[asterisk-users] unable to call AT&T audio conference bridge

Wed Oct 4 13:00:56 MST 2006

Hello,
Can someone help me with this please?
Attached is the log file.

thank you

-------- Original Message --------
Subject: 	[Fwd: asterisk-users Digest, Vol 26, Issue 166]
Date: 	Fri, 29 Sep 2006 10:31:21 -0400
From: 	asterisk-user <myacc at roundbox.com>
To: 	asterisk-users at lists.digium.com

I tried by adding "answer()" to the dial plan but the problem still exists.
I am not sure if I am doing this right.
Attached is the log file from asterisk while making the call to the conf 
bridge after adding "answer()"
Could you please let me know if you find anything out of this log file?

thanks for your help.

-------- Original Message --------
Subject: 	asterisk-users Digest, Vol 26, Issue 166
Date: 	Thu, 28 Sep 2006 07:42:43 -0700 (MST)
From: 	asterisk-users-request at lists.digium.com
Reply-To: 	asterisk-users at lists.digium.com
To: 	asterisk-users at lists.digium.com

Message: 19
Date: Thu, 28 Sep 2006 10:30:25 -0400
From: "BJ Weschke" <bweschke at gmail.com>
Subject: Re: [asterisk-users] unable to call AT&T audio conference
	bridge
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID:
	<79cf6330609280730y619006a6mab5194b394dd040b at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

On 9/28/06, asterisk-user <myacc at roundbox.com> wrote:
> Hello,
> I have a problem with asterisk and trying to see if someone can help me
> fix the issue...
>
> Problem:
> I couldn't join AT&T's Tele Conference bridge directly without their
> customer service interaction.
> Instead of getting the automated prompts to join the conference, it
> takes me to the customer support and then I got to give them the bridge
> number and pincode to add me into the conference call.
>
> The reason given by AT&T was that their conference system is unable to
> identify our tone.
> This happens only with AT&T conference bridges... not sure what the
> problem is.
>
> This problem started after I installed trixbox on a new hardware.
> Previous setup with asterisk at home <mailto:asterisk at home> did not have
> this issue and I even switched back to asterisk at home
> <mailto:asterisk at home> (a different box) and called the same conf
> bridge... that worked fine.
>
> I am running trixbox with the following versions:
> asterisk - 1.2.9.1
> zaptel - 1.2.8
> libpri - 1.2.3-1.349
> using zap over a 8 channel pri
>
> Thanks in advance.
>

 AT&T's IVR to collect the passcode is coming through as "early media"
and since you haven't signaled to the phones that the phone is
"answered" they're probably not letting you send DTMF through the
bridge that isn't technically supposed to be there yet.

 Put an Answer() in your dial plan prior to sending the call out to
the Dial() application to reach the bridge for these types of calls
and this generally fixes your problems caused by someone else not
signaling correctly.

-- 
Bird's The Word Technologies, Inc.
http://www.btwtech.com/

------------------------------

-------------- next part --------------
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '1'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '208'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '208'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is ''
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '"" <208>'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '208'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '0'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'AGI'
Sep 28 19:30:04 DEBUG[32330] app_queue.c: Device 'SIP/208' changed to state '2' (In use) but we don't care because they're not a member of any queue.
Sep 28 19:30:04 VERBOSE[32329] logger.c:   recordingcheck|20060928-193004|1159486204.0: Outbound recording not enabled
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Macro'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '1'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is ''
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] db.c: Unable to find key '208/emergency_cid' in family 'DEVICE'
Sep 28 19:30:04 DEBUG[32329] func_db.c: DB: DEVICE/208/emergency_cid not found in database.
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is ''
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '1'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '1'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '"" '
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'NoOp'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is '1'
Sep 28 19:30:04 WARNING[32329] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
 1 >  
      ^
Sep 28 19:30:04 WARNING[32329] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source.
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'AGI'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Function result is 'ZAP/17'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Set'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Expression result is '0'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'GotoIf'
Sep 28 19:30:04 DEBUG[32329] pbx.c: Not taking any branch
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Answer'
Sep 28 19:30:04 DEBUG[32329] chan_sip.c: sip_answer(SIP/208-b621)
Sep 28 19:30:04 DEBUG[32329] pbx.c: Launching 'Dial'
Sep 28 19:30:04 DEBUG[32329] chan_zap.c: Using channel 17
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-dialout-trunk-s-15.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable MACRO_DEPTH.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-dialout-trunk-s-14.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-dialout-trunk-s-13.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable custom.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-dialout-trunk-s-12.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable OUTNUM.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-dialout-trunk-s-11.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-dialout-trunk-s-10.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable DIAL_TRUNK.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-dialout-trunk-s-9.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable DIAL_NUMBER.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-dialout-trunk-s-8.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-dialout-trunk-s-7.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable GROUP.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-dialout-trunk-s-6.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable MACRO_PRIORITY.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable MACRO_CONTEXT.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable MACRO_EXTEN.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable ARG1.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-outbound-callerid-s-15.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-outbound-callerid-s-13.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-outbound-callerid-s-12.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-outbound-callerid-s-11.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-outbound-callerid-s-7.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable TRUNKOUTCID.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-outbound-callerid-s-6.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable EMERGENCYCID.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-outbound-callerid-s-5.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable USEROUTCID.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-outbound-callerid-s-4.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable DB_RESULT.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-outbound-callerid-s-3.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-outbound-callerid-s-1.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-dialout-trunk-s-5.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable ARG2.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-record-enable-s-5.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-record-enable-s-4.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-record-enable-s-1.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-dialout-trunk-s-4.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-user-callerid-s-9.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-user-callerid-s-8.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-user-callerid-s-7.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable AMPUSERCIDNAME.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-user-callerid-s-6.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable AMPUSER.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-user-callerid-s-5.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-user-callerid-s-4.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable REALCALLERIDNUM.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-user-callerid-s-3.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-user-callerid-s-2.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-user-callerid-s-1.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-dialout-trunk-s-3.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-macro-dialout-trunk-s-1.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable ARG4.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable ARG3.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable STACK-from-internal-918666032932-1.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable SIPCALLID.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable SIPUSERAGENT.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable SIPDOMAIN.
Sep 28 19:30:04 DEBUG[32329] channel.c: Not copying variable SIPURI.
Sep 28 19:30:04 DEBUG[32329] channel.c: Driver for channel 'SIP/208-b621' does not support indication 3, emulating it
Sep 28 19:30:04 DEBUG[32329] channel.c: Set channel SIP/208-b621 to write format slin
Sep 28 19:30:04 DEBUG[32329] channel.c: Scheduling timer at 160 sample intervals
Sep 28 19:30:04 DEBUG[32335] app_queue.c: Device 'SIP/208' changed to state '2' (In use) but we don't care because they're not a member of any queue.
Sep 28 19:30:04 DEBUG[32336] app_queue.c: Device 'Zap/17' changed to state '2' (In use) but we don't care because they're not a member of any queue.
Sep 28 19:30:04 DEBUG[32337] app_queue.c: Device 'Zap/17' changed to state '2' (In use) but we don't care because they're not a member of any queue.
Sep 28 19:30:04 DEBUG[32329] rtp.c: Ooh, format changed from unknown to ulaw
Sep 28 19:30:05 DEBUG[32054] chan_sip.c: Stopping retransmission on 'ba997448b97b99243b18509fe040e641 at 192.168.4.199' of Response 1144871604: Match Found
Sep 28 19:30:05 DEBUG[32056] chan_zap.c: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/17 span 1
Sep 28 19:30:06 DEBUG[32054] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP)
Sep 28 19:30:06 DEBUG[32054] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP)
Sep 28 19:30:06 DEBUG[32054] chan_sip.c: Stopping retransmission on '1a924a323e2a22e80ac8e15c26eec112 at 192.168.4.201' of Request 102: Match Found
Sep 28 19:30:06 DEBUG[32056] chan_zap.c: Queuing frame from PRI_EVENT_PROGRESS on channel 0/17 span 1
Sep 28 19:30:06 DEBUG[32054] chan_sip.c: Stopping retransmission on '72e9d2601cf98cd4489be1af63885663 at 192.168.4.201' of Request 102: Match Found
Sep 28 19:30:06 DEBUG[32054] chan_sip.c: Allocating new SIP dialog for f23b0ac4-ad649df2-4c7b8863 at 192.168.4.179 - REGISTER (No RTP)
Sep 28 19:30:06 DEBUG[32349] app_queue.c: Device 'SIP/209' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.