[asterisk-users] Tim's band DEEPFALL NOT SPAM!!

[Tim King]

First of all I apologize for emailing everyone in one mass email like this,
but it is the only logical way to get this done. We have restarted the
Kickstarter campaign in hopes of raising the funds needed to get us into
the studio with a national producer.

PLEASE DONATE IF YOU CAN!

No Donation is too small.

It only takes 3 minutes.

Here is the Link:
https://www.kickstarter.com/projects/424887562/new-ep-music-development-30?ref=nav_search

Please share this link with anyone you might know that could spare $5
toward a good cause.

Thanks

TimK, TimH, Sully, Chris, and Greg
-- 
_
-- 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] G729 and voice mail

[Tim King]

The G729 is coming from a Sangoma D100-030 card and the G729 transcoding is
working, the only issue I have is not hearing prompts from the system.

On Tue, Jun 5, 2012 at 2:42 PM, Eric Wieling  wrote:

> What does the output of "g729 show licenses" show?  If it doesn't show
> licenses then Asterisk is not licensed for G729 codec.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Tim King
> Sent: Tuesday, June 05, 2012 2:32 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] G729 and voice mail
>
> I am trying to figure out the best way to deal with this. I want all of
> the calls in the network to be G729 and this is working. I do have hardware
> that provides me 30 g729 licenses. I am setting each extensions to
> disallow=all and allow=g729. However when I have this setup, I get no voice
> mail prompts. I tried setting to disallow=all and allow=g729,alaw and I
> still have no audio when calling voice mail. If I remove the disallow=all I
> do have voice mail prompts, but the calls do not seem to be always using
> g729 when possible.
>
>
> --
> _
> -- 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] G729 and voice mail

[Tim King]

I am trying to figure out the best way to deal with this. I want all of the
calls in the network to be G729 and this is working. I do have hardware
that provides me 30 g729 licenses. I am setting each extensions to
disallow=all and allow=g729. However when I have this setup, I get no voice
mail prompts. I tried setting to disallow=all and allow=g729,alaw and I
still have no audio when calling voice mail. If I remove the disallow=all I
do have voice mail prompts, but the calls do not seem to be always using
g729 when possible.
--
_
-- 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] Sangoma D100 Transcoder Asterisk 1.6

[Tim King]

I have installed and configures this card in asterisk 1.6. When trying to
load the module codec_sangoma.so I see the following in the asterisk log.

[2012-06-04 15:50:31] WARNING[18168] loader.c: Error loading module
'codec_sangoma.so': /usr/lib/asterisk/modules/codec_sangoma.so: undefined
symbol: ast_config_load
[2012-06-04 15:50:31] WARNING[18168] loader.c: Module 'codec_sangoma.so'
could not be loaded.

Has anyone had a similar issue with this card or have any idea what the
undefined symbol: ast_config_load might mean to figure out what direction
to head for further debugging?
--
_
-- 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 1.8.3.3 T.38 Gateway

[Tim King]

I have found numerous claims that 1.8 can do T.38 gateway with a patch,
however I am yet to find the patch or any instructions on implementing it.
Anyone have a link?
--
_
-- 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] Faxes suddenly failing

[Tim King]

I realize that faxing is not great with voip but here is my confusion. I
have been working on a web based fax system for 2 weeks. During this time I
have sent over 100 2 page faxes without any errors. Now today as things are
finally completed I can not seem to get any fax to go through unless it is a
1 page cover only. Anyone able to tell the issue from this debug output?

   -- Channel 'SIP/MyVoipProvider-0046' FAX session '12' started
-- FAX handle 0: [ 000.38 ], STAT_EVT_STRT_RX   st: IDLE
rt: IDLENSRX
-- FAX handle 0: [ 000.000184 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY
rt: RRDYNHRY
-- FAX handle 0: [ 000.000504 ], P30EVN_RECEIVE_STARTED
-- FAX handle 0: [ 000.000538 ], STAT_INFO_CSI
-- FAX handle 0: [ 000.000568 ], STAT_INFO_DIS
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
000.091837 ], stack sent 5 frames (100 ms) of energy.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
000.160248 ], stack sent 3 frames (60 ms) of silence.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
000.960201 ], channel sent 48 frames (960 ms) of silence.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
000.979464 ], channel sent 1 frames (20 ms) of energy.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
003.157848 ], stack sent 150 frames (3000 ms) of energy.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
003.219814 ], stack sent 3 frames (60 ms) of silence.
-- FAX handle 0: [ 005.240927 ], STAT_EVT_TX_V21_DONE   st: WT_DIS_RSP
rt: WDSRNT21
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
005.579811 ], stack sent 118 frames (2360 ms) of energy.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
006.481179 ], channel sent 275 frames (5500 ms) of silence.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
007.801045 ], channel sent 66 frames (1320 ms) of energy.
-- FAX handle 0: [ 007.800554 ], STAT_FRM_CRP
-- FAX handle 0: [ 007.800586 ], STAT_EVT_CRP   st: WT_DIS_RSP
rt: NT4X
-- FAX handle 0: [ 007.800602 ], STAT_EVT_FSC_ERR   st: WT_DIS_RSP
rt: UNEXPECT
-- FAX handle 0: [ 011.012832 ], STAT_EVT_RX_TRN_ENDst: WT_DIS_RSP
rt: RXXXNFRX
-- FAX handle 0: [ 011.012878 ], STAT_INFO_CSI
-- FAX handle 0: [ 011.012905 ], STAT_INFO_DIS
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
011.152812 ], stack sent 279 frames (5580 ms) of silence.
-- FAX handle 0: [ 013.179561 ], STAT_EVT_TX_V21_DONE   st: WT_DIS_RSP
rt: WDSRNT21
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
013.471827 ], stack sent 116 frames (2320 ms) of energy.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
014.260642 ], channel sent 323 frames (6460 ms) of silence.
-- FAX handle 0: [ 016.119786 ], STAT_INFO_TSI
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
016.460661 ], channel sent 110 frames (2200 ms) of energy.
-- FAX handle 0: [ 016.460394 ], STAT_INFO_DCS
-- FAX handle 0: [ 016.460431 ], STAT_EVT_DCS   st: WT_DIS_RSP
rt: WDSRNDCS
-- FAX handle 0: [ 016.460449 ], STAT_NEG_V17_14400
-- FAX handle 0: [ 016.460464 ], STAT_NEG_MH
-- FAX handle 0: [ 016.460476 ], STAT_NEG_A4
-- FAX handle 0: [ 016.460488 ], STAT_NEG_RES_204x196
-- FAX handle 0: [ 016.460500 ], STAT_NEG_ECM
-- FAX handle 0: [ 016.460514 ], STAT_EVT_SW_ECMst: WT_DIS_RSP
rt: WDSRNSWE
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
016.540315 ], channel sent 4 frames (80 ms) of silence.
-- FAX handle 0: [ 016.800906 ], STAT_EVT_RX_IMG_STRT   st: RCV_ECM_TRN
rt: UNEXPECT
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
019.700543 ], channel sent 158 frames (3160 ms) of energy.
-- FAX handle 0: [ 019.759984 ], STAT_EVT_RX_TRN_ENDst: RCV_ECM_TRN
rt: RTCFNERT
-- FAX handle 0: [ 019.760071 ], STAT_FRM_CFR
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
019.912812 ], stack sent 322 frames (6440 ms) of silence.
-- FAX handle 0: [ 020.957834 ], STAT_EVT_TX_V21_DONE   st: RCV_ECM_STRT
rt: RECMNT21
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
021.278809 ], stack sent 68 frames (1360 ms) of energy.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
022.261160 ], channel sent 128 frames (2560 ms) of silence.
-- FAX handle 0: [ 022.517880 ], STAT_EVT_RX_IMG_STRT   st: RCV_ECM_STRT
rt: RECMNSRI
-- FAX handle 0: [ 022.517982 ], P30EVN_PHASE_C
-- FAX handle 0: [ 022.517998 ], P30EVN_DOC_START
-- FAX handle 0: [ 022.518429 ], P30EVN_PAGE_START
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
031.102000 ], channel sent 442 frames (8840 ms) of energy.
   > Channel 'SIP/MyVoipProvider-0046' fax session '12', [
031.160415 ], channel sent 3 frames (60 ms) of silence.
-- FAX handle 0: [ 031.160196 

Re: [asterisk-users] System Command not executing php

[Tim King]

What does this mean? The suggestion from Jian did resolve the issue. Thank
You.

On Wed, Aug 24, 2011 at 3:07 PM, Steve Edwards wrote:

> Un-top-posting...
>
>
>  On 11-08-24 10:21 AM, Tim King wrote:
>>
>
>  I have been testing this for a week now, and I am still struggling to make
>> it work. Here is the output from extension 11 just to show that permissions
>> are correct and asterisk can access faxnotify.php
>>
>
> On Wed, 24 Aug 2011, J Gao wrote:
>
>  Try put the php in the AGI directory and call it via AGI() in Asterisk.
>>
>
> If the script does not conform to the AGI protocol, do not call it with
> agi().
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
>
> --
> __**__**_
> -- 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<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] System Command not executing php

[Tim King]

Jian,

I tried that and I get the results seem the same. It appears to run but does
not.


exten => 777,1,AGI(faxnotify.php,NOTIFY "tim.compnetw...@gmail.com"
"616555" "24/08/11 : 09:00:00" "FAX_SUCCESS" "1")
exten => 777,n,Playback(vm-goodbye)
exten => 777,n,Hangup()

 -- Executing [777@outb2:1] AGI("SIP/616818-0310",
"faxnotify.php,NOTIFY "tim.compnetw...@gmail.com" "616555" "24/08/11 :
09:00:00" "FAX_SUCCESS" "1"") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/faxnotify.php
-- AGI Script faxnotify.php completed,
returning 0
-- Executing [777@outb2:2] Playback("SIP/616818-0310",
"vm-goodbye") in new stack
--  Playing 'vm-goodbye.alaw' (language 'en')
-- Executing [777@outb2:3] Hangup("SIP/616818-0310", "") in new
stack
  == Spawn extension (outb2, 777, 3) exited non-zero on
'SIP/616818-0310'
--
_
-- 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] System Command not executing php

[Tim King]

*I have been testing this for a week now, and I am still struggling to make
it work. Here is the output from extension 11 just to show that permissions
are correct and asterisk can access faxnotify.php*

-- Executing [11@outb2:1] Verbose("SIP/616818-02f6",
"Testing") in new stack
Testing
-- Executing [11@outb2:2] Set("SIP/616818-02f6",
"FNAME=/var/lib/asterisk/bin/faxnotify.php") in new stack
-- Executing [11@outb2:3] Verbose("SIP/616818-02f6", "Size:
2553") in new stack
Size: 2553
-- Executing [11@outb2:4] Verbose("SIP/616818-02f6", "Mode:
100777") in new stack
Mode: 100777
-- Executing [11@outb2:5] Verbose("SIP/616818-02f6",
"#!/usr/bin/php
 11,1,Verbose(Testing)
exten => 11,n,Set(FNAME=/var/lib/asterisk/bin/faxnotify.php)
exten => 11,n,Verbose(Size: ${STAT(s,${FNAME})})
exten => 11,n,Verbose(Mode: ${STAT(m,${FNAME})})
exten => 11,n,Verbose(${FILE(${FNAME},,100)})
exten => 11,n,TrySystem(${FNAME})
exten => 11,n,Verbose(${SYSTEMSTATUS})
exten => 11,n,Hangup()

exten => 777,1,System(/var/lib/asterisk/bin/test.php)
exten => 777,n,Playback(vm-goodbye)
exten => 777,n,Hangup()

*[ext-fax]*
exten => s,1,Noop(Receiving Fax for: ${FAX_RX_EMAIL} , From:
${CALLERID(all)})
exten => s,n(receivefax),StopPlaytones
exten => s,n,ReceiveFAX(${ASTSPOOLDIR}/fax/${UNIQUEID}.tif,f)
exten => s,n,ExecIf($["${FAXOPT(error)}"=""]?Set(FAXSTATUS=FAILED LICENSE
EXCEEDED))
exten => s,n,ExecIf($["${FAXOPT(error)}"!="" &&
"${FAXOPT(error)}"!="NO_ERROR"]?Set(FAXSTATUS="FAILED FAXOPT: error:
${FAXOPT(error)} status: ${FAXOPT(status)} statusstr: ${FAX
exten => s,n,Hangup
exten => h,1,GotoIf($["${FAXSTATUS:0:6}" = "FAILED"]?failed)
exten => h,n(process),GotoIf($[${LEN(${FAX_RX_EMAIL})} = 0]?end)
exten => h,n,System(${ASTVARLIBDIR}/bin/fax-process.pl --to
"${FAX_RX_EMAIL}" --from "nore...@mydomain.com" --dest "${FROM_DID}"
--subject "New fax from ${URIENCODE(${CALLERI
exten => h,n(end),Macro(hangupcall,)
exten => h,process+101(failed),Noop(FAX ${FAXSTATUS} for: ${FAX_RX_EMAIL} ,
From: ${CALLERID(all)})
exten => h,n,Macro(hangupcall,)

*[app-fax]*
include => app-fax-custom
exten => 6169804602,1,Set(FAX_RX_EMAIL=tim.compnetw...@gmail.com)
exten => 6169804602,n,Goto(ext-fax,s,1)
exten => h,1,Macro(hangupcall,)


*test.php*

";

// end setting up

if ($messtype == "INIT") { // SendFAX called successfully in the dialplan...
$emailSubject = "Your fax to $dest has been initiated";
$notice = "Your fax to $dest sent on $timestamp has been initiated.
You will get a notification " .
  "email when the transmission is complete. ";
$emailBody = "Hi!  \n\n" .  $notice . " \n\n " .
 "If you have any queries, please contact us
at:  $CONTACT_EMAIL";
mail($email, $emailSubject, $emailBody, $headers);
}
else {  // meaning $messtype = "NOTIFY" ... sending of fax is complete. Need
to check if SUCCEEDED
$tech_details = "-- \n".
"DESTINATION = $dest\n".
"TIMESTAMP = $timestamp \n".
"FAXOPTS_STATUSSTRING = $status \n".
"NUM_PAGES = $numpages  \n".
"-- \n";


echo "Sending fax notification email to: $email from $FROM_EMAIL
\n";

if($status == $SUCCESS_STATUS) {
  $emailSubject = "Your fax to $dest was delivered successfully";
  $notice = "This is an automated response to let you know that your
fax to " .
"$dest sent on $timestamp was delivered
successfully. \n";
} else {
  $emailSubject = "Your fax to $dest could not be sent";
  $notice = "This is an automated response to let you know that your
fax to " .
"$dest sent on $timestamp could not be
delivered. \n";
}

$emailBody = "Hi!  \n\n" .  $notice . "\n\n" . $tech_details . "
\n\n " .
 "If you beleive there was an error please
contact:  $CONTACT_EMAIL";

// echo $emailSubject . "\n";
// echo $emailBody . "\n";

// mail
mail($email, $emailSubject, $emailBody, $headers );
// exec("echo $email $timestamp $emailSubject >>
/var/log/asterisk/webfax.log");
// exec("echo $emailBody >> /var/log/asterisk/webfax.log");
// exec("echo  >>
/var/log/asterisk/webfax.log");
}
?>
--
_
-- 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] Sytem Commands not executing

[Tim King]

When using the following my sytems commands to not seem to execute.

[outboundfax]
; exten => s,1,NoOp(send a fax)
exten => s,1,Set(FAXOPT(filename)=${FAXFILE})
exten => s,n,Set(FAXOPT(ecm)=yes)
exten => s,n,Set(FAXOPT(headerinfo)=${FAXHEADER})
exten => s,n,Set(FAXOPT(localstationid)=${LOCALID})
exten => s,n,Set(FAXOPT(maxrate)=14400)
exten => s,n,Set(FAXOPT(minrate)=2400)
exten => s,n,SendFAX(${FAXFILE},d)
exten => s,n,System(/usr/bin/php /var/lib/asterisk/bin/faxnotify.php INIT
"${EMAIL}" "${DESTINATION}" "${TIMESTAMP}" "NO_STATUS"
exten => h,1,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten => h,n,NoOp(FaxStatus : ${FAXSTATUS})
exten => h,n,NoOp(FaxStatusString : ${FAXSTATUSSTRING})
exten => h,n,NoOp(FaxError : ${FAXERROR})
exten => h,n,NoOp(RemoteStationID : ${REMOTESTATIONID})
exten => h,n,NoOp(FaxPages : ${FAXPAGES})
exten => h,n,NoOp(FaxBitRate : ${FAXBITRATE})
exten => h,n,NoOp(FaxResolution : ${FAXRESOLUTION})
exten => h,n,System(/usr/bin/php /var/lib/asterisk/bin/faxnotify.php NOTIFY
"${EMAIL}" "${DESTINATION}" "${TIMESTAMP}" "${FAXSTAT
; end of outboundfax context
I added this to make sure that it was not permissions.

exten => 777,1,System(/bin/touch /var/lib/asterisk/bin/tesfile.txt)
exten => 777,n,Playback(vm-goodbye)
exten => 777,n,Hangup()
When I dial 777 the test file is created in the folder where faxnotify.php
exists.

If I copy the data I see in the console and paste it to command prompt the
faxnotify script does execute and everything works.
/var/lib/asterisk/bin/faxnotify.php NOTIFY "t...@compnetwork.net"
"6165551212" "20/08/11 : 06:55:11" "FAX_SUCCESS" "1"
--
_
-- 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] One Way Audio

[Tim King]

Still not working now that audio is restored jitter makes it inaudible?  I
am ready to move this to commercial if someone can tell me how I need to pay
for support,

Thanks

Tim

On Thu, Mar 10, 2011 at 10:19 AM, Tim King  wrote:

> It looks like the issue was my provider enforcing a codec translation that
> was not working.
>
>
> On Thu, Mar 10, 2011 at 9:21 AM, Satish Patel wrote:
>
>> Also it could be the routing issue as well.
>>
>> --
>> Sent from my iPhone
>>
>> On Mar 9, 2011, at 7:43 PM, Duncan Turnbull 
>> wrote:
>>
>> So that suggests audio is flowing as follows in a unidirectional manner
>>
>> 199.173.66.22.53102 > 74.204.4.5.11732  IP 74.204.4.5.11732 >
>> 209.216.2.203.60362
>>
>>
>> Somewhere near the end this pops up which is slightly different, I am
>> guessing 74.204.4.5 is your asterisk box
>>
>> 19:18:36.389548 IP 74.204.4.5.11732 > 174.133.195.194.18364: UDP, length
>> 172
>>
>> I am not sure why this is happening or if its still part of the same
>> conversation
>>
>> Overall it looks a bit like the asterisk box thinks it needs to send rtp
>> to a different location than perhaps its meant to i.e. its asymmetric - you
>> can check the sdp in the sip invite to see where media is expected to be
>> sent to
>>
>> There is no rtp coming back from 209.216.2.203 so possibly this is device
>> that isn't meant to be part of the conversation and either doesn't exist or
>> is not expecting anything and thus not responding
>>
>> What are the addresses of the devices in this conversation? so that you
>> can match the traffic to device
>>
>> Cheers Duncan
>>
>> On 10/03/2011, at 1:20 PM, Tim King wrote:
>>
>> It looks like this:
>> 19:18:34.782016 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.789527 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.802064 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.809757 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.821855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.829598 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.842015 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.849764 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.861902 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.869568 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.881882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.889739 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.901882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.909612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.921984 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.929664 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.941855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.949589 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.962003 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.969592 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.981851 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.989543 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.002006 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.009973 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.022008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.029539 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.042071 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.049561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.062008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.069612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.081986 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.089519 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.101918 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.109722 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.122021 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18

Re: [asterisk-users] One Way Audio

[Tim King]

It looks like the issue was my provider enforcing a codec translation that
was not working.

On Thu, Mar 10, 2011 at 9:21 AM, Satish Patel  wrote:

> Also it could be the routing issue as well.
>
> --
> Sent from my iPhone
>
> On Mar 9, 2011, at 7:43 PM, Duncan Turnbull  wrote:
>
> So that suggests audio is flowing as follows in a unidirectional manner
>
> 199.173.66.22.53102 > 74.204.4.5.11732  IP 74.204.4.5.11732 >
> 209.216.2.203.60362
>
>
> Somewhere near the end this pops up which is slightly different, I am
> guessing 74.204.4.5 is your asterisk box
>
> 19:18:36.389548 IP 74.204.4.5.11732 > 174.133.195.194.18364: UDP, length
> 172
>
> I am not sure why this is happening or if its still part of the same
> conversation
>
> Overall it looks a bit like the asterisk box thinks it needs to send rtp to
> a different location than perhaps its meant to i.e. its asymmetric - you can
> check the sdp in the sip invite to see where media is expected to be sent to
>
> There is no rtp coming back from 209.216.2.203 so possibly this is device
> that isn't meant to be part of the conversation and either doesn't exist or
> is not expecting anything and thus not responding
>
> What are the addresses of the devices in this conversation? so that you can
> match the traffic to device
>
> Cheers Duncan
>
> On 10/03/2011, at 1:20 PM, Tim King wrote:
>
> It looks like this:
> 19:18:34.782016 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.789527 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.802064 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.809757 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.821855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.829598 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.842015 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.849764 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.861902 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.869568 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.881882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.889739 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.901882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.909612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.921984 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.929664 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.941855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.949589 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.962003 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.969592 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.981851 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.989543 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.002006 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.009973 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.022008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.029539 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.042071 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.049561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.062008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.069612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.081986 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.089519 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.101918 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.109722 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.122021 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.129590 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.141878 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.149709 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.161886 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.169561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.181879 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.189710 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 17

Re: [asterisk-users] One Way Audio

[Tim King]

My message with the configuration attached is awaiting moderator approval. I
will try to paste relevant data here.

*sip.conf*
[general]
context=inbound ;
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
dtmfmode = rfc2833
directmedia=no

[castlewire]
type=user
host=74.204.4.206
context=outb2
dtmfmode=rfc2833
username=castlewire
secret=1234
quallify=yes
canreinvite=no

[equity]
type=friend
host=dynamic
context=outb2
dtmfmode=rfc2833
username=equity
secret=1234
quallify=yes
canreinvite=no

[3000]
type=friend
host=dynamic
nat=yes
context=inbound
dtmfmode=rfc2833
username=3000
secret=1234
quallify=yes
canreinvite=no

[6168182996]
type=friend
host=dynamic
nat=yes
context=outb2
dtmfmode=rfc2833
username=6168182996
secret=1234
quallify=yes
canreinvite=no

[VITELITY]
type=friend
host=64.2.142.93
port=5060
dtmfmode=auto
context=inbound

[QWEST_OUT]
type=friend
host=67.135.79.80
port=5060
dtmfmode=inband

[QWEST8XX_IN]
type=friend
host=67.135.79.199
port=5060
context=qwest800

[DIDX1]
type=peer
host=67.15.128.14
context=inbound
canreinvite=no

[DIDX2]
type=peer
host=67.15.128.18
context=inbound
canreinvite=no

[DIDX3]
type=peer
host=208.44.220.237
context=inbound
canreinvite=no

[DIDX4]
type=peer
host=208.44.220.234
context=inbound
canreinvite=no

[DIDX5]
type=peer
host=209.62.66.242
context=inbound
canreinvite=no

[DIDX6]
type=peer
host=64.246.22.119
context=inbound
canreinvite=no

[DIDX7]
type=peer
host=70.84.58.18
context=inbound
canreinvite=no

[DIDX8]
type=peer
host=174.133.195.194
context=inbound
canreinvite=no

*iax.conf*

[general]
bandwidth=low
disallow=all
allow=ulaw
allow=alaw
jitterbuffer=no
forcejitterbuffer=no
autokill=yes

register=equity_out:1234@74.204.4.166
;register => IAX2/castlewire_trix:1234@74.204.4.206

[CASTLEWIRE]
type=friend
;host=74.204.4.206
host=dynamic
trunk=yes
auth=md5,plaintext,rsa
secret=1234
username=CASTLEWIRE
qualify=yes
context=outb2

[castlewire_trix]
type=friend
;host=74.204.4.206
host=dynamic
trunk=yes
auth=md5,plaintext,rsa
secret=1234
username=castlewire_trix
qualify=yes
context=outb2
requirecalltoken=no

[equity]
type=friend
host=dynamic
context=equity-fix
secret=1234
username=default
channels=10
trunk=yes
timezone=America/Detroit
qualify=yes
requirecalltoken=no

[equity_out]
type=friend
host=dynamic
context=outb2
secret=1234
username=equity_out
channels=10
trunk=yes
timezone=America/Detroit
qualify=yes
requirecalltoken=no

*extensions.conf*

[inbound]
;Equity Logistics
;exten => 6168182400,1,Dial(IAX2/equity/${EXTEN})
;exten => 6168182400,n,Hangup()
;exten => 8182400,1,Dial(IAX2/equity/${EXTEN})
;exten => 8182400,n,Hangup()

exten => 6168182400,1,Dial(SIP/equity/${EXTEN})
exten => 6168182400,n,Hangup()

exten => 6168182996,1,Dial(SIP/${EXTEN})
exten => 6168182996,n,Hangup()
;exten => 6168182996,1,Answer()
;exten => 6168182996,n,Milliwatt()

exten => 3000,1,Dial(SIP/${EXTEN})
exten => 3000,n,Hangup()

;CASTELWIRE NUMBERS
exten => 6168182000,1,Dial(IAX2/castlewire_trix/${EXTEN})
exten => 6168182000,n,Hangup()

;exten => 6168182000,1,Dial(SIP/4403712250@12.194.10.18)
;exten => 6168182000,n,Hangup()


exten => 6168182999,1,Set(portnum=${CALLERID(rdnis)})
exten => 6168182999,n,Set(cutNum=${CUT(portnum|\-|6)})
exten => 6168182999,n,Dial(SIP/${cutNum})
exten => 6168182999,n,Hangup()
--
_
-- 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] One Way Audio

[Tim King]

th 172
19:18:36.049761 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.061939 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:36.069719 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.082035 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:36.089685 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.102040 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:36.109568 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.122062 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:36.129698 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.142063 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:36.149589 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.162024 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:36.169637 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.181959 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:36.189543 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.202018 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:36.202027 IP 199.173.66.22.53103 > 74.204.4.5.11733: UDP, length 60
19:18:36.209751 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.222036 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:36.229544 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.249522 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.269699 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.289586 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.309528 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.329544 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.349672 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.369715 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:36.389548 IP 74.204.4.5.11732 > 174.133.195.194.18364: UDP, length 172
19:18:36.409578 IP 74.204.4.5.11732 > 174.133.195.194.18364: UDP, length 172
19:18:36.429550 IP 74.204.4.5.11732 > 174.133.195.194.18364: UDP, length 172


On Wed, Mar 9, 2011 at 7:01 PM, Duncan Turnbull wrote:

> Can you do a tcpdump to look at the rtp streams on your box and check they
> are both generating and aiming at the right places
>
> IAX will have no issue with NAT/firewall but SIP / RTP can.
>
> tcpdump -nn udp and portrange 1-2
> (pick your portrange if its operating on something else)
>
> Should show you mad lines of rtp going backwards and forwards (like below)
> when there is a conversation in place. If you can see it being sent from the
> asterisk box but not heard by the client then either try a different client,
> or something is blocking the return leg to your client
>
> 13:00:21.309139 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length
> 172
> 13:00:21.328703 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length
> 172
> 13:00:21.348572 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length
> 172
> 13:00:21.369096 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length
> 172
> 13:00:21.388572 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length
> 172
>
> Cheers Duncan
>
> On 10/03/2011, at 12:26 PM, Tim King wrote:
>
> Thank you I have also tried those settings. The main thing is coming from
> my voip provider all I am doing is bridging the calls to two other devices
> (1 trixbox and 1 digium aa50) via IAX trunks. Both devices are answering
> with an IVR and when I call in I can not hear the IVR. However if I call
> directly to a SIP client the person answering the SIP phone can hear me but
> I can not hear them at all.  Its definately not a NAT issue which is what
> makes it even more confusing. When the call is in place a sip show channels
> shows me both lefs of the call and they are both using either alaw or ulaw
> so it should not be a codec translation issue either.
>
> On Wed, Mar 9, 2011 at 6:19 PM, Satish Patel wrote:
>
>
>
> --
> _
> -- 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] One Way Audio

[Tim King]

Thank you I have also tried those settings. The main thing is coming from my
voip provider all I am doing is bridging the calls to two other devices (1
trixbox and 1 digium aa50) via IAX trunks. Both devices are answering with
an IVR and when I call in I can not hear the IVR. However if I call directly
to a SIP client the person answering the SIP phone can hear me but I can not
hear them at all.  Its definately not a NAT issue which is what makes it
even more confusing. When the call is in place a sip show channels shows me
both lefs of the call and they are both using either alaw or ulaw so it
should not be a codec translation issue either.

On Wed, Mar 9, 2011 at 6:19 PM, Satish Patel  wrote:

> What about your  sip clients? Are they on public network?
>
> Try on sip.conf
>
> Nat=no/yes
>
> conreinvite=yes/no
>
> --
> Sent from my iPhone
>
> On Mar 9, 2011, at 6:11 PM, Tim King  wrote:
>
> IPTBALES is off and I have all firewalls disabled. All network elements
> currently involved have public IP's assigned to them. My main asterisk box
> has a public IP. I have multiple trunks to voip peers for inbound and
> outbound calls which are also all public IP's. My two clients are trunked
> via IAX and one is a Trixbox and the other is a digium AA50 which both also
> have public IP's assigned to them.
>
> On Wed, Mar 9, 2011 at 6:04 PM, Adolphe Cher-Aime < 
> achera...@gmail.com> wrote:
>
>> How is your network is organized? Is your server behind a firewal, about
>> iptables ?
>>
>>
>>
>>
>> On Wed, Mar 9, 2011 at 5:40 PM, Tim King < 
>> t...@compnetwork.net> wrote:
>>
>>> I am having trouble with no return audio on inbound calls. I have been
>>> working on this for 18 hours and even built a fresh server and moved
>>> everything over and I am getting the same results. I need someone that can
>>> help get this resolved tonight. I can provide access to all machines
>>> involved.
>>>
>>> Please email me at tim.compnetwork@gmail.comif 
>>> you can help.
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by <http://www.api-digital.com>
>>> http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>   <http://www.asterisk.org/hello>
>>> http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   <http://lists.digium.com/mailman/listinfo/asterisk-users>
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>>
>> --
>> *Adolphe CHER-AIME
>> Network / VoIP  Engineer
>> CCNA, CCNA VOICE, Global VSAT Forum Certified
>> (509) 3449-4280*
>>
>
> --
> _
> -- 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>
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
><http://lists.digium.com/mailman/listinfo/asterisk-users>
> 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] One Way Audio

[Tim King]

IPTBALES is off and I have all firewalls disabled. All network elements
currently involved have public IP's assigned to them. My main asterisk box
has a public IP. I have multiple trunks to voip peers for inbound and
outbound calls which are also all public IP's. My two clients are trunked
via IAX and one is a Trixbox and the other is a digium AA50 which both also
have public IP's assigned to them.

On Wed, Mar 9, 2011 at 6:04 PM, Adolphe Cher-Aime wrote:

> How is your network is organized? Is your server behind a firewal, about
> iptables ?
>
>
>
>
> On Wed, Mar 9, 2011 at 5:40 PM, Tim King  wrote:
>
>> I am having trouble with no return audio on inbound calls. I have been
>> working on this for 18 hours and even built a fresh server and moved
>> everything over and I am getting the same results. I need someone that can
>> help get this resolved tonight. I can provide access to all machines
>> involved.
>>
>> Please email me at tim.compnetw...@gmail.com if you can help.
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> *Adolphe CHER-AIME
> Network / VoIP  Engineer
> CCNA, CCNA VOICE, Global VSAT Forum Certified
> (509) 3449-4280*
>
--
_
-- 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] One Way Audio

[Tim King]

I am having trouble with no return audio on inbound calls. I have been
working on this for 18 hours and even built a fresh server and moved
everything over and I am getting the same results. I need someone that can
help get this resolved tonight. I can provide access to all machines
involved.

Please email me at tim.compnetw...@gmail.com if you can help.
--
_
-- 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] SIP Load Balancing

[Tim King]

It seems that the
GotoIf(${SET(DB(avoics/route)=$[!0${DB(avoics/route)}])}?route1:route2) is
always returning false as if the SET command is not returning a value nor is
it changing the value in the DB.
Will this not work because I am running Asterisk 1.4.25.1??

On Thu, Oct 28, 2010 at 3:15 PM, Tim King  wrote:

> I updated it as follows and I am still only getting the SayNumber(2)
>
> [tim]
>
> exten =>
> _X.,1,GotoIf(${SET(DB(avoics/route)=$[!0${DB(avoics/route)}])}?route1:route2)
> exten => _X.,n(route1),SayNumber(1)
> exten => _X.,n,Hangup()
> exten => _X.,n(route2),SayNumber(2)
> exten => _X.,n,Hangup()
>
>
>
>
> On Thu, Oct 28, 2010 at 3:05 PM, Tilghman Lesher wrote:
>
>> On Thursday 28 October 2010 13:32:51 Tim King wrote:
>> > Thanks For The replies. I have tried piecing the samples together. Just
>> > for testing purposes i have created the following.
>> >
>> > [test]
>> > exten =>
>> > _X.,1,GotoIf(${SET(DB(avoics/route)=$[!0${DB(avoics/route)}])}?route1:ro
>> > ute2) exten => _X.,n(route1),Set(DB(avoics/route)=1)
>> > exten => _X.,n,SayNumber(1)
>> > exten => _X.,n,Hangup()
>> > exten => _X.,n(route2),Set(DB(avoics/route)=0)
>> > exten => _X.,n,SayNumber(2)
>> > exten => _X.,n,Hangup()
>> >
>> > The idea is if I continue dialing any number into this context I should
>> > hear 1 2 1 2 1 2
>> >
>> > Currently it is skipping to 2 as it should be since my database shows:
>> > /avoics/route  : 1
>> >
>> > It appears there is something wrong with my set command?
>>
>> You can drop your separate Set application.  The SET() dialplan function
>> does the alternation for you.
>>
>> --
>> 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] SIP Load Balancing

[Tim King]

I updated it as follows and I am still only getting the SayNumber(2)

[tim]
exten =>
_X.,1,GotoIf(${SET(DB(avoics/route)=$[!0${DB(avoics/route)}])}?route1:route2)
exten => _X.,n(route1),SayNumber(1)
exten => _X.,n,Hangup()
exten => _X.,n(route2),SayNumber(2)
exten => _X.,n,Hangup()



On Thu, Oct 28, 2010 at 3:05 PM, Tilghman Lesher  wrote:

> On Thursday 28 October 2010 13:32:51 Tim King wrote:
> > Thanks For The replies. I have tried piecing the samples together. Just
> > for testing purposes i have created the following.
> >
> > [test]
> > exten =>
> > _X.,1,GotoIf(${SET(DB(avoics/route)=$[!0${DB(avoics/route)}])}?route1:ro
> > ute2) exten => _X.,n(route1),Set(DB(avoics/route)=1)
> > exten => _X.,n,SayNumber(1)
> > exten => _X.,n,Hangup()
> > exten => _X.,n(route2),Set(DB(avoics/route)=0)
> > exten => _X.,n,SayNumber(2)
> > exten => _X.,n,Hangup()
> >
> > The idea is if I continue dialing any number into this context I should
> > hear 1 2 1 2 1 2
> >
> > Currently it is skipping to 2 as it should be since my database shows:
> > /avoics/route  : 1
> >
> > It appears there is something wrong with my set command?
>
> You can drop your separate Set application.  The SET() dialplan function
> does the alternation for you.
>
> --
> 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] SIP Load Balancing

[Tim King]

Its not so much me load balancing but the carrier requires that every other
call I send goes to the other address..

On Thu, Oct 28, 2010 at 2:36 PM, Barry Miller wrote:

> On Thu, Oct 28, 2010 at 11:08:12AM -0400, Tim King wrote:
> > I have a very simple setup with two SIP routes to my carrier. I need to
> have
> > every other phone call placed to that carrier go to a different address.
> >
> > This is what I need the call flow to look like. I have spent many hours
> > searching and have not found a working example.
> > Call1  exten => 
> > NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8
> >
> > )
> > Call2  exten => 
> > NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4
> >
> > )
> > Call3  exten => 
> > NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8
> >
> > )
> > Call4  exten => 
> > NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4
> >
> > )
> > Call5  exten => 
> > NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8
> >
> > )
> > Call6  exten => 
> > NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4
> >
> > )
> > Call7  exten => 
> > NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8
> >
> > )
> > Call8  exten => 
> > NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4
> >
> > )
> > ..
>
> If your goal is really load balancing, not just alternating between
> providers, you might look at the GROUP* functions.  Otherwise, if you
> hit a stretch where you have, e.g., several even-numbered calls of long
> duration mixed with short odd-numbered calls, most of your traffic will
> wind up on the same route.
>
> --
> Barry
>
> --
> _
> -- 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] SIP Load Balancing

[Tim King]

Thanks For The replies. I have tried piecing the samples together. Just for
testing purposes i have created the following.

[test]
exten =>
_X.,1,GotoIf(${SET(DB(avoics/route)=$[!0${DB(avoics/route)}])}?route1:route2)
exten => _X.,n(route1),Set(DB(avoics/route)=1)
exten => _X.,n,SayNumber(1)
exten => _X.,n,Hangup()
exten => _X.,n(route2),Set(DB(avoics/route)=0)
exten => _X.,n,SayNumber(2)
exten => _X.,n,Hangup()

The idea is if I continue dialing any number into this context I should hear
1 2 1 2 1 2

Currently it is skipping to 2 as it should be since my database shows:
/avoics/route  : 1

It appears there is something wrong with my set command?





On Thu, Oct 28, 2010 at 2:15 PM, Tilghman Lesher  wrote:

> On Thursday 28 October 2010 13:06:00 Gordon Henderson wrote:
> > On Thu, 28 Oct 2010, Tim King wrote:
> > > On Thu, Oct 28, 2010 at 11:24 AM, Roger Burton West
> wrote:
> > >> On Thu, Oct 28, 2010 at 11:08:12AM -0400, Tim King wrote:
> > >>> I have a very simple setup with two SIP routes to my carrier. I need
> > >>> to
> > >>
> > >> have
> > >>
> > >>> every other phone call placed to that carrier go to a different
> > >>> address.
> > >>
> > >> I think what you need to do here is check/set a variable in the
> > >> astdb.
> > >>
> > >> (If the variable is 1, set it to 2 and route via A; otherwise, set it
> > >> to 1 and route via B.)
> > >>
> > >> Translation of this to dialplan logic is left as an exercise for the
> > >> student.
> > >
> > > Sorry for the confusion, but the last sentence throws me off.
> > > "Translation of this to dialplan logic is left as an exercise for the
> > > student." Is this example from some sort of book or is this a way of
> > > saying I am left to figure the rest out??
> > >
> > > I was hoping to find a simple example of how this works.
> >
> > It's a way of leafing you to figure the rest out.
> >
> > It's a bastardised version of a quote from many textbooks - along the
> > lines of "implementation is left as an excercise to the student" - ie.
> > this is the method in general terms, you write nuts & bolts of the code.
> >
> > One reference to it might be:
> >
> >http://catb.org/jargon/html/E/exercise--left-as-an.html
> >
> > Roger has told you how to do it - use a variable kept in the astdb and
> > alternate it
> >
> > In pseudo code:
> >
> >if (switch == 1)
> >  Dial (SIP/provider1/number)
> >  switch = 0
> >else
> > Dial (SIP/provider2/number
> > switch = 1
> >endif
> >
> > Now your task is write the actual dialplan. Or you can pay me or Roger
> > to do it for you if you like, but really, it's only a few lines of
> > dialplan.
>
> GotoIf(${SET(DB(sw/provider)=$[!0${DB(sw/provider)}])}?provider1:provider2)
>
> --
> 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] SIP Load Balancing

[Tim King]

Sorry for the confusion, but the last sentence throws me off. "Translation
of this to dialplan logic is left as an exercise for the
student." Is this example from some sort of book or is this a way of saying
I am left to figure the rest out??

I was hoping to find a simple example of how this works.

On Thu, Oct 28, 2010 at 11:24 AM, Roger Burton West wrote:

> On Thu, Oct 28, 2010 at 11:08:12AM -0400, Tim King wrote:
> >I have a very simple setup with two SIP routes to my carrier. I need to
> have
> >every other phone call placed to that carrier go to a different address.
>
> I think what you need to do here is check/set a variable in the astdb.
>
> (If the variable is 1, set it to 2 and route via A; otherwise, set it to
> 1 and route via B.)
>
> Translation of this to dialplan logic is left as an exercise for the
> student.
>
> R
>
> --
> _
> -- 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] SIP Load Balancing

[Tim King]

I have a very simple setup with two SIP routes to my carrier. I need to have
every other phone call placed to that carrier go to a different address.

This is what I need the call flow to look like. I have spent many hours
searching and have not found a working example.
Call1  exten => NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8
)
Call2  exten => NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4
)
Call3  exten => NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8
)
Call4  exten => NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4
)
Call5  exten => NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8
)
Call6  exten => NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4
)
Call7  exten => NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8
)
Call8  exten => NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4
)
..
-- 
_
-- 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 SS7 Sigtran Protocol

[Tim King]

These guys are pretty close. http://www.ss7box.com/


On Wed, Nov 4, 2009 at 4:16 AM, Khaled W Chehab wrote:

>  Dears,
>
>
>
> Do Asterisk support SS7 SIGTRAN(SS7 over IP) protocol ?
>
> And how to integrate
>
>
>
>
>
> Regards
>
>
>
>
>
> *Khaled  Chehab*
>
> *   NGN Eng.*
>
>
>
>  [image: Untitled]
>
> * Operations Office - Lebanon*
>
>  Office : +961 1 868686 ext 115
>
>  Mobile: +961 3 045212
>
>  E-mail:  kche...@xplorium.com 
>
>  MSN ID :khalidche...@hotmail.com
>
>  Web Site: http://www.Xplorium.com
>
>
>
>
>
>
> --
> *
>
> No employee or agent is authorized to conclude any binding agreement on 
> behalf of Xplorium with another party by e-mail without express written 
> confirmation by an officer of Xplorium. Any views expressed by an individual 
> in this electronic message do not necessarily reflect views of Xplorium or 
> its subsidiaries and associates.
>
>
> This electronic message and its attachments are solely addressed to the 
> addressee(s), and contain confidential information protected from disclosure 
> belonging to Xplorium.
>
>
> If you are not the intended addressee of this electronic message and its 
> attachments, kindly delete it immediately from your system and notify the 
> sender by electronic mail. You must not copy this message or attachment or 
> disclose its content to any other person.
>
>
> Xplorium does not guarantee the integrity of this electronic message and any 
> of its attachments, or that they are free from computer viruses or other 
> defects.
> *
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> 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 --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP 18x Messages

[Tim King]

I thought that was it and tried each setting and did not see any change on
the line.

On Wed, Oct 28, 2009 at 3:58 PM, Kevin P. Fleming wrote:

> Tim King wrote:
> > When I make an outbound call I hear a half of a ring and than silence
> > until the call opens up.
> >
> > It seems asterisk is sending a 183 after the 180 message. My CPE device
> > does not support multiple 18x messages in the same call setup.  When we
> > receive the 180 we present ring back to the phone, but when we receive
> > the 183 we get confused and stop the ring back tone, but do not open up
> > the early media path for the ring back to be played from the network.
> >
> > In Metaswitch the configuration knob to correct this is “Superfluous 18x
> > messages”, I don’t know what it takes to configure Asterisk that way.
> > Can anyone help with this.
>
> Check out the 'progressinband' configuration option.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kpflem...@digium.com
> Check us out at www.digium.com & www.asterisk.org
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> 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 --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Having a heck of a time

[Tim King]

Did you use ./Setup dahdi when installing the wanpipe drivers?

http://wiki.sangoma.com/wanpipe-linux-asterisk-dahdi
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SIP 18x Messages

[Tim King]

When I make an outbound call I hear a half of a ring and than silence until
the call opens up.

It seems asterisk is sending a 183 after the 180 message. My CPE device does
not support multiple 18x messages in the same call setup.  When we receive
the 180 we present ring back to the phone, but when we receive the 183 we
get confused and stop the ring back tone, but do not open up the early media
path for the ring back to be played from the network.

In Metaswitch the configuration knob to correct this is “Superfluous 18x
messages”, I don’t know what it takes to configure Asterisk that way. Can
anyone help with this.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] DS3 Interface

[Tim King]

I have started the open source project to get this going. I am working
directly with the manufacture to form agreements and gain access to the
hardware and source code for their drivers. The list price for the card is
$4,995.00 USD. I will keep everyone posted and will have site for
development and forums up soon.

Thanks for the support


Tim King
CEO

7589 Cottonwood Drive   Suite C
Jenison, MI  49428

Phone 616.301.3290Fax: 616.667.1104

Website: http://www.compnetwork.net



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Baji
Panchumarti
Sent: Tuesday, October 09, 2007 12:07 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DS3 Interface

  On 10/9/07, Brian West  wrote:

> http://www.imagestream.com/PCI_921-CDS.html
>
> [...]

 off-topic :

 I saw Imagestream at the Ohio Linuxfest a weekend ago.

 Also picked up a few literature bags by Digium  :-)

--

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

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--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] DS3 Interface

[Tim King]

If it hasn't already been done I am looking to put together a team to write
drivers for this DS3 card to interface asterisk.

 

http://www.imagestream.com/PCI_921-CDS.html

 

The card itself seems reasonable and I believe we can make it work. As soon
as I have positive feedback to begin the project I will put a server on the
net with a card in it.

 

 

Let's make this happen.

 

 

 

Tim King

CEO

 <http://www.compnetwork.net/> CNS_LOGO_Beveled

7589 Cottonwood Drive   Suite C

Jenison, MI  49428

 

Phone 616.301.3290Fax: 616.667.1104

 

Website:  <http://www.compnetwork.net/> http://www.compnetwork.net

 

 

<>___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Assistance needed.

[Tim King]

I am looking for help from someone familiar with using asterisk and openser
to build a rather large VOIP network. I have 6 servers in place each with
their own purpose. I will give a brief summary and hopefully someone out
there is able to help be finalize this dialplan. I have six servers in
place. One has 4 port T1 to handle trunks to PSTN. One has 2 port T1 and is
to be used for SS7 links. Third is to be used as primary sip machine running
openser. 4th is voicemail server using asterisk. 5th is asterisk machine to
be used for VPBX customers. 6th is to hold all mysql databases with
configuration from all servers to providing a single point of provisioning.
If anyone out here thinks they can help please advise.


___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Configuration assistance needed.

[Tim King]

I have a fairly large system to configure. I was hoping to find someone
locally to employ for this project but remote configuration is considerable.
Pleas let me know if you are interested and have the time.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Seeking Asterisk Solution For mid sized corp.

[Tim King]

Hey Guys,

 

   
I have a new task to tackle. I need to make asterisk save me as much $$$ From
Ma Bell As possible. Here is the Scenario. I have 50 storefronts throughout the
state of Michigan.
Each location has 2 voice lines in a hunt group and a FAX Line with DSL on it.
I also have a large Asterisk Box here at the corporate office Currently with
one PRI Terminated to it. I obviously need to keep at least one analog line on
site at each location for the DSL to work. But how could I cut the other two
lines. I was trying to figure out how to keep my hunt groups working though.
Would I have to port my existing numbers over and have asterisk do all the
hunting and forward to a new unlisted number using a 2ns PRI Channel. Or would
it be easier to just use VOIP for everything and only use the Pots line to
bring in the DSL. For e911 purposes I do need the stores to still be able to
pick up the local POTS line and call out. The other issue is the faxes, if I
could use asterisk to distribute the faxes and use voip from the stores to send
them out via asterisk I could save thousands there alone. Let me know if anyone
has had a similar setup. Thanks in advance!!

 

Regards

 

Tim King

 

 






___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk CMD MySQL

[Tim King]

What version of asterisk do I need to be running for this to
work? I have 1.0.9 running and when I try to install asterisk-addons from CVS I
get app_addon_sql_mysql.c23:19: mysql.h: No such file or directory.

So of course it fails to install that add-on. What am I
missing? I can find info on how to use it, but not much on getting it working.

 

Thanks Again

 

Tim






___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Problem setting up TDM22B card

[Tim King]

Try lspci -vb

See if you can find you digium card and what interrupt it is running on.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of somesh s
Sent: Friday, September 23, 2005 5:49 AM
To: Asterisk Users
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Problem setting up TDM22B card

Hi All,

I have the problem setting up TDM22B card.

Steps what I have followed are:

[1] compiled zaptel-1.0.9.2 & installed the same.

[2] modprobe wcfxo
/lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect module
parameters, including invalid IO or IRQ parameters.
  You may find more information in syslog or the
output from dmesg
/lib/modules/2.4.20-8/misc/wcfxo.o: insmod
/lib/modules/2.4.20-8/misc/wcfxo.o failed
/lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo
failed

I tried to put the card in another PCI slot also...but
same result!

What should I do? Please help me in this regard.

Regards,
Somesh S. Shanbhag




__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] AGI Script to interact with ACCESS Databse and Set CID info on the fly.

[Tim King]

Well guys here comes the fun part. I have a Microsoft access
(VBA) application that interfaces with my SQL database. This app pulls of info from
the SQL record and than picks up the phone and dials that locations number. I
have purchased a few hundred NpaNxx’s for my own use. I want get into too
much detail there but no worries this is legal. I need to change my CID info on
the fly. So I am thinking it should be easy to make an AGI script that just sets
the CID info on a particular line using two variables being passed to it
$Line_No to tell it what line to set and than $CID to be the number to set on
that extension for that call. It also should be relatively simple to have the
access app take a look at the area code and phone number for the location being
called and pull a phone number from the NUMBERS table which has all of my
numbers in it and pass that over. The real question is how do we get Access to
speak to an AGI script. Has anyone done anything like this? Thanks a lot for
reading but this will be a fun one. 






___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Need Local HELP!!!

[Tim King]

I need to find someone to work with me in the Grand Rapids
Michigan Area. Someone good with Linux and Asterisk would be ideal. Please get
me contact info if you are interested.

 

Thanks

 

Tim






___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Outbound Extension problem

[Tim King]

The inbound call is answered by the digital receptionist and is than
transferred to this extension.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday, August 04, 2005 7:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Outbound Extension problem

Have to answer your inbound call first - I suspect

On Aug 4, 2005, at 5:28 PM, Tim King wrote:

> [macro-dialout-trunk]
>
> exten => s,1,GotoIf($[foo${ARG3} = foo]?3:2))   ; arg3 is pattern  
> password
>
> exten => s,2,Authenticate(${ARG3})
>
> exten => s,3,Macro(record-enable,${CALLERIDNUM},OUT)
>
> exten => s,4,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?7)  ;check  
> for CID override for exten
>
> exten => s,5,SetCallerID(${ECID${CALLERIDNUM}})
>
> exten => s,6,Goto(9)
>
> exten => s,7,GotoIf($[foo${OUTCID_${ARG1}} = foo]?9)  ;check for  
> CID override for trunk
>
> exten => s,8,SetCallerID(${OUTCID_${ARG1}})
>
> exten => s,9,SetGroup(OUT_${ARG1})
>
> exten => s,10,CheckGroup(${OUTMAXCHANS_${ARG1}})
>
> ; if we've used up the max channels, continue at 109 (n+101)
>
> exten => s,11,SetVar(DIAL_NUMBER=${ARG2})
>
> exten => s,12,SetVar(DIAL_TRUNK=${ARG1})
>
> exten => s,13,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the  
> proper dial string for this trunk
>
> exten => s,14,SetVar(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER})  ;  
> OUTNUM is the final dial number
>
> exten => s,15,Cut(custom=OUT_${ARG1},:,1)  ; Custom trunks are  
> prefixed with "AMP:"
>
> exten => s,16,GotoIf($[${custom} = AMP]?19)
>
> exten => s,17,Dial(${OUT_${ARG1}}/${OUTNUM})  ; Regular Trunk Dial
>
> exten => s,18,Goto(s-${DIALSTATUS},1)
>
>
>
> From: [EMAIL PROTECTED] [mailto:asterisk- 
> [EMAIL PROTECTED] On Behalf Of Jason Walker
> Sent: Thursday, August 04, 2005 6:23 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Outbound Extension problem
>
>
>
> Can you post your macro?
>
>
>
> Thanks.
>
>
>
> From: [EMAIL PROTECTED] [mailto:asterisk- 
> [EMAIL PROTECTED] On Behalf Of Tim King
> Sent: Thursday, August 04, 2005 2:56 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] Outbound Extension problem
>
>
>
> New problem, I figured out how to get the extension working and  
> internally it works just fine. If I pick up a phone and hit 501 my  
> cell starts ringing. However if an inbound caller dials that  
> extension Everything seems to stop when it trys to bridge the two  
> trunks together. Sound familiar to anyone?
>
>
>
> exten => 501,1,Macro(dialout-trunk,1,5551212)
>
> exten => 501,2,Wait,1
>
> exten => 501,3,Voicemail(300)
>
>
>
>
>
> Thanks
>
>
>
> Tim
>
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Outbound Extension problem

[Tim King]

[macro-dialout-trunk]

exten => s,1,GotoIf($[foo${ARG3} =
foo]?3:2))   ; arg3 is pattern password

exten => s,2,Authenticate(${ARG3})

exten =>
s,3,Macro(record-enable,${CALLERIDNUM},OUT)

exten =>
s,4,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?7)  ;check for CID override for
exten

exten =>
s,5,SetCallerID(${ECID${CALLERIDNUM}})

exten => s,6,Goto(9)

exten =>
s,7,GotoIf($[foo${OUTCID_${ARG1}} = foo]?9)  ;check for CID override for
trunk

exten =>
s,8,SetCallerID(${OUTCID_${ARG1}})

exten => s,9,SetGroup(OUT_${ARG1})

exten => s,10,CheckGroup(${OUTMAXCHANS_${ARG1}})

; if we've used up the max channels,
continue at 109 (n+101)

exten =>
s,11,SetVar(DIAL_NUMBER=${ARG2})

exten =>
s,12,SetVar(DIAL_TRUNK=${ARG1})

exten => s,13,AGI(fixlocalprefix) ;
this sets DIAL_NUMBER to the proper dial string for this trunk

exten =>
s,14,SetVar(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER})  ; OUTNUM is the
final dial number

exten =>
s,15,Cut(custom=OUT_${ARG1},:,1)  ; Custom trunks are prefixed with
"AMP:"

exten => s,16,GotoIf($[${custom} =
AMP]?19)

exten =>
s,17,Dial(${OUT_${ARG1}}/${OUTNUM})  ; Regular Trunk Dial

exten => s,18,Goto(s-${DIALSTATUS},1)

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Walker
Sent: Thursday, August 04, 2005
6:23 PM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
Outbound Extension problem



 

Can you post your macro?

 

Thanks.

 









From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim King
Sent: Thursday, August 04, 2005
2:56 PM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Outbound
Extension problem



 

New problem, I figured out how to get the extension working
and internally it works just fine. If I pick up a phone and hit 501 my cell
starts ringing. However if an inbound caller dials that extension Everything
seems to stop when it trys to bridge the two trunks together. Sound familiar to
anyone?

 

exten => 501,1,Macro(dialout-trunk,1,5551212)

exten => 501,2,Wait,1

exten => 501,3,Voicemail(300)

 

 

Thanks

 

Tim






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Outbound Extension problem

[Tim King]

New problem, I figured out how to get the extension working and
internally it works just fine. If I pick up a phone and hit 501 my cell starts
ringing. However if an inbound caller dials that extension Everything seems to
stop when it trys to bridge the two trunks together. Sound familiar to anyone?

 

exten => 501,1,Macro(dialout-trunk,1,5551212)

exten => 501,2,Wait,1

exten => 501,3,Voicemail(300)

 

 

Thanks

 

Tim






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Transfer to outside line.

[Tim King]

I tried this solution, although ti acts
like it is working it only rings once and than the call is just dead air. The
number I am forwarding to never rings. Anything else I may need to try?

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ariel Batista
Sent: Wednesday, August 03, 2005
3:40 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Transfer to outside line.



 



This is simple since your using AMP, you can create a
ring group to dial that number out for you.  First create your ring group
lets put number 200 for it (you can call it any number you want). where the
extension number goes just put there the phone number you want like
301212#  don't for get the # key after the number.





 





Then if it does not pickup you can send it to a voicemail
box or any other place you want it.





 





Ariel







- Original Message - 





From: Tim King 





To: 'Asterisk Users Mailing List -
Non-Commercial Discussion' 





Sent: Wednesday, August
03, 2005 10:12 AM





Subject: [Asterisk-Users]
Transfer to outside line.





 



Finally got everything up and run with the help of Manny
Wise last night. So I am setting up my digital assistant and getting down to
the task I need this box to perform the most. I need to have a custom app that
I can call that will take me pressing 2 at the menu and have it transfer the
call to a offsite phone number utilizing my Zap Trunk. I’m sure someone
has done this already. Anyone want to point me in the right direction?

 

Tim King







___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users








___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Transfer to outside line.

[Tim King]

Actually what I am wanting is for the
digital receptionist. So that if a user selects option 2 for “Internet Technical
support” It transfers the call to my offsite call center. Thanks

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Wednesday, August 03, 2005
11:56 AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Transfer to outside line.



 

I think what you want is called DISA
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA

DISA (Direct Inward System Access) Allows someone from outside the
telephone switch (PBX) to obtain an "internal" system dialtone and to
place calls from it as if they were placing a call from within the switch. A
user calls a number that connects to the DISA application and is given dialtone
and context.

Doug

At 09:12 AM 8/3/2005, you wrote:



Finally got everything up and run with the help of Manny
Wise last night. So I am setting up my digital assistant and getting down to
the task I need this box to perform the most. I need to have a custom app that
I can call that will take me pressing 2 at the menu and have it transfer the
call to a offsite phone number utilizing my Zap Trunk. Im sure someone has
done this already. Anyone want to point me in the right direction?
 
Tim King






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] WHat does it take

[Tim King]

Ok everyone, I received help from Manny last night and it would seem that
asterisk at home failed to properly configure itself from the 1.3 ISO. All
of the configurations were correct. We downloaded the AAH package and
reinstalled it over the top of itself and than recompiled the kernel. Upon
doing that everything just started working with all of the existing
configurations.  Thank for the help. On to trying to figure out how to
create a custom app to transfer calls to outside lines. Anyone have a
script?


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Transfer to outside line.

[Tim King]

Finally got everything up and run with the help of Manny
Wise last night. So I am setting up my digital assistant and getting down to
the task I need this box to perform the most. I need to have a custom app that
I can call that will take me pressing 2 at the menu and have it transfer the
call to a offsite phone number utilizing my Zap Trunk. I’m sure someone has
done this already. Anyone want to point me in the right direction?

 

Tim King






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: Re: [Asterisk-Users] WHat does it take

[Tim King]

M})
exten => s,6,GotoIf($[${lastcaller}]?7:13)
exten => s,7,SayDigits(${lastcaller})
exten => s,8,DigitTimeout(3)
exten => s,9,ResponseTimeout(7)
exten => s,10,Background(loligo/to-call-this-number)
exten => s,11,Background(allison7/press-1)
exten => s,12,Goto(15)
exten => s,13,Playback(loligo/from-unknown-caller)
exten => s,14,Macro(hangupcall)
exten => s,15,NoOp
exten => 1,1,Goto(from-internal,${lastcaller},1);
exten => i,1,Playback(vm-goodbye)
exten => i,2,Macro(hangupcall)
exten => t,1,Playback(vm-goodbye)
exten => t,2,Macro(hangupcall)


;

; Inbound Contexts [from]
;


[from-sip-external]

;give external sip users congestion and hangup
exten => _.,1,AbsoluteTimeout(15)
exten => _.,2,Congestion 
exten => _.,3,Hangup

[from-internal]
;allow phones to use applications
include => app-directory
include => app-dnd
include => app-callforward
include => app-callwaiting
include => app-messagecenter
include => app-calltrace
include => parkedcalls
include => from-internal-custom
;allow phones to dial other extensions
include => ext-fax
include => ext-local
include => ext-group
include => ext-queues
include => ext-zapbarge
include => ext-meetme
include => ext-record
include => ext-test
;allow phones to access trunks
include => outbound-allroutes
exten => s,1,Macro(hangupcall)
exten => h,1,Macro(hangupcall)

;

; Extension Contexts [ext]
;


[ext-zapbarge]
exten => 888,1,SetGroup(${CALLERIDNUM})
exten => 888,2,Answer
exten => 888,3,Wait(1)
exten => 888,4,ZapBarge
exten => 888,5,Hangup

[ext-meetme]
exten => _8X,1,Answer
exten => _8X,2,Wait(1)
exten => _8X,3,GotoIf($[${CALLERIDNUM} = ${EXTEN:1}]?5:4)
exten => _8X,4,MeetMe(${EXTEN}|sM)
exten => _8X,5,MeetMe(${EXTEN}|asM)

exten => _8XX,1,Answer
exten => _8XX,2,Wait(1)
exten => _8XX,3,GotoIf($[${CALLERIDNUM} = ${EXTEN:1}]?5:4)
exten => _8XX,4,MeetMe(${EXTEN}|sM)
exten => _8XX,5,MeetMe(${EXTEN}|asM)

exten => _8XXX,1,Answer
exten => _8XXX,2,Wait(1)
exten => _8XXX,3,GotoIf($[${CALLERIDNUM} = ${EXTEN:1}]?5:4)
exten => _8XXX,4,MeetMe(${EXTEN}|sM)
exten => _8XXX,5,MeetMe(${EXTEN}|asM)

exten => _8,1,Answer
exten => _8,2,Wait(1)
exten => _8,3,GotoIf($[${CALLERIDNUM} = ${EXTEN:1}]?5:4)
exten => _8,4,MeetMe(${EXTEN}|sM)
exten => _8,5,MeetMe(${EXTEN}|asM)


[ext-fax]
exten => s,1,Answer
exten => s,2,Goto(in_fax,1)
exten => in_fax,1,GotoIf($[${FAX_RX} = system]?2:analog_fax,1)
exten => in_fax,2,Macro(faxreceive)
exten => in_fax,3,system(tiff2ps -2eaz -w 8.5 -h 11 ${FAXFILE} | ps2pdf -
${FAXFILE}.pdf)
exten => in_fax,4,system(mime-construct --to ${EMAILADDR} --subject "Fax
from ${CALLERIDNUM} ${CALLERIDNAME}" --attachment ${CALLERIDNUM}.pdf --type
application/pdf --file ${FAXFILE}.pdf)
exten => in_fax,5,system(rm ${FAXFILE} ${FAXFILE}.pdf)
exten => in_fax,6,Hangup
exten => analog_fax,1,GotoIf($[${FAX_RX} = disabled]?3:2)  ;if fax is
disabled, just hang up
exten => analog_fax,2,Dial(${FAX_RX},20,d)
exten => analog_fax,3,Hangup
;exten => out_fax,1,wait(7)
exten => out_fax,1,txfax(${TXFAX_NAME}|caller)
exten => out_fax,2,Hangup
exten => h,1,Hangup()

[ext-record]
exten => *77,1,Wait(2)
exten => *77,2,Record(${CALLERIDNUM}ivrrecording:wav) 
exten => *77,3,Wait(2)
exten => *77,4,Hangup
exten => *99,1,Playback(${CALLERIDNUM}ivrrecording) 
exten => *99,2,Wait(2) 
exten => *99,3,Hangup 

;this is where parked calls go if they time-out.  Should probably re-ring
[default]
include => ext-local
exten => s,1,Playback(vm-goodbye)
exten => s,2,Macro(hangupcall)

[ext-test]
exten => ,1,Goto(from-pstn,s,1)
exten => 666,1,Goto(ext-fax,in_fax,1)
exten => h,1,Macro(hangupcall)

;echo test
exten => *43,1,Answer
exten => *43,2,Wait(2)
exten => *43,3,Playback(demo-echotest)
exten => *43,4,Echo
exten => *43,5,Playback(demo-echodone)
exten => *43,6,Hangup

extensions_addtional.conf

[globals]
ZAPCHAN_300 = 1
ZAPCHAN_201 = 2
VM_PREFIX = *
RINGTIMER = 15
REGTIME = *
REGDAYS = *
RECORDEXTEN = ""
PARKNOTIFY = SIP/200
OUT_1 = ZAP/1
OUTPREFIX_1 = 
OUTMAXCHANS_1 = 
OUTCID_1 = 
OPERATOR = 
NULL = ""
IN_OVERRIDE = forcereghours
INCOMING = EXT-2007
FAX_RX_EMAIL = [EMAIL PROTECTED]
FAX_RX = system
FAX = 
E3007 = SIP
E300 = ZAP
E203 = SIP
E202 = SIP
E201 = ZAP
E2007 = IAX2
DIRECTORY_OPTS = e
DIRECTORY = last
DIAL_OUT = 9
DIAL_OPTIONS = tr
DIALOUTIDS = 1/
CALLFILENAME = ""
AFTER_INCOMING = EXT-3007

[ext-group]
include => ext-group-custom
exten => 1,1,Macro(rg-group,3

[Asterisk-Users] Zaptel.conf question

[Tim King]

# It must be in the module loading order

 

 

# Span 1: WCTDM/1 "Wildcard TDM400P REV I Board 2"


fxoks=1

fxoks=2

fxoks=3

fxoks=4

 

# Span 2: WCTDM/2 "Wildcard TDM400P REV I Board 3"


fxsks=5

fxsks=6

fxoks=7

fxoks=8

 

# Global data

 

loadzone   = us

defaultzone   = us

 

Is this creating a problem because of the two FXO ports
being in the middle of the FXS ports?

 

Thanks

 

Tim King

Network Engineer

Computer & Network Solutions LLC

1331 Plainfield Ave

Grand Rapids MI  49505

 

Phone: 800-669-3290



 






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] WHat does it take

[Tim King]

How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?

 






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] No Extensions

[Tim King]

Can somebody please help here. At least respond and call me
a moron. I have tried everything. I finally gave up and installed [EMAIL PROTECTED]
from the iso and I am back to the exact same problem. Everything seems to work
but my extensions are all busy. I used the AMP setup tool to add my zap
extensions. If I view the console this is what happens when I call form one
extension to the next. In the extensions setup when it asks what channel do I have
to use a 2 digit number or something? This is a Digium TDM22B card. All the
zaptel stuff seems to be working. And I can call out as well.

 

Thanks

 

Tim King

Network Engineer

Computer & Network Solutions LLC

1331 Plainfield Ave

Grand Rapids MI  49505

 

Phone: 800-669-3290



 






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Busy Extensions.

[Tim King]

Thanks for the response they are zap
extensions on Digium TDM40B and TDM22B

 

pbx*CLI> zap show channel 3

Channel: 3

File Descriptor: 13

Span: 1

Extension:

Dialing: no

Context: from-internal

Caller ID string: "Tim King"
<200>

Destroy: 0

InAlarm: 0

Signalling Type: FXO Kewlstart

Owner: 

Real: 

Callwait: 

Threeway: 

Confno: -1

Propagated Conference: -1

Real in conference: 0

DSP: no

Relax DTMF: no

Dialing/CallwaitCAS: 0/0

Default law: ulaw

Fax Handled: no

Pulse phone: no

Echo Cancellation: 128 taps unless TDM
bridged, currently OFF

Actual Confinfo: Num/0, Mode/0x

Actual Confmute: No

ctual Hookstate: Onhook

 

pbx*CLI> zap show channels

   Chan Extension  Context
Language   MusicOnHold

 pseudo    from-pstn   en

  1    from-pstn   en

  2    from-pstn   en

  3    from-internal   en

  4    from-internal   en

  5    from-internal   en

pbx*CLI>

 









From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bates, Curtis
Sent: Thursday, July 21, 2005 4:23
PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Busy
Extensions.



 



I am not familiar with your configuration,
so I am assuming that they are SIP extensions.  If they are and you do a
sip show peers, what is the status of the peers?





-Original Message-
From: Tim King [mailto:[EMAIL PROTECTED]]
Sent: Thursday, July 21, 2005 2:23
PM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Busy
Extensions.
Importance: High

I seem to have almost everything working now. The only
problem is all of my extensions seem to be busy. I can call out, but not in.
Can someone point me to the settings in the extensions file that could cause
this.

 

Thanks in advance guys.

 

Tim King

Network Engineer

Computer & Network Solutions LLC

1331 Plainfield Ave

Grand Rapids MI  49505

 

Phone: 800-669-3290

 

 





-
A.G. Edwards & Sons' outgoing and incoming e-mails are electronically
archived and subject to review and/or disclosure to someone other 
than the recipient.

-






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Busy Extensions

[Tim King]

Here is the output. These are Panasonic KX-TG2564’s.
Does something need to be set for the phones? I can call out fine, but all of
the extensions seem to be busy.

Starting simple switch on 'Zap/5-1'

    -- Executing Macro("Zap/5-1",
"exten-vm|[EMAIL PROTECTED]|200") in new stack

    -- Executing SetVar("Zap/5-1",
"FROMCONTEXT=exten-vm") in new stack

    -- Executing Macro("Zap/5-1",
"record-enable|200|IN") in new stack

    -- Executing GotoIf("Zap/5-1", "0 >
0?2:4") in new stack

    -- Goto (macro-record-enable,s,4)

    -- Executing GotoIf("Zap/5-1",
"0?5:8") in new stack

    -- Goto (macro-record-enable,s,8)

    -- Executing GotoIf("Zap/5-1",
"0?9:12") in new stack

    -- Goto (macro-record-enable,s,12)

    -- Executing DBget("Zap/5-1",
"RecEnable=RECORD-IN/200") in new stack

    -- DBget: varname=RecEnable, family=RECORD-IN, key=200

    -- DBget: Value not found in database.

    -- Executing SetVar("Zap/5-1",
"CALLFILENAME=20050721-034056-1121931654.5") in new stack

    -- Executing GotoIf("Zap/5-1",
"0?15:99") in new stack

    -- Goto (macro-record-enable,s,99)

    -- Executing NoOp("Zap/5-1", "NO
RECORDING NEEDED") in new stack

    -- Executing Macro("Zap/5-1",
"dial|15|tr|200") in new stack

    -- Executing GotoIf("Zap/5-1",
"0?4:2") in new stack

    -- Goto (macro-dial,s,2)

    -- Executing GotoIf("Zap/5-1", "0?4:3")
in new stack

    -- Goto (macro-dial,s,3)

    -- Executing SetCIDName("Zap/5-1",
"") in new stack

    -- Executing AGI("Zap/5-1",
"dialparties.agi") in new stack

    -- Launched AGI Script
/var/lib/asterisk/agi-bin/dialparties.agi

    -- AGI Script dialparties.agi completed, returning 0

    -- Executing NoOp("Zap/5-1", "Returned
from dialparties with no extensions to call") in new stack

    -- Executing SetVar("Zap/5-1",
"DIALSTATUS=BUSY") in new stack

    -- Executing GotoIf("Zap/5-1",
"0?s-BUSY|1") in new stack

    -- Executing GotoIf("Zap/5-1",
"0?s-BUSY|1") in new stack

    -- Executing NoOp("Zap/5-1", "Sending to
Voicemail box [EMAIL PROTECTED]") in new stack

    -- Executing Macro("Zap/5-1",
"vm|[EMAIL PROTECTED]|BUSY") in new stack

    -- Executing Goto("Zap/5-1",
"s-BUSY|1") in new stack

    -- Goto (macro-vm,s-BUSY,1)

    -- Executing VoiceMail("Zap/5-1",
"[EMAIL PROTECTED]") in new stack

    -- Playing 'vm-theperson' (language 'en')

    -- Playing 'digits/2' (language 'en')

    -- Playing 'digits/0' (language 'en')

    -- Playing 'digits/0' (language 'en')

    -- Playing 'vm-isonphone' (language 'en')

    -- Playing 'vm-intro' (language 'en')

  == Spawn extension (macro-vm, s-BUSY, 1) exited non-zero
on 'Zap/5-1' in macro 'vm'

  == Spawn extension (macro-exten-vm, s, 7) exited non-zero
on 'Zap/5-1' in macro 'exten-vm'

  == Spawn extension (from-internal, 200, 1) exited non-zero
on 'Zap/5-1'

    -- Executing Macro("Zap/5-1",
"hangupcall") in new stack

    -- Executing ResetCDR("Zap/5-1",
"w") in new stack

    -- Executing NoCDR("Zap/5-1", "") in
new stack

    -- Executing Wait("Zap/5-1", "5") in
new stack

  == Spawn extension (macro-hangupcall, s, 3) exited
non-zero on 'Zap/5-1' in macro 'hangupcall'

  == Spawn extension (from-internal, h, 1) exited non-zero
on 'Zap/5-1'

    -- Hungup 'Zap/5-1'

 






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Busy Extensions.

[Tim King]

I seem to have almost everything working now. The only
problem is all of my extensions seem to be busy. I can call out, but not in.
Can someone point me to the settings in the extensions file that could cause
this.

 

Thanks in advance guys.

 

Tim King

Network Engineer

Computer & Network Solutions LLC

1331 Plainfield Ave

Grand Rapids MI  49505

 

Phone: 800-669-3290

 

 






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] System Jsut hangs Up

[Tim King]

I took care of my earlier problem. But now if I call in it just
says goodbye, And on my extension no matter what I do it seems to just hang up
on me immediately. It’s a slackware 10.1 box with Digium 22b card. I am
running AMP so its mysql driven. I’m not seeing any errors. It just hangs
up.

 

Tim King

Network Engineer

Computer & Network Solutions LLC

1331 Plainfield Ave

Grand Rapids MI  49505

 

Phone: 800-669-3290



 






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] modprobe wcfxo fails.

[Tim King]

I was reading a thread where you were helping someone out
and noticed it ended without resolve. Was this issue ever taken care of?I seem
to be having the exact same problem.

 

Thanks

 

 

Tim King

Network Engineer

Computer & Network Solutions LLC

1331 Plainfield Ave

Grand Rapids MI  49505

 

Phone: 800-669-3290



 






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users