helper script.
dahdi_dynamic: Release reference count on network device when destroying
dynamic spans.
dahdi: struct file.f_dentry macro was removed in kernel 3.19
Tzafrir Cohen (3):
xpp: FPGA_1161.201.hex: module types detection
xpp: firmware: 203 as alias to (newer) 201
xpp: firmware: a stray ^Z in FPGA_1161.201.hex
Shortlog of dahdi-tools changes since v2.10.0.1:
Oron Peled (7):
xpp: astribank_is_starting: improve '-v' output
xpp: waitfor_xpds: expansion error with no ABs
xpp: waitfor_xpds: assume astribank_is_starting exists
xpp: can use modern Asterisk hotplug support
xpp: waitfor_xpds: documentation
xpp/astribank_hook: remove Astribank initialization
xpp: waitfor_xpds: Always remove Astribank semaphore
Tzafrir Cohen (2):
astribank_hook: remove useless 'time'
no astribank_is_starting with hotplug asterisk
The diffstat from the dahdi-linux v2.10.0.1 release:
build_tools/dkms-helper | 2 +-
drivers/dahdi/dahdi-base.c| 4 +
drivers/dahdi/dahdi_dynamic_eth.c | 5 +-
drivers/dahdi/wcaxx-base.c| 2 +-
drivers/dahdi/wcte12xp/base.c | 4 +-
drivers/dahdi/wcte13xp-base.c | 4 +-
drivers/dahdi/wcte43x-base.c | 2 +-
drivers/dahdi/xpp/firmwares/FPGA_1161.201.hex | 31193
drivers/dahdi/xpp/firmwares/Makefile | 9 +-
include/dahdi/kernel.h|12 +
10 files changed, 15583 insertions(+), 15654 deletions(-)
The diffstat from the dahdi-tools v2.10.0.1 release:
dahdi.init | 8 +++
init.conf.sample| 1 +
xpp/astribank_hook | 117 +++-
xpp/astribank_is_starting.c | 2 +
xpp/waitfor_xpds| 22 +
5 files changed, 117 insertions(+), 33 deletions(-)
For a full list of changes in these releases, please see the shortlog at:
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.10.1-rc2
http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.10.1-rc2
Issues found in this release can be reported in the DAHDI-Linux [1] and
DAHDI-Tools [2] projects at https://issues.asterisk.org/jira
[1] https://issues.asterisk.org/jira/browse/DAHLIN
[2] https://issues.asterisk.org/jira/browse/DAHTOOL
Thank you for your continued support of Asterisk!
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[asterisk-users] Maintenance for community services tonight (January 22nd)
Tonight a few community services will have intermittent availability due to maintenance. This maintenance will begin at 8:00 PM CST and should last no longer than one hour, ending around 9:00 PM CST. The affected services include: * bamboo.asterisk.org * code.asterisk.org * wiki.asterisk.org Thank you for your support! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR and confbridge
Good evening all. I am having issues with CDR and confbridge. When the first call is placed into conference CDR stops tracking time. If I hang the call up the billsec reported is only up to the the time before the call enters the bridge. However if a second call joins the bridge the full amount of time is reported for the first call but not for the second. This looks inconsistent and more like a bug Has anyone else experienced issues with CDR and confbridge? Thanks-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CALLERID(ani2) inserting
Hi
According to this:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables
It is read only.
On 22 January 2015 at 16:22, CDR vene...@gmail.com wrote:
I checked
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
But I cannot find a way to insert CALLERID(ani2), which I can read, but
when I try to set it for a new call, I get a runtime error.
This information, known as isup-oli comes embedded in the From header,like
this
sip:9087311878@64.45.157.104:5060;isup-oli=62;tag=sansay1724414rdb124
and it can be read by using
Set(var=${CALLERID(ani2)}
But how do we add that information to the outbound INVITE? This is
critical in the toll-free industry and call-from-jail industries.
Thanks for your help.
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