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salah.elharit200 at gm... Guest
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Posted: Thu Mar 26, 2015 6:03 am Post subject: [asterisk-users] call between snom 300 and aastra 6731i |
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hello list
i need your help please regarding an issue with snom300 and aastra6731i using asteriskÂ
11.13.0 Â asterisk
snom 300 Â 8.7.3.25
astra 6731i 2.6.0.2019
i have configured the trunks like below
100 in snom300
200 in snom300
300 in aastra6731i
400 in x-lite
the calls between x-lite and aastra ====ok inbound and outbound
the calls between x-lite and snom300====> ok inbound and outbound
the issue just between snom and aastra i can call from aastra to snom without issue
but when itry to call from snom300 to aastra6731i  i get bad request all the time
i test with 3 snom300 i get the same result
please any body have the snom and aastra can help me in order to fixe this issue
thanks and regards. |
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salah.elharit200 at gm... Guest
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Posted: Fri Mar 27, 2015 12:06 pm Post subject: [asterisk-users] call between snom 300 and aastra 6731i |
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please no body has som with aastra can help me in this issue
2015-03-26 11:02 GMT+00:00 Salaheddine Elharit <salah.elharit200@gmail.com (salah.elharit200@gmail.com)>:
Quote: | hello list
i need your help please regarding an issue with snom300 and aastra6731i using asteriskÂ
11.13.0 Â asterisk
snom 300 Â 8.7.3.25
astra 6731i 2.6.0.2019
i have configured the trunks like below
100 in snom300
200 in snom300
300 in aastra6731i
400 in x-lite
the calls between x-lite and aastra ====ok inbound and outbound
the calls between x-lite and snom300====> ok inbound and outbound
the issue just between snom and aastra i can call from aastra to snom without issue
but when itry to call from snom300 to aastra6731i  i get bad request all the time
i test with 3 snom300 i get the same result
please any body have the snom and aastra can help me in order to fixe this issue
thanks and regards.
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mailinglist+asterisk a... Guest
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Posted: Fri Mar 27, 2015 12:08 pm Post subject: [asterisk-users] call between snom 300 and aastra 6731i |
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You would need to give more information really.
Your sip.conf file listing the entries for the phones especially which codecs are permitted.
A copy of the 'asterisk -rvvv' console output when you make the call.
On 27/03/15 17:05, Salaheddine Elharit wrote:
Quote: | please no body has som with aastra can help me in this issue
2015-03-26 11:02 GMT+00:00 Salaheddine Elharit <salah.elharit200@gmail.com (salah.elharit200@gmail.com)>:
Quote: | hello list
i need your help please regarding an issue with snom300 and aastra6731i using asterisk
11.13.0 asterisk
snom 300 8.7.3.25
astra 6731i 2.6.0.2019
i have configured the trunks like below
100 in snom300
200 in snom300
300 in aastra6731i
400 in x-lite
the calls between x-lite and aastra ====ok inbound and outbound
the calls between x-lite and snom300====> ok inbound and outbound
the issue just between snom and aastra i can call from aastra to snom without issue
but when itry to call from snom300 to aastra6731i i get bad request all the time
i test with 3 snom300 i get the same result
please any body have the snom and aastra can help me in order to fixe this issue
thanks and regards.
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salah.elharit200 at gm... Guest
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Posted: Fri Mar 27, 2015 12:44 pm Post subject: [asterisk-users] call between snom 300 and aastra 6731i |
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thank you for your response below the asterisk -vvvr
Using SIP RTP TOS bits 184
 == Using SIP RTP CoS mark 5
  -- Executing [0176XXXXXX@from-internal:1] Macro("SIP/300-00000192", "user-callerid,LIMIT,EXTERNAL,") in new stack
  -- Executing [s@macro-user-callerid:1] Set("SIP/300-00000192", "TOUCH_MONITOR=1427481319.470") in new stack
  -- Executing [s@macro-user-callerid:2] Set("SIP/300-00000192", "AMPUSER=300") in new stack
  -- Executing [s@macro-user-callerid:3] GotoIf("SIP/300-00000192", "0?report") in new stack
  -- Executing [s@macro-user-callerid:4] ExecIf("SIP/300-00000192", "1?Set(REALCALLERIDNUM=300)") in new stack
  -- Executing [s@macro-user-callerid:5] Set("SIP/300-00000192", "AMPUSER=300") in new stack
  -- Executing [s@macro-user-callerid:6] GotoIf("SIP/300-00000192", "0?limit") in new stack
  -- Executing [s@macro-user-callerid:7] Set("SIP/300-00000192", "AMPUSERCIDNAME=300") in new stack
  -- Executing [s@macro-user-callerid:8] GotoIf("SIP/300-00000192", "0?report") in new stack
  -- Executing [s@macro-user-callerid:9] Set("SIP/300-00000192", "AMPUSERCID=300") in new stack
  -- Executing [s@macro-user-callerid:10] Set("SIP/300-00000192", "__DIAL_OPTIONS=tr") in new stack
  -- Executing [s@macro-user-callerid:11] Set("SIP/300-00000192", "CALLERID(all)="300" <300>") in new stack
  -- Executing [s@macro-user-callerid:12] GotoIf("SIP/300-00000192", "0?limit") in new stack
  -- Executing [s@macro-user-callerid:13] ExecIf("SIP/300-00000192", "1?Set(GROUP(concurrency_limit)=300)") in new stack
  -- Executing [s@macro-user-callerid:14] ExecIf("SIP/300-00000192", "0?Set(CHANNEL(language)=)") in new stack
  -- Executing [s@macro-user-callerid:15] GotoIf("SIP/300-00000192", "1?continue") in new stack
  -- Goto (macro-user-callerid,s,2
  -- Executing [s@macro-user-callerid:28] Set("SIP/300-00000192", "CALLERID(number)=300") in new stack
  -- Executing [s@macro-user-callerid:29] Set("SIP/300-00000192", "CALLERID(name)=300") in new stack
  -- Executing [s@macro-user-callerid:30] Set("SIP/300-00000192", "CDR(cnum)=300") in new stack
  -- Executing [s@macro-user-callerid:31] Set("SIP/300-00000192", "CDR(cnam)=300") in new stack
  -- Executing [s@macro-user-callerid:32] Set("SIP/300-00000192", "CHANNEL(language)=en") in new stack
  -- Executing [0176XXXXXX@from-internal:2] Set("SIP/300-00000192", "MOHCLASS=default") in new stack
  -- Executing [0176XXXXXX@from-internal:3] ExecIf("SIP/300-00000192", "0?Set(TRUNKCIDOVERRIDE=0176XXXXXX)") in new stack
  -- Executing [0176XXXXXX@from-internal:4] Set("SIP/300-00000192", "_NODEST=") in new stack
  -- Executing [0176XXXXXX@from-internal:5] Gosub("SIP/300-00000192", "sub-record-check,s,1(out,0176XXXXXX,)") in new stack
  -- Executing [s@sub-record-check:1] Set("SIP/300-00000192", "REC_POLICY_MODE_SAVE=") in new stack
  -- Executing [s@sub-record-check:2] GotoIf("SIP/300-00000192", "1?check") in new stack
  -- Goto (sub-record-check,s,7)
  -- Executing [s@sub-record-check:7] Set("SIP/300-00000192", "__MON_FMT=wav") in new stack
  -- Executing [s@sub-record-check:8] GotoIf("SIP/300-00000192", "1?next") in new stack
  -- Goto (sub-record-check,s,11)
  -- Executing [s@sub-record-check:11] ExecIf("SIP/300-00000192", "0?Return()") in new stack
  -- Executing [s@sub-record-check:12] ExecIf("SIP/300-00000192", "0?Set(__REC_POLICY_MODE=)") in new stack
  -- Executing [s@sub-record-check:13] GotoIf("SIP/300-00000192", "0?out,1") in new stack
  -- Executing [s@sub-record-check:14] Set("SIP/300-00000192", "__REC_STATUS=INITIALIZED") in new stack
  -- Executing [s@sub-record-check:15] Set("SIP/300-00000192", "NOW=1427481319") in new stack
  -- Executing [s@sub-record-check:16] Set("SIP/300-00000192", "__DAY=27") in new stack
  -- Executing [s@sub-record-check:17] Set("SIP/300-00000192", "__MONTH=03") in new stack
  -- Executing [s@sub-record-check:18] Set("SIP/300-00000192", "__YEAR=2015") in new stack
  -- Executing [s@sub-record-check:19] Set("SIP/300-00000192", "__TIMESTR=20150327-183519") in new stack
  -- Executing [s@sub-record-check:20] Set("SIP/300-00000192", "__FROMEXTEN=300") in new stack
  -- Executing [s@sub-record-check:21] Set("SIP/300-00000192", "__CALLFILENAME=out-0176XXXXXX-300-20150327-183519-1427481319.470") in new stack
  -- Executing [s@sub-record-check:22] Goto("SIP/300-00000192", "out,1") in new stack
  -- Goto (sub-record-check,out,1)
  -- Executing [out@sub-record-check:1] ExecIf("SIP/300-00000192", "1?Set(__REC_POLICY_MODE=always)") in new stack
  -- Executing [out@sub-record-check:2] GosubIf("SIP/300-00000192", "1?record,1(exten,0176XXXXXX,300)") in new stack
  -- Executing [record@sub-record-check:1] Set("SIP/300-00000192", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
  -- Executing [record@sub-record-check:2] MixMonitor("SIP/300-00000192", "2015/03/27/out-0176XXXXXX-300-20150327-183519-1427481319.470.wav,,") in new stack
  -- Executing [record@sub-record-check:3] Set("SIP/300-00000192", "__REC_STATUS=RECORDING") in new stack
  -- Executing [record@sub-record-check:4] Set("SIP/300-00000192", "CDR(recordingfile)=out-0176XXXXXX-300-20150327-183519-1427481319.470.wav") in new stack
  -- Executing [record@sub-record-check:5] Return("SIP/300-00000192", "") in new stack
  -- Executing [out@sub-record-check:3] Return("SIP/300-00000192", "") in new stack
  -- Executing [0176XXXXXX@from-internal:6] Macro("SIP/300-00000192", "dialout-trunk,5,00XX17621XXXX,,off") in new stack
 == Begin MixMonitor Recording SIP/300-00000192
  -- Executing [s@macro-dialout-trunk:1] Set("SIP/300-00000192", "DIAL_TRUNK=5") in new stack
  -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/300-00000192", "0?sub-pincheck,s,1()") in new stack
  -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/300-00000192", "0?disabletrunk,1") in new stack
  -- Executing [s@macro-dialout-trunk:4] Set("SIP/300-00000192", "DIAL_NUMBER=00XX17621XXXX") in new stack
  -- Executing [s@macro-dialout-trunk:5] Set("SIP/300-00000192", "DIAL_TRUNK_OPTIONS=tr") in new stack
  -- Executing [s@macro-dialout-trunk:6] Set("SIP/300-00000192", "OUTBOUND_GROUP=OUT_5") in new stack
  -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/300-00000192", "0?nomax") in new stack
  -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/300-00000192", "0?chanfull") in new stack
  -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/300-00000192", "0?skipoutcid") in new stack
  -- Executing [s@macro-dialout-trunk:10] Set("SIP/300-00000192", "DIAL_TRUNK_OPTIONS=") in new stack
  -- Executing [s@macro-dialout-trunk:11] Macro("SIP/300-00000192", "outbound-callerid,5") in new stack
  -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/300-00000192", "0?Set(CALLERPRES()=)") in new stack
  -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/300-00000192", "0?Set(REALCALLERIDNUM=300)") in new stack
  -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/300-00000192", "1?normcid") in new stack
  -- Goto (macro-outbound-callerid,s,6)
  -- Executing [s@macro-outbound-callerid:6] Set("SIP/300-00000192", "USEROUTCID=300") in new stack
  -- Executing [s@macro-outbound-callerid:7] Set("SIP/300-00000192", "EMERGENCYCID=") in new stack
  -- Executing [s@macro-outbound-callerid:8] Set("SIP/300-00000192", "TRUNKOUTCID=") in new stack
  -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/300-00000192", "1?trunkcid") in new stack
  -- Goto (macro-outbound-callerid,s,14)
  -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/300-00000192", "0?Set(CALLERID(all)=)") in new stack
  -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/300-00000192", "1?Set(CALLERID(all)=300)") in new stack
  -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/300-00000192", "0?Set(CALLERID(all)=)") in new stack
  -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/300-00000192", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
  -- Executing [s@macro-outbound-callerid:18] Set("SIP/300-00000192", "CDR(outbound_cnum)=300") in new stack
  -- Executing [s@macro-outbound-callerid:19] Set("SIP/300-00000192", "CDR(outbound_cnam)=") in new stack
  -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/300-00000192", "1?sub-flp-5,s,1()") in new stack
  -- Executing [s@sub-flp-5:1] ExecIf("SIP/300-00000192", "0?Set(TARGET_FLP_5=00XX0XX176XXXXXX)") in new stack
  -- Executing [s@sub-flp-5:2] GotoIf("SIP/300-00000192", "0?match") in new stack
  -- Executing [s@sub-flp-5:3] Return("SIP/300-00000192", "") in new stack
  -- Executing [s@macro-dialout-trunk:13] Set("SIP/300-00000192", "OUTNUM=00XX17621XXXX") in new stack
  -- Executing [s@macro-dialout-trunk:14] Set("SIP/300-00000192", "custom=SIP/FD") in new stack
  -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/300-00000192", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
  -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/300-00000192", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack
  -- Executing [s@macro-dialout-trunk:17] Macro("SIP/300-00000192", "dialout-trunk-predial-hook,") in new stack
  -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/300-00000192", "") in new stack
  -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/300-00000192", "0?bypass,1") in new stack
  -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/300-00000192", "1?Set(CONNECTEDLINE(num,i)=00XX17621XXXX)") in new stack
  -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/300-00000192", "1?Set(CONNECTEDLINE(name,i)=CID:300)") in new stack
  -- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/300-00000192", "0?customtrunk") in new stack
  -- Executing [s@macro-dialout-trunk:22] Dial("SIP/300-00000192", "SIP/FD/00XX17621XXXX,300,") in new stack
 == Using SIP RTP TOS bits 184
 == Using SIP RTP CoS mark 5
  -- Called SIP/FD/00XX17621XXXX
  -- Got SIP response 480 "No address found" back from 217.195.XX.XXX:5060
  -- SIP/FD-00000193 is circuit-busy
 == Everyone is busy/congested at this time (1:0/1/0)
  -- Executing [s@macro-dialout-trunk:23] NoOp("SIP/300-00000192", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 19") in new stack
  -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/300-00000192", "0?continue,1:s-CONGESTION,1") in new stack
  -- Goto (macro-dialout-trunk,s-CONGESTION,1)
  -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/300-00000192", "RC=19") in new stack
  -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/300-00000192", "19,1") in new stack
  -- Goto (macro-dialout-trunk,19,1)
  -- Executing [19@macro-dialout-trunk:1] Goto("SIP/300-00000192", "continue,1") in new stack
  -- Goto (macro-dialout-trunk,continue,1)
  -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/300-00000192", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 19 - failing through to other trunks") in new stack
  -- Executing [continue@macro-dialout-trunk:2] Set("SIP/300-00000192", "CALLERID(number)=300") in new stack
  -- Executing [0176XXXXXX@from-internal:7] Macro("SIP/300-00000192", "outisbusy,") in new stack
  -- Executing [s@macro-outisbusy:1] Progress("SIP/300-00000192", "") in new stack
  -- Executing [s@macro-outisbusy:2] GotoIf("SIP/300-00000192", "0?emergency,1") in new stack
  -- Executing [s@macro-outisbusy:3] GotoIf("SIP/300-00000192", "0?intracompany,1") in new stack
  -- Executing [s@macro-outisbusy:4] Playback("SIP/300-00000192", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
[2015-03-27 18:35:19] WARNING[350][C-000000f3]: file.c:701 ast_openstream_full: File all-circuits-busy-now does not exist in any format
[2015-03-27 18:35:19] WARNING[350][C-000000f3]: file.c:1017 ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)): No such file or directory
[2015-03-27 18:35:19] WARNING[350][C-000000f3]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/300-00000192 for all-circuits-busy-now&pls-try-call-later, noanswer
[2015-03-27 18:35:19] WARNING[350][C-000000f3]: file.c:701 ast_openstream_full: File pls-try-call-later does not exist in any format
[2015-03-27 18:35:19] WARNING[350][C-000000f3]: file.c:1017 ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No such file or directory
[2015-03-27 18:35:19] WARNING[350][C-000000f3]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/300-00000192 for all-circuits-busy-now&pls-try-call-later, noanswer
  -- Executing [s@macro-outisbusy:5] Congestion("SIP/300-00000192", "20") in new stack
[2015-03-27 18:35:19] WARNING[350][C-000000f3]: channel.c:4862 ast_prod: Prodding channel 'SIP/300-00000192' failed
 == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/300-00000192' in macro 'outisbusy'
 == Spawn extension (from-internal, 0176XXXXXX, 7) exited non-zero on 'SIP/300-00000192'
  -- Executing [h@from-internal:1] Hangup("SIP/300-00000192", "") in new stack
 == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/300-00000192'
 == MixMonitor close filestream (mixed)
 == End MixMonitor Recording SIP/300-00000192
[2015-03-27 18:35:28] WARNING[18275]: chan_sip.c:23527 handle_response_register: Got 423 Interval too brief for service fdmaroc2@sip.serveurcom.com (fdmaroc2@sip.serveurcom.com), minimum is 480 seconds
thanks nd regards
2015-03-27 17:08 GMT+00:00 Gareth Blades <mailinglist+asterisk@dns99.co.uk ([email]mailinglist+asterisk@dns99.co.uk[/email])>:
Quote: | You would need to give more information really.
Your sip.conf file listing the entries for the phones especially which codecs are permitted.
A copy of the 'asterisk -rvvv' console output when you make the call.
On 27/03/15 17:05, Salaheddine Elharit wrote:
Quote: | please no body has som with aastra can help me in this issue
2015-03-26 11:02 GMT+00:00 Salaheddine Elharit <salah.elharit200@gmail.com (salah.elharit200@gmail.com)>:
Quote: | hello list
i need your help please regarding an issue with snom300 and aastra6731i using asteriskÂ
11.13.0 Â asterisk
snom 300 Â 8.7.3.25
astra 6731i 2.6.0.2019
i have configured the trunks like below
100 in snom300
200 in snom300
300 in aastra6731i
400 in x-lite
the calls between x-lite and aastra ====ok inbound and outbound
the calls between x-lite and snom300====> ok inbound and outbound
the issue just between snom and aastra i can call from aastra to snom without issue
but when itry to call from snom300 to aastra6731i  i get bad request all the time
i test with 3 snom300 i get the same result
please any body have the snom and aastra can help me in order to fixe this issue
thanks and regards.
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--
_____________________________________________________________________
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mailinglist+asterisk a... Guest
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Posted: Fri Mar 27, 2015 1:04 pm Post subject: [asterisk-users] call between snom 300 and aastra 6731i |
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That is your issue.
You can enable a 'sip debug' and make the call again and get a trace of
the SIP message asterisk is sending to the phone.
We can take a look here to see if anything looks wrong.
If you could post a trace from a phone that can call that destination it
might be easier to spot the differences.
I am off work for the weekend now but someone else might be able to take
a look in the next couple of days.
On 27/03/15 17:44, Salaheddine Elharit wrote:
Quote: | -- Called SIP/FD/00XX17621XXXX
-- Got SIP response 480 "No address found" back from
217.195.XX.XXX:5060
|
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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