# Transfering call to ConfBridge drops customer call **URL:** \ **Category:** Asterisk Support\ **Created:** [May 6, 2011, 3:41pm UTC](https://community.asterisk.org/t/transfering-call-to-confbridge-drops-customer-call/35150 "2011-05-06T15:41:12Z")\ **Posts on this page:** 2\ **Page:** 1
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [May 6, 2011, 3:41pm UTC](https://community.asterisk.org/t/transfering-call-to-confbridge-drops-customer-call/35150/1 "2011-05-06T15:41:12Z")
Hi, I have this situation : 1. Customer A from the outside calls asterisk, goes trough a queue and then gets his call answered by an operator B 2. When B wants to do a conference with another operator C and the customer, he clicks a button in our interace 3. A and B get redirected over a ConfBridge room, C gets originated to the same ConfBridge room This is my extensions.conf : [queue\_base] exten =\> 78,1,Answer exten =\> 78,2,MixMonitor(${CALLERID(num)}-${UNIQUEID}-${EPOCH}-${STRFTIME(${EPOCH},%Y%m%d-%H%M%S)}.wav,b) exten =\> 78,3,Set(incoming=1) exten =\> 78,n,Queue(centralino-low) exten =\> 78,n,Hangup exten =\> 78,n,StopMixMonitor() [to\_conference] exten =\> \_.X,1,Answer exten =\> \_.X,n,ConfBridge(${EXTEN},1q) exten =\> h,n,Hangup [add\_to\_conference] exten =\> \_.X,1,Answer exten =\> \_.X,n,ConfBridge(${EXTEN},1q) exten =\> h,n,Hangup And what I do over AMI command is this : 1. First send an originate to have C called and put into add\_to\_conference context, with these parameters "Originate", “Channel” =\> SIP/2001, “Context” =\> “add\_to\_conference”, “Exten” =\> “468”, “Priority” =\> “1” 2. Then send a redirect to have A and B redirected to the to\_conference context, with these parameters “Redirect”, “Channel” =\> “SIP/2000-00000002”, “ExtraChannel” =\> “SIP/noverca-out-00000000”, “Context” =\> “to\_conference”, “ExtraContext” =\> “to\_conference”, “Exten” =\> “468”, “ExtraExten” =\> “468”, “Priority” =\> “1”, “ExtraPriority” =\> “1” It all goes well for B and C, the operators, but A, the customer gets his phone hangup. This is what I get in asterisk console : – Executing [78@queue\_base:1] Answer(“SIP/noverca-out-00000000”, “”) in new stack – Executing [78@queue\_base:2] MixMonitor(“SIP/noverca-out-00000000”, “00393355286159-1304695086.0-1304695087-20110506-171807.wav,b”) in new stack – Executing [78@queue\_base:3] Set(“SIP/noverca-out-00000000”, “incoming=1”) in new stack – Executing [78@queue\_base:4] Queue(“SIP/noverca-out-00000000”, “centralino-low”) in new stack – Music class default requested but no musiconhold loaded. == Using SIP RTP CoS mark 5 == Begin MixMonitor Recording SIP/noverca-out-00000000 – SIP/2001-00000001 is ringing \> doing dnsmgr\_lookup for ‘[noverca.com](http://noverca.com)’ \> ast\_get\_srv: SRV lookup for ‘\_sip.\_udp.noverca.com’ mapped to host [voip5.noverca.com](http://voip5.noverca.com), port 443 \> doing dnsmgr\_lookup for ‘[noverca.com](http://noverca.com)’ \> ast\_get\_srv: SRV lookup for ‘\_sip.\_udp.noverca.com’ mapped to host [voip1.noverca.com](http://voip1.noverca.com), port 5060 – SIP/2001-00000001 answered SIP/noverca-out-00000000 == Using SIP RTP CoS mark 5 \> Channel SIP/2000-00000002 was answered. – Executing [468@add\_to\_conference:1] Answer(“SIP/2000-00000002”, “”) in new stack – Executing [468@add\_to\_conference:2] ConfBridge(“SIP/2000-00000002”, “468,1q”) in new stack – Executing [468@to\_conference:1] Answer(“SIP/2001-00000001”, “”) in new stack – Executing [468@to\_conference:2] ConfBridge(“SIP/2001-00000001”, “468,1q”) in new stack == Spawn extension (to\_conference, 468, 1) exited non-zero on ‘SIP/noverca-out-00000000’ – Executing [468@to\_conference:1] Answer(“SIP/noverca-out-00000000”, “”) in new stack – Executing [468@to\_conference:2] ConfBridge(“SIP/noverca-out-00000000”, “468,1q”) in new stack – Executing [h@to\_conference:1] Answer(“SIP/noverca-out-00000000”, “”) in new stack == Spawn extension (to\_conference, h, 1) exited non-zero on ‘SIP/noverca-out-00000000’ == MixMonitor close filestream == End MixMonitor Recording SIP/noverca-out-00000000 I can’t understand what I’m doing wrong. I tried swapping redirect with originate, swapping redirect channels… it just drops the call. It just works randomly sometimes, but like 1 in a 100 times… Can someone help me pls? Thanks. ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [May 6, 2011, 3:51pm UTC](https://community.asterisk.org/t/transfering-call-to-confbridge-drops-customer-call/35150/2 "2011-05-06T15:51:42Z")
Just now I tried another scenario, Operator B calls Customer A and then starts the conference with Operator C. This seems to work, until I voluntarily hangup : ``` > Channel SIP/2000-00000005 was answered. -- Executing [470@add_to_conference:1] Answer("SIP/2000-00000005", "") in new stack -- Executing [470@add_to_conference:2] ConfBridge("SIP/2000-00000005", "470,1q") in new stack -- Executing [470@to_conference:1] Answer("SIP/noverca-out-00000004", "") in new stack -- Executing [470@to_conference:2] ConfBridge("SIP/noverca-out-00000004", "470,1q") in new stack -- AGI Script [agi://127.0.0.1](agi://127.0.0.1) completed, returning -1 -- Executing [470@to_conference:1] Answer("SIP/2001-00000003", "") in new stack -- Executing [470@to_conference:2] ConfBridge("SIP/2001-00000003", "470,1q") in new stack -- Executing [h@to_conference:1] Answer("SIP/noverca-out-00000004", "") in new stack -- Executing [h@to_conference:2] ConfBridge("SIP/noverca-out-00000004", "h,1q") in new stack -- Executing [h@to_conference:1] Answer("SIP/2001-00000003", "") in new stack -- Executing [h@to_conference:2] ConfBridge("SIP/2001-00000003", "h,1q") in new stack ``` == MixMonitor close filestream == End MixMonitor Recording SIP/2001-00000003 [May 6 17:35:18] WARNING[7095]: translate.c:162 framein: no samples for gsmtolin – Executing [h@add\_to\_conference:1] Answer(“SIP/2000-00000005”, “”) in new stack – Executing [h@add\_to\_conference:2] ConfBridge(“SIP/2000-00000005”, “h,1q”) in new stack I also tried once again the scenario in the opening post and noted that customer call gets hangup just as operator C answers his phone…