[asterisk-users] Asterisk DTMF RFC2833 issues
Hi all I have posted a question on the asterisk dev board about this issue but I want to see if any users have run up against this. This issue is that when calls are run through Broadvox and Level 3 the in-call rfc2833 dtmf is not reliable. This occured for me on asterisk version 1.6.1.18, 1.6.1.20 it appears to have been fixed when I went to 1.6.2.11 but broken again in 1.6.2.12-rc1. I have tested with Grandstream and SNOM phones and both fail 90% of the time Unidata phones fail 10% of the time Audiocodes and Grandstream ATA's appear to not suffer from the issue on any version of asterisk. What happens is when a caller trys to enter DTMF keys durring a call the far end routed through these carriers do not detect all of the digits. We did captures with broadvox and here is what they have said. Hello, Per our phone conversation I have attached our signaling capture. The issue is that after we receive a RTP packet, the RTP event that follows needs to be sent within 100 ms. Anything greater than 100 ms will not be received. Thank you, Broadvox Network Operations Center Any one else seen this? Any ideas? Please note you must be being proxied directly to the carrier so your RTP flows direct other wise you will not see the issue. Thanks Bryant -- _ -- 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] Protect yourself
Hey all We are seeing intrusion attempts coming from address 201.47.236.122 today They were hitting our switches trying to get in. So we blocked them at our firewall. Just wanted to put the word out so you all can protect your self. Bryant -- _ -- 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] Migrating 1.4 to 1.6.2
From: Bruce Ferrell bferr...@baywinds.org much static testing of my realtime configuration and applications I'm almost ready to pull the trigger. The one thing I've been able to determine is what I need to do to migrate my g729 licenses. Has anyone got any advice for me on this? The Digium site is... difficult to navigate TIA Bruce Ferrell--- If you are not changing servers you just download the correct binary for 1.6.2 for your machine. If your are moving machines then you must re-register the license on the new box. If you have moved them before you must call Digium and have them increment the count on the licenses. Here is a link to the general install instructions. http://downloads.digium.com/pub/telephony/codec_g729/README It is not really hard to do you just need to follow the steps. Bryant -- _ -- 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] only part of dialplan available
I have found it best when doing remarks to not use the ;- combination as I have seen it cause failuers on dialplan reload. Bryant What I saw was that Asterisk stumbles when putting a comment like this : ;-- bla bla !!! It should be : ; -- bla bla !!! So with a space between ; and -- The rest of my dialplan came available when doing this... So problem solved. Jonas. On 08/28/2010 11:25 AM, kisho...@techroutes.com wrote: First of all explan your dial plan and extensions. i will resolve that... Regards, Kishor kumarHello list, yesterday I finished work having my whole dialplan available... Today I want to make a call from one local phone to another and I get this : [Aug 28 10:48:57] NOTICE[1895]: chan_sip.c:15144 handle_request_invite: Call from 'test2' to extension '60' rejected because extension not found. Although I have this context : [from-TEST] where all my local extensions are defined... Yesterday all went fine, today it no longer works. With the command dialplan show [tab], I also see only a small part of all my defined contexts... Reloading, restarting... it all does not help... When I look at my file extensions.conf, it has not changed !! Asterisk just only loads 20% of the total dialplan... Using asterisk 1.4.30. Don't know which nightmare Asterisk had last night, but it's all messed up this morning ! Anyone has had the same experience yet ?! Any solution ?! Kind regards, Jonas. -- -- _ -- 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 does not translate from wav to alaw
On Sat, Aug 28, 2010 at 3:22 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/zip-code.wav does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open /var/lib/asterisk/sounds/vprompts/zip-code.wav (format 0x8 (alaw)): No such file or directory [Aug 28 11:16:29] WARNING[2705]: pbx.c:5752 pbx_builtin_background: ast_streamfile failed on SIP/test1-000f for /var/lib/asterisk/sounds/vprompts/zip-code.wav Am I missing a module to translate from wav to alaw/gsm/g726/... ?? My guess is that your .wav file is NOT 8khz. The system doesn't accept anything but wav files at 8khz. Use sox to downsample to 8khz (and 1 chan), and the problems should go away. While you are at it, you could use sox to convert to the target format in a single operation. The scripts that Digium uses to take Allison's voice prompts (at 48khz) to the different formats, convert things to slin (raw) as a central format, but in my experience, the fewer steps the better. But I doubt that anyone could detect the difference in the end result... Here's what I do with CD-qual sounds to turn them into the common Asterisk formats: Assume $i is the name of the .wav file you want to convert: x=`basename $i .wav` sox -v 0.7 $i -r 16000 -c 1 -t sw $x.sln16 sox -v 0.7 $i -r 8000 -c 1 -t sw $x.raw sox -r 8000 -c 1 -t sw $x.raw -t gsm $x.gsm## OR ### sox -v 0.7 $i -r 8000 -t gsm $x.gsm sox -r 8000 -c 1 -t sw $x.raw -t ul $x.ulaw## OR ### sox -v 0.7 $i -r 8000 -t ul $x.ulaw sox -r 8000 -c 1 -t sw $x.raw -t al $x.alaw ## OR ### sox -v 0.7 $i -r 8000 -t wav $x.wav rm $x.raw y=`pwd` sudo asterisk -rx file convert $y/$i $y/$x.g722 I'm ignoring the siren g729 formats; use asterisk for those in like format, depending on your asterisk version and codecs. Allison normalizes the volume of sounds she distributes; use the -v 0.7 to bring the volume down a bit to the standard, and your sounds won't stick out against rest of Allison's existing recordings in Asterisk. Digium uses a filter program to 'heighten' the sounds a little; That's the main reason, I think, that they use the .raw format as an in-between. I've been skipping this step, as it doesn't seem critical, in which case the direct conversion is probably preferable. I suggest, that if you are converting sounds for Asterisk's sake, that you convert to all the possible formats. Disk space is cheap, and you'll squeeze a little extra performance out of Asterisk by allowing it to pick the 'best' format. Dahdi type interfaces would prefer the ulaw/alaw formats; High-def phones like Snom (and appropriate Polycoms, etc) could use the g722. Ulaw and gsm transcodings are cheap, but no transcoding is cheaper still. murf Steve Thanks for sharing I appericate your insight as this is something I run up against as well. What about g729 we use this coded a lot what is the best method to transcode it it? Thanks Bryant -- _ -- 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] help with dialplan
Todd
How do you have the context in the phones sip configs set?
Bryant
From: Todd Reese trees...@gmail.com
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.
[globals]
QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17
ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27
GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37
EXTERNPHONE0=SIP/150
CPHONE1=SIP/1678000
CPHONE2=SIP/177
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133
[from-pstn]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn1]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn2]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn3]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn4]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming3,s,1)
[from-pstn5]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming2,s,1)
[from-pstn6]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn7]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn8]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[incoming1]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${QPHONE0}${QPHONE1}${QPHONE2}${QPHONE3}${QPHONE4}${QPHONE5}$
{QPHONE6}${QPHONE7},40,Ttr)
exten = s,n,Hangup
[incoming2]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${ACAPHONE0}${ACAPHONE1}${ACAPHONE2}${ACAPHONE3}${ACAPHONE4}${
ACAPHONE5}${ACAPHONE6}${ACAPHONE7},40,TTr)
exten = s,n,Hangup
[incoming3]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =
s,n,Dial(${GMNETPHONE0}${GMNETPHONE1}${GMNETPHONE2}${GMNETPHONE3}${GMNET
PHONE4}${GMNETPHONE5}${GMNETPHONE6}${GMNETPHONE7},40,Ttr)
exten = s,n,Hangup
[from-interal]
include = dialout1
include = dialout2
include = dialout3
include = parkedcalls
include = intercom
exten = 10,1,Macro(oneline,${QPHONE0})
exten = 11,1,Macro(oneline,${QPHONE1})
exten = 12,1,Macro(oneline,${QPHONE2})
exten = 13,1,Macro(oneline,${QPHONE3})
exten = 14,1,Macro(oneline,${QPHONE4})
exten = 15,1,Macro(oneline,${QPHONE5})
exten = 16,1,Macro(oneline,${QPHONE6})
exten = 17,1,Macro(oneline,${QPHONE7})
exten = 20,1,Macro(oneline,${ACAPHONE0})
exten = 21,1,Macro(oneline,${ACAPHONE1})
exten = 22,1,Macro(oneline,${ACAPHONE2})
exten = 23,1,Macro(oneline,${ACAPHONE3})
exten = 24,1,Macro(oneline,${ACAPHONE4})
exten = 25,1,Macro(oneline,${ACAPHONE5})
exten = 26,1,Macro(oneline,${ACAPHONE6})
exten = 27,1,Macro(oneline,${ACAPHONE7})
exten = 30,1,Macro(oneline,${GMNETPHONE0})
exten = 31,1,Macro(oneline,${GMNETPHONE1})
exten = 32,1,Macro(oneline,${GMNETPHONE2})
exten = 33,1,Macro(oneline,${GMNETPHONE3})
exten = 34,1,Macro(oneline,${GMNETPHONE4})
exten = 35,1,Macro(oneline,${GMNETPHONE5})
exten = 36,1,Macro(oneline,${GMNETPHONE6})
exten = 37,1,Macro(oneline,${GMNETPHONE7})
exten = 40,1,Macro(oneline,${QPHONE0})
exten = 41,1,Macro(oneline,${QPHONE1})
exten = 42,1,Macro(oneline,${QPHONE2})
exten = 43,1,Macro(oneline,${QPHONE3})
exten = 44,1,Macro(oneline,${QPHONE4})
exten = 45,1,Macro(oneline,${QPHONE5})
exten = 46,1,Macro(oneline,${QPHONE6})
exten = 47,1,Macro(oneline,${QPHONE7})
exten = 150,1,Macro(oneline,${EXTERNPHONE0})
[macro-oneline]
exten = s,1,Set(CHANNEL(musicclass)=default)
exten = s,n,Dial(${ARG1},20,Ttr)
exten = s,n,Voicemail(${MACRO_EXTEN})
exten = s,n,Hangup
exten = s,102,Voicemail(${MACRO_EXTEN})
exten =
Re: [asterisk-users] help with dialplan
Todd
Your context must be set to where you want your extension to start each
time it dials out. Without getting into your dialplan code too much try
changing the context to point to dialout1
context=dialout1
If dialout1 is working you should be able to dial.
The best way to handle this is to create a context that when you dial from
your phones it decieds if you have dialed an extension or an external
number and then routes the call correclty. This way you can pickup an
extension and dial either and get the desired results.
Bryant
From: Todd Reese trees...@gmail.com
Sent: Monday, August 30, 2010 11:20 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] help with dialplan
Here is the sip.conf portion for extension 150
[150]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234567890
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/150
context=from-trunk
canreinvite=no
callgroup=
callerid=device 150
accountcode=
call-limit=50
On 8/30/2010 10:37 AM, Bryant Zimmerman wrote: Todd
How do you have the context in the phones sip configs set?
Bryant
From: Todd Reese trees...@gmail.com
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.
[globals]
QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17
ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27
GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37
EXTERNPHONE0=SIP/150
CPHONE1=SIP/1678000
CPHONE2=SIP/177
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133
[from-pstn]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn1]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn2]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn3]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn4]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming3,s,1)
[from-pstn5]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming2,s,1)
[from-pstn6]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn7]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn8]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[incoming1]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${QPHONE0}${QPHONE1}${QPHONE2}${QPHONE3}${QPHONE4}${QPHONE5}$
{QPHONE6}${QPHONE7},40,Ttr)
exten = s,n,Hangup
[incoming2]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${ACAPHONE0}${ACAPHONE1}${ACAPHONE2}${ACAPHONE3}${ACAPHONE4}${
ACAPHONE5}${ACAPHONE6}${ACAPHONE7},40,TTr)
exten = s,n,Hangup
[incoming3]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =
s,n,Dial(${GMNETPHONE0}${GMNETPHONE1}${GMNETPHONE2}${GMNETPHONE3}${GMNET
PHONE4}${GMNETPHONE5}${GMNETPHONE6}${GMNETPHONE7},40,Ttr)
exten = s,n,Hangup
[from-interal]
include = dialout1
include = dialout2
include = dialout3
include = parkedcalls
include = intercom
exten = 10,1,Macro(oneline,${QPHONE0})
exten = 11,1,Macro(oneline,${QPHONE1})
exten = 12,1,Macro(oneline,${QPHONE2})
exten = 13,1,Macro(oneline,${QPHONE3})
exten = 14,1,Macro(oneline,${QPHONE4})
exten = 15,1,Macro(oneline,${QPHONE5})
exten = 16,1,Macro(oneline,${QPHONE6})
exten = 17,1,Macro(oneline,${QPHONE7})
exten = 20,1,Macro(oneline,${ACAPHONE0})
exten = 21,1,Macro(oneline,${ACAPHONE1})
exten = 22,1,Macro(oneline,${ACAPHONE2
Re: [asterisk-users] How to tell if there is a transfer from CDR?
On blind transfers I believe the two cdr's have the same unique id . On
attended transfers there is no real way I have found to address this issue.
CDR's with transfers really suck the way they are right now. On blind transfers
you can do some flagging of the second CDR by checking in your dialing contexts
to confirm it is a blind transfer ${BLINDTRANSFER}. On attended transfers you
are just out of luck. You have to sort them out with CDR's. This cost us some
money with inbound toll free calls because we did not know this occurred this
way for some time.
Bryant
From: C F shma...@gmail.com
Sent: Saturday, September 04, 2010 10:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to tell if there is a transfer from CDR?
Last time I analyzed this (I believe back in 1.2) there was no way of
telling. However a blind transfered call would generate 2 CDR
recoreds:
1. For the part of the call with the transferrer and transfered.
2. For the part of the call with the transferee and transfered.
The call duration for the 2nd record would include the time of the 1st
record as well. So if part one took 20 seconds and part 2 40 seconds,
then the 2nd record would have 60 seconds as billable.
The only workaround was to check the BLINDTRANSFER var and reset cdr
if it was populated.
Please members of this list, I would love to hear more input as I'm
sure this has changed. Also I would not be surprised that I'm wrong in
my analysis as more than 4 years has passed since and I might have
forgotten.
TIA
On Fri, Sep 3, 2010 at 5:06 PM, Carlos Chavez cur...@telecomabmex.com wrote:
Is there any way to know if a call was transferred from reading the
CDR? Any relation in fields like UNIQUEID? Something that can be
scripted to make a special report?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Re: [asterisk-users] How to tell if there is a transfer from CDR?
Nic How stable is 1.8 really? It sounds like you are running it in production is this the case? CDR Transfer issues and rfc2833 DTMF issues are hitting us hard with 1.6.2.x. We want to move as soon as 1.8 is stable enough. Thanks Bryant From: Nic Colledge n...@njcolledge.net Hi, I use CEL or Call Event Logging in 1.8 to get a more concise picture of what happened in a call. We use it for a bunch of stuff including billing attended and unattended transfers differently. If you are thinking of upgrading, it's worth a try. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: 05 September 2010 03:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to tell if there is a transfer from CDR? Last time I analyzed this (I believe back in 1.2) there was no way of telling. However a blind transfered call would generate 2 CDR recoreds: 1. For the part of the call with the transferrer and transfered. 2. For the part of the call with the transferee and transfered. The call duration for the 2nd record would include the time of the 1st record as well. So if part one took 20 seconds and part 2 40 seconds, then the 2nd record would have 60 seconds as billable. The only workaround was to check the BLINDTRANSFER var and reset cdr if it was populated. Please members of this list, I would love to hear more input as I'm sure this has changed. Also I would not be surprised that I'm wrong in my analysis as more than 4 years has passed since and I might have forgotten. TIA On Fri, Sep 3, 2010 at 5:06 PM, Carlos Chavez cur...@telecomabmex.com wrote: Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 -- _ -- 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] Losing first DTMF digit (with ASR)
Richard Who is the carrier that the calls are flowing in from? Bruamt From: ken...@gnat.com (Richard Kenner) I'm having a wierd problem. Somewhere around 1-2% of the time, the first DTMF digit dialed gets dropped. This is occurring during a SpeechBackground application call. If the caller reenters the digits when given a second chance, all is OK. Any suggestions how to debug this intermittent 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
Re: [asterisk-users] Losing first DTMF digit (with ASR)
I have seen simalar issues with some audiocodes gear. I adjusted the early media options on the pri in the audiocodes gateway and that fixed my issues. We have also seen this when calls come in from Level 3 toll free some times their gatways screw with things. We found adding an manual answer and then a few seconds before the prompts solve the issue. Somthing about how Level 3 injects rfc dtmf and some packet time stamp issues. Bryant From: ken...@gnat.com (Richard Kenner) Sent: Tuesday, September 07, 2010 3:56 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Losing first DTMF digit (with ASR) Who is the carrier that the calls are flowing in from? It's a Paetec PRI into an NEC SV8300, then QSIG from there to Asterisk. -- _ -- 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] Issues with in-call DTMF using Broadvox and Level 3
The issue we are having is that in-call RFC2833 DTMF digits are being dropped with Broadvox and Level 3. This is happening with Grandstream GXP and Snom phones. We did some testing with the vendors and here is one of the responses we got back. Is there any way to force asterisk to modify the DTMF so that these phones will work with the carriers at issue. This is a big compatibility issue with SONUS. Hello, We have reviewed both captures, one of a good call with proper DTMF passing and one with DTMF failing to send properly. The issue still remains the same. On the good call, the time between the last RTP packet and the first DTMF event was less than 2ms. For the bad call, the time was over 200ms. Also, to clarify, the 100ms requirement is not per RFC but a standard that our switch vendor has put in front of us in order to guarantee proper DTMF passing. We have had them troubleshoot this in the past, but unfortunately it is something they cannot rectify on their end. The next course of action now is to see what can be done to work around this issue. Here are the following options: 1. Get the time between the packets down to ~100ms or lower. 2. Send DTMF via SIP INFO 3. Send DTMF via Inband. Please advise our NOC how you would like to proceed. Thank you, -- _ -- 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] Force ip disconnect after register?
Is there a way to drop a ip connection to asterisk after a number of register attempts. I have been having issues with hackers doing registration scanning against our server. We block their address at the fire wall but since asterisk does not force a drop of the connect after so many bad reg attempts I can't enforce the block until they drop and try again. This allows them to run the box with reg attempts as long as they maintain their initial connection or I reset the state tables on the firewall. This is very bad. Is there a way to force the connection to drop and reconnect after let's say 50 attempts. Thanks for any input. Bryant -- _ -- 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] High volume BLF - Suggestions?
Steve Grandstream has a new services GXP-21XX coming out they may work for your. We have been a beta tester and the BLF on these seem to work much better then the GXP-20XX units. I do not have the side cars in stock right now so I don't know how they work with it but you can put at least two for about 112 addtional blf keys. Bryant From: Olivier oza_4...@yahoo.fr Sent: Monday, September 13, 2010 6:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] High volume BLF - Suggestions? 2010/9/13 Steve Davies davies...@gmail.com Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite low-end hardware, but we cannot find any handsets that can cope with it longer term. Our test involves about 10 BLF-NOTIFY messages per second to each handset with a 5-second pause every 5 seconds. This will either crash or render unusable all of the following combinations: snom360 + 1 x sidecar As Snom phones have a parameter to express a time period during which BLF's SUBSCRIBE messages are spread and sent to Asterisk, I thought Snom phones would handle this load more easily. Yealink T28 + 1 x sidecar Yealink T28 + 2 x sidecar Cisco SPA504g + 1 x sidecar Cisco SPA504g + 2 x sidecar Cisco SPA525g + 1 x sidecar (reboots often) Cisco SPA525g + 2 x sidecar (reboots quickly) Aastra 55i + non-LCD sidecar Did not try Polycom as they do not do directed pickup and only small sidecars. Linksys SPA962 with one sidecar is OK but is discontinued hardware. Help? Thanks, Steve -- -- _ -- 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] PostgreSQL is asterisk friendly with it?
As I look to move our systems to version 1.8 I am looking at making a change from mySQL to PostgreSQL. I love mySQL but am getting very concerned about i'ts new owners. Should I be able to move all my realtime stuff to PostgreSQL is it fully supported with asterisk? Is there any down side to PostgreSQL over mySQL or will it be a big win? Our database servers are linux but we access them from asterisk as well as windows are there any thing to be concerned with there? I use c#, vb.net and mono to do a lot of our stuff are there any issues I should know about? Thanks Bryant -- _ -- 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 1.8 and CEL logging
Is there the ability in the Asterisk 1.8 CEL logging to log the SIP endpoint IP as weell as the medie enpoint's ID's? Thanks Bryant -- _ -- 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 and Digium TC400B
We can resell the Sangoma card. They have some higher license counts as well. They are also offering a step up offering. If you buy at one level and need to move to the next. They will offer you a trade back on the old card. Bryant From: Tim Nelson tnel...@rockbochs.com Sent: Thursday, September 23, 2010 10:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk and Digium TC400B - Tarek Sawah tareksa...@hotmail.com wrote: Greetings, Because of the heavy load and the high expectations of an asterisk server offered as a solution integrated with our CRM software.. we were looking into other possibilities than software Licenses for G729 and G723 codecs.. to lower the pressure on the processor giving it more space to do more work. We heard of a hardware (PCI CARDS) can be used with Asterisk that does the work. And we stumbled with Digium TC400B. Could be a newbie's question.. but does that serve our needs? As we have not pressured a server before up to 1400 extensions with 600 outbound SIP calls (customer's needs). The server in question is Core I7 16 GB ram and Raid 10 SAS drives. We need to know how many calls with G729 or G723 can this server handle? And as far as we can see this Digium card can be a cheaper solution If calculating the CPU cost plus the licenses for each channel. One more question.. can we add two of those cards to the server? Will it be efficient? Sangoma also has a transcoding card: http://sangoma.com/products/hardware_products/transcoding.html My understanding of both the Digium and Sangoma offerings is that you can use multiple cards in your system. --Tim -- _ -- 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] Asterisk and Digium TC400B
On 09/23/2010 06:48 AM, Tarek Sawah wrote: Greetings, Because of the heavy load and the high expectations of an asterisk server offered as a solution integrated with our CRM software.. we were looking into other possibilities than software Licenses for G729 and G723 codecs.. to lower the pressure on the processor giving it more space to do more work. We heard of a hardware (PCI CARDS) can be used with Asterisk that does the work. And we stumbled with Digium TC400B. Could be a newbie's question.. but does that serve our needs? As we have not pressured a server before up to 1400 extensions with 600 outbound SIP calls (customer's needs). The server in question is Core I7 16 GB ram and Raid 10 SAS drives. We need to know how many calls with G729 or G723 can this server handle? And as far as we can see this Digium card can be a cheaper solution If calculating the CPU cost plus the licenses for each channel. One more question.. can we add two of those cards to the server? Will it be efficient? Hi Tarek, I have TC400B cards installed and they work fine. You get up to 120 channels per card. You can install multiple cards and they work good. The new sangoma G729 cards have the ability to do up to 2400 channels per card depending on the configuration purchased. The sangoma option is really good option once you get the 120 channel level. The real question is how many channels do you need to transcode. In certian combinations asterisk can eat g729 channels. Let's say you are coming g729 and recording and then going back out g729 you may eat up to 4 encode/decode license. For some calls if your end points are g729 and your carrier is g729 you many not need any license. It comes down to how many of your calls will need access to non g729 prompts and how many calls will need to be converted due to differing source formats. If you can convert your ivr prompts to g729 you get a win here. But playing voicemail files will never use g729 as it is not currently a supported record format as far as I have found My guss is one of the sangoma 400 license cards would likely meet your needs That will range between $2200 and $2300 A single sangoma 240 license card is between $1550 and $1650 The digium TC400B selles for beteen $1025 and $1200.00 If you want more details you can contact me off list. Bryant -- _ -- 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] Asterisk 1.6.2.13 Audio Prompts Stopping
Version 1.6.2.13 is having issues with audio prompts dieing. When users call in to get voicemail the prompts start and then stop about 6 to 10 seconds in. On hold music plays for 6 to 10 seconds and then stops. In meet me conference rooms hold music will stop about 6 to 10 seconds in. Audio playback in IVR's start to play and then stops. It happes with both g729 and g711 calls. This does not happen on every call but more then 50%. This is a big issue any ideas? I need to fix this ASAP. Thanks Bryan -- _ -- 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] RTP Read too short
Hi All In the console I am seeing warring rtp.c:1635 ast_rtp_read: RTP Read too short I get these all of the time things seem to be working fine but I am trying to figure out if there is a way to resolve these Warnings. I am running asterisk 1.6.2.13 Any direction is appreciated. Thanks Bryant -- _ -- 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] Weird stalling of playback on IAX2 channels on 1.8 svn
Tim I am actually seeing this on a 1.6.2.13 box as well. For some reason durring prompt playbacks they some times stall mid file. The call stays up but no audio comes in. Bryant From: Tim Panton t...@westhawk.co.uk Sent: Friday, October 08, 2010 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Weird stalling of playback on IAX2 channels on 1.8 svn I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- 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] Weird stalling of playback on IAX2 channels on 1.8svn
On 8 Oct 2010, at 15:37, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton Sent: Friday, October 08, 2010 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Weird stalling of playback on IAX2 channels on 1.8svn I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk Are both Asterisk's 1.8? I had unhappy results doing IAX between 1.4 and 1.6 (1.8 is built on 1.6???) The far end is our voip supplier's asterisk - no Idea what version. But it also happens when talking to our Java IAX stack which isn't asterisk at all, so it isn't specific to a particular asterisk :-) What's more, if a call makes it past the announcement and gets bridged, it works fine. I've had several half hour calls through it. So it seems to me that it is an interaction between playback and iax2. T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk Tim my issues with 1.6.13 have been on sip Bryant -- _ -- 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] looking for a better ATA
I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF talkoff and T38 Any advise before I start testing with these brands would be apperciated. Any better option you may know of. Thanks for any input Bryant -- _ -- 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] looking for a better ATA
Us too. Tons of SPA2102's out there working fine! On Fri, Oct 8, 2010 at 4:36 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Fri, 8 Oct 2010, Bryant Zimmerman wrote: I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF talkoff and T38 Any advise before I start testing with these brands would be apperciated. Any better option you may know of. Thanks for any input Bryant I'm curious which of the above ills you attribute to the Linksys (assuming an SPA2102? The PAP2T does have the T38 problem I believe). Its basically the defacto standard for all the giant ITSPs. Perhaps your problem is one that could be rectified in some way. I have also tried Grandstream and Audiocodes (still use the MP-124s in certain situations) and have found that the SPA2102s work the best for us... Cheers, j Jayson The big issue with me and the SPA2102 is the DTMF talk off. Thanks Bryant -- _ -- 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] GXP-21XX
Anyone used the new Grandstream GXP-21XX series phones. We have been testing these phones and like what we see. We are looking for a greater cross section of testing before we roll them to production. Any feed back would be appreciated. We are talking with Grandstream engineering and they are looking for feed back as well. Any input is appreciated. Thanks Bryant -- _ -- 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] innomedia ATA's
We are testing the innomedia ATA's to possibly replace our current line up of ATA's that we are using. Has anyone used their product? What is their track record on stability, voice quality, DTMF talkoff, T.38 Thanks Bryant From: Zeeshan Zakaria zisha...@gmail.com Sent: Wednesday, October 13, 2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DMTF Mode I would suggest first to make sure that asterisk is receiving DTMF fine from your IP devices/phones. Do you have a test IVR where you can dial and press digits and verify that asterisk is responding? Once you are sure that asterisk is receiving DTMF fine, then you should ask your provider what DTMF setting you should have on your system. Usually all of them support RFC2833, so if in your sip.conf where you have defined the trunk, dtmfmode is set to rfc2833, your provider should receive it and pass on to the next carrier or trunk. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-13 10:19 AM, Dan Journo d...@keshercommunications.com wrote: It depends upon whether you are receiving DTMF or sending, and whether you are using a VoIP protoc... Sorry about the lack of info. It's a simple SIP only setup. A handful of sip phones, an asterisk server, and a sip provider. The DTMF signals from the sip phones are received by Asterisk because they can access features like *1. The DTMF signal from the called party are received by Asterisk because they can also access features like *1. But, the DTMF tones are not passed through from the Sip Phone to the Called Party. The same happens regardless of whether its an incoming or outgoing call. That means, if any of my users try to call a company with a menu system, they can't select any options. How can I tell if Asterisk is sending the tones through to the provider? I need to find out whether its something I'm doing, or something the provider is doing. Any ideas? Thanks Dan -- _ -- 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] GXP-21XX
Gordon Thanks for the reply. Grandstream has three new phones that will replace the GXP-20XX series as some point. GXP-2000 - GXP-2100, GXP-2010 - GXP-2110, GXP-2020 - GXP-2120. The GXP-2110 has been released the others appear to be on the cusp of release. We have been testing the GXP-2110 for several months now and are looking to see if there is any one else that has used them in production since their release last month. Are there any other early adopters out there? Based on your reply you have used several of the new GXP-2110's with operators. Have you had any issues with screen display issues. What version of the firmware are you on with them. Thanks Bryant From: Gordon Henderson gordon+aster...@drogon.net Sent: Wednesday, October 13, 2010 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] GXP-21XX On Wed, 13 Oct 2010, Bryant Zimmerman wrote: Anyone used the new Grandstream GXP-21XX series phones. We have been testing these phones and like what we see. We are looking for a greater cross section of testing before we roll them to production. Any feed back would be appreciated. We are talking with Grandstream engineering and they are looking for feed back as well. Any input is appreciated. That's the replacement for the GXP2000 - which I've deployed a great many of. Only deployed a small number of GXP2110s as reception console phones though and I've not had issues. Grandstream seem to suffer from buggy early software though, so do check their releases and when you find a stable version - stick to it - although I have to say, all the GXP2000 releases over the past couple of years have been stable, so maybe they're learning :) Gordon -- _ -- 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] checking CDR
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias Sent: Wednesday, October 13, 2010 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] checking CDR Hello Asterisk Community, Is there a way to check in asterisk cdrs and extension forwarded? I mean, i'm calling to a ISDN number, wich goes to extension 8222, but this extension is forwarded to another one, the problem is that in CDRs i am able to see the the first step of the call, but never see the forwarded extension, how can i do that? Thanks! The CDR is only going to record all legs on incoming calls. As you state above, your outgoing call is going to show as one leg regardless of how many bounces it takes. The way I have addressed this issue is using flag variables that determine how the call has originated. Inbound calls set one state and outbounds calling checks for that state if it exists we assume that it is either a call forward or a transfer. We then check headers and variables to see what state it is. We then forward the outbound call through a call to LOCAL/customeroutbund/number~trackingvars. This will cause the system to create a sperate channel leg for that part of the call. We have found it to very tricky to get this right for both blind and attended transfers as well as call forwards but you can get very close. We were loosing on 100% of our transfers and forwards and now we are down to 3%-5% of the cases where our method does not work. Or 100% method is to use an additional asterisk box that routes all toll bearing inbound and outbound calls we disable forwards and transfers there. That is where we bill from so we don't loose and $$$. Asterisk 1.8 is looking good with the CEL logging but you have to sift the records to create billing CDR's Bryant -- _ -- 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] checking CDR
The real question is are you having the phone forward the calls or is your dial plan redirecting to outbound calling? Bryant From: Zeeshan Zakaria zisha...@gmail.com Sent: Wednesday, October 13, 2010 2:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] checking CDR Hi, (Following is for asterisk 1.4) For the forwarded calls, you should see two entries in the cdr, and this is because a forwarded call is actually two separate calls. You have to look in the channel and dstchannel fields of the cdr to match the call ids of the calls to figure out which calls were forwarded. Incoming call's channel value and outgoing call's dstchannel value will be the same, except a comma and digit at the end, showing if it was the first call on that id, second, third or more. I have programmed two billing systems, and this is how I catch forwarded calls and bill them, works perfectly fine. Though it is confusing. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-13 1:21 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello Asterisk Community, Is there a way to check in asterisk cdrs and extension forwarded? I mean, i'm calling to a ISDN number, wich goes to extension 8222, but this extension is forwarded to another one, the problem is that in CDRs i am able to see the the first step of the call, but never see the forwarded extension, how can i do that? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Audiocodes firmware
From: Paul Belanger paul.belan...@polybeacon.com Sent: Thursday, October 14, 2010 5:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Audiocodes firmware On Thu, Oct 14, 2010 at 3:25 PM, Mark Murawski markm-li...@intellasoft.net wrote: Does anyone have links to the most recent audiocodes firmware? Why not contact Audiocodes? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- You have to normally get the Audiocodes firmware from your reseller, or you have to buy a support contract on the device to get current firmware. Audiocodes for some reason does not offer simple just download the current version and install it as an option. They have stated that too many people tend to mess up firmware upgrades so they want you to have the support contract from them or your resellar. It is really hard to select a bin file and hit update without shutting off your device until it's done. $$$ -- _ -- 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] Audiocodes firmware
We are being forced to move away from audiocodes ATA's because they refuse to fix a few minor bugs unless we commit to a 1000 piece order. This is on their 2 port ATA's. Their response to us is that ATA's are intended for serious carriers that are using them in conjunction with their higher end gateways. And we use their PRI gateways and a few of their 4 and 8 port gateways but we can't user their 2 ports. From: Paul Belanger paul.belan...@polybeacon.com Sent: Thursday, October 14, 2010 6:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Audiocodes firmware On Thu, Oct 14, 2010 at 5:27 PM, Mark Murawski markm-li...@intellasoft.net wrote: Because audiocodes does not provide support to end users and will tell you to contact your vendor that sold you the box. That is ridiculous, how hard is it to provide a download link and disclaimer about no support. Unless Audiocodec's simply wants to charge you more money. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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 -- _ -- 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] fraud advice (Also advice on using ipbanning)
When we designed our systems on asterisk we designed it to me multi-tenant. Se we use customer prefixes on all extensions. This allows us to have multiple customers using the same extension pools. It also reduces the hack foot print as hackers must know the prefix for a customer to try and brute force things. All passwords use 8+ characters with alfa/numeric and special characters. As I see it Asterisk does very good keeping out the hackers if you use a solid design in your peer and dialplans. At the least put an alpha character post or pre other wise you are just asking for it. Use your head you can be smarter then they are. We are looking into ipban as well. If any one has an example of ipban I would love to see how best to implement it. In a 4 year period we have not had a breach but we do get about 10 to 15 hack attempts a week. We have blocking scripts that block ip's at the primary firewall but I would like to trigger the ipban at each switch level. Could I also use the ipban method to trigger the audo updates to our primary firewalls? Any advice is appreciated. Bryant From: Steve Totaro stot...@totarotechnologies.com Sent: Friday, October 15, 2010 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] fraud advice On Fri, Oct 15, 2010 at 10:29 AM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 14 Oct 2010, bruce bruce wrote: But it also sickens me at how badly Asterisk is made to not cope with situations like this and worse than that is FreePBX. Kind of like blaming the gun manufacturer instead of the criminal with their finger on the trigger? Is there some gaping hole in Asterisk security or are you just asleep at the wheel? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 This is nothing new. Trunk to trunk transfers and other exploits could be used on old school phone systems to do the same thing. I would start with getting the current balance, if over $10k call the FBI, call them anyways, it couldn't hurt. You want the Feds to check things out before local police if possible. Gather as much info as possible, along with police and FBI case numbers and then call the carrier and see what can be done. A friend of mine took what was supposed to be my one month rotation to Iraq. I had too much going on to be in Iraq for a month and a half and had taken the last rotation so it wasn't even my turn. The phone bill came for his cell (company provided on Asia Cell) for $4k in just a couple weeks. It turns out that he was not using the cell and one of the cleaning people stole his SIM. After contacting Asia Cell a few times about the matter, they credited the whole amount back. So you never know. As for security, I assume you need to allow these extensions to register from outside the LAN? If not, then only allow them to register via a LAN IP, I would do it with iptables, only allow the provider IP through. I am curious what your user:pass was? something like 1000:1000, I see many systems setup like this and am surprised they haven't been hit yet. In the future, you could use a scheme that makes it much more secure and also pretty easy to maintain. The username could be the MAC and the pass could be the serial number or asset tags if you use them. I know there must be dozens of people reading this that have had the same issue but are embarrassed to speak up. (BTW Sierra Leone is in West Africa, not the Middle East.) Thanks, Steve T -- _ -- 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] Asterisk to switch on electric heaters remotely?
I would look at x10 triggered switches. There are some command line tools you could call from an IVR. I did a lot of x10 development on windows back in the day. I have seen some things for linux as well. http://www.heyu.org/ Bryant From: C F shma...@gmail.com Sent: Monday, October 18, 2010 7:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk to switch on electric heaters remotely? Ah Sandman http://sandman.com use a relay that goes onto an fxs port, call that fxs port and you have a connection. Since that only work momentary you will need a flip flop relay, the advantage is that by calling it again you can turn it off. Ring relay: http://sandman.com/wizard.html#UniversalRingRelay flip flop relay: http://altronix.com/index.php?pid=2model_num=RBR1224 On Mon, Oct 18, 2010 at 7:09 AM, Gilles codecompl...@free.fr wrote: Hello I'm sure someone has already tried this: I use a couple of electric heaters to heat my office. I'd like to somehow connect them to Asterisk so that I could switch them on remotely by either calling the IVR or sending an e-mail to the Asterisk host, so that the room is warm when I get to the office :-) Any information appreciated. Thank you. -- _ -- 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 -- _ -- 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] Audio Playback randomly stops
We are having issues with asterisk 1.6.2.12-rc1 and 1.6.2.13 with audio playback randomly stopping during calls. A caller goes to voice mail and the prompts stop playing back. IVR prompts stop playing in mid stream. This occurs randomly and is causing quite a problem. I do not see any errors or warring when the playback stops. It has occurred with sip endpoints running both g711 and g729. Any ideas? Bryant -- _ -- 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] Pop-up for MS Outlook 2007 recommended
Bria is a full SIP soft client. It works ok if you have a very good sound card and good wired headset. It is not a dialer application in the sense that you would dial your desk phone using it. Some of my clients love the Bria and some say the quality is poor. You must have a computer that can handle it the supporting sound and headsets. Bryant From: unsero...@aol.com Sent: Monday, October 25, 2010 3:27 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Pop-up for MS Outlook 2007 recommended Did you already check Bria? I have not tested it yet but it seems to be very powerful. Unfortunately there is no trial version available. If you will give it a try I would be very interested in your opinion. http://www.counterpath.com/bria-for-microsoft-outlook.html Oliver -Original Message- From: Bruce B bruceb...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Mon, Oct 25, 2010 9:10 pm Subject: Re: [asterisk-users] Pop-up for MS Outlook 2007 recommended Great suggestion but unfortunately for this client a proven technology is needed and we don't mind paying a bit for it so once the time is available we might do this the way you suggested. Thanks On Mon, Oct 25, 2010 at 2:20 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Monday, October 25, 2010 1:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Pop-up for MS Outlook 2007 recommended Hi Everyone, Which paid or unpaid commercial plugin is available out there for Asterisk that would do Outlook contacts pop-up that is proven to work great with MS Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well through the Outlook. Thanks, Bruce Not specifically what you are looking for, but it is very simple to use Apache/Ajax to make AMI links to launch calls from anywhere. I would invest 30-240 minutes into this method before bothering with the other stuff that is out there. Also, will make it easier when you eventually jump to 1.8/1.10. -- _ -- 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 -- _ -- 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] Migration from 1.2 to 1.8 in production
I have used 1.4 1.6. I am testing 1.8 for production and it is looking very good. I am making some changes to accommodate some minor dialplan changes from 1.6. Our 1.4 is very solid 1.6 has some issues with DTMF issues when used with Sonus on the back end. 1.8 is looking very good and we hope to go production before the end of the year. If you have to change righ now are you using custom dialplan code? If you are I would roll the dice and go for 1.8 this will give you the longest life span. If not there is no real big hit for stepping from 1.4 to 1.8. The other issue is if you want really detailed logging for call records the CEL method in 1.8 is the way to go. You will need to be able to boil the data down but it is there. I have seen a few kinks in the current version but it looks like they will be worked out with some incremental updates. Our hope is to be fully 1.8 on all of our backbone production units by the end of Jan 2011 with our first unit by December 2010. I would shy away of 1.6.x based on our experience. Our 1.6.x boxes will move before our 1.4.x boxes. Thanks Bryant From: Tilghman Lesher tles...@digium.com Sent: Wednesday, November 03, 2010 11:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Migration from 1.2 to 1.8 in production On Wednesday 03 November 2010 09:32:10 Danny Nicholas wrote: satish patel wrote: We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? I would like if you suggest me which version would be good for production since asterisk 1.8 still in beta process. 1.8 will introduce many features and is the supported standard, which will be important to you since you are on a 5 year upgrade plan. It also has more opportunities than the 1.4 version since it is under active development and 1.4 is in a patch only state. This is not the case. Both 1.8 and 1.4 are in the same state right now. The only difference in support level is that 1.4's EOL is much sooner than the EOL for 1.8. 1.6.2 will EOL at approximately the same time as 1.4. See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for the most up-to-date schedule. If immediate stability is your goal, you may want to stick with 1.4. If I were going to bite the bullet on 1.6, I'd jump straight to 1.8 since there is no end-of-life advantage. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) 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 -- _ -- 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] A few questions regarding Asterisk 1.8.0
From: Mark Scholten m...@streamservice.nl Hello, I have a few questions regarding Asterisk 1.8.0. If you can answer a question, please do so. Is Asterisk 1.8.0 stable enough for production environments? It appars to be so far we are testing and hoping to go production before the end of the year. Is it possible (and if yes what is the best option) to use CDR MySQL with Asterisk 1.8.0? With 1.6.x we use the add-on package for that, however we could do something with scripts to do it (but I don't like the idea). You can use the same MySQL method you are use to but if you want to use the new more extensive CEL method you will likely need to use ODBC to write to MySQL for now. You will also need to parse the new CEL format for the info you need. It is looking realy cool but it is taking a bit of work to intagrate it into our system. We will go live using the old CDR to MySQL for now. Please not that the addons are part of the main package now use menuselect to choose which ones you want to build. If it is stable and there is a good option for CDR with MySQL we will startusing it very soon. Good luck as with any new version there may be some bugs so if you bump up against ones report them so they can be fixed. Also don't just drop it into production with out testing it on a box for a bit. 1.8 has a lot of changes. Most appear to be for the better. Regards, Mark Regards Bryant -- _ -- 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] Issues with 1.8 and BlindTransfer
I am having issues with Blind Transfer on asterisk 1.8 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use the # to initiate the transfer everything works. But our customers are use to using the transfer key on the phone. I found several bugs out there on the bug tracker but do not see if there is any work around. Any ideas or help would be appreciated. Thanks Bryant -- _ -- 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] Issues with 1.8 and BlindTransfer
Replys from Bryant On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com wrote: I am having issues with Blind Transfer on asterisk 1.8 What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS? Verison 1.8.0, Suse 11.1 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. Does the remote party (being transferred) initially hear hold music, then the line go silent after completing the transfer? No the call just drops and nothing happens in the dial plan. Does the Grandstream show the call still on hold, but you are unable to pick it up? The call just goes a way. Are you using Realtime and/or Direct media? Not using Realtime. I don't think I am using Direct media. Our switch should be handling all of the rtp traffic It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use the # to initiate the transfer everything works. But our customers are use to using the transfer key on the phone. I found several bugs out there on the bug tracker but do not see if there is any work around. Any ideas or help would be appreciated. I have been chasing a deadlock (issue #18403) on blind transfers with 1.8SVN and have not found a work-around yet. While I can deadlock every time (Polycom and Cisco handsets), at least one other has reported different results with the Bria Softphone and Grandstream handsets. You could try a softphone and see if you get the same results as the physical phones. I have a version of Bria I can try later today. -Jonathan Bryant -- _ -- 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] Issues with 1.8 and BlindTransfer
Karsten I do not see it in the changlog for the 1.8.1 rc version. How would I get the SVN version to test? Thanks for your help. Bryant From: Karsten Wemheuer k...@gmx.de Sent: Thursday, December 02, 2010 11:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Issues with 1.8 and BlindTransfer Hi, Am Donnerstag, den 02.12.2010, 11:02 -0500 schrieb Bryant Zimmerman: Replys from Bryant On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com wrote: I am having issues with Blind Transfer on asterisk 1.8 What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS? Verison 1.8.0, Suse 11.1 There was an issue with blind transfer in 1.8.0, fixed in SVN (and maybe in 1.8.1 ?) See https://issues.asterisk.org/view.php?id=18185 HTH, Karsten -- _ -- 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] Callee side blind transfer is failing in 1.8
Nikhil Known bug. there is a patch that is in the SVN trunk. I just downloaded the trunk version last night and will be testing in a bit. I will keep you posted. Bryant From: Nikhil d.nik...@cem-solutions.net Sent: Monday, December 06, 2010 6:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Callee side blind transfer is failing in 1.8 HI callee side blind transfer is failed in 1.8 but caller side blind transfer is succes,Transfer doing by refer method,please help me on this Nikhil -- _ -- 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] Wireless Desktop VoIP Phone?
I belive the WBP54g cisco/LINKSYS adapter is what we are using with the Grandstream phones. You have to buy a Cisco/Linksys power supply but it works great. I have over 200 of them out there. Bryant From: Jeremy Betts jer...@freevoicepbx.com Sent: Friday, December 17, 2010 4:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Wireless Desktop VoIP Phone? Cisco also make a wireless adapter for the 500 series phones. On Fri, Dec 17, 2010 at 7:40 AM, Matt mhop...@gmail.com wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? -- _ -- 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] cdr_mysql stopped working
I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql table for CDR's today there are no entries since the update. I have rebuilt and re-installed and re-started asterisk still no CDR's flowing to mysql. I did not change any configs. I checked to make sure that the cdr_mysql option was selected under the make menu options. The module shows it is there when I do a modules show. I don't get any errors saying it can't write to the table. My voicemail settings are pulling from the same server. Any ideas on what I could try to fix this or how I could test to see what is causing it? Thanks Bryant -- _ -- 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] Include ${HANGUPCAUSE} in CDR
I am trying to include the ${HANGUPCAUSE} in my mySQL cdr tables. I have a
field called cause_code but it won't write. I belive it is because the
record has already been written by the time I hit the h section of the
code. How might I get this info into the CDR. I need this info for Quality
of Service as well as route checking. Any ideas would be apperciated.
Here is my dial line and my h lines. I also use the g option so if the
other party hangs up and that is not working either.
exten =
doDialStd,n,Dial(${siteDefaultOutboundTrunk}/${c_DialArg}${c_DialExten},120,
ge)
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
Bryant exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})Bryant
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Re: [asterisk-users] DIALSTATUS on CANCEL
I see the same thing. Why is there an CANCEL status if it is never set. The only way I have been able to capture a Cancel status is with the h extensions using the 'e' option under dial. But this leaves no way to tell what the DIALSTATUS state was as it is blank. I belive it is a bug as well. Bryant From: Michael voip.quest...@gmail.com Sent: Wednesday, December 22, 2010 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DIALSTATUS on CANCEL Hi Nikhil, Both debug and verbose are set to 20. That's all I got, but as you can see, for the other types of reasons, the DIALSTATUS got a value (and we see the events). I'm pretty sure it's a bug. Michael On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions.net wrote: Hi Enable debug level to more than 1 ,you may get something. Thanks Nikhil On 12/22/2010 11:26 AM, Michael wrote: Spawn extension (incoming-private, , 3) exited non-zero on 'SIP/Proxy-0031' -- _ -- 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] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
To my knowledge there is currently no free version of the g729 codec. There were some spec builds but those were just for testing if I recall correctly. Each version of the codec that we have always gotten has been compiled for each version of asterisk. I would just buy the Digium licenses for the codec and not mess with it. That way you are legal and have support if you need it. From: Joel Maslak jmas...@antelope.net Sent: Wednesday, December 22, 2010 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!) I'm going to guess you aren't going to get a lot of help on a list hosted by Digium on how to use a potentially illegal codec... That said, ast14 in the filename might signify what the problem is. The APIs likely changed for modules between 1.4 and 1.8. On Wed, Dec 22, 2010 at 7:58 AM, Giorgio Incantalupo gincantal...@fgasoftware.com wrote: pbx18*CLI module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]: loader.c:852 load_resource: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' could not be loaded. It worked on Asterisk 1.4, but not anymore on my Asterisk 1.8...why??? :( Thank you Giorgio Incantalupo -- _ -- 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 -- _ -- 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] DIALSTATUS on CANCEL
The Dial Status is not set when accessing it from the h extension. Bryant From: Vardan Harutyunyan hvarda...@gmail.com Sent: Wednesday, December 22, 2010 10:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DIALSTATUS on CANCEL Try to use h extension -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com Michael wrote: Hi Nikhil, Both debug and verbose are set to 20. That's all I got, but as you can see, for the other types of reasons, the DIALSTATUS got a value (and we see the events). I'm pretty sure it's a bug. Michael On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions.net mailto:d.nik...@cem-solutions.net wrote: Hi Enable debug level to more than 1 ,you may get something. Thanks Nikhil On 12/22/2010 11:26 AM, Michael wrote: Spawn extension (incoming-private, , 3) exited non-zero on 'SIP/Proxy-0031' -- _ -- 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 -- _ -- 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] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
Giorgio You could buy just a couple of licenses 3 to 5. It would get rid of the messages for the most part and it would give you the ability to transcode for voicemails and other items requiring transcode. The reason you are likely getting the messages is there is some kind of transcode required that it can't do and you are getting the warring. If you shut off all in the middle functions like recording, voicemail, and feature codes you may be able to get rid of them but you would also loose the functions. You will likely waste more than the $30 to $50 dollars in time and you get the option to transcode to boot. Just my 2 cents. From: Giorgio Incantalupo gincantal...@fgasoftware.com Sent: Wednesday, December 22, 2010 11:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!) Hi all, thanks for answering. You all are right but I do not really need the codec because my phones and my Voip lines are all working using g729. Asterisk is working fine without transcoding as well.the problem is my CLI is flooded with messages like: WARNING[7831] translate.c: No translator path from alaw to unknown which are quite annoying...aren't they? Should I pay to avoid a CLI message? That doesn't sound fair to me. I know I should report the problem but the fake codec seemed the faster way. Giorgio Incantalupo Giorgio Incantalupo wrote: pbx18*CLI module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]: loader.c:852 load_resource: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' could not be loaded. It worked on Asterisk 1.4, but not anymore on my Asterisk 1.8...why??? :( Thank you Giorgio Incantalupo -- _ -- 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 -- _ -- 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] Possible Bug (Include ${HANGUPCAUSE} in CDR)
Ok I can't get my CDR values to set from the h extension in either 1.6.2 or
1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may be
; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Really what I need to be able to do is in the h quickly store some values
to the CDR then.
For the write of the CDR and stopping the billing seconds. Then continue
to process some cleanup funcitons.
How can I work arround asterisk not honoring the endbeforehexten=no.
Is there some other way to achieve this?
Bryant I need the cause code stored.Really what I need to be able to do is
in the h quickly store some values to the CDR then.For the write of the CDR
and stopping the billing seconds. Then continue to process some cleanup
funcitons.How can I work arround asterisk not honoring the
endbeforehexten=no.Is there some other way to achieve this?Bryant
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Re: [asterisk-users] CDR on MySQL
What would it do if you
exten = h,1,ResetCDR(w)
exten = h,2,NoCDR()
exten = h,3,DEADAGI(get-unqiueid.php)
I have not tried it but in theory it should write the first CDR and then
kill the write of the second NO ANSWER CDR.
Let me know if it works for you as I may need to do it on some of my h
exten code as well.
Bryant
From: Ron nha...@gmail.com
Sent: Wednesday, December 22, 2010 9:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CDR on MySQL
Hi I have tried setting endbeforehexten=yes but still CDR does not get
inserted before h exten. what i tried is setting ResetCDR(w) before the
DEADAGI. Like this:
exten = h,1,ResetCDR(w)
exten = h,2,DEADAGI(get-unqiueid.php)
it seems to work but it's inserting 2 record on the CDR, one with
disposition ANSWERED and one with NO ANSWER.
any ideas? thanks again.
regards
Ron
On 12/22/2010 7:29 PM, Ishfaq Malik wrote:
On Wed, 2010-12-22 at 18:10 +0800, Ron wrote:
Hi All,
I've got this dialplan:
[macro-callout-intl]
exten = s,1,ResetCDR(w)
exten = s,2,Dial(IAX2/${ARG1}/018${OUTBOUND}||t|L(${OUTTIME}00:6000))
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s,4,Hangup(19)
exten = s-BUSY,1,NoCDR()
exten = s-BUSY,n,Playback(useris-curntly-busy)
exten = s-BUSY,n,Hangup(19)
exten = s-CONGESTION,1,NoCDR()
exten = s-CONGESTION,n,Playback(useris-curntly-busy)
exten = s-CONGESTION,n,Hangup(19)
exten = s-CHANUNAVAIL,1,NoCDR()
exten = s-CHANUNAVAIL,n,Playback(useris-curntly-unavail)
exten = s-CHANUNAVAIL,n,Hangup(19)
exten = s-NOANSWER,1,NoCDR()
exten = s-NOANSWER,n,Playback(number-not-answering)
exten = s-NOANSWER,n,Hangup(19)
;exten = s-ANSWER,1,ResetCDR(w)
;exten = s-ANSWER,n,Set(CDR(UserField)=${SIP_HEADER(From)})
;exten = s-ANSWER,n,Hangup(19)
exten = h,1,DEADAGI(get-unqiueid.php)
on the last line...i would like to get the uniqueid of the call and use
it to compute cost of the call. unfortunately with this setup, after i
hangup, it does not insert the CDR yet. so my AGI get-unqiueid.php does
not find any record. have i placed my ResetCDR(w) correctly?
thank you in advanced.
regards
Ron
Make sure you set
endbeforehexten=yes
in cdr.conf
Ish
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Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
My h extension is in the same context as my Dial commands. Here is a cut
back version of the code.
The cause_code value is never stored in the mysql databae even when set in
the h extension or the
when set in rc-ANSWER' OR doDialStd
[macro-OBD-DoOutboundDial]
exten = s,1,Macro(${ARG1})
exten = s,n,Set(CALLERID(name)=${siteDefaultCIDName})
exten = s,n,Set(CALLERID(number)=${siteDefaultCIDNumber})
exten = s,n,SipAddHeader(X-interNetGR-linetype:${gbl_ibclinetype})
exten = s,n,SipAddHeader(X-interNetGR-actlineid:${gbl_actlineid})
exten = s,n,Set(GROUP()=${siteGrpLineCount})
exten = s,n,Set(c_DialArg=${ARG2})
exten = s,n,Set(c_DialExten=${MACRO_EXTEN})
exten = s,n,GoSub(DoLineCountCheck,1)
exten = s,n,GotoIf($[${siteOverLineCount}=1]?OverLineCount,1)
exten = s,n,GosubIf($[${c_DialExten}=${siteDirSer}]?OverLineCount,1)
exten = s,n,GosubIf($[${c_DialExten}=411]?nofeature,1)
exten = s,n,GosubIf($[${siteUseE164}=1]?doDialE164,1:doDialStd,1)
exten = s,n,Goto(rc-${DIALSTATUS},1)
exten = s,n,Busy(60)
exten = s,n,Hangup()
exten = h,1,NoOp(Cause Code = ${HANGUPCAUSE})
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
exten = h,n,Goto(rc-${DIALSTATUS},1)
exten = doDialStd,1,NoOp(Calling Using No E164)
exten =
doDialStd,n,Macro(OBD-CheckOutboundNumber,${c_DialArg}${c_DialExten})
exten =
doDialStd,n,Dial(${siteDefaultOutboundTrunk}/${c_DialArg}${c_DialExten},120,
ge${siteDialOptionsPublic})
exten = doDialStd,n,Set(CDR(cause_code)=${HANGUPCAUSE})
exten = doDialStd,n,Return
exten = rc-ANSWER,1,NoOp(Do Return ANSWER)
exten = rc-ANSWER,n,Set(CDR(cause_code)=${HANGUPCAUSE})
exten = rc-ANSWER,n,Hangup()
exten = rc-BUSY,1,NoOp(Do Return BUSY)
exten = rc-BUSY,n,Busy()
exten = rc-BUSY,n,Hangup()
exten = rc-NOANSWER,1,NoOp(Do Return NOANSWER)
exten = rc-NOANSWER,n,NoOp(Cause Code = ${HANGUPCAUSE})
exten = rc-NOANSWER,n,Hangup()
Any more feed back would be appercaited.
Bryant
From: Tilghman Lesher tilgh...@meg.abyt.es
Sent: Wednesday, December 22, 2010 3:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wednesday 22 December 2010 11:42:33 Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either 1.6.2
or 1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may
be ; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Sounds like your h extension is in the wrong context. Try including some
information about where you are putting the h extension and what
includes
you're doing.
--
Tilghman
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Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
No this is just a snip of the much larger code.
The h extension is runing but no values port dial function aer being written.
If I do a Set(CDR(field)=Value) before the dial
The value is stored. See my other response for a larger snip of code.
Bryant
From: Carlos Chavez cur...@telecomabmex.com
Sent: Wednesday, December 22, 2010 2:46 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wed, 2010-12-22 at 12:42 -0500, Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either
1.6.2 or 1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may
be
; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Really what I need to be able to do is in the h quickly store some
values to the CDR then.
For the write of the CDR and stopping the billing seconds. Then
continue to process some cleanup funcitons.
How can I work arround asterisk not honoring the endbeforehexten=no.
Is there some other way to achieve this?
Bryant
I need the cause code stored.Really what I need to be able to do is in
the h quickly store some values to the CDR then.For the write of the
CDR and stopping the billing seconds. Then continue to process some
cleanup funcitons.How can I work arround asterisk not honoring the
endbeforehexten=no.Is there some other way to achieve this?Bryant
Is the CDR line your only h line? I ask because if you only have one
priority for h then you MUST have:
exten = h,1,Set(CDR(cause_code)=${HANGUPCAUSE})
This is because the dialplan will not use n for the first priority and
thus will never run.
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Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
Tilghman
This does not make any sense. In the voip-info posting for the h
extension it specifically states that to handle h while in a macro that
the macro needs an h extension. The h extension runs inside the macro
but the CDR data is not being updated correctly. Also the rc-ANSWER entry
in the macro does not update the CDR with the ${HANGUPCAUSE} either after
the far end hangs up. This is diffently inconsistent behavior here. Both
the DIAL and h extension are inside the macro so the behaivior should be
consistent. If I am understanding you correctly the only way we can get a
CDR to update after a dial is to not do any DIAL calls in a MACRO is this
what you are saying? Otherwise your logic may be flawed or we have a very
big logic bug in the Asterisk Macro system.
From: http://www.voip-info.org/wiki/view/Asterisk+h+extension
Be aware: Macros require their own h extension as they do not make use
of the calling context's h extension!
Tilghamn thanks for the feed back the back and forth here is great and
helps a lot it is giving me more ideas to test against.
Bryant
From: Tilghman Lesher tilgh...@meg.abyt.es
Sent: Thursday, December 23, 2010 12:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wednesday 22 December 2010 21:08:56 Bryant Zimmerman wrote:
My h extension is in the same context as my Dial commands. Here is a
cut back version of the code.
The cause_code value is never stored in the mysql databae even when set
in the h extension or the
when set in rc-ANSWER' OR doDialStd
[macro-OBD-DoOutboundDial]
exten = h,1,NoOp(Cause Code = ${HANGUPCAUSE})
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
exten = h,n,Goto(rc-${DIALSTATUS},1)
There's the problem. The h extension should be in whatever context is
calling the Macro, not in the Macro context itself.
--
Tilghman
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Re: [asterisk-users] DIALSTATUS on CANCEL
Vardan
I have not use AEL so it is a bit hard to follow with the formatting the
way it is but it looks like correct.
Please note the h extension only appears to run if a call is connected so
I do not know when the CANCEL would ever be set.
There may be someone else who can speak to this. It also appears thet
${DIALSTATUS} may not be set if the call is not allowed to time out or
dialed. To me it would make sense to set the inital state of the
${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but I
may be missing the point on this can anyone else speak to it?
Bryant
From: Vardan Harutyunyan hvarda...@gmail.com
Sent: Thursday, December 23, 2010 2:11 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DIALSTATUS on CANCEL
I have make test in AEL.
context fu {
_000./userN = {
Dial(SIP/${EXTEN:3...@prov);
Noop(${DIALSTATUS});
};
h = {
Noop(${DIALSTATUS});
};
};
And look CLI
-- Executing [00018185402...@fu:1] NoOp(SIP/userN-b6317738, )
in new stack
-- Executing [00018185402...@fo:2] Dial(SIP/user3-b6317738,
SIP/18185402...@prov) in new stack
-- Called 18185402...@prov
-- SIP/Prov-082a83b8 is making progress passing it to
SIP/userN-b6317738
== Spawn extension (fu, 00018185402020, 2) exited non-zero on
'SIP/user3-b6317738'
-- Executing [...@fu:1] NoOp(SIP/userN-b6317738, CANCEL) in new stack
I think, I am right
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Bryant Zimmerman wrote:
The Dial Status is not set when accessing it from the h extension.
Bryant
*From*: Vardan Harutyunyan hvarda...@gmail.com
*Sent*: Wednesday, December 22, 2010 10:39 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
Try to use h extension
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Michael wrote:
Hi Nikhil,
Both debug and verbose are set to 20. That's all I got, but as you can
see, for the other types of reasons, the DIALSTATUS got a value (and we
see the events). I'm pretty sure it's a bug.
Michael
On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions.net
mailto:d.nik...@cem-solutions.net wrote:
Hi
Enable debug level to more than 1 ,you may get something.
Thanks
Nikhil
On 12/22/2010 11:26 AM, Michael wrote:
Spawn extension (incoming-private, , 3) exited non-zero
on 'SIP/Proxy-0031'
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Re: [asterisk-users] cdr_mysql stopped working
David I got the svn trunk again and did a make clean and rebuilt the install and all started to work again. My guess is that it looks like the mysql client code was out of sync with the server version. All is good again. Bryant From: David Backeberg dbackeb...@gmail.com Sent: Thursday, December 23, 2010 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] cdr_mysql stopped working On Mon, Dec 20, 2010 at 5:02 PM, Bryant Zimmerman brya...@zktech.com wrote: I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql table for CDR's today there are no entries since the update. I have rebuilt and re-installed and re-started asterisk still no CDR's flowing to mysql. I did not change any configs. I checked to make sure that the cdr_mysql option was selected under the make menu options. The module shows it is there when I do a modules show. I don't get any errors saying it can't write to the table. My voicemail settings are pulling from the same server. Any ideas on what I could try to fix this or how I could test to see what is causing it? Rebooting is a good clue. You could check your firewall settings. Firewalls can stop mysql connections. Try manually connecting to the mysql server from the asterisk system and see what happens. -- _ -- 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] cdr_mysql stopped working
Jose Thanks for your response. It appears that the issue was that the mysql client drivers were updated when I installed some mono updates and I had to recompile asterisk the system was actually writing completely blank entries for every call. Once asterisk was compiled using the newer mysql client lib things started to work again. The moral of the story is if you update anything on the box that may change mysql at all you should do a complete make clean and recompile. Bryant From: Jose P. Espinal j...@slackware-es.com Sent: Thursday, December 23, 2010 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] cdr_mysql stopped working You can also enter into the CLI in order to see if you can spot any error regarding cdr_mysql, or 'duplicated value for key...' after hangin up a call. There might be a corruption in the 'cdr' table (I've seen this sometimes). You could try a 'repair table cdr' from the MySQL CLI. Note: Sometimes, corrumptions in myISAM tables not always produce the data to be unaccessible, but just make it impossible to insert new records. David Backeberg wrote: On Mon, Dec 20, 2010 at 5:02 PM, Bryant Zimmerman brya...@zktech.com wrote: I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql table for CDR's today there are no entries since the update. I have rebuilt and re-installed and re-started asterisk still no CDR's flowing to mysql. I did not change any configs. I checked to make sure that the cdr_mysql option was selected under the make menu options. The module shows it is there when I do a modules show. I don't get any errors saying it can't write to the table. My voicemail settings are pulling from the same server. Any ideas on what I could try to fix this or how I could test to see what is causing it? Rebooting is a good clue. You could check your firewall settings. Firewalls can stop mysql connections. Try manually connecting to the mysql server from the asterisk system and see what happens. -- _ -- 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 -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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] CEL and custom values.
I am setting up CEL with asterisk 1.8 and so far so good. The issue I was hoping to address here was also being able to get storage of other values such as HANGUPCAUSE and other variables that are used for billing and quality of service. The CEL documentation starts out by saying that we can not store any other variables but then at the top of that section it says this is incorrect and that section of the documentation needs to be changed. So how can I set a variable for storage when a CEL log event is fired. I want to be able to add some additional fields to my database so when a CEL storage event is fired that the values of variables are stored to my database or CSV if the variable is set. Is there something like the CDR(field)=value but for CEL(field)=value. Any help is appreciated. Bryant -- _ -- 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] Find media and sip endpoints IP address durring h extension
How can I get the media and sip endpoints IP address durring h extension? I need to write these to my CEL logs. Any ideas? Thanks Bryant -- _ -- 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] Users of CEL Please comment on Bug
If you are using CEL in asterisk 1.8 can you please look at the issue tracker and comment. On how this might effect you. https://issues.asterisk.org/view.php?id=18559 Thanks Bryant -- _ -- 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] DIALSTATUS on CANCEL
Vandar
I know understand what you are saying here. Once I turned on CEL I was able
to see when and where each hangup was firing for each channel and the order
of operations here. I am now moving very aggressively to get to CEL as I
now see why CDR's are so broken. I have my CEL to CDR translator in testing
and this is looking very promising.
Thanks for your help.
Bryant
From: brya...@zktech.com
Sent: Friday, December 24, 2010 9:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DIALSTATUS on CANCEL
If a call is hung up before an answer our h extension is not running in
our dial macro
Bryant
On Dec 24, 2010, at 3:47 AM, Vardan Harutyunyan hvarda...@gmail.com
wrote:
Hello Bryant
Extension h is worked in any case of hangup.
It not important to that the call was answered or no.
It also be more flexible, if you use instead of ${DIALSTATUS}use
${HANGUPCAUSE}, while in most of cause ${DIALSTATUS} is returned the same
return code.
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Bryant Zimmerman wrote:
Vardan
I have not use AEL so it is a bit hard to follow with the formatting
the
way it is but it looks like correct.
Please note the h extension only appears to run if a call is
connected
so I do not know when the CANCEL would ever be set.
There may be someone else who can speak to this. It also appears thet
${DIALSTATUS} may not be set if the call is not allowed to time out or
dialed. To me it would make sense to set the inital state of the
${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but
I may be missing the point on this can anyone else speak to it?
Bryant
*From*: Vardan Harutyunyan hvarda...@gmail.com
*Sent*: Thursday, December 23, 2010 2:11 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
I have make test in AEL.
context fu {
_000./userN = {
Dial(SIP/${EXTEN:3...@prov);
Noop(${DIALSTATUS});
};
h = {
Noop(${DIALSTATUS});
};
};
And look CLI
-- Executing [00018185402...@fu:1] NoOp(SIP/userN-b6317738, )
in new stack
-- Executing [00018185402...@fo:2] Dial(SIP/user3-b6317738,
SIP/18185402...@prov) in new stack
-- Called 18185402...@prov
-- SIP/Prov-082a83b8 is making progress passing it to
SIP/userN-b6317738
== Spawn extension (fu, 00018185402020, 2) exited non-zero on
'SIP/user3-b6317738'
-- Executing [...@fu:1] NoOp(SIP/userN-b6317738, CANCEL) in new stack
I think, I am right
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Bryant Zimmerman wrote:
The Dial Status is not set when accessing it from the h extension.
Bryant
*From*: Vardan Harutyunyan hvarda...@gmail.com
*Sent*: Wednesday, December 22, 2010 10:39 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
Try to use h extension
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Michael wrote:
Hi Nikhil,
Both debug and verbose are set to 20. That's all I got, but as you
can
see, for the other types of reasons, the DIALSTATUS got a value (and
we
see the events). I'm pretty sure it's a bug.
Michael
On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions..net
mailto:d.nik...@cem-solutions.net wrote:
Hi
Enable debug level to more than 1 ,you may get something.
Thanks
Nikhil
On 12/22/2010 11:26 AM, Michael wrote:
Spawn extension (incoming-private, , 3) exited non-zero
on 'SIP/Proxy-0031'
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Re: [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)
Use a combination of ${EPOCH} with a format string and the unique call /
channel id.
Example:
exten = s,1,Set(MY_TIMEVAR=:${STRFTIME(${EPOCH},,%d%mNaVH:NaVS)})
exten = s,n,Monitor(wav,${MY_TIMEVAR}~${CHANNEL},m)
From: bilal ghayyad bilmar...@yahoo.com
Sent: Saturday, January 01, 2011 1:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Saving the monitor file on new file always using
Monitor(wav, Record1, m)
Dear List;
For each call (in specific case), I need to do a record and save in a
spearated file, so I am thinking the best thing is to save based on the
time.
Monitor(wav,Record1,m)
So, how can I make the file name to be based on the current time (which is
changed always, or based on the some unique paramter (related to the call
it self).
Any advise?
Regards
Bilal
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Re: [asterisk-users] Add Privacy: id to SIP-invite
Jonas
This is how we are doing it.
exten = s,n,SipAddHeader(P-Asserted-Identity: :${siteDefaultCIDNumber})
exten = s,n,GosubIf($[${gbl_CallPrivacy}=id]?rfc-3325-CPN,1)
exten = rfc-3325-CPN,1,NoOp(Set Call Privacy)
exten = rfc-3325-CPN,n,NoOp(From ${SIP_HEADER(From)})
exten = rfc-3325-CPN,n,NoOp(To ${SIP_HEADER(To)})
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(SIP_HEADER(From),@,2)})
exten = rfc-3325-CPN,n,GotoIf($[${l_sipheaderfromip} != ]?hasat)
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(CUT(SIP_HEADER(From),,1),:,2)})
exten = rfc-3325-CPN,n,Goto(gotip)
exten =
rfc-3325-CPN,n(hasat),Set(FROM_IP=${CUT(CUT(CUT(SIP_HEADER(From),@,2),,1),:
,1)})
exten = rfc-3325-CPN,n(gotip),NoOp(Gateway IP is ${FROM_IP})
exten =
rfc-3325-CPN,n,SIPAddHeader(P-Preferred-Identity:${CALLERID(name)}
sip:+1${CALLERID(num)}...@${from_ip}\;user=phone)
exten = rfc-3325-CPN,n,SIPAddHeader(Privacy: id)
;exten = rfc-3325-CPN,n,SetCallerPres(prohib_not_screened) ; this might
not be needed --- needs further testing
exten = rfc-3325-CPN,n,Set(CALLERPRES()=prohib_not_screened)
exten = rfc-3325-CPN,n,Set(CALLERID(num)=Anonymous)
exten = rfc-3325-CPN,n,Set(CALLERID(name)=Anonymous)
exten = rfc-3325-CPN,n,Return()
Good Luck
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003 Good LuckBryant Zimmerman (ZK Tech Inc.)616-855-1030
Ext. 2003
From: Jonas Kellens jonas.kell...@telenet.be
Sent: Wednesday, January 05, 2011 9:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] Add Privacy: id to SIP-invite
Hello list,
is it possible to add the field Privacy: id to a SIP INVITE message ?
INVITE sip:32444666...@1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/UDP1 .2.3.4:5060
From: R321113 sip:3211133...@1.2.3.4;tag=2096790244
To: sip:32444666...@1.2.3.4
Call-ID: 3b040826e909d311880a009033060...@192.168.12.40
CSeq: 34677 INVITE
Contact: sip:32444666...@1.2.3.4:5060
Allow:
REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPD
ATE
Content-Length: 203
Content-Type: application/sdp
Max-Forwards: 69
Supported: replaces,answermode,100rel
User-agent: (innovaphone IP800/6.00 sr2-hotfix16 [09-60901.35/424/110])
Privacy: id
How can I do this in the Asterisk dialplan ?? SIPAddHeader ??
Kind regards,
Jonas.
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Re: [asterisk-users] DTMF-troubles with Snom
Jonas What is the dtmf setting on all peers involved in the call? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Jonas Kellens jonas.kell...@telenet.be Sent: Wednesday, January 05, 2011 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF-troubles with Snom Hello list, I'm having DTMF-troubles with a Snom phone. I want to know if it's the Snom or Asterisk that makes the trouble. I'm playing a prompt, then make a choice for 2 : [Jan 5 17:06:38] VERBOSE[29172] file.c: [Jan 5 17:06:38] -- SIP/test1-0701 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' (language 'nl') [Jan 5 17:06:39] VERBOSE[29172] pbx.c: [Jan 5 17:06:39] -- Executing [...@sub-routing:52] WaitExten(SIP/test1-0701, 15) in new stack [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF begin '2' received on SIP/test1-0701 [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF begin ignored '2' on SIP/test1-0701 [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on SIP/test1-0701, duration 160 ms [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on SIP/test1-0701 What follows is a prompt again, and it automatically chooses option 2 : [Jan 5 17:06:41] VERBOSE[29172] file.c: [Jan 5 17:06:41] -- SIP/test1-0701 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl') [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on SIP/test1-0701, duration 160 ms [Jan 5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on SIP/test1-0701 Even without pressing 2 on the Snom phone, option 2 is chosen in the menu. The above is different when I do the same with a Grandstream device : [Jan 5 17:14:15] VERBOSE[29384] file.c: [Jan 5 17:14:15] -- SIP/test6-0714 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' ( language 'nl') [Jan 5 17:14:17] VERBOSE[29384] pbx.c: [Jan 5 17:14:17] -- Executing [...@sub-routing:52] WaitExten(SIP/test6-0714, 15) in new stack [Jan 5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan 5 17:14:18] doing dnsmgr_lookup for 'ssw4.brussels.weepee.org' [Jan 5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan 5 17:14:18] doing dnsmgr_lookup for 'ssw4.brussels.weepee.org' [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF begin '2' received on SIP/test6-0714 [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF begin ignored '2' on SIP/test6-0714 [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF end '2' received on SIP/test6-0714, duration 100 ms [Jan 5 17:14:21] DTMF[29384] channel.c: DTMF end passthrough '2' on SIP/test6-0714 [Jan 5 17:14:38] VERBOSE[29384] file.c: [Jan 5 17:14:38] -- SIP/test6-0714 Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl') [Jan 5 17:14:39] VERBOSE[29384] pbx.c: [Jan 5 17:14:39] -- Executing [...@sub-routing:52] WaitExten(SIP/test6-0714, 15) in new stack [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF begin '2' received on SIP/test6-0714 [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF begin ignored '2' on SIP/test6-0714 [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF end '2' received on SIP/test6-0714, duration 100 ms [Jan 5 17:14:44] DTMF[29384] channel.c: DTMF end passthrough '2' on SIP/test6-0714 Here I explicitly chose option 2 by pressing on button 2. What is going on with the Snom ? There is a difference in duration (160ms vs 100ms). Is that the problem ?? Kind regards, Jonas. -- _ -- 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] ReceiveFax
From: William Stillwell will...@stillwellsoft.com
Sent: Thursday, January 20, 2011 11:26 AM
This is new to me, I have a fax server using Receive Fax and gets way over 5
calls at a time. [fax-in] exten = s,1,Answer() exten = s,n,Wait(1) exten
= s,n,Set(BASEFILE=fax-${CALLERID(dnid)}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
;exten = s,n,Set(${LOCALSTATIONID}) exten =
s,n,MixMonitor(/mnt/ramdisk/${BASEFILE}.wav) exten =
s,n,ReceiveFAX(/mnt/ramdisk/${BASEFILE}.tif) exten = s,n,Hangup() exten =
h,1,System(/home/asterisk/dofax.sh ${EMAILADDRESS} ${FAXSTATUS}
${CALLERID(num)} ${snip From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, January 20, 2011 10:49 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] ReceiveFax From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Flavio Miranda
Sent: Thursday, January 20, 2011 9:00 AM
Hi all,I realize that the application Receivefax can't handle with
more than one fax at the same time. In a environment with a lot of fax, some
caller get the signal but the operation can't be completed.Is there a way
to send busy tone to the second caller?
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda My guess is no. A possible work-around would be to
set a global variable to indicate that the line is busy and to play a message
and hang-up immediately or to just hangup. Something like this: - exten
= s,1,answer - exten = s,n,AGI(checkstat.agi) - reset variable if
receivefax died or hungup - exten = s,n,Gotoif($[ ${FAXINUSE} =
YES]?byebye) - exten = s,n,Set(GLOBAL(FAXINUSE)=YES) - exten =
s,n,receivefax - exten = s,n,Set(GLOBAL(FAXINUSE)=NO) - exten =
s,n,hangup - exten = s,n(byebye),playback(im-busy) - exten =
s,n,hangup
Why can't receivefax handle more then 5 faxes at the same time? Are you using
the res_fax_spandsp.so or the res_fax_digium.so modules? It was my
understanding that the res_fax_spandsp.so did not have a limit and the
res_fax_digium.so was the commercial offering that is based on a per channel
license.
Am I wrong on the res_fax_spandsp.so module is there a limit other than
hardware performance?
Bryant
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Re: [asterisk-users] res_fax
On 01/20/2011 11:47 AM, Steve Underwood On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: On 01/19/2011 02:30 PM, Bryant Zimmerman wrote: On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. *From*: Jason Parker jpar...@digium.com *Sent*: Wednesday, January 19, 2011 3:19 PM There was a typo in the res_fax documentation. Application_SendeFax should be the correct documentation. I don't know where Application_SendFax is coming from - it's probably old. When the next import happens, Application_SendFax should be replaced by the correct version (then I'll try to remember to remove the bogus SendeFax copy). Jason thanks for the clarification on this. If I start my development with the res_fax_spandsp.so module. Should all of my code be compatible with the res_fax_digium.so module? I want to be able to get things running and tested and move to the digium supported option in the future. The choice of technology module is mostly irrelevant; that was the whole point of splitting res_fax out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. Well, people do get problems with the Digum FAX software, which go away when they switch to spandsp. Its best to test with the code you intend to deploy. Steve Steve is there any real compelling reason to res_fax_digium.so over the res_fax_spandsp.so? I was thinking Digium module was likely to be better is this wrong based on what people are seeing? -- _ -- 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 to asterisk t.38
Amit Make sure that the trunk you have between the two servers has the t.38 enabled on it. Do you have any NAT between the two servers or are they on the same lan. We do the t.38 faxing between 1.4 and 1.6 asterisk boxes all of the time. Our audio codes gateway dumps into a 1.4 box and all faxes calls are then sent to either 1.6.x or 1.8.x boxes and then on to the final ata. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Amit Nepal ami...@phoenixinternet.net Sent: Thursday, January 20, 2011 4:27 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk to asterisk t.38 Hi, I have an Audio code gateway between two asterisk servers. The audio code has PRI connected for PSTN. I can send faxes and receive faxes in ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and receive faxes. The only problem I am having is sending/receiving between ast 1.4 and ast 1.6. ATA (T.38 capable) AST 1.6 AUDIO CODEAST 1.4ATA (t.38 Capable) Thank You Amit Nepal On 1/20/2011 1:56 PM, David Backeberg wrote: On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepalami...@phoenixinternet.net wrote: I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? You don't say what's between the boxes as the medium over which the faxes are going. Try a fax between them without t.38 and see if it goes through. It might be a connection that is not reliable for any kind of faxing. That would not be an asterisk problem, it would be a faxing over a bad connection 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 -- _ -- 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] spandsp download
Where can I get the latest stable version of spandsp. That will work with res_fax_spandsp.so. The link listed on the voip-info website is dead. Any other location for download? http://www.soft-switch.org/ Thanks Bryant Zimmerman -- _ -- 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] ReceiveFAX issue.
On 01/24/2011 2:54PM Bryant Zimmerman wrote
I am testing out inbound faxing using res_fax and res_fax_spandsp.so My
system answers the call but then sets there on the ReseiveFax line then
comes back with an error that it exceeded the maximum retries. How
would I go about debugging this? Below is my very simple dialplan code I am
using, and the fax show version gives the following as well. FAX For
Asterisk Components: Applications: SVN-branch-1.8-r297535M
Spandsp FAX Driver: 20110122 075024
[fax_inbound] exten = ProcessFax,1,Answer() exten =
ProcessFax,n,Wait(2) exten =
ProcessFax,n,Set(TIFF=/var/spool/fax_in/fax_${STRFTIME(,,%Y%m%d-%H%M)}_${RAN
D(1)}.tiff) exten = ProcessFax,n,ReceiveFAX(${TIFF}
,d) exten = ProcessFax,n,NoOp(Error =
${FAXOPT(error)}) exten = ProcessFax,n,NoOp(Status =
${FAXOPT(status)}) exten = ProcessFax,n,NoOp(Header =
${FAXOPT(headerinfo)}) exten = ProcessFax,n,NoOp(
RemoteID = ${FAXOPT(remotestationid)}) exten = ProcessFax,n,NoOp(
Result = ${FAXOPT(statusstr)})
I can't figure out how to turn on any debuging for this nor can I
understand why this should be failing. I am using the spandsp version that
Steve recommended (0.0.6pre18) Any ideas or feedback would be
appreciated. Thanks Bryant Zimmerman (ZK Tech
Inc.) Ok I got a
inbound fax to work with our audiocodes pri gateway but I am having no luck
with a number on Level 3.
I did figure out that I can use fax set debug on/off and I found
that I can add the fax keyword in logging.conf to see the fax debugging.
The issue now is I am unsure how to figure out the cause of the fail. I
have attached a text file with the debug logs.
The attached file was too large so I am putting in a link to the file.
It is a virus free text file.
Fax Debug.txt
Thanks for any help.
Bryant Zimmerman (ZK Tech Inc.)
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Re: [asterisk-users] ReceiveFAX issue.
From: David Backeberg dbackeb...@gmail.com Sent: Tuesday, January 25, 2011 1:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ReceiveFAX issue. On Tue, Jan 25, 2011 at 1:45 PM, Bryant Zimmerman brya...@zktech.com wrote: Do you know how to force off T.38 in res_fax? it's in sip.conf take a look for t38pt_udptl=yes change it to no reload sip on your console that should force it to either fail entirely or do audio passthrough. Ok If I set t38pt_udptl = no on the trunk the fax comes in t.30 but I can't make t.38 work I keep getting the following error Disconnected after permitted retries Any ideas on this? Thanks Bryant -- _ -- 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] ReceiveFAX issue.
Has anyone else seen an issue with t.38 faxing on Level 3 with res_fax and res_fax_spandsp.so What we are seeing in the packet captuers is that the call is trying to do t.38 but does not appear to be completing the handshaking. No data is being transmitted. I have included a link to my pcap of this. Can anyone give me some more insight? cap-t38.pcap Thanks Bryant -- _ -- 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] Regarding error in Asterisk dail plan:
From: viswavardhanreddy karna viswavardhanre...@gmail.com Sent: Wednesday, January 26, 2011 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Regarding error in Asterisk dail plan: Hi all, i am doing my master thesis on server perfromance evaluation i am using asterisk as sip proxy server and sipp tool as traffic generator... i have run basic testing of asterisk like as shown in website http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_usi ng_SIPp when i have copied sip.conf and extensions.conf from the site and run the client and server i am getting error like this NOTICE[2715]: chan_sip.c:20314 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default' i dont know y this is coming its really troubling me a lot... please any one send me some xml, dial plan and sip.conf files for registering and for inviting. I have been trying for this a lot if any one help me i would be more thankful . BR viswavardhanreddy - viswavardhanreddy Your inbound request is not being sent with any target context or it is not matching the ip found in your sip peers. This causes the default context to trying and handle the call and you don't have anthing in it that can complete the call. The three options are 1 if you are doing registration make sure that the sending device is specifiying a context. (It does not look like you are based on your link) 2 make sure that the sending ip matches your peer account or change the peer account to friend (also change your peers to use insecure=port,invite and see if that helps) 3 add a universal handler to the [default] contect to direct the call to your test contects (exten = _.X,1,Goto(test,s,1) One of these ideas may help you if I am understanding your issue. Bryant -- _ -- 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] res_fax
Steve Are there any undocumented options available with ReceiveFAX and the res_fax_spandsp module. I am having issues with getting t.38 to negotiate with Level 3 faxes but if I force t.30 the fax comes in. But the fax does not fall back t.30 if the t.38 fails Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Wednesday, January 26, 2011 1:50 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/26/2011 12:42 PM, Bryant Zimmerman wrote: Steve Are there any undocumented options available with ReceiveFAX and the res_fax_spandsp module. I am having issues with getting t.38 to negotiate with Level 3 faxes but if I force t.30 the fax comes in. But the fax does not fall back t.30 if the t.38 fails You haven't posted any logs of the failing attempts, or packet captures of the SIP traffic, so it's pretty much impossible for anyone to help you debug this (anyone who tried would just be guessing). Steve did not write res_fax (which where SendFAX and ReceiveFAX come from), and there are no 'undocumented' options available for it, because it's open source and the source code shows all the options that are available. If you would like to try to figure out what is going on, start by posting a *complete* log file from Asterisk for a failed inbound FAX attempt, with 'core set debug 10' and 'core set verbose 10' and all logger levels (including 'fax') enabled. -- Kevin These were attached to another post. Here are the links again Fax Debug.txt cap-t38.pcap And by the way thank you for your response it is appreciated. Thanks Bryant Zimmerman (ZK Tech Inc.) -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Wednesday, January 26, 2011 2:29 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/26/2011 01:19 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Wednesday, January 26, 2011 1:50 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/26/2011 12:42 PM, Bryant Zimmerman wrote: Steve Are there any undocumented options available with ReceiveFAX and the res_fax_spandsp module. I am having issues with getting t.38 to negotiate with Level 3 faxes but if I force t.30 the fax comes in. But the fax does not fall back t.30 if the t.38 fails You haven't posted any logs of the failing attempts, or packet captures of the SIP traffic, so it's pretty much impossible for anyone to help you debug this (anyone who tried would just be guessing). Steve did not write res_fax (which where SendFAX and ReceiveFAX come from), and there are no 'undocumented' options available for it, because it's open source and the source code shows all the options that are available. If you would like to try to figure out what is going on, start by posting a *complete* log file from Asterisk for a failed inbound FAX attempt, with 'core set debug 10' and 'core set verbose 10' and all logger levels (including 'fax') enabled. -- Kevin These were attached to another post. Here are the links again Fax Debug.txt http://webmail.zktech.com/public/downloadfile.aspx?f=KERoF6PWf6e2FK8S5zgEDs 02rFGdd7zE0AIG7tjbCR9a06oFY1NwFap58zDWva3BcdOp%2b%2f%2fuBo8%3d cap-t38.pcap http://webmail.zktech.com/public/downloadfile.aspx?f=ulHIhepag5qoKm0cTUmljm T%2f7YCcOPvzlyZcnZg%2fG2B25W%2fsSr6Uwbu%2bET3kbKw84pTJjtuqrPQ%3d Unfortunately that log capture is incomplete; it doesn't include any of the messages that res_fax emits as it goes through T.38 negotiations. Please ensure that your 'console' channel in logger.conf has 'debug,verbose,warning,notice,error,fax' enabled and that you have 'core set verbose 10' and 'core set debug 10' set before the call attempt begins (or at least before ReceiveFAX is executed). If the server is only processing this particular call, then 'sip set debug on' would also be helpful. - Kevin I will get the additional debugs done when there is no other load on the fax. Is there a way for me to force t.38 off for a call but to allow t.38 for other calls. What I am thinking is if a t.38 fails to flag the next call from that number to g711 audio. This would at least let me work arround the issue for now where t.38 fails with some endpoints but not others and the g711 audio will work. The issue I am seeing is it appears that with some endpoinds on Level 3 that the t.38 tunnel comes up fine but no fax data starts flowing but this only is happening with faxes coming from some Cisco gateways sending out via PRI using t.30 Thanks Bryant -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Wednesday, January 26, 2011 4:52 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/26/2011 03:14 PM, Bryant Zimmerman wrote: Is there a way for me to force t.38 off for a call but to allow t.38 for other calls. What I am thinking is if a t.38 fails to flag the next call from that number to g711 audio. This would at least let me work arround the issue for now where t.38 fails with some endpoints but not others and the g711 audio will work. The issue I am seeing is it appears that with some endpoinds on Level 3 that the t.38 tunnel comes up fine but no fax data starts flowing but this only is happening with faxes coming from some Cisco gateways sending out via PRI using t.30 No, unfortunately there isn't a way to do that that I can see. It wouldn't be terribly hard to add to res_fax.c, but I don't think we ever thought of doing that before. With out this I have no way to force the fall back then and the faxes will always fail in this case because t38 successfully negotiates.. Do you have any other ideas? If I pick arround in the source what might it take to add another option to the ReceiveFAX to only do g711 audio? Is this somthing that I could get submitted back into the tree if I can figure it out? Thanks Bryant -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Wednesday, January 26, 2011 5:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/26/2011 04:16 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Wednesday, January 26, 2011 4:52 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/26/2011 03:14 PM, Bryant Zimmerman wrote: Is there a way for me to force t.38 off for a call but to allow t.38 for other calls. What I am thinking is if a t.38 fails to flag the next call from that number to g711 audio. This would at least let me work arround the issue for now where t.38 fails with some endpoints but not others and the g711 audio will work. The issue I am seeing is it appears that with some endpoinds on Level 3 that the t.38 tunnel comes up fine but no fax data starts flowing but this only is happening with faxes coming from some Cisco gateways sending out via PRI using t.30 No, unfortunately there isn't a way to do that that I can see. It wouldn't be terribly hard to add to res_fax.c, but I don't think we ever thought of doing that before. With out this I have no way to force the fall back then and the faxes will always fail in this case because t38 successfully negotiates.. Do you have any other ideas? If I pick arround in the source what might it take to add another option to the ReceiveFAX to only do g711 audio? Is this somthing that I could get submitted back into the tree if I can figure it out? Most definitely; I can see cases like yours where someone would want to be able to forcibly disable T.38 for specific calls for troubleshooting purposes. In fact... if you give me about 15 minutes, I'll commit a patch to Asterisk trunk to add an option to do that, and you can backport it to the version you are using :-) Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? Much thanks on this. Bryant -- _ -- 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] res_fax
Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 - Kevin I downloaded 1.8.2.3 and copied the modified version of res_fax.c into my the res folder. I built and installed the version of asterisk. When I use the new n option with ReceiveFAX I get a bunch of WARNING messages on the console that state. [Jan 26 20:43:38] WARNING[23393]: chan_sip.c:6047 sip_write: Asked to transmit frame type slin, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw) If I shut of the n option it goes back to the normal behavior. It appears that there is somthing missing in the n option and it is not causing it to fall back to audio only mode. as it would if t38pt_udptl=no Bryant -- _ -- 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] res_fax
Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? Thanks Bryant -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Thursday, January 27, 2011 10:31 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? My schedule is pretty full today, but I will take another look over the code and see what might be going on. -- Kevin Thanks I am continuing with other parts of my fax code as well for now. I will test any changes as you are able to make them. Bryant -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Thursday, January 27, 2011 3:08 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? Revision 304599 should fix this (and I also changed the option letter from 'n' to 'F' since it really means 'force audio'). - Kevin I will rebuild and test in a bit. Thanks Bryant -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Thursday, January 27, 2011 3:08 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? Revision 304599 should fix this (and I also changed the option letter from 'n' to 'F' since it really means 'force audio'). _ Kevin The 304599 rev does seem to work good. I just finished my testing on it and the F option works great. I have three more test to do and if they pass it should be good to go. When could it get into the releases? Thanks Bryant -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Monday, January 31, 2011 5:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/31/2011 02:08 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Thursday, January 27, 2011 3:08 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? Revision 304599 should fix this (and I also changed the option letter from 'n' to 'F' since it really means 'force audio'). _ Kevin The 304599 rev does seem to work good. I just finished my testing on it and the F option works great. I have three more test to do and if they pass it should be good to go. When could it get into the releases? It's a new feature, so it won't go into any existing release branches; the first release that will have this addition is Asterisk 1.10.1. Of course, the patch is quite small as you've seen, so it will be easy for you to apply it to your installations. _ Kevin I just replaced the res_fax.c file with the one from 304599. Would I just keep doing that as I step forward on versions of 1.8.x? If this is the case how would I get any other critical changes to res_fax.c that may occur after rev 304599? How would I create a patch that would allow me to apply it to additional release version of asterisk. Sorry for the simple questions I do most of my dev on windows machines and this process is a still a bit confusing to me. Thanks Bryant -- _ -- 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 1.8.3 BLF stopped working
I am running 1.8.3 and my BLF lights have stopped working. The hints appear to be intact when I use core show hints. But none of the phones are getting the BLF updates. This has happend in the past and I have had to restart my server. What could be causing this to occur. It did not do this with the 1.6.x builds. Is there a way to reload the hints or force a refresh without re-starting Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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] Voicemail email attachment as MP3, with tags containing sender name, number, message number
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Tuesday, February 15, 2011 1:16 PM To: Asterisk Users List Subject: [asterisk-users] Voicemail email attachment as MP3, with tags containing sender name, number, message number I found some great pieces of script on the internet that I've combined to allow Asterisk to send voicemails as an MP3 file, and encode the sender name and number as well as message number as tags into the MP3 file. I even include a cover art image which has our company logo and PBX symbol in it. Mobile phone users love it, and Android phones can now play the attachments (without having to move to the larger WAV format). If anyone wants to try it out let me know! Michelle That sounds like a nice implimentation. I would love to take a look. I have tried to figure out how to do things before the e-mail is sent and this sounds like it would allow for that. Bryant -- _ -- 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] Recieve_Fax caused crash 1.8.2.3
I had an issue today where receive_fax caused an asterisk switch to crash. The switch is 1.8.2.3 version. The call was coming from a fax machine. The call started receive_fax answered and then asterisk stopped responding. I was able to log into asterisk but it would not do a core restart now nor would it take any calls or show an peer registrations. I had to kill the asterisk process and restart it. As best we can tell there was no attempt by the sender to intentionally send any malformed packets that should have caused this. I see there is a security patch 1.8.2.4 that lists some RTP security issues. is it possible that this fix may address what I ran into as well? Thanks zktech -- _ -- 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] SIPAddHeader not working
From: Jonas Kellens jonas.kell...@telenet.be
Sent: Wednesday, March 09, 2011 4:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
exten = s,n,SIPAddHeader(Privacy: id)
in SIP invite no trace of this header :
Using Asterisk 1.6.2.16.1
How do I correctly add the Privacy header ?!
Kind regards,
Jonas.
Jonas
Here is the way we add the rfc-3325 privacey header so our vendors pick it
up correctly. This is what we use in 1.6.x and 1.8.x
When I check on my versions the privacy header appears to be there.
exten = rfc-3325-CPN,1,NoOp(Set Call Privacy)
exten = rfc-3325-CPN,n,NoOp(From ${SIP_HEADER(From)})
exten = rfc-3325-CPN,n,NoOp(To ${SIP_HEADER(To)})
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(SIP_HEADER(From),@,2)})
exten = rfc-3325-CPN,n,GotoIf($[${l_sipheaderfromip} != ]?hasat)
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(CUT(SIP_HEADER(From),,1),:,2)})
exten = rfc-3325-CPN,n,Goto(gotip)
exten =
rfc-3325-CPN,n(hasat),Set(FROM_IP=${CUT(CUT(CUT(SIP_HEADER(From),@,2),,1),:
,1)})
exten = rfc-3325-CPN,n(gotip),NoOp(Gateway IP is ${FROM_IP})
exten =
rfc-3325-CPN,n,SIPAddHeader(P-Preferred-Identity:${CALLERID(name)}
sip:+1${CALLERID(num)}@${FROM_IP}\;user=phone)
exten = rfc-3325-CPN,n,SIPAddHeader(Privacy: id)
exten = rfc-3325-CPN,n,Set(CALLERPRES()=prohib_not_screened)
exten = rfc-3325-CPN,n,Set(CALLERID(num)=Anonymous)
exten = rfc-3325-CPN,n,Set(CALLERID(name)=Anonymous)
exten = rfc-3325-CPN,n,Return()
--
_
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Re: [asterisk-users] Multiple SIP endpoint registrations
From: --[ UxBoD ]-- ux...@splatnix.net Sent: Wednesday, March 09, 2011 6:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Multiple SIP endpoint registrations Hi, With Asterisk 1.8 is it now possible to register the same SIP account at multiple endpoints and for both to ring when the associated extension is dialed ? -- Thanks, Phil Phil Based on what we have seen you must have a sip account per end point. If you want to ring multiple endpoints you can specify them in the dial command exten = exp,n,Dial(SIP/Account1SIP/Account2SIP/Account3, options). This is the only way we know of to do this as you must have an IP and port number to send traffic to and we have seen no method of having two IP's and Ports per account. The only other way I could think of is some outside the box multicast method and the endpoints would need to be set to receive any SIP traffice without registration. This would not be secure and to my knowledge would be beyond basic asterisk at this time. Thanks Bryant -- _ -- 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] call file for page auto-call
From: satish patel satish...@hotmail.com Sent: Tuesday, March 15, 2011 2:31 PM To: asterisk-users asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call file for page auto-call Thanks for you input but how to do SIPAddHeader(Alert-Info: Ring Answer) for auto answer my polycom phones and how to create group in .call file I am reading at http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out but didn't found anything related group calling. may be i am missing something could point me out.. -S Hey Support, I am planing to implement new page system with asterisk 1.8 we have 200 SIP calls and page() will overkill my system if announce in one shot. so i am planing to record and play page over 50...50...50 chunk.. I am planing to do with .call file for auto calling after record message but i don't know how to call multiple extension ? and how to use page() with .call file for auto-answer and auto-call? Appreciate your help.. -S One suggestion - set up 4 call groups. Group 1 calls first 50 phones, Group 2 51-100, etc. If you set it up like 601, 602, etc. then in your call file you can test with 101 to get what you want, then change it to 601. satish We have a page group offering in our systems. We do not use call files to handle this we do it as direct processing. If I were to use a call file. I would create a custom context to use from the call file. The first thing I would do is build a string list of the phones being paged. The second is I would add the auto answer headers for the different types of phones that are in my network. This process is really quite straight forward. The flow would be somthing like this.. Call Page Record. Call in. Record Message. Select page groups to send the message to. Write a call file with the message name, page groups and the page handling context. Call file would contain. Custom page handling/processing context. List of page groups and message file name stored in vars. In your Custom page handling/processing context. Read and parse the page groups list from a variable set in the .call file Read the recorded message from the .call file Loop for each page group. Build your paging group in a string (This should be able to be done using some kind of list. Either stright. csv or database you choose) Set the correct page headers Call the page command with the correct list. Play the recorded message Hangup Loop back and do next group. This is really just a coding project. You have to break the entire issue down into it's base parts and then solve each one. Good luck. Bryant Z -- _ -- 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 1.8 question
From: Bob Beers bob.be...@gmail.com Sent: Friday, March 25, 2011 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk 1.8 question On Fri, Mar 25, 2011 at 9:51 AM, Jerry Geis ge...@pagestation.com wrote: In 1.4 there was core show channels concise This seems to be gone from 1.8. When I am using the AMI interface to get a listing of all channels my listing names are cut short. SIP/devcentos5x64_to notice above. In 1.4 it would have given me SIP/devcentos5x64_to_am2mm How in 1.8 do I get the FULL listing of the channels. I think you should try all three below and see which gives you what you like: core show channels core show channels concise core show channels verbose From my experience, they all work in 1.8, but do give different output. -- HTH, - Bob Beers -- They work for me in 1.8 as well. Bryant -- _ -- 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] Why shouldn't I use 1.8?
From: Jonathan Thurman jonat...@thurmantech.com Sent: Friday, March 25, 2011 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Why shouldn't I use 1.8? On Fri, Mar 25, 2011 at 7:36 AM, Douglas Mortensen d...@impalanetworks.com wrote: But I would like specific reasons why I shouldn't use 1.8 in a production environment if anyone has some? That is a loaded question, in that no two environments are likely to be the same. Some bugs are major issues for 1% of the install base and take time to get merged into the code base. You should read through the open issues for the 1.8 branch and see if there are any show stoppers for your environment. If not, try it in the lab and validate that it works for you. Check out https://issues.asterisk.org For my environment specifically, this issue is currently preventing me from migrating from 1.6.2: - 18818 [patch] Crashing when using local channels and realtime on asterisk There are a lot of benefits to the 1.8 branch (Long term support, Called party id, Multicast RTP, etc) but only you can say if it will work with your configuration in your environment. -Jonathan -- Doug I agree with Jonathan. I have moved all but one of our production switches to 1.8 the only thing holding me back is a minor bug so I have to keep the 1.6.2 box around until that patch is released into the 1.8 branches. When that is done I will no longer be on the 1.6. I have over 98% of our load on the 1.8 switches and we are doing multi tenant pbx hosting and sip trunking. A point of note I just turned down my last 1.4 box 2 weeks ago. It was not because it was not working but because I need more volume and 1.8 on the new hardware meets that need and I get the bonus of not having to support three versions of asterisk now. It is very likely that most of the time I will have at least two versions in production at a time. This is so I can offer the newest features with a stable build and I can offer a more long term support for the customers that the newest features are not as important. Most of my switch hardware has a planned 4 year life span. The better asterisk gets the longer I can stretch that investment. My recommendation is if 1.8 does not have any bugs that are issues for you try 1.8 out of the gate and test, test, test offer feed back from your testing and the bugs will get fixed. I would not spend to much time worrying spend more time doing. Good luck Bryant -- _ -- 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] call-limit bypass
From what I understand on the newer versions of asterisk call-limit does not limit calls anymore. You have to limit them from your code using call groups. From what I have seen on the 1.6x and 1.8 versions call-limit does not limit your call counts. We use code and the GROUP_COUNT to limit calls. If you use it right it is rock solid. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Rizwan Hisham rizwanhas...@gmail.com Sent: Monday, April 04, 2011 12:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] call-limit bypass Hi everyone, one of our users last night bypassed asterisk call-limit limitation. I have no Idea how. Is it possible? Is there a bug in asterisk that can be manipulated for this purpose? The call-limit variable was to 2, and the user initiated 169 calls in 2 minutes each has duration at least 8 minutes. Please comment... Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.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
[asterisk-users] Asterisk 1.8.3
I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a core restart now cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2 was poor but it is worse with 1.8.3.2 any ideas of how I can approach solving this. Thanks Bryant -- _ -- 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 1.8.3
On 4/5/11 6:10 PM, Bryant Zimmerman wrote: I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a core restart now cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2 was poor but it is worse with 1.8.3.2 any ideas of how I can approach solving this. From: Edwin Lam edwin@officegeneral.com Sent: Wednesday, April 06, 2011 5:37 PM We've upgraded our system over the weekend from 1.4.35 to 1.8.3.2 For the past couple of days, we had several random hangs(most of the time core stop now didn't work, I had to kill -9 the process) Also the PRI behavior seems to be slightly different, we can't hear any early media sounds on 800 numbers that goes through ATT. I finally downgraded it back to 1.6.2.17, now everything work. Edwin Thanks for your response. I have added the patch for 18818 per Michel Verbrask's recomendation. It appers that it has made quite a difference. I don't have an PRI connections as all of our PRI's are connected via SIP gateways. I did run into serveral instances wher I had to kill -9 the process as well but post patch I have been in good shape know on wood. I hope there will be a new release that will address the stability issues very soon if they release 1.8.4 without cleaning this up I won't move unitl it is addressed. For Now 1.8.3..2 is very bad. Thanks Bryant -- _ -- 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 1.8.3
On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@officegeneral.com wrote: On 4/6/11 3:02 PM, Bryant Zimmerman wrote: Thanks for your response. I have added the patch for 18818 per Michel Verbrask's recomendation. It appers that it has made quite a difference. I don't have an PRI connections as all of our PRI's are connected via SIP gateways. I did run into serveral instances wher I had to kill -9 the process as well but post patch I have been in good shape know on wood. I hope there will be a new release that will address the stability issues very soon if they release 1.8.4 without cleaning this up I won't move unitl it is addressed. looking back at the messages file for the past 2 days. it just hanged on totally different events none of which related to Local channels. as far as the PRI not hearing early media issue. here's the excerpt from the messages file after pri debug on command: * -- Executing [18008291011@out_going_x:1] Dial(SIP/ ... Parts Removed see origional response -- Processing IE 30 (cs0, Progress Indicator) -- PROGRESS with cause code 127 received -- DAHDI/34-1 is making progress passing it to SIP/4988-6-0b45 *** i used the same SIP station to dial the same 800 number on both versions (1.8.3.2 1.6.2.17). the output are pretty much identical except on 1.8.3.2, after the PROGRESS with cause code 127... message. i would hear nothing until the other side timed out hang up, whereas on 1.6.2.17. i got the DAHDI/... is making progress passing it to SIP... message and can hear the early media from the other side. For Now 1.8.3..2 is very bad. agreed... From: Satish Patel satish...@hotmail.com Sent: Thursday, April 07, 2011 8:22 AM Oh! Boy, Is it ture 1.8.3 is unstable? We are planning to put this in production. Please suggest me what should I do? Satish For me 1.8.3.2 has been the worst build that I have tried to use as far a stability in a very long time. We are having issues with deadlocks and voicemail. I don't have a good option for you if you want to run 1.8 currently the most stable release version I have found is 1.8.2.3 but I am having the Voicemail issues there as well. Things like messages not deleting propperly and hanging up the mail box so users can't check them. -- _ -- 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 1.8.3
On Apr 7, 2011, at 8:51 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote: On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@officegeneral.com wrote: On 4/6/11 3:02 PM, Bryant Zimmerman wrote: Thanks for your response. I have added the patch for 18818 per Michel Verbrask's recomendation. It appers that it has made quite a difference. I don't have an PRI connections as all of our PRI's are connected via SIP gateways. I did run into serveral instances wher I had to kill -9 the process as well but post patch I have been in good shape know on wood. I hope there will be a new release that will address the stability issues very soon if they release 1.8.4 without cleaning this up I won't move unitl it is addressed. looking back at the messages file for the past 2 days. it just hanged on totally different events none of which related to Local channels. as far as the PRI not hearing early media issue. here's the excerpt from the messages file after pri debug on command: * -- Executing [18008291011@out_going_x:1] Dial(SIP/ ... Parts Removed see origional response -- Processing IE 30 (cs0, Progress Indicator) -- PROGRESS with cause code 127 received -- DAHDI/34-1 is making progress passing it to SIP/4988-6-0b45 *** i used the same SIP station to dial the same 800 number on both versions (1.8.3.2 1.6.2.17). the output are pretty much identical except on 1.8.3.2, after the PROGRESS with cause code 127... message. i would hear nothing until the other side timed out hang up, whereas on 1.6.2.17. i got the DAHDI/... is making progress passing it to SIP... message and can hear the early media from the other side. For Now 1.8.3..2 is very bad. agreed... From: Satish Patel satish...@hotmail.com Sent: Thursday, April 07, 2011 8:22 AM Oh! Boy, Is it ture 1.8.3 is unstable? We are planning to put this in production. Please suggest me what should I do? Satish For me 1.8.3.2 has been the worst build that I have tried to use as far a stability in a very long time. We are having issues with deadlocks and voicemail. I don't have a good option for you if you want to run 1.8 currently the most stable release version I have found is 1.8.2.3 but I am having the Voicemail issues there as well. Things like messages not deleting propperly and hanging up the mail box so users can't check them. 1.8.2 is unusable if you use RealTime without the patch in this issue https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 From: Satish Patel satish...@hotmail.com Sent: Thursday, April 07, 2011 9:06 AM We don't have realtime configuration everything is in plain text file. Is 1.8.3 has realtime issue or general issue? Satish I have seen my issues with the realtime disabled and using just plain text. The issues get worse for me when we move to our realtime confgs. So from my perspective I would say you might get farther with realtime off but I would not bank on it. -- _ -- 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 1.8.3
From: Chris Owen ow...@hubris.net Sent: Thursday, April 07, 2011 9:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.3 Best I can tell, multi-tenant parking also hasn't worked in any of the 1.8.x releases. Chris Chris I have not been able to get multi-tenant parking stable there either. I gave up yesterday on 1.8.3.2 as I could not get it stable with any number of patches I could find. I fell back to 1.8.2.3 as that is the last version that I have been able to run production with. My customers have now been happy for the last 24 hours. I also tried 1.8.4 rc and the stability did not appear to be much better then 1.8.3.2 I hope they don't release 1.8.4 until the stability issues are addressed more rc version with fixes would be ideal. The longer these items drag out the worse it gets for users to know what to use. I would ask the developers to hold 1.8.4 until some of these items can be fixed and rolled in. Bryant -- _ -- 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] Meetme Time Limit?
Is there a way to place a hangup time on a dynamic Meetme conference. I am using Page() with a Meetme conf and I have had a few instances where someone from a wifi voip phone looses ip while doing a page and the page never hangs up. I have to kill it. I need to somehow limit the page so after a worse case 2Min timeout it hangs up. Thanks Bryant -- _ -- 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] Discussion: Are we ready to leave 1.4 behind?
I will throw in my 2 cents on this. I agree that 1.8 is not as stable as it needs to be. From my perspective I will continue to use the 1.4.x or 1.6.2.x release that is the best fit for me and it should continue to do what it does and it get's it's security releases. If the primary development focus is moved to 1.8 to get the lead out and stabilize it than that is what I want. New work on 1.10 should only be under taken after 1.8.x is stable then we can tinker with the newer stuff. Making it stable makes it stronger. As far as I can see 1.4.x is stable and that is what people want use it until 1.8.x is where you want it but test 1.8.x help find the bugs so you can make the move otherwise stay with the solid 1.4.x and wait for others to find the bugs in the newer versions. I know of several companies that are on 1.2 and will make the move to a new version only if 1.2 fails them and it has not for their needs. Again we do need 1.8 to be stabilized quickly the stuck voicemail issues and system crashes are driving me crazy. Thanks to all of the developers who work on asterisk. The core makes my business possible. Keep up the good work Thanks Bryant -- _ -- 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] Asterisk 10 / Trunk and RecieveFax F Option
I have been using sendfax and recievefax with 1.8.x.x version I have a patch that Kevin Fleming wrote to allow the forced shutoff of T.38 F option. This was considered a new feature so it is not in new releases of 1.8.x and I have not been able to get a patch working for the current releases. How can I get the current 10/trunk version as I really need this feature. Anyone used the 10/trunk build and had any success with it? Here is a link to the revision. http://svnview.digium.com/svn/asterisk?view=revrev=304342 Any ideas of feed back would be apperciated. Bryant -- _ -- 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] Discussion: Are we ready to leave 1.4 behind?
From: Ira i...@extrasensory.com Sent: Thursday, May 05, 2011 12:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind? At 07:56 AM 5/5/2011, you wrote: So how can we fix this? How can we get more people involded? What makes projects like FedoraTesting[3] and DebianTesting[4] popular? How can the Asterisk project reproduce their success? Well, it's not a lot of people willing to run beta software on their phone system. Phones need to work and for most people they need to work perfectly all the time. I'm one of those oddities that will always run beta software if given the chance but my experience is that quite rare. As I've said before, I'm more then willing to help with answering questions about the testsuite or reviewing code that people want to get merged in. We also have an IRC channel, #asterisk-testing available for people to join, ask question, idle, lurk, etc, or if you want to reply to this thread, feel free. But get involved! :) So I'm the person who has never been able to keep 1.8 alive on my system for more than a minute or two and I've probably tried more than 10 different betas and release versions. I posted a bug report which was closed in minutes, I posted the problem on this list every few tries and zero response. I tried to figure out mIRC. It's installed on my machine but I've never got past that. I just don't get the instructions. I know that all the people involved in the project are Linux heads, but some of us, like me, have a Linux box only because of Asterisk and if you want my help, you need to make being involved accessible and stop assuming we all know what you know. I see the words, jut post a bug report on Mantis posted all the time and I'm sure it means as little to others as it means to me. Maybe there needs to be a web page somewhere, Asterisk beta testing for dummies so that you can point us to so you don't have to answer the stupid questions over and over. I've beta tested enough and had enough beta testers to understand the kinds of things that make it possible to get bugs fixed, but it's usually a very small percentage of users that understand that. Ira --- Ira Contact me off list and we can have a conversation. We are running 1.6.2.x boxes and 1.8.x boxes very successfully. We have had issues with 1.8.x but that is to be expected as it has been bleeding edge at times. I am not a linux expert either but if I may be ableo to point you in the right direction. Your determination to support Asterisk is what the community needs if I can help foster that I would be happy to do so. I am only where I am at because others invested some time in me. examples: The power of IRC chats. I spend three days on the freenas forums and could not solve a problem I was just pulling my hair out. I took 20 min to get up to speed with irc using IceChat and after 1.5 hours on the freenas board the problem was solved and I was diving deep into the guts of the freenas 8 system. it was a game changer for me. Bryant -- _ -- 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
