Re: [Asterisk-Users] VoicePulse broken?
Inbound is working here, no problems that I know of. Scott - Original Message - From: C. Sullivan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 20, 2004 12:52 PM Subject: [Asterisk-Users] VoicePulse broken? Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Asterisk on a dual processor machine
Try upgrading your kernel... This is the original shipped kernel and is very buggy. I would suggest getting yum It can be retrieved from: http://ftp.freshrpms.net/pub/freshrpms/redhat/9/yum/yum-2.0.4-1.rh.fr.i386.rpm Once installed you can just type yum update kernel as root at a bash shell. Or if you would like to update your whole system type: yum update This should fix that problem.. Scott - Original Message - From: Carlos Medina [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 10, 2004 7:11 PM Subject: [Asterisk-Users] Asterisk on a dual processor machine Hi there, i have a problem installing asterisk on a dual processor machine.I have Red Hat 9.0 with kernel-smp-2.4.20-6. I did the installation process with no problem, i used the asterisk stable version 1.0. The problem is that the machine has some troubles after Asterisk goes up, the CPU performance goes to 99%, and its all consumed by the asterisk process. I dont know if there is a special process to compile asterisk using dual-processor or maybe a special version or what steps do i have to follow to make asterisk works fine on that machine. The only message error that i have, is when i tried to load the card module...when i put modprobe wct4xxp it shows me the following message: NMI received. Dazed and confused, but trying to continue. You probably have a hardware problem with your RAM chips. After that i follow the rest of loading steps and asterisk goes up just fine. But how i mentioned above after a few minutes the CPU performance goes to 99% and the machine is impossible to handle. Thanks for your time. Carlos Andres Medina CVCOL S.A - Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] How do you close a VoicePulse Connect! account?
As best I can tell you remove your credit card info, cancel any phone numbers you have, and turn off the automatic billing stuff and when your account hits 0 your canceled. Scott - Original Message - From: Brian Cuthie [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 03, 2004 8:46 PM Subject: [Asterisk-Users] How do you close a VoicePulse Connect! account? Anybody figured out how to close a VoicePulse Connect! account? As bad as their web site is at most other things, the notion of actually closing an account doesn't appear to have even been contemplated. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] module help?
Simple solution on redhat machines In the zaptel source tree (At least the CVS one) there is a file called zaptel.init. This is a script that will allow you to install all needed modules. To use it do this: cd /usr/src/zaptel cp zaptel.init /etc/init.d/zaptel chkconfig --add zaptel chkconfig --level 2345 zaptel on Now every time you reboot all the zaptel modules will be install automatically. PS Why this is not done in the make install script is beyond me. Scott 700-297-0469 - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 02, 2004 6:59 PM Subject: Re: [Asterisk-Users] module help? I've installed the new TDM04B 4-port FXO card and its working. After a reboot, when I do lsmod I see the wcfxo module but not the wcfxs even though both are listed modules.conf. If I modprobe wcfxs, then lsmod has both modules showing. why you need wcfxs on a quad-fxo ? Because the support folks at digium said on Friday the supporting routines for the new fxo card are actually in wcfxs. The wcfxs module is the last one in the modules.conf. Is the order of entries sensitive in modules.conf? modules.conf != loaded modules. as the name suggest, it contains only configuration params for modules Do I need to be concerned with wcfxs not showing before starting asterisk? Any suggestions? sure. learn something more about kernel, modules and what is modules.conf bug us with asterisk related questions, not with what-are-kernel-modules? questions. Okay, then let me reword this just for you. Is there a problem with the asterisk make install process that might be considered the root-cause for wcfxs not showing up in lsmod? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] New TDM04B 4-port FXO card problems
Just installed the new 4-port FXO card and moved two pstn lines from existing x100p cards to ports on the FXO card. All zapata.conf entries that were functional on the x100p's were copied to the new FXO channels (including callprogress=no). Observations thus far: 1. asterisk will spontanously decide a pstn call has arrived, and ring the sip phone designated in the dailplan. Verified callprogress=no is in place, and monitoring the two pstn lines with an external analog phone (with line lamps) indicates no one was on the phone and no ringing actually occurred. I saw this a few times today on my X100P.. Problem in zaptel code perhaps.. 2. Incoming CallerID seems to be slightly less reliable on the FXO compared to the x100p. (Eg, there seems to be more cases of the callerid showing up as asterisk.) Yes I see errors more here too.. 3. The echo issues that have been so well documented with the x100p's seem to be identical on the new FXO card. 4. Incoming pstn calls that either go to IVR menues or VM do not properly sense disconnect supervision. Again, monitoring the pstn line via the LEDs on an analog phone does indicate approximately .5 second of no-battery (disconnect). After the disconnect, * does not release the pstn line which then causes dial tone from the Central Office to be recorded in the VM, etc. The CLI does not indicate a hangup until _after_ the sip phone hangs up. MAJOR Problem, and reported to support. I worked with Mark today and this problem is now fixed in the CVS. Additional testing is being conducted. Anyone seeing these same problems? Rich Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] New TDM04B 4-port FXO card problems
Here is the bug ID I posted: Add comments I guess http://bugs.digium.com/bug_view_page.php?bug_id=0001522 - Original Message - From: Roger Gulbranson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: Roger Gulbranson [EMAIL PROTECTED] Sent: Saturday, May 01, 2004 11:57 PM Subject: Re: [Asterisk-Users] New TDM04B 4-port FXO card problems On Sat, 2004-05-01 at 23:28, Scott Weis wrote: 4. Incoming pstn calls that either go to IVR menues or VM do not properly sense disconnect supervision. Again, monitoring the pstn line via the LEDs on an analog phone does indicate approximately .5 second of no-battery (disconnect). After the disconnect, * does not release the pstn line which then causes dial tone from the Central Office to be recorded in the VM, etc. The CLI does not indicate a hangup until _after_ the sip phone hangs up. MAJOR Problem, and reported to support. I worked with Mark today and this problem is now fixed in the CVS. I just downloaded the most recent CVS, compiled, removed modules, and reloaded. Problem still persists. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Using Exchange to send voicemail message
Simple answer, Without using sendmail to send the message to the exchange server the answer is no. Asterisk is a linux based PBX with no build in MTA. It needs an external program to send mail. So unless you have a linux program that can speak exchange you need sendmail (Or any program that compiles on linux with a sendmail like interface(Postfix, courier, etc)). Scott - Original Message - From: Paul Tyreman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, April 25, 2004 3:37 PM Subject: Re: [Asterisk-Users] Using Exchange to send voicemail message Re: [Asterisk-Users] Using Exchange to send voicemail messageAn of course, its SMTP that I need. I don't want sendmail to send mail to the exchange server, I want to use the exchange server to send the mail in the first place ! What I want to do is forget about sendmail, and make an account on the exchange server for asterisk to send mail from. Can that be done ? Thanks, Paul. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Hajime Lanning Posted At: 25 April 2004 19:40 Posted To: Asterisk-Users Conversation: [Asterisk-Users] Using Exchange to send voicemail message Subject: Re: [Asterisk-Users] Using Exchange to send voicemail message IMAP and POP3 are used for the MUA to get access to a mailbox. They are not used for delivering messages to a mailbox, but for reading message out of a mailbox. What you are looking for, is an SMTP gateway. Sendmail is an SMTP MTA that can be configured to send the email (via SMTP/ESMTP) to the Exchange server. All you really need to do is have the DNS MX records for foo.com pointing to your Exchange server. Then, in voicemail.conf you would have the email address set to [EMAIL PROTECTED] Of course, change foo.com to whatever your domain, for the Exchange server, is. And, make sure you have the SMTP connector configured for Exchange. quote who=Paul Tyreman Hi, I run a local exchange server and would like asterisk to send voicemail notification messages via exchange. I have had a look at the voicemail.conf file, but I can't see how I would go about configuring it to use an account set up on exchange ? The exchange account would have both POP3 and IMAP access, so how can I tell Asterisk to use the exchange account rather then sendmail ? Thanks, Paul. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Ok, Im confused
The simple answer probably is, If you have a NAT firewall (like a linksys,
netgear, dlink, etc) it will not work.
If your linux machine is directly connected follow the instructions on the
wiki and it will work no problems. I could not get FWD to work at all until
I made my linux box the outside edge of my network.
Scott
- Original Message -
From: James H. Thompson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, April 21, 2004 8:58 PM
Subject: Re: [Asterisk-Users] Ok, Im confused
You can post your .conf files.
But here is a guess at what you may need
replace FWD## with your freeworlddialup number and mypassword with
your freeworlddialup
password.
in sip.conf
context = from-fwd
register=FWD##:[EMAIL PROTECTED]/FWD##
[fwd]
type=friend
secret=mypassword
username=FWD##
host=fwd.pulver.com
in phone.conf
...
context=from-phone
...
in extensions.conf
[from-fwd]
exten = FWD##,1,Dial(Phone/phone0)
exten = i,2,Playback(invalid)
[from-phone]
exten = _.,1,SetCallerID(FWD##)
exten = _.,2,SetCIDName(FWD##)
exten = _.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _.,4,Playback(invalid)
exten = _.,5,Hangup
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: tmpm [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, April 21, 2004 1:24 PM
Subject: Re: [Asterisk-Users] Ok, Im confused
Thanks Jim,
But that page started my trip off to confusionbeen theretried it
10
different ways...still no joy.
I'll go through it once again, maybe Im missing something, I dont know.
Im
about ready to boot the penguin to the curb...
I know its in there...I think Ive got it all configured, and I dial the
outbound strings, and get a fast busy...I know one stinking letter off,
and
its whacked...
HOW for example do I specify my one and only extension is the Internet
phone jack? Phone0?
Somehow theres got to be a tie-in...I cant find it.
been thru extensions.conf, phones.conf, sip.conf..etc.
groan..
At 18:40 4/21/2004, you wrote:
Look here:
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: tmpm [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, April 21, 2004 11:50 AM
Subject: [Asterisk-Users] Ok, Im confused
Im totally a newbee at *
Im confused.
Ive got a FWD account, and it works on the winboxen. Ive got * up
and can
do the echotest etc, so its working.
I want to get FWD working, and all the pages ive seen on setup are
most
confusing.
Is FWD setup like IAXTEL? Do i plug in my FWD info in the same
places as
the IAXTEL stuff?
Ive been trying for a week now, and Im more lost than before.
Ive got a Internet phonejack card in the penguin, phone0, and all I
want to
do at this point is make and receive calls thru FWD using that
jackIll
plug the house in later...Ill learn the other stuff later. No
voicemail, no
BS, no dial thru least cost routing, or nightlines just make it
work as
a phone.
Im either more stupid than I think, or Im missing something major
here.
Ive got to the point the CLI shows me connected to FWD fine.(I
think)
Sip show users
Username Secret Authen Def. Context a/c
fwd.pulver.com secret md5,plaintext default no
Need some basic, stupidly simple scripts here...I dont need or want
to
dial
1-700 or *9 or any other crap, just make it work like the stupid
winbox
phone for now...Ill read the wiki for a couple years, and then maybe
I can
do voicemail or whatever...
frustrated...and I know its showing...sri
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Re: [Asterisk-Users] Basic Answering Machine Function?
Here is how I do the same thing: exten = 1234,1,Dial(Zap/2,30) exten = 1234,2,Answer exten = 1234,3,DigitTimeout,5 exten = 1234,4,ResponseTimeout,3 exten = 1234,5,SetMusicOnHold(random) exten = 1234,6,BackGround(2) exten = 1234,7,BackGround(vm-nobodyavail) exten = 1234,8,Voicemail(21) This of course does require a device to dial... Scott - Original Message - From: Jeff Rush [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 31, 2004 1:45 PM Subject: [Asterisk-Users] Basic Answering Machine Function? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAXtel Broken?
Here is what I get... -- Starting simple switch on 'Zap/2-1' -- Executing Dial(Zap/2-1, IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called username:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 69.73.19.178 (format GSM) -- Format for call is GSM -- IAX2[iaxtel]/3 stopped sounds Mine stops here too.. - Original Message - From: Hans-Henrik Andresen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, March 09, 2004 12:48 AM Subject: [Asterisk-Users] Re: IAXtel Broken? Greate, I was thought I had done something to my installation, I cant use iaxtel's 1-8XX numbers. So they might be down. -- Executing Dial(SIP/hha2-bf35, IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 69.73.19.178 (format GSM) -- Format for call is GSM -- IAX2[iaxtel]/5 stopped sounds And here it stops. /HHA Scott Weis [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I anyone able to get calls from IAXtel, I have been trying to call between to * systems all day with no luck. Worked fine Friday. Thanks, Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXtel Broken?
I anyone able to get calls from IAXtel, I have been trying to call between to * systems all day with no luck. Worked fine Friday. Thanks, Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Free World Dialup
Anyone know what this means? Mar 8 12:28:50 NOTICE[-112661]: chan_sip.c:3150 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again Mar 8 12:28:50 WARNING[-112661]: chan_sip.c:455 __sip_xmit: sip_xmit of 0x82ac7b4 (len 366) to 192.246.69.223 returned -1: Bad file descriptor Mar 8 12:28:51 WARNING[-112661]: chan_sip.c:455 __sip_xmit: sip_xmit of 0x82ac7b4 (len 366) to 192.246.69.223 returned -1: Bad file descriptor Mar 8 12:28:52 WARNING[-112661]: chan_sip.c:455 __sip_xmit: sip_xmit of 0x82ac7b4 (len 366) to 192.246.69.223 returned -1: Bad file descriptor Mar 8 12:28:53 WARNING[-112661]: chan_sip.c:455 __sip_xmit: sip_xmit of 0x82ac7b4 (len 366) to 192.246.69.223 returned -1: Bad file descriptor Mar 8 12:28:54 WARNING[-112661]: chan_sip.c:455 __sip_xmit: sip_xmit of 0x82ac7b4 (len 366) to 192.246.69.223 returned -1: Bad file descriptor Mar 8 12:28:55 WARNING[-112661]: chan_sip.c:455 __sip_xmit: sip_xmit of 0x82ac7b4 (len 366) to 192.246.69.223 returned -1: Bad file descriptor Mar 8 12:28:56 WARNING[-112661]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Request) ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Voicepulse error
Looking at the # dialed, it looks like you need to strip the 9 off of your
${EXTEN} like this ${EXTEN:1}
Scott
- Original Message -
From: oliver vermeulen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 12:53 PM
Subject: [Asterisk-Users] Voicepulse error
Hi Guys,
I anybody having problems with voicepulse out/in bound call ?
On the outbound calls im getting this error : (removed the username)
-- Executing Dial(SIP/103-296e,
IAX2/[EMAIL PROTECTED]/917707840009) in new stack
-- Called [EMAIL PROTECTED]/917707840009
Mar 4 12:51:31 WARNING[131081]: chan_iax2.c:4515 socket_read: Call
rejected
by 66.234.228.132: No such context/extension
-- Hungup 'IAX2[voicepulse-out]/3'
And on the inbound call im getting fast busy?
Thanks,
Oliver
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[Asterisk-Users] FXO Gateway of choice is?
I have a need to purchase a 2-4 port FXO gateway for use with *. I have no PCI slots left in my * machine so I can't use a X100P. So what is the best FXO gateway to get? Thanks, Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iaxtel to asterisk config help.
I am looking for a working set of config files for IAXTEL. When I dial a 1700 number I get a busy back and see no IAX debug messages. When I dial in to my 1700 number I get thses messages: Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00044ms SCall: 1 DCall: 00030 [69.73.19.178:4569] Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 1ms SCall: 00237 DCall: 0 [69.73.19.178:4569] VERSION : 2 CALLED NUMBER : s CALLING NUMBER : 403259 LANGUAGE: en USERNAME: iaxtel FORMAT : 2 CAPABILITY : 258 ADSICPE : 2 DATE TIME : 139833750 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 1ms SCall: 2 DCall: 00237 [69.73.19.178:4569] AUTHMETHODS : 4 CHALLENGE : 93159559 USERNAME: iaxtel Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00052ms SCall: 00237 DCall: 2 [69.73.19.178:4569] RSA RESULT : I38ugnoTcX4eynf3wVru7XzvsJJWbIsFarIlPKhqoYzQDD4epCrwki6zjVbn4HUpdl8V78pKbiXW Dmo -- Accepting AUTHENTICATED call from 69.73.19.178, requested format = 2, actual format = 2 Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00051ms SCall: 2 DCall: 00237 [69.73.19.178:4569] FORMAT : 2 -- Executing Wait([EMAIL PROTECTED]:4569]/2, 1) in new stack Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00051ms SCall: 00237 DCall: 2 [69.73.19.178:4569] -- Executing Answer([EMAIL PROTECTED]:4569]/2, ) in new stack Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: CONTROL Subclass: ANSWER Timestamp: 01051ms SCall: 2 DCall: 00237 [69.73.19.178:4569] -- Executing DigitTimeout([EMAIL PROTECTED]:4569]/2, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout([EMAIL PROTECTED]:4569]/2, 10) in new stack -- Set Response Timeout to 10 -- Executing BackGround([EMAIL PROTECTED]:4569]/2, demo-congrats) in new stack Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 002 Type: VOICE Subclass: 2 Timestamp: 01053ms SCall: 2 DCall: 00237 [69.73.19.178:4569] -- Playing 'demo-congrats' (language 'en') Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 01051ms SCall: 00237 DCall: 2 [69.73.19.178:4569] Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 01053ms SCall: 00237 DCall: 2 [69.73.19.178:4569] Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass: HANGUP Timestamp: 07107ms SCall: 00237 DCall: 2 [69.73.19.178:4569] Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 07107ms SCall: 2 DCall: 00237 [69.73.19.178:4569] == Spawn extension (default, s, 5) exited non-zero on '[EMAIL PROTECTED]:4569]/2' -- Hungup '[EMAIL PROTECTED]:4569]/2' Thanks for any help... Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
