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_voip版 - Asterisk on RT-N16 +Sipura1001 对方根本听不清
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话题: gtalk话题: sip话题: asterisk话题: executing
1 (共1页)
e**********i
发帖数: 108
1
按照fiu的dummy guide装的,打进打出都能响铃,我这边也很清楚,可是对方根本听不
见,声音微弱,还延迟很长,根本没法用,不知道是什么原因,请板上各位给诊断一下
。下面是ssh的信息:打入打出的都在
Tomato v1.28.7500 MIPSR2Toastman-RT K26 USB VPN
root@RT-N16USB:/tmp/home/root# asterisk -rvvv
Asterisk 1.8.17.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/opt/etc/asterisk/asterisk.conf': == Found
== Parsing '/opt/etc/asterisk/extconfig.conf': == Found
Connected to Asterisk 1.8.17.0 currently running on RT-N16USB (pid = 518)
Verbosity is at least 4
-- Remote UNIX connection
== Using SIP RTP CoS mark 5
-- Executing [10123456789@outbound:1] Dial("SIP/101-0000000a", "Gtalk/MY
_GV/1*********[email protected]") in new stack
-- Called Gtalk/MY_GV/1*********[email protected]
-- Gtalk/1*********[email protected] is ringing
-- Gtalk/1*********[email protected] answered SIP/101-0000000a
== Spawn extension (outbound, 10123456789, 1) exited non-zero on 'SIP/101-
0000000a'
-- Executing [M***[email protected]@google-in:1] GotoIf("Gtalk/+10123456789-
3903", "0?bridged") in new stack
-- Executing [M***[email protected]@google-in:2] NoOp("Gtalk/+10123456789-3903
", "Callerid +**********[email protected]/srvres-MTAuMTMuMTcxLjM6OTg4Ng=="
) in new stack
-- Executing [M***[email protected]@google-in:3] Set("Gtalk/+10123456789-3903"
, "CALLERID(num)=+10123456789") in new stack
-- Executing [M***[email protected]@google-in:4] Set("Gtalk/+10123456789-3903"
, "CALLERID(name)=") in new stack
-- Executing [M***[email protected]@google-in:5] Dial("Gtalk/+10123456789-3903
", "SIP/101, 180, D(:1)") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/101
-- SIP/101-0000000b is ringing
-- SIP/101-0000000b answered Gtalk/+10123456789-3903
-- Sending DTMF '1' to the calling party.
== Spawn extension (google-in, M***[email protected], 5) exited non-zero on '
Gtalk/+10123456789-3903'
RT-N16USB*CLI>
c**s
发帖数: 771
2
you have to find it by yourself, unfortunately.
If it is 对方根本听不见,声音微弱, you could change "gain" (don't remember
the exact word now) in your SPA1001.
Or it could be your internet connection speed is slow (what is your upload
speed?). If several people/devices are sharing the same pipeline, you may
need to set up a good QoS rule in your router.
It is natural to have a little delay/hiccup 延迟 in the very beginning of
every call.
e**********i
发帖数: 108
3
按照fiu的dummy guide装的,打进打出都能响铃,我这边也很清楚,可是对方根本听不
见,声音微弱,还延迟很长,根本没法用,不知道是什么原因,请板上各位给诊断一下
。下面是ssh的信息:打入打出的都在
Tomato v1.28.7500 MIPSR2Toastman-RT K26 USB VPN
root@RT-N16USB:/tmp/home/root# asterisk -rvvv
Asterisk 1.8.17.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/opt/etc/asterisk/asterisk.conf': == Found
== Parsing '/opt/etc/asterisk/extconfig.conf': == Found
Connected to Asterisk 1.8.17.0 currently running on RT-N16USB (pid = 518)
Verbosity is at least 4
-- Remote UNIX connection
== Using SIP RTP CoS mark 5
-- Executing [10123456789@outbound:1] Dial("SIP/101-0000000a", "Gtalk/MY
_GV/1*********[email protected]") in new stack
-- Called Gtalk/MY_GV/1*********[email protected]
-- Gtalk/1*********[email protected] is ringing
-- Gtalk/1*********[email protected] answered SIP/101-0000000a
== Spawn extension (outbound, 10123456789, 1) exited non-zero on 'SIP/101-
0000000a'
-- Executing [M***[email protected]@google-in:1] GotoIf("Gtalk/+10123456789-
3903", "0?bridged") in new stack
-- Executing [M***[email protected]@google-in:2] NoOp("Gtalk/+10123456789-3903
", "Callerid +**********[email protected]/srvres-MTAuMTMuMTcxLjM6OTg4Ng=="
) in new stack
-- Executing [M***[email protected]@google-in:3] Set("Gtalk/+10123456789-3903"
, "CALLERID(num)=+10123456789") in new stack
-- Executing [M***[email protected]@google-in:4] Set("Gtalk/+10123456789-3903"
, "CALLERID(name)=") in new stack
-- Executing [M***[email protected]@google-in:5] Dial("Gtalk/+10123456789-3903
", "SIP/101, 180, D(:1)") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/101
-- SIP/101-0000000b is ringing
-- SIP/101-0000000b answered Gtalk/+10123456789-3903
-- Sending DTMF '1' to the calling party.
== Spawn extension (google-in, M***[email protected], 5) exited non-zero on '
Gtalk/+10123456789-3903'
RT-N16USB*CLI>
c**s
发帖数: 771
4
you have to find it by yourself, unfortunately.
If it is 对方根本听不见,声音微弱, you could change "gain" (don't remember
the exact word now) in your SPA1001.
Or it could be your internet connection speed is slow (what is your upload
speed?). If several people/devices are sharing the same pipeline, you may
need to set up a good QoS rule in your router.
It is natural to have a little delay/hiccup 延迟 in the very beginning of
every call.
e**********i
发帖数: 108
5
谢谢回复,折腾了一下,搞不定,买了个obitalk了事。通话质量好太多了
t*******f
发帖数: 2634
6
试试自己两个帐号对打。
我不是很喜欢Asterisk,因为功能太老旧。好像好多功能可以
自己想法设置,但是我已经没兴趣鼓捣乐。以前有个milkfish,
不知道现在怎么样。虽然简单,但是非常好使。

Public

【在 e**********i 的大作中提到】
: 按照fiu的dummy guide装的,打进打出都能响铃,我这边也很清楚,可是对方根本听不
: 见,声音微弱,还延迟很长,根本没法用,不知道是什么原因,请板上各位给诊断一下
: 。下面是ssh的信息:打入打出的都在
: Tomato v1.28.7500 MIPSR2Toastman-RT K26 USB VPN
: root@RT-N16USB:/tmp/home/root# asterisk -rvvv
: Asterisk 1.8.17.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
: Created by Mark Spencer
: Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
: details.
: This is free software, with components licensed under the GNU General Public

1 (共1页)
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谁能共享一个asterisk dialplan?拨的是0118613312345678,asterisk给我拨少了一个号。
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问一下,版上有没大牛分享过asterisk的配置文件?obi100 deal
asterisk防火墙的设置最进用obi + GV 好像有问题
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完美Asterisk+GV,折腾ipkall/sipgate的可以休矣asterisk 1.8+gtalk总结
asterisk gtalk接不起来的,你们有没有试过Asterisk 1.8 多google voice用户一点心得
相关话题的讨论汇总
话题: gtalk话题: sip话题: asterisk话题: executing