Shad Mortazavi
2005-Jul-12 03:48 UTC
[Asterisk-Users] Possible Agent Bug found while upgrading from Asterisk 0.7.1 to 1.0.9
Dear All, I have been running an Asterisk 0.7.1 (patched with various agent applications) server for almost 2 years. We have a data center in the USA and a call center in the UK. All calls are routed to a group of central call queues in the USA. Agents from the data center, call center and from remote locations (London, Scotland, LA, Florida, and Maine) can log in, join the call queue and pick up calls This function has worked well since implementing the system and works well using SNOM 200's (data center and call center) and SJ Phone Build 1.50.271d, Mar 11 2005. I have rebuilt an identical test environment in my test lab and I can run version 0.7.1 (patched). I log in as an agent using my softphone, make a call from a second phone, I get greeted, put in a queue, given my position, the call goes through to my soft phone, I accept the call, press # and I'm on the call. I run the upgrade to version 1.0.9 and run the same test; I get greeted, put in a queue, given my position, the call goes through to my soft phone, and I accept the call, press #... I then get a message telling me that the system saying transfer? I see nothing on the CLI except the usual waiting for '#' to acknowledge To discount the SJ Phone I installed the version of X-Ten light that some of our agents/staff use and I got the same result. I checked the DTMF setting in sio.conf and these appear correct. I downgrade to 0.7.1 and the function works on both SJ Phone and X-ten light. I have included the CLI captures below; Sip show agents; (Angela Holt) available at '0401@sip' (musiconhold is 'default')>From the CLI>-- outgoing agentcall, to agent '1031', on 'Local/0401@sip-bcc2,1' -- Called Agent/1031 -- Executing Dial("Local/0401@sip-bcc2,2", "SIP/phone6&SIP/0401|20|tr") in new stack -- Called phone6 -- Called 0401 -- Agent/1031 is ringing -- SIP/phone6-1d2b is ringing -- Agent/1031 is ringing -- SIP/phone6-1d2b answered Local/0401@sip-bcc2,2 -- Local/0401@sip-bcc2,1 answered, waiting for '#' to acknowledge I need to move to the latest version of Asterisk to enable me to measure the number of minutes a user has been held in a queue. This function was not available in version 0.7.1. I remember a similar problem with version 0.7.2. Anyone else run into the same issue? Is it a known issue/bug? What is the fix? Thanks and Regards Shad Mortazavi ------------------------------------------------------ Nexus Global Technical Manager n|m Nexus Management Inc
Adam Goryachev
2005-Jul-12 06:12 UTC
[Asterisk-Users] Possible Agent Bug found while upgrading from Asterisk 0.7.1 to 1.0.9
I get greeted, put in a queue, given my position, the call goes through> to my soft phone, and I accept the call, press #... I then get a message > telling me that the system saying transfer? I see nothing on the CLI > except the usual waiting for '#' to acknowledge >Send the complete extensions.conf for the incoming call portion, and the agentcallbacklogin section. Also send the complete CLI from the call arriving into the PABX through to the call being sent to the agent. I suspect somewhere you are including the t or T option to the queue or dial which allows # to transfer a call. Of course, perhaps someone should check this, as we can't transfer a call until after we accept it... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395 adam@websitemanagers.com.au Fax: +61 2 9345 4396 www.websitemanagers.com.au
Shad Mortazavi
2005-Jul-12 09:36 UTC
[Asterisk-Users] RE: Possible Agent Bug found while upgrading from Asterisk 0.7.1 to 1.0.9
Good day Adam, I have about 30 Queues configured so at the risk of boring everyone I have included one of the lines; exten => _812108,1,Playback(nexus/wel-helpdesk-interwise) exten => _812108,2,SetCIDName(Client1) exten => _812108,3,Queue(Client1|Tt|||) exten => _812108,4,Playback(nexus/im-sorry) exten => _812108,5,Voicemail(1500) The _812108 is the DNIS number on the T1. I did have Tt configured in the queue. I followed your suggestion and changed this to; exten => _812108,1,Playback(nexus/wel-helpdesk-interwise) exten => _812108,2,SetCIDName(Client1) exten => _812108,3,Queue(Client1||||) exten => _812108,4,Playback(nexus/im-sorry) exten => _812108,5,Voicemail(1500) Same issue. I looked at the Agent's extension. It was configured as; ; Angela Holt exten => 0420,1,Dial(SIP/phone21,20,tr) exten => 0420,2,VoiceMail,u1021 exten => 0420,3,MusicOnHold(default) I changed this to; ; Angela Holt exten => 0420,1,Dial(SIP/phone21,20) exten => 0420,2,VoiceMail,u1021 exten => 0420,3,MusicOnHold(default) Removing the tr has done the trick. And the problem is gone. The agent can still transfer the call. Thanks for the idea. Warm Regards Shad Mortazavi ------------------------------------------------------ Nexus Global Technical Manager n|m Nexus Management Inc I get greeted, put in a queue, given my position, the call goes through> to my soft phone, and I accept the call, press #... I then get amessage> telling me that the system saying transfer? I see nothing on the CLI > except the usual waiting for '#' to acknowledge >Send the complete extensions.conf for the incoming call portion, and the agentcallbacklogin section. Also send the complete CLI from the call arriving into the PABX through to the call being sent to the agent. I suspect somewhere you are including the t or T option to the queue or dial which allows # to transfer a call. Of course, perhaps someone should check this, as we can't transfer a call until after we accept it... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395 adam@websitemanagers.com.au Fax: +61 2 9345 4396 www.websitemanagers.com.au