asterisk不能外呼的问题

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weichenqi

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Dec 11, 2011, 2:48:24 AM12/11/11
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我已经按文档配置好sip trunk和外呼路由。
outgoing settings如下
host=xx.xx.xx.xx
username=XXX22
secret=XXX
type=peer
context=from-trunk
dtmfmode=rfc2833
allow=g729&alaw&ulaw
fromuser=XXX22

现在拨号日志如下
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [013385214578@from-internal:1]
Macro("SIP/2911-0000000a", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/2911-0000000a",
"AMPUSER=2911") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/2911-0000000a",
"0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/2911-0000000a",
"1?Set(REALCALLERIDNUM=2911)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/2911-0000000a",
"AMPUSER=2911") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/2911-0000000a",
"AMPUSERCIDNAME=2911") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/2911-0000000a",
"0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/2911-0000000a",
"AMPUSERCID=2911") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/2911-0000000a",
"CALLERID(all)="2911" <2911>") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/2911-0000000a",
"1?continue") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [s@macro-user-callerid:18] NoOp("SIP/2911-0000000a",
"Using CallerID "2911" <2911>") in new stack
-- Executing [013385214578@from-internal:2]
Set("SIP/2911-0000000a", "_NODEST=") in new stack
-- Executing [013385214578@from-internal:3]
Macro("SIP/2911-0000000a", "record-enable,2911,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/2911-0000000a",
"1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/2911-0000000a",
"0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/2911-0000000a",
"0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,16)
-- Executing [s@macro-record-enable:16] GotoIf("SIP/2911-0000000a",
"0?IN") in new stack
-- Executing [s@macro-record-enable:17] ExecIf("SIP/2911-0000000a",
"1?MacroExit()") in new stack
-- Executing [013385214578@from-internal:4]
Macro("SIP/2911-0000000a", "dialout-trunk,2,13385214578,,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/2911-0000000a",
"DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/2911-0000000a",
"0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/2911-0000000a",
"0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/2911-0000000a",
"DIAL_NUMBER=13385214578") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/2911-0000000a",
"DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/2911-0000000a",
"OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/2911-0000000a",
"1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/2911-0000000a",
"0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/2911-0000000a",
"DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/2911-0000000a",
"outbound-callerid,2") in new stack
-- Executing [s@macro-outbound-callerid:1]
ExecIf("SIP/2911-0000000a", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2]
ExecIf("SIP/2911-0000000a", "0?Set(REALCALLERIDNUM=2911)") in new stack
-- Executing [s@macro-outbound-callerid:3]
GotoIf("SIP/2911-0000000a", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/2911-0000000a",
"USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/2911-0000000a",
"EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/2911-0000000a",
"TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9]
GotoIf("SIP/2911-0000000a", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12]
ExecIf("SIP/2911-0000000a", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:13]
ExecIf("SIP/2911-0000000a", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14]
ExecIf("SIP/2911-0000000a", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15]
ExecIf("SIP/2911-0000000a", "0?Set(CALLERPRES()=prohib_passed_screen)")
in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/2911-0000000a",
"0?AGI(fixlocalprefix)") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/2911-0000000a",
"OUTNUM=13385214578") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/2911-0000000a",
"custom=SIP/122sip") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/2911-0000000a",
"0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/2911-0000000a",
"dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1]
MacroExit("SIP/2911-0000000a", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/2911-0000000a",
"0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/2911-0000000a",
"0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/2911-0000000a",
"SIP/122sip/13385214578,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 122sip/13385214578
-- SIP/122sip-0000000b is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] NoOp("SIP/2911-0000000a",
"Dial failed for some reason with DIALSTATUS = CONGESTION and
HANGUPCAUSE = 21") in new stack
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/2911-0000000a",
"s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1]
Set("SIP/2911-0000000a", "RC=21") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2]
Goto("SIP/2911-0000000a", "21,1") in new stack
-- Goto (macro-dialout-trunk,21,1)
-- Executing [21@macro-dialout-trunk:1] Goto("SIP/2911-0000000a",
"continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1]
GotoIf("SIP/2911-0000000a", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3]
NoOp("SIP/2911-0000000a", "TRUNK Dial failed due to CONGESTION
HANGUPCAUSE: 21 - failing through to other trunks") in new stack
-- Executing [013385214578@from-internal:5]
Macro("SIP/2911-0000000a", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/2911-0000000a",
"") in new stack
-- Executing [s@macro-outisbusy:2] Playback("SIP/2911-0000000a",
"all-circuits-busy-now,noanswer") in new stack
-- <SIP/2911-0000000a> Playing 'all-circuits-busy-now.gsm'
(language 'en')
-- Executing [s@macro-outisbusy:3] Playback("SIP/2911-0000000a",
"pls-try-call-later,noanswer") in new stack
-- <SIP/2911-0000000a> Playing 'pls-try-call-later.gsm' (language 'en')
-- Executing [s@macro-outisbusy:4] Macro("SIP/2911-0000000a",
"hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/2911-0000000a",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/2911-0000000a",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/2911-0000000a",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/2911-0000000a", "")
in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/2911-0000000a' in macro 'hangupcall'
== Spawn extension (macro-outisbusy, s, 4) exited non-zero on
'SIP/2911-0000000a' in macro 'outisbusy'
== Spawn extension (from-internal, 013385214578, 5) exited non-zero
on 'SIP/2911-0000000a'
-- Executing [h@from-internal:1] Macro("SIP/2911-0000000a",
"hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/2911-0000000a",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/2911-0000000a",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/2911-0000000a",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/2911-0000000a", "")
in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/2911-0000000a' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/2911-0000000a'

我感觉是sip trunk的参数没有配置好，求大大帮忙。万分感谢。

A-Lang, Hsu - Asterisk/Linux/IT Consultant

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Dec 11, 2011, 10:05:01 PM12/11/11
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執行 sip show peer 122sip，檢查 codec 是否與設定的相同。


2011/12/11 weichenqi <weic...@gmail.com>

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weichenqi

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Dec 12, 2011, 4:25:40 AM12/12/11
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����������Ѿ�����������и��µ����⣺
��������sip��أ��ں�̨�������� ��sip trunk��Maximum Channels����1��extensions_additional.conf����ļ�����Ӧ��λ��Ҳ����Ч
OUT_3 = SIP/out31
OUTCID_3 =
OUTMAXCHANS_3 = 1

OUT_2 = SIP/out
OUTCID_2 =
OUTMAXCHANS_2 = 1

���Ǻ��е�ʱ����·�绰��ͨ��Ȼ��� ѯcdr������Ȼ���Ǵ���һ��sip��س�ȥ�ġ���������˵��bug���и�����˵Ҫ��� extensions_additional.conf����ļ��е����У�����û��˵����ġ���̴�ţ����ν����

On 2011/12/12 11:05, A-Lang, Hsu - Asterisk/Linux/IT Consultant wrote:
���� sip show peer 122sip���z�� codec �Ƿ��c�O������ͬ��


2011/12/11 weichenqi <weic...@gmail.com>
���Ѿ����ĵ����ú�sip trunk�����·�ɡ�
outgoing settings����

host=xx.xx.xx.xx
username=XXX22
secret=XXX
type=peer
context=from-trunk
dtmfmode=rfc2833
allow=g729&alaw&ulaw
fromuser=XXX22

���ڲ�����־����
�Ҹо���sip trunk�IJ���û�����úã������æ����ָ�л��

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A-Lang, Hsu - Asterisk/Linux/IT Consultant

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Dec 14, 2011, 3:29:52 AM12/14/11
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能說說是如何解決的嗎？


2011/12/12 weichenqi <weic...@gmail.com>
外呼的问题已经解决，现在有个新的问题：
我有两个sip落地，在后台设置两个 个sip trunk的Maximum Channels都是1，extensions_additional.conf这个文件中相应的位置也已生效

OUT_3 = SIP/out31
OUTCID_3 =
OUTMAXCHANS_3 = 1

OUT_2 = SIP/out
OUTCID_2 =
OUTMAXCHANS_2 = 1

但是呼叫的时候两路电话接通，然后查 询cdr还是依然都是从这一个sip落地出去的。网上有人说是bug，有个老外说要更改 extensions_additional.conf这个文件中的两行，但是没有说具体的。请教大牛，如何解决。


On 2011/12/12 11:05, A-Lang, Hsu - Asterisk/Linux/IT Consultant wrote:
執行 sip show peer 122sip，檢查 codec 是否與設定的相同。


2011/12/11 weichenqi <weic...@gmail.com>
我已经按文档配置好sip trunk和外呼路由。
outgoing settings如下

host=xx.xx.xx.xx
username=XXX22
secret=XXX
type=peer
context=from-trunk
dtmfmode=rfc2833
allow=g729&alaw&ulaw
fromuser=XXX22

现在拨号日志如下
我感觉是sip trunk的参数没有配置好，求大大帮忙。万分感谢。

--
您已訂閱「Google 網上論壇」的「Taiwan Asterisk Users' Group - www.osslab.com.tw」群組，因此我們特別傳送這封郵件通知您。
如要在此群組張貼留言，請傳送電子郵件至 aster...@googlegroups.com。
如要取消訂閱此群組，請傳送電子郵件至 asterisk-tw...@googlegroups.com。
如需更多選項，請造訪此群組：http://groups.google.com/group/asterisk-tw?hl=zh-TW。


--
您已訂閱「Google 網上論壇」的「Taiwan Asterisk Users' Group - www.osslab.com.tw」群組，因此我們特別傳送這封郵件通知您。
如要在此群組張貼留言，請傳送電子郵件至 aster...@googlegroups.com。
如要取消訂閱此群組，請傳送電子郵件至 asterisk-tw...@googlegroups.com。
如需更多選項，請造訪此群組：http://groups.google.com/group/asterisk-tw?hl=zh-TW。

--
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weichenqi

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Dec 14, 2011, 3:48:12 AM12/14/11
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��extensions_custom.conf�ļ��м����ˣ�
[from-internal]
include => from-internal-xfer
include => bad-number
#############
exten => _X.,1,dial(SIP/sip31/${EXTEN},60)
exten => _X.,2,dial(SIP/sip40/${EXTEN},60)

����Ϳ�������ˡ�
���ij��sip trunk ����max channels��������Ȼû�н���������������ط�����channels������Ȼ��sip31����·����ˣ��ͻ���sip40�ߡ����ǻ�ʵ���ҵ� �����˰ɡ�
����ڱ�������sip trunk��max channels�����⣬A-Lang����н���취ô��


On 2011/12/14 16:29, A-Lang, Hsu - Asterisk/Linux/IT Consultant wrote:
���f�f����ν�Q�Ć᣿


2011/12/12 weichenqi <weic...@gmail.com>
����������Ѿ�����������и��µ����⣺
�� ������sip��أ��ں�̨�������� ��sip trunk��Maximum Channels����1��extensions_additional.conf����ļ�����Ӧ��λ��Ҳ����Ч
OUT_3 = SIP/out31
OUTCID_3 =
OUTMAXCHANS_3 = 1

OUT_2 = SIP/out
OUTCID_2 =
OUTMAXCHANS_2 = 1

�� �Ǻ��е�ʱ����·�绰��ͨ��Ȼ��� ѯcdr������Ȼ���Ǵ���һ��sip��س�ȥ�ġ���������˵��bug���и�����˵Ҫ��� extensions_additional.conf����ļ��е����У�����û��˵����ġ���̴�ţ����ν����


On 2011/12/12 11:05, A-Lang, Hsu - Asterisk/Linux/IT Consultant wrote:
���� sip show peer 122sip���z�� codec �Ƿ��c�O������ͬ��


2011/12/11 weichenqi <weic...@gmail.com>
���Ѿ����ĵ����ú�sip trunk�����·�ɡ�
outgoing settings����
host=xx.xx.xx.xx
username=XXX22
secret=XXX
type=peer
context=from-trunk
dtmfmode=rfc2833
allow=g729&alaw&ulaw
fromuser=XXX22

���ڲ�����־����
�Ҹо���sip trunk�IJ���û�����úã������æ����ָ�л��

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A-Lang, Hsu - Asterisk/Linux/IT Consultant

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Dec 20, 2011, 12:32:13 AM12/20/11
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系統如果正常運作下，不應該需要這兩行才能外撥，通常只要透過 FreePBX 介面就可以完成所有設定。

至於 max channels 的問題，你需要檢查 dialplan，一行行作驗證，說不定是 Bug。


2011/12/14 weichenqi <weic...@gmail.com>
在extensions_custom.conf文件中加入了：

[from-internal]
include => from-internal-xfer
include => bad-number
#############
exten => _X.,1,dial(SIP/sip31/${EXTEN},60)
exten => _X.,2,dial(SIP/sip40/${EXTEN},60)

这样就可以外呼了。
针对某个sip trunk 设置max channels的问题依然没有解决。不过可以让落地方限制channels的量，然后sip31如果电路溢出了，就会往sip40走。算是基本实现我的 需求了吧。
这个在本机限制sip trunk的max channels的问题，A-Lang大大有解决办法么。


On 2011/12/14 16:29, A-Lang, Hsu - Asterisk/Linux/IT Consultant wrote:
能說說是如何解決的嗎？


2011/12/12 weichenqi <weic...@gmail.com>
外呼的问题已经解决，现在有个新的问题：
我 有两个sip落地，在后台设置两个 个sip trunk的Maximum Channels都是1，extensions_additional.conf这个文件中相应的位置也已生效

OUT_3 = SIP/out31
OUTCID_3 =
OUTMAXCHANS_3 = 1

OUT_2 = SIP/out
OUTCID_2 =
OUTMAXCHANS_2 = 1

但 是呼叫的时候两路电话接通，然后查 询cdr还是依然都是从这一个sip落地出去的。网上有人说是bug，有个老外说要更改 extensions_additional.conf这个文件中的两行，但是没有说具体的。请教大牛，如何解决。


On 2011/12/12 11:05, A-Lang, Hsu - Asterisk/Linux/IT Consultant wrote:
執行 sip show peer 122sip，檢查 codec 是否與設定的相同。


2011/12/11 weichenqi <weic...@gmail.com>
我已经按文档配置好sip trunk和外呼路由。
outgoing settings如下

host=xx.xx.xx.xx
username=XXX22
secret=XXX
type=peer
context=from-trunk
dtmfmode=rfc2833
allow=g729&alaw&ulaw
fromuser=XXX22

现在拨号日志如下
我感觉是sip trunk的参数没有配置好，求大大帮忙。万分感谢。

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chq.wei

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Dec 21, 2011, 10:09:09 AM12/21/11
to aster...@googlegroups.com

我安装的是elastix集成的iso。无论哪个版本都有这个问题，必须要加两行才能外拨。而且也不能通过freepbx 界面调整sip落地的优先级。只能修改这两行的sip 名字的字段才行。

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