Call being dropped when calling HP Support.

cmsinternet
Posts: 1
Member Since:
2008-02-29

I know this is a strange one and it will require some leg work for others but any help would be VERY appreciated.

When anyone on our phone system calls HP support at 800-474-6836 after you go thru the prompts you are put on hold and there is no hold music, after about 4 minutes you get disconnected. If you call the number just give thier automated system false answers to get put on hold. Here is the output from Trixbox when this happens.

== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/278-b6 224468' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/278-b6 224468'
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/278-b6224468", "hangupcall |") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/278-b6224468", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/278-b6224468", "") in new sta ck
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/278-b6224468", "1?skiprg") i n new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/278-b6224468", "1?skipblkvm" ) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/278-b6224468", "1?theend") i n new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/278-b6224468", "") in new s tack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/278-b6224 468' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/278-b6224 468'

I am told by another coworked it has happened to them too only with Microsoft and no hold music.

Does anyone have an idea why a call would be disconnected when there is no hold music?



timtmcc
Posts: 27
Member Since:
2008-09-26
Dropped calls automated system

I get the same issue when calling random vendors. We will be on hold and when the vendor's automated system transfers to a human the call drops. It's random, calling back a second time and it'll sometimes transfer correctly. I don't have any logs of those specific times as of yet.

I am running:

Trixbox v2.6.2.1
Aastra 57i with firmware version 2.5.1.2000
Sangoma A104d QUAD T1/E1 AFT card
PRI

Any input would be appreciated.



vccs
Posts: 19
Member Since:
2008-09-04
There is a default timeout

There is a default timeout with RTP. When there is no audio heard for a set amount of time asterisk drops the call and considers it dead. I forget exactly where the setting is but I would check into that, as well as canreinvite settings. Hope this will be a start point.

--

Matthew Martin
Chief Consultant
VCCS
http://www.vccs.ca
trixbox Hosting, Customization and Support|Business VoIP



jfinstrom
Posts: 2013
Member Since:
2007-03-07
@vccs

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