[asterisk-users] AEL problem: bug or feature?
Hi! I have a problem with jump in AEL:
_+43123456789! = jump +22;
+22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
_22 = { NoOp(); }
- OK
_+43123456789! = jump +22;
_+22 = { NoOp(); }
-- AEL compile error:
LOG: lev:4 file:pbx_ael.c line:1234 func: check_goto Error: file
./ofis/extensions.ael_trunking, line 525-525: goto: no label +22|1
exists in the current context, or any of its inclusions!
Is this is some special feature/limitation or just a bug?
thanks
Klaus
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Re: [asterisk-users] Problems using chan_sebi and Huawei E169G
Martin Stubbs wrote: Hi, If I connect to the USB modem with minicom and issue the ATDxxx; command with a semicolon at the end to signify a voice call I get the same error response. Could someone else with this type of USB modem tell me if that command should work in minicom? I had exactly this problem with a 3UK E169, in the end after trying a million and one things (including crossflashing to various different providers), I replaced it with an unlocked Vodafone UK one. (apparently vodafone ES ones work fine as well, I also have what I think is a german vodafone one, that works). Thanks for the patch, I also have a little addition that allows the dongle to roam. (very simple change but essential on the network I am using it on). Do you know if there's anywhere set up that people can collaborate on this? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GXW4024 experience
Hi, In this http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/204725/focus=221375dating from 2008, experiences with Grandstream GXW4024 were asked. Has anyone something up-to-date to share about this ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
When I call in , there is nothing, because I did not set any inbound routes, I am using the box only to forward calls to PSTN, so when I use softphone to call outside here is the log: -- Executing [966505103...@from-internal:1] Macro(SIP/100-08fba098, user-callerid|SKIPTTL|) in new stack -- Executing [...@macro-user-callerid:1] Set(SIP/100-08fba098, AMPUSER=100) in new stack -- Executing [...@macro-user-callerid:2] GotoIf(SIP/100-08fba098, 0?report) in new stack -- Executing [...@macro-user-callerid:3] ExecIf(SIP/100-08fba098, 1|Set|REALCALLERIDNUM=100) in new stack -- Executing [...@macro-user-callerid:4] Set(SIP/100-08fba098, AMPUSER=100) in new stack -- Executing [...@macro-user-callerid:5] Set(SIP/100-08fba098, AMPUSERCIDNAME=100) in new stack -- Executing [...@macro-user-callerid:6] GotoIf(SIP/100-08fba098, 0?report) in new stack -- Executing [...@macro-user-callerid:7] Set(SIP/100-08fba098, AMPUSERCID=100) in new stack -- Executing [...@macro-user-callerid:8] Set(SIP/100-08fba098, CALLERID(all)=100 100) in new stack -- Executing [...@macro-user-callerid:9] Set(SIP/100-08fba098, REALCALLERIDNUM=100) in new stack -- Executing [...@macro-user-callerid:10] ExecIf(SIP/100-08fba098, 0|Set|CHANNEL(language)=) in new stack -- Executing [...@macro-user-callerid:11] GotoIf(SIP/100-08fba098, 1?continue) in new stack -- Goto (macro-user-callerid,s,20) -- Executing [...@macro-user-callerid:20] NoOp(SIP/100-08fba098, Using CallerID 100 100) in new stack -- Executing [966505103...@from-internal:2] Set(SIP/100-08fba098, _NODEST=) in new stack -- Executing [966505103...@from-internal:3] Macro(SIP/100-08fba098, record-enable|100|OUT|) in new stack -- Executing [...@macro-record-enable:1] GotoIf(SIP/100-08fba098, 1?check) in new stack -- Goto (macro-record-enable,s,4) -- Executing [...@macro-record-enable:4] AGI(SIP/100-08fba098, recordingcheck|20091005-123740|1254735460.0) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20091005-123740|1254735460.0: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing [...@macro-record-enable:5] MacroExit(SIP/100-08fba098, ) in new stack -- Executing [966505103...@from-internal:4] Macro(SIP/100-08fba098, dialout-trunk|2|966505103250||) in new stack -- Executing [...@macro-dialout-trunk:1] Set(SIP/100-08fba098, DIAL_TRUNK=2) in new stack -- Executing [...@macro-dialout-trunk:2] GosubIf(SIP/100-08fba098, 0?sub-pincheck|s|1) in new stack -- Executing [...@macro-dialout-trunk:3] GotoIf(SIP/100-08fba098, 0?disabletrunk|1) in new stack -- Executing [...@macro-dialout-trunk:4] Set(SIP/100-08fba098, DIAL_NUMBER=966505103250) in new stack -- Executing [...@macro-dialout-trunk:5] Set(SIP/100-08fba098, DIAL_TRUNK_OPTIONS=tr) in new stack -- Executing [...@macro-dialout-trunk:6] Set(SIP/100-08fba098, OUTBOUND_GROUP=OUT_2) in new stack -- Executing [...@macro-dialout-trunk:7] GotoIf(SIP/100-08fba098, 0?nomax) in new stack -- Executing [...@macro-dialout-trunk:8] GotoIf(SIP/100-08fba098, 0?chanfull) in new stack -- Executing [...@macro-dialout-trunk:9] GotoIf(SIP/100-08fba098, 0?skipoutcid) in new stack -- Executing [...@macro-dialout-trunk:10] Set(SIP/100-08fba098, DIAL_TRUNK_OPTIONS=) in new stack -- Executing [...@macro-dialout-trunk:11] Macro(SIP/100-08fba098, outbound-callerid|2) in new stack -- Executing [...@macro-outbound-callerid:1] ExecIf(SIP/100-08fba098, 0|SetCallerPres|) in new stack -- Executing [...@macro-outbound-callerid:2] ExecIf(SIP/100-08fba098, 0|Set|REALCALLERIDNUM=100) in new stack -- Executing [...@macro-outbound-callerid:3] GotoIf(SIP/100-08fba098, 1?normcid) in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [...@macro-outbound-callerid:6] Set(SIP/100-08fba098, USEROUTCID=) in new stack -- Executing [...@macro-outbound-callerid:7] Set(SIP/100-08fba098, EMERGENCYCID=) in new stack -- Executing [...@macro-outbound-callerid:8] Set(SIP/100-08fba098, TRUNKOUTCID=) in new stack -- Executing [...@macro-outbound-callerid:9] GotoIf(SIP/100-08fba098, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [...@macro-outbound-callerid:12] ExecIf(SIP/100-08fba098, 0|Set|CALLERID(all)=) in new stack -- Executing [...@macro-outbound-callerid:13] GotoIf(SIP/100-08fba098, 1?exit) in new stack -- Goto (macro-outbound-callerid,s,11) -- Executing [...@macro-outbound-callerid:11] MacroExit(SIP/100-08fba098, ) in new stack -- Executing [...@macro-dialout-trunk:12] ExecIf(SIP/100-08fba098, 0|AGI|fixlocalprefix) in new stack -- Executing [...@macro-dialout-trunk:13] Set(SIP/100-08fba098, OUTNUM=966505103250) in new stack -- Executing [...@macro-dialout-trunk:14] Set(SIP/100-08fba098, custom=DAHDI/DGTDM24) in new stack -- Executing [...@macro-dialout-trunk:15] ExecIf(SIP
Re: [asterisk-users] Drop calls when using Flash Operator Panel
hin lee wrote: Whenever I try to drag calls to the Parking Lot or On Hold, FOP would drop my calls. I have searched online and have found similar problem, such as the link below. I have tried their solution but still the FOP is not working correctly. I even installed the HUDLite server and is getting the same results. http://www.freepbx.org/forum/freepbx/users/flas...ot-transfering-calls My experience shows, that if you're getting dropped calls when using FOP, then your grabbing the wrong leg of the call. I've (often) grabbed the phone instead of the incoming line. We have a PRI, call comes in on channel 1, rings to the operator at 4100. Go to transfer a call, I (mistakenly) grab 4100 and drop it off on the destination, call drops. What I should have done is grab channel 1 on the inbound call. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
Are you series??? My card is FXO TDM2400, I am sure its designed to forward calls to pstn!!! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira Sent: Monday, October 05, 2009 5:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm outgoing At 04:32 PM 10/4/2009, you wrote: Hi I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls to that trunk, I am getting all circuits are busy now, do I have to do something specific?? I am using elastix. Sometimes you can't make a call on DAHDI until a call has been received. At least I can't. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
DAHDI/DGTDM24/966505103250 This (DGTDM24) is strange. Could you provide the setup of the DAHDI trunk? You should have something like DAHDI/g0/96 or DAHDI/10/96 Here are more info on this subject: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg226642.html HTH, Ioan (Nini) Indreias www.modulo.ro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
Man, thanks a lot! I just changed the name to g0 instead of DGTDM24 and it worked!! I would like to know where I can set the configuration for line tones( dial tone, call and busy tone) and where I can change different setting for polarity / current disconnect etc.. of the line? I cant find Zapata.cfg Thanks again! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ioan Indreias Sent: Monday, October 05, 2009 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm outgoing DAHDI/DGTDM24/966505103250 This (DGTDM24) is strange. Could you provide the setup of the DAHDI trunk? You should have something like DAHDI/g0/96 or DAHDI/10/96 Here are more info on this subject: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg226642.html HTH, Ioan (Nini) Indreias www.modulo.ro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem sending a DTMF remotely. Please need help!!
Hello,
I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.
isb177*CLI features show
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #8
Attended Transfer *2
One Touch Monitor *1
Disconnect Call * **
Park Call
One Touch MixMonitor
Dynamic Feature Default Current
--- --- ---
(none)
Call parking
Parking extension : 70
Parking context : parkedcalls
Parked call extensions: 71-78
My script is:
#!/usr/bin/php -q
?php
error_reporting (E_ALL);
set_time_limit(60);
ob_implicit_flush(false);
$ip_asterisk = 127.0.0.1;
$user_asterisk = admin;
$pass_asterisk = forward;
$canal = SIP/1000-0a292360; //hardcodeado
$oSocket = fsockopen($ip_asterisk, 5038, $errnum, $errdesc) or
die(Connection to host failed);
fputs($oSocket, Action: login\r\n);
fputs($oSocket, Username: $user_asterisk\r\n);
fputs($oSocket, Secret: $pass_asterisk\r\n\r\n);
fputs($oSocket, Action: PlayDTMF\r\n);
fputs($oSocket, Channel: $canal\r\n);
fputs($oSocket, Digit: #\r\n\r\n);
usleep(50);
fputs($oSocket, Action: PlayDTMF\r\n);
fputs($oSocket, Channel: $canal\r\n);
fputs($oSocket, Digit: 8\r\n\r\n);
usleep(50);
fputs($oSocket, Action: Logoff\r\n\r\n);
$loaded = ;
while (!feof($oSocket)){
$buffer = fgets($oSocket, 4096);
$loaded .= $buffer;}
$vec = explode(\n, $loaded);
$len = count($vec);
print_r($vec);
?
The script output is:
Array
(
[0] = Asterisk Call Manager/1.1
[1] = Response: Success
[2] = Message: Authentication accepted
[3] =
[4] = Response: Success
[5] = Message: DTMF successfully queued
[6] =
[7] = Response: Success
[8] = Message: DTMF successfully queued
[9] =
[10] = Response: Goodbye
[11] = Message: Thanks for all the fish.
[12] =
[13] =
)
When I run the script I can hear the two digit (only the audio) but nothing
happens, the Transfer menu doesnt start. The Cli shows:
[Oct 2 11:14:46] DEBUG[30054]: manager.c:2776 process_message: Manager
received command 'login'
== Manager 'admin' logged on from 127.0.0.1
[Oct 2 11:14:46] DEBUG[30054]: manager.c:2776 process_message: Manager
received command 'PlayDTMF'
[Oct 2 11:14:46] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread
-1216881776 Blocking 'SIP/1000-0a292360', already blocked by thread
-1217414256 in procedure ast_waitfor_nandfds
[Oct 2 11:14:47] DEBUG[29533]: channel.c:3341 ast_write: Deadlock avoided
for write to channel 'SIP/1000-0a292360'
[Oct 2 11:14:47] DEBUG[30054]: manager.c:2776 process_message: Manager
received command 'PlayDTMF'
[Oct 2 11:14:47] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread
-1216881776 Blocking 'SIP/1000-0a292360', already blocked by thread
-1217414256 in procedure ast_waitfor_nandfds
[Oct 2 11:14:47] DEBUG[29533]: channel.c:3341 ast_write: Deadlock avoided
for write to channel 'SIP/1000-0a292360'
[Oct 2 11:14:47] DEBUG[30054]: manager.c:2776 process_message: Manager
received command 'Logoff'
== Manager 'admin' logged off from 127.0.0.1
BUT, if I press #8 in the softphone, I can hear the two digit and
inmediately the Transfer menu begins playing 'pbx-transfer.gsm'. And the Cli
output in this case is:
[Oct 2 11:09:17] DEBUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of
60 bytes
[Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#),
at 192.168.0.148
[Oct 2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#'
received on SIP/1000-0a292360
[Oct 2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read: DTMF begin
passthrough '#' on SIP/1000-0a292360
[Oct 2 11:09:20] DEBUG[29533]: channel.c:4806 ast_generic_bridge: Got DTMF
begin on channel (SIP/1000-0a292360)
[Oct 2 11:09:20] DEBUG[29533]: channel.c:5150 ast_channel_bridge: Bridge
stops bridging channels SIP/1000-0a292360 and SIP/1001-0a026408
[Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event:
000b (len = 4)
[Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#),
at 192.168.0.148
[Oct 2 11:09:20] DTMF[29533]: channel.c:2768 __ast_read: DTMF end '#'
received on SIP/1000-0a292360, duration 80 ms
[Oct 2 11:09:20] DTMF[29533]: channel.c:2808 __ast_read: DTMF end accepted
with begin '#' on SIP/1000-0a292360
[Oct 2
Re: [asterisk-users] AEL problem: bug or feature?
forgot to mention this happens on Asterisk 1.4.26.1
Klaus Darilion schrieb:
Hi! I have a problem with jump in AEL:
_+43123456789! = jump +22;
+22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
_22 = { NoOp(); }
- OK
_+43123456789! = jump +22;
_+22 = { NoOp(); }
-- AEL compile error:
LOG: lev:4 file:pbx_ael.c line:1234 func: check_goto Error: file
./ofis/extensions.ael_trunking, line 525-525: goto: no label +22|1
exists in the current context, or any of its inclusions!
Is this is some special feature/limitation or just a bug?
thanks
Klaus
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Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling
Danny Nicholas schrieb: Sipregisterattempts would seem to be the simplest way to do this. It is 0 by default, changing it to 5 would stop the hacker after 5 tries. wrong. registerattempts wokrs the other way round - if Asterisk is the client and registers to another SIP proxy. regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] digium fax: can't indicate condition 19?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Kevin P. Fleming I am working on getting this situation resolved and should have new releases of FFA out at the end of this week, but in the meantime if you want to use FFA with T.38 support you'll have to use one of the versions of Asterisk listed on the download selector page. Any update on when the new FFA modules will be out? Thanks. sl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL problem: bug or feature?
On Mon, 5 Oct 2009, Klaus Darilion wrote:
forgot to mention this happens on Asterisk 1.4.26.1
Klaus Darilion schrieb:
Hi! I have a problem with jump in AEL:
_+43123456789! = jump +22;
+22 = { NoOp(); }
Don't you need another semi-colon after the closing brace?
Since AEL compiles to dialplan, what does the resulting dialplan look
like?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] AEL problem: bug or feature?
Klaus Darilion schrieb:
forgot to mention this happens on Asterisk 1.4.26.1
Klaus Darilion schrieb:
Hi! I have a problem with jump in AEL:
_+43123456789! = jump +22;
+22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
_22 = { NoOp(); }
- OK
_+43123456789! = jump +22;
_+22 = { NoOp(); }
-- AEL compile error:
LOG: lev:4 file:pbx_ael.c line:1234 func: check_goto Error: file
./ofis/extensions.ael_trunking, line 525-525: goto: no label +22|1
exists in the current context, or any of its inclusions!
Not that it should make a difference (as + is not a special
character in Asterisk's patterns) but did you try
_+43123456789! = jump +22;
_[+]22 = { NoOp(); }
just in case?
Is this is some special feature/limitation or just a bug?
Looks like a bug to me.
Philipp Kempgen
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[asterisk-users] Questions about app_jack.c
Hello,
My configuration is :
Card 0 - kernel dummy sound card
Card 1 - my soundcard
I have a jackd running in background. My jackd launch command is :
jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0
--capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2
--outchannels 2 --dither triangular
1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to the
kernel dummy sound card (allow me dial command). I do a call with a
JACK_HOOK from app_jack.so, sound is sent but no one is received.
My extensions.conf :
exten = _0.,1,Answer
exten =
_0.,n,Set(JACK_HOOK(manipulate,c(asterisk))i(from_voip:input)o(to_voip:outpu
t)))=on)
exten = _0.,n,Dial(SIP/freephonie-out/${EXTEN:1})
Asterisk command :
console dial 0
2) Jackd works well with anothers applications when I force them to use jack
as input/output. - probably not a jack configuration problem.
3) If I kill jackd and I use chan_alsa.so with the real soundcard, it works.
- probably not a network or sip configuration problem.
4) If I replace f_buf[i] = s_buf[i] * (1.0 / SHRT_MAX); with f_buf[i] =
0.5 * sin(0.3454 * ((float) i)); in app_jack.c and I retry the test 2, I
get test sound.
It looks like no sound was read in channel...
Do you have any idea ?
Fabien
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Re: [asterisk-users] digium fax: can't indicate condition 19?
Scott L. Lykens wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Kevin P. Fleming I am working on getting this situation resolved and should have new releases of FFA out at the end of this week, but in the meantime if you want to use FFA with T.38 support you'll have to use one of the versions of Asterisk listed on the download selector page. Any update on when the new FFA modules will be out? Still working on fixing bugs. Hopefully tomorrow. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
I cant find Zapata.cfg You have a DAHDI installation thus you have to find chan_dahdi.conf. it should be located under /etc/asterisk Regarding the configuration for tones you have to check indications.conf file Best regards, Nini ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What dahdi_dynamic and dahdi_transcode modules are for?
I want to know what dahdi_dynamic and dahdi_transcode modules are for. What are they purpose?. I have read in this thread: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg180595.html That dahdi_transcode is for the TC400B transcoder card. But this does not seems to be true, because in the oficial documentation of the card, does not mention at all to load this module. ...and dadhi_dynamic? Where i can find more info about this two modules does? Thank you. Gonzalo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OriginateResponse Event
Hi people, I'm executing some parallel Originate async, is there a way to know the result of each originate?... I was looking at the OriginateResponse event, but I don't know how to get it from my web service. Also, if I have 3 originate in parallel, how can I identify the OriginateResponse of each one? Thanks in advance... Anahi Ludueña _ Nuevo Windows Live, un mundo lleno de posibilidades. Descúbrelo. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL problem: bug or feature?
Steve Edwards schrieb:
On Mon, 5 Oct 2009, Klaus Darilion wrote:
forgot to mention this happens on Asterisk 1.4.26.1
Klaus Darilion schrieb:
Hi! I have a problem with jump in AEL:
_+43123456789! = jump +22;
+22 = { NoOp(); }
Don't you need another semi-colon after the closing brace?
No. IIRC this was changed from AEL to AEL2.
Since AEL compiles to dialplan, what does the resulting dialplan look
like?
it failes during compilation, thus there is no dialplan generated.
regards
klaus
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Re: [asterisk-users] OriginateResponse Event
Each response set has a uniqueid field that designates the start time and call sequence of the call. Unless you manage to start 36K calls simultaneously, you can track each call with this. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Monday, October 05, 2009 9:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] OriginateResponse Event Hi people, I'm executing some parallel Originate async, is there a way to know the result of each originate?... I was looking at the OriginateResponse event, but I don't know how to get it from my web service. Also, if I have 3 originate in parallel, how can I identify the OriginateResponse of each one? Thanks in advance... _ Anahi Ludueña _ Nuevo Windows Live, un mundo lleno de posibilidades Descúbrelo. http://www.microsoft.com/windows/windowslive/default.aspx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL problem: bug or feature?
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
forgot to mention this happens on Asterisk 1.4.26.1
Klaus Darilion schrieb:
Hi! I have a problem with jump in AEL:
_+43123456789! = jump +22;
+22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
_22 = { NoOp(); }
- OK
_+43123456789! = jump +22;
_+22 = { NoOp(); }
-- AEL compile error:
LOG: lev:4 file:pbx_ael.c line:1234 func: check_goto Error: file
./ofis/extensions.ael_trunking, line 525-525: goto: no label +22|1
exists in the current context, or any of its inclusions!
Not that it should make a difference (as + is not a special
character in Asterisk's patterns) but did you try
_+43123456789! = jump +22;
_[+]22 = { NoOp(); }
just in case?
Indeed, that works. Also jump +22${FOO}; works
Is this is some special feature/limitation or just a bug?
Looks like a bug to me.
Me too :-)
https://issues.asterisk.org/view.php?id=16019
regards
klaus
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[asterisk-users] DTMF problem during read()
I have a simple dialplan. Using the read cmd, I ask caller for his passcode. If the caller is calling from an plain old analog phone, his dtmf is not heard until the prompt stops playing. SIP phones work correctly. I've trird everything I found searching the net. I've tried all the dtmfmode. I'm using 1.4.26 Currently my vitelity sip account is setup: insecure=very canreinvite=no host=xx.xx.xx.xx qualify=yes dtmfmode=rfc2833 disallow=all allow=ulaw rfc2833compensate=yes I need to trouble shoot this furthur. I read I can enable rtp debug IP but I can't find any output. Thanks, Bart___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and QSIG
Hello, I'm looking for info about interconnecting asterisk to QSIG GF enabled PABX over PRI . Any information and pointers will be helpful. The very first first question: does asterisk support QSIG BC and GF natively i see that it is supported through CAPI enabled cards but what about support through librip/dahdi? Thanks Vadim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *****SPAM***** DTMF problem during read()
You are playing the prompt with Background or Playback? Please post the dialplan snippet. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bart Fisher Sent: Monday, October 05, 2009 10:03 AM To: asterisk-users@lists.digium.com Subject: *SPAM* [asterisk-users] DTMF problem during read() I have a simple dialplan. Using the read cmd, I ask caller for his passcode. If the caller is calling from an plain old analog phone, his dtmf is not heard until the prompt stops playing. SIP phones work correctly. I've trird everything I found searching the net. I've tried all the dtmfmode. I'm using 1.4.26 Currently my vitelity sip account is setup: insecure=very canreinvite=no host=xx.xx.xx.xx qualify=yes dtmfmode=rfc2833 disallow=all allow=ulaw rfc2833compensate=yes I need to trouble shoot this furthur. I read I can enable rtp debug IP but I can't find any output. Thanks, Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi dies with No more room in scheduler
Hi, I noticed that Dahdi starting producing these error messages: ERROR[29250] chan_dahdi.c: No more room in scheduler ERROR[29250] chan_dahdi.c: Asked to delete sched id -1??? during which time I could not send any calls or receive calls on at least one of my Dahdi spans. The only way to clear the problem seemed to be to restart Asterisk. It appears to start after the following message ERROR[29250] chan_dahdi.c: T200 counter expired, nothing to send... This is with dahdi 2.2.0 and asterisk 1.6.0.10. Any ideas on this issue? Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OriginateResponse Event
Thanks Danny, but how can I get it from my web service? Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 5 Oct 2009 10:03:41 -0500 Subject: Re: [asterisk-users] OriginateResponse Event Each response set has a uniqueid field that designates the start time and call sequence of the call. Unless you manage to start 36K calls simultaneously, you can track each call with this. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Monday, October 05, 2009 9:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] OriginateResponse Event Hi people, I'm executing some parallel Originate async, is there a way to know the result of each originate?... I was looking at the OriginateResponse event, but I don't know how to get it from my web service. Also, if I have 3 originate in parallel, how can I identify the OriginateResponse of each one? Thanks in advance... Anahi Ludueña Nuevo Windows Live, un mundo lleno de posibilidades Descúbrelo. _ Nuevo Windows Live, un mundo lleno de posibilidades. Descúbrelo. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXW4024 experience
Don't use them for Fax... I spent too much time trying to use one for a faxing ATA. (We went with the AudioCodes MP-124 instead, which rocks). We to have some analog phones and an analog IVR system hooked up to one with no issues. They are easy to configure if you just need to hook up some analog handsets. -Jonathan On Mon, Oct 5, 2009 at 2:14 AM, Olivier oza-4...@myamail.com wrote: Hi, In this http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/204725/focus=221375 dating from 2008, experiences with Grandstream GXW4024 were asked. Has anyone something up-to-date to share about this ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi dies with No more room in scheduler
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 James Lamanna wrote: This is with dahdi 2.2.0 and asterisk 1.6.0.10. Any ideas on this issue? Check to see if this is a bug that has been fixed in 1.6.0.10. I think the current is 1.6.0.15 and there has been significant bug fixes since your version. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKyhoUCFu3bIiwtTARAsUjAJ44sqcqVk3VtrecJVVgjr3LlwtTJgCeP4qH jxsq2fuwqEb+qlrxziOEye4= =CcCX -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drop calls when using Flash Operator Panel
Doug, I have tested both ends and got the same results. I was able to using FOP to drag to Conferences, just not the Parking Lot. Another strange thing I found is this: - On ext 5134, I call ext 5334 - 5334 picks up the call - using FOP, I drag 5334 and drop it back to 5334. - 5334 gets disconnected - 5134 gets the message Call Forward on no answer on extension 34 cancel. Somehow FOP strips the first two digits off the extension. It should be 5334 or 5134, not 34. I don't know if that's the problem or not. From: Doug Lytle supp...@drdos.info To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Mon, October 5, 2009 2:40:50 AM Subject: Re: [asterisk-users] Drop calls when using Flash Operator Panel hin lee wrote: Whenever I try to drag calls to the Parking Lot or On Hold, FOP would drop my calls. I have searched online and have found similar problem, such as the link below. I have tried their solution but still the FOP is not working correctly. I even installed the HUDLite server and is getting the same results. http://www.freepbx.org/forum/freepbx/users/flas...ot-transfering-calls My experience shows, that if you're getting dropped calls when using FOP, then your grabbing the wrong leg of the call. I've (often) grabbed the phone instead of the incoming line. We have a PRI, call comes in on channel 1, rings to the operator at 4100. Go to transfer a call, I (mistakenly) grab 4100 and drop it off on the destination, call drops. What I should have done is grab channel 1 on the inbound call. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OriginateResponse Event
You would have to be able to query an AMI interface for results (PHP, PERL, etc) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Monday, October 05, 2009 10:46 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] OriginateResponse Event Thanks Danny, but how can I get it from my web service? _ Anahi Ludueña _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 5 Oct 2009 10:03:41 -0500 Subject: Re: [asterisk-users] OriginateResponse Event Each response set has a uniqueid field that designates the start time and call sequence of the call. Unless you manage to start 36K calls simultaneously, you can track each call with this. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Monday, October 05, 2009 9:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] OriginateResponse Event Hi people, I'm executing some parallel Originate async, is there a way to know the result of each originate?... I was looking at the OriginateResponse event, but I don't know how to get it from my web service. Also, if I have 3 originate in parallel, how can I identify the OriginateResponse of each one? Thanks in advance... _ Anahi Ludueña _ Nuevo Windows Live, un mundo lleno de posibilidades Descúbrelo. http://www.microsoft.com/windows/windowslive/default.aspx _ Nuevo Windows Live, un mundo lleno de posibilidades Descúbrelo. http://www.microsoft.com/windows/windowslive/default.aspx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi dies with No more room in scheduler
On Mon, 5 Oct 2009, James Lamanna wrote: Hi, I noticed that Dahdi starting producing these error messages: ERROR[29250] chan_dahdi.c: No more room in scheduler ERROR[29250] chan_dahdi.c: Asked to delete sched id -1??? during which time I could not send any calls or receive calls on at least one of my Dahdi spans. The only way to clear the problem seemed to be to restart Asterisk. It appears to start after the following message ERROR[29250] chan_dahdi.c: T200 counter expired, nothing to send... This is with dahdi 2.2.0 and asterisk 1.6.0.10. Any ideas on this issue? This appears to be issue: https://issues.asterisk.org/view.php?id=15892nbn=4 reported a few weeks ago. Doesn't look like there is an update yet. One of the reporters mentions that dahdi restart also fixes the problem and allows calls again. Might be better to do this then restart asterisk, as you won't interrupt any non-dahdi conversations. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] web module for video calls
Anyone working on this? Would love to have a click to talk that would operate with my Grandstream video phones. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
At 02:47 AM 10/5/2009, you wrote: TDM04. The original 4 channel card with 4 red cards installed. Are you series??? My card is FXO TDM2400, I am sure its designed to forward calls to pstn!!! At 04:32 PM 10/4/2009, you wrote: Hi I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls to that trunk, I am getting all circuits are busy now, do I have to do something specific?? I am using elastix. Sometimes you can't make a call on DAHDI until a call has been received. At least I can't. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Method to downgrade asterisk
I currently have asterisk-1.4.26.2 installed and working. It was sugguested I try asterisk-1.4.25 to see if it fixes my SIP dtmf problems. What is the method to downgrade? Do I just do in the asterisk-1.4.25 folder: make clean ./configure make install Or do I need to 'make clean' in the asterisk-1.4.26.2 first then move to the asterisk-1.4.25 folder and do ./configure make install? Thanks, Bart___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *****SPAM***** Method to downgrade asterisk
Each is independent of the other. The important things are to make sure asterisk is not running when doing make install and to clean /usr/lib/asterisk/modules before make install. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bart Fisher Sent: Monday, October 05, 2009 11:54 AM To: asterisk-users@lists.digium.com Subject: *SPAM* [asterisk-users] Method to downgrade asterisk I currently have asterisk-1.4.26.2 installed and working. It was sugguested I try asterisk-1.4.25 to see if it fixes my SIP dtmf problems. What is the method to downgrade? Do I just do in the asterisk-1.4.25 folder: make clean ./configure make install Or do I need to 'make clean' in the asterisk-1.4.26.2 first then move to the asterisk-1.4.25 folder and do ./configure make install? Thanks, Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OriginateResponse Event
On Mon, 2009-10-05 at 14:55 +, Anahi Ludueña wrote: I'm executing some parallel Originate async, is there a way to know the result of each originate?... I was looking at the OriginateResponse event, but I don't know how to get it from my web service. Also, if I have 3 originate in parallel, how can I identify the OriginateResponse of each one? Whenever you send an action through AMI, you should also provide an ActionID string, which is something you create and should be unique for each action you send. The response from that action should contain that same ActionID, so that you can identify the responses with the corresponding action based on the ActionID. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OriginateResponse Event
What are the limitations of ActionID? In all of the examples I see, it is usually 1 or some integer. Can it be a timestamp like uniqueid? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith Sent: Monday, October 05, 2009 12:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OriginateResponse Event On Mon, 2009-10-05 at 14:55 +, Anahi Ludueña wrote: I'm executing some parallel Originate async, is there a way to know the result of each originate?... I was looking at the OriginateResponse event, but I don't know how to get it from my web service. Also, if I have 3 originate in parallel, how can I identify the OriginateResponse of each one? Whenever you send an action through AMI, you should also provide an ActionID string, which is something you create and should be unique for each action you send. The response from that action should contain that same ActionID, so that you can identify the responses with the corresponding action based on the ActionID. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OriginateResponse Event
Danny Nicholas schrieb: What are the limitations of ActionID? In all of the examples I see, it is usually 1 or some integer. Can it be a timestamp like uniqueid? AFAICR ActionID is a string. Probably limited to 255 characters or something. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith On Mon, 2009-10-05 at 14:55 +, Anahi Ludueña wrote: I'm executing some parallel Originate async, is there a way to know the result of each originate?... I was looking at the OriginateResponse event, but I don't know how to get it from my web service. Also, if I have 3 originate in parallel, how can I identify the OriginateResponse of each one? Whenever you send an action through AMI, you should also provide an ActionID string, which is something you create and should be unique for each action you send. The response from that action should contain that same ActionID, so that you can identify the responses with the corresponding action based on the ActionID. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Receptionist GUI?
Hey, all. Just wondering if there's a GUI out there -- preferably OSS, but I'll take what-have-you -- that a) can run on an Ubuntu/Debian box, and b) allows a receptionist to see what calls are in-process, and forward calls from their phone to somewhere else. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OriginateResponse Event
On Mon, 2009-10-05 at 12:33 -0500, Danny Nicholas wrote: What are the limitations of ActionID? In all of the examples I see, it is usually 1 or some integer. Can it be a timestamp like uniqueid? It is simply a unique string. You can make it a timestamp if you'd like, but I doubt that means you can guarantee that it's going to be unique across concurrent calls. Otherwise, it's not likely to be very useful to you in the long run. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receptionist GUI?
On Mon, Oct 5, 2009 at 2:34 PM, Ken D'Ambrosio k...@jots.org wrote: Hey, all. Just wondering if there's a GUI out there -- preferably OSS, but I'll take what-have-you -- that a) can run on an Ubuntu/Debian box, and b) allows a receptionist to see what calls are in-process, and forward calls from their phone to somewhere else. Thanks! -Ken Flash Operator Panel is very nice. I think recently the newer versions are not free or open source but the older versions are, and if the pricing is reasonable, and better features, I would certainly look at purchasing some licenses. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
Thanks, I made the zone, and now call disconnect works ok! i have one last problem, I have defined the card g0 to have 24 channels, but every time I try to call, if all ports are off the call always go to the first port, how can I balance the calls over all ports??? Any suggestions appreciated. Thanks all for the help. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ioan Indreias Sent: Monday, October 05, 2009 5:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm outgoing I cant find Zapata.cfg You have a DAHDI installation thus you have to find chan_dahdi.conf. it should be located under /etc/asterisk Regarding the configuration for tones you have to check indications.conf file Best regards, Nini ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Method to downgrade asterisk
You don't need to run make clean the 1.4.26.2 folder. Just do ./configure make install in the 1.4.25 folder. When you run make you are just compiling the source into binaries in that folder. You can have a number of these source folders and they won't conflict. Make install is what actually copies the files to the install point. Running make install on a different version will just overwrite all installed files with those files. Ryan On Mon, Oct 5, 2009 at 12:53 PM, Bart Fisher b...@icpage.com wrote: I currently have asterisk-1.4.26.2 installed and working. It was sugguested I try asterisk-1.4.25 to see if it fixes my SIP dtmf problems. What is the method to downgrade? Do I just do in the asterisk-1.4.25 folder: make clean ./configure make install Or do I need to 'make clean' in the asterisk-1.4.26.2 first then move to the asterisk-1.4.25 folder and do ./configure make install? Thanks, Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
B.Masoud @ SH schrieb: I have defined the card g0 to have 24 channels, but every time I try to call, if all ports are off the call always go to the first port, how can I balance the calls over all ports??? http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels#DialingaGroup Dial(Dahdi/r0/...) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receptionist GUI?
We use iSymphony Asterisk Operator Panel with a great deal of success. http://www.i9technologies.com/index.php?option=com_contenttask=viewid=19Itemid=40 Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com Ken D'Ambrosio wrote: Hey, all. Just wondering if there's a GUI out there -- preferably OSS, but I'll take what-have-you -- that a) can run on an Ubuntu/Debian box, and b) allows a receptionist to see what calls are in-process, and forward calls from their phone to somewhere else. Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receptionist GUI?
There are plenty of good products out there, but I use my own PERL/Apache/AMI interface for this _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Baker Sent: Monday, October 05, 2009 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Receptionist GUI? We use iSymphony Asterisk Operator Panel with a great deal of success. http://www.i9technologies.com/index.php?option=com_content http://www.i9technologies.com/index.php?option=com_contenttask=viewid=19; Itemid=40 task=viewid=19Itemid=40 Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com http://www.glastender.com/ Ken D'Ambrosio wrote: Hey, all. Just wondering if there's a GUI out there -- preferably OSS, but I'll take what-have-you -- that a) can run on an Ubuntu/Debian box, and b) allows a receptionist to see what calls are in-process, and forward calls from their phone to somewhere else. Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OriginateResponse Event
On Monday 05 October 2009 12:33:47 Danny Nicholas wrote: What are the limitations of ActionID? In all of the examples I see, it is usually 1 or some integer. Can it be a timestamp like uniqueid? I use AMI as part of an external bash application and I usually specify the ActionID to the something unique outside of Asterisk itself, such as as the external bash process id $$ or the process id combined with the date in nanoseconds. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (OT) Zaptel, SuSE 9.3, Debian
Suse 11.1 for some reason won't install on the VIA box. After installing get garbled text on screen. I want to fix this as a learning experience. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: 04 October 2009 23:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (OT) Zaptel, SuSE 9.3, Debian Just detecting this tread... Moving to Debian is quite a big step. How about updating to openSUSE_11.1 and use the prebuild asterisk packages (either zaptel or dahdi) . On the OBS they are available for 1.4.x, 1.6.0, 1.6.1 hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel problems on SUSE 9.3
Core show channeltypes:
SIP Session Initiation Protocol (SIP)yes yes
yes
Console OSS Console Channel Driver no yes
no
OOH323 Objective Systems H323 Channel Driverno yes
no
Skinny Skinny Client Control Protocol (Skinny) no yes
no
Phone Standard Linux Telephony API Driver no yes
no
Agent Call Agent Proxy Channel yes yes
no
IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes
yes
Local Local Proxy Channel Driver yes yes
no
MGCPMedia Gateway Control Protocol (MGCP)yes yes
no
--LI
9 channel drivers registered.
Which one in above signifies the Zaptel channel?
asterisk:~ # cat /proc/zaptel/*
Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 (MASTER)
1 WCTDM/0/0 RED
2 WCTDM/0/1 RED
3 WCTDM/0/2 RED
4 WCTDM/0/3 RED
lspci:
:00:14.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
I fiddled an it is working now - all green lights on on board.
It seems that the zaptel startup script does not work. I noticed at startup
the line:
/etc/init.d/zaptel: line 40: /etc/init.d/functions: No such file or
directory
Line 40:
# Source function library.
if [ $system = redhat ]; then
. $initdir/functions || exit 0
Fi
The . %initdir... is line 40.
Any ideas how to fix this file on suse?
I think if I can fix this everything should be ok.
Angus
Could the system be confusing the Zaptel device for Tiger Jet? Ie loading
wrong driver?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: 04 October 2009 22:47
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Zaptel problems on SUSE 9.3
On Sun, Oct 04, 2009 at 11:28:23AM +0100, Angus Asterisk wrote:
Hi
My asterisk output is:
chan_sip.so = (Session Initiation Protocol (SIP))
Asterisk Ready.
-- Registered SIP '201' at 192.168.0.55 port 33906
-- Saved useragent X-Lite release 1011s stamp 41150 for peer 201
-- Executing [907768385...@default:1] Dial(SIP/201-083e75c0,
ZAP/g1/907768385144|60) in new stack
[Oct 4 11:54:27] WARNING[6255]: channel.c:3388 ast_request: No channel
type
registered for 'ZAP'
Looks like chan_zap failed to load or something similar.
What is the output of: (in Asterisk)
core show channeltypes
(In Linux)
cat /proc/zaptel/*
[Oct 4 11:54:27] WARNING[6255]: app_dial.c:1275 dial_exec_full: Unable to
create channel of type 'ZAP' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [907768385...@default:2] Hangup(SIP/201-083e75c0, )
in
new stack
== Spawn extension (default, 907768385144, 2) exited non-zero on
'SIP/201-083e75c0'
when I make a call from a sip device to my outbound analog trunk using a
Digium TDM card.
My /etc/zaptel.conf file:
loadzone=uk
defaultzone=uk
fxsks=1-4
I am in the uk by the way.
Relevant part of /etc/astersk/zapata.conf:
signalling=v23 ; added for UK CLI detection
cidstart=polarity ; added for UK CLI detection
context=frompstnanalog
group=1
callgroup=1
pickupgroup=1
signalling=fxs_ks
channel=1-4
part of extensions.conf:
exten = _X.,1,Dial(ZAP/g1/${EXTEN},60)
exten = _X.,2,Hangup
I am running suse 9.3 on via and read article regarding old version of
zaptel driver and fixed as per script -
http://www.voip-info.org/wiki/view/Asterisk+Linux+SuSE
So now running dmesg reveals:
zaptel: unsupported module, tainting kernel.
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.4.12.1
Zaptel Echo Canceller: MG2
Have you actually loaded the module wctdm ?
So that looks encouraging
But still getting problem dialing out.
Also quite worrying is that there are no lights on the Digium card.
This used to work on same box and same operating system. I just can't
remember how I got it to work last time.
Anyone have any suggestions?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] Receptionist GUI?
On Mon, Oct 5, 2009 at 11:34 AM, Ken D'Ambrosio k...@jots.org wrote: Hey, all. Just wondering if there's a GUI out there -- preferably OSS, but I'll take what-have-you -- that a) can run on an Ubuntu/Debian box, and b) allows a receptionist to see what calls are in-process, and forward calls from their phone to somewhere else. Thanks! -Ken I can add a recommendation for iSymphony - cheaper than dirt, easy to configure, and the users like it. CP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receptionist GUI?
2009/10/5 CunningPike cunningp...@gmail.com: I can add a recommendation for iSymphony - cheaper than dirt, easy to configure, and the users like it. CP Hi , but this is free? regardss -- rickygm http://gnuforever.homelinux.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receptionist GUI?
$595 US. Cheap, but depends on the price of local dirt. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of troxlinux Sent: Monday, October 05, 2009 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Receptionist GUI? 2009/10/5 CunningPike cunningp...@gmail.com: I can add a recommendation for iSymphony - cheaper than dirt, easy to configure, and the users like it. CP Hi , but this is free? regardss -- rickygm http://gnuforever.homelinux.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel problems on SuSE 9.3
Angus Asterisk schrieb: It seems that the zaptel startup script does not work. I noticed at startup the line: /etc/init.d/zaptel: line 40: /etc/init.d/functions: No such file or directory Line 40: # Source function library. if [ $system = redhat ]; then . $initdir/functions || exit 0 Fi The . %initdir... is line 40. Any ideas how to fix this file on suse? /etc/init.d/functions might be available as /lib/lsb/init-functions so the snippet in /etc/init.d/zaptel could be changed to something like # Source function library. if [ -e /lib/lsb/init-functions ]; then . /lib/lsb/init-functions || exit 0 elif [ -e $initdir/functions ]; then . $initdir/functions || exit 0 fi (untested) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Networking Concept
Hello, I would like to know how Asterisk deal in this case: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk will deal with this call?? Will his latency be JAPN-UK + UK-China (around 1000ms !) or only from Japan to China??? Please let me know. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Networking Concept
B.Masoud @ SH wrote: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk will deal with this call?? Will his latency be JAPN-UK + UK-China (around 1000ms !) or only from Japan to China??? In the case of the SIP protocol, the audio (RTP) traffic can be re-routed on the fly from A(jp) to C(ch), reducing the audio latency, (and sometimes increasing your headaches). This is calling re-INVITE, and can be turned on on asterisk. For other protocols there are similar features. I think your latency figures are a little bit exaggerated if you speak about the network latency. I am in Spain and my latency to China at my home ADSL is arround 80ms for mainland. 250ms to Tokio tough. Regards -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receptionist GUI?
On Mon, Oct 5, 2009 at 6:31 PM, Danny Nicholas da...@debsinc.com wrote: $595 US. Cheap, but depends on the price of local dirt. LOL... dirt in Argentina is cheaper. -- Nicolás Gudiño Buenos Aires - Argentina ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] got stuck at 150 calls, above that not working in stress test
Hi Matt, Thanks so much for your help. I tried lot of ways to trouble shoot the issue, but finally I figured out that it was from the carrier side that they had set the limit of 150. Till now I under the impression that they provide just the bandwidth for the trunk, but they have the ability to limit the concurrent calls. Thanks Sandesh On Sun, Oct 4, 2009 at 9:06 PM, Matt Riddell li...@venturevoip.com wrote: On 3/10/09 3:55 AM, das sandesh wrote: I am using the command: ./sipp -sn uac -d 200 -s repective context pattern IP Address -l 200 Its 10 calls per second and 200 concurrent calls, similarly I used 2 ssh sessions each sending 100 concurrent calls. But this was limiting to only 150 calls. Start with 5 calls per second. Also, I don't notice anything to set it to this, are you sure you're not trying to start all those calls concurrently? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXW4024 experience
2009/10/5 Jonathan Thurman jthurma...@gmail.com Don't use them for Fax... I spent too much time trying to use one for a faxing ATA. (We went with the AudioCodes MP-124 instead, which rocks). We to have some analog phones and an analog IVR system hooked up to one with no issues. They are easy to configure if you just need to hook up some analog handsets. I'm planning to use them as a temporary device (while cabling is densified to support IP telephony) for analog phones. As such, price per port is a major requirement. -Jonathan On Mon, Oct 5, 2009 at 2:14 AM, Olivier oza-4...@myamail.com wrote: Hi, In this http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/204725/focus=221375 dating from 2008, experiences with Grandstream GXW4024 were asked. Has anyone something up-to-date to share about this ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T38 REINVITe issue
Hi
My call flow is
T38 static IP gateway -- Asterisk -- Sip Provider-- PSTN
Call is placed in reverse direction - from PSTN to T38 Gateway.
T38 GW reinvites with T38, and asterisk passes it along to the SIP provider.
The SIP provider challenges it and asterisk reponds to the Challenge with
INVITE with Auth credentials...however, the Asterisk changes the SDP and
replaces the T38 info in SDP with G711uLawand fax fails. How do I configure
the host entry in users.conf such that it maintains the T38 reinvite as it
responds to the SIP INVITE challenge from the Sip Provider.
Her eis my users.conf entry for Asterisk registration to the Sip Provider. (I
know I don't have T38 as allowed codecs, not sure what to add for T38)
[trunk_66]
;register
allow = ulaw
dialformat = ${EXTEN:1}
canreinvite = no
hasexten = no
hasiax = no
hassip = yes
host = provider.com
insecure = very
port = 5060
registeriax = no
registersip = yes
trunkname = abc
username = abc
disallow = gsm,g726,alaw
contact = abc
secret = abc
Any ideas appreciated.
Thx
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