Summary:	ASTERISK-17526: No Audio When Using Pickup() Application
Reporter:	Pawel Kucmus (mcabra)	Labels:
Date Opened:	2011-03-09 01:24:59.000-0600	Date Closed:	2011-07-26 14:19:54
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Features
Versions:	1.6.2.16	Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	I've tried to set up a remote inbound call pickup, so that one sales phone can answer another that is ringing. I'm using SPA922 phones. In my extentions.conf I have an: exten => druk95,1,Set(_PICKUPMARK=50) exten => druk95,n,Dial(SIP/50,20,rwt) exten => _850,1,Pickup(50@PICKUPMARK) when druk95 is called from ex. cell phone my SPA922 ext.50 is ringing, when I dial 850 number form an other SPA922 ex.52 I can pickup that call but I dont hear anthing but both cell and SPA are connected. When I hold the cell call and dial another extention from 52, talk to that extention, hang up and unhold the pickedup call from my cell, then i have the connection and i can talk. ****** ADDITIONAL INFORMATION ****** When I pickup no voice:  == Using SIP RTP CoS mark 5    -- Executing [druk95@default:1] Set("SIP/trunk_80-00001236", "_PICKUPMARK=50") in new stack    -- Executing [druk95@default:2] Dial("SIP/trunk_80-00001236", "SIP/50,20,rwt") in new stack  == Using SIP RTP CoS mark 5    -- Called 50  == Extension Changed 50[default] new state Ringing for Notify User 29    -- SIP/50-00001237 is ringing  == Using SIP RTP CoS mark 5    -- Executing [850@default:1] Pickup("SIP/52-00001238", "50@PICKUPMARK") in new stack    -- Auto fallthrough, channel 'SIP/52-00001238' status is 'UNKNOWN'    -- SIP/52-00001238 answered SIP/trunk_80-00001236  == Extension Changed 50[default] new state Idle for Notify User 29  == Spawn extension (default, druk95, 2) exited non-zero on 'SIP/trunk_80-00001236' When I pickup, conference then return to pickedup call:  == Using SIP RTP CoS mark 5    -- Executing [druk95@default:1] Set("SIP/trunk_80-0000123b", "_PICKUPMARK=50") in new stack    -- Executing [druk95@default:2] Dial("SIP/trunk_80-0000123b", "SIP/50,20,rwt") in new stack  == Using SIP RTP CoS mark 5    -- Called 50  == Extension Changed 50[default] new state Ringing for Notify User 29    -- SIP/50-0000123c is ringing  == Using SIP RTP CoS mark 5    -- Executing [850@default:1] Pickup("SIP/52-0000123d", "50@PICKUPMARK") in new stack    -- Auto fallthrough, channel 'SIP/52-0000123d' status is 'UNKNOWN'    -- SIP/52-0000123d answered SIP/trunk_80-0000123b  == Extension Changed 50[default] new state Idle for Notify User 29    -- Started music on hold, class 'default', on SIP/trunk_80-0000123b  == Using SIP RTP CoS mark 5    -- Executing [50@default:1] Dial("SIP/52-0000123e", "SIP/50") in new stack  == Using SIP RTP CoS mark 5    -- Called 50  == Extension Changed 50[default] new state Ringing for Notify User 29    -- SIP/50-0000123f is ringing    -- Registered SIP '50' at 62.148.79.98 port 25139    -- SIP/50-0000123f answered SIP/52-0000123e    -- Packet2Packet bridging SIP/52-0000123e and SIP/50-0000123f  == Extension Changed 50[default] new state InUse for Notify User 29  == Spawn extension (default, 50, 1) exited non-zero on 'SIP/52-0000123e'  == Extension Changed 50[default] new state Idle for Notify User 29    -- Stopped music on hold on SIP/trunk_80-0000123b  == Spawn extension (default, druk95, 2) exited non-zero on 'SIP/trunk_80-0000123b'
Comments:	By: Leif Madsen (lmadsen) 2011-03-31 12:14:36 You'll need to supply a SIP trace, SIP configuration for the devices (along with the [general] section) and a PCAP trace which contains the RTP data. From there we can see why the audio is not getting redirected appropriately. By: Pawel Kucmus (mcabra) 2011-04-08 04:18:48 How can I provide all that? By: Leif Madsen (lmadsen) 2011-05-10 14:13:05 You provide it with the appropriate tools, such as tshark/wireshark, and following instructions in the Debugging section of the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Debugging By: Russell Bryant (russell) 2011-07-26 14:19:44.160-0500 Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4 and 1.6.x branches has ended. For continued maintenance support please move to the 1.8 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions If this is still an issue, please open a new issue so it can be re-triaged appropriately. Thanks!