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Topic: asterisk 1.4.5 plus addons

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Hi,

I updated the Asterisk 1.4 repositories to the most recent versions as follows:

  Asterisk 1.4.5
  - app_fax (app_rxfax, app_txfax, depending on libspandsp 0.0.4pre2 which is also provided)
  - chan_gtalk (includes the recent stability fixes and NAT support)
  - chan_alsa
  - chan_oss
  - chan_h323
  - incl. the usual modules such as app_meetme, sqllite, res_crypto, res_agi etc.

  Asterisk Addons 1.4.2
  - chan-mobile (formerly called chan_cellphone; backported from trunk to 1.4.2)
  - chan-ooh323
  - format-mp3
  - mysql

  Asterisk GUI
  - svn version 1146

Repositories for WhiteRussian/Kamikaze and additional information can be found via: http://zandbelt.dyndns.org/asterisk.html .
Please test these releases and report your results back here.

Hans.

(Last edited by zandbelt on 22 Jun 2007, 23:04)

Asus wl-500gP + whiterussian + Asterisk 1.4.5 + ooh323 works fine without any problems till now.

Than you Hans!!!!
Sincerely
Kirill

Its working great for me . on wrtsl54gs

Thank you.

One question.. is there anything in * and openwrt that is read/write intensive and  can short the life of the router? That i should consider moving to the USB stick?

thanks again.

a1jatt wrote:

One question.. is there anything in * and openwrt that is read/write intensive and  can short the life of the router? That i should consider moving to the USB stick?

from the core that would be the asterisk database with eg. sip registrations; that one is now taken care of via the startup script: it is/will-be placed on ram space, so no need do do this manually anymore.

other modules such as voicemail and cdr logging can potentially write flash many times depending on its usage; otoh, I haven't heard of any worn out cases yet... (nb: if you are in the position to use a USB device, I would rather use a harddrive than a USB stick)

Hans.

^^^ thank you smile

Can't seem to make it work here, prob the router I have WRT54G ver 3 with kamikazee loaded.
everytime I go to load the package it locks my session up and doesn't do anything. Is this a memory issue or could I actually be doing something wrong.

Thank You...

P.s looks like a great package for what I have read so far :-)

bobsj2000 wrote:

Can't seem to make it work here, prob the router I have WRT54G ver 3 with kamikazee loaded.
everytime I go to load the package it locks my session up and doesn't do anything. Is this a memory issue or could I actually be doing something wrong.

i had the same problem on asus wl500gx on Kamikaze; I managed to work around it by doing individual installs per package instead of installing the whole lot at once; you could also try and download first on local media and then install it

update: forgot to mention: a reboot may also help

looks like this is a Kamikaze related problem probably showing up only with large size packages; perhaps you could take this up in another topic/forum

Hans.

(Last edited by zandbelt on 28 Jun 2007, 21:04)

Hello,

Yesterday I installed Kamikaze 7.06 on my Asus WL500gP and asterisk 1.4.5 with the GUI.
I followed the instructions I found on http://zandbelt.dyndns.org/asterisk.html .

Asterisk is working fine, however I encountered some issues with the GUI.

The gui installation wizard was never showing the setup steps .. it kept reloading the page; connecting to the asterisk console there were a lot of errors and that was solved by loading the app_system.so module in asterisk.

After doing this the wizard was loading and I was able to complete some step; however the trunk configuration gave me an error, saying that a file was missing (extensions.conf), while that file was actually in the /etc/asterisk directory.
I tried to comment the "TRUNKMSD=1" at the beginning of the file and.. the wizard was able to complete and the GUI seemed to work fine.

However, it was not working fine as the gui was able to create the extensions and the dialplans correctly, but the trunks were incomplete because for instance the variable "TRUNK_1=SIP/trunk_1" was not set and the dialplan was not working (as it tried to call  "/123456" instead of "SIP/trunk_1/123456").
When I set the "TRUNK_1=SIP/trunk_1" manually.. then the GUI stopped working again, redirecting me to the wizard.

How can I fix this?
Am I the only one experiencing this issue?

Thank you!!

Regards,
Federico

federkicco wrote:

How can I fix this?
Am I the only one experiencing this issue?

Most of the problems with the GUI are not related to OpenWRT but are due to the fact that the GUI is not production quality (yet)
In fact I have doubts that the way forward is really the path that asterisk-gui has chosen.

You should really ask these questions on the asterisk-gui mailinglist and find out there what the cause of your problems is (if that points back to OpenWRT we might be able to do something about it).

Concluding: the asterisk14-gui package is an alpha release and is there merely for testing and playing with it; don't use it for "production", it is not yet really supported on OpenWRT yet, at least not by me ... (I'll add a statement along those lines on the webpages for the Asterisk 1.4 repositories).

Update: if anyone wants to put some effort in supporting and maintaining asterisk14-gui, please contact me.

Hans.

(Last edited by zandbelt on 30 Jun 2007, 10:59)

zandbelt wrote:

Please test these releases and report your results back here.

Hans.

Thank you Hans,

I know the limitations about the gui.. no issues, I'm just playing with it! smile
The main purpose of my questions was actually to report the results of the tests with your packages and to understand if someone else is experiencing similar issues with the same packages in OpenWrt?
Perhaps I did something wrong with my setup..?

In the meanwhile I will try to ask the same questions to the asterisk-gui mailing lists.. wink

Cheers,
Federico

federkicco wrote:
zandbelt wrote:

Please test these releases and report your results back here.

Hans.

The main purpose of my questions was actually to report the results of the tests with your packages and to understand if someone else is experiencing similar issues with the same packages in OpenWrt?

Right, thanks for reporting! I justed wanted describe the current state of matters as far as I'm concerned; unfortunately I can't help you with your specific question because I don't used the GUI myself.

Good luck,

Hans.

Hello Hans!

Is there any plans to port asterisk14 package to RouterBoard 532(A) platform?

As we know it can run at 400 MHz and has 32 (64) Mb of RAM. So I think it could be very impressive appliance...

Sincerely
Kirill

(Last edited by iam on 15 Jul 2007, 23:19)

Hi everybody,

I'm still playing with asterisk 1.4.5 on my Asus WL-500gP.
The GUI doesn't work for me... anyway, I tried to install and configure the chan_mobile.
The BT dongle is configured on the system, an I can see my mobile from openwrt and viceversa.
However if I load chan_mobile in asterisk, I got a nice segmentation fault.
I was following the guides I found on the web; does anybody played with this stuff and got it working?

Cheers,
Fede

Hi ppl !

First of all, my very thanks to you, Zandbelt, for this superb work. I have 2 wrt54gl 1.1 with your 1.2 packages: for what i use, flawless !

But talk with my friends that use gtalk sure will be a plus ! Thats's why i'm here to ask if it's possible to point a link where i can dl the latest patches, because i want to build it from inside whiterussian buildroot, as i did with 1.2 (my whiterussian is "a bit" custom smile

TIA for my ask, and again for your generous masterpieces,

Mauro

federkicco ,

I think you've forgotten to run sdpd before starting asterisk

When i don't run this , mine asterisk segfaults too....

Hi zandbelt,

chan_cellphone is driving me mad... I simply get no audio (whiterussian, kemikaze 7.06 and 7.07), even thou A* says 'incomin SCO connection' and sco/l2cap/hci_usb debugging outputs look fine. Only got audio with your A* 1.4.4 patches with kernel 2.6.19.2 (same as kamikaze 7.06) on a laptop (same cell phone and sip phone).

Did you get it working on your wl500? Which A*, openwrt version?

Andy

Hi,

I have updated the repositories to Asterisk version 1.4.9.
NB: chan_cellphone users: chan_cellphone in asterisk14 is obsoleted by chan_mobile included in asterisk14-addons.
asterisk14-addons-1.4.2-2 contains a backported chan_mobile with SDP fixes so it should no longer crash when sdpd is not running.

Hans.

zandbelt wrote:

Hi,

I have updated the repositories to Asterisk version 1.4.9.
NB: chan_cellphone users: chan_cellphone in asterisk14 is obsoleted by chan_mobile included in asterisk14-addons.
asterisk14-addons-1.4.2-2 contains a backported chan_mobile with SDP fixes so it should no longer crash when sdpd is not running.

Hans.

Hans,

When using chan_mobile on wl500gp with kamikaze 7.07 2.6 kernel

asterisk: can't resolve symbol 'ast_debug'

when starting asterisk.

Grts Roel

cuppie wrote:

asterisk: can't resolve symbol 'ast_debug'

forgot to backport ast_debug with chan_mobile; fixed now in an updated release of asterisk14-addons including asterisk14-addons-chan-mobile_1.4.2-4

good luck

Hans.

zandbelt wrote:
cuppie wrote:

asterisk: can't resolve symbol 'ast_debug'

forgot to backport ast_debug with chan_mobile; fixed now in an updated release of asterisk14-addons including asterisk14-addons-chan-mobile_1.4.2-4

good luck

Hans.

Thank you very much !

Now I get a segmentation fault when a call from gsm is bridged.
I found a possible solution , but it seems more a kind of workaround....
Its about changing channel.c
I'm using a CSR bluetooth dongle (Sitecom CN-521)
The bug is about another one...

http://bugs.digium.com/view.php?id=10163


Debug output :

Asterisk Ready.
*CLI> [Jan  1 00:01:38] DEBUG[766]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for YjFjNWNiMGZlMTkwNjM2ZmYwMDM0NDgwZjYxNjQzZDY. - SUBSCRIBE (No RTP)
    -- Bluetooth Device gsm-roel has connected.
]Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:852 rfcomm_write: rfcomm_write() (gsm-roel) [AT+BRSF=4
[Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [+BRSF: 47]
[Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [OK]
]Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:852 rfcomm_write: rfcomm_write() (gsm-roel) [AT+CIND=?
[Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [+CIND: ("call",(0,1)),("service",(0,1)),("call_setup",(0-3)),("callsetup",(0-3))]
[Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:1177 do_monitor_phone: CIEV_CALL=1 CIEV_CALLSETUP=4
[Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [OK]
]Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:852 rfcomm_write: rfcomm_write() (gsm-roel) [AT+CIND?
[Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [+CIND: 0,1,0,0]
[Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [OK]
]Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:852 rfcomm_write: rfcomm_write() (gsm-roel) [AT+CMER=3,0,0,1
[Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [OK]
]Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:852 rfcomm_write: rfcomm_write() (gsm-roel) [AT+CLIP=1
[Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [OK]
]Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:852 rfcomm_write: rfcomm_write() (gsm-roel) [AT+CMGF=1
[Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [OK]
]Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:852 rfcomm_write: rfcomm_write() (gsm-roel) [AT+CNMI=2,1,0,1,0
[Jan  1 00:01:41] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [OK]
    -- Bluetooth Device gsm-roel initialised and ready.
[Jan  1 00:01:44] NOTICE[765]: chan_mobile.c:1646 do_sco_listen: sco_socket returns 15...
[Jan  1 00:01:44] DEBUG[765]: chan_mobile.c:1652 do_sco_listen: Incoming Audio Connection from device 00:13:FD:82:C5:58 MTU is 64
[Jan  1 00:01:44] NOTICE[765]: chan_mobile.c:1660 do_sco_listen: about to close the pvt-sco_socket and set it ns
[Jan  1 00:01:44] NOTICE[765]: chan_mobile.c:1644 do_sco_listen: About to accept the sco_socket...
[Jan  1 00:02:24] DEBUG[766]: chan_sip.c:2624 do_setnat: Setting NAT on RTP to Off
[Jan  1 00:02:24] DEBUG[766]: chan_sip.c:4375 sip_alloc: Allocating new SIP dialog for 3C9035FC805A1AC4@192.168.249.2 - INVITE (With RTP)
[Jan  1 00:02:24] DEBUG[766]: chan_sip.c:2624 do_setnat: Setting NAT on RTP to Off
[Jan  1 00:02:24] DEBUG[766]: chan_sip.c:13547 handle_request_invite: Checking SIP call limits for device openwrt
[Jan  1 00:02:24] DEBUG[774]: pbx.c:1809 pbx_extension_helper: Launching 'Dial'
    -- Executing [06xxxxxxxx@default:1] Dial("SIP/openwrt-100645f8", "MOBILE/gsm-roel/06xxxxxxxx|45") in new stack
[Jan  1 00:02:24] DEBUG[774]: rtp.c:1574 ast_rtp_make_compatible: Channel 'Mobile/gsm-roel-b195' has no RTP, not doing anything
[Jan  1 00:02:24] DEBUG[774]: channel.c:3470 ast_channel_inherit_variables: Not copying variable STACK-default-06xxxxxxxx-1.
[Jan  1 00:02:24] DEBUG[774]: channel.c:3470 ast_channel_inherit_variables: Not copying variable SIPCALLID.
[Jan  1 00:02:24] DEBUG[774]: channel.c:3470 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT.
[Jan  1 00:02:24] DEBUG[774]: channel.c:3470 ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
[Jan  1 00:02:24] DEBUG[774]: channel.c:3470 ast_channel_inherit_variables: Not copying variable SIPURI.
[Jan  1 00:02:24] DEBUG[774]: chan_mobile.c:511 mbl_call: Calling gsm-roel/06xxxxxxxx on Mobile/gsm-roel-b195
    -- Called gsm-roel/06xxxxxxxx
[Jan  1 00:02:24] DEBUG[774]: channel.c:2999 set_format: Set channel Mobile/gsm-roel-b195 to read format ulaw
[Jan  1 00:02:24] DEBUG[774]: channel.c:2999 set_format: Set channel SIP/openwrt-100645f8 to read format slin
]Jan  1 00:02:25] DEBUG[771]: chan_mobile.c:852 rfcomm_write: rfcomm_write() (gsm-roel) [ATD06xxxxxxxx;
[Jan  1 00:02:25] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [OK]
[Jan  1 00:02:25] DEBUG[749]: chan_mobile.c:757 mbl_devicestate: Checking device state for device gsm-roel
[Jan  1 00:02:25] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [+CIEV: 3,2]
[Jan  1 00:02:25] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [+CIEV: 4,2]
[Jan  1 00:02:32] NOTICE[765]: chan_mobile.c:1646 do_sco_listen: sco_socket returns 19...
[Jan  1 00:02:32] DEBUG[765]: chan_mobile.c:1652 do_sco_listen: Incoming Audio Connection from device 00:13:FD:82:C5:58 MTU is 64
[Jan  1 00:02:32] NOTICE[765]: chan_mobile.c:1660 do_sco_listen: about to close the pvt-sco_socket and set it ns
[Jan  1 00:02:32] NOTICE[765]: chan_mobile.c:1644 do_sco_listen: About to accept the sco_socket...
[Jan  1 00:02:34] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [+CIEV: 3,3]
[Jan  1 00:02:34] DEBUG[771]: chan_mobile.c:1136 do_monitor_phone: rfcomm_read() (gsm-roel) [+CIEV: 4,3]
    -- Mobile/gsm-roel-b195 is ringing
[Jan  1 00:02:34] DEBUG[774]: rtp.c:1499 ast_rtp_early_bridge: Channel 'Mobile/gsm-roel-b195' has no RTP, not doing anything
[Jan  1 00:02:34] DEBUG[774]: chan_sip.c:6499 transmit_response_with_sdp: Setting framing from config on incoming call
[Jan  1 00:02:34] DEBUG[774]: rtp.c:2727 ast_rtp_write: Ooh, format changed from unknown to ulaw
[Jan  1 00:02:34] DEBUG[774]: rtp.c:2744 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160
Segmentation fault
root@OpenWrt:~#

Hello Hans!

Is it possible for you to share the script for building Asterisk 1.4.9 AND Add-ons for building it in Kamikaze environment.
I'd like to compile it for my RB532

Thanks in advance
Kirill

Hi,

I've imported the asterisk14 package from Hans into the kamikaze 7.07 build enviroment.
Applied the path mentioned in http://bugs.digium.com/view.php?id=10163

chan_cellphone / mobile seems to work with a nokia 6230i !

Thanks !

Hi,

Just another fix for bad audio.
You can check this by setting debug in the console logging in logger.conf.
start asterisk with -cvd option (more v's means more verbose logging , btw -cvvvd)
when the are messages about 'Overrun on sco_out_buf detected'
read following:
http://bugs.digium.com/view.php?id=8919#62973

after this, good two way audio

but excessive delay in audio, about one or two seconds.

Grts Roel

iam wrote:

Is it possible for you to share the script for building Asterisk 1.4.9 AND Add-ons for building it in Kamikaze environment.
I'd like to compile it for my RB532

Just updated the build-from-source script to Asterisk 1.4.9 and addons 1.4.2.
You can use this for compiling against Kamikaze too.

Hans.

The discussion might have continued from here.