Re: [asterisk-users] Asterisk Release 20.3.1
Le 07/07/2023 à 12:49, Joshua C. Colp a écrit : On Fri, Jul 7, 2023 at 6:40 PM Jean-Denis Girard <mailto:jd.gir...@sysnux.pf>> wrote: There seems to be a problem with the tar.gz archive on github. It's correct on downloads.asterisk.org <http://downloads.asterisk.org>. Can you be more specific? They are identical and the same tarball. I just downloaded both from each place and confirmed that, and confirmed they both extract fine. Downloading from github (I tried 5 times), I get: 10353870 7 juil. 13:44 'asterisk-20.3.1(1).tar.gz' 10353870 7 juil. 13:46 'asterisk-20.3.1(2).tar.gz' sha256sum is: aec7271fda5eb1e185bb94f3f52977c636783bd456e9c361dd853cd0eba10203 Extracting is fine. Downloading from asterisk.org, I get: 28176262 7 juil. 11:34 asterisk-20.3.1.tar.gz 5d7dea82b11ce97eec294ba0234c3a68fe2f05065c04a4279baa4a4442f4f628 Bien cordialement, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Release 20.3.1
There seems to be a problem with the tar.gz archive on github. It's correct on downloads.asterisk.org. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 Le 07/07/2023 à 10:10, Asterisk Development Team a écrit : The Asterisk Development Team would like to announce security release Asterisk 20.3.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.3.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security advisories were resolved in this release: https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm Change Log for Release 20.3.1 Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: - apply_patches: Use globbing instead of file/sort. - apply_patches: Sort patch list before applying - pjsip: Upgrade bundled version to pjproject 2.13.1 User Notes: - ### res_http_media_cache: Introduce options and customize The res_http_media_cache module now attempts to load configuration from the res_http_media_cache.conf file. The following options were added: * timeout_secs * user_agent * follow_location * max_redirects * protocols * redirect_protocols * dns_cache_timeout_secs - ### format_sln: add .slin as supported file extension format_sln now recognizes '.slin' as a valid file extension in addition to the existing '.sln' and '.raw'. - ### bridge_builtin_features: add beep via touch variable Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval) Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds will result in a periodic beep being played to the monitored channel upon MixMontior/Monitor feature start. If an interval less than 5 seconds is specified, the interval will default to 5 seconds. If the value is set to an invalid interval, the default of 15 seconds will be used. - ### app_senddtmf: Add SendFlash AMI action. The SendFlash AMI action now allows sending a hook flash event on a channel. - ### res_mixmonitor: MixMonitorMute by MixMonitor ID It is now possible to specify the MixMonitorID when calling the manager action: MixMonitorMute. This will allow an individual MixMonitor instance to be muted via ID. The MixMonitorID can be stored as a channel variable using the 'i' MixMonitor option and is returned upon creation if this option is used. As part of this change, if no MixMonitorID is specified in the manager action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor audiohooks on the channel. Previous behavior would set the flag on the first MixMonitor audiohook found. - ### pbx_dundi: Add PJSIP support. DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the pjsip_outgoing_endpoint option to be set in dundi.conf. - ### test.c: Fix counting of tests and add 2 new tests The "tests" attribute of the "testsuite" element in the output XML now reflects only the tests actually requested to be executed instead of all the tests registered. The "failures" attribute was added to the "testsuite" element. Also added two new unit tests that just pass and fail to be used for testing CI itself. - ### cli: increase channel column width This change increases the display width on 'core show channels' amd 'core show channels verbose' For 'core show channels', the Channel name field is increased to 64 characters and the Location name field is increased to 32 characters. For 'core show channels verbose', the Channel name field is increased to 80 characters, the Context is increased to 24 characters and the Extension is increased to 24 characters. Upgrade Notes: Closed Issues: - #193: [bug]: third-party/apply-patches doesn't sort the patch file list before applying -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Compile Asterisk without CPU specific extensions/optimizations
Le 30/03/2020 à 10:52, Telium Technical Support a écrit : > Is it possible to configure Asterisk to NOT use CPU specific > instructions/optimizations so that the executable is portable? Menuselect -> Compiler Flags -> BUILD_NATIVE not selected. Bien cordialement, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP wizard reload not reloading ?
Le 31/07/2019 à 08:44, George Joseph a écrit : > Go ahead and open an issue for this at https://issues.asterisk.org. > > We have an internal issue for something similar so I'll link them together. https://issues.asterisk.org/jira/browse/ASTERISK-28492 Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP wizard reload not reloading ?
Le 25/07/2019 à 18:49, Jean-Denis Girard a écrit :
> Hi list,
>
> I'm having a strange problem when using pjsip wizard and reloading
> ("pjsip reload" on CLI): some data (specifically endpoint/pickup_group)
> is not modified.
Am I the only one experiencing this problem? Or nobody uses call_group /
pickup_group on Asterisk-16?
Thanks,
--
Jean-Denis Girard
SysNux Systèmes Linux en Polynésie française
https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527
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[asterisk-users] PJSIP wizard reload not reloading ?
Hi list,
I'm having a strange problem when using pjsip wizard and reloading
("pjsip reload" on CLI): some data (specifically endpoint/pickup_group)
is not modified.
For example, initially I have empty pickup group:
tiare*CLI> pjsip show endpoint xxx
...
pickup_group :
...
Then, I add endpoint/pickup_group = 0,3 to pjsip_wizard.conf, and
reload: pickup_group remains empty.
Then, if I change the line in pjsip_wizard.conf to
endpoint/pickup_group = 0, 3
^ note the space here!
then reload, and I get what was expected:
tiare*CLI> pjsip show endpoint xxx
...
pickup_group : 0, 3
...
I have seen this problem on Asterisk-16 only (up to latest 16.5.0).
The modified configuration file is included from
/etc/asterisk/pjsip_wizard.conf:
#include astportal/pjsip_wizard.conf
pjsip reload has default definition in cli_aliases.conf:
pjsip reload=module reload res_pjsip.so
res_pjsip_authenticator_digest.so res_pjsip_endpoint_identifier_ip.so
res_pjsip_mwi.so res_pjsip_notify.so res_pjsip_outbound_publish.so
res_pjsip_publish_asterisk.so res_pjsip_outbound_registration.so
Did I miss something, or should I open an issue?
Thanks,
--
Jean-Denis Girard
SysNux Systèmes Linux en Polynésie française
https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527
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Re: [asterisk-users] ARI libraries?
Le 20/07/2019 à 12:21, Tony Mountifield a écrit :
> What is the bug with channel variables? Do you have a fix for it?
Channels variables caused an error, my fix is in aioswagger11/client.py
(line 80) :
elif param['paramType'] == 'body':
<if not data:
<data = {}
<data[pname] = value
>if isinstance(value, dict):
>if data:
>data.update(value)
>else:
>data = value
>else:
>raise TypeError("Parameters of type 'body'
require dict input")
I know, I should fork and make a pull request...
Best regards,
--
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SysNux Systèmes Linux en Polynésie française
https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527
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Re: [asterisk-users] ARI libraries?
Hi Tony, Le 20/07/2019 à 06:29, Tony Mountifield a écrit : > Are there any other languages/libraries I should be considering? Same here, after years of AGI / AMI, I recently made my first project using ARI on Asterisk-16. I love Python, and was disappointed to find that Python ARI looks abandoned. Then I found aioari (https://github.com/M-o-a-T/aioari), an asyncio version of Python ARI, which looked newer, and supported modern Python. Apart from a bug with channel variables, aioari works for me. Hope that helps. Best regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Usage Survey
Hi Matt, I would have loved to participate to the survey, but I feel it does apply to my situation: as an integrator, I'm installing Asterisk for call centers, PBX, IVR... so I can not answer the first question of the survey ;) I also have dfferent versions installed. This is not a negative comment, I just want to express that the survey does not seem to apply to me; and many people on the Asterisk lists may be in a situation similar as mine. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 Le 08/03/2019 à 05:35, Matthew Fredrickson a écrit : > Hey All, > > For those of you that do not know me, my name is Matthew Fredrickson > and I’m the project lead for the Asterisk project. First off, I wanted > to thank all of you that contribute in various ways to the project – > whether it be at a developmental level, answering questions on forums > and mailing lists, contributing documentation, or just generally > advocating for it within your sphere of influence. It takes so many > people’s efforts to make the project what it is and to sustain such a > large and vibrant user and developer community. > > We created a general survey inquiring how people utilize Asterisk. It > should only take about 10-15 minutes, but would help us understand > better how our users are utilizing Asterisk and help us to understand > if there are important areas of Asterisk that we underemphasize from a > development perspective. If you don’t mind filling it out, it would be > greatly appreciated. > > Thanks *so* much again for your time, and best wishes to each of you > in your efforts. > > https://goo.gl/forms/xL1VUHRsf95saly13 > > Matthew Fredrickson > signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with the DB() function
Le 02/03/2019 à 18:10, Ira a écrit :
> exten => 1,1,set(DB(forwards/calls)=${home_in})
>same => n,set(DB(forwards/number)=1)
>same => n,verbose(${DB(forwards/calls)})
>same => n,return
>
> I can see the code running on the console and it prints out the first line
> with ${home_in} replaced with the expected 80 or so characters that variable
> contains.
>
> But the third line which should print those 80 characters back to the screen
> prints an empty string. What might I be doing wrong. It's worked from
> version 2 or 3 through 13 but it seems to be broken in 16.
This looks correct to me, and should be working with asterisk-16: could
you paste the console output with verbose=3?
> Also, when I installed asterisk it did not set itself up to start when the
> machine boots. Is there something else I need to do? It's the most recent
> Fedora.
The service manager in Fedora 29 is systemd: you probably need to
install the unit file, see asterisk-16.2.1/contrib/systemd/
Best regards,
--
Jean-Denis Girard
SysNux Systèmes Linux en Polynésie française
https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527
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Re: [asterisk-users] Can't find uuid!
Hi Ira, Le 28/02/2019 à 12:10, Ira a écrit : > Hello Asterisk, > > New install. > > Current Fedora server. > Current Asterisk V16 > > Installed everything necessary to run configure including uuid and uuid-devel. > > I know everything but uuid works as I commented out the uuid test and > configure finished. > > Make fails with uuid/uuid.h not found. > > uuid/uuid.h is not there, but uuid.h is there. > > Tried changing to in uuid.c but Asterisk will not > compile. > > ./configure fails with something about unable to find uuid_generate_random. > > Any suggestions? Seems like maybe it looking for an old version of uuid. Asterisk-16.2.1 builds without error here on Fedora 29, with stock libuuid-2.32.1-1.fc29.x86_64 and libuuid-devel-2.32.1-1.fc29.x86_64 installed. Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-app-dev] IAX2 protocol documentation
Hi Wojciech, The IAX2 RFC is available here: https://tools.ietf.org/html/rfc5456 I once developped an IAX2 softphone based on libiax: http://downloads.asterisk.org/pub/telephony/libiax/ Maybe it would be easyer to start from here. Best regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 Le 01/02/2019 à 09:09, Wojciech Puchar a écrit : > where can i find IAX2 protocol documentation - including encryption. > > I want to make IAX2 compatible mobile phone (over GPRS). > > ___ > asterisk-app-dev mailing list > asterisk-app-...@lists.digium.com > http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev signature.asc Description: OpenPGP digital signature ___ asterisk-app-dev mailing list asterisk-app-...@lists.digium.com http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tel URI
Hi Matt, Thanks for prompt reply. Le 30/01/2019 à 22:38, Matthew Fredrickson a écrit : > Right now, chan_pjsip does not properly handle tel: URIs. If you need > them you might need to use chan_sip. ok, I'm back on chan_sip, but I still do not see how I can send outgoing calls with tel: uri scheme. Is it supported? Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] tel URI
Hi list, Using Asterisk 16.1.1, with PJSIP, I'm asked to build a SIP trunk to a system that uses exclusively tel: uri on inbound and outbound calls. I could not find documentation or sample config about tel:uri. Is this doable? If not possible with PJSIP, is chan_sip a better option? Any pointer would be greatly appreciated. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iridium integration / gateway
Hi Tim, The PotsDOCK is what I was looking for. I'll propose that to my customer. My use case is an airway company, who needs Iridium for safety reasons, but does not want to be stuck with only one handset, so the idea is to connect the handset to the asterisk PBX, so they can call from any desk phone. Many thanks for detailed answer. Best regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 Le 03/04/2018 à 20:25, Tim S a écrit : > Hi, I use an Iridium 9555 handset and a "POTSdock". > >This takes a standard 9555 handset, gives it a fixed antenna > mount, and a telco service jack. Interfacing to the POTSdock, is a > matter of providing an analog POTS interface as if you were attaching > to a standard POTS phone provider. The dock generated standard > dial-tone and DTMF control. Iridium provides a test phone number you > can make calls from, and there are serial commands you can issue to > the handset to get service status, and account information, or Iridium > voicemail status. Be aware that rates are and order or two or three > of magnitude higher than most POTS providers. Minutes can be in the > multiple USD each range. > > I generally do my VoIP as a cost routing: > 1st, Cable modem (unlimited data, business internet) > 2nd, Cellular Data (data unlimited to 25GB/month, then rate reduced to > 5GB/month rate) > 3rd, Cellular Voice (unlimited minutes, lower voice quality due to CODEC > change) > 4th, Iridium Voice (expensive minutes, lowest voice quality due to > CODEC change, very high voice lag 1-3 seconds, more likely to work > while everything else is down locally) > > If you want to roll your own embedded VoIP system, the Iridium 9523 > modem engine is available as a module at retail, and takes the same > basic serial commands to establish voice or data calls. The > development manuals are available with a Google search. Be aware that > FCC or ITU certification is required if you embed an RF module in an > end product - which is why I went the POTSdock route. > > I also use the SBD modem modules (both 9603 and 9602) to do basic > telemetry and status reporting when conventional network access is > limited. The rates for sending a partial packet of data are very > reasonable, and you can message from modem directly to another modem > without traversing the commercial internet or phone infrastructure. > Very useful for rebooting a remote router that has crashed and is > blocking phone or internet access. ;-) > > If you've been following the news, SpaceX has been lofting the > IridiumNEXT constellation to replace the legacy constellation from the > 1990's (STILL running a good 20 years past design life) - and they are > expecting to modernize their subscriber equipment and rate plans. > They have three more launches to get the complete replacement > constellation in orbit, and they will be fixing the periodic dead > spots in their network as a result. > > I can try to answer questions on my use case. > > Best, > > -Tim > > On Tue, Apr 3, 2018 at 10:01 PM, Jean-Denis Girard <jd.gir...@sysnux.pf> > wrote: >> Hi list, >> >> I have a request to integrate Iridium in a Asterisk system. A quick >> search didn't return much: I expected to find products similar to GSM >> gateways, but this does not seem to exist. so I'd be very interested >> about possible solutions. Has it be done already, how? >> >> >> Thanks, >> -- >> Jean-Denis Girard >> >> SysNux Systèmes Linux en Polynésie française >> https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iridium integration / gateway
Hi Harry, Yes, it seems to be what I was looking for. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 Le 03/04/2018 à 22:25, Harry McGregor a écrit : > Hi, > > This looks like it may work for you. > > https://www.iridium.com/products/beam-potsdock-extreme-docking-station/ > > Harry > > On April 3, 2018 9:35:53 PM MST, Bertrand Lupart > <bertrand.lup...@linkeo.com> wrote: > > Hello, > > >> Thanks for reply, but this is irrelevant, I'm looking for an *Iridium* >> gateway. > > I guess calling to/from Iridium via GSM network is not an option and > you’re looking for on boarding asterisk on a boat or somewhere GSM > is not available. > > http://www.groundcontrol.com/Iridium_Making_Calls.htm > > > I guess you could connect to your Iridium handset via USB : > > https://www.iridium.com/support/help-desk/ > > > Greetings, > > — > Bertrand > > > -- > Sent from my Android device with K-9 Mail. Please excuse my brevity. > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iridium integration / gateway
Thanks for reply, but this is irrelevant, I'm looking for an *Iridium* gateway. Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 Le 03/04/2018 à 16:05, albert zhang a écrit : > http://www.dinstar.cn/en/index.php/GSM/ > > 2018-04-04 10:01 GMT+08:00 Jean-Denis Girard <jd.gir...@sysnux.pf > <mailto:jd.gir...@sysnux.pf>>: > > Hi list, > > I have a request to integrate Iridium in a Asterisk system. A quick > search didn't return much: I expected to find products similar to GSM > gateways, but this does not seem to exist. so I'd be very interested > about possible solutions. Has it be done already, how? > > > Thanks, > -- > Jean-Denis Girard > > SysNux Systèmes Linux en Polynésie française > https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ <https://community.asterisk.org/> > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > <https://wiki.asterisk.org/wiki/display/AST/Getting+Started> > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > <http://lists.digium.com/mailman/listinfo/asterisk-users> > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Iridium integration / gateway
Hi list, I have a request to integrate Iridium in a Asterisk system. A quick search didn't return much: I expected to find products similar to GSM gateways, but this does not seem to exist. so I'd be very interested about possible solutions. Has it be done already, how? Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi get latest
Le 18/10/2017 à 02:11, Jerry Geis a écrit : > I am trying to use dahdi complete 2.11.1 with a 4.13 kernel. - NOT > working for know reasons. > I tried applying two patches but still get compile errors. AHHH! > > How do I just use git to get the latest with the fixes > > This command did not work - I still get the errors. > git clone git://git.asterisk.org/dahdi/linux > <http://git.asterisk.org/dahdi/linux> dahdi-linux Hi Jerry, Maybe you missed this patch: https://issues.asterisk.org/jira/browse/DAHLIN-356 Or you can try my fork: git clone -b next https://github.com/sysnux/dahdi-linux.git Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP missing objects (Saint Michael)
Hi, Le 03/12/2016 à 08:35, Saint Michael a écrit : > create_new_socket: Unable to allocate RTP socket: Address family not > supported by protocol You are probably running a krnel without IPv6, this is a known problem on 13.13.0, see: https://issues.asterisk.org/jira/browse/ASTERISK-26617 There is a fix available, it will be in next release. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk now available with bundled pjproject!
Hi George, It seems configure with --disable-pa, and configuration "#define PJSIP_MAX_PKT_LEN 6000" did not make it to 13.8.0-rc1, do you still intend to add include these modifications? Thanks, -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 Le 13/03/2016 17:32, George Joseph a écrit : > > > On Sat, Mar 12, 2016 at 10:48 PM, Jean-Denis Girard <jd.gir...@sysnux.pf > <mailto:jd.gir...@sysnux.pf>> wrote: > > Hi George, > > Le 07/03/2016 12:53, George Joseph a écrit : > > Le 07/03/2016 09:28, George Joseph a écrit : > > > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is > released. > > I don't think this is related to the bundled version, but I got > PJSIP_ERXOVERFLOW when initiating a WebRTC video call from Chrome: > > [Mar 12 19:08:37] ERROR[9071]: pjproject:0 : sip_endpoint.c > Error processing packet from 192.168.10.88:50072 > <http://192.168.10.88:50072>: Rx buffer overflow > (PJSIP_ERXOVERFLOW) [code 171062]: > INVITE sip:*9...@sysnux.pf <mailto:9...@sysnux.pf> SIP/2.0 > Via: SIP/2.0/WSS ca4cqpd5cv2h.invalid;branch=z9hG4bK2286368 > Max-Forwards: 70 > To: <sip:*9...@sysnux.pf <mailto:9...@sysnux.pf>> > From: <sip:webs...@sysnux.pf > <mailto:sip%3awebs...@sysnux.pf>>;tag=q1ejnhm074 > Call-ID: l7rivm3clnebl6om63eb > CSeq: 1487 INVITE > Authorization: Digest algorithm=MD5, username="websip2", > realm="asterisk", nonce="1457845717/bfbd52f55e31f89cda00a1305c272bd6", > uri="sip:*9...@sysnux.pf <mailto:9...@sysnux.pf>", > response="d30a2f2b4d5d25e81dded44b7d98e336", > opaque="639fdd14224f0290", qop=auth, cnonce="r0d44vjitbof", nc=0001 > Contact: <sip:cldsr32v@ca4cqpd5cv2h.invalid;transport=ws;ob> > Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY > Content-Type: application/sdp > Supported: outbound > User-Agent: SIP.js/0.7.3 > Content-Length: 3335 > ... > > This can be solved by adding the following line to config_site.h: > #define PJSIP_MAX_PKT_LEN 6000 > > Would you consider adding it? > > > > Yes. I'll add it this week. > > > > > Thanks, > -- > Jean-Denis Girard > > SysNuxSystèmes Linux en Polynésie française > http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 > > signature.asc Description: OpenPGP digital signature -- _ -- 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] Asterisk now available with bundled pjproject!
Hi George, Le 07/03/2016 12:53, George Joseph a écrit : > Le 07/03/2016 09:28, George Joseph a écrit : > > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is released. I don't think this is related to the bundled version, but I got PJSIP_ERXOVERFLOW when initiating a WebRTC video call from Chrome: [Mar 12 19:08:37] ERROR[9071]: pjproject:0 : sip_endpoint.c Error processing packet from 192.168.10.88:50072: Rx buffer overflow (PJSIP_ERXOVERFLOW) [code 171062]: INVITE sip:*9...@sysnux.pf SIP/2.0 Via: SIP/2.0/WSS ca4cqpd5cv2h.invalid;branch=z9hG4bK2286368 Max-Forwards: 70 To: <sip:*9...@sysnux.pf> From: <sip:webs...@sysnux.pf>;tag=q1ejnhm074 Call-ID: l7rivm3clnebl6om63eb CSeq: 1487 INVITE Authorization: Digest algorithm=MD5, username="websip2", realm="asterisk", nonce="1457845717/bfbd52f55e31f89cda00a1305c272bd6", uri="sip:*9...@sysnux.pf", response="d30a2f2b4d5d25e81dded44b7d98e336", opaque="639fdd14224f0290", qop=auth, cnonce="r0d44vjitbof", nc=0001 Contact: <sip:cldsr32v@ca4cqpd5cv2h.invalid;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY Content-Type: application/sdp Supported: outbound User-Agent: SIP.js/0.7.3 Content-Length: 3335 ... This can be solved by adding the following line to config_site.h: #define PJSIP_MAX_PKT_LEN 6000 Would you consider adding it? Thanks, -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 signature.asc Description: OpenPGP digital signature -- _ -- 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] 2 devices same *actual* extension - can it be done
Hi,Le 09/03/2016 08:40, Kevin Long a écrit :
> At this time their system allows 2 devices (for example iPhone + desktop
> computer) using the same software license per user , which many of our users
> require.
With PJSIP, you can have multiple devices registered with same SIP
account, and then use function PJSIP_DIAL_CONTACTS to call all of them,
for example:
exten => 123,1,Dial(${PJSIP_DIAL_CONTACTS(sip123)})
Regards,
--
Jean-Denis Girard
SysNuxSystèmes Linux en Polynésie française
http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
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Re: [asterisk-users] Asterisk now available with bundled pjproject!
Hi, Le 07/03/2016 09:28, George Joseph a écrit : > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is released. I have tried GIT-master-ee5a944M on my Fedora 23 test server, and got: [pjproject] Unpacking /tmp/pjproject-2.4.5.tar.bz2 [pjproject] Applying patches and custom files [pjproject] Configuring with --prefix=/opt/pjproject --with-external-speex --with-external-gsm --with-external-srtp --with-external-pa --disable-video --disable-v4l2 --disable-sound --disable-resample --disable-opencore-amr --disable-ilbc-codec --without-libyuv --disable-g7221-codec --enable-epoll aconfigure: error: Unable to use PortAudio. If PortAudio development files are not available in the default locations, use CFLAGS and LDFLAGS env var to set the include/lib paths Makefile:57: recipe for target 'build.mak' failed make: *** [build.mak] Error 1 failed So I installed portaudio-devel (it was not needed before), and then compilation / installation were ok. When restarting Asterisk, SELinux blocked an access to /usr/bin/portaudio. Can't we simply disable portaudio? I have changed --with-external-pa to --disable-pa in third-party/pjproject/Makefile.rules, and it seems to compile / work fine. I have a question for servers without Internet access : is it enough to copy pjproject-2.4.5.tar.bz2 to /tmp or will there be other dependencies? I made a couple of test calls without problem (with or without portaudio). Thanks for your work, -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 signature.asc Description: OpenPGP digital signature -- _ -- 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] Grandstream Early Dial
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 19/02/2016 12:24, Bryant Zimmerman a écrit : > Jean > > If you moved the exten => _. Lines to the bottom of the context then > you should like be able to get away from having to have two separate > contexts. I use that method quiet often, but was in a hurry to get you a > response and did not think remember that nuance. No problem. I have tried different combinations, but could not make it work within a single context, but that's not a big deal anyway. Thanks, - -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlbLYLcACgkQuu7Rv+oOo/gD8ACgpEC4BsQRr0Rf+wPbGS1ShJWl xIIAoIlowl4EhoOEng0tWJTrDIEYBKR7 =+Sln -END PGP SIGNATURE- -- _ -- 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] Grandstream Early Dial
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Bryant,
Thanks for your reply.
It didn't work immediately, I had to create a second context, or else it
was looping between the second and first line. This seems to work:
[earlydial] ; Test Early Dial
exten => _.,1,Set(l_Extension=${EXTEN})
exten => _.,n,Goto(earlydial2,${l_Extension},1)
[earlydial2]
exten => _.,n,Goto(noMatch,1)
exten => noMatch,1, Incomplete(n)
exten => i,1,Goto(noMatch,1)
exten => t,1,Goto(noMatch,1)
exten => _1XX,1,Verbose(2, Dialed ${EXTEN})
same => n,Playback(extension)
same => n,SayDigits(${EXTEN})
same => n,Hangup()
Best regards,
- --
Jean-Denis Girard
SysNux Systèmes Linux en Polynésie française
http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
Le 19/02/2016 03:31, Bryant Zimmerman a écrit :
> Jean-Denis Girard
>
> I have not used the Incomplete yet, but you might be able to do
> something like this.
>
> [earlydial]
>
> exten => _.,1,Set(l_Extension = ${EXTEN})
> exten => _.,n,Goto(${l_Extension},1)
> exten => _.,n,Goto(noMatch,1)
>
> exten => i,1,Goto(noMatch,1)
>
> exten => noMatch,1, Incomplete(n)
>
> exten => _1XX,1,Verbose(2, Dialed ${EXTEN})
> same => n,Playback(extension)
> same => n,SayDigits(${EXTEN})
> same => n,Hangup()
>
>
> I wrote this in this message and have not tested this so use with
> caution. There may be syntactical issues, but the concept might work f
or
> you.
>
> Bryant
>
> --
- --
> *From*: "Jean-Denis Girard" <jd.gir...@sysnux.pf>
> *Sent*: Thursday, February 18, 2016 8:02 PM
> *To*: asterisk-users@lists.digium.com
> *Subject*: Re: [asterisk-users] Grandstream Early Dial
>
> Le 18/02/2016 11:03, Richard Mudgett a écrit :
>> I've been using Grandstream phones for more than 10 years, but onl
> y
>> yesterday tried to use Early Dial... and I failed. What is needed
> on the
>> Asterisk side to reply 484 to INVITE? Phones are talking to chan_p
> jsip
>> on Asterisk-13.7.1.
>
>
>> Look into the Incomplete application.
>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_In
c
> omplete
>
> Thanks for prompt answer Richard.
>
> Actually I had already tried the Incomplete application, but failed to
> add the "n" option, and this seems mandatory for SIP. I find the help
> text misleading : "NOTE: Most channel types need to be in Answer state
> in order to receive DTMF".
>
> This is my test dialplan:
>
> [earlydial] ; Test Early Dial
> exten => 1,1,Verbose(2,Incomplete 1 test)
> same => n,Incomplete(n)
>
> exten => _1X,1,Verbose(2,Incomplete 1X test)
> same => n,Incomplete(n)
>
> exten => _1XX,1,Verbose(2, Dialed ${EXTEN})
> same => n,Playback(extension)
> same => n,SayDigits(${EXTEN})
> same => n,Hangup()
>
> It works, but seems a bit complicated: is this the correct way to use
> Incomplete ?
>
>
> 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
>
>
>
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Re: [asterisk-users] Grandstream Early Dial
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Le 18/02/2016 11:03, Richard Mudgett a écrit :
> I've been using Grandstream phones for more than 10 years, but onl
y
> yesterday tried to use Early Dial... and I failed. What is needed
on the
> Asterisk side to reply 484 to INVITE? Phones are talking to chan_p
jsip
> on Asterisk-13.7.1.
>
>
> Look into the Incomplete application.
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Inc
omplete
Thanks for prompt answer Richard.
Actually I had already tried the Incomplete application, but failed to
add the "n" option, and this seems mandatory for SIP. I find the help
text misleading : "NOTE: Most channel types need to be in Answer state
in order to receive DTMF".
This is my test dialplan:
[earlydial] ; Test Early Dial
exten => 1,1,Verbose(2,Incomplete 1 test)
same => n,Incomplete(n)
exten => _1X,1,Verbose(2,Incomplete 1X test)
same => n,Incomplete(n)
exten => _1XX,1,Verbose(2, Dialed ${EXTEN})
same => n,Playback(extension)
same => n,SayDigits(${EXTEN})
same => n,Hangup()
It works, but seems a bit complicated: is this the correct way to use
Incomplete ?
Thanks,
- --
Jean-Denis Girard
SysNuxSystèmes Linux en Polynésie française
http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
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=g2Jy
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[asterisk-users] Grandstream Early Dial
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi list, I've been using Grandstream phones for more than 10 years, but only yesterday tried to use Early Dial... and I failed. What is needed on the Asterisk side to reply 484 to INVITE? Phones are talking to chan_pjsip on Asterisk-13.7.1. Thanks, - -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlbGLL8ACgkQuu7Rv+oOo/g55ACeON0aeNt9TFGw5lcUb1FhN7rH XqAAn2HRmx65LRP4hFUsCOvlAoV7/y8R =4iVe -END PGP SIGNATURE- -- _ -- 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] PJSIP T.38 issues
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Le 30/07/2015 13:20, Larry Moore a écrit :
Was it enabling alaw/ulaw which helped or did you need to use another
method to route the IAX channel through PJSIP or some other
configuration setting such as 'faxdetect' which may have been disabled
?
Well, first, I had SELinux enabled, which blocked Hylafax, and I didn't
notice :( I disabled it during testing.
Then, I had inconsistencies in my asterisks configurations: working
configuration is shown below. faxdetect is only needed when you want to
redirect the call to the fax extension. faxgateway is obviously needed
on both Asterisks.
With this configuration, I'm able to send faxes from Hylafax to the
PSTN. And receive fax from the PSTN on the same extension as my phone.
And T.38 is used on the network between the 2 asterisk, so faxing
reliability should be good.
Here is the relevant configuration on the gateway (Asterisk-11.18.0):
* chan_dahdi.conf:
context=incoming_isdn
switchtype = euroisdn
faxdetect = no
faxbuffers = 64,full
* sip.conf:
[general]
faxdetect = no
t38pt_udptl=yes,fec
[tiare] ; Real IPBX
type = friend
context = outgoing
host = tiare.sysnux.pf
disallow = all
allow = alaw
qualify = 153
* extensions.conf:
[incoming_isdn]
exten = s,1,Goto(1040,1)
exten = _104[01234],1,NoOp(Appel entrant sur ligne RNIS)
same = n,Set(FAXOPT(gateway)=yes)
same = n,Dial(SIP/tiare/${EXTEN})
[outgoing] ; Real IPBX !
include = local
include = gsm
include = international
[local]
exten = _NXX.,1,Set(FAXOPT(gateway)=yes)
same = n,Dial(${rnis}/${EXTEN})
Here is the configuration on the IPBX (Asterisk-13 git-309dd2a):
* pjsip.conf:
[t0gw]
type = endpoint
transport = udp
context = incoming
allow = alaw
aors = t0gw
language=fr
fax_detect = no
t38_udptl=yes
t38_udptl_ec=fec
* iax.conf
[iaxmodem0]
type=friend
secret=
context=fax-outgoing
host=dynamic
disallow=all
allow=slin
qualify=200
jitterbuffer=no
forcejitterbuffer=no
requirecalltoken=no
auth=md5
port=4570
* extensions.conf
[fax-outgoing]
include = local
include = international
[local]
exten = _4[09]XX,1,NoOp(${CALLERID()} calls ${EXTEN} (local))
same = n,Set(FAXOPT(gateway)=yes)
same = n,Dial(PJSIP/${EXTEN}@t0gw)
[stdexten] ; Extension normale
include = faxin
exten = _X.,1,NoOp(STDEXTEN ${EXTEN})
same = n,Set(sip=${DB(exten/${EXTEN})}) ; Convert extension to SIP
user
same = n,GotoIf($[${sip} != ]?sip_ok)
same = n,Return
same = n(sip_ok),Set(ext=${EXTEN}) ; Save extension
same = n,Set(FAXOPT(faxdetect)=yes)
same = n,Set(cfvm=${DB(CFVM/${sip})}) ; Check CFVM
same = n,GotoIf(${cfvm}?:nocfvm)
same = n,NoOp(Forward to voicemail ${ext})
same = n,Voicemail(${ext}@astportal,u)
same = n(nocfvm),Set(cfim=${DB(CFIM/${sip})})
same = n,GotoIf($[${LEN(${cfim})} 3]?${CHANNEL:4:8},${cfim},1)
same = n,GotoIf(${cfim}?:nocfim)
; If caller is CFIM, pass the call (call screening)
same = n,GotoIf($[${cfim} = ${DB(netxe/${CHANNEL:4:8})}]?nocfim)
same = n,Set(sip=${DB(exten/${cfim})})
same = n,NoOp(Forward immediate to ${sip})
same = n,Dial(PJSIP/${sip},,)
same = n(nocfim),Set(sip=${DB(exten/${ext})})
same = n,Dial(${PJSIP_DIAL_CONTACTS(${sip})},25)
[faxin]
exten = fax,1,NoOp(FAXIN (${FAXEXTEN}) ${CALLERID(all)})
same = n,Set(FAXOPT(gateway)=yes)
same = n,Dial(IAX2/iaxmodem0/${FAXEXTEN})
* /etc/iaxmodem/iaxmodem-cfg.ttyIAX
evice /dev/ttyIAX
owner uucp:uucp
mode660
port4570
refresh 300
server 127.0.0.1
peernameiaxmodem0
secret hyla
cidname SysNux
cidnumber +689 40.50.10.40
codec slin
Thanks,
- --
Jean-Denis Girard
SysNuxSystèmes Linux en Polynésie française
http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
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Re: [asterisk-users] PJSIP T.38 issues
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Thanks for your reply Larry. Le 27/07/2015 01:22, Larry Moore a écrit : I think the 488 Not acceptable here is occurring because the channel connecting through is not T.38 capable, that will be the IAX channel from iaxmomdem. This is what T38gateway is supposed to do. And I'm very happy to report that after one more day of efforts, I have everything working as I wante d. Thanks, - -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlW4UuAACgkQuu7Rv+oOo/hN8gCeLE74bX+LCMHF/GQf3GkpaDph 47sAnienQ5m2Fm2AI4BPIaqcYbMuiQZv =S4VT -END PGP SIGNATURE- -- _ -- 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] PJSIP T.38 issues
To: sip:40zzz...@gw.sysnux.pf;tag=as7bba6b0d Call-ID: 57283616-94e0-4052-bdff-491b31fdd229 CSeq: 31693 INVITE Server: Asterisk PBX 11.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:40ZZ@192.168.0.10:5060 Content-Type: application/sdp Require: timer Content-Length: 236 v=0 o=root 2087714374 2087714374 IN IP4 192.168.0.10 s=Asterisk PBX 11.18.0 c=IN IP4 192.168.0.10 t=0 0 m=audio 16834 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- Transmitting SIP request (412 bytes) to UDP:192.168.0.10:5060 --- ACK sip:40ZZ@192.168.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPj8504e505-1222-4747-955f-4788fef f58d1 From: SysNux sip:+68940XX@192.168.0.200;tag=22d8369f-061d-4232-9c4d-5068e81bc5c 5 To: sip:40zzz...@gw.sysnux.pf;tag=as7bba6b0d Call-ID: 57283616-94e0-4052-bdff-491b31fdd229 CSeq: 31693 ACK Max-Forwards: 70 User-Agent: Asterisk GPL PBX Content-Length: 0 -- PJSIP/t0gw-001a answered IAX2/iaxmodem0-7838 -- Channel PJSIP/t0gw-001a joined 'simple_bridge' basic-bridge 56a7726f-44a3-4df3-aee0-d21020aa5be1 -- Channel IAX2/iaxmodem0-7838 joined 'simple_bridge' basic-bridge 56a7726f-44a3-4df3-aee0-d21020aa5be1 --- Received SIP request (954 bytes) from UDP:192.168.0.10:5060 --- UPDATE sip:63035284-ad7d-484f-8e54-f5ea54f39104@192.168.0.200:5060 SIP/2 .0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK4fd84f17;rport Max-Forwards: 70 From: sip:40zzz...@gw.sysnux.pf;tag=as7bba6b0d To: SysNux sip:+68940XX@192.168.0.200;tag=22d8369f-061d-4232-9c4d-5068e81bc5c 5 Contact: sip:40ZZ@192.168.0.10:5060 Call-ID: 57283616-94e0-4052-bdff-491b31fdd229 CSeq: 102 UPDATE User-Agent: Asterisk PBX 11.18.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 287 v=0 o=root 2087714374 2087714375 IN IP4 192.168.0.10 s=Asterisk PBX 11.18.0 c=IN IP4 192.168.0.10 t=0 0 m=image 5720 udptl t38 c=IN IP4 192.168.0.10 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:849 a=T38FaxUdpEC:t38UDPFEC --- Transmitting SIP response (376 bytes) to UDP:192.168.0.10:5060 --- SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK4fd84f1 7 Call-ID: 57283616-94e0-4052-bdff-491b31fdd229 From: sip:40zzz...@gw.sysnux.pf;tag=as7bba6b0d To: SysNux sip:+68940XX@192.168.0.200;tag=22d8369f-061d-4232-9c4d-5068e81bc5c 5 CSeq: 102 UPDATE Server: Asterisk GPL PBX Content-Length: 0 Is anyone successfully using chan_pjsip and iaxmodem? Thanks, - -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlW1ojoACgkQuu7Rv+oOo/iU1gCglmxl6Pe3igseOwpbWfWtZdqg qzAAoJ8/zxzP3Eg79DxT7cjyXJj2oP9h =FR59 -END PGP SIGNATURE- -- _ -- 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] PJSIP, T.38 fax gateway
request (444 bytes) from UDP:192.168.0.10:5060 --- BYE sip:192.168.0.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK66a545b1;rport Max-Forwards: 70 From: sip:40483527@192.168.0.10;tag=as40626b30 To: sip:1041@192.168.0.200;tag=c0f238c0-f4de-4c5b-b8b4-3c8570b40620 Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060 CSeq: 103 BYE User-Agent: Asterisk PBX 11.16.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Transmitting SIP response (353 bytes) to UDP:192.168.0.10:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK66a545b 1 Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060 From: sip:40483527@192.168.0.10;tag=as40626b30 To: sip:1041@192.168.0.200;tag=c0f238c0-f4de-4c5b-b8b4-3c8570b40620 CSeq: 103 BYE Server: Asterisk GPL PBX Content-Length: 0 -- Channel PJSIP/t0gw-073b left 'simple_bridge' basic-bridge 0328da69-07f5-4270-8fa4-8178649c9906 -- Channel IAX2/iaxmodem0-7773 left 'simple_bridge' basic-bridge 0328da69-07f5-4270-8fa4-8178649c9906 == Spawn extension (stdexten, fax, 2) exited non-zero on 'PJSIP/t0gw-073b' -- Hungup 'IAX2/iaxmodem0-7773' Thanks, - -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlWeAjQACgkQuu7Rv+oOo/hS4ACfULyb3xNAOFvJpM6X/HQGhhfx gvcAoIYp4KZsEbU7zlBrbIV1vLfUmxs6 =lqkJ -END PGP SIGNATURE- -- _ -- 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] Seeking advice about ISDN BRI Cards
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 HI, I'm happy with OpenVox BRI ISDN cards, using Dahdi. Thanks, - -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 Le 26/05/2015 00:17, Lukasz Sokol a écrit : Hi, please whoever has some expertise in choice of BRI ISDN cards, please restore my faith in community support :) (on private email I can probably explain more than fits for a public f orum) Most I'd like to ask is about what to choose, out of what is available ... My locality is United Kingdom, lines from British Telecom (BT), but any advice / pointers (I googled around already) are welcome... My system to fit this card into, is FreePBX Distro with Asterisk 11, already running with incoming SIP trunk(s); I wish to extend it to accept incoming 'landline' ISDN BRI (6 channels / 3 ports). So far the interesting option(s) were Sangoma A500 and Digium B410P... (the appliance is adopted from an old desktop that still only ever has PCI2.0 slots, no PCIE) (there are also OpenVOX's cards, although their installation guide is somewhat, well... in the old kernel era...) Anyone who use(d) any of the above, not necessarily on a FreePBX - you 're welcome... :) Kind Regards, Lukasz -BEGIN PGP SIGNATURE- iEYEARECAAYFAlVl7mYACgkQuu7Rv+oOo/hgywCfQhrVWrcdjjhkN0fOaKVf3fJR o04AnRgubgRlu6UbsFiBZXJ/il0Cx42W =dyBq -END PGP SIGNATURE- -- _ -- 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] PJSIP CCSS
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 21/05/2015 00:16, Joshua Colp a écrit : If CCSS is needed then the only option is to use chan_sip. The chan_pjsip module does not implement CCSS in any way. Is CCSS support planned for PJSIP? chan_sip is in extended state in asterisk-13, so chan_pjsip should be preferred for new installations, ri ght? Thanks, - -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlVeAOkACgkQuu7Rv+oOo/jHggCfZ2pxnWrwPwWmKWApX5sDonUg wX8An0zPldZqvKUUdwmQ542oP9ZMaTK/ =BFyM -END PGP SIGNATURE- -- _ -- 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] PJSIP CCSS
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 21/05/2015 06:39, Ludovic Gasc a écrit : If you really want CCSS support and to be fancy with PJSIP, you can easily implement a similar feature with AMI events, I already did tha t a long time ago before the integration of CCSS in Asterisk. I think it's possible to implement that only with dialplan and call f iles. Yes, that's what I'm going to do. Thanks, - -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlVeEUUACgkQuu7Rv+oOo/jvIACdFoCaX3QCT5d8SLXb8zuyrW7G rwEAn3gy934zwRmlR5uGoIWXlI/irr9D =T6/q -END PGP SIGNATURE- -- _ -- 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] CHANNEL(aor) CHANNEL(contact) return nothing
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi list,
I'm trying to use CHANNEL(aor) and CHANNEL(contact) on PJSIP channel, on
asterisk-13.3.2, but they don't return anything. Is this a bug, or did I
miss something?
Here is my test dialplan:
exten = *98,1,Answer
same = n,NoOp(Channel=${CHANNEL(name)},type=
${CHANNEL(channeltype)})
same = n,NoOp(AOR=${CHANNEL(aor)}, contact=${CHANNEL(contact)})
same = n,Set(aor=${CHANNEL(name):$[LEN(CHANNEL(channeltype)) +1]:-9})
same = n,Set(contact=${PJSIP_AOR(${aor},contact)})
same = n,NoOp(URI=${PJSIP_CONTACT(${contact},uri)})
same = n,NoOp(Expiration time=
${PJSIP_CONTACT(${contact},expiration_time)})
same = n,NoOp(Qualify frequency=
${PJSIP_CONTACT(${contact},qualify_frequency)})
same = n,NoOp(Outbound proxy=
${PJSIP_CONTACT(${contact},outbound_proxy)})
same = n,NoOp(Path=${PJSIP_CONTACT(${contact},path)})
same = n,NoOp(User-Agent=${PJSIP_CONTACT(${contact},user_agent)})
same = n,PJSIPNotify(,jdg,gs-idle-screen-refresh)
same = n,Hangup
And here is the result in Asterisk CLI:
x220*CLI
-- Executing [*98@public:1] Answer(PJSIP/jdg-001f, ) in new
stack
-- Executing [*98@public:2] NoOp(PJSIP/jdg-001f,
Channel=PJSIP/jdg-001f, type=PJSIP) in new stack
-- Executing [*98@public:3] NoOp(PJSIP/jdg-001f, AOR=,
contact=) in new stack
[May 19 18:44:11] NOTICE[1476][C-001f]: ast_expr2.y:763
compose_func_args: argbuf allocated 12 bytes;
[May 19 18:44:11] NOTICE[1476][C-001f]: ast_expr2.y:782
compose_func_args: argbuf uses 11 bytes;
[May 19 18:44:11] NOTICE[1476][C-001f]: ast_expr2.y:763
compose_func_args: argbuf allocated 6 bytes;
[May 19 18:44:11] NOTICE[1476][C-001f]: ast_expr2.y:782
compose_func_args: argbuf uses 5 bytes;
-- Executing [*98@public:4] Set(PJSIP/jdg-001f, aor=jdg) in
new stack
-- Executing [*98@public:5] Set(PJSIP/jdg-001f,
contact=jdg;@sip:jdg@192.168.10.131:5062) in new stack
-- Executing [*98@public:6] NoOp(PJSIP/jdg-001f,
URI=sip:jdg@192.168.10.131:5062) in new stack
-- Executing [*98@public:7] NoOp(PJSIP/jdg-001f, Expiration
time=1432098235) in new stack
-- Executing [*98@public:8] NoOp(PJSIP/jdg-001f, Qualify
frequency=60) in new stack
-- Executing [*98@public:9] NoOp(PJSIP/jdg-001f, Outbound
proxy=) in new stack
-- Executing [*98@public:10] NoOp(PJSIP/jdg-001f, Path=)
in new stack
-- Executing [*98@public:11] NoOp(PJSIP/jdg-001f,
User-Agent=Grandstream GXP2130 1.0.4.23) in new stack
-- Executing [*98@public:12] PJSIPNotify(PJSIP/jdg-001f,
,jdg,gs-idle-screen-refresh) in new stack
-- Executing [*98@public:13] Hangup(PJSIP/jdg-001f, ) in new
stack
== Spawn extension (public, *98, 13) exited non-zero on
'PJSIP/jdg-001f'
What is wrong ?
Thanks,
- --
Jean-Denis Girard
SysNux Systèmes Linux en Polynésie française
http://www.sysnux.pf/Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
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=n/qH
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Re: [asterisk-users] CHANNEL(aor) CHANNEL(contact) return nothing
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 20/05/2015 00:50, Joshua Colp a écrit : It looks like this is an incoming leg, in which case that information isn't available. There is no association of an AOR and Contact on incoming legs (it MAY be possible to deduce but it certainly wouldn't work in all cases). Since you specify one explicitly on outgoing, that 's when it is available. When you say it may be possible, could you be more specific: is there another dialplan function / application to use ? Thanks, - -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlVctnwACgkQuu7Rv+oOo/jgWgCgsaDSC7i/Am9OGPFeM7GwsLL7 NrQAnRKgV5QVVOTWllRjF7zROpCb5/XF =FPOG -END PGP SIGNATURE- -- _ -- 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] PJSIP CCSS
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi list, It looks like Call Completion Supplementary Services is not available for PJSIP channels, am I right? Is there another solution? Thanks, - -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlVdWzsACgkQuu7Rv+oOo/gWpQCgiUry8ccsoZvcs40cra5f+iUx fs0An2xwUBilfpwAju18fQEy7tpLbDzT =MgvT -END PGP SIGNATURE- -- _ -- 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] PJSIP some AMI events is absent?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I made some tests with asterisk-13.2.0 and chan_pjsip this weekend myself, and came to the same conclusion: some peerstatus events are missing (eg. when contacts become unreachable / unavailable, IIRC), and I could not find a way to get contacts status through AMI. It looks a bit similar to issues 23172, 23173: PJSip missing functionalities. Thanks, - -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 Le 10/03/2015 21:27, Dmitriy Serov a écrit : Hello. Asterisk 13.2, PJSIP. Problem: I do not get any AMI events when changing the status of the contact. When using chan_sip I got peerstatus event. When using res_pjsip and devices (endpoint configuration) I got peerstatus event. When using res_pjsip.so and OUTBOUND REGISTRATION WITH OUTBOUND AUTHENTICATION i got registry event. When using ENDPOINT CONFIGURED AS A TRUNK, OUTBOUND AUTHENTICATION and status on contact changed I do not get any AMI event. I missed something? Tell me how to determine the change in the status of the contact (or endpoint/trunk) through AMI? Simple config: [srv_dev] type=auth auth_type=userpass username=login password=secret [srv_dev] type=aor contact=sip:sip.example.com:5060 qualify_frequency=5 default_expiration=10 max_contacts=1 remove_existing=yes [srv_dev] type=endpoint from_domain=example.com aors=srv_dev outbound_auth=srv_dev rewrite_contact=yes allow=!all,alaw Dmitriy Serov -BEGIN PGP SIGNATURE- iEYEARECAAYFAlUA2aoACgkQuu7Rv+oOo/g8VgCfejBsC/X/47QnKrsoNjl3pewn YRIAnA/IzoxGm/2sQgKEI4gFv6TpwN9Q =Db9Z -END PGP SIGNATURE- -- _ -- 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] [OT] IP Phone with Braille console for blind reseptionist
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi list, I'm currently working on a project where the customer is employing a blind person as receptionist: she has an Alcatel phone with extensions and a Braille console connected via serial port. I've searched for something similar for IP telephony, but found nothing. Has someone used such equipment? Else I'm thinking of using a PC with receptionist software and Braille console attached, but does that exist? Thanks, - -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlTToxAACgkQuu7Rv+oOo/gnkACdHESfvDUWYGh6Spo301BT1gIA shUAn2qiJcC1K0PWh715vc+Ns9uz0d8+ =XQVi -END PGP SIGNATURE- -- _ -- 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] DUNDI or ENUM or ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 20/01/2014 12:03, Jean-Denis Girard a écrit : Hi list, I'm looking for the best / recommended solution for automatic discovery of phone numbers for a multiple Asterisk system. This would be for an administration, with many branches (~30), but a common infrastructure (DNS, LDAP). Most branches would have Asterisk servers for various reasons (location, administrative). All contacts would be in LDAP, and Asterisk servers would have DNS entries. The problem is contacting other Asterisk without setting static routes in dialplan. I think DUNDI would be ideal, but is it still recommended for new installations or is it deprecated? dundi.com is dead, and redirects to the profile page on Digium website (https://my.digium.com/en/users/viewprofile/). ENUM could be another solution. What would you suggest? No recommendation ? Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlLep2QACgkQuu7Rv+oOo/h1YwCgnhs5Pioo0vr5wuWB4yZeDVuJ S+oAnj1GGr7JXtc3wDVyc4wOSN5GZZcw =0O+Z -END PGP SIGNATURE- -- _ -- 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] DUNDI or ENUM or ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi list, I'm looking for the best / recommended solution for automatic discovery of phone numbers for a multiple Asterisk system. This would be for an administration, with many branches (~30), but a common infrastructure (DNS, LDAP). Most branches would have Asterisk servers for various reasons (location, administrative). All contacts would be in LDAP, and Asterisk servers would have DNS entries. The problem is contacting other Asterisk without setting static routes in dialplan. I think DUNDI would be ideal, but is it still recommended for new installations or is it deprecated? dundi.com is dead, and redirects to the profile page on Digium website (https://my.digium.com/en/users/viewprofile/). ENUM could be another solution. What would you suggest? Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlLdnUgACgkQuu7Rv+oOo/jd6QCffhNne0yiNnfrgcS+cRQziz1/ dL0Anipk2Qqj2pCWbLIorW+Z8qff3q4L =DzaL -END PGP SIGNATURE- -- _ -- 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] CEL for attented transfer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Jairo, Le 20/11/2013 07:15, Jairo a écrit : peer: Local/321@entrada-canal-0001;1 How do you link Local/321@entrada-canal-0001;1 to the real original (physical) channel ? Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlKNXwoACgkQuu7Rv+oOo/iT+QCeIBPQZSr6urbiwUMiMZOzE5tJ Mu0Anjp/v33dPi+CZtTnEQR1JbxlUS13 =rb3h -END PGP SIGNATURE- -- _ -- 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] CEL for attented transfer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Paul, Le 20/11/2013 09:04, Paul Belanger a écrit : Well, it is a way lot harder to figure out because you used features.conf. Because of this, local channels are involved. Right, but this is sometimes necessary, and it's an asterisk feature. Specifically, you are going to have to track the channel IDs and look at the sequence of events. Then make an educated guess about what is happening. ok, but it could be so much easier (and reliable) if we had a CEL (or AMI event) that tracks the requested atxfer, like the CLI debug message: [Nov 20 13:09:28] DEBUG[12002][C-0174]: features.c:2701 builtin_atxfer: Executing Attended Transfer SIP/CROcqu0s-02b5, SIP/ngqckJos-02b6 (sense=2) Or adding the physical channel in the extra field of the CEL when CHAN_START event is fired for Local channels. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlKNeQoACgkQuu7Rv+oOo/isFQCeMTifSJNZ4IKS469dQFIkb+ti FNEAnR2pRTbNcCAIbOp5FS83/gGq1Fk9 =O7tw -END PGP SIGNATURE- -- _ -- 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] CEL for attented transfer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Jairo, Le 19/11/2013 01:36, Jairo a écrit : https://wiki.asterisk.org/wiki/display/AST/CEL+Events+and+Fields Thanks for your reply, but I have read this page of the wiki, I know what the fields mean. What I don't understand is how the events in my example can be used to determine 107 was attended transferred to 103 by 100. Or I do know that Local/103@100-0042;1 and Local/103@100-0042;2 were created by asterisk when SIP/100-0275 asked for atxfer? How does the event ATTENDEDTRANSFER/ SIP/107-0274/ Local/103@100-0042;1 show that 107 is transferred to 103? Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlKLi74ACgkQuu7Rv+oOo/hbXQCfbznLU4gcYBlz2OTATRQxrZlv 0DoAoLLYF+ykKuQtRPusTsVOrk/BcQQI =eO3S -END PGP SIGNATURE- -- _ -- 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] CEL for attented transfer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nobody, really? Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 Le 17/11/2013 19:03, Jean-Denis Girard a écrit : Hi list, I'm trying to use CEL to display channel information in real time. It works fine for simple calls, blind transfers, or SIP attended transfers (initiated from the SIP phone). My problem is for Asterisk attended transfers (atxfer as configured in features.conf). The scenario is: . phone 107 calls phone 100, . 100 dials the atxfer code, . 107 is on hold, and 100 hears the transfer message, . 100 dials phone 103, . 103 answers, . 100 hangups, . 107 and 103 are connected, . 107 hangups. CEL is configured with apps=all and events=ALL, and events are stored in a database via cel_pgsql. This is the list of events in the database for this call: eventtype |channame| peer - -++--- CHAN_START | SIP/107-0274 | CHAN_START | SIP/100-0275 | ANSWER | SIP/100-0275 | ANSWER | SIP/107-0274 | BRIDGE_START | SIP/107-0274 | SIP/100-0275 CHAN_START | Local/103@100-0042;1 | CHAN_START | Local/103@100-0042;2 | CHAN_START | SIP/103-0276 | ANSWER | SIP/103-0276 | ANSWER | Local/103@100-0042;2 | BRIDGE_START | Local/103@100-0042;2 | SIP/103-0276 ANSWER | Local/103@100-0042;1 | BRIDGE_START | SIP/100-0275 | Local/103@100-0042;1 BRIDGE_END | SIP/100-0275 | Local/103@100-0042;1 ATTENDEDTRANSFER | SIP/107-0274 | Local/103@100-0042;1 CHAN_START | Transfered/SIP/107-0274| BRIDGE_END | Transfered/SIP/107-0274ZOMBIE| SIP/100-0275 BRIDGE_START | SIP/107-0274 | Local/103@100-0042;1 HANGUP | SIP/100-0275 | CHAN_END | SIP/100-0275 | HANGUP | Transfered/SIP/107-0274ZOMBIE| CHAN_END | Transfered/SIP/107-0274ZOMBIE| BRIDGE_END | SIP/107-0274 | Local/103@100-0042;1 HANGUP | Local/103@100-0042;1 | CHAN_END | Local/103@100-0042;1 | HANGUP | SIP/107-0274 | CHAN_END | SIP/107-0274 | BRIDGE_END | Local/103@100-0042;2 | SIP/103-0276 HANGUP | SIP/103-0276 | CHAN_END | SIP/103-0276 | HANGUP | Local/103@100-0042;2 | CHAN_END | Local/103@100-0042;2 | LINKEDID_END | Local/103@100-0042;2 | (33 lignes) How should these events be interpreted? Asterisk version is 11.6.0. Thanks, -BEGIN PGP SIGNATURE- iEYEARECAAYFAlKK6p0ACgkQuu7Rv+oOo/heAACeN0eMR1qwRLcdV+Tsgn9fA+6c RKcAn246hmNUU2dxivPFEziueHYRTWcS =q196 -END PGP SIGNATURE- -- _ -- 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] CEL for attented transfer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi list, I'm trying to use CEL to display channel information in real time. It works fine for simple calls, blind transfers, or SIP attended transfers (initiated from the SIP phone). My problem is for Asterisk attended transfers (atxfer as configured in features.conf). The scenario is: . phone 107 calls phone 100, . 100 dials the atxfer code, . 107 is on hold, and 100 hears the transfer message, . 100 dials phone 103, . 103 answers, . 100 hangups, . 107 and 103 are connected, . 107 hangups. CEL is configured with apps=all and events=ALL, and events are stored in a database via cel_pgsql. This is the list of events in the database for this call: eventtype |channame| peer - -++--- CHAN_START | SIP/107-0274 | CHAN_START | SIP/100-0275 | ANSWER | SIP/100-0275 | ANSWER | SIP/107-0274 | BRIDGE_START | SIP/107-0274 | SIP/100-0275 CHAN_START | Local/103@100-0042;1 | CHAN_START | Local/103@100-0042;2 | CHAN_START | SIP/103-0276 | ANSWER | SIP/103-0276 | ANSWER | Local/103@100-0042;2 | BRIDGE_START | Local/103@100-0042;2 | SIP/103-0276 ANSWER | Local/103@100-0042;1 | BRIDGE_START | SIP/100-0275 | Local/103@100-0042;1 BRIDGE_END | SIP/100-0275 | Local/103@100-0042;1 ATTENDEDTRANSFER | SIP/107-0274 | Local/103@100-0042;1 CHAN_START | Transfered/SIP/107-0274| BRIDGE_END | Transfered/SIP/107-0274ZOMBIE| SIP/100-0275 BRIDGE_START | SIP/107-0274 | Local/103@100-0042;1 HANGUP | SIP/100-0275 | CHAN_END | SIP/100-0275 | HANGUP | Transfered/SIP/107-0274ZOMBIE| CHAN_END | Transfered/SIP/107-0274ZOMBIE| BRIDGE_END | SIP/107-0274 | Local/103@100-0042;1 HANGUP | Local/103@100-0042;1 | CHAN_END | Local/103@100-0042;1 | HANGUP | SIP/107-0274 | CHAN_END | SIP/107-0274 | BRIDGE_END | Local/103@100-0042;2 | SIP/103-0276 HANGUP | SIP/103-0276 | CHAN_END | SIP/103-0276 | HANGUP | Local/103@100-0042;2 | CHAN_END | Local/103@100-0042;2 | LINKEDID_END | Local/103@100-0042;2 | (33 lignes) How should these events be interpreted? Asterisk version is 11.6.0. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlKJn7EACgkQuu7Rv+oOo/gM+wCeMuUit/qQDBH3ymnxkngZAPUs FKUAn3wYVXKqtyd/xEZQM5u4tVqAwyZ9 =6SQH -END PGP SIGNATURE- -- _ -- 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] How to know the conflict in the dependencies?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 31/05/2013 15:10, bilal ghayyad a écrit : Hello; When I type make menuselect and finding the channels that has the sign XXX before it (this at the driver), how can I know the dependencies that are causing this conflict? Dependencies are detailed at the bottom of menuselect screen, for example XXX chan_motif Motif Jingle Channel Driver Depends on: iksemel(E), res_xmpp(M) Can use: openssl(E) Support Level: core Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlGpT0IACgkQuu7Rv+oOo/hkcwCgrMjVmIu1ftcMi9T1HSa6wTpZ upMAnixIgG4wYusuf02/zp9EByUgQcJY =Quzq -END PGP SIGNATURE- -- _ -- 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] [OT] Mediatrix Euro ISDN hangup problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I finally found the option in the Call Router - Route Config menu: when choosing Early disconnect, SIP BYE is sent as soon as ISDN Disconnect is received. That solved my problem. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 Le 12/02/2013 08:25, giovanni.v a écrit : On 12/02/2013 17.43, Jean-Denis Girard wrote: You're right, when someone answers it's not a problem. But if the caller is sent to voicemail and he hangups, we get 30 seconds disconnect tone. Yes, I imagined Your problem was born on such scenario. Yes, I already looked for such a configuration option, but unfortunately couldn't find any. I hoped someone on the list had experience with a Mediatrix on Euro ISDN. Unfortunately I don't have such knowledge, despite the fact some collegue suggested to try mediatrix devices I never do seriously because by reading their documentation I had the feeling of limited versatility. If You are not locked into this device I would suggest a Vega gateway from Sangoma. Their firmware has a configuration option that does just what you need: ---8-- disc_with_progress=0 O: Disconnect SIP call if disconnect , even if disconnect with progress 1 .. 6000: Enable passage of in-band (audio) information on call disconnect – pass media through for a maximum of this number of seconds. ---8-- ... or pass this info to the Mediatrix support so they can be inspired by competitors ;-) Regards, Giovanni -BEGIN PGP SIGNATURE- iEYEARECAAYFAlEpY4IACgkQuu7Rv+oOo/jFmgCffTT/bABwLFAqY3euwJ9vcKYG kb8An08ZvJBAaSwbIdL32UV0jOO7/Wxl =MQrf -END PGP SIGNATURE- -- _ -- 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] Faxdetect + T38gateway
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi list, I'm using faxdetect so that users can receive faxes on their phone numbers. It works fine. Fax is actually received by Hylafax through iaxmodem. I'm also using T.38 between asterisk and ATA (HT502/503) for better reliability. Sending from / to Hylafax works fine too; I checked with udptl set debug on that T.38 is actually used. The problem I see is when other end tries to use T.38: re-invite is to the SIP phone, which obviously rejects it (488 not applicable here), so T.38 is not used when hitting fax extension. Is there a solution to combine fax detect and t38getaway on the same call? This is with asterisk-11.2.1. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlEhWL0ACgkQuu7Rv+oOo/j6MgCdFZd9vOWvyFJDwbfDsWROEa9q pZcAn1l211ekkYcHKI7j8CLTy38UJ76b =IfLb -END PGP SIGNATURE- -- _ -- 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] Faxdetect + T38gateway
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 17/02/2013 13:17, Larry Moore a ￩crit : You have not provided any information relating to your configurations. Thanks for your reply. I've tried your settings, no luck. I'll try to better describe my problem. T.38 from HT503 to Asterisk + iaxmodem + Hylafax works fine when calling directly the fax extension (UDPTL traffic is displayed on the Asterisk console). faxdetect also works fine, call is redirected to the fax extension (same as above), and fax is indeed received by Asterisk + iaxmodem + Hylafax. But in the second case T.38 is not used (no UDPTL trafic on the Asterisk console), presumably because the T.38 re-invite from the HT503 was sent and declined by the phone, so HT503 continues sending fax as voice. This is the SIP dialog, when ht503 calls the phone (100) --- SIP read from UDP:192.168.10.170:5060 --- INVITE sip:100@192.168.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.170:5060;branch=z9hG4bK107821504;rport From: sip:ht...@asterisk.sysnux.pf;tag=874861461 To: sip:1...@asterisk.sysnux.pf;tag=as702b1ac8 Call-ID: 1925950287-506...@bjc.bgi.ba.bha CSeq: 42 INVITE Contact: sip:ht503@192.168.10.170:5060 Max-Forwards: 70 Supported: replaces, path, timer, eventlist User-Agent: Grandstream HT-502 V1.1B 1.0.9.1 chip V2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 274 v=0 o=ht503 8000 8001 IN IP4 192.168.10.170 s=SIP Call c=IN IP4 192.168.10.170 t=0 0 m=image 5004 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:280 a=T38FaxUdpEC:t38UDPRedundancy - - --- (14 headers 12 lines) --- Sending to 192.168.10.170:5060 (no NAT) == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 Got T.38 offer in SDP in dialog 1925950287-506...@bjc.bgi.ba.bha Capabilities: us - (gsm|ulaw|alaw|h263|h264|testlaw), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Peer doesn't provide video == Redirecting 'SIP/ht503-0038' to fax extension due to peer T.38 re-INVITE --- Transmitting (no NAT) to 192.168.10.170:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.170:5060;branch=z9hG4bK107821504;received=192.168.10.170;rport=5060 From: sip:ht...@asterisk.sysnux.pf;tag=874861461 To: sip:1...@asterisk.sysnux.pf;tag=as702b1ac8 Call-ID: 1925950287-506...@bjc.bgi.ba.bha CSeq: 42 INVITE Server: Asterisk PBX 11.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@192.168.0.10:5060 Content-Length: 0 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Spawn extension (stdexten, fax, 1) exited non-zero on 'SIP/ht503-0038' -- Executing [fax@stdexten:1] NoOp(SIP/ht503-0038, R￩ception FAX) in new stack -- Executing [fax@stdexten:2] Dial(SIP/ht503-0038, IAX2/iaxmodem0/100) in new stack -- Called IAX2/iaxmodem0/100 -- Call accepted by 127.0.0.1 (format alaw) -- Format for call is (alaw) -- IAX2/iaxmodem0-2681 is ringing -- IAX2/iaxmodem0-2681 answered SIP/ht503-0038 --- Reliably Transmitting (no NAT) to 192.168.10.170:5060 --- SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.10.170:5060;branch=z9hG4bK107821504;received=192.168.10.170;rport=5060 From: sip:ht...@asterisk.sysnux.pf;tag=874861461 To: sip:1...@asterisk.sysnux.pf;tag=as702b1ac8 Call-ID: 1925950287-506...@bjc.bgi.ba.bha CSeq: 42 INVITE Server: Asterisk PBX 11.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 Are you sure T.38 is actually used when call is redirected to fax extension? Thanks, - -- Jean-Denis Girard SysNux Syst￨mes Linux en Polyn￩sie franaise http://www.sysnux.pf/ T￩l: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlEhk6kACgkQuu7Rv+oOo/hqVwCeK8fb+yLOPgR3gwEIW2LscObb oIMAnjJyaItM5KOM6MUX0J6EAVgwoGdC =ELV+ -END PGP SIGNATURE- -- _ -- 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] [OT] Mediatrix Euro ISDN hangup problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 11/02/2013 12:40, giovanni.v a ￩crit : On 11/02/2013 17.01, Jean-Denis Girard wrote: I believe the first one will be not a viable option at all, no telco will change any important protocol compliance rule on a per subscriber basis. Well, in this case the subscriber is also the telco! Now forget your gateway for a moment and make a call on an imaginary phone connected to your PRI, after that call successfully answered let the called party hang up before you do. What you expect to hear? Sure, a disconnect tone... so you will put your phone on hook and the phone will send a disconnect immediately. Your pri-gateway-asterisk should work the same, even if the gateway does not send a disconnect immediately the user who started that call will hang up at least when hearing the disconnect tone (good feedback for humans, no?) and asterisk will send a bye to the sip gateway then that one shall initiate a disconnect on the user side. You're right, when someone answers it's not a problem. But if the caller is sent to voicemail and he hangups, we get 30 seconds disconnect tone. Or if the caller is sent to a queue and he hangs up, the agent takes a call which is already hung up. Check also if your gateway allow for customization to remap isdn/q.931 messages to sip. Yes, I already looked for such a configuration option, but unfortunately couldn't find any. I hoped someone on the list had experience with a Mediatrix on Euro ISDN. Sorry, hope you will be able to understand because English isn't my native language. No problem understanding, maybe because English is not my native language either ;) Thanks a lot, - -- Jean-Denis Girard SysNux Syst￨mes Linux en Polyn￩sie franaise http://www.sysnux.pf/ T￩l: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlEacSkACgkQuu7Rv+oOo/g+cgCeMPAPVplRp2o/QvxnWGdoux5q BTwAnArY230E3pPU8pJ4/OprCUmax8Gs =kWdl -END PGP SIGNATURE- -- _ -- 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] [OT] Mediatrix Euro ISDN hangup problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 11/02/2013 03:44, giovanni.v a ￩crit : I think the gateway is working in compliance to ETS 300 102-1 (5.3.4.1 Clearing when tones/announcements provided). 30 Sec. is the time assigned to T306 on the network side. Ok, thanks for your analysis. So the solution would be that the network does not send the progress indicator in the Disconnect message, or find a configuration parameter on the gateway so that it ignores the progress indicator, right? Thanks, - -- Jean-Denis Girard SysNux Syst￨mes Linux en Polyn￩sie franaise http://www.sysnux.pf/ T￩l: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlEZFccACgkQuu7Rv+oOo/g05ACfSCgt6+FVHSub4K1HMymJe6fx 7tkAnA/XWj9fC/xqNrv2+/THW+HfVt9T =2IzF -END PGP SIGNATURE- -- _ -- 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] [OT] Mediatrix Euro ISDN hangup problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi list, I'm getting a strange problem with a Mediatrix 3631 Gateway connected to the PSTN via an E1 PRI link configured for Euro ISDN signaling. The Mediatrix sends incoming calls from the PSTN to an Asterisk server via SIP: this works fine. But when the caller hangs up, the Mediatrix doesn't send Bye to Asterisk, so the call is not finished immediately from the Asterisk point of view: the delay is exactly 30 seconds. The telco made a trace on the ISDN, hangup is sent immediately, everything looks fine on their side. Then we took a trace on the gateway (see attached file). Line 345, ISDN Disconnect is received from the PSTN. According to the ISDN specs (and a trace I made on an Asterisk server connected via an ISDN card), I think the gateway should reply with a Release, then the network would reply with Release complete. But the Mediatrix never sends Release. After 30 sec, the *network* sends Release (line 445), then the Mediatrix immediately sends Release complete (line 448). Then, the Mediatrix sends SIP / BYE (line 475), and Asterisk immediately hangs up. It seems to me that there is something wrong with the Mediatrix or am I wrong? It could obviously be a mistake in my configuration, but I could not find what is wrong. Has anyone successfully used a 3631 with Euro ISDN? Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlEYcRoACgkQuu7Rv+oOo/ixzACfQYghIjClC/sDIcBLB7Xp/VBI eFYAn17Kya8EIvAMF+xJvjmBTDfoMrWv =xc1L -END PGP SIGNATURE- trace_Mediatrix_anon.txt.bz2 Description: BZip2 compressed data -- _ -- 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] Google Voice Routing
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 26/11/2012 04:26, Joshua Colp a écrit : To others using chan_motif - are you experiencing the same issue? I didn't use chan_motif since testing a few weeks ago, so I may I have broke my configuration, but Google Voice seems to be broken now. Call is received, but Asterisk does nothing: --- XMPP received from 'google-cathy' --- iq type=set to=cathy.fou...@gmail.com/asterisk-x217D1B44 id=078099D69B89C046 from=jeandenis.gir...@gmail.com/gmail.3027C461jin:jingle action=session-initiate sid=c1654741541 initiator=jeandenis.gir...@gmail.com/gmail.3027C461 xmlns:jin=urn:xmpp:jingle:1jin:content name=audio creator=initiatorrtp:description media=audio ssrc=731587560 xmlns:rtp=urn:xmpp:jingle:apps:rtp:1rtp:payload-type id=103 name=ISAC clockrate=16000/rtp:payload-type id=104 name=ISAC clockrate=32000/rtp:payload-type id=107 name=speex clockrate=16000rtp:parameter name=bitrate value=22000//rtp:payload-typertp:payload-type id=9 name=G722 clockrate=16000rtp:parameter name=bitrate value=64000//rtp:payload-typertp:payload-type id=102 name=ILBC clockrate=8000rtp:parameter name=bitrate value=13300//rtp:payload-typertp:payload-type id=108 name=speex clockrate=8000rtp:parameter name=bitrate value=11000//rtp: - --- XMPP received from 'google-cathy' --- payload-typertp:payload-type id=0 name=PCMU clockrate=8000rtp:parameter name=bitrate value=64000//rtp:payload-typertp:payload-type id=8 name=PCMA clockrate=8000rtp:parameter name=bitrate value=64000//rtp:payload-typertp:payload-type id=127 name=red clockrate=8000/rtp:payload-type id=126 name=telephone-event clockrate=8000/rtp:rtcp-mux/rtp:encryptionrtp:crypto crypto-suite=AES_CM_128_HMAC_SHA1_80 key-params=inline:t/ni1bJ62BAh0CYQgH0LebZabWx47cG7iou0/OsJ tag=1/rtp:crypto crypto-suite=AES_CM_128_HMAC_SHA1_32 key-params=inline:ysx82SVYw1H61YGmaV2d0b32zxvRBtf6PvBMlhwR tag=2//rtp:encryption/rtp:descriptionp:transport xmlns:p=http://www.google.com/transport/p2p//jin:content/jin:jingleses:session type=initiate id=c1654741541 initiator= - --- XMPP received from 'google-cathy' --- jeandenis.gir...@gmail.com/gmail.3027C461 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=103 name=ISAC clockrate=16000/pho:payload-type id=104 name=ISAC clockrate=32000/pho:payload-type id=107 name=speex bitrate=22000 clockrate=16000/pho:payload-type id=9 name=G722 bitrate=64000 clockrate=16000/pho:payload-type id=102 name=ILBC bitrate=13300 clockrate=8000/pho:payload-type id=108 name=speex bitrate=11000 clockrate=8000/pho:payload-type id=0 name=PCMU bitrate=64000 clockrate=8000/pho:payload-type id=8 name=PCMA bitrate=64000 clockrate=8000/pho:payload-type id=127 name=red clockrate=8000/pho:payload-type id=126 name=telephone-event clockrate=8000/pho:rtcp-mux/pho:src-id731587560/pho:src-idrtp:encryption xmlns:rtp= - --- XMPP received from 'google-cathy' --- urn:xmpp:jingle:apps:rtp:1rtp:crypto crypto-suite=AES_CM_128_HMAC_SHA1_80 key-params=inline:t/ni1bJ62BAh0CYQgH0LebZabWx47cG7iou0/OsJ tag=1/rtp:crypto crypto-suite=AES_CM_128_HMAC_SHA1_32 key-params=inline:ysx82SVYw1H61YGmaV2d0b32zxvRBtf6PvBMlhwR tag=2/pho:usage//rtp:encryption/pho:description/ses:session/iq - --- XMPP received from 'google-cathy' --- iq type=set to=cathy.fou...@gmail.com/asterisk-x217D1B44 id=7B548BACBF5495D3 from=jeandenis.gir...@gmail.com/gmail.3027C461jin:jingle action=session-terminate sid=c1654741541 xmlns:jin=urn:xmpp:jingle:1ses:reason xmlns:ses=http://www.google.com/session;ses:connectivity-error//ses:reasonpho:call-ended xmlns:pho=http://www.google.com/session/phone//jin:jingleses:session type=terminate id=c1654741541 initiator=jeandenis.gir...@gmail.com/gmail.3027C461 xmlns:ses=http://www.google.com/session;ses:reasonses:connectivity-error//ses:reasonpho:call-ended xmlns:pho=http://www.google.com/session/phone//ses:session/iq - Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEUEARECAAYFAlCzs60ACgkQuu7Rv+oOo/iAvQCYlWFMToLIl3CFtYLhCCpQBbZx WACeJ6xBAn1c/JU+U7kqqlvAZvPr+lk= =DOBH -END PGP SIGNATURE- -- _ -- 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] Google Voice and back (chan_motif)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 06/11/2012 02:16, Joshua Colp a écrit : You've found a bug! I've fixed it now, though. It'll go out in the next Asterisk 11 release or you can check out Asterisk 11 from subversion to get it. I have applied the patch, it now works as I expected: I can make calls from sip phone1 connected to Asterisk, through my Google Voice account to another Google Voice account, and receive on sip phone2, connected to the same Asterisk. Awesome! Sorry for the inconvenience! No problem Joshua, thanks for very prompt fix! Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlCZOpMACgkQuu7Rv+oOo/jJdwCaAyw+unmXEpH8vHYBQiiBDe4z 9ygAnjNQKFmuUvMdLnv7/sblJNr0k5oW =V11X -END PGP SIGNATURE- -- _ -- 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] Google Voice and back (chan_motif)
type=host/ - --- XMPP received from 'google-cathy' --- candidate component=2 foundation=583378294 generation=0 id=cedf ip=192.168.0.10 port=16119 priority=2130706430 protocol=udp type=host/candidate component=2 foundation=192809686 generation=0 id=76d2 ip=123.50.122.114 port=16119 priority=2130706430 protocol=udp type=host//transport/content/jingle/iq - [Nov 5 18:30:15] ERROR[28652][C-0005]: chan_motif.c:1971 jingle_interpret_ice_udp_transport: Incomplete ICE-UDP candidate received on session '7e44df781ce623b6' --- XMPP sent to 'google-cathy' --- iq type='result' from='cathy.fou...@gmail.com/asterisk-xD2C13566' to='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' id='p'/ - == Spawn extension (incoming-motif, s, 2) exited non-zero on 'Motif/jeandenis.girard-646f' --- XMPP sent to 'google-cathy' --- iq to='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' type='set' id='k'jingle action='session-terminate' sid='7e44df781ce623b6' xmlns='urn:xmpp:jingle:1'reasonfailed-transport//reason/jingle/iq - --- XMPP received from 'google-cathy' --- iq type=result from=jeandenis.gir...@gmail.com/asterisk-xBEE0DF06 to=cathy.fou...@gmail.com/asterisk-xD2C13566 id=j/ - --- XMPP received from 'google-cathy' --- iq to=cathy.fou...@gmail.com/asterisk-xD2C13566 type=set id=q from=jeandenis.gir...@gmail.com/asterisk-xBEE0DF06jingle action=session-terminate sid=7e44df781ce623b6 xmlns=urn:xmpp:jingle:1reasonfailed-transport//reason/jingle/iq - --- XMPP sent to 'google-cathy' --- iq type='error' from='cathy.fou...@gmail.com/asterisk-xD2C13566' to='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' id='q'error type='cancel'item-not-found xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'/unknown-session xmlns='urn:xmpp:jingle:errors:1'//error/iq - --- XMPP received from 'google-jd' --- iq type=result from=cathy.fou...@gmail.com/asterisk-xD2C13566 to=jeandenis.gir...@gmail.com/asterisk-xBEE0DF06 id=p/ - --- XMPP received from 'google-jd' --- iq to=jeandenis.gir...@gmail.com/asterisk-xBEE0DF06 type=set id=k from=cathy.fou...@gmail.com/asterisk-xD2C13566jingle action=session-terminate sid=7e44df781ce623b6 xmlns=urn:xmpp:jingle:1reasonfailed-transport//reason/jingle/iq - --- XMPP sent to 'google-jd' --- iq type='error' from='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' to='cathy.fou...@gmail.com/asterisk-xD2C13566' id='k'error type='cancel'item-not-found xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'/unknown-session xmlns='urn:xmpp:jingle:errors:1'//error/iq - --- XMPP received from 'google-jd' --- iq type=error from=cathy.fou...@gmail.com/asterisk-xD2C13566 to=jeandenis.gir...@gmail.com/asterisk-xBEE0DF06 id=qerror type=cancelitem-not-found xmlns=urn:ietf:params:xml:ns:xmpp-stanzas/unknown-session xmlns=urn:xmpp:jingle:errors:1//error/iq - --- XMPP received from 'google-cathy' --- iq type=error from=jeandenis.gir...@gmail.com/asterisk-xBEE0DF06 to=cathy.fou...@gmail.com/asterisk-xD2C13566 id=kerror type=cancelitem-not-found xmlns=urn:ietf:params:xml:ns:xmpp-stanzas/unknown-session xmlns=urn:xmpp:jingle:errors:1//error/iq - Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlCYk4sACgkQuu7Rv+oOo/igqgCdG4lbXgV9/3in9jOqDK6UQwpM rSgAoKYYKU4ste9lV8zLLBLaJOanEZ4X =UJR7 -END PGP SIGNATURE- -- _ -- 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] Google Voice and back (chan_motif)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 05/11/2012 18:55, Co-op Vacation Rentals a écrit : Try adding transport=google-v1 to motif.conf [google-jd] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw *transport=google-v1* connection=google-jd ; - xmpp.conf [google-cathy] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw *transport=google-v1* connection=google-cathy ; - xmpp.conf Thanks for your reply, unfortunately that makes no difference, I still get: [Nov 5 19:45:16] ERROR[30664][C-0005]: chan_motif.c:1971 jingle_interpret_ice_udp_transport: Incomplete ICE-UDP candidate received on session '14ec70fb484b5700' Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlCYpNoACgkQuu7Rv+oOo/imrgCgrDUi0VdhCbspzA7SUtFQWpDK iEAAn3X5x/eX96eSRj8PsXqpk4SYFpA5 =98GL -END PGP SIGNATURE- -- _ -- 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] GSM gateway or PCI Card recommendation?
Hi, Le 12/05/2012 06:09, Shahid H a écrit : I am looking for a GSM Gateway or GSM PCI Card with minimum of 6 Sim Cards slots. Which one do you recommend and easier to setup? As long it work on UK mobile network and make 6 calls simultaneously. I never used GSM cards, but had good success with 2N Voiceblue Lite gateways: http://www.2n.cz/en/products/gsm-gateways/voip/voiceblue-lite/ Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -- _ -- 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] T.38 troubles
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Steve, Le 26/03/2012 14:50, Steve Underwood a écrit : Spandsp has some workarounds for bugs in Mediatrix boxes. They usually work OK. I just made a couple of tests with spandsp snapshot 20120328: I now get The call dropped prematurely. Network capture is attached, hope that helps. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAk9zrHoACgkQuu7Rv+oOo/gFeACeN+xlKqUOsduA8132kiGP7Lbc gpIAn2n/b6DoN/80G0kQv/Q8MkraSJRW =NVE/ -END PGP SIGNATURE- fax-ast10-2.pcap.bz2 Description: BZip2 compressed data -- _ -- 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] T.38 troubles
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Hash: SHA1
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Hi list,
I'm having difficulties when receiving faxes from the PSTN with this
relatively simple installation:
PSTN --PRI-- GW --T.38-- Asterisk
The gateway is a Mediatrix 3301 (firmware Dgw 2.0.14.251). It's
configured to transmit faxes as T.38. I may have missed something in its
configuration, but it does switch to T.38 when a fax is detected. On the
Asterisk side, I'm using 10.2.1 with spandsp-0.0.6-pre20 and ReceiveFax
from res_fax_spandsp. ${FAXSTATUS} returns FAILED and ${FAXERROR}
Disconnected after permitted retries.
I did a network capture, attached to this mail: from my understanding,
T.38 is accepted by Asterisk, then there seems to be some UDPTL traffic,
which I don't understand...
Why does it fail, and what is wrong? I'd appreciate if someone could
send me advice / suggestions.
Thanks,
- - --
Jean-Denis Girard
SysNux Systèmes Linux en Polynésie française
http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27
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fax-ast10.pcap.bz2
Description: BZip2 compressed data
--
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Re: [asterisk-users] T.38 troubles
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Steve, Le 26/03/2012 14:50, Steve Underwood a écrit : Your log shows the Mediatrix GW has problems. It sends a DCS signal to the Asterisk box, but doesn't following it with TCF as it should. The asterisk box times out waiting for TCF and tries to take recovery action which fails. Thanks for your analysis. Could it be a configuration problem on the Mediatrix? Spandsp has some workarounds for bugs in Mediatrix boxes. They usually work OK. What would you suggest then, is there anything to do to enable the workarounds? spandsp-0.0.6pre20 is the latest available, or should I try spandsp-20120324? As far as I know, there is no firmware update for the Mediatrix... Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAk9xFUUACgkQuu7Rv+oOo/jp+QCeMH14GwjoNTWjF7JpTVntv8nr wbYAn1dik/g+uurccqat/KcQwS8cxxZw =MQNr -END PGP SIGNATURE- -- _ -- 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] Major changes between 1.4/1.6.1.8/10?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 06/02/2012 04:04, Gilles a écrit : Hello Is there a document that sums up the major changes made to the four main releases available (1.4, 1.6, 1.8, and 10), to check if it's worth upgrading? This link also presents changes between Asterisk versions: http://linuxinnovations.com/applications1.4-1.6.2.html Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAk8xTP0ACgkQuu7Rv+oOo/gmlgCeKdH/TPWOAM5cIG+Ee0L9cG1e exEAn3ThS2K+JUvxUNhcuNd3GAsdzZKq =6T+v -END PGP SIGNATURE- -- _ -- 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] Grandstream GXP2000 - copy configuration from handset
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Le 09/10/2011 03:40, Silverthorne Wystead a écrit : I have a Grandstream GXP2000 and I would like to use tftp or some other utility to grab the configuration from it. Anyone have any bright ideas? gsutil works for me: http://www.pkts.ca/gsutil.shtml Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAk6R6woACgkQuu7Rv+oOo/iLoQCfa22qoGXgca5yjkykbamzAzDL 8K4An1LOB8owlQdyhLAqZp5YIArsL/BM =yC6f -END PGP SIGNATURE- -- _ -- 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] Integration of OpenVXI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Le 20/06/2011 04:40, Gopal krishnan a écrit : Have anybody integrated OpenVXI http://www.speech.cs.cmu.edu/openvxi/ with Asterisk? Voiceglue works for me: http://www.voiceglue.org/ Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAk3/dbgACgkQuu7Rv+oOo/hemACdEN4qLhxLl9LJGpdGIfd8zZ0B PAsAnRxitrzwt5RhWPeo/iwVuYqfeKNh =LpwD -END PGP SIGNATURE- -- _ -- 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] Cross compiling Asterisk, Dahdi..
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Gordon Gordon Henderson a écrit : Is there a proper, documented way to cross compile DAHDI and Asterisk for a processor/system other than the one you're currently typing on? Here is what I'm doing for building dahdi modules on my x86_64 system, for geode target. In dahdi linux directory: make KVERS=2.6.33-rc3-git3-sysnux KSRC=/home/jdg/RPM/BUILD/linux Then install in /tmp/dahdi: make DESTDIR=/tmp/dahdi ARCH=i386 KVERS=2.6.33-rc3-git3-sysnux KSRC=/home/jdg/RPM/BUILD/linux install-modules Then I make a tar of /tmp/dahdi, and extract that archive on the geode target. I don't know if it's the proper way to do it, but it works fine for me. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAktR8KoACgkQuu7Rv+oOo/jTVgCcDGNON+UlFwz59ykMFV1aQWEy gpEAn1MNPrqYfUoTowdjUbMexGqTvkz9 =RD6g -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cross compiling Asterisk, Dahdi..
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen a écrit : On Sat, Jan 16, 2010 at 07:00:26AM -1000, Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Gordon Gordon Henderson a écrit : Is there a proper, documented way to cross compile DAHDI and Asterisk for a processor/system other than the one you're currently typing on? Here is what I'm doing for building dahdi modules on my x86_64 system, for geode target. In dahdi linux directory: make KVERS=2.6.33-rc3-git3-sysnux KSRC=/home/jdg/RPM/BUILD/linux Then install in /tmp/dahdi: make DESTDIR=/tmp/dahdi ARCH=i386 KVERS=2.6.33-rc3-git3-sysnux KSRC=/home/jdg/RPM/BUILD/linux install-modules Is an explicit ARCH needed? It shouldn't have been there in the first place. The ARCH is caculated by Kbuild from your config (in the kernel tree) and there should be no need to provide it (at least as of dahdi 2.2). Likewise: is KVERS really needed in that line? ARCH seems to be needed: [...@tiare dahdi-linux.svn]$ make DESTDIR=/tmp/dahdi KSRC=/home/jdg/RPM/BUILD/linux CC [M] /home/jdg/RPM/BUILD/dahdi-linux.svn/drivers/dahdi/dahdi_echocan_mg2.o LD [M] /home/jdg/RPM/BUILD/dahdi-linux.svn/drivers/dahdi/dahdi_vpmadt032_loader.o ld: Relocatable linking with relocations from format elf64-x86-64 (/home/jdg/RPM/BUILD/dahdi-linux.svn/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o) to format elf32-i386 (/home/jdg/RPM/BUILD/dahdi-linux.svn/drivers/dahdi/dahdi_vpmadt032_loader.o) is not supported make[2]: *** [/home/jdg/RPM/BUILD/dahdi-linux.svn/drivers/dahdi/dahdi_vpmadt032_loader.o] Erreur 1 make[1]: *** [_module_/home/jdg/RPM/BUILD/dahdi-linux.svn/drivers/dahdi] Erreur 2 make[1]: quittant le répertoire « /home/jdg/RPM/BUILD/linux-2.6 » make: *** [modules] Erreur 2 KVERS is not needed. This is with today svn tree. [...@tiare dahdi-linux.svn]$ svnversion 7918 Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAktSVksACgkQuu7Rv+oOo/gxEQCeJSJDm9LwbYqaNN/3rSmvES11 QqUAnRuMY7NNRAIJcwlZMTvxJuuJ7KFx =E+1B -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - mISDN and B410P questions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Olivier a écrit : | | | 2009/10/24 Jean-Denis Girard jd.gir...@sysnux.pf | mailto:jd.gir...@sysnux.pf | | Olivier a écrit : | | Hello, | | | | I'm evaluating to possibility to use chan_misdn as a short term | | workaround, in case latest Dahdi is not stable enough for what we are | | planning to do (we wish to use Junghanns and Digium BRI hardware with | | Asterisk 1.6) . | | Dahdi has been working fine for me for a few months, using a Junghanns | DuoBRI and asterisk-1.6. I have used bristuff before, it usually worked | fine, but separate patch. I also tried chan_misdn but quickly abandoned. | | | Do you mean you abondoned chan_misdn (1.1.9-1) with 1.6.x on Junghanns | hardware or are you thinking about another combination ? | | Did you change because revision-2822-enabled Dahdi ran successfully | enough that you didn't need to further try with misdn or because you met | bloking issues ? Sorry if that my post was not clear. I'm using Dahdi because I'm happy with it. It works and is simple to set up: it does not need any external patch / software. Before BRI support in Dahdi, I used BRIstuff, which worked fine, but didn't follow Asterisk releases. And it was more work to set up. I tried mISDN, but then I depended on specific kernel versions. I also had echo problems. So I quickly abandoned mISDN. The conclusion is that I'm very happy with BRI support in Dahdi. I've been waiting for it, probably for more than 5 years ! Regards, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkrlyAsACgkQuu7Rv+oOo/hvTgCfaPXjHD90VpeNlSMH/GPY9bzk jvMAn2hzE3JWJ/iY+LXf7Zj6hfHPcT2R =caOJ -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - mISDN and B410P questions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Olivier a écrit : | Hello, | | I'm evaluating to possibility to use chan_misdn as a short term | workaround, in case latest Dahdi is not stable enough for what we are | planning to do (we wish to use Junghanns and Digium BRI hardware with | Asterisk 1.6) . Dahdi has been working fine for me for a few months, using a Junghanns DuoBRI and asterisk-1.6. I have used bristuff before, it usually worked fine, but separate patch. I also tried chan_misdn but quickly abandoned. Regards, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkrjbg0ACgkQuu7Rv+oOo/hvxACggSiVlumqoSBtbcvUHKDHj1SG B8kAnicGGenJgcQXYq5RrkX7DCzYhpbR =jXB8 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - mISDN and B410P questions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen a écrit : | In dahdi-linux trunk those cards should be supported by the wcbrxxp | driver. Though I'd welcome more testing regarding those cards, as I | specifically suspect that LED handling there is incorrect. I think you mean wcb4xxp: dahdi_echocan_mg2 6336 2 wcb4xxp40036 3 dahdi 206704 10 dahdi_echocan_mg2,wcb4xxp I confirm that LEDs are not working. | Note that you have to use 'hardhdlc' rather than 'dchan' in | /etc/dahdi/system.conf . Right. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkrj1sQACgkQuu7Rv+oOo/gT2ACgkwgcg3YA9QQpOMriPL6iAKPe OqEAn1rGwwdJr+iAnNzKiJ3/wUUcghHc =P3yz -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Nuance Vocalizer TTS Engine
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Vela Sivasankaran a écrit : | Hi, | How can I integrate Asterisk to Nuance TTS engine instead of | Cepstral? Has anybody done this? How is the architecture and can Java | AGI be used to communicate between them? I have made app_realspeak which integrates RealSpeak in Asterisk, so it's usable from the dialplan or from AGI. It has been discussed a few times on the list, eg in this thread: http://lists.digium.com/pipermail/asterisk-users/2008-December/222734.html Contact me if interested. Regards, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkrgfhgACgkQuu7Rv+oOo/jr+ACfWJ7PtqZftJcEPQrRdyj9Jocu UhAAnj7uICjaiAQsLeGtW/gDno2pk4zZ =/j8E -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX for internet file transfer?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Tzafrir Cohen a écrit : | On Fri, Jun 26, 2009 at 07:50:08PM -0500, Moises Silva wrote: | On Fri, Jun 26, 2009 at 6:48 PM, Maris maris@vdi.de wrote: | | I'm dealing with an idea to exchange data in a socket connection style | or a sort of ftp transfer with IAX2 as the transport medium. | | So while you can basically encode your data into an audio file, send it | to the other side and decode it there Asterisk (or just about any VoIP | software) will opt for timely delivery rather than a reliable delivery. | IAX does support sending / receiving images, and so does asterisk, so I guess it could support any file. tiare*CLI ~ -= Info about application 'SendImage' =- [Synopsis] Send an image file [Description] ~ SendImage(filename): Sends an image on a channel. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkpFiL4ACgkQuu7Rv+oOo/gjXACfY7DkEdXFVYLG564mc3Nk1pD6 63MAn3d6va2Zo5Xa+ztD0boIyLVkayRa =bQnJ -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alcatel OmniPCX Enterprise + Asterisk with E1
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Sebastian Milioto a écrit : | I see.. My hardware provider offers me TE122P with echo cancellation | module VPMADT032. | Would it be a good choice? Probably, yes. | | Now, since my costumer already has its Alcatel PBX connected to E1 ILEC | (with MFC-R2 signaling). Should I configure my Asterisk card to work | with MFC-R2 for direct replacing?, or just configure the Alcatel PBX to | work with default TEXXXP card configuration?. | Is there any issues with MFC-R2 and Asterisk cards? I'm only have experience with EuroISDN, but I would indeed use the same protocol than the telco. Best regards, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAknqFx4ACgkQuu7Rv+oOo/iLxgCfUi0GdbrXHX7SoMrXuIN8W3ad 7iIAnjp+Eq/RIk70qihRL8WxVxcfNujC =usfb -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alcatel OmniPCX Enterprise + Asterisk with E1
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Sebastian, Sebastian Milioto a écrit : | 1. Anybody has done this interconection? How does Asterisk and PBX | OmniPCX work together through an E1 interface? Any problems or bugs? I have two similar installations (OmniPCX-E1-Asterisk); they have been running, one of them since early 2005, passing more than 700 000 calls. | | 2. What E1 card should I buy for Asterisk? Is the physical interface | (conectors) E1 identical as T1? Digium TE110P or newer. | | 3. If cost wasn't a problem, do you suggest another interconection way | technically better? May be replacing Asterisk with another device with | an in-box E1? I now prefer to put asterisk between the telco and the PBX, using a dual E1 card. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkno3RoACgkQuu7Rv+oOo/iHZACfRSsheLqcnq5SDaiQkNs31Mfr cZYAoI5J6Bs8ld7nDG0IyFqrhFaubQ4w =51dT -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AOC-E pass through
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Olivier a écrit : I've got a customer (a University) who should be interested in advanced call cost control techniques. Beside limiting dialing, reading AOC-D messages would be help to keep those costs down. Are these AOC-D data reliable for any route (fixed to mobile, fixed to international, ...) and up-to-date (price variation) ? Is it easy to add this data in CDR ? Well, I guess AOC messages should be reliable; they are sent by the telco so they should be ! I have zero experience with AOC-D, they don't send it here. Concerning AOC-E, they are visible in the Asterisk console, but not logged, so unfortunately useless. From my understanding this is because AOC-E message is sent when the channel is down and CDR already written by Asterisk. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkmdhuUACgkQuu7Rv+oOo/iOwwCdEQ6ReNIe8FiT8yVDvEwutWj4 gf8An10+Z4cS7jKtRO/+OpkOW1L8Qoh8 =Jvh1 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AOC-E pass through
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Olivier a écrit : just for curiosity, is AOC-E messages sending included in telco basic subscription or is an option needed for that ? cheers It's included on PRI (even partial) subscriptions, optional on BRI. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkmc89wACgkQuu7Rv+oOo/jc6ACgopAymdTqm6cI8vnTre0uZr5T +F0AoJGzU1sgpbDiWdK8ZLVckLMmJUba =0Mgk -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AOC-E pass through
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Klaus Darilion a écrit : Take a look at http://bugs.digium.com/view.php?id=7494 Thanks for the pointer; I'm already monitoring this issue, but there seems to be no progress on that, unfortunately. Unfortunately it is not yet included in Asterisk, as the patch is somehow a workaround (e.g. faking AOC-E based on last AOC-D). Here the telco is not sending AOC-D, just AOC-E. Nevertheless a customer of us uses it for some years now (Astersik 1.2) without any problems. regards klaus Jean-Denis Girard schrieb: Hi, I'd like to know what is the current situation with regard to AOC-E, when Asterisk is inserted between the telco and an existing PBX, using E1 / EuroISDN. Can Asterisk pass the AOC-E information received from the telco to the PBX, so that billing system still works? The system would be for a hotel, so breaking billing system is not possible. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkmJuusACgkQuu7Rv+oOo/iHggCghWlXnKBZ+plXZdiHQTM8kyIi QQsAn3+O2kq2jPpcoyMAcReXltDOnQ8t =uh9L -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AOC-E pass through
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I'd like to know what is the current situation with regard to AOC-E, when Asterisk is inserted between the telco and an existing PBX, using E1 / EuroISDN. Can Asterisk pass the AOC-E information received from the telco to the PBX, so that billing system still works? The system would be for a hotel, so breaking billing system is not possible. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkmJMs0ACgkQuu7Rv+oOo/gFwwCgkO0LFaJ4uOQXifeGajhZAXOe pDkAoJDnClPDX16ZuT27XXYUU02n5Uw1 =L5/q -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 John Todd a écrit : My results: The RealSpeak sample was more clear than the Cepstral. But by how much? I should probably test with more than just that one phrase, but I can't say I'd prefer RealSpeak significantly over Cepstral in this extremely limited case. Does RealSpeak get better long-term test results and comprehension/retention? I know that Cepstral is $50/port - the RealSpeak pricing is un-findable, which tells me that it's significantly higher than Cepstral. (Personal peeve: at least put your list pricing on the website! grumble) For French language, I find the quality of RealSpeak to be very good. Festival was unusable (for French); I tried Cepstral but was deceived. The price of RealSpeak is not far from an order of magnitude higher compared to Cepstral. That being said, I'd really be interested in hearing if anyone has done a RealSpeak-to-Asterisk conduit. I wasn't able to quickly uncover how they interact with third-party systems - is it VoIP? A C library? Some sort of HTTP socket? The more methods we can get working with Asterisk, the better, because not every implementation of a voice system has the same requirements... That's a C library. I bought RealSpeak SDK, and developed app_realspeak for Asterisk (1.2, then ported to 1.4). I've been using it since 2005 for my IVR projects, including telcos/banks/airlines :) Regards, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkk10kAACgkQuu7Rv+oOo/gK2ACfXedtJ8k7cmVRpOqTU+rYpbVy PcIAnjbXbDPuicE29673TQY3CritOksQ =vvB7 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MozIAX - Mozilla IAX2 soft-phone 3sec delay
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Joseph a écrit : | Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone. | http://moziax.mozdev.org/ | | I tried it yesterday on eee pc, connected to asterisk on local LAN and the performance is terrible! | The delay is about 2sec or 3sec. and very bad echo. | I think it is the implementation of their IAX2 in their add on, as I have tried external mic. and the same delay problem. | | As a comparison I've tried DIAX over dial-up connection and the voice quality was acceptable with very little delay. | MozIAX and DIAX share the same IAX2 implementation (libiaxclient, see http://iaxclient.wiki.sourceforge.net/projects), so I doubt this is the problem. I'm not aware of such delay problems; maybe there is a sound daemon running on the EEE PC. Maybe you could ask on the MozIAX mailing list. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mandriva - http://enigmail.mozdev.org iEYEARECAAYFAkkoXk8ACgkQuu7Rv+oOo/im4gCfUEimQ33BMHsjyNR4fygwdOBm kP8Anj7Ei+FQNLiKslJepE2hV8xRI6fq =voCP -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium B410P, bristuff and BRI support in 1.6
Hi, Tzafrir Cohen a écrit : On Mon, Apr 07, 2008 at 06:29:09PM +0200, Olivier wrote: Do you think it could work with a Bewan Gazel 128 USB ? I could get a hand on one and before diving into it, I would be very curious to get your opinion on it. I read carefully the thread you pointed me to and specifically I read : http://lists.digium.com/pipermail/asterisk-dev/2008-March/032164.html From it, I understood Bristuff's zaphfc was key for that. I'm not aware of any zaptel driver for such HFC USB modem (some Xorcom's products use USB, so ...) so I'm inclined to think it's not possible but it's better to ask ... Just to clarify: Our device does not use the HFC-USB chip. I'm indeed not aware of any zaptel driver for such a device. I agree with Tzafir, I'm not aware of zaptel support for HFC-USB; I checked bristuff, it doesn't support it. Would be interesting to see one. +1 Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium B410P, bristuff and BRI support in 1.6
Olivier a écrit : Would you mind if I asked you this : - Which card did you include in your home system ? Are you using an ISDN BRI access ? This is a basic BRI card with HFC chipset (Bewan Gazel 128) - Is libpri necessary for ISDN BRI access ? I thought libpri was mostly dedicated to E1/T1 access Development version of libpri (libpri-trunk) does include prliminary support for BRI. From above, do you understand that Digium is committed to support BRI cards in 1.6 ? If positive, which cards will be supported and with which feature set ? I can't speak for Digium ! Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium B410P, bristuff and BRI support in 1.6
Olivier a écrit : 2008/4/2, Jean-Denis Girard [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Development version of libpri (libpri-trunk) does include prliminary support for BRI. I took a look at : http://svn.digium.com/view/libpri/trunk/ Though BRI support is mentioned several times but I couldn't find any supported hardware list. I'll open a new thread on this last and specific point. One last question Jean-Denis, when you wrote your system has been running fine for nearly a month with libpri-trunk, asterisk-1.6.0, zaptel-1.4 (patched) and zaphfc, does zaptel-1.4 (patched) also relates to BRI support ? I thought that (without bristuff) zaptel was dedicated to analog lines. Did you have to install zaptel for BRI support ? Zaptel also has always supported digital (PRI) cards from Digium. Though I don't have experience with this setting and other BRI cards at the moment, I believe B410P is supported, as well as other cards supported by bristuff. Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium B410P, bristuff and BRI support in 1.6
Hi, Olivier a écrit : Hi, Is it possible to both use Digium B410P and bristuff'ed 1.4 Asterisk, now ? I've heard BRI support in Asterisk is about to change with 1.6 but I'm not sure I understood what the plan is. If someone has a clue, l would delighted to learn about it. You should have a look at this thread: http://lists.digium.com/pipermail/asterisk-dev/2008-March/032142.html I'm not sure about the B410, but my home asterisk system has been running fine for nearly a month with libpri-trunk, asterisk-1.6.0, zaptel-1.4 (patched) and zaphfc. I have the following error in system logs, but it seems harmless. Apr 1 09:16:59 tiare kernel: zaphfc: dropped audio (z1=6992, z2=6958, wanted 8 got 34, dropped 26). Quality is great (no echo). Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Softphones and Citrix ?
d4rk f1br a écrit : Anyone aware of any SIP softphones that might virtualize well with Citrix presentation server? I suspect I know the answer already as I MozPhone (moziax.mozdev.org) has been designed to be run in thin client environment. I don't know much about Citrix, but I have customers running MozPhone on thin clients in a Terminal Server environment. The idea is that sound and IAX communication are managed locally on the thin client, while user interface (Firefox extension) runs on the terminal server. Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Softphones and Citrix ?
Steve Totaro a écrit : Awesome! Thanks for pointing out this link. I was looking for a thinclient softphone a couple of years ago. Any feedback on how well this works (thinclient and/or PC browser plugin)? Disclaimer: as the main developper of MozPhone I'm obviously biased :) I used it myself in a Linux LTSP environment years ago without trouble. My customers using it in the windows TS environment did lot of testing. The audio part of the thin client is obviously important. They use Neoware thin clients, and were getting good results with onboard sound card. Then they tried a first usb headset which gave very bad sound, and finally settled on the Plantronics CS60-usb headset. They contracted me to add managment of CS60 buttons (off/on hook, mute) to MozPhone, so I guess they are satisfied. Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crash related to asterisk -rx ?
Atis Lezdins a écrit : Yup, it's also a problem for me, but it haven't ever crashed server. It just makes specific remote process unresponsive. There's a patch for 1.4, but i guess it wouldn't be hard to backport it for 1.2 http://bugs.digium.com/view.php?id=10847 you might also want the one mentioned in comments: http://bugs.digium.com/view.php?id=10888 Regards, Atis Atis, Thanks for the reply and pointers. Best regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Crash related to asterisk -rx ?
Hi list, Last Friday, an Asterisk server became unresponsive after ~8,5 months of smooth operation (~32 calls). Server did reply to pings, but no ssh, no more console login. Also Asterisk no longer took calls, but ISDNguard watchdog was still alive. Looking at the logs after reboot, I could not find anything significant, except in a file created by the following command via a cron job: date /var/log/asterisk/calls.log ; asterisk -rx show channels concise /var/log/asterisk/calls.log Two days before the crash, the calls.log file started to be filled with the Asterisk console messages. I suspect this is what caused the server crash. Anybody seen this before, is this a known problem with asterisk -rx commands? Asterisk is version 1.2.15 (I can provide more details if needed). Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where to download Junghanns ISDNguard software?
Nick Richardson a écrit : What if you don't use or want to use bristuff? We use Digium PRI cards and don't need any of the BRIstuff As was said in previous posts, you don't need the full bristuff, just res_watchdog. Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where to download Junghanns ISDNguard software?
Nick Richardson a écrit : so is there no way to use this thing without recompiling? No. Also where does one get the /usr/sbin/ISDNguard binary from? By compiling ? Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where to download Junghanns ISDNguard software?
Nick Richardson a écrit : Hi list, I recently purchased an ISDNguard from Junghanns. It came with no software and there is no sign on their website or in any of their documentation where to download it. I have looked in http://www.junghanns.net/downloads/ and there is no sign of it there either. The only thing remotly close ther is isdnguard-asterisk-1.2.13.patch. Their documentation refers to /usr/sbin/ISDNguard. Where does one get this mysterious binary from? I have emailed their support a few times and get no response, needless to say I am NOT a happy customer. Can anyone help me with a download link? It is in their bristuff package: you'll have to pick res_watchdog and include it in your asterisk build. Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Java Softphone?
Matthew Rubenstein a écrit : Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. Did you try JIAXClient ? http://www.hem.za.org/jiaxclient/ Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Java Softphone?
Dean Collins a écrit : As far as I know Jiaxclient is dead - the developer hasn't touched it in at least 18 months. Correct, but this is free software, anybody with the skills can revive it :) Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcing - AstJax click2call Firefox greasemonkey script - click and dial phone numbers in any webpage
Matthew Rubenstein a écrit : Is there any FireFox plugin that contains an entire (SIP or IAX) softphone, that can also be scripted in the page's HTML/Javascript? Have you looked at MozPhone (http://moziax.mozdev.org/) ? It's a Firefox VoIP extension IAX softphone, and Asterisk manager interface. It does include click to dial, click to transfer, and could do more from a web page through javascript. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ReceiveText()?
Olle E Johansson a écrit : 24 feb 2007 kl. 03.15 skrev Yuan LIU: How do I receive text sent from SendText() application? Asterisk lists text capability, so SendText() is successful. But I don't see an application to actually use it. EyeBeam and several SIP phones does receive those messages. IAX softphones also can display text messages and MozPhone has a chat functionnality . Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mini-ITX board + FXO PCI card?
Vincent Delporte a écrit : At 10:09 11/02/2007 -0500, Michelle Dupuis Henderson wrote: We use a lot of mini-itx pc's, including the pCI slot. I don't think any of the systems have shared an irq with the PCI slot Thanks for the tip. In that case, I have a couple of questions for you :-) 1. The smallest mini-ITX case I found that accepts a PCI card is the Travla C138: If you used a mini-ITX with a Digium TDM400P, do you know if it fits? I didn't find its width, and apparently, the C138 will not accept a PCI card bigger than 17,52cm. I'm using these cases for my Mini-TX projects: http://www.emko.cz/emko_en/produkty/skrine/miniitx/em-141.html A TDM400P fits nicely. 2. Can the Via motherboards boot from a USB drive, so I can install Linux from this and fetch the install files from an FTP server? Yes, they also boot from USB CDROM, and directly from LAN (PXE). Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] yellow alarm after weeks without trouble
Hi list, I'm getting an error on a E1 link to the telco, after some weeks of operation without trouble. I have an asterisk with a TE405 in passtrough mode: two E1 are connected to the Telco, two E1 are connected to 2 Siemens PaBX. Only 15 channels are used on each E1 (conf is attached).The system has been in production for nearly a year, and does work flawlessly for weeks, then I mysteriously get errors on the first PRI span, which is used as primary clocking source. It happened again this Sunday (while the system was idle), after six weeks of operation without trouble. Call are dropped, and users are angry. It has happened 4-5 times during one year. Reloading zaptel modules and restarting asterisk does bring the system back to normal. As far as I can tell, pri debug span 1 shows nothing special (see attached log). The Telco says that there is drift in the synchronization. System load as shown by top is very low, vmstat shows no CPU spikes, TE405P has its on interrupt. I already changed the TE405P. I have other similar systems running without any trouble, so any idea about improving this installation would be welcome. I can provide more info if needed. Main differences with my other similar installations are: two E1 are used, and system is 64 bit (Asus MB with Opteron). Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 # Zaptel.conf: fichier de configuration bas niveau de zaptel # SysNux 07/12/2005 # Jean-Denis Girard [EMAIL PROTECTED] loadzone=fr defaultzone=fr # 4 canaux T2 (2 sur OPT, 2 sur PaBX). Chaque T2 utilise seulement # les 15 premiers canaux B span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 #bchan=17-31 span=2,2,0,ccs,hdb3 bchan=32-46 dchan=47 #bchan=48-62 span=3,0,0,ccs,hdb3 bchan=63-77 dchan=78 #bchan=79-93 span=4,0,0,ccs,hdb3 bchan=94-108 dchan=109 #bchan=110-124 ; Zaptata.conf: fichier de configuration des interfaces zap ; 4 T2: 2 connectés à l'OPT, 2 connectés au PaBX ; sur chaque T2 seulement 15 canaux B utilisés ; SysNux 07/12/2005 ; Jean-Denis Girard [EMAIL PROTECTED] [channels] language=fr switchtype=EuroISDN usecallerid=yes callwaiting=no callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes pridialplan=national prilocaldialplan=national internationalprefix=00 nationalprefix=0689 overlapdial=yes callerid=asreceived immediate=no ; XXX ; 2 premiers T2 connectés à l'OPT context=entrant signalling=pri_cpe group=1 channel = 1-15 ;channel = 17-31 channel = 32-46 ;channel = 48-62 ; 2 derniers T2 connectés au PaBX context=sortant signalling=pri_net immediate=no group=2 channel = 63-77 ;channel = 79-93 channel = 94-108 ;channel = 110-124 debug.bz2 Description: BZip2 compressed data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] yellow alarm after weeks without trouble
Hi list, I'm getting an error on a E1 link to the telco, after some weeks of operation without trouble. I have an asterisk with a TE405 in passtrough mode: two E1 are connected to the Telco, two E1 are connected to 2 Siemens PaBX. Only 15 channels are used on each E1 (conf is attached).The system has been in production for nearly a year, and does work flawlessly for weeks, then I mysteriously get errors on the first PRI span, which is used as primary clocking source. It happened again this Sunday (while the system was idle), after six weeks of operation without trouble. Call are dropped, and users are angry. It has happened 4-5 times during one year. Reloading zaptel modules and restarting asterisk does bring the system back to normal. As far as I can tell, pri debug span 1 shows nothing special (see attached log). The Telco says that there is drift in the synchronization. System load as shown by top is very low, vmstat shows no CPU spikes, TE405P has its on interrupt. I already changed the TE405P. I have other similar systems running without any trouble, so any idea about improving this installation would be welcome. I can provide more info if needed. Main differences with my other similar installations are: two E1 are used, and system is 64 bit (Asus MB with Opteron). Versions are: Asterisk 1.2.13 built by root @ asterisk.xxx.xx on a x86_64 running Linux on 2006-11-10 16:12:43 UTC Zaptel Version: 1.2.11 Echo Canceller: KB1 Found TE4XXP at base address fdefe000, remapped to c201 TE4XXP version c01a0164, burst OFF, slip debug: OFF FALC version: 0005, Board ID: 00 Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 # Zaptel.conf: fichier de configuration bas niveau de zaptel # SysNux 07/12/2005 # Jean-Denis Girard [EMAIL PROTECTED] loadzone=fr defaultzone=fr # 4 canaux T2 (2 sur OPT, 2 sur PaBX). Chaque T2 utilise seulement # les 15 premiers canaux B span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 #bchan=17-31 span=2,2,0,ccs,hdb3 bchan=32-46 dchan=47 #bchan=48-62 span=3,0,0,ccs,hdb3 bchan=63-77 dchan=78 #bchan=79-93 span=4,0,0,ccs,hdb3 bchan=94-108 dchan=109 #bchan=110-124 ; Zaptata.conf: fichier de configuration des interfaces zap ; 4 T2: 2 connectés à l'OPT, 2 connectés au PaBX ; sur chaque T2 seulement 15 canaux B utilisés ; SysNux 07/12/2005 ; Jean-Denis Girard [EMAIL PROTECTED] [channels] language=fr switchtype=EuroISDN usecallerid=yes callwaiting=no callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes pridialplan=national prilocaldialplan=national internationalprefix=00 nationalprefix=0689 overlapdial=yes callerid=asreceived immediate=no ; XXX ; 2 premiers T2 connectés à l'OPT context=entrant signalling=pri_cpe group=1 channel = 1-15 ;channel = 17-31 channel = 32-46 ;channel = 48-62 ; 2 derniers T2 connectés au PaBX context=sortant signalling=pri_net immediate=no group=2 channel = 63-77 ;channel = 79-93 channel = 94-108 ;channel = 110-124 Jan 31 12:41:45 VERBOSE[9968] logger.c: -- Hungup 'Zap/68-1' Jan 31 12:41:46 VERBOSE[32105] logger.c: Protocol Discriminator: Q.931 (8) len=39 Jan 31 12:41:46 VERBOSE[32105] logger.c: Call Ref: len= 2 (reference 25024/0x61C0) (Terminator) Jan 31 12:41:46 VERBOSE[32105] logger.c: Message type: RELEASE (77) Jan 31 12:41:46 VERBOSE[32105] logger.c: [1c 20 91 a1 1d 02 02 01 8d 02 01 24 30 14 30 12 a1 10 30 06 02 01 01 02 01 01 30 06 02 01 00 02 01 02] Jan 31 12:41:46 VERBOSE[32105] logger.c: Facility (len=34, codeset=0) [ Jan 31 12:41:46 VERBOSE[32105] logger.c: Facility (len=34, codeset=0) [ 0x91, 0xa1, 0x1d, 0x02, 0x02, 0x01, 0x8d, 0x02, 0x01, 0x24, '0', 0x14, '0', 0x12, 0xa1, 0x10, '0', 0x06, 0x02, 0x01, 0x01, 0x02, 0x01, 0x01, '0', 0x06, 0x02, 0x01, 0x00, 0x02, 0x01, 0x02Jan 31 12:41:46 VERBOSE[32105] logger.c: Facility (len=34, codeset=0) [ 0x91, 0xa1, 0x1d, 0x02, 0x02, 0x01, 0x8d, 0x02, 0x01, 0x24, '0', 0x14, '0', 0x12, 0xa1, 0x10, '0', 0x06, 0x02, 0x01, 0x01, 0x02, 0x01, 0x01, '0', 0x06, 0x02, 0x01, 0x00, 0x02, 0x01, 0x02 ] Jan 31 12:41:46 VERBOSE[32105] logger.c: -- Processing IE 28 (cs0, Facility) Jan 31 12:41:46 VERBOSE[32105] logger.c: Handle Q.932 ROSE Invoke component Jan 31 12:41:46 VERBOSE[32105] logger.c: !! Don't know how to handle 0x30 in AOC-E RecordedUnitsList Jan 31 12:41:46 VERBOSE[32105] logger.c: NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Jan 31 12:41:46 VERBOSE[32105] logger.c: Protocol Discriminator: Q.931 (8) len=9 Jan 31 12:41:46 VERBOSE[32105] logger.c: Call Ref: len= 2 (reference 25024/0x61C0) (Originator) Jan 31 12:41:46 VERBOSE[32105] logger.c: Message type: RELEASE COMPLETE (90) Jan 31 12:41:46 VERBOSE[32105] logger.c: [08 02 81 90] Jan 31 12:41:46 VERBOSE[32105] logger.c: Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private
Re: [asterisk-users] popups, queue agents
Todd- Asterisk a écrit : Hi everyone - I have a nicely working system to which I'd like to add popups for incoming calls. Calls go into a queue, then all extensions ring. I'd like the agent that answers to call to get the popup on screen. I'm currently using Flash Operator Panel to get a popup (other suggestions welcome). Currently, all users get a popup when the call first goes into the queue which obviously isn't that great Where in the dial-plan do I put the code for the popup and specify only the agent to whom the call is connected? Use the URL option of the Queue application, and a softphone that supports URL: MozPhone is my favorite :) Jean-Denis ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE:Asterisk and dialer Running on Thin Clients
Ignacio Ortega A. a écrit : *Vitaly,* could you please be more spesific about all you did in order to get tis done, ill do anithing to aconplish this. Have a look at the mailing list archive of MozPhone (moziax.mozdev.org): back in August, Machula Viach made modifications in order to run MozPhone in TSE environment. I plan to integrate his changes but it's not done yet. MozPhone was specifically developped with thin client in mind (complete separation of IAX / sound processing running on the client and user interface running on the server); and I have successfully used it with LTSP (www.ltsp.org), though not for hundreds of clients. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
