asterisk-user
2006-Sep-28 06:36 UTC
[asterisk-users] unable to call AT&T audio conference bridge
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@home <mailto:asterisk@home> did not have this issue and I even switched back to asterisk@home <mailto:asterisk@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.
BJ Weschke
2006-Sep-28 07:30 UTC
[asterisk-users] unable to call AT&T audio conference bridge
On 9/28/06, asterisk-user <myacc@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@home <mailto:asterisk@home> did not have > this issue and I even switched back to asterisk@home > <mailto:asterisk@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/
asterisk-user
2006-Oct-04 13:00 UTC
[asterisk-users] unable to call AT&T audio conference bridge
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@roundbox.com> To: asterisk-users@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@lists.digium.com Reply-To: asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Message: 19 Date: Thu, 28 Sep 2006 10:30:25 -0400 From: "BJ Weschke" <bweschke@gmail.com> Subject: Re: [asterisk-users] unable to call AT&T audio conference bridge To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <79cf6330609280730y619006a6mab5194b394dd040b@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 9/28/06, asterisk-user <myacc@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@home <mailto:asterisk@home> did not have > this issue and I even switched back to asterisk@home > <mailto:asterisk@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@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@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@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@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.