نوشته اصلی توسط Iman.gh
من همين فايل شما رو دارم تست ميكنم خطا ميگيرم ديگه
بالا رو ببينيد ll grep گرفتم
بعد مگه اين خروجي نميگه كه استريكس من 1.6 هستش ؟
core show version
Asterisk 1.6.2.10 built by root @ rpmbuild32-2.elastix.palosanto.com on a i686 running Linux on 2010-07-29 08:45:42 UTC
و سوال ديگه اينكه اگه g729 نداريم اون چيه تو خروجي نشون ميده ! كل codec ها هستش ؟
سلام
از این چیزی که گذاشتید بله 1.6 هستش.
service asterisk restart
wget http://asterisk.hosting.lv/bin162/co...glibc-core2.so
mv codec_g729-ast16-icc-glibc-core2.so /usr/lib/asterisk/modules
chmod +x /usr/lib/asterisk/modules/codec_g729-ast16-icc-glibc-core2.so
wget http://asterisk.hosting.lv/bin162/codec_g723-ast16-icc-glibc-core2.so
mv codec_g723-ast16-icc-glibc-core2.so /usr/lib/asterisk/modules
chmod +x /usr/lib/asterisk/modules/codec_g723-ast16-icc-glibc-core2.so
asterisk -rx "module load codec_g729-ast16-icc-glibc-core2.so"
asterisk -rx "module load codec_g723-ast16-icc-glibc-core2.so"
service asterisk restart
sleep 2
asterisk -rx "core show codecs"
vi /etc/asterisk/sip_general_custom.conf
allow=g729
allow=G723
پس چرا 1.4 دارید تست می کنید ؟
این که من گذاشتم برای Core2 هستش از این سایت می تونید کدکی که با cpu یکی هست را DL کنید.
Asterisk codecs G.729 G.723.1 binaries
ویرایش توسط Iman.gh : 2011-03-06 در ساعت 01:46 PM
آره ايمان جان ! پس من اشتباه كرده بودم ديگه ! 1.4 رو ميخواستم روي 1.6 نصب كنم
سرور فيزيكي من DL 380 G5 هستش كه اين سرور مجازي روش نصب شده و CPU اش از نوع XEON هست كه در Note هاي لينك شما آمده
Use Pentium 4 build for Pentium D. Use Pentium 3 for VIA C3 and Pentium 4 for VIA C7. XEON is Pentium3/Pentium4/Core2 with larger cache. Celeron is Pentium3/Pentium4/Core2 with smaller cache. Basically, there are 4 software cores: MMX, SSE, SSE2, SSE3. Pick closest match from Intel CPU line and try it.
ولي بدبختي اينجا پر است از Core2 و Pentium4 براي اين نوع CPU
من كه همش رو گرفتم ولي كدوم رو تست كنم بالاخره ! گيج شدم
اين كه حل بشه بايد برم سراغ سوال بعدي كه چطور الستيكس و bria رو مجبور كنم از اين كدك استفاده كنند
سلام
از همین که من گذاشتم استفاده کن خدا کنه جواب بده
سلام مجدد
آقا من Core2 رو تست كردم
نتيجه اين شد
Loaded codec_g729-ast16-icc-glibc-core2.so
== Registered translator 'g729tolin' from format g729 to slin, cost 1999
== Registered translator 'lintog729' from format slin to g729, cost 4999
Loaded codec_g729-ast16-icc-glibc-core2.so => (g729 Coder/Decoder, based on IPP)
حله ؟
ولي وقتي رفتم تو bria و زنگ زدم تست كردم خروجي اين شد :
(راستي اون allow=g729 رو هم تو فايل اضافه كردم)
(خروجي sip show cahnnels)
- Executing [s@macro-exten-vm:1] Macro("SIP/110-00000000", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/110-00000000", "AMPUSER=110") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/110-00000000", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/110-00000000", "1?Set(REALCALLERIDNUM=110)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/110-00000000", "AMPUSER=110") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/110-00000000", "AMPUSERCIDNAME=Mohammad GAnji") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/110-00000000", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/110-00000000", "AMPUSERCID=110") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/110-00000000", "CALLERID(all)="Mohammad GAnji" <110>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/110-00000000", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/110-00000000", "0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/110-00000000", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/110-00000000", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/110-00000000", "Using CallerID "Mohammad GAnji" <110>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/110-00000000", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/110-00000000", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/110-00000000", "EXTTOCALL=200") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/110-00000000", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/110-00000000", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/110-00000000", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/110-00000000", "record-enable,200,IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/110-00000000", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/110-00000000", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/110-00000000", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/110-00000000", "1?IN") in new stack
-- Goto (macro-record-enable,s,20)
-- Executing [s@macro-record-enable:20] ExecIf("SIP/110-00000000", "1?MacroExit()") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/110-00000000", "dial,,tr,200") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/110-00000000", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/110-00000000", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'Mohammad GAnji' number is '110'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 200 to extension map
-- dialparties.agi: Extension 200 cf is disabled
-- dialparties.agi: Extension 200 do not disturb is disabled
dialparties.agi: EXTENSION_STATE: 4 (UNAVAILABLE)
dialparties.agi: Extension 200 has ExtensionState: 4
-- dialparties.agi: Checking CW and CFB status for extension 200
-- dialparties.agi: dbset CALLTRACE/200 to 110
-- dialparties.agi: Filtered ARG3: 200
-- <SIP/110-00000000>AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/110-00000000", "SIP/200,,tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dial:8] Set("SIP/110-00000000", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-dial:9] GosubIf("SIP/110-00000000", "0?CHANUNAVAIL,1") in new stack
-- Executing [s@macro-exten-vm:10] GotoIf("SIP/110-00000000", "0?exit,return") in new stack
-- Executing [s@macro-exten-vm:11] Set("SIP/110-00000000", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/110-00000000", "0?docfu,1") in new stack
-- Executing [s@macro-exten-vm:13] GosubIf("SIP/110-00000000", "0?docfb,1") in new stack
-- Executing [s@macro-exten-vm:14] Set("SIP/110-00000000", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:15] NoOp("SIP/110-00000000", "Voicemail is 'novm'") in new stack
-- Executing [s@macro-exten-vm:16] GotoIf("SIP/110-00000000", "1?s-CHANUNAVAIL,1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-exten-vm:1] NoOp("SIP/110-00000000", "IVR_RETVM: IVR_CONTEXT: ") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:2] GotoIf("SIP/110-00000000", "0?exit,1") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:3] PlayTones("SIP/110-00000000", "congestion") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:4] Congestion("SIP/110-00000000", "10") in new stack
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 4) exited non-zero on 'SIP/110-00000000' in macro 'exten-vm'
== Spawn extension (from-internal, 200, 1) exited non-zero on 'SIP/110-00000000'
-- Executing [h@from-internal:1] Macro("SIP/110-00000000", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/110-00000000", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/110-00000000", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/110-00000000", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/110-00000000", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/110-00000000", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/110-00000000", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/110-00000000' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/110-00000000'
بعد رفتم تو bria
اون ulaw رو از كدك ها در اوردم و گذاشتم فقط g729 و چند تا ديگه بمونه
ديگه اصلا تماس برقرار نشد
فكر كنم تا الان اوضاع خوبه و كدك بايد نصب شده باشه
فقط چطوري به bria و eybeam حالي كنيم كه از اين كدك استفاده كنند
والبته چه تنظيم ديگري روي سرور لازمه (غير از اون خط allow) - راستي فايله خالي بودا ! من فقط همين يك خط رو اضافه كردم
پيشاپيش تشكر
سلام
یعنی زنگ هم نمی خوره ؟
روش نصب همینه و چیزه دیگری ندارد.
میگم شاید این کدک با CPU کار نمی کنه.
با Winscp به این مسیر برید /usr/lib/asterisk/modules و کدک را پاک کنید و یکی دیگر را تست کنید.
بله اون خالی هستش
نه اجازه بديد
سرويس رو ريستارت نكرده بودم
البته ظاهرا Eyebeam با اينكه نشون ميده اما ساپورت نميكنه
رفتم تو بريا
دو تا اكانت ست كردم
همه كدك ها رو هم پاك كردم فقط G729 رو گذاشتم
الان زنگ ميخوره
البته يه ذره قاطي بازي داره
مثلا با اين اكانت اون يكي رو ميگيرم زنگ ميخوره اما با اون يكي اين يكي رو ميگيرم ارور 503 ميده و service not available
پس به نظر مياد نصب باشه
چطور ميشه تستش كرد ! راهي داره ؟
برای تست بهترین روش اینه که یک تماس برقرار کنید و دستور sip show channels را اجرا کنید . اگر از کدک g729 استفاده کنه اونجا نمایش میده .. اینجوری می توانید مطمئن بشید .
امیدجان من همین خروجی رو بالا گذاشتم دیگه
امروز یکبار دیگه تست میکنم
راستی اون ارور علتش چیه ؟ که یکی اون یکی رو میگیره ولی اون یکی میخواد این یکی رو بپیره بعضا ارور 503 میده و بعضا 486 اگه اشتباه نکنم (یا 468)
اون دستورات بالا که نوشته registered نشون میده نصب شده دیگه نه ؟ اون خط allow رو هم که نوشتم
راستی disallow=all نمیخواد ؟
سلام
میتونید disallow=all کنید و بعد G.711 را allow کنید که برای فکس این کدک هستش.
ولی من این کار را کردم ولی فکس کار نکرد تا disallow=all را برداشتم و فقط G.729 را Allow کردم.
جسارتا ممكنه مشكل از ترتيب تعريفتون بوده باشه ؟سلام
میتونید disallow=all کنید و بعد G.711 را allow کنید که برای فکس این کدک هستش.
ولی من این کار را کردم ولی فکس کار نکرد تا disallow=all را برداشتم و فقط G.729 را Allow کردم.
يعني بايد اول اون allow رو مينوشتيد بعد disallow=all رو
دوستان ما دو طرف Bria نصب كرديم و تو Codec ها همه چيز رو غير از G729 حذف كرديم
تماس ها برقرار شد و هيچ مشكلي نبود
يعني ما بالاخره G729 رو نصب كرديم يا نه ؟
خروجي رو ببينيد :
-- Executing [s@macro-exten-vm:9] Macro("SIP/110-00000025", "dial,15,tr,112") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/110-00000025", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/110-00000025", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'Mohammad GAnji' number is '110'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 112 to extension map
-- dialparties.agi: Extension 112 cf is disabled
-- dialparties.agi: Extension 112 do not disturb is disabled
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
dialparties.agi: Extension 112 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 112
-- dialparties.agi: dbset CALLTRACE/112 to 110
-- dialparties.agi: Filtered ARG3: 112
-- <SIP/110-00000025>AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/110-00000025", "SIP/112,15,tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 112
-- SIP/112-00000026 is ringing
== Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/110-00000025' in macro 'dial'
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/110-00000025' in macro 'exten-vm'
== Spawn extension (from-internal, 112, 1) exited non-zero on 'SIP/110-00000025'
-- Executing [h@from-internal:1] Macro("SIP/110-00000025", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/110-00000025", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/110-00000025", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/110-00000025", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/110-00000025", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/110-00000025", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/110-00000025", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/110-00000025' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/110-00000025'
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [110@from-internal:1] Macro("SIP/110-00000027", "exten-vm,novm,110") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/110-00000027", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/110-00000027", "AMPUSER=110") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/110-00000027", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/110-00000027", "1?Set(REALCALLERIDNUM=110)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/110-00000027", "AMPUSER=110") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/110-00000027", "AMPUSERCIDNAME=Mohammad GAnji") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/110-00000027", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/110-00000027", "AMPUSERCID=110") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/110-00000027", "CALLERID(all)="Mohammad GAnji" <110>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/110-00000027", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/110-00000027", "0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/110-00000027", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/110-00000027", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/110-00000027", "Using CallerID "Mohammad GAnji" <110>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/110-00000027", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/110-00000027", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/110-00000027", "EXTTOCALL=110") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/110-00000027", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/110-00000027", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/110-00000027", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/110-00000027", "record-enable,110,IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/110-00000027", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/110-00000027", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/110-00000027", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/110-00000027", "1?IN") in new stack
-- Goto (macro-record-enable,s,20)
-- Executing [s@macro-record-enable:20] ExecIf("SIP/110-00000027", "1?MacroExit()") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/110-00000027", "dial,,tr,110") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/110-00000027", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/110-00000027", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'Mohammad GAnji' number is '110'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 110 to extension map
-- dialparties.agi: Extension 110 cf is disabled
-- dialparties.agi: Extension 110 do not disturb is disabled
dialparties.agi: EXTENSION_STATE: 1 (INUSE)
dialparties.agi: Extension 110 has ExtensionState: 1
-- dialparties.agi: Checking CW and CFB status for extension 110
dialparties.agi: Extension 110 is not available to be called
dialparties.agi: Extension 110 has call waiting disabled
-- dialparties.agi: Filtered ARG3:
-- <SIP/110-00000027>AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:4] NoOp("SIP/110-00000027", "Returned from dialparties with no extensions to call and DIALSTATUS: BUSY") in new stack
-- Executing [s@macro-exten-vm:10] GotoIf("SIP/110-00000027", "0?exit,return") in new stack
-- Executing [s@macro-exten-vm:11] Set("SIP/110-00000027", "SV_DIALSTATUS=BUSY") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/110-00000027", "0?docfu,1") in new stack
-- Executing [s@macro-exten-vm:13] GosubIf("SIP/110-00000027", "0?docfb,1") in new stack
-- Executing [s@macro-exten-vm:14] Set("SIP/110-00000027", "DIALSTATUS=BUSY") in new stack
-- Executing [s@macro-exten-vm:15] NoOp("SIP/110-00000027", "Voicemail is 'novm'") in new stack
-- Executing [s@macro-exten-vm:16] GotoIf("SIP/110-00000027", "1?s-BUSY,1") in new stack
-- Goto (macro-exten-vm,s-BUSY,1)
-- Executing [s-BUSY@macro-exten-vm:1] NoOp("SIP/110-00000027", "Extension is reporting BUSY and not passing to Voicemail") in new stack
-- Executing [s-BUSY@macro-exten-vm:2] GotoIf("SIP/110-00000027", "0?exit,1") in new stack
-- Executing [s-BUSY@macro-exten-vm:3] PlayTones("SIP/110-00000027", "busy") in new stack
-- Executing [s-BUSY@macro-exten-vm:4] Busy("SIP/110-00000027", "20") in new stack
== Spawn extension (macro-exten-vm, s-BUSY, 4) exited non-zero on 'SIP/110-00000027' in macro 'exten-vm'
== Spawn extension (from-internal, 110, 1) exited non-zero on 'SIP/110-00000027'
-- Executing [h@from-internal:1] Macro("SIP/110-00000027", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/110-00000027", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/110-00000027", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/110-00000027", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/110-00000027", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/110-00000027", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/110-00000027", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/110-00000027' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/110-00000027'
-- Unregistered SIP '110'
-- Unregistered SIP '112'
Elastix-Server*CLI>
login as: root
root@140.100.30.2's password:
Last login: Sun Feb 20 15:15:09 2011 from 192.168.60.11
Welcome to Elastix
----------------------------------------------------
To access your Elastix System, using a separate workstation (PC/MAC/Linux)
Open the Internet Browser using the following URL:
http://140.100.30.2
[root@Elastix-Server ~]# asterisk -r
Asterisk 1.6.2.10, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
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.
================================================== =======================
Connected to Asterisk 1.6.2.10 currently running on Elastix-Server (pid = 31166)
Verbosity is at least 3
Elastix-Server*CLI> sip show ch
channels channelstats channel
Elastix-Server*CLI> sip show channel
Usage: sip show channel <call-id>
Provides detailed status on a given SIP dialog (identified by SIP call-id).
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [110@from-internal:1] Macro("SIP/200-0000003d", "exten-vm,novm,110") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/200-0000003d", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/200-0000003d", "AMPUSER=200") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/200-0000003d", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/200-0000003d", "1?Set(REALCALLERIDNUM=200)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/200-0000003d", "AMPUSER=200") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/200-0000003d", "AMPUSERCIDNAME=noori") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/200-0000003d", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/200-0000003d", "AMPUSERCID=200") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/200-0000003d", "CALLERID(all)="noori" <200>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/200-0000003d", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/200-0000003d", "0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/200-0000003d", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/200-0000003d", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/200-0000003d", "Using CallerID "noori" <200>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/200-0000003d", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/200-0000003d", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/200-0000003d", "EXTTOCALL=110") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/200-0000003d", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/200-0000003d", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/200-0000003d", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/200-0000003d", "record-enable,110,IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/200-0000003d", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/200-0000003d", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/200-0000003d", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/200-0000003d", "1?IN") in new stack
-- Goto (macro-record-enable,s,20)
-- Executing [s@macro-record-enable:20] ExecIf("SIP/200-0000003d", "1?MacroExit()") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/200-0000003d", "dial,,tr,110") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/200-0000003d", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/200-0000003d", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'noori' number is '200'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 110 to extension map
-- dialparties.agi: Extension 110 cf is disabled
-- dialparties.agi: Extension 110 do not disturb is disabled
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
dialparties.agi: Extension 110 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 110
-- dialparties.agi: dbset CALLTRACE/110 to 200
-- dialparties.agi: Filtered ARG3: 110
-- <SIP/200-0000003d>AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/200-0000003d", "SIP/110,,tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Extension Changed 110[ext-local] new state Ringing for Notify User 200
-- Called 110
-- SIP/110-0000003e is ringing
Elastix-Server*CLI>
به نظر همه چيز اكي هستش اما Sip Show Channels رو كه ميگيرم چيزي از G729 نميبينم
ببينيد
بچه ها چي شد ؟ بالاخره G729 هستيم يا نه
بفرماييد چون تازه سوالاتم شورع شده و ميخوام از اين مطمئن بشم و برم پرسشهاي بعدي رو بپرسم
سلام
آره disallow=all اول گذاشتم و بعد G.729 و G.711 را Allow کردم ولی G.711 فقط کار نکرد.
اگه فقط G.729 گذاشتی آره دیگه موفق شدی
خوب اين از G729
مراحل بعدي رو بفرماييد
اين عوض كردن TOS بسته ها در سافتفون و يا الستيكس و راه انداختن QoS Policy ها روي سوييچ ميتونه گام بعدي باشه ؟
خلاصه قضيه اينه كه اول پست هم گفتم ! يك خط 2 مگ بايد بشه روش 6 تا تماس همزمان داشت كه ترافيكشون در اولويت باشه
اين وسط هم از مخابرات (بستر اينترانت كشوري ) مي گذرند
مرسی ایمان جان که یا جواب نمیدی یا اگر جواب بدی تا آخرش و کامل با تفصیل و توضیحات هست. همیشه موفق باشی.
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