Re: [asterisk-users] working example of t38 fax w/ 1.6.2?
Hello!
I use similar setup.
Probably you need Answer() in receiving end. And wait(3) before
receiving fax.
T.38 works fine with 1.6.2.
Ilmars.
On 2010.05.05. 0:17, sean darcy wrote:
On 5/4/2010 7:32 AM, Miguel Amez wrote:
App_fax? I didn't hear about that. What's that?
Could you please explain that a little bit better?
I'm experiencing some troubles with T38modem and would like to solve on
the better way.
regards,
Miguel Amez
2010/5/4 sean darcyseandar...@gmail.commailto:seandar...@gmail.com
Miguel Amez wrote:
Hi Sean,
Do you know about t38modem and hylafax?
There are lots of wonderfull options with both of them.
If you need config files with both of them tell me.
See ya
2010/5/2 sean darcyseandar...@gmail.com
mailto:seandar...@gmail.com mailto:seandar...@gmail.com
mailto:seandar...@gmail.com
I can't get a test T.38 fax between 2 1.6.2 machines, using app
_fax and spandsp pre17 and 20100501. The machines can't seem
to get
connected.
send side extensions.conf:
[fax-tx-test]
exten=s,1,NoOp(Context fax-tx-test)
exten=s,n,SendFAX(${FaxFile}.tif)
exten=s,n,HangUp()
exten=h,1,NoOp(FAXSTATUS: ${FAXSTATUS} FAXERROR: ${FAXERROR}
FAXMODE:
${FAXMODE})
Channel:SIP/side-sip-fax
Context:fax-tx-test
Extension:s
Priority:1
Set:FaxFile=/var/spool/asterisk/fax/20091113_1455
receive side:
[incoming-fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)})
exten = s,n,ReceiveFAX(${FAXFILE}.tif)
exten = s,n,Hangup()
There's a bunch more stuff at
https://issues.asterisk.org/view.php?id=17105
But does anyone have a setup that Just Works? I'd love to
find a setup
that works for someone else and just copy it.
Thanks,
sean
Yes, I am familiar with Hylafax. But I'm trying to Keep It Simple, and
just use app_fax. Is it working for anyone? Does anybody have a simple
working example?
sean
It's the fax module built into 1.6.2.
sean
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Re: [asterisk-users] Asterisk Query
Hi Juan,
Thanks for your inputs, I tried with changes you suggested and find my
observation.
After adding context and extension able to make an outgoing call
[Digium-fxs to X-lite2000].
But not able to make incoming call [X-lite2000 to Digium-fxs]. Call
failed with,
(1) “*Call failed: 503 Service Unavailat *” error message on X-lite
(2) “CHANUNAVAIL” on asterisk CLI.
**CLI Saved useragent X-Lite release 1105d for peer 2000*
* == Using SIP RTP CoS mark 5*
*-- Executing [3...@my-phones:1] Dial(SIP/2000-, Zap/1/)
in new stack*
*[May 6 13:02:44] WARNING[20496]: channel.c:4003 ast_request: No channel
type registered for 'Zap'*
*[May 6 13:02:44] WARNING[20496]: app_dial.c:1745 dial_exec_full: Unable to
create channel of type 'Zap' (cause 66 - Channel not implemented)*
* == Everyone is busy/congested at this time (1:0/0/1)*
*-- Auto fallthrough, channel 'SIP/2000-' status is
'CHANUNAVAIL'*
Please find conf files below.
chan_dahdi.conf
[channels]
context=my-phones
usecallerid=yes
hidecallerid=no
immediate=no
signaling=fxo_ks
echocancel=yes
group=1
channel=1
sip.conf
==
[general]
port=5060
bindaddr=0.0.0.0
context=my-phones
[2000]
type=friend
context=my-phones
secret=1234
host=dynamic
extensions.conf
===
[my-phones]
exten = 2000,1,Dial(SIP/2000)
exten = ,1,Dial(Zap/1/)
system.conf
fxoks=1
loadzone=us
defaultzone=us
Please let me know any other configuration needs to be done.
On Fri, Apr 30, 2010 at 1:12 AM, Juan David Diaz juanch...@gmail.comwrote:
2010/4/29 garge rama garge.r...@gmail.com
Hi,
I am new to asterisk and trying to make calls with TDM400P asterisk digium
card.
I am using asterisk-1.6.2.4, dahdi-linux-complete-2.3.0+2.3.0 and
libpri-1.4.10.2 packages which are downloaded from asterisk website (
www.asterisk.org)
and able to compile successfully. TDM400P Digium card (having only one FXS
connected to J4) has installed successfully in PC.
I would like to make calls across SIP [x-lite] to analog phone connected
to TDM400P Digium card (fxs-j4).
For this the following four conf files are modified as shown below.
* chan_dahdi.conf*
*==*
[channels]
context=test
usecallerid=yes
hidecallerid=no
immediate=no
signaling=fxo_ks
echocancel=yes
group=1
channel=1
*extensions.conf***
*=*
[my-phones] ---*EXTEN does not exists for your
sip peer context*
exten = 2000,1,Dial(SIP/2000)
; Should look like:
*exten = ,1,Asterisk_Application(Action)* ;Dial(Zap/1/${Phone_Number_you
want})
[test]
exten = ,1,Dial(Zap/1)
exten = ,2,HangUp()
*sip.conf***
*===*
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
*context=**my-phones *
secret=1234
host=dynamic
*system.conf*
*==*
fxoks=1
loadzone = be
defaultzone = be
With those changes x-lite getting registered with asterisk and analog
device/phone is getting ring tone with off-hook and also getting debug
prints on cli, but not able to make calls.
Test Setup:
X-lite [configured as 2000, password… other info] running on asterisk PC
à registered with asterisk.
Analog phone connected to TDM400P Digium card - FXS-J4 running on same
asterisk PC à getting ring tone
Test Result:
=
Tried by calling from x-lite à getting message on CLI “call from
‘2000’ to ‘’ rejected because extension not found”
Tried by calling 2000 from analog phone [Digium-FXS-J4] - getting some
engage/disconnected tone while pressing digts [2000] on phone itself.
Welcome for your valuable suggestions and comments. Thank You in advance.
Regards,
Garge.
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Linux User #441131
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Re: [asterisk-users] problem with ringinuse=no, queue members receive randomly two calls
i get may debug messages like this: DEBUG[30684] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=-1) Is because dahdi is not installed? Can this be a possible cause of this behaviour? On Tue, May 4, 2010 at 9:54 PM, nik600 nik...@gmail.com wrote: Dear all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the user goes in a status Ring+Inuse. Any idea? Why the call is still dispatched to the user if it is not in the Not in use status? Thanks to all in advance * * LOG (core debug and verbose set to 5) * * #grep PL1038 full [May 4 16:21:08] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '6' (Ringing) [May 4 16:21:08] DEBUG[3035] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:08] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:08] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:08] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 6 (Ringing) [May 4 16:21:08] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '6' (Ringing) [May 4 16:21:08] VERBOSE[30453] logger.c: -- SIP/PL1038-5f7d is ringing [May 4 16:21:08] DEBUG[3035] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:08] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:08] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:08] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 6 (Ringing) [May 4 16:21:08] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '6' (Ringing) [May 4 16:21:08] VERBOSE[30268] logger.c: -- SIP/PL1038-5f7e is ringing [May 4 16:21:10] DEBUG[3035] chan_sip.c: T38 state changed to 0 on channel SIP/PL1038-5f7e [May 4 16:21:10] DEBUG[3035] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:10] DEBUG[3035] chan_sip.c: build_route: Contact hop: sip:pl1...@10.192.37.119 [May 4 16:21:10] DEBUG[30268] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:10] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:10] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:10] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 7 (Ring+Inuse) [May 4 16:21:10] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '7' (Ring+Inuse) [May 4 16:21:10] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:10] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:10] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 7 (Ring+Inuse) [May 4 16:21:10] VERBOSE[30268] logger.c: -- SIP/PL1038-5f7e answered SIP/192.168.55.32-5f59 [May 4 16:21:10] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '7' (Ring+Inuse) [May 4 16:21:14] VERBOSE[30268] logger.c: -- Native bridging SIP/192.168.55.32-5f59 and SIP/PL1038-5f7e [May 4 16:21:14] DEBUG[3035] chan_sip.c: T38 state changed to 0 on channel SIP/PL1038-5f7e [May 4 16:21:14] DEBUG[3035] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:14] DEBUG[3035] chan_sip.c: T38 state changed to 0 on channel SIP/PL1038-5f7e [May 4 16:21:14] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:14] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:14] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 7 (Ring+Inuse) [May 4 16:21:14] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '7' (Ring+Inuse) [May 4 16:21:15] DEBUG[29938] app_queue.c: Trying 'SIP/PL1038' with metric 0 [May 4 16:21:15] DEBUG[29938] app_queue.c: SIP/PL1038 in use, can't receive call [May 4 16:21:16] DEBUG[30097] app_queue.c: Trying 'SIP/PL1038' with metric 0 [May 4 16:21:16] DEBUG[30097] app_queue.c: SIP/PL1038 in use, can't receive call [ * * config * * sip users: [PL1039] context=mycontext callerid=PhoneLine1039 1039 secret=pwd1039 type=peer host=dynamic call-limit=3
Re: [asterisk-users] CDR to MS-SQL via ODBC issue
Tilghman Lesher wrote:
Okay, second idea is that you should very carefully examine your CDR table
layout and ensure that the columns that you have match EXACTLY what the
module expects you to have. If Asterisk expects you to have a column that you
don't (or the column type is wrong), that is another reason that the prepare
might fail. You might consider using the cdr_adaptive_odbc driver, instead,
as it is designed to create the insert based upon the structure of the table.
Ya, after thinking a bit more I had the same thought. Here are the table
formats
I found in cdr_odbc.c.
if (loguniqueid) {
snprintf(sqlcmd,sizeof(sqlcmd),INSERT INTO %s
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,
lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield)
VALUES (?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?), table);
} else {
snprintf(sqlcmd,sizeof(sqlcmd),INSERT INTO %s
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,
duration,billsec,disposition,amaflags,accountcode)
VALUES (?,?,?,?,?,?,?,?,?,?,?,?,?,?), table);
}
So if in cdr_odbc.conf you have:
loguniqueid=yes
Looks like the system expects the first table layout, otherwise it expects to
use the 2nd table layout.
Based on the table layout you emailed me previously, the fields missing are
'calldate' and 'userfield'.
Thanks!
Leif.
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[asterisk-users] Make the call finish after executing Dial(G())
Dear List,
My Dial command:
exten = _X.,n,Dial(SIP/PBX2/1234,60,G(connect-jack^${EXTEN}^1))
exten = h,1,
[connect-jack]
exten = _X.,1,NoOp(${CHANNEL}) ; Leg A
exten = _X.,2,NoOp(${CHANNEL}) ; Leg B
The problem is: after answering, [connect-jack] both priorities are
executed, and right after executing them call drops.
Log:
-- Executing [123...@npdb2:76] Dial(SIP/1001-0004,
SIP/PBX2/000123456,60,G(connect-jack^123456^1)) in new stack
== Using SIP RTP CoS mark 5
-- Called PBX2/000123456
== Begin MixMonitor Recording SIP/1001-0004
-- SIP/PBX2-0005 is ringing
-- SIP/PBX2-0005 answered SIP/1001-0004
-- Executing [123...@connect-jack:1] NoOp(SIP/1001-0004,
SIP/1001-0004) in new stack
-- Executing [123...@connect-jack:2] NoOp(SIP/1001-0004,
SIP/1001-0004) in new stack
-- Auto fallthrough, channel 'SIP/1001-0004' status is 'ANSWER'
-- Executing [123...@connect-jack:2] NoOp(SIP/PBX2-0005,
SIP/PBX2-0005) in new stack
-- Auto fallthrough, channel 'SIP/PBX2-0005' status is 'UNKNOWN'
== End MixMonitor Recording SIP/1001-0004
The question: how to execute G() right after answering (purpose behind
this: I will need to set some leg B variables), and then continue the
conversation?
Thank you,
Motiejus
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[asterisk-users] problem with trustrpid
Hi everyone, I am trying to figure out the behavior of trustrpid Basically its not behaving the way I expected it to or maybe I am missing a configuration option or something else. When a call from a phone is sent to the * box it has the following sip headers: From: From Phone sip:1...@10.0.0.29;tag=4bf4bb4e11e92476. Remote-Party-ID: Cloutier sip:5147714...@10.0.0.29;privacy=off;screen=no;party=calling;id-type=subscriber;screen=yes. And when the second leg of the call from the * box to our voip provider is setup the call has the following sip headers: From: From Phone sip:5147714...@10.0.0.24;tag=as73d69a8f. Remote-Party-ID: From Phone sip:5147714...@10.0.0.24;privacy=off;screen=no. Shouldnt the Remote-Party-ID stay what I origionally set it to? in sip.conf for the phone I have set trustrpid=yes and for our provider I have sendrpid=yes I am using asterisk 1.6.0.10 Thanks Alot Jesse Cloutier -- _ -- 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: NAT in SPA922
Ok..So what ip phone model do NAT? Sebastian On Wed, May 5, 2010 at 12:26 PM, Luki lugos...@gmail.com wrote: However, when I connect a PC to that port, SPA922 works as bridge. Exactly. The SPA9x2 has a 2-port switch; no NAT, no routing (unlike the SPA2102, etc). I think the 5.1 series is the latest firmware for the 922; the the 942, there is 6.1.5a. Luki -- _ -- 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 -- _ -- 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] Make the call finish after executing Dial(G())
Hi!
[connect-jack]
exten = _X.,1,NoOp(${CHANNEL}) ; Leg A
exten = _X.,2,NoOp(${CHANNEL}) ; Leg B
The problem is: after answering, [connect-jack] both priorities are
executed, and right after executing them call drops.
The call legs drop because you do not do anything with them, since your
dialplan is finished at this point. You could put them into a MeetMe
conference, for example, or use Transfer() or ChannelRedirect().
[connect-jack]
exten = _X.,1,Goto(bridging,meet,1); Leg A
exten = _X.,2,NoOp(${CHANNEL}) ; Leg B
exten = _X.,n,Goto(bridging,meet,1); Leg B
[bridging]
exten = meet,1,MeetMe(1234)
exten = meet,n,Hangup ; handle a hangup after meetme cleanly
Another way to address this: Use Dial option M instead of G
Philipp
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[asterisk-users] REALTIME in 1.2
I am trying to change a 1.6 realtime statement into a 1.2 realtime
statement and I know much has changed. I wish I could just upgrade, but
alas not right now.
exten =x,n,Set(NULL1=${REALTIME(schedules,id,${SCHEDULE})})
comes back with
pbx.c:1371 ast_func_read: Function REALTIME not registered
I am not stuck with realtime, I just have a mysql database with info
that changes and needs to update the dialplan accordingly.
Jason
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Re: [asterisk-users] Make the call finish after executing Dial(G())
Hi,
Great! I thought I won't see leg B channel while using M(), but I do!
:) M() did my day.
Thanks.
On Thu, May 6, 2010 at 4:29 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
[connect-jack]
exten = _X.,1,NoOp(${CHANNEL}) ; Leg A
exten = _X.,2,NoOp(${CHANNEL}) ; Leg B
The problem is: after answering, [connect-jack] both priorities are
executed, and right after executing them call drops.
The call legs drop because you do not do anything with them, since your
dialplan is finished at this point. You could put them into a MeetMe
conference, for example, or use Transfer() or ChannelRedirect().
[connect-jack]
exten = _X.,1,Goto(bridging,meet,1); Leg A
exten = _X.,2,NoOp(${CHANNEL}) ; Leg B
exten = _X.,n,Goto(bridging,meet,1); Leg B
[bridging]
exten = meet,1,MeetMe(1234)
exten = meet,n,Hangup ; handle a hangup after meetme cleanly
Another way to address this: Use Dial option M instead of G
Philipp
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Re: [asterisk-users] OT: NAT in SPA922
On 6 May 2010, at 14:16, Sebastian Milioto wrote: Ok..So what ip phone model do NAT? I think you'd struggle to find one. If it's a requirement you're probably doing something wrong... S -- _ -- 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 Query
Hi Garge -
exten =
,1,Asterisk_Application(Action) ;Dial(Zap/1/${Phone_Number_you want})
Two things:
1. There is no such thing as Zap anymore. Zap has been renamed to
Dahdi because of a trademark issue. So your extension should look
like:
exten = ,Dial(Dahdi/1/)
2. Do you really mean to dial ''? This number should be a valid
phone number.
- Noah
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Re: [asterisk-users] VoIP Termination in Japan
On 5/5/10, Adrian Marsh adrian.ma...@ubiquisys.com wrote: Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -- Sent from my mobile device -- _ -- 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: NAT in SPA922
Ok..So what ip phone model do NAT? I think you'd struggle to find one. If it's a requirement you're probably doing something wrong... Definitely get a router. Plug the IP phone into the router, and then you can plug the computer into the phone or the router. - Noah -- _ -- 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] AMR codec for Asterisk 1.6.1.X
Dear list, i have re-compiled again the source code of amr patch for 1.6 (https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/asterisk-1.6-AMR.patch) The patch does not compile with the static function into frame.c called : static int amr_samples(unsigned char *data, int datalen) i have removed the static and used like int amr_samples(unsigned char *data, int datalen) Anyone else got this issue ??? In this way the patch compile . It also show right format name when i try lo load codec_amr.so load codec_amr.so The 'load' command is deprecated and will be removed in a future release. Please use 'module load' instead. == Parsing '/etc/asterisk/codecs.conf': == Found -- codec_amr: parsing codecs.conf -- codec_amr: set octed-aligned mode to 1 -- codec_amr: set dtx mode to 0 -- codec_amr: AMR mode set to MR122 (7) codec_amr: enc_mode = 7, dtx = 0 == Registered translator 'amrtolin' from format amr to slin, cost 2000 == Registered translator 'lintoamr' from format slin to amr, cost 17997 Loaded codec_amr.so = (AMR Coder/Decoder) Also i have into the config file asterisk.conf the following value to filed transcode_via_sln = yes so transcode_via_sln = yes If i try to make a call to echotest by dialing 600 '600' = 1. Answer() [pbx_config] 2. Playback(demo-echotest) [pbx_config] 3. Echo() [pbx_config] 4. Playback(demo-echodone) [pbx_config] with a client that have only enabled amr codec i got this output: [May 6 17:51:11] WARNING[9684]: chan_sip.c:7654 process_sdp: Unsupported SDP media type in offer: audio 4002 RTP/SAVP 114 18 113 0 8 101 Anyone know how to get this AMR codec doing transcoding on asterisk 1.6? Many thanks in advantage Andrea Il 05/05/2010 18:13, Adrian Marsh ha scritto: It says in the readme from that link you provided: This patch adds AMR-NB support to Asterisk 1.4 (for Asterisk 1.6 check out asterisk 1.6 branch and use the asterisk-1.6-AMR.patch patch (provided by Ivelin Ivanov)) Did you use the 1.6 branch and patch ?? I'll have to try this myself at some point. Thanks, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrea Cristofanini Sent: 05 May 2010 14:22 To: asterisk-users@lists.digium.com Subject: [asterisk-users] AMR codec for Asterisk 1.6.1.X Hi list, Anyone have successfully compiled amr codec for asterisk 1.6.1.X ? I still have no problem compiling and playing with it on Asterisk 1.4.X. I have used the following patch : https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/ Hare is what i get while loading codec_amr.so debbi*CLI load codec_amr.so == Parsing '/etc/asterisk/codecs.conf': == Found -- codec_amr: parsing codecs.conf -- codec_amr: set octed-aligned mode to 1 -- codec_amr: set dtx mode to 0 -- codec_amr: AMR mode set to MR122 (7) codec_amr: enc_mode = 7, dtx = 0 == Registered translator 'amrtolin' from format unknown to slin, cost 4000 == Registered translator 'lintoamr' from format slin to unknown, cost 32002 Loaded codec_amr.so = (AMR Coder/Decoder) debbi*CLI core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - -- - - - - - - - - - gsm - - 2 22 2 1 4001 12002 - - 2 2 4003 ulaw - 12002 - 12 2 1 4001 12002 - - 2 2 4003 alaw - 12002 1 -2 2 1 4001 12002 - - 2 2 4003 g726aal2 - 12002 2 2- 2 1 4001 12002 - - 2 2 4003 adpcm - 12002 2 22 - 1 4001 12002 - - 2 2 4003 slin - 12001 1 11 1 - 4000 12001 - - 1 1 4002 lpc10 - 16001 4001 4001 4001 4001 4000 - 16001 - - 4001 4001 8002 g729 - 16001 4001 4001 4001 4001 4000 8000 - - - 4001 4001 8002 speex - - - -- - - - - - - - - - ilbc - - - -- - - - - - - - - - g726 - 16001 4001 4001 4001 4001 4000 8000 16001 - - - 4001 8002 g722 - 20001 8001 8001 8001 8001 8000 12000 20001 - - 8001 - 4001 slin16 - 24001 12001 1200112001 12001 12000 16000 24001 - - 12001 4000 - debbi*CLI core show file formats
Re: [asterisk-users] VoIP Termination in Japan
On 5/5/10, Adrian Marsh adrian.ma...@ubiquisys.com wrote: Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -- Sent from my mobile device -- _ -- 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: NAT in SPA922
It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Sebastian On Thu, May 6, 2010 at 12:36 PM, Noah Miller noahisaacmil...@gmail.comwrote: Ok..So what ip phone model do NAT? I think you'd struggle to find one. If it's a requirement you're probably doing something wrong... Definitely get a router. Plug the IP phone into the router, and then you can plug the computer into the phone or the router. - Noah -- _ -- 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 -- _ -- 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] IAX2 Auto-congesting call due to slow response
Hi all, I have been testing several asterisk versions and I found out that all the previus version of asterisk worked fine. After 1.4.22 it cease to work. In the change log referring to iax from 1.4.22 to 1.4.23 I found this: / 2009-01-06 20:48 + [r167260] Tilghman Lesher tles...@digium.com * /, channels/chan_iax2.c: Merged revisions 167259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06 Jan 2009) | 2 lines Security fix AST-2009-001. 2008-12-03 17:55 + [r160480-160570] Tilghman Lesher tles...@digium.com * channels/chan_iax2.c: During bridge code, the channel bridge may return a retry code, if a transfer was initiated but not yet completed. If the bridge is immediately retried, then we may send a storm of TXREQ packets, even though the first set is sent reliably (retransmitted). Fixes AST-137. 2008-12-01 17:27 + [r160003] Russell Bryant russ...@digium.com * channels/chan_iax2.c: Apply some logic used in iax2_indicate() to iax2_setoption(), as well, since they both have the potential to send control frames in the middle of call setup. We have to wait until we have received a message back from the remote end before we try to send any more frames. Otherwise, the remote end will consider it invalid, and we'll get stuck in an INVAL/VNAK storm. 2008-11-25 21:56 + [r159246-159269] Tilghman Lesher tles...@digium.com * channels/chan_iax2.c: Don't try to send a response on a NULL pvt. (closes issue #13919) Reported by: barthpbx Patches: chan_iax2.c.patch uploaded by eliel (license 64) Tested by: barthpbx * /, channels/chan_iax2.c: Merged revisions 159245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008) | 7 lines Regression fix for last security fix. Set the iseqno correctly. (closes issue #13918) Reported by: ffloimair Patches: 20081119__bug13918.diff.txt uploaded by Corydon76 (license 14) Tested by: ffloimair iax2.c: Revert revision 132506, since it occasionally caused IAX2 HANGUP packets not to be sent, and instead, schedule a task to destroy the iax2 pvt structure 10 seconds later. This allows the IAX2 HANGUP packet to be queued, transmitted, and ACKed before the pvt is destroyed. (closes issue #13645) Reported by: dzajro Patches: 2008__bug13645__3.diff.txt uploaded by Corydon76 (license 14) Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/ 2008-11-04 20:49 + [r154365] Tilghman Lesher tles...@digium.com * channels/chan_iax2.c: On busy systems, it's possible for the values checked within a single line of code to change, unless the structure is locked to ensure a consistent state. (closes issue #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma Reported by: jaroth Patch by: me (modified jaroth's patch) * main/rtp.c: Remove the potential for a division by zero error. (Closes issue #13810) / can this happen because of the sequence number?? Thanks in advance, Alex 2010/5/5 Alexandre Rodrigues alex...@gmail.com Hi all, I am trying to connect to a softphone application using an Iax channel on Asterisk 1.4.30. I can do outbound calls, from softphone to asterisk, but not inbound from asterisk to softphone. I get the following Debug: -- -- Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00018ms SCall: 04825 DCall: 0 [10.20.0.201:41764] VERSION : 2 CALLED NUMBER : s CODEC_PREFS : (ulaw) CALLING NUMBER : 2000 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: athens_user LANGUAGE: en USERNAME: wtgpl FORMAT : 4 CAPABILITY : 4 ADSICPE : 2 DATE TIME : 2010-05-04 18:48:48 Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00018ms SCall: 04825 DCall: 0 [10.20.0.201:41764] VERSION : 2 CALLED NUMBER : s CODEC_PREFS : (ulaw)
Re: [asterisk-users] OT: NAT in SPA922
On Thu, 6 May 2010, Sebastian Milioto wrote: It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Sebastian Put each PC in its own VLAN. Keep all the phones in one VLAN. Although having a $30 router in each room hanging off the phone would accomplish what you want also. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is billsec in CDR?
Philipp von Klitzing wrote:
Hi!
apps like playback do an implicit answer and this fires up the billsec
counter.
OK, here is my dialplan:
exten = _011X.,n,Playback(this-call-will-end-in)
exten =
_011X.,n,Dial(SIP/${ext...@${ldtrunk1},60,L(${ms}:3))
Is there any way that Asterisk will record the correct billsec? Or, is
there a different approach?
Place a ResetCDR() after your Playback() statement and before Dial().
Philipp
ResetCDR() works!
Thank you very much!
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jian@sjgeophysics.com mailto:jian@sjgeophysics.com
Tel: (604)582-1100
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Re: [asterisk-users] Asterisk 1.6.2.7 Now Available
On 05/04/2010 07:41 PM, Leif Madsen wrote: OK, I got sufficiently curious to make sure Skype for Asterisk still loaded on 1.6.2.7. It does for me, but I had to run make install in my Skype source directory. One of the modules loaded, but the 'skype' CLI command was not available until after I ran make install again, so one of the Skype for Asterisk components must not have been compatible. Since Skype For Asterisk includes source code components, just like any other add-on module that is distributed as source code it must always (or nearly always) be recompiled when the version of Asterisk installed on the system is changed, or even when Asterisk is recompiled and significant compile-time options have been changed (like DEBUG_THREADS, MALLOC_DEBUG and the like). Digium's binary-only modules are built in such a way to avoid this requirement, but that is not possible for a channel driver like chan_skype, so it must be distributed as source code and compiled against the configured and installed copy of Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: NAT in SPA922
It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Put each PC in its own VLAN. Keep all the phones in one VLAN. Although having a $30 router in each room hanging off the phone would accomplish what you want also. Take j's suggestion to use VLANs. This is not a good situation for NAT. Cisco 2950's can do VLANs. - Noah -- _ -- 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: NAT in SPA922
I see the following in SPA922 System tab (new firmware) VLAN Settings Enable VLAN:yesnoEnable CDP:yesno VLAN ID:PC Port VLAN Highest Priority:01234567No Limit Enable PC Port VLAN Tagging:yesnoPC Port VLAN ID: VLAN ID:1 for all Phones, and VLAN 2, 3, 4, 5..,24 for each PC. This should work, right? Sebastian On Thu, May 6, 2010 at 2:25 PM, Jeff LaCoursiere j...@jeff.net wrote: On Thu, 6 May 2010, Sebastian Milioto wrote: It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Sebastian Put each PC in its own VLAN. Keep all the phones in one VLAN. Although having a $30 router in each room hanging off the phone would accomplish what you want also. j -- _ -- 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 -- _ -- 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: NAT in SPA922
-Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Noah Miller Sent: Thu 5/6/2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: NAT in SPA922 It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Put each PC in its own VLAN. Keep all the phones in one VLAN. Although having a $30 router in each room hanging off the phone would accomplish what you want also. Take j's suggestion to use VLANs. This is not a good situation for NAT. Cisco 2950's can do VLANs. to be clear, the only way this will work with the PCs is if each PC vlan is *also* a unique ip subnet (else how do all the vlans access a common default gw?) place the phones in a voice vlan, and the phone problem is solved. as for the PC isolation, you might get better feedback on a cisco or other networking forum. -david -- _ -- 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: NAT in SPA922
On Thu, 6 May 2010, Sebastian Milioto wrote: I see the following in SPA922 System tab (new firmware) VLAN Settings Enable VLAN:yesnoEnable CDP:yesno VLAN ID:PC Port VLAN Highest Priority:01234567No Limit Enable PC Port VLAN Tagging:yesnoPC Port VLAN ID: VLAN ID:1 for all Phones, and VLAN 2, 3, 4, 5..,24 for each PC. This should work, right? Sebastian Then you will have to do some work on the gateway and layout all your IP ranges. One for the phones and presumably your asterisk server, then one range for each PC. Your gateway will end up with 25 networks. j -- _ -- 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] long return times from System() calls with 1.6.2.6?
In case anybody was following this thread, wanted to let people know that the fix made it into SVN, and is packaged into 1.6.2.8-rc1 Huge thanks to Kevin and Tilghman On Wed, Apr 21, 2010 at 3:40 PM, David Backeberg dbackeb...@gmail.com wrote: issue opened. https://issues.asterisk.org/view.php?id=17223 -- _ -- 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: NAT in SPA922
Hi! Should I take another approach on that? Put each PC in its own VLAN. Keep all the phones in one VLAN. Note: VLANs are an organisational tool, and do not really add security. If you want to go with VLANs in thise case then rather consider port based VLAN (configured in the switch only) instead of the typical tagged VLAN (802.1Q). There are various midly cheap smart switches around that support this type of poor-man's VLAN. Note: You will then have to have PC and IP phone on the same static VLAN. Philipp -- _ -- 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] Questions About Fax for Asterisk
Yes, I purchased licenses for Fax for Asterisk and yes I called tech support and had the WORST experience I have ever had with any technical support call. I am running Asterisk 1.6.2.6 and: FAX For Asterisk Components: Applications: 1.6.2.0_1.2.0 voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32) The guy was arrogant and absolutely a jerk and I don't like to call people names, but call it as I see it. This has not been my experience the five or six times I have had to call Digium over the years, but it has been many years since my last call so I have no idea what the general support staff is like. I could not get any questions answered by the tech that took hours to call me back to tell me to read the readme. That would be all well and good if I didn't pay money. He could not explain Digium's math as far as faxing and failed to offer to get back to me with any kind of answer. Maybe someone on the list can make sense of this Enron style of accounting: voipgw01*CLI fax show stats voipgw01*CLI FAX Statistics: --- Current Sessions : 1 Transmit Attempts: 0 Receive Attempts : 336 Completed FAXes : 320 Failed FAXes : 57 Digium G.711 Licensed Channels: 4 Max Concurrent : 1 Success : 0 Switched to T.38 : 0 Canceled : 0 No FAX : 1 Partial : 0 Negotiation Failed : 0 Train Failure: 3 Protocol Error : 0 IO Partial : 0 IO Fail : 0 voipgw01*CLI Digium T.38 Licensed Channels: 4 Max Concurrent : 4 Success : 175 Canceled : 0 No FAX : 6 Partial : 19 Negotiation Failed : 0 Train Failure: 83 Protocol Error : 33 IO Partial : 0 IO Fail : 0 Thanks, Steve Totaro -- _ -- 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] Possible bug in chan_sip:add_sdp
Am I missing something here? I see
if (needvideo) { /* only if video response is appropriate */
add_line(resp, m_video-str);
add_line(resp, a_video-str);
add_line(resp, hold); /* Repeat hold for the video stream */
} else if (p-offered_media[SDP_VIDEO].offered) {
snprintf(dummy_answer, sizeof(dummy_answer), m=video 0 RTP/AVP\
%s\r\n, p-offered_media[SDP_VIDEO].text);
add_line(resp, dummy_answer);
But len, which was used to set Content-Length, isn't updated to onclide
that dummy. Doesn't it need to be?
I think this may be a problem with a connection to my Polycom VSX.
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Re: [asterisk-users] Possible bug in chan_sip:add_sdp
I can confirm that the following fixes my problem: --- chan_sip.c (revision 261450) +++ chan_sip.c (working copy) @@ -10357,12 +10357,22 @@ strlen(connection) + strlen(session_time); if (needaudio) len += m_audio-used + a_audio-used + strlen(hold); + else if (p-offered_media[SDP_AUDIO].offered) + len += strlen(m=audio 0 RTP/AVP \r\n) + strlen(p-offered_media[SDP_AUDIO].text); + if (needvideo) /* only if video response is appropriate */ len += m_video-used + a_video-used + strlen(bandwidth) + strlen(hold); + else if (p-offered_media[SDP_VIDEO].offered) + len += strlen(m=video 0 RTP/AVP \r\n) + strlen(p-offered_media[SDP_VIDEO].text); + if (needtext) /* only if text response is appropriate */ len += m_text-used + a_text-used + strlen(hold); + else if (p-offered_media[SDP_TEXT].offered) + len += strlen(m=text 0 RTP/AVP \r\n) + strlen(p-offered_media[SDP_TEXT].text); if (add_t38) len += m_modem-used + a_modem-used; + else if (p-offered_media[SDP_IMAGE].offered) + len += strlen(m=image 0 udptl t38\r\n); add_header(resp, Content-Type, application/sdp); add_header_contentLength(resp, len); -- _ -- 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] CDR to MS-SQL via ODBC issue
On Wednesday 05 May 2010 18:29:26 Neeraj Chand wrote: --- Message: 10 Date: Wed, 5 May 2010 10:26:34 -0500 From: Tilghman Lesher tles...@digium.com Subject: Re: [asterisk-users] CDR to MS-SQL via ODBC issue To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 201005051026.34929.tles...@digium.com Content-Type: text/plain; charset=iso-8859-1 On Wednesday 05 May 2010 06:51:48 Neeraj Chand wrote: I can connect to the database and run via isql, and also use func_odbc, etc with res_odbc configured with the same database / freetds, but I cannot write CDRs. Are you writing to the database with func_odbc, or just reading? My gut says that you need to check your permissions on the database to ensure that you're allowed to write to the CDR table. Hi Tilghman, yeah I thought so too at first but then, using the same permissions I'm doing both read writes as well. On the database end, the user is setup as database_owner and has db_read db_write permissions. I got Leif to check this with me last night, we couldn't figure it out. The error that pops up is: cdr_odbc: Connected to asterisk-freetds-connector cdr_odbc: Error in PREPARE -1 cdr_odbc: Query FAILED Call not logged! __ Okay, second idea is that you should very carefully examine your CDR table layout and ensure that the columns that you have match EXACTLY what the module expects you to have. If Asterisk expects you to have a column that you don't (or the column type is wrong), that is another reason that the prepare might fail. You might consider using the cdr_adaptive_odbc driver, instead, as it is designed to create the insert based upon the structure of the table. -- Tilghman Lesher - That did the trick. Had calldate userfield missing I had loguniqueid=yes. Thanks Leif / Tilghman. :) -- _ -- 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 Fax With Flowroute SIP Provider
Does anybody have T.38 faxing working with Flowroute? I am running Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in sip.conf. When I receive a fax it tries to negotiate T.38 and Flowroute sends back a Bad Request response saying I have a SIP syntax error. Flowroute support is recommending that I try again after removing externip and localnet from sip.conf. They state that their service will recognize the private IP and rewrite the SIP packets. However this is going to cause issues for my remote SIP phones. Thanks, Ryan DEBUG[32389] app_fax.c: Negotiating T.38 for receive on SIP/flowroute- INVITE sip:+num...@xx.xx.xx.xx:5060 SIP/2.0 ... CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.7-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 293 v=0 o=root 2048302926 2048302927 IN IP4 xx.xx.xx.xx s=Asterisk PBX 1.6.2.7-rc3 c=IN IP4 xx.xx.xx.xx t=0 0 m=image 4575 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPFEC SIP/2.0 400 Bad Request ... CSeq: 102 INVITE Error-Info: sip:+num...@xx.xx.xx.xx;cause=[line 023] SIP syntax error Content-Length: 0 WARNING[32389] app_fax.c: Transmission error -- _ -- 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 Fax With Flowroute SIP Provider
On 05/06/2010 05:46 PM, Ryan Wagoner wrote: Does anybody have T.38 faxing working with Flowroute? I am running Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in sip.conf. When I receive a fax it tries to negotiate T.38 and Flowroute sends back a Bad Request response saying I have a SIP syntax error. Flowroute support is recommending that I try again after removing externip and localnet from sip.conf. They state that their service will recognize the private IP and rewrite the SIP packets. However this is going to cause issues for my remote SIP phones. Thanks, Ryan DEBUG[32389] app_fax.c: Negotiating T.38 for receive on SIP/flowroute- INVITE sip:+num...@xx.xx.xx.xx:5060 SIP/2.0 ... CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.7-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 293 v=0 o=root 2048302926 2048302927 IN IP4 xx.xx.xx.xx s=Asterisk PBX 1.6.2.7-rc3 c=IN IP4 xx.xx.xx.xx t=0 0 m=image 4575 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPFEC SIP/2.0 400 Bad Request ... CSeq: 102 INVITE Error-Info: sip:+num...@xx.xx.xx.xx;cause=[line 023] SIP syntax error Content-Length: 0 Which line is 'line 23' of the T.38 re-INVITE? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 Fax With Flowroute SIP Provider
On Thu, May 6, 2010 at 5:54 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 05/06/2010 05:46 PM, Ryan Wagoner wrote: Does anybody have T.38 faxing working with Flowroute? I am running Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in sip.conf. When I receive a fax it tries to negotiate T.38 and Flowroute sends back a Bad Request response saying I have a SIP syntax error. Flowroute support is recommending that I try again after removing externip and localnet from sip.conf. They state that their service will recognize the private IP and rewrite the SIP packets. However this is going to cause issues for my remote SIP phones. Last I checked with Flowroute, they weren't yet supporting T.38. Has this changed in the last month or so? -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- 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 Fax With Flowroute SIP Provider
I wasn't sure how the lines were counted. Here is the debug output from Asterisk where it is building the invite packet. I looked at the a=T38 lines and nothing is standing out to me. Ryan [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 0 [ 47]: INVITE sip:+num...@x.x.x.x:5060 SIP/2.0 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK2837f4cf;rport [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 2 [ 54]: Route: sip:x.x.x.x;lr,sip:x.x.x.x;lr [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 3 [ 16]: Max-Forwards: 70 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 4 [ 59]: From: sip:+num...@x.x.x.x:5060;tag=as7d21d6f3 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 5 [ 53]: To: sip:+num...@x.x.x.x:5060;tag=gK0d4c48f7 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 6 [ 39]: Contact: sip:num...@x.x.x.x [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 7 [ 39]: Call-ID: 302861516_123483...@x.x.x.x [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 8 [ 16]: CSeq: 102 INVITE [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 9 [ 36]: User-Agent: Asterisk PBX 1.6.2.7-rc3 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 12 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 14 [ 19]: Content-Length: 293 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 15 [ 0]: [May 6 13:29:05] DEBUG[32389] chan_sip.c:Body 0 [ 3]: v=0 [May 6 13:29:05] DEBUG[32389] chan_sip.c:Body 1 [ 48]: o=root 2048302926 2048302927 IN IP4 x.x.x.x [May 6 13:29:05] DEBUG[32389] chan_sip.c:Body 2 [ 26]: s=Asterisk PBX 1.6.2.7-rc3 [May 6 13:29:05] DEBUG[32389] chan_sip.c:Body 3 [ 21]: c=IN IP4 x.x.x.x [May 6 13:29:05] DEBUG[32389] chan_sip.c:Body 4 [ 5]: t=0 0 [May 6 13:29:05] DEBUG[32389] chan_sip.c:Body 5 [ 22]: m=image 4575 udptl t38 [May 6 13:29:05] DEBUG[32389] chan_sip.c:Body 6 [ 17]: a=T38FaxVersion:0 [May 6 13:29:05] DEBUG[32389] chan_sip.c:Body 7 [ 21]: a=T38MaxBitRate:14400 [May 6 13:29:05] DEBUG[32389] chan_sip.c:Body 8 [ 22]: a=T38FaxFillBitRemoval [May 6 13:29:05] DEBUG[32389] chan_sip.c:Body 9 [ 37]: a=T38FaxRateManagement:transferredTCF [May 6 13:29:05] DEBUG[32389] chan_sip.c:Body 10 [ 24]: a=T38FaxMaxDatagram:1400 [May 6 13:29:05] DEBUG[32389] chan_sip.c:Body 11 [ 23]: a=T38FaxUdpEC:t38UDPFEC On Thu, May 6, 2010 at 6:54 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/06/2010 05:46 PM, Ryan Wagoner wrote: Does anybody have T.38 faxing working with Flowroute? I am running Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in sip.conf. When I receive a fax it tries to negotiate T.38 and Flowroute sends back a Bad Request response saying I have a SIP syntax error. Flowroute support is recommending that I try again after removing externip and localnet from sip.conf. They state that their service will recognize the private IP and rewrite the SIP packets. However this is going to cause issues for my remote SIP phones. Thanks, Ryan DEBUG[32389] app_fax.c: Negotiating T.38 for receive on SIP/flowroute- INVITE sip:+num...@xx.xx.xx.xx:5060 SIP/2.0 ... CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.7-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 293 v=0 o=root 2048302926 2048302927 IN IP4 xx.xx.xx.xx s=Asterisk PBX 1.6.2.7-rc3 c=IN IP4 xx.xx.xx.xx t=0 0 m=image 4575 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPFEC SIP/2.0 400 Bad Request ... CSeq: 102 INVITE Error-Info: sip:+num...@xx.xx.xx.xx;cause=[line 023] SIP syntax error Content-Length: 0 Which line is 'line 23' of the T.38 re-INVITE? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --
[asterisk-users] Contact header gets url decoded?
I'm migrating an application running on a fairly old 1.4 (or 1.2?) version of Asterisk to some boxes running 1.6.0.27 The application takes an inbound INVITE like: mumble-fratz-sip%3afoo%40bar@asteriskbox.abc.com:5062 The older version of asterisk replies with a 200 OK and a Contact: header that looks like: Contact: sip:mumble-fratz-sip%3afoo%40bar@asteriskbox.abc.com:5062 Newer 1.6 Asterisk (I've tried 1.6.0.9 and 1.6.0.27) take the identical call and reply with a 200 OK and a Contact header of: Contact: sip:mumble-fratz-sip:f...@bar.com@asteriskbox.abc.com:5062 And the calling applications appear to not recognize this 200 OK and never send an ACK and Asterisk eventually throws in the towel on the call setup Is there a knob I can adjust this behavior? The original To: is never molested in the same way, just the Contact header. Thanks in advance, Tom -- _ -- 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] Channels In Use
Hi Luki, Thank you so much.. The soft xx worked perfectly. The rtptimeout is excellent also. Regards, S. On 5 May 2010 23:59, Luki lugos...@gmail.com wrote: Are there any CLI commands to free this up or any other ways without having to restart asterisk. Did you try soft hangup channel? Or set an RTP timeout to avoid abandoned channels? Luki -- _ -- 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 -- _ -- 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 Fax With Flowroute SIP Provider
On Thu, May 6, 2010 at 7:11 PM, Warren Selby wcse...@selbytech.com wrote: On Thu, May 6, 2010 at 5:54 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/06/2010 05:46 PM, Ryan Wagoner wrote: Does anybody have T.38 faxing working with Flowroute? I am running Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in sip.conf. When I receive a fax it tries to negotiate T.38 and Flowroute sends back a Bad Request response saying I have a SIP syntax error. Flowroute support is recommending that I try again after removing externip and localnet from sip.conf. They state that their service will recognize the private IP and rewrite the SIP packets. However this is going to cause issues for my remote SIP phones. Last I checked with Flowroute, they weren't yet supporting T.38. Has this changed in the last month or so? -- Thanks, --Warren Selby http://www.selbytech.com I found some websites mentioning they supported it. Plus when I receive a fax with t38 turned off I get the following in the log WARNING[27824] chan_sip.c: Unsupported SDP media type in offer: image 19738 udptl t38 Ryan -- _ -- 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] problem with trustrpid
I am trying to figure out the behavior of trustrpid Basically its not behaving the way I expected it to or maybe I am missing a configuration option or something else. When a call from a phone is sent to the * box it has the following sip headers: From: From Phone sip:1001 at 10.0.0.29;tag=4bf4bb4e11e92476. Remote-Party-ID: Cloutier sip:5147714203 at 10.0.0.29;privacy=off;screen=no;party=calling;id-type=subscriber;screen=yes. And when the second leg of the call from the * box to our voip provider is setup the call has the following sip headers: From: From Phone sip:5147714203 at 10.0.0.24;tag=as73d69a8f. Remote-Party-ID: From Phone sip:5147714203 at 10.0.0.24;privacy=off;screen=no. Shouldnt the Remote-Party-ID stay what I origionally set it to? in sip.conf for the phone I have set trustrpid=yes and for our provider I have sendrpid=yes I am using asterisk 1.6.0.10 Thanks Alot Jesse Cloutier I had a similar situation in which playing with trustrpid did not seem to have any effect. The solution was to use the I option to Dial. (that is an upper case i). In the end I could not say whether or not trustpid is broken because it is not entirely clear what it is supposed to do. -crjw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with trustrpid
On Fri, May 07, 2010 at 02:20:24AM +, crjw wrote:
| I am trying to figure out the behavior of trustrpid
|
| Basically its not behaving the way I expected it to or maybe I am
| missing a configuration option or something else.
|
| I had a similar situation in which playing with trustrpid did not seem to
have any effect.
| The solution was to use the I option to Dial. (that is an upper case i).
| In the end I could not say whether or not trustpid is broken because it is
not entirely clear what it is supposed to do.
| -crjw
I think in the end I patched asterisk directly for this. This is
unlikely to apply cleanly but here you go, you get the general idea:
Index: chan_sip.c
===
--- chan_sip.c (revision 182593)
+++ chan_sip.c (working copy)
@@ -9721,6 +9722,8 @@
/* replace callerid if rpid found, and not restricted */
if (!ast_strlen_zero(rpid_num) ast_test_flag(p-flags[0],
SIP_TRUSTRPID)) {
char *tmp;
+ memset(calleridname, 0, sizeof(calleridname));
+ get_calleridname(rpid, calleridname,
sizeof(calleridname));
if (*calleridname)
ast_string_field_set(p, cid_name, calleridname);
tmp = ast_strdupa(rpid_num);
Hope it helps!
-d
--
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Re: [asterisk-users] OT: NAT in SPA922
I think this is a motel kind of situation and a PVLAN serves the situation right. Put all the ipphones in the voice vlan as suggested, make a seperate isolated vlan for the PCs, this will restrict traffic between the clients. Rgds, Vineet Bhojnagarwala RCDD, NTS, OSP Spear Networks Pvt Ltd Integration Consultancy +91-9831436607 On May 6, 2010, at 11:30 PM, David White david.wh...@watchguard.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Noah Miller Sent: Thu 5/6/2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: NAT in SPA922 It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Put each PC in its own VLAN. Keep all the phones in one VLAN. Although having a $30 router in each room hanging off the phone would accomplish what you want also. Take j's suggestion to use VLANs. This is not a good situation for NAT. Cisco 2950's can do VLANs. to be clear, the only way this will work with the PCs is if each PC vlan is *also* a unique ip subnet (else how do all the vlans access a common default gw?) place the phones in a voice vlan, and the phone problem is solved. as for the PC isolation, you might get better feedback on a cisco or other networking forum. -david -- _ -- 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 -- _ -- 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: NAT in SPA922
Alternatively, if using normal vlans, this can also be achieved by enabling access list on the switch and restrict traffic flows. Generally this is done on a layer 3 switch, don't think it will support on your switch model. Rgds, Vineet Bhojnagarwala RCDD, NTS, OSP Spear Networks Pvt Ltd Integration Consultancy +91-9831436607 On May 7, 2010, at 8:39 AM, Vineet Bhojnagarwala vbho...@gmail.com wrote: I think this is a motel kind of situation and a PVLAN serves the situation right. Put all the ipphones in the voice vlan as suggested, make a seperate isolated vlan for the PCs, this will restrict traffic between the clients. Rgds, Vineet Bhojnagarwala RCDD, NTS, OSP Spear Networks Pvt Ltd Integration Consultancy +91-9831436607 On May 6, 2010, at 11:30 PM, David White david.wh...@watchguard.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Noah Miller Sent: Thu 5/6/2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: NAT in SPA922 It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Put each PC in its own VLAN. Keep all the phones in one VLAN. Although having a $30 router in each room hanging off the phone would accomplish what you want also. Take j's suggestion to use VLANs. This is not a good situation for NAT. Cisco 2950's can do VLANs. to be clear, the only way this will work with the PCs is if each PC vlan is *also* a unique ip subnet (else how do all the vlans access a common default gw?) place the phones in a voice vlan, and the phone problem is solved. as for the PC isolation, you might get better feedback on a cisco or other networking forum. -david -- _ -- 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 -- _ -- 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] Problem of Playing 'pbx-transfer'
Dear all, We have been using asterisk for 4 years. Now we have got problems which occurs during the attended transfer. During attended transfer, sometimes we cannot hear the sound of 'pbx-transfer'. I cannot understand why this is happening? log is : -- Started music on hold, class 'default', on SIP/113.34.235.13-b7a3f110 -- SIP/185148-092db338 Playing 'pbx-transfer' (language 'jp') Although it is showing Playing 'pbx-transfer' (language 'jp'), but it cannot hear 'pbx-transfer' sound Sometimes we can hear the sound of 'pbx-transfer'. is it the problem of network load or phone-set or something else? Please let me know. I am using x-lite and snom 300. Before i tested it for memory load, And found out that it is not a memory problem. Our system is as like as: The number of User agent is: 1650 The number of Actual registered user agent is: 600 Our System configuration is : IBM X3550 CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz HDD: 3.5 SATA 1TB x 2 version of asterisk: 1.4.23.1 our memory size is 4GB. concurrent calls no : 30. Our memory condition is below : Cpu(s): 0.3%us, 0.7%sy, 0.0%ni, 98.5%id, 0.0%wa, 0.1%hi, 0.3%si, 0.0%st Mem: 4147888k total, 3986540k used, 161348k free,76852k buffers Swap: 2031608k total, 56k used, 2031552k free, 3170396k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 23160 root 15 0 440m 415m 5688 S 4.3 10.3 398:13.93 asterisk Our disk space condition is below: FilesystemSize Used Avail Use% Mounted on /dev/mapper/VolGroup00-LogVol00 901G 245G 610G 29% / /dev/sda1 99M 18M 77M 19% /boot tmpfs 2.0G 0 2.0G 0% /dev/shm Asterisk and the User-Agent is connected through the Internet. ..And Is there any solution to solve this problem? I have investigated in several places but I cannot find out the reason? I need this solution very urgently. Is there any one who can solve this problem? -- _ -- 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] Video in Skype for Asterisk
Is there anything special that has to be done to make video calls work? It doesn't seem to work for me (no video). What CODECS are supported? -- _ -- 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
