[asterisk-users] detect if call to device is from queue
hi,
do you have idea if is possible detect if a call to device(1) is from
queue? (i.e. if app_queue set some variable)
exten => 800,1,queue(sales) ; queue pick exten 20
exten => 20,1,noop("detect variables")
exten => 20,n,Dial(SIP/20)
(1) its through a local interface i.e Local/20@phones
tnx
Marek
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[asterisk-users] asterisk libsrtp 2.x status
hi, what's your experience with asterisk compiled with libsrtp 2.x and WebRTC(pjsip)? issues/crashes/speed/cpu usage? Marek official status https://wiki.asterisk.org/wiki/display/AST/libsrtp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pjsip aor stays in status created
hi, i have webrtc client chrome69/jssip which is connecting to asterisk 13.23.1/pjsip i have strange problem where pjsip aor stays in status "created" sip trace on asterisk looks ok. do you think if this can be bug? test*CLI> pjsip show aors Aor: Contact: == Aor: vr1k50 1 Contact: vr1k50/sip:6i2b9766@1.1.1.1:34434;tran b2ad914030 Created 0.000 <--- Received SIP request (566 bytes) from WSS:1.1.1.1:34434 ---> REGISTER sip:sip.example.com SIP/2.0 Via: SIP/2.0/WSS v0i0at11ojbn.invalid;branch=z9hG4bK2155317 Max-Forwards: 69 To: From: "vr1k50" ;tag=d56ij3vuo3 Call-ID: 0mm678kf72bc9b5ur7ea8d CSeq: 13 REGISTER Contact: ;+sip.ice;reg-id=1;+sip.instance="";expires=60 Expires: 60 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO Supported: path,gruu,outbound User-Agent: JsSIP 3.2.9 Content-Length: 0 <--- Transmitting SIP response (484 bytes) to WSS:1.1.1.1:34434 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/WSS v0i0at11ojbn.invalid;rport=34434;received=1.1.1.1;branch=z9hG4bK2155317 Call-ID: 0mm678kf72bc9b5ur7ea8d From: "vr1k50" ;tag=d56ij3vuo3 To: ;tag=z9hG4bK2155317 CSeq: 13 REGISTER WWW-Authenticate: Digest realm="asterisk",nonce="1540467808/121f72ae15612cc46a72e2861657a940",opaque="3060464337b28725",algorithm=md5,qop="auth" Server: Asterisk PBX 13.23.1 Content-Length: 0 <--- Received SIP request (837 bytes) from WSS:1.1.1.1:34434 ---> REGISTER sip:sip.example.com SIP/2.0 Via: SIP/2.0/WSS v0i0at11ojbn.invalid;branch=z9hG4bK9799804 Max-Forwards: 69 To: From: "vr1k50" ;tag=d56ij3vuo3 Call-ID: 0mm678kf72bc9b5ur7ea8d CSeq: 14 REGISTER Authorization: Digest algorithm=MD5, username="vr1k50", realm="asterisk", nonce="1540467808/121f72ae15612cc46a72e2861657a940", uri="sip:sip.example.com", response="376b4ac58b01dde2e043931467bba55a", opaque="3060464337b28725", qop=auth, cnonce="v8i7444gio8r", nc=0001 Contact: ;+sip.ice;reg-id=1;+sip.instance="";expires=60 Expires: 60 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO Supported: path,gruu,outbound User-Agent: JsSIP 3.2.9 Content-Length: 0 <--- Transmitting SIP response (446 bytes) to WSS:1.1.1.1:34434 ---> SIP/2.0 200 OK Via: SIP/2.0/WSS v0i0at11ojbn.invalid;rport=34434;received=1.1.1.1;branch=z9hG4bK9799804 Call-ID: 0mm678kf72bc9b5ur7ea8d From: "vr1k50" ;tag=d56ij3vuo3 To: ;tag=z9hG4bK9799804 CSeq: 14 REGISTER Date: Thu, 25 Oct 2018 11:43:28 GMT Contact: ;expires=59 Expires: 60 Server: Asterisk PBX 13.23.1 Content-Length: 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_pjsip_transport_management.c: Shutting down transport
hello,
i met with this interesting situation
[Jan 24 13:48:37] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport '8' since no request was received in 32 seconds
[Jan 24 13:48:37] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport '8' since no request was received in 32 seconds
[Jan 24 13:48:41] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport '' since no request was received in 32 seconds
[Jan 24 13:48:44] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport
'e="";expires=60
u▒l^' since no request was received in 32 seconds
[Jan 24 13:48:44] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport '' since no request was received in 32 seconds
[Jan 24 13:48:45] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport '' since no request was received in 32 seconds
[Jan 24 13:48:47] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport
'e="";expires=60
' since no request was received in 32 seconds
[Jan 24 13:48:49] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport '' since no request was received in 32 seconds
[Jan 24 13:48:50] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport '.0
Date: Wed, 24 Jan 2018 12:48:18 GMT
Allow: INVITE, ACK, CAN' since no request was received in 32 seconds
[Jan 24 13:48:53] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport '' since no request was received in 32 seconds
[Jan 24 13:48:53] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport ' SUBSCRIBE, INFO' since no request was received
in 32 seconds
[Jan 24 13:48:53] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport 'c732305e-f905-489a-a6f4-5164f0809c8a>";expires=60
Expires: 60
@u▒^' since no request was received in 32 seconds
[Jan 24 13:49:27] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport
'▒▒<%▒▒*W▒▒▒$@▒▒▒{▒X_DL▒▒▒1▒▒"▒`$▒zC▒l▒o▒O▒3▒▒c:133 idle_sched_cb:
Shutting down transport
'="";expires=60
asterisk went crazy and had to be restarted
topology
asterisk 13.18.2 + pjsip realtime + mariadb (mariadb is on different
network!) + jssip via wss as client
extconfig.conf
ps_endpoints => odbc,configDb
ps_auths => odbc,configDb
ps_aors => odbc,configDb
ps_domain_aliases => odbc,configDb
sorcery.conf
[res_pjsip] ; Realtime PJSIP configuration wizard
endpoint/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes
endpoint=realtime,ps_endpoints
auth/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes
auth=realtime,ps_auths
aor/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes
aor=realtime,ps_aors
domain_alias=realtime,ps_domain_aliases
there was net interruption on ~13:48
do you have any ideas what can be cause of
"res_pjsip_transport_management.c: Shutting down transport" ?
my idea was that Asterisk with cache doesnt need realtime connectivity
with mariadb (can survive short internet interruptions)
Marek
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Re: [asterisk-users] asterisk pjsip as voip client with multiple registrations
Dne 26/09/2017 v 22:33 Joshua Colp napsal(a): On Tue, Sep 26, 2017, at 05:29 PM, marek cervenka wrote: hi, i want use asterisk+pjsip as voip client with multiple registrations (perf testing) i'm using this example configuration for one account [308] type=registration outbound_auth=308 server_uri=sip:3...@example.com:5060 client_uri=sip:3...@example.com:5060 [308](auth-userpass) username=308 password=pass [308](aor-single-reg) contact=sip:example.com:5060 [308](endpoint-basic) outbound_auth=308 aors=308 [308] type=identify endpoint=308 match=example.com my problem is contact on the other side (is same for all endpoints) Addr->IP : 1.1.1.1:5060 Reg. Contact : sip:s@1.1.1.1:5060 all incoming calls from PBX to my Asterisk are routed to only one account (because of same ip address/port ?) how can i specify different source port or different contact address for asterisk pjsip client with registration? The "contact_user" option configures the user portion of the Contact that is sent in the REGISTER. You can set it to a different value for each registration. ok i have this configuration now client - asterisk+pjsip (public ip 1.1.1.1) pjsip/307 pjsip/308 server - asterisk+chan_sip (public ip 2.2.2.2) sip/307 Addr->IP : 1.1.1.1:5060 Reg. Contact : sip:307@1.1.1.1:5060 sip/308 Addr->IP : 1.1.1.1:5060 Reg. Contact : sip:308@1.1.1.1:5060 now, every call from server to client is received through pjsip/307 . but i need receive call for pjsip/308 through registration of pjsip/308. is it possible? is it possible configure different source port other than 5060? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk pjsip as voip client with multiple registrations
hi, i want use asterisk+pjsip as voip client with multiple registrations (perf testing) i'm using this example configuration for one account [308] type=registration outbound_auth=308 server_uri=sip:3...@example.com:5060 client_uri=sip:3...@example.com:5060 [308](auth-userpass) username=308 password=pass [308](aor-single-reg) contact=sip:example.com:5060 [308](endpoint-basic) outbound_auth=308 aors=308 [308] type=identify endpoint=308 match=example.com my problem is contact on the other side (is same for all endpoints) Addr->IP : 1.1.1.1:5060 Reg. Contact : sip:s@1.1.1.1:5060 all incoming calls from PBX to my Asterisk are routed to only one account (because of same ip address/port ?) how can i specify different source port or different contact address for asterisk pjsip client with registration? Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load testing with media in batch mode
my perftest suite call generator sipp, but creating sipp scenario is not easy. i'm using more user friendly(but its for win) - http://startrinity.com (REST API available) device emulation using asterisk as SIP client in docker - 30 SIP endpoints per instance reports pbx cpu/load/.. - grafana call details - export from startrinity my plan for automation is node.js app connected via ARI to Asterisk and via REST API to startrinity. endpoints configuration via ARI Push to PJSIP realtime Dne 20/09/2017 v 13:49 Olivier napsal(a): Hello, I am currently tasked on how to load test both signal and media from a couple of Asterisk machines which are doing corporate SIP trunking (no phone endpoint). If that matters, ecah machine will host debian Stretch, Asterisk 13 with either classic SIP or PJSIP. For instance, I can generate from a given source machine to a destination machine, 1000 calls passing an Asterisk instance under test. This under test Asterisk instance generate CDRs in which a testing program can seach for successful calls (reading disposition and/or RTCP stats in userfield). Most probably, this under test Asterisk instance will also log SIP capture to a remote Homer server, using Capagent and Homer itself. A testing program can also search this Homer/Capture database to evaluate "testing exit code". My question are: 1. Which (preferably available on Debian) tool(s) would you use to assert a single captured call, recorded on purpose by the system under test, has met call quality requirements ? (This one-call tool is needed to calibrate next tool). 2. Which tool(s) would you use to do the same, on whole testing campaign generating 1000 or 2000 simultaneous calls ? 3. How can you automate such tests ? Cheers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] current cpu recommendation for asterisk 13 + app_queue
hi, i know about architecture limits of app_queue https://issues.asterisk.org/jira/browse/ASTERISK-25806 what CPUs are you actually using for asterisk + app_queue ? (my actual scenario 90simult calls, 50agents, call recording to SSD (mixmonitor), no transcoding, CDR/CEL via odbc to MariaDB) customer offers Intel Xeon E5-2680v3 - 2,5GHz@9,6GT 30MB cache, 12core,HT, 120W,LGA2011 i think for app_queue will be better Intel Xeon E5-2637v3 - 3,5GHz@9,6GT 15MB cache, 4core,HT, 135W,LGA2011,tray thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SayUnixTime plays nothing if say.conf mode=new and a format is specified
hi, is there somebody who is using say.conf mode=new in Asterisk 13? i'm searching for tips what to try in https://issues.asterisk.org/jira/browse/ASTERISK-15421 Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk ari dialer
i solved problem of missing incoming channel using local channel
curl -X POST
"http://my_pbx:8088/ari/channels?endpoint=Local%2F300%40originate=555666777=originate=1=7=30_key=apikey;
(%2F is /, %40 is @)
extensions.conf
[originate]
exten => 300,1,noop(originate)
same => n,answer
same => n,MusicOnHold(10)
exten => _X.,1,noop(stasis)
same => n,Stasis(originate-example)
same => n,Hangup()
my actual problem is, howto call specific number in stasis application?
e.g. 12345678
var ENDPOINT ='PJSIP/my_sip_trunk';
return outgoing.originate({
endpoint:ENDPOINT,
app:'originate-example',
appArgs:'dialed',
callerId:'7' });
can i specify it in endpoint somehow?
Dne 30/06/2017 v 10:45 marek cervenka napsal(a):
my use case is for performace testing
scenario
asterisk14 - sip - tested asterisk - sip - clients (asterisk 14)
i have working ari push configuration
now i want create a call where call leg A will be some media file.
call leg B will be channel to tested asterisk
i dont have an incoming call e.g. for this example
https://github.com/asterisk/node-ari-client/blob/master/examples/promises/originate.js
Dne 29/06/2017 v 13:38 marek cervenka napsal(a):
hi,
do you have someone example of
http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/
in node.js asterisk-ari ?
thanks
Marek
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Re: [asterisk-users] asterisk ari dialer
my use case is for performace testing scenario asterisk14 - sip - tested asterisk - sip - clients (asterisk 14) i have working ari push configuration now i want create a call where call leg A will be some media file. call leg B will be channel to tested asterisk i dont have an incoming call e.g. for this example https://github.com/asterisk/node-ari-client/blob/master/examples/promises/originate.js Dne 29/06/2017 v 13:38 marek cervenka napsal(a): hi, do you have someone example of http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/ in node.js asterisk-ari ? thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk ari dialer
hi, do you have someone example of http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/ in node.js asterisk-ari ? thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pjsip configuration realtime+static
hi, i have mix of realtime and static configuration of pjsip https://pastebin.com/YVFwVsMD pjsip.conf [global] endpoint_identifier_order=username,ip,anonymous user_agent=ipbx ... transport definition extconfig.conf [settings] ps_endpoints => odbc,configDb ps_auths => odbc,configDb ps_domain_aliases => odbc,configDb ps_aors => odbc,configDb sorcery.conf [res_pjsip] endpoint/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes endpoint=realtime,ps_endpoints auth/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes auth=realtime,ps_auths aor/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes aor=realtime,ps_aors domain_alias=realtime,ps_domain_aliases but the section [global] is not honored CLI> pjsip show settings Global Settings: ParameterName : ParameterValue === contact_expiration_check_interval : 30 debug : no default_from_user : asterisk default_outbound_endpoint : default_outbound_endpoint default_realm : asterisk default_voicemail_extension : disable_multi_domain: false endpoint_identifier_order : ip,username,anonymous --snip-- user_agent : Asterisk PBX 13.16.0 any ideas? Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call hangup after leaving app_queue
can you someone confirm https://issues.asterisk.org/jira/browse/ASTERISK-27065 its easy to replicate Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hangup handlers & unwanted cdr
hi, i'm using hangup handlers on Asterisk13 with standard answered calls i have 1 CDR per call with scenario call from voip->mobile, call rejected on mobile i have 2 CDRs i dont want the second CDR without hangup handlers i have 1 CDR do you think its bug or its feature of hangup handlers? *** 1. row *** calldate: 2017-05-31 15:50:28 clid: "voip_number" src: voip_number dst: mobile_number dcontext: route_phones_1 channel: SIP/vr1a915-001e dstchannel: SIP/siptrunk-001f channtype: lastapp: Dial lastdata: SIP/siptrunk/mobile_number,120,tTb(pre_dial_handler^callee^1)B(pre_dial_handler^cal duration: 10 billsec: 0 disposition: NO ANSWER amaflags: 3 accountcode: uniqueid: 1496238628.30 hangupcause: stamp: 2017-05-31 15:50:38 linkedid: 1496238628.30 sequence: 30 *** 2. row *** calldate: 2017-05-31 15:50:28 clid: "mobile_number" src: mobile_number dst: mobile_number dcontext: trunk_context_1 channel: SIP/siptrunk-001f dstchannel: channtype: lastapp: Return lastdata: duration: 9 billsec: 0 disposition: FAILED amaflags: 3 accountcode: uniqueid: 1496238628.31 hangupcause: stamp: 2017-05-31 15:50:38 linkedid: 1496238628.30 sequence: 31 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 13.15.0 stopping/crashing
"worker_start"
#16 0x005f03fe in dummy_start (data=) at utils.c:1235
__clframe = {__cancel_routine = , __cancel_arg =
0x7f19a9c25700, __do_it = 1, __cancel_type = }
ret =
a = {start_routine = 0x5e65e0 , data =
0x7f1a1800ae00, name = }
#17 0x7f1a1e89ddc5 in start_thread () from /lib64/libpthread.so.0
No symbol table info available.
#18 0x7f1a1db7d73d in clone () from /lib64/libc.so.6
No symbol table info available.
Dne 09/05/2017 v 14:57 marek cervenka napsal(a):
when run from console without systemd i found its segfaulting
turned core dump on because it was off
Dne 09/05/2017 v 13:52 marek cervenka napsal(a):
hi,
i have strange problem with asterisk 13.15.0+pjsip bundled/centos
7/systemd start script
we are using chan_pjsip only for webrtc endpoints . switched from
sipml5 to jssip with upgrade to 13.15.0(from 13.9.0) few days ago
today i have problems with stopping/crashing asterisk
/var/log/asterisk/messages dont show any clues
[May 9 12:10:52] WARNING[25762] pjproject: tsx0x7fbb28024088 ..Error
sending Response msg 200/REGISTER/cseq=4 (tdta0x7fbb080d29e8):
Unknown Error (PJ_EUNKNOWN)
[May 9 12:10:54] WARNING[6458] pjproject: tsx0x7fbb28024088 ..Error
sending Response msg 200/REGISTER/cseq=4 (tdta0x7fbb382be318):
Unknown Error (PJ_EUNKNOWN)
[May 9 12:10:54] WARNING[20014] pjproject: tsx0x7fbb2c4a93e8 ..Error
sending Response msg 200/REGISTER/cseq=4 (tdta0x1e44748): Unknown
Error (PJ_EUNKNOWN)
[May 9 12:10:59] WARNING[6458] pjproject: tsx0x7fbb2c4a93e8 ..Error
sending Response msg 200/REGISTER/cseq=4 (tdta0x7fbb1cc358a8):
Unknown Error (PJ_EUNKNOWN)
[May 9 12:11:00] NOTICE[19165] chan_sip.c: Received SIP subscribe
for peer without mailbox: vr1a86
[May 9 12:15:27] Asterisk 13.15.0 built by root @ 45ba17aca47d on a
x86_64 running Linux on 2017-04-10 12:10:44 UTC
--
[May 9 12:16:41] WARNING[30730] pjproject: tsx0x7f1a9c027ae8 ..Error
sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1a982abe68):
Unknown Error (PJ_EUNKNOWN)
[May 9 12:16:41] NOTICE[30762] chan_sip.c: Received SIP subscribe
for peer without mailbox: vr1a99
[May 9 12:16:42] NOTICE[30762] chan_sip.c: Received SIP subscribe
for peer without mailbox: vr1a86
[May 9 12:16:42] NOTICE[30762] chan_sip.c: Received SIP subscribe
for peer without mailbox: vr1a99
[May 9 12:16:43] WARNING[30726] pjproject: tsx0x7f1a9401b4f8 ..Error
sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1a7402cc58):
Unknown Error (PJ_EUNKNOWN)
[May 9 12:17:33] Asterisk 13.15.0 built by root @ 45ba17aca47d on a
x86_64 running Linux on 2017-04-10 12:10:44 UTC
--
[May 9 12:55:00] WARNING[31091] pjproject: tsx0x7f1b08036368 ..Error
sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1afc024278):
Unknown Error (PJ_EUNKNOWN)
[May 9 12:55:06] NOTICE[31144] chan_sip.c: Received SIP subscribe
for peer without mailbox: vr1a86
[May 9 12:55:07] NOTICE[31144] chan_sip.c: Received SIP subscribe
for peer without mailbox: vr1a86
[May 9 12:55:09] WARNING[2964] pjproject: tsx0x7f1afc017c18 ..Error
sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1afc0370e8):
Unknown Error (PJ_EUNKNOWN)
[May 9 12:55:09] WARNING[31089] pjproject: tsx0x7f1afc017c18 ..Error
sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1ae40193f8):
Unknown Error (PJ_EUNKNOWN)
[May 9 12:57:16] Asterisk 13.15.0 built by root @ 45ba17aca47d on a
x86_64 running Linux on 2017-04-10 12:10:44 UTC
--
[May 9 12:58:25] NOTICE[6235] chan_sip.c: Received SIP subscribe for
peer without mailbox: vr1a99
[May 9 12:58:26] NOTICE[6235] chan_sip.c: Received SIP subscribe for
peer without mailbox: vr1a99
[May 9 12:58:34] WARNING[6190] pjproject: tsx0x7f89f401b398 ..Error
sending Response msg 200/REGISTER/cseq=4 (tdta0x7f89c80046b8):
Unknown Error (PJ_EUNKNOWN)
[May 9 12:58:48] Asterisk 13.15.0 built by root @ 45ba17aca47d on a
x86_64 running Linux on 2017-04-10 12:10:44 UTC
any tips? known issues?
thanks
Marek
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Re: [asterisk-users] asterisk 13.15.0 stopping/crashing
when run from console without systemd i found its segfaulting turned core dump on because it was off Dne 09/05/2017 v 13:52 marek cervenka napsal(a): hi, i have strange problem with asterisk 13.15.0+pjsip bundled/centos 7/systemd start script we are using chan_pjsip only for webrtc endpoints . switched from sipml5 to jssip with upgrade to 13.15.0(from 13.9.0) few days ago today i have problems with stopping/crashing asterisk /var/log/asterisk/messages dont show any clues [May 9 12:10:52] WARNING[25762] pjproject: tsx0x7fbb28024088 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7fbb080d29e8): Unknown Error (PJ_EUNKNOWN) [May 9 12:10:54] WARNING[6458] pjproject: tsx0x7fbb28024088 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7fbb382be318): Unknown Error (PJ_EUNKNOWN) [May 9 12:10:54] WARNING[20014] pjproject: tsx0x7fbb2c4a93e8 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x1e44748): Unknown Error (PJ_EUNKNOWN) [May 9 12:10:59] WARNING[6458] pjproject: tsx0x7fbb2c4a93e8 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7fbb1cc358a8): Unknown Error (PJ_EUNKNOWN) [May 9 12:11:00] NOTICE[19165] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a86 [May 9 12:15:27] Asterisk 13.15.0 built by root @ 45ba17aca47d on a x86_64 running Linux on 2017-04-10 12:10:44 UTC -- [May 9 12:16:41] WARNING[30730] pjproject: tsx0x7f1a9c027ae8 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1a982abe68): Unknown Error (PJ_EUNKNOWN) [May 9 12:16:41] NOTICE[30762] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a99 [May 9 12:16:42] NOTICE[30762] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a86 [May 9 12:16:42] NOTICE[30762] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a99 [May 9 12:16:43] WARNING[30726] pjproject: tsx0x7f1a9401b4f8 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1a7402cc58): Unknown Error (PJ_EUNKNOWN) [May 9 12:17:33] Asterisk 13.15.0 built by root @ 45ba17aca47d on a x86_64 running Linux on 2017-04-10 12:10:44 UTC -- [May 9 12:55:00] WARNING[31091] pjproject: tsx0x7f1b08036368 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1afc024278): Unknown Error (PJ_EUNKNOWN) [May 9 12:55:06] NOTICE[31144] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a86 [May 9 12:55:07] NOTICE[31144] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a86 [May 9 12:55:09] WARNING[2964] pjproject: tsx0x7f1afc017c18 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1afc0370e8): Unknown Error (PJ_EUNKNOWN) [May 9 12:55:09] WARNING[31089] pjproject: tsx0x7f1afc017c18 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1ae40193f8): Unknown Error (PJ_EUNKNOWN) [May 9 12:57:16] Asterisk 13.15.0 built by root @ 45ba17aca47d on a x86_64 running Linux on 2017-04-10 12:10:44 UTC -- [May 9 12:58:25] NOTICE[6235] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a99 [May 9 12:58:26] NOTICE[6235] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a99 [May 9 12:58:34] WARNING[6190] pjproject: tsx0x7f89f401b398 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7f89c80046b8): Unknown Error (PJ_EUNKNOWN) [May 9 12:58:48] Asterisk 13.15.0 built by root @ 45ba17aca47d on a x86_64 running Linux on 2017-04-10 12:10:44 UTC any tips? known issues? thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 13.15.0 stopping/crashing
hi, i have strange problem with asterisk 13.15.0+pjsip bundled/centos 7/systemd start script we are using chan_pjsip only for webrtc endpoints . switched from sipml5 to jssip with upgrade to 13.15.0(from 13.9.0) few days ago today i have problems with stopping/crashing asterisk /var/log/asterisk/messages dont show any clues [May 9 12:10:52] WARNING[25762] pjproject: tsx0x7fbb28024088 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7fbb080d29e8): Unknown Error (PJ_EUNKNOWN) [May 9 12:10:54] WARNING[6458] pjproject: tsx0x7fbb28024088 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7fbb382be318): Unknown Error (PJ_EUNKNOWN) [May 9 12:10:54] WARNING[20014] pjproject: tsx0x7fbb2c4a93e8 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x1e44748): Unknown Error (PJ_EUNKNOWN) [May 9 12:10:59] WARNING[6458] pjproject: tsx0x7fbb2c4a93e8 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7fbb1cc358a8): Unknown Error (PJ_EUNKNOWN) [May 9 12:11:00] NOTICE[19165] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a86 [May 9 12:15:27] Asterisk 13.15.0 built by root @ 45ba17aca47d on a x86_64 running Linux on 2017-04-10 12:10:44 UTC -- [May 9 12:16:41] WARNING[30730] pjproject: tsx0x7f1a9c027ae8 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1a982abe68): Unknown Error (PJ_EUNKNOWN) [May 9 12:16:41] NOTICE[30762] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a99 [May 9 12:16:42] NOTICE[30762] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a86 [May 9 12:16:42] NOTICE[30762] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a99 [May 9 12:16:43] WARNING[30726] pjproject: tsx0x7f1a9401b4f8 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1a7402cc58): Unknown Error (PJ_EUNKNOWN) [May 9 12:17:33] Asterisk 13.15.0 built by root @ 45ba17aca47d on a x86_64 running Linux on 2017-04-10 12:10:44 UTC -- [May 9 12:55:00] WARNING[31091] pjproject: tsx0x7f1b08036368 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1afc024278): Unknown Error (PJ_EUNKNOWN) [May 9 12:55:06] NOTICE[31144] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a86 [May 9 12:55:07] NOTICE[31144] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a86 [May 9 12:55:09] WARNING[2964] pjproject: tsx0x7f1afc017c18 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1afc0370e8): Unknown Error (PJ_EUNKNOWN) [May 9 12:55:09] WARNING[31089] pjproject: tsx0x7f1afc017c18 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7f1ae40193f8): Unknown Error (PJ_EUNKNOWN) [May 9 12:57:16] Asterisk 13.15.0 built by root @ 45ba17aca47d on a x86_64 running Linux on 2017-04-10 12:10:44 UTC -- [May 9 12:58:25] NOTICE[6235] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a99 [May 9 12:58:26] NOTICE[6235] chan_sip.c: Received SIP subscribe for peer without mailbox: vr1a99 [May 9 12:58:34] WARNING[6190] pjproject: tsx0x7f89f401b398 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7f89c80046b8): Unknown Error (PJ_EUNKNOWN) [May 9 12:58:48] Asterisk 13.15.0 built by root @ 45ba17aca47d on a x86_64 running Linux on 2017-04-10 12:10:44 UTC any tips? known issues? thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] best kernel for Asterisk
hi, what kernel version are you using for asterisk? are you satisfied with distro kernel (centos 6 2.6.32, centos 7 3.10, ...) ? are you using newer kernels from elrepo.org? which kernel features are most critical for Asterisk performance pattern? thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk13+app_queue scalability
hi, i have similar problem to https://issues.asterisk.org/jira/browse/ASTERISK-25806 do you know about some workarounds/patches for better scalability? thanks marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no rtp after dns query (SOLVED)
thanks for confirmation dns name in /etc/hosts & dnsmgr enabled solved my problem Dne 14/12/2016 v 13:50 Joshua Colp napsal(a): On Wed, Dec 14, 2016, at 08:47 AM, marek cervenka wrote: i found this issue https://issues.asterisk.org/jira/browse/ASTERISK-26280 but its not clear if this problem can be in chan_sip/udp created channels & pjsip module is active only for wss transport This was in RTP, so it was applicable to every channel driver that uses RTP including chan_sip. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_queue missed calls per agent - caller hangup before timeout
hi,
i'm trying get report about missed calls per agent. i'm using queue_log
and RINGNOANSWER event
but i found problem described here
---
https://www.thirdlane.com/forum/queue-log-problem
RINGNOANSWER only happens if the call TIMES OUT ringing the agent and it
returns to the queue. If your agent has a 30 second timeout and the
caller ABANDONS the call in 5 seconds it will log an ABANDON not a
RINGNOANSWER.
This is the only time ast_queue_log is executed with RINGNOANSWER. The
subsequent code of this function goes on to autopause the agent/member
if autopause is enabled. Not something that happens when callers hang up
when ringing the agents.
/*! \brief RNA == Ring No Answer. Common code that is executed when we
try a queue member and they don't answer. */
static void rna(int rnatime, struct queue_ent *qe, char *interface, char
*membername)
{
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "Nobody picked up in %d ms\n", rnatime);
ast_queue_log(qe->parent->name, qe->chan->uniqueid, membername,
"RINGNOANSWER", "%d", rnatime);
---
any tips howto detect missed calls where caller hangup before timeout?
tnx
Marek
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Re: [asterisk-users] no rtp after dns query
i found this issue https://issues.asterisk.org/jira/browse/ASTERISK-26280 but its not clear if this problem can be in chan_sip/udp created channels & pjsip module is active only for wss transport Dne 14/12/2016 v 12:14 marek cervenka napsal(a): hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4468, Time=716240 1172 25.045629000 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60990, Time=716240 1173 25.048134000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=274, Time=1442112421 1174 25.048207000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49319, Time=1442112416 1175 25.065362000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4469, Time=716400 1176 25.065441000 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60991, Time=716400 1177 25.068138000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=275, Time=1442112581 1178 25.068214000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49320, Time=1442112576 1179 25.085427000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4470, Time=716560 1180 25.08550 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60992, Time=716560 1181 25.088133000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=276, Time=1442112741 1182 25.088207000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49321, Time=1442112736 1183 25.099395000 172.23.5.2 -> 172.23.0.3 RTCP 94 Sender Report 1184 25.099569000 172.23.0.3 -> 172.16.1.20 DNS 78 Standard query 0xd853 A pbx.somewhere.com 1185 25.099591000 172.23.0.3 -> 172.16.1.20 DNS 78 Standard query 0xeb9f pbx.somewhere.com 1186 25.100211000 172.16.1.20 -> 172.23.0.3 DNS 94 Standard query response 0xd853 A 172.23.0.3 1187 25.105456000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4471, Time=716720 1188 25.108153000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=277, Time=1442112901 1189 25.125115000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4472, Time=716880 1190 25.128169000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=278, Time=1442113061 1191 25.145232000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4473, Time=717040 1192 25.148169000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=279, Time=1442113221 1193 25.165214000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4474, Time=717200 1194 25.168169000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=280, Time=1442113381 1195 25.185212000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4475, Time=717360 1196 25.188194000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=281, Time=1442113541 1197 25.205216000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4476, Time=717520 1198 25.208164000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=282, Time=1442113701 1199 25.225149000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4477, Time=717680 1200 25.228177000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=283, Time=1442113861 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no rtp after dns query
hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4468, Time=716240 1172 25.045629000 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60990, Time=716240 1173 25.048134000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=274, Time=1442112421 1174 25.048207000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49319, Time=1442112416 1175 25.065362000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4469, Time=716400 1176 25.065441000 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60991, Time=716400 1177 25.068138000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=275, Time=1442112581 1178 25.068214000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49320, Time=1442112576 1179 25.085427000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4470, Time=716560 1180 25.08550 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60992, Time=716560 1181 25.088133000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=276, Time=1442112741 1182 25.088207000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49321, Time=1442112736 1183 25.099395000 172.23.5.2 -> 172.23.0.3 RTCP 94 Sender Report 1184 25.099569000 172.23.0.3 -> 172.16.1.20 DNS 78 Standard query 0xd853 A pbx.somewhere.com 1185 25.099591000 172.23.0.3 -> 172.16.1.20 DNS 78 Standard query 0xeb9f pbx.somewhere.com 1186 25.100211000 172.16.1.20 -> 172.23.0.3 DNS 94 Standard query response 0xd853 A 172.23.0.3 1187 25.105456000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4471, Time=716720 1188 25.108153000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=277, Time=1442112901 1189 25.125115000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4472, Time=716880 1190 25.128169000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=278, Time=1442113061 1191 25.145232000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4473, Time=717040 1192 25.148169000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=279, Time=1442113221 1193 25.165214000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4474, Time=717200 1194 25.168169000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=280, Time=1442113381 1195 25.185212000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4475, Time=717360 1196 25.188194000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=281, Time=1442113541 1197 25.205216000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4476, Time=717520 1198 25.208164000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=282, Time=1442113701 1199 25.225149000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4477, Time=717680 1200 25.228177000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=283, Time=1442113861 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_queue ringall - 2 agents answer same time problem
upgrade to ast 13.13.0 doesnt help switch from local channel to SIP help ;member => Local/2000@route_phones_1,1,2000,hint:2000@subscribe_1 member => SIP/vr1a2000 load average is around 2 (4 core, vmware with 1Ghz per core), generated by 2x yes > /dev/null & [route_phones_1] is around 10 dialplan commands (execif,set) + 1x fastAGI do you think it's bug or timing "limit" of Asterisk? Dne 30/11/2016 v 22:17 marek cervenka napsal(a): hmm. i think customer will not agree this is correct behavior from pcap it looks like there is missing CANCEL to the second device Dne 30/11/2016 v 19:42 Sam Basan napsal(a): Your second call is not without sound, there is simply no call at all. As the first answer the call his channel and the external call channel connected. The second device simply off hook but his channel have no external channel to connect. It's looks like a simple telephony glare. Sam בתאריך 30 בנוב' 2016 7:00 PM, "marek cervenka" <cerva...@gmail.com <mailto:cerva...@gmail.com>> כתב: hi, our customer reports problem when 2 agents answer the call in the same time faster operator (device) answer the call, but the second is showed up (on device) and call is without sound asterisk 13.9/app_queue with strategy ringall/operators via Local channel with sip device (chan_sip) do you have any tips/info before i will dig deep into logs/debug? checked google <http://issues.asterisk.org> without any clue marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ <https://community.asterisk.org/> New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started <https://wiki.asterisk.org/wiki/display/AST/Getting+Started> asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_queue ringall - 2 agents answer same time problem
hmm. i think customer will not agree this is correct behavior from pcap it looks like there is missing CANCEL to the second device Dne 30/11/2016 v 19:42 Sam Basan napsal(a): Your second call is not without sound, there is simply no call at all. As the first answer the call his channel and the external call channel connected. The second device simply off hook but his channel have no external channel to connect. It's looks like a simple telephony glare. Sam בתאריך 30 בנוב' 2016 7:00 PM, "marek cervenka" <cerva...@gmail.com <mailto:cerva...@gmail.com>> כתב: hi, our customer reports problem when 2 agents answer the call in the same time faster operator (device) answer the call, but the second is showed up (on device) and call is without sound asterisk 13.9/app_queue with strategy ringall/operators via Local channel with sip device (chan_sip) do you have any tips/info before i will dig deep into logs/debug? checked google <http://issues.asterisk.org> without any clue marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ <https://community.asterisk.org/> New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started <https://wiki.asterisk.org/wiki/display/AST/Getting+Started> asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_queue ringall - 2 agents answer same time problem
hi, our customer reports problem when 2 agents answer the call in the same time faster operator (device) answer the call, but the second is showed up (on device) and call is without sound asterisk 13.9/app_queue with strategy ringall/operators via Local channel with sip device (chan_sip) do you have any tips/info before i will dig deep into logs/debug? checked google without any clue marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: recommended helpdesk OSS with Asterisk integration
hi, can you recommend open source helpdesk solution with working Asterisk integration? marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue_log/cel sqlite
Dne 20/10/2016 v 16:32 Matt Fredrickson napsal(a):
On Thu, Oct 20, 2016 at 4:50 AM, marek cervenka <cerva...@gmail.com> wrote:
i tested this
# cat /etc/asterisk/extconfig.conf
[settings]
queue_log => sqlite3,cdrDb
# cat /etc/asterisk/res_config_sqlite3.conf
[cdrDb]
dbfile = /var/lib/asterisk/realtime.sqlite3
sqlite3 /var/lib/asterisk/realtime.sqlite3
CREATE TABLE "queue_log" ("time" TEXT, "data1" TEXT, "data2" TEXT, "data3"
TEXT, "data4" TEXT, "data5" TEXT, "event" TEXT, "agent" TEXT, "queuename"
TEXT, "callid" TEXT);
and it works
sqlite> select * from queue_log;
2016-10-20 11:40:36.628804||QUEUESTART|NONE|NONE|NONE
2016-10-20 11:40:36.690313||CONFIGRELOAD|NONE|NONE|NONE
column types needs modification to something more appropriate
can someone with confluence access ad info to
https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration ?
Which info are you referring to? The table schema?
ideally add "correct" sql schema for sqlite to asterisk repo and link it to
https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
it was hard for me to find if queue_log can be logged with sqlite. imho
it will be usefull document the example configuration for others
but i'm not sure where is the best place
maybe https://wiki.asterisk.org/wiki/display/AST/Queue+Logs ?
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Re: [asterisk-users] queue_log/cel sqlite
i tested this
# cat /etc/asterisk/extconfig.conf
[settings]
queue_log => sqlite3,cdrDb
# cat /etc/asterisk/res_config_sqlite3.conf
[cdrDb]
dbfile = /var/lib/asterisk/realtime.sqlite3
sqlite3 /var/lib/asterisk/realtime.sqlite3
CREATE TABLE "queue_log" ("time" TEXT, "data1" TEXT, "data2" TEXT,
"data3" TEXT, "data4" TEXT, "data5" TEXT, "event" TEXT, "agent" TEXT,
"queuename" TEXT, "callid" TEXT);
and it works
sqlite> select * from queue_log;
2016-10-20 11:40:36.628804||QUEUESTART|NONE|NONE|NONE
2016-10-20 11:40:36.690313||CONFIGRELOAD|NONE|NONE|NONE
column types needs modification to something more appropriate
can someone with confluence access ad info to
https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration ?
is there somebody using it in production?
thanks
Dne 20/10/2016 v 10:16 marek cervenka napsal(a):
hi,
is it possible log cel/queue_log to sqlite?
via odbc?
any experience?
marek
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[asterisk-users] queue_log/cel sqlite
hi, is it possible log cel/queue_log to sqlite? via odbc? any experience? marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CONNECTEDLINE endpoint support
hi, i'm testing CONNECTEDLINE function https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information example dialplan same => n,set(CONNECTEDLINE(name,i)=aastra) same => n,set(CONNECTEDLINE(name-pres,i)=allowed) same => n,Set(CONNECTEDLINE(num,i)=5551212) same => n,Set(CONNECTEDLINE(num-pres)=allowed) same => n,dial(SIP/sipline501,,I) it only works with mitel(aastra 6767i) phones i tested - grandstream 2130, jitsi, blink, microsip, zoiper - nothing worked what devices working for you with CONNECTEDLINE function? thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuration management and update deployment - what do you use?
ansible.com Dne 18/10/2016 v 11:46 Duncan napsal(a): Hi All We have about 15 different asterisk boxes around the place and on my list has been automate deployment updates and keep a revision history. They are mostly not publicly accessible, and external SIP access is closely firewalled , so updates happen straight away when its something like heartbleed, but take a while to trust/test new releases. Our boxes are Ubuntu LTS - mostly 14.04 at the moment. We use Freebpx as the configuration front end and so that tends to be a more manual update, although there is an API we could use to keep things in step. We run backups from freepbx and archive those as well as any specific asterisk settings missed. At the moment our scale means manual is okay, but automation would make it easier if the learning curve and new issues aren't too high. We compile asterisk from source as the packages aren't usually quite what we want. I was just curious how people deploy asterisk across multiple platforms and keep them all up to date? What tools are good for this sort of thing? Thanks very much Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk security framework
hi, i'm trying configure $subj https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger but there is a ton of "informational" messages [Sep 30 14:40:16] SECURITY[18311] res_security_log.c: SecurityEvent="SuccessfulAuth",EventTV="2016-09-30T14:40:16.833+0200",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="webrtc",SessionID="e74a4f71-4f0a-14ee-6373-a53d0540dd12",LocalAddress="IPV4/WSS/1.1.1.1/39512",RemoteAddress="IPV4/WSS/1.1.1.1/39512",UsingPassword="1" is there possibility log only "important" things? i.e. by severity or by category from https://wiki.asterisk.org/wiki/display/AST/Security+Events+to+Log thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 and WebRTC
using in production last asterisk 13 + pjsip bundled + pjsip patch for RTP/SAVPF (search pjsip conf) + sipml5 version from roginvs https://github.com/DoubangoTelecom/sipml5/pull/238 Dne 08/09/2016 v 23:36 Annus Fictus napsal(a): Hello list, before to lost my time, I'd like know if someone have a WebRTC working configuration on Asterisk 13.11.0 SIP or PJSIP channel. Thank you Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] switch from fastAGI to CURL
hi, i want switch my application server(dynamic routing) in node.js from fastAGI to CURL because of - easier development of REST API server - testing and debuging - AGI is not known in the web dev world what do you think about curl from performance view? (10cps, 500 simult calls per node) what do you think about using curl for PJSIP realtime vs ODBC? Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk queue - skills based routing (patch updated)
Le 2015-08-10 13:54, Marek Cervenka a écrit : Dne 6.8.2015 v 21:00 Sylvain Boily napsal(a): Hello, Le 2015-08-06 09:24, Marek Cervenka a écrit : hi, there is updated skills based routing patch for asterisk queue please test if you have time https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22 You can find the latest version we maintain here : https://github.com/xivo-pbx/asterisk/blob/master/debian/patches/xivo_skill_queues (asterisk 13.5) We originally wrote this patch for xivo and it's included by default. Sylvain that's great! do you have plan merge it to the asterisk master? At the astricondev 2012, there was a decision to not merged this patch on app_queue because nobody really wanted to add new features. So, no there is no plan to merge this patch on the master, but we maintain it on xivo with the latest asterisk version and if someone want to work with us and people would like this patch into the master, we will be enjoy to contribute. ---from the ticket--- Kevin Harwell https://issues.asterisk.org/jira/secure/ViewProfile.jspa?name=kharwell added a comment - 21/Jul/15 4:55 PM Supplying an updated patch and submitting it for review would certainly expedite the process. Please see https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process for more information on how to do that. - i think it's very important feature for call center. can you please try upload actual patch to the issue tracker? it's working perfectly thank you marek -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] webrtc no audio
hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) BUT call from person2(chrome) to person 3(Jitsi sip client) - works! any tips howto find the problem? -- --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] asterisk queue - skills based routing (patch updated)
Dne 6.8.2015 v 21:00 Sylvain Boily napsal(a): Hello, Le 2015-08-06 09:24, Marek Cervenka a écrit : hi, there is updated skills based routing patch for asterisk queue please test if you have time https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22 You can find the latest version we maintain here : https://github.com/xivo-pbx/asterisk/blob/master/debian/patches/xivo_skill_queues (asterisk 13.5) We originally wrote this patch for xivo and it's included by default. Sylvain that's great! do you have plan merge it to the asterisk master? -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] asterisk queue - skills based routing (patch updated)
hi, there is updated skills based routing patch for asterisk queue please test if you have time https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22 -- --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] Req help regarding webRTC : Attempted Attempted to set an invalid DTLS-SRTP configuration on RTP instance
paste sip.conf or pjsip.conf on pastebin and post link here Dne 16.6.2015 v 7:46 Kantharuban Ruban napsal(a): Hi List, I am trying to setup a Asterisk setup in AWS instance Centos6.5 . I have installed Asterisk 13.4 with srtp,pjproject. I have configured two numbers for webRTC clients, when i try to call from a client (sipml5) to another client (sipml5) it throws the following error: chan_sip.c:5851 dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP instance '0x7f3ccc020718' I am struck here. Please throw some light to go further. Thanks in advance. Best regards, Ruban.S -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] queue periodic-announce without stopping ringing
hello, is it possible to play queue periodic-announce without stopping agents ringing? actual situation is sequential - ring agents, play announce (for 15 sec), ring agents , ... (i need to connect agent with caller asap when agent is free) is it possible with ARI? -- --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] sslv3 alert unexpected message
Dne 3.6.2015 v 17:57 Marek Cervenka napsal(a): hello, my webrtc calls ends after ~60seconds with res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0xb6c02a94' due to reason 'sslv3 alert unexpected message', terminating. any ideas where can be problem? or howto debug this problem? asterisk13.4.0-rc1 + sipml5 latest (chrome,firefox) upgrade from libsrtp-1.4.4-4.20101004cvs.el6.i686 to libsrtp-1.5.0-2.el6.i686 - doesnt help it's on centos6 - openssl-1.0.1e-30.el6.8.i686 the problem is in dtlsrekey=60 if i change it to dtlsrekey=120 it hangs after 120seconds do you think it's a bug? do you recommend fill bug in issues.asterisk.org? -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] sslv3 alert unexpected message
hello, my webrtc calls ends after ~60seconds with res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0xb6c02a94' due to reason 'sslv3 alert unexpected message', terminating. any ideas where can be problem? or howto debug this problem? asterisk13.4.0-rc1 + sipml5 latest (chrome,firefox) -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] [SOLVED] Re: asterisk 13 webrtc
dtlsenable=yes was missing thank you joshua Dne 21.5.2015 v 22:53 Marek Cervenka napsal(a): hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia canreinvite=no encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=yes transport=wss,ws dtlsrekey=60 dtlsverify=no dtlscertfile=/etc/pki/tls/certs/rapidssl.crt dtlsprivatekey=/etc/pki/tls/private/rapidssl.key dtlssetup=actpass -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] asterisk 13 webrtc
[May 19 16:47:43] DEBUG[14160][C-0007]: chan_sip.c:10458
process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[May 19 16:47:43] DEBUG[14160][C-0007]: chan_sip.c:10458
process_sdp: Processing media-level (audio) SDP a=setup:actpass...
UNSUPPORTED OR FAILED.
[May 19 16:47:43] DEBUG[14160][C-0007]: chan_sip.c:10458
process_sdp: Processing media-level (audio) SDP a=ssrc:1181629171
cname:{dc854b06-da58-45b3-8185-bbc6a57746c0}...
UNSUPPORTED OR FAILED.
[May 19 16:47:43] WARNING[14160][C-0007]: chan_sip.c:10496
process_sdp: Can't provide secure audio requested in SDP offer
--- Reliably Transmitting (NAT) to 2.2.2.2:8558 ---
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKyhVsmJdAtwRvuOWz9BHiNo1DGcqp4grQ;received=2.2.2.2;rport=8558
From: cervenkasip:vr1a...@vhxxx.example.com;tag=RDmpGm2Mubc5xQQ8NMli
To: sip:887@ipbx;tag=as5d30f0ef
Call-ID: cf2990ba-3f12-3d9e-adb6-52889c414ed3
CSeq: 41942 INVITE
Server: ipbx 3.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
[May 19 16:47:43] DEBUG[14160][C-0007]: chan_sip.c:3696 __sip_xmit:
Trying to put 'SIP/2.0 488' onto WS socket destined for
2.2.2.2:8558
Scheduling destruction of SIP dialog
'cf2990ba-3f12-3d9e-adb6-52889c414ed3' in 32000 ms (Method: INVITE)
[May 19 16:47:43] DEBUG[14160][C-0007]: chan_sip.c:25557
handle_request_invite: No compatible codecs for this SIP call.
[May 19 16:47:43] DEBUG[14160][C-0007]: chan_sip.c:28297
handle_request_do: SIP message could not be handled, bad request:
cf2990ba-3f12-3d9e-adb6-52889c414ed3
--
---
Marek Cervenka
===
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[asterisk-users] Local channel + queue
hi, i'm facing problem with multiple calls to one agent when Local channels are used wireshark shows multiple invites to the agent's phone used versions asterisk 1.8/asterisk 13 agents are logged dynamically. interface state based on hints queue configuration ... ringinuse=no autofill = yes ... member = Local/99@route_phones_1,2,mila_jojovich,SIP/virtual_99 member = Local/88@route_phones_1,3,angelina_jolie,SIP/virtual_88 time between call to local channel and call to SIP device can be in seconds //Without local channel queue works good, but i need local channel for additional settings/actions i need working BLF (multiple states was problem), i need working transfers (cannot limit call to 1 via GROUP_COUNT) howto change state in queue immediately after calling local channel (similiary to after calling sip device) ? any tips? -- --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] Asterisk 13. Writing call quality parameters to CDR. How?
because of problems you are facing i decided to go way with second table
CREATE TABLE `cdr_extended` (
`id` int(11) unsigned NOT NULL AUTO_INCREMENT,
`uniqueid` varchar(32) NOT NULL DEFAULT '',
`callid` varchar(256) NOT NULL DEFAULT '' COMMENT 'sip call-id',
`hangupcause` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci
NOT NULL COMMENT 'info about hangup',
`peerip` varchar(15) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`recvip` varchar(15) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`from_u` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`uri` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`useragent` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT
NULL,
`codec1` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`codec2` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`llp` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
COMMENT 'lost packets by local end',
`rlp` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
COMMENT 'lost packets by remote end ',
`ljitt` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
COMMENT 'the same for jitter ',
`rjitt` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
COMMENT 'the same for jitter ',
PRIMARY KEY (`id`),
KEY `uniqueid` (`uniqueid`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8;
in hangup handler or h exten i will use func_odbc
like
insert into cdr_extended (uniqueid,hangupcause,peerip,...) values
('${UNIQUEID}',...);
Dne 18.3.2015 v 20:37 Dmitriy Serov napsal(a):
Hello.
Voice quality when calling - this is one of the most important in the
PBX.
You need to record the quality parameters for each call to improve.
Because the overall quality of a call can only be determined upon
completion, I did it in the HangUp handler and wrote in custom fields
of CDR.
This worked well in asterisk 11.
In asterisk 13 I did not find a handler after the call, but before
finalizing the CDR.
I tried to call the AGI and there to update the CDR record by unique
identifiers. But faced with the fact that there are no needed record
in the table yet.
To write the data into a separate table and join them may be an
option. But do not want to resort to such a decision
How do you solve this problem?
Dmitriy Serov.
--
---
Marek Cervenka
===
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Re: [asterisk-users] [BOUNTY] ASTERISK-22708 ODBC failover
bounty offer prolonged to 31.4.2015 (end of april) Dne 3.3.2015 v 16:22 Marek Cervenka napsal(a): hi, i'm offering bounty[1] $500 (five hundred) US dollars for resolving https://issues.asterisk.org/jira/browse/ASTERISK-22708 fix must be available for asterisk 11.x and asterisk 13.x and accepted to upstream As part of this fix we should see seamless fail down the ODBC database stack regardless of the database type (Must cleanup MySQL). No more lockups due to a down database server when other accessible database servers are available. this bounty offer expire on 10.3.2015 00:00 please contact me privately i'm on linkedin for identity verification i prefer people with known identity Marek Cervenka [1] Bounty rules https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties -- --- Marek Cervenka === -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] second BOUNTY donor for ASTERISK-22708 (ODBC failover)
hello, i'm searching second BOUNTY donor ($250) for https://issues.asterisk.org/jira/browse/ASTERISK-22708 if you want participate, please contact me privately -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] static realtime vs config files
hi, is it possible use asterisk static realtime and config files simultaneously in asterisk 11? i want [globals] from extensions.conf in database, but dialplan in extensions.conf config file i saw this can be configured in stasis.conf in asterisk 13 thanks -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] convert asterisk extensions to single numbers
hi, i'm converting extensions.conf to DB routing. can you help me with regexp or something which converts dialplan to single numbers like _3X0 to 310,320,330,340,... ? i found only https://pypi.python.org/pypi/asterisk_dialplan/0.1.2 but i need the opposite direction thanks -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] connect call to queue to specified agent
hi, is it possible connect call to queue to specified agent? like Mr. Neo called helpdesk queue, call picked by agent Smith Mr. Neo is calling again and i want connect him with agent Smith -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] higher cpu usage 1.8 - 11
hi, i upgraded few asterisk systems from last asterisk 1.8 to 11 and i see in graph that cpu usage is ~50% higher any ideas? configuration, modules, .. is the same -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] howto cancel simultaneous calls - dial(sip/phone1sip/phone2)
hi. i have dialplan with 2 simultaneous calls - dial(sip/phone1sip/phone2). when i cancel call on phone1 (push reject button), the call is still ringing on phone2 can i cancel call on both phones from one place(one phone)? thanks -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] mixmonitor - convert wav to mp3/aac
hi, i want convert mixmonitor recorded speech audio from wav to mp3 or aac can you recommend your settings for speech audio? filters, noise elimination, compression ratio, ... i will probably use lame thank you -- --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] Tutorial: compiling and installing Asterisk 13
Dne 12.9.2014 v 11:27 Lenz Emilitri napsal(a): Hi all, I just prepared a little tutorial on installing Asterisk 13 on CentOS 6.5 64-bit. See http://astrecipes.net/index.php?n=668 Hope you like. :) l. you can shrink it by: - srtp is in EPEL repo [root@dev6 ~]# yum list|grep srtp libsrtp.i686 1.4.4-4.20101004cvs.el6@epel libsrtp-devel.i686 1.4.4-4.20101004cvs.el6@epel - jansson is in EPEL repo [root@dev6 ~]# yum list|grep jansson jansson.i686 2.6-1.el6 @epel jansson-devel.i686 2.6-1.el6 @epel - pjproject spec file https://bugzilla.redhat.com/show_bug.cgi?id=1140324 -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] opus 11.12.0
hi, any plans update patch for 11.12.0? |https://github.com/meetecho/asterisk-opus https://github.com/netaskd/asterisk-opus/ | patching file build_tools/menuselect-deps.in patching file channels/chan_sip.c Hunk #1 succeeded at 7659 (offset -98 lines). Hunk #2 succeeded at 11011 (offset -34 lines). Hunk #3 succeeded at 11050 (offset -34 lines). Hunk #4 succeeded at 7 with fuzz 1 (offset -34 lines). Hunk #5 FAILED at 12722. 1 out of 6 hunks FAILED -- saving rejects to file channels/chan_sip.c.rej patching file codecs/codec_opus.c patching file codecs/ex_opus.h patching file configure.ac Hunk #2 succeeded at 2150 (offset 31 lines). patching file formats/format_vp8.c patching file include/asterisk/format.h patching file main/channel.c patching file main/format.c Hunk #6 succeeded at 1098 (offset 12 lines). patching file main/frame.c patching file main/rtp_engine.c Hunk #1 succeeded at 2326 (offset 37 lines). Hunk #2 succeeded at 2370 (offset 37 lines). patching file makeopts.in patching file res/res_rtp_asterisk.c Hunk #1 succeeded at 95 with fuzz 1 (offset 4 lines). Hunk #2 FAILED at 349. Hunk #3 succeeded at 3011 (offset 394 lines). Hunk #4 succeeded at 3097 (offset 394 lines). 1 out of 4 hunks FAILED -- saving rejects to file res/res_rtp_asterisk.c.rej thanks -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] asterisk multiple ip
hi, i need migrate customers from severeal to one asterisk server with multiple ip aliases like eth0 192.168.10.1 eth0:1 192.168.10.20 eth0:2 192.168.10.30 i must preserve endpoint configuration to these ip adressess the problem is if i register to 192.168.10.30, the answer is from 192.168.10.1 are there some ways for this scenario? 1) chan_pjsip? 2) kamailio in front of asterisk on the same server? 3) iptables magic? 4) ... thanks -- --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] asterisk multiple ip
it looks like i found solution with chan_pjsip /etc/asterisk/pjsip.conf [transport-udp-net1] type=transport protocol=udp bind=192.168.10.20 [transport-udp-net2] type=transport protocol=udp bind=192.168.10.30 [net1_user1] type=endpoint transport=transport-udp-net1 [net2_user1] type=endpoint transport=transport-udp-net2 can you someone confirm this solution? Dne 29.8.2014 v 11:26 Marek Cervenka napsal(a): hi, i need migrate customers from severeal to one asterisk server with multiple ip aliases like eth0 192.168.10.1 eth0:1 192.168.10.20 eth0:2 192.168.10.30 i must preserve endpoint configuration to these ip adressess the problem is if i register to 192.168.10.30, the answer is from 192.168.10.1 are there some ways for this scenario? 1) chan_pjsip? 2) kamailio in front of asterisk on the same server? 3) iptables magic? 4) ... thanks -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] asterisk SugarCrm integration
hello, can you recommend good asterisk-SugarCrm integration plugin? i googled a lot, but i want something what is used on daily basis thank you -- --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] asterisk SugarCrm integration
it's old. sugarcrm v7 is not supported Dne 28.8.2014 v 14:54 Scott Griepentrog napsal(a): I've used this before, and it appears to still be an active project. https://github.com/blak3r/yaai On Thu, Aug 28, 2014 at 7:29 AM, Marek Cervenka cerv...@fpf.slu.cz mailto:cerv...@fpf.slu.cz wrote: hello, can you recommend good asterisk-SugarCrm integration plugin? i googled a lot, but i want something what is used on daily basis thank you -- _ -- 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
Re: [asterisk-users] mixmonitor extension
for the record. info about opus from Lorenzo Mniero (author of Opus patch for asterisk) with his permission --cite-- Opus is just a codec. In order to save an audio file using Opus, you need a container, which for Opus is OGG. Asterisk supports OGG, but I think it is implemented to only dump Vorbis audio, and so the existing module would need to be extended to support Opus as well. I haven't checked how complex this could be, to be honest, so I have no idea about how much effort would be needed for this. Right now we don't need it, so I really can't say if and when we'll start working on this. Lorenzo --cite-- Dne 24.1.2014 10:42, Gareth Blades napsal(a): On 23/01/14 23:37, Marek Cervenka wrote: can someone confirm that mp3 is unsupported? is patch available? what about patch for Opus? uncle google doesnt know MP3 is only supported for reading not writing. Its a patent issue as Asterisk cannot distribute the software to write to mp3 under its own license. Its a similar issue with Opus as the codec is covered by a couple of patents in the USA. What most people do is use MixMonitor to record to .wav (alaw) and then in the 'h' extension call a program which runs a background task to convert the .wav file to whatever format they wish. Thats what we do but we actually use the Monitor application and we end up with both legs of the call and multiple sets of recordings if people pause and unpause. We then move these files off to a different server when they get mixed and converted to mp3 and then emailed out to our customers. We do it this way to reduce the load on the Asterisk boxes but also keep all the call recordings in a central location. -- --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] mixmonitor extension
i'm talking about native mp3,opus support in mixmonitor application. read the first answer from Gareth Blades Dne 24.1.2014 1:39, Patrick Lists napsal(a): On 24-01-14 00:37, Marek Cervenka wrote: can someone confirm that mp3 is unsupported? is patch available? Iirc mp3 was in asterisk-addons in asterisk 1.4 and iirc in later versions of asterisk you can enable format_mp3 in make menuselect. what about patch for Opus? uncle google doesnt know Did you really google? http://lmgtfy.com/?q=asterisk+opus Regards, Patrick -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] mixmonitor extension
hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? -- --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] mixmonitor extension
can someone confirm that mp3 is unsupported? is patch available? what about patch for Opus? uncle google doesnt know Dne 23.1.2014 16:31, Gareth Blades napsal(a): On 23/01/14 15:21, Marek Cervenka wrote: hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? core show file formats will give you a list of formats your system supports together with the filename extension. Not all may be supported for writing (mp3 being one example I believe). core show file formats Format Name Extensions -- -- slin mp3mp3 h264 h264 h264 g729 g729 g729 g719 g719 g719 gsmgsmgsm g726 g726-16g726-16 g726 g726-24g726-24 g726 g726-32g726-32 g726 g726-40g726-40 h263 h263 h263 gsmwav49 WAV|wav49 g722 g722 g722 ulaw au au alaw alaw alaw|al|alw ulaw pcmpcm|ulaw|ul|mu|ulw siren14siren14siren14 siren7 siren7 siren7 slin192sln192 sln192 slin96 sln96 sln96 slin48 sln48 sln48 slin44 sln44 sln44 slin32 sln32 sln32 slin24 sln24 sln24 slin16 sln16 sln16 slin12 sln12 sln12 slin slnsln|raw slin16 wav16 wav16 slin wavwav g723 g723sf g723|g723sf ilbc iLBC ilbc 30 file formats registered. -- --- Marek Cervenka Centrum Vypocetni Techniky jabber - cerve...@slu.cz CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz RHCE,RHCVA 100-175-678 === -- _ -- 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
[asterisk-users] groupcount fraud problem
hi, i have strange problem with call-limit/groupcount limiting. i set up limit of 2 calls. i'm using both methods but a for few times i have problem with successfull fraud with more calls than 2 asterisk is 1.8.22 someone with the same problem? any ideas how to solve or debug this problem? -- --- Marek === -- _ -- 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
Re: [asterisk-users] groupcount fraud problem
Dne 14.8.2013 13:35, Marek Cervenka napsal(a): hi, i have strange problem with call-limit/groupcount limiting. i set up limit of 2 calls. i'm using both methods but a for few times i have problem with successfull fraud with more calls than 2 asterisk is 1.8.22 someone with the same problem? any ideas how to solve or debug this problem? it's seems like they are using some transfer or system code to modify call flow in CDR i see | calldate | duration| billsec | peerip | recvip | useragent | uniqueid | uri | 2013-08-10 17:12:52 |7 | 2 | attacker_ip | attacker_ip | eyeBeam release 3006o stamp 17 | 1375679572.17728 | sip:clid_number@attacker_ip:14932 | | 2013-08-10 17:13:03 | 666 | 660 | siptrunk_ip | siptrunk_ip | operator_switch | 1375679583.17730 | sip:called_number@siptrunk_ip:5060 | -- --- Marek === -- _ -- 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
[asterisk-users] sip video endpoint with asterisk
hi, i need some small sip video endpoint for cloud videoconference (like bluejeans) i have this idea VIDEO OUT TV or projector with HDMI VIDEO IN cameras with h264 hw enconding - http://downloads.element14.com/raspberry-pi-camera/ http://downloads.element14.com/raspberry-pi-camera/ - logitech C920 - Creative Live! Cam Connect HD - ??? ENDPOINT - raspberry - miniPC linux + asterisk ? https://wiki.asterisk.org/wiki/display/AST/Video+Console https://wiki.asterisk.org/wiki/display/AST/Video+Console AUDIO IN + AUDIO OUT microphone with integrated speakers for the table http://www.jabra.com/Products/PC_Headsets/Jabra_SPEAK__510_Series/Jabra_Speak_510 http://www.jabra.com/Products/PC_Headsets/Jabra_SPEAK__510_Series/Jabra_Speak_510 (bluetooth connection!!!) http://www.phnxaudio.com/quattro3 http://www.phnxaudio.com/quattro3 http://www.yamaha.com/products/en/communication/usb_conference_speakerphones/ http://www.yamaha.com/products/en/communication/usb_conference_speakerphones/ http://www.dev-audio.com/products/microcone/ http://www.dev-audio.com/products/microcone/ http://www.clearone.com/products_chat160 http://www.clearone.com/products_chat160 do you think it is possible? any recommendations? -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] WebM / VP8 support
hello, any news about WebM/VP8 support in asterisk? some bounty where can i contribute? -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] salesforce opencti
hello, do you have someone connector to salesforce? http://wiki.developerforce.com/page/Open_CTI i need it for Mac OS X (the web/javascript way, not the old CTI Adapter way) i'm using Asterisk 1.8 thanks -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] AGI not generating sip 180/183 status
hello, i have strange problem with AGI (asterisk 1.8.10.0) when i use Dial from dialplan everything is ok when i dial from AGI script there is missing SIP Status 180 ringing and 183 session progress any ideas? DIAL without AGI 196.356479 10.0.0.193 - 10.0.0.213 SIP/SDP Request: INVITE sip:222333...@some.pbx.org, with session description 196.356768 10.0.0.213 - 10.0.0.193 SIP Status: 401 Unauthorized 196.365709 10.0.0.193 - 10.0.0.213 SIP Request: ACK sip:222333...@some.pbx.org 196.370028 10.0.0.193 - 10.0.0.213 SIP/SDP Request: INVITE sip:222333...@some.pbx.org, with session description 196.370503 10.0.0.213 - 10.0.0.193 SIP Status: 100 Trying 199.797325 10.0.0.213 - 10.0.0.193 SIP Status: 180 Ringing 199.797932 10.0.0.213 - 10.0.0.193 SIP/SDP Status: 183 Session Progress, with session description 199.878441 10.0.0.193 - 10.0.0.213 RTCP Receiver Report Source description 199.988259 10.0.0.193 - 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xD2C6DEB8, Seq=7289, Time=3171500, Mark 200.004139 10.0.0.213 - 10.0.0.193 RTP PT=ITU-T G.711 PCMA, SSRC=0x279E385A, Seq=50775, Time=28960 200.008118 10.0.0.193 - 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xD2C6DEB8, Seq=7290, Time=3171660 201.504218 10.0.0.213 - 10.0.0.193 RTP PT=ITU-T G.711 PCMA, SSRC=0x279E385A, Seq=50850, Time=40960 201.519477 10.0.0.193 - 10.0.0.213 SIP Request: BYE sip:222333444@10.0.0.213:5060 201.519611 10.0.0.213 - 10.0.0.193 SIP Status: 487 Request Terminated 201.519800 10.0.0.213 - 10.0.0.193 SIP Status: 200 OK 201.528465 10.0.0.193 - 10.0.0.213 SIP Request: ACK sip:222333...@some.pbx.org DIAL from AGI 66.581752 10.0.0.193 - 10.0.0.213 SIP/SDP Request: INVITE sip:222333...@some.pbx.org, with session description 66.581958 10.0.0.213 - 10.0.0.193 SIP Status: 401 Unauthorized 66.590738 10.0.0.193 - 10.0.0.213 SIP Request: ACK sip:222333...@some.pbx.org 66.59 10.0.0.193 - 10.0.0.213 SIP/SDP Request: INVITE sip:222333...@some.pbx.org, with session description 66.596167 10.0.0.213 - 10.0.0.193 SIP Status: 100 Trying 66.652571 10.0.0.213 - 10.0.0.193 SIP/SDP Status: 200 OK, with session description 66.676485 10.0.0.193 - 10.0.0.213 RTCP Receiver Report Source description 66.750371 10.0.0.193 - 10.0.0.213 SIP Request: ACK sip:222333444@10.0.0.213:5060 66.844392 10.0.0.193 - 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xE842E26F, Seq=3869, Time=1120100, Mark 66.854430 10.0.0.193 - 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xE842E26F, Seq=3870, Time=1120260 ... 69.404625 10.0.0.193 - 10.0.0.213 RTP PT=ITU-T G.711 PCMA, SSRC=0xE842E26F, Seq=3998, Time=1140740 69.516390 10.0.0.193 - 10.0.0.213 SIP Request: BYE sip:222333444@10.0.0.213:5060 69.516669 10.0.0.213 - 10.0.0.193 SIP Status: 200 OK -- --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk
Dne 21.6.2012 9:52, Ishfaq Malik napsal(a): On Wed, 2012-06-20 at 20:04 +0200, Stefan at WPF wrote: Hello, 1) I am wondering what is the best practice to monitor if there are or were problems with SIP calls on my Asterisk box. E.g. how about a software that extracts all calls from the /var/log/asterisk/full (I have permanently enabled verbose 10 and sip debug) log and tells me on which of them were problems? Checking the logs manually is very hard, but as SIP is a standardized protocoll, there should be tools doing that for you? As an example, a person calling me recently got a 488 Not acceptable error as reply from my Asterisk box. Nothing came through to my SIP phone, so I didn't know anything about the call or the problems (which were on his phone btw). I would like to be informed about such cases, know that there was a call to my Asterisk box that made problems. 2) How about monitoring speech quality? E.g. sometimes it seems like a packet is missing (I then have a short pause during the call), how to monitor such things and create statistics out of this data? So basically I want to monitor my Asterisk installation proactively for reliability/problems and (speech) quality. check asterisk testsuite https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation thereis scenarios for console sip client pjsua(from pjproject) which can perform speech quality measurement marek cervenka -- _ -- 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
Re: [asterisk-users] attended transfer with CEL
https://wiki.asterisk.org/wiki/display/AST/CEL+Function on this wiki is THIS IS NO LONGER TRUE REWRITE is there some way to write userfield,accountcode to the cel? Dne 5.6.2012 13:21, Marek Cervenka napsal(a): hello, is there someone who successfully get info about attended transfer from CEL? if yes, can you post some hints/algorithm/...? scenario A - customer B - secretary C - consultant1 D - consultant2 A - B B axfer C C axfer D i need to know time B with C (consultation) time A with C time C with D (consultation) time A with D time A with everyone (full time - from start to the end of call) -- --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] attended transfer with CEL
Dne 20.6.2012 18:40, Marek Cervenka napsal(a): https://wiki.asterisk.org/wiki/display/AST/CEL+Function on this wiki is THIS IS NO LONGER TRUE REWRITE is there some way to write userfield,accountcode to the cel? solved. it's set(CHANNEL(userfield)=something) another question i'm using patch from https://issues.asterisk.org/jira/browse/ASTERISK-18037 it works great but there is problem(bug?) in second axfer A - call - B - axfer(AtoC) - C - axfer(AtoD) D in cel is eventtype, cid_num, exten HOLD_START, A, B HOLD_STOP, A, B BUT second axfer is HOLD_START, B, C HOLD_STOP, B, C this is strange because on hold is A. is it a bug? very big problem is that, i cant get info about A - D call (after second axfer). there is no info about bridged channel A after axfer Dne 5.6.2012 13:21, Marek Cervenka napsal(a): hello, is there someone who successfully get info about attended transfer from CEL? if yes, can you post some hints/algorithm/...? scenario A - customer B - secretary C - consultant1 D - consultant2 A - B B axfer C C axfer D i need to know time B with C (consultation) time A with C time C with D (consultation) time A with D time A with everyone (full time - from start to the end of call) -- --- Marek Cervenka Centrum Vypocetni Techniky jabber - cerve...@slu.cz CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz RHCE,RHCVA 100-175-678 === -- _ -- 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
[asterisk-users] attended transfer with CEL
hello, is there someone who successfully get info about attended transfer from CEL? if yes, can you post some hints/algorithm/...? scenario A - customer B - secretary C - consultant1 D - consultant2 A - B B axfer C C axfer D i need to know time B with C (consultation) time A with C time C with D (consultation) time A with D time A with everyone (full time - from start to the end of call) -- --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] axfer with simple CDR
Dne 29.5.2012 18:23, Kevin P. Fleming napsal(a): On 05/29/2012 07:57 AM, Marek Cervenka wrote: is it possible with simple CDR fully describe axfer? (axfer is asterisk native, not phone function) No, it is not. CDRs (Asterisk or otherwise) are only capable of directly (simply) describing a call from party A to party B. They have no ability to describe call treatments, in-call features, or any other advanced features. Asterisk's CDRs *attempt* to represent such information, but as you've seen, they don't satisfy everyone, and it seems that many parties have conflicting ideas as to how things like transfers should be represented in CDRs. ok ok. i tried it :) i'll try it the right way - CEL (centos6,unixODBC,cel_odbc,mysql) any sql views,scripts,sql triggers someone? is it implemented in switchvox,asterisknow,trixbox,elastix? -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] axfer with simple CDR
hi, i read a lot about CDR problems this document is the best description of CDRs problem in Asterisk http://svn.digium.com/svn/asterisk/team/murf/RFCs/CDRfix2.rfc.docx i found but i cant still answer my question is it possible with simple CDR fully describe axfer? (axfer is asterisk native, not phone function) scenario A - customer B - secretary C - consultant1 D - consultant2 A - B B axfer C C axfer D i need to know time B with C (consultation) time A with C time C with D (consultation) time A with D time A with everyone (full time - from start to the end of call) (what about ring time?) is it possible? if yes, can you post some example? -- --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] Calendar Integration Problem
Dne 30.4.2012 11:09, Bharat Lalcheta napsal(a): Hiii all, I am using asterisk 1.8.9.2 and compile all modules related to calendar. neon version is 0.29.6. OS is ubuntu 11.10. I configured ical for zimbra, caldav for google mail and ews for exchange 2010 calendar. ical and caldav setup working fine and i am getting my calendar events perfectly. But for exchange 2010 calendar i am getting following error. Unable to communicate with Exchange Web Service at 'https://ex1.domain.com/EWS/Exchange.asmx': Could not authenticate to server: ignored NTLM challenge, GSSAPI authentication error: Unspecified GSS failure. Minor code may provide more information: Credentials cache file '/tmp/krb5cc_0' not found my calendar.conf is as follows [calendar3] type = ews ; type of calendar--currently supported: ical, caldav, exchange, or ews url = https://ex1.domain.com/EWS/Exchange.asmx ; URL to MS Exchange EWS user = myn...@domain.com mailto:myn...@domain.com ; Exchange username secret = xx ; Exchange password refresh = 10 ; refresh calendar every n minutes timeframe = 20 try user = domain.com/myname mailto:myn...@domain.com ; Exchange username -- _ -- 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
[asterisk-users] cdr documentation - new fields
hi, there are 3 new cdr fields in asterisk 1.8 (https://wiki.asterisk.org/wiki/display/AST/New+in+1.8#Newin1.8-CDR) linkedid - is based on uniqueID, but spreads to other channels as transfers, dials, etc are performed. Thus the pieces of CDR can be grouped into multilegged sets. sequence - can be combined with linkedid or uniqueid to uniquely identify a CDR. peeraccount - ? can someone with write permissions fix this doc? https://wiki.asterisk.org/wiki/display/AST/CDR+Fields https://wiki.asterisk.org/wiki/display/AST/MySQL+CDR+Backend thanks -- --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] call pickup
On 10/05/2011 11:21 PM, A. M. Hoffmeister wrote: Am 05.10.2011 20:42, schrieb Marek Cervenka: hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? You can have that with subscriptions/hints, for example Snom phones can display not only a call to one of the peers but also the caller and callee identification. can you point me to some doc/examples? how this is implemented in SIP? i think about sending some notify from dialplan (i have incoming call, i know who is in pickup group, i can send call to callee and before send some NOTIFY to other phones in the pickupgroup) i found only one app like this - jabbersend. but i need this notification on phone screen This works jaw to cheek with BLF (busy lamp field) which allows to monitor other extensions' status (in_use, ringing...). Of course you can be member of a pickup group without monitoring the status of any of the peers, and you can monitor a peer's status without being in the same pickup group (although not pickup the call then, obviously :-) -- --- Marek Cervenka Centrum Vypocetni Techniky jabber - cerv...@njs.netlab.cz CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz RHCE 100-175-678 === -- _ -- 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
[asterisk-users] call pickup
hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? thanks -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] asterisk rpm build problem
hi, i'm trying build asterisk rpm normal compilation is ok but rpm building always fail centos6/asterisk 1.8.5.0 any ideas? gcc -o recno/rec_utils.o -c recno/rec_utils.c -MD -MT recno/rec_utils.o -MF .recno_rec_utils.o.d -MP -pthread -Wall -D__DBINTERFACE_PRIVATE -I. -I.. -Iinclude -Ihash -Ibtree -Irecno -I/root/rpmbuild/BUILD/asterisk-1.8.5.0/include -O2 -g -march=i386 -mtune=i686 -Werror-implicit-function-declaration -I/usr/include/libxml2 -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations-Wno-strict-aliasing -O2 -g -march=i386 -mtune=i686 -Werror-implicit-function-declaration ar cr libdb1.a hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o mpool/mpool.o recno/rec_close.o recno/rec_delete.o recno/rec_get.o recno/rec_open.o recno/rec_put.o recno/rec_search.o recno/rec_seq.o recno/rec_utils.o ranlib libdb1.a make[2]: Leaving directory `/root/rpmbuild/BUILD/asterisk-1.8.5.0/main/db1-ast' /root/rpmbuild/BUILD/asterisk-1.8.5.0/build_tools/make_linker_version_script asterisk gcc -o asterisk -Wl,--export-dynamic -Wl,--version-script,asterisk.exports -Wl,--dynamic-list,asterisk.dynamics abstract_jb.o acl.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o chanvars.o cli.o config.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o logger.o manager.o md5.o netsock.o netsock2.o pbx.o plc.o poll.o privacy.o rtp_engine.o say.o sched.o security_events.o sha1.o slinfactory.o srv.o ssl.o stdtime/localtime.o strcompat.o strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o xml.o xmldoc.o db1-ast/libdb1.a buildinfo.o -lssl -lcrypto -lc -lxml2 -lz -lm -ldl -lpthread -ltermcap -lm -lresolv -ledit -lcurses astobj2.o: In function `ast_atomic_fetchadd_int': /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: undefined reference to `__sync_fetch_and_add_4' /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: undefined reference to `__sync_fetch_and_add_4' ccss.o: In function `ast_atomic_fetchadd_int': /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: undefined reference to `__sync_fetch_and_add_4' /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: undefined reference to `__sync_fetch_and_add_4' /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: undefined reference to `__sync_fetch_and_add_4' cdr.o:/root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: more undefined references to `__sync_fetch_and_add_4' follow utils.o: In function `ast_atomic_dec_and_test': /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:646: undefined reference to `__sync_sub_and_fetch_4' utils.o: In function `ast_atomic_fetchadd_int': /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: undefined reference to `__sync_fetch_and_add_4' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make[1]: Leaving directory `/root/rpmbuild/BUILD/asterisk-1.8.5.0/main' make: *** [main] Error 2 error: Bad exit status from /var/tmp/rpm-tmp.FeglRm (%build) -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] sip trunk balancing
hi, is there some way to balance accross sip trunks by the number of calls? example 3 trunks alfa(max 30 channels, priority 1),delta(60,priority 2),omega(90,priority 3) alfa have 25 calls now i want next call terminate to delta. how to find in asterisk the current calls number on sip trunk alfa? 1) set call-limit in sip.conf. then in the dialplan sip show peer inuse|grep alfa - parse - if numcalls 25 then dial(sip/delta) 2) groupcount ? 3) what else? thanks Marek -- _ -- 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
Re: [asterisk-users] Sharing Fail2ban data
I've been doing a little work that I wanted to share. We've had a number of Asterisk systems that have been under heavier than normal attack. We use fail2ban but we either have to let each system be exposed or keep all the data synchronized which is a bit of a chore. I wrote a little server that assists in keeping data synchronized across sites. If you're interested in using it to assist in managing your own fail2ban sharing list I'll gladly share it. I also am offering it as a free service for those who are interested in contributing to a blacklist. If you're interested the information is available here: http://fail2ban.aleph-com.net/fail2ban_sharing If you're interested in the server code just drop me an email. i'm interested in the server code. thanks -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] resending cause codes
hello,
i'm testing sending ISDN cause codes to customer pbx (test scenario for
unallocated number)
topology:
PSTN-E1-AsteriskA-AsteriskB-SOMEPBX
INVITE from SOMEPBX to PSTN
AsteriskA sends to AsteriskB
Status-Line: SIP/2.0 503 Service Unavailable
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
how can i resend HangupCauseCode from AsteriskB to SOMEPBX?
i'm tried this on AsteriskB
exten = _X.,1,Dial(SIP/AsteriskA,${EXTEN})
exten = _X.,n,Hangup(${SIP_HEADER(X-Asterisk-HangupCauseCode)})
thanks
--
---
Marek Cervenka
===
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Re: [asterisk-users] Asterisk T.38 Gateway code testing
asterisk t38 gw patch updated to 1.6.2.9 https://issues.asterisk.org/view.php?id=13405 i made page for Asterisk T.38 Gateway code testing http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway Final T.38 Gateway API for asterisk 1.8 will be posted by Kevin Fleming later BUT Asterisk 1.8 is too far and we need t.38 gw now (for testing etc) if you would like help/test current code(last patch from https://issues.asterisk.org/view.php?id=13405), please contact me i have 2 public testing machines connected over E1 my jabber is cerv...@njs.netlab.cz --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] UDPTL T38 via NAT
On 06/22/2010 02:51 PM, Johann Steinwendtner wrote: On 2010-06-22 12:36, Remco Bressers wrote: Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29) This means my outgoing udptl traffic is correctly translated, but somehow i'm sending 172.16.0.156 instead of my public IP address on the firewall. try asterisk 1.6.2.9 --- Marek Cervenka === -- _ -- 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
Re: [asterisk-users] UDPTL T38 via NAT
On 06/22/2010 04:38 PM, marek cervenka wrote: On 06/22/2010 02:51 PM, Johann Steinwendtner wrote: On 2010-06-22 12:36, Remco Bressers wrote: Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29) This means my outgoing udptl traffic is correctly translated, but somehow i'm sending 172.16.0.156 instead of my public IP address on the firewall. try asterisk 1.6.2.9 What would be the reason to do that? Is there any change on this in 1.6.2.9? yes 1.6.2.x branch is a lot better in T.38 area --- Marek Cervenka === -- _ -- 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
[asterisk-users] Asterisk T.38 Gateway code testing
hi, i made page for Asterisk T.38 Gateway code testing http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway Final T.38 Gateway API for asterisk 1.8 will be posted by Kevin Fleming later BUT Asterisk 1.8 is too far and we need t.38 gw now if you would like help/test current code(last patch from https://issues.asterisk.org/view.php?id=13405), please contact me i have 2 public testing machines connected over E1 PLEASE do not post bug reports to https://issues.asterisk.org/view.php?id=13405 because this patch cannot be included in 1.6.2 (digium rules) i'm in contact with klaus darilion and daniel ferenci(asterisk t.38 developers) and i can arrange fixing bugs my jabber is cerv...@njs.netlab.cz look forward for better t.38 days --- Marek Cervenka jabber - cerv...@njs.netlab.cz === -- _ -- 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
[asterisk-users] ANNOUNCE: New version of Activa TAPI driver
hello, there is new version of the best open source TAPI driver for Asterisk - Activa 1.6.1 * NEW: Asterisk 1.6 compatibility (partially sponsored by IPEX a.s. http://www.ipex.cz) * NEW: FEATURE_CODES standardization for AgentACD integration login, logout, ready, notReady. * NEW: ActivaTSP x64 version. * New: Windows 2008 Server compatibility. * CHANGE: Some performance optimization. * FIX: SIP/ Dns can generate void extensions. * FIX: in process dn expresion, the duplicate filter deletes non duplicate entries. download: http://sourceforge.net/projects/activa/files/ doc: http://activa.sourceforge.net/readme.html many thanks to Activa Team --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk cdr - remote ip address - SOLVED
for the record
(added to http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql)
some_context
;dial trunk
exten = _X.,1,Dial(SIP/trunk/${EXTEN})
;exten h must be in same context!
exten = h,1,noop(extended CDR)
exten = h,n,set(CDR(hangupcause)=${HANGUPCAUSE}) ; hangupcause
exten = h,n,set(CDR(peerip)=${CHANNEL(peerip)}) ; like
10.0.0.5 if behind nat
exten = h,n,set(CDR(recvip)=${CHANNEL(recvip)}) ; like
194.79.52.192 - public ip
exten = h,n,set(CDR(from)=${CHANNEL(from)}) ; like
sip:1...@sip.proxy.cz
exten = h,n,set(CDR(uri)=${CHANNEL(uri)}) ; like
sip:1...@10.0.0.5
exten = h,n,set(CDR(useragent)=${CHANNEL(useragent)}) ; useragent
like Aastra_57i
exten = h,n,set(CDR(codec1)=${CHANNEL(audioreadformat)}) ; codec *
exten = h,n,set(CDR(codec2)=${CHANNEL(audiowriteformat)}) ;
exten = h,n,set(CDR(llp)=${CHANNEL(rtpqos,audio,local_lostpackets)}) ; lost
packets by local end **
exten = h,n,set(CDR(rlp)=${CHANNEL(rtpqos,audio,remote_lostpackets)}) ; lost
packets by remote end
exten = h,n,set(CDR(ljitt)=${CHANNEL(rtpqos,audio,local_jitter)}) ; the
same for jitter
exten = h,n,set(CDR(rjitt)=${CHANNEL(rtpqos,audio,remote_jitter)})
* i dont know if the same channel can have different audioreadformat and
audiowriteformat. imho not
** RTPAUDIOQOS isnt ok. check
http://lists.digium.com/pipermail/asterisk-biz/2009-November/031910.html
known problem:
it is only for caller. i dont know how to log call leg B
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of marek cervenka
Sent: Monday, November 16, 2009 8:50 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk cdr - remote ip address
hi,
i want add info about remote party ip address to the asterisk cdr table
can you recommend me the system way?
---
Marek Cervenka
===
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[asterisk-users] asterisk cdr - remote ip address
hi, i want add info about remote party ip address to the asterisk cdr table can you recommend me the system way? thanks --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk cdr - remote ip address
- exten = s,x,Set(CDR(userfield) = information) - replace information
with the information like ${remoteip}
${remoteip} variable doesnt exist in asterisk (for remote voip phone)
SIPURI=sip:6...@192.168.1.184:5061 doesnt have public ip
i'm only found way
- check ${CHANNEL} for name
- check astDB SIP/Registry
- set some variable
really doesnt exist some cleaner way?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of marek cervenka
Sent: Monday, November 16, 2009 8:50 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk cdr - remote ip address
hi,
i want add info about remote party ip address to the asterisk cdr table
can you recommend me the system way?
---
Marek Cervenka
===
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] [Asterisk 0013405]: [patch] T38 gateway (fwd)
testers needed -- Forwarded message -- Date: Wed, 11 Nov 2009 17:48:04 -0600 Subject: [Asterisk 0013405]: [patch] T38 gateway A NOTE has been added to this issue. == https://issues.asterisk.org/view.php?id=13405 == Reported By:dafe_von_cetin Assigned To: == Project:Asterisk Issue ID: 13405 Category: Applications/app_fax Reproducibility:N/A Severity: feature Priority: normal Status: confirmed Asterisk Version: SVN Regression: No Reviewboard Link: SVN Branch (only for SVN checkouts, not tarball releases): trunk SVN Revision (number only!): 140548 == Date Submitted: 2008-08-30 16:44 CDT Last Modified: 2009-11-11 17:47 CST == Summary:[patch] T38 gateway Description: Hi all, I'm sending you patch containing new application app_faxgateway.c (FaxGateway) which is able to mediate T30 to T38 and vice versa. Feature is using spands library (I used spandsp-0.0.4pre18 and spandsp-0.0.5pre4). Best regards Daniel. == -- (0113693) dafe_von_cetin (reporter) - 2009-11-11 17:47 https://issues.asterisk.org/view.php?id=13405#c113693 -- Hi, I've just uploaded the patch update for the newest trunk. The patch is still without previously mentioned transparency. Daniel. Issue History Date ModifiedUsername FieldChange == 2009-11-11 17:47 dafe_von_cetin Note Added: 0113693 == ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] supermicro hardware + sangoma
hi, i have new supermicro server (centos5, 2.6.30.9/2.6.27.34/2.6.18-distro kernels, wanpipe 3.5.6) card is: 1 . AFT-A101-SH : SLOT=4 : BUS=8 : IRQ=11 : CPU=A : PORT=1 : HWEC=32 : V=36 and i have this in log irq 17: nobody cared (try booting with the irqpoll option) Pid: 0, comm: swapper Not tainted 2.6.30.9rh #1 Call Trace: [c0465b07] __report_bad_irq+0x27/0x90 [c0465caa] note_interrupt+0x13a/0x180 [c04665af] handle_fasteoi_irq+0x9f/0xd0 [c0466510] ? handle_fasteoi_irq+0x0/0xd0 IRQ [c0404506] ? do_IRQ+0x46/0xb0 [c0588234] ? acpi_hw_write_port+0x27/0x71 [c0403469] ? common_interrupt+0x29/0x30 [c05943d4] ? acpi_idle_enter_bm+0x218/0x241 [c062bf8e] ? cpuidle_idle_call+0x6e/0xc0 [c0401e45] ? cpu_idle+0x35/0x60 [c06d42f2] ? start_secondary+0x182/0x1e0 handlers: [f872d9b0] (sdla_isr+0x0/0x310 [wanpipe]) Disabling IRQ #17 dou you have idea what is the problem? irqpoll doesnt help i have tried this supermicro motherboards http://www.supermicro.com/products/motherboard/QPI/5500/X8DTU-F.cfm http://www.supermicro.com/products/motherboard/QPI/5500/X8DTi-F.cfm http://www.supermicro.com/products/motherboard/QPI/5500/X8DTL-iF.cfm do you have someone working sangoma card with Tylersburg(intel 5520/5500) chipset? thanks p.s. sorry for offtopic :( --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ekiga, Twinkle and from where to start with open source
Hi All; I am looking to start develop an Softphone that has messanger feature (voice and text, who is online also), anyone can advise for the best link to start with it, so they have open source for softphone that we can start on it from there? http://www.qutecom.org (platform - windows,linux,mac) --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] playing media(moh,prompts) from flash player
hi, i'm searching solution for playing media(moh,prompts,voicemail,recordings - wav format) from adobe flash player (web browser) flash cannot play wav directly (imho) i must convert files to any other format on-the-fly - i cannot use mp3 because of royalties - next option is swf (with ffmpeg), supported free audio codecs (http://en.wikipedia.org/wiki/Flash_Video#Format_details) * uncompressed PCM * ADPCM * AAC can you someone recommend solution/combination which works? tnx --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open source SIP client
can anybody help me to give Opensource SIP client information which can be modified as per our requirment http://www.qutecom.org --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SRTP testers needed
please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compilerun clients with srtp (linksys,grandstream,aastra, qutecom, eyebeam, ...) digium need feedback for srtp inclusion to 1.6.3.0 http://bugs.digium.com/view.php?id=5413 if you need additional info, i'm on jabber - cerv...@njs.netlab.cz thanks! --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
