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一位用户最近通知我,每当他们尝试拨入另一家公司的电话会议时,电话会在 5 秒左右后挂断。他们还表示,当使用手机拨打同一个号码时,没有问题。我在日志文件中找到了以下条目。

[May 4 11:58:20] VERBOSE[24063] app_dial.c: -- DAHDI/1-1 is ringing
[May 4 11:58:20] VERBOSE[24063] app_dial.c: -- DAHDI/1-1 answered SIP/145-00000005
[May 4 11:58:24] WARNING[24063] rtp.c: Don't know how to represent 'f'
[May 4 11:58:24] VERBOSE[24063] chan_dahdi.c: -- Redirecting DAHDI/1-1 to fax extension
[May 4 11:58:24] VERBOSE[24063] pbx.c: -- Executing [h@macro-dialout-trunk:1] Macro("SIP/145-00000005", "hangupcall,") in new stack
[May 4 11:58:24] VERBOSE[24063] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/145-00000005", "1?theend") in new stack

我无法确定解决方案。任何有关解决此问题的见解或建议表示赞赏。(使用 FreePBX v2.9;Asterisk v1.6.2.15.1;CentOS 5.5(最终版);Sangoma A102)

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2 回答 2

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尝试添加到文件中

/etc/asterisk/sip_general_custom.conf

faxdetect=no
于 2012-05-04T18:52:34.797 回答
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还尝试修改 chan_dahdi.conf,但没有奏效。最终解决方案是在 /etc/wanrouter/wanpipe1.conf 中修改这些设置(从 YES 更改为 NO)

TDMV_HW_DTMF         = NO            # YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT   = NO            # YES: receive fax 1100hz events from hardware
于 2012-05-07T12:59:22.683 回答