Hi,

I created a new package for Asterisk 1.6.x (asterisk16). The "minimized" approach was applied here: it includes only very basic stuff, similar to the current asterisk-mini packages for 1.4.x. At the moment it only contains support for chan_sip. Additional features could be added, preferably as separate packages (if not required for basic operation).

Feel free to test and send your comments in this topic. I created a mipsel package for 8.09 only so far, it is in the repository at:
http://members.home.nl/hans.zandbelt/op … l/packages

I'd like to submit the package source to SVN when reaching rough consensus about dropping 1.2.x in favour of 1.6.x.

Regards,

Hans.

Hello Zandbelt,

Thanks for your work.
I was going to compile 1.6.x for testing purposes. There is no need to do so right now since you are tackling this task.
Excellent work, my friend

Hello Zandbelt,

can you please share your Makefile on https://dev.openwrt.org/ticket/3895 or create a new ticket?

Thank you.

I will submit it as a new package to the packages repository; it needs some cleansing first; you can mail me directly if you really need it right away

Hans.

Hi! Great work! How did you build the package? I need Asterisk on my OpenWRT powered AR7161 board, but I need a more complete Asterisk configuration (bristuff, sql+ldap support, etc.) so it's better if I try to rebuild it myself...

Thank you... and work correctly with 8.09 RC1 and Fonera 2100?

p.s. this is for mipsel, Fonera is mips, not work...
... can you compile for mips?

Regards,
Elia

(Last edited by elfo78 on 27 Jan 2009, 10:57)

I'm interested in installing the a version of OpenWRT with Asterisk 1.6.x integrated.  Is this the file i'm looking for: http://downloads.openwrt.org/snapshots/ … uashfs.bin

And is the webgui included or do I have to install it using opkg?  If that's the case can someone show me which package contains the webgui?

LA Dude:

There is no web gui for this at the moment but it does work if you know how to configure the various files.

There is work going on at the moment on a simplified LuCI gui for Asterisk. It should be available shortly.

marc.

Please keep us updated on when the LuCI gui for Asterisk is available.  Will it be included with OpenWRT or is it an addon?

LA Dude:

It's currently in the LuCI trunk. The plan is to include it in the main trunk builds shortly.

Currently, it is configured to work with Asterisk 1.4.

marc.

I have installed asterisk1.4.23.1 on openwrt (ASUS 500gp).
Actually, my asterisk is running perfectly. I've tested it, and I called some computer in my LAN.
but I've experiencing a problem. it can't load codec_gsm.so

I google this problem, someone said: it is version confuse, may I used prevenient codec_gsm.so file.
but i install it from http://downloads.openwrt.org/kamikaze/8 … 4/packages

The foolowing is my problem display:

root@OpenWrt:/# asterisk -r
Asterisk 1.4.23.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.4.23.1 currently running on OpenWrt (pid = 1032)
Verbosity is at least 6
OpenWrt*CLI> module load codec_gsm.so
[Jan  1 21:40:32] WARNING[1099]: loader.c:372 load_dynamic_module: Module 'codec_gsm.so' did not register itself during load
[Jan  1 21:40:32] WARNING[1099]: loader.c:653 load_resource: Module 'codec_gsm.so' could not be loaded.
OpenWrt*CLI>

Anybody can help me? thanks

please follow https://dev.openwrt.org/ticket/4592

Thanks, zandbelt.

I just rebulding the openwrt img, everything is ok. the following is my step:
1, install kamikaze_8.09
2, make menuconfig and select the asterisk14 to build a module.
3. make it
4, use tar  command extract the codec_gsm.so from new  asterisk14_1.4.23.1-1_mipsel.ipk file
5. replace codec_gsm.so, everything is ok.

Thanks for the great job, zandbelt.

The package, asterisk14 (ver 1.4.22) has some problem with sip CANCEL not forwarded properly (without Contact info, etc.) Many docile carrier proxies work quite well with it, but with some, you might have the 'never disconnecting' problem. Of course, this may not be a asterisk bug, but if it doesn't work, it's darn broken to me. Anybody has similar problem, get a 1.6

Your asterisk16 package does not have this problem and works well. Though when I installed asterisk16, it said libssl.so was missing. I had to install libssl seperately. (maybe it's due to the fact that I had removed asterisk14 package first. maybe libssl had gone with it?) BTW, I am using this with FON2100 as a home pbx.

(Last edited by test011 on 16 Mar 2009, 23:24)

1. asterisk16rc1 seems to have a problem with dnsmgr (it looks up too often when failed, souldn't lookup section name in register string, etc.) Well, I guess one can live with it, though.

2. With asterisk16rc1 package does not have chan_IAX by default. Could somebody generate the package please?

----------- edit: dnsmgr problem is said to be fixed with 1.6.1.0 release according to change.log at asterisk.org

-------------- however, it does not seem so.

(Last edited by test011 on 14 May 2009, 03:25)

I fond there have no NVFaxDetect application in Openwrt asterisk1.4.
app_nv_faxdetect-1.0.6_1.4.so   (Detects fax sounds on all channel types)
How I got this applications? Thanks

I believe Callerid function is pretty much the basic stuff with SIP configuration. Without it asterisk is not a very useful in real world environment. End-user confusion would be, as always, a nightmare to system admins.

Current Asterisk 1.6.1rc1 package does not have this module, so rather simple extension.conf would not run on it. Can't hardly test anything but a test call. Please, please have it available.

btw, there is a 1.6.1.rc3 on the Asterisk site.

I just updated the asterisk-1.6.x trunk package in SVN to 1.6.1.0-rc3, including chan_iax2, func_callerid and libopenssl dependencies.
That should address the issues raised above.

Hans.

Hi Hans,
Is it possible to extract full list of modules available in asterisk-1.6 package?
I'm specially interested in such functions like format_g729.so,cdr_csv.so,app_db.so and func_db.so
thanks a lot

so far I have:

app_adsiprog.so         app_nbscat.so           cdr_custom.so    format_sln16.so           func_strings.so
app_alarmreceiver.so    app_parkandannounce.so  cdr_manager.so   format_sln.so             func_sysinfo.so
app_amd.so              app_playback.so         cdr_pgsql.so     format_vox.so             func_timeout.so
app_authenticate.so     app_privacy.so          cdr_sqlite.so    format_wav_gsm.so         func_uri.so
app_cdr.so              app_queue.so            chan_agent.so    format_wav.so             func_version.so
app_chanisavail.so      app_readexten.so        chan_alsa.so     func_audiohookinherit.so  func_vmcount.so
app_channelredirect.so  app_readfile.so         chan_iax2.so     func_base64.so            func_volume.so
app_chanspy.so          app_read.so             chan_local.so    func_blacklist.so         pbx_ael.so
app_controlplayback.so  app_record.so           chan_mgcp.so     func_callerid.so          pbx_config.so
app_db.so               app_sayunixtime.so      chan_oss.so      func_cdr.so               pbx_dundi.so
app_dial.so             app_senddtmf.so         chan_phone.so    func_channel.so           pbx_loopback.so
app_dictate.so          app_sendtext.so         chan_sip.so      func_config.so            pbx_realtime.so
app_directed_pickup.so  app_setcallerid.so      chan_skinny.so   func_cut.so               pbx_spool.so
app_directory.so        app_sms.so              chan_unistim.so  func_db.so                res_adsi.so
app_disa.so             app_softhangup.so       codec_adpcm.so   func_devstate.so          res_ael_share.so
app_dumpchan.so         app_speech_utils.so     codec_alaw.so    func_dialgroup.so         res_agi.so
app_echo.so             app_stack.so            codec_a_mu.so    func_dialplan.so          res_clioriginate.so
app_exec.so             app_system.so           codec_g722.so    func_enum.so              res_config_pgsql.so
app_externalivr.so      app_talkdetect.so       codec_g726.so    func_env.so               res_config_sqlite.so
app_festival.so         app_test.so             codec_gsm.so     func_extstate.so          res_convert.so
app_followme.so         app_transfer.so         codec_lpc10.so   func_global.so            res_crypto.so
app_forkcdr.so          app_url.so              codec_ulaw.so    func_groupcount.so        res_indications.so
app_getcpeid.so         app_userevent.so        format_g723.so   func_lock.so              res_limit.so
app_ices.so             app_verbose.so          format_g726.so   func_logic.so             res_monitor.so
app_image.so            app_voicemail.so        format_g729.so   func_math.so              res_musiconhold.so
app_macro.so            app_waitforring.so      format_gsm.so    func_md5.so               res_phoneprov.so
app_milliwatt.so        app_waitforsilence.so   format_h263.so   func_module.so            res_realtime.so
app_minivm.so           app_waituntil.so        format_h264.so   func_rand.so              res_smdi.so
app_mixmonitor.so       app_while.so            format_ilbc.so   func_realtime.so          res_speech.so
app_morsecode.so        app_zapateller.so       format_jpeg.so   func_sha1.so              res_timing_pthread.so
app_mp3.so              cdr_csv.so              format_pcm.so    func_shell.so             test_heap.so

I guess format_g729.so could go in a separate package.
Would the *db* and *csv* stuff naturally fit into a single package?

Hans.

Hans
normally all this packages I mentioned + app_dial where previously in asterisk14 full package ( not mini one) they are not so big. I might be  wrong but they used a lot.
I would say format_g729 is worth separating but *db* *csv* + *cdr* would  be nice to see in some kind of asterisk_file_utils.
Thanks

1.6.1rc3 works well. So far so good. Great job, Hans.
I also would like to see more of those modules. Maybe till Digium releases 1.6.1 full?

btw, If any users want to deploy the call transfer feature with current asterisk 1.6.1rc3 on openwrt, I suggest to wait until there is res_musiconhold module available.

When someone inside does a call transfer, because callers hear nothing during the process (due to no musiconhold module) they go panic and after a couple of 'hello?' they just hang up. Yes, people can't wait for mere 10 sec. these days without any music on them. Anyway no way to educate outside callers, I had to suspend all call transfer to a halt. kinda embarrassing.

Finally long awaited Asterisk 1.6.1.0 is released at asterisk.org

I am currently using svn version of rc3, thanks to Hans.
Though rc3 is not very stable release - since they had put on rc4 and rc5 so fast - and I had to put some cron script to boot my openwrt every two hours, openwrt + asterisk is a great promise to soho pbx community.

I am looking forward to see 1.6.1.0 port for openwrt soon. Thanks again guys.

------
MHO for essential (very incomplete) package includes, (Using as a soho pbx, receive&call, blind transfer, and 'who-called-me-yesterday?' style call log)

chan_local, iax2, sip
func_callerid, string, timeout, cdr
app_dial, echo, macro, gosub, playback, cdr(for simple call log)
various codec&format (ulaw, gsm, if possible 729)
res_musiconhold (simple call transfer needs this)

(Last edited by test011 on 30 Apr 2009, 08:25)

I was wondering where can I get  precompiled app_playback and app_macro for asterisk 1.6 (I am currently using asterisk 1.6.1rc1-1.).

I was configuring  Asterisk 1.4 but run into a problem (pstn sip provider not detecting my hangups) which I did not find when trying 1.6.

Thanks,
Roberto.

most of the stuff referred to above should be in the packages trunk now, which was updated to release 1.6.1.0

Hans.