Re: [asterisk-users] Need stronger SRTP ciphers (256 bit)
Some more information (would love some thoughts on this, I have never submitted
a patch yet).
Groundwire (Popular SIP app) supports the following cipher suites for SRTP:
AES_CM_128_HMAC_SHA1_32
AES_CM_128_HMAC_SHA1_80
AES_CM_192_HMAC_SHA1_32
AES_CM_192_HMAC_SHA1_80
AES_CM_256_HMAC_SHA1_32
AES_CM_256_HMAC_SHA1_80
AEAD_AES_128_GCM
AEAD_AES_256_GCM
I see in the asterisk 13.9.1 source tarsal, in res/res_srtp.c :
Could adding support for the above cipher suites be as simple as adding more
options to this switch/case statement with the appropriate parameters or is
there more to it?
Thank you!
static int policy_set_suite(crypto_policy_t *p, enum ast_srtp_suite suite)
{
switch (suite) {
case AST_AES_CM_128_HMAC_SHA1_80:
p->cipher_type = AES_128_ICM;
p->cipher_key_len = 30;
p->auth_type = HMAC_SHA1;
p->auth_key_len = 20;
p->auth_tag_len = 10;
p->sec_serv = sec_serv_conf_and_auth;
return 0;
case AST_AES_CM_128_HMAC_SHA1_32:
p->cipher_type = AES_128_ICM;
p->cipher_key_len = 30;
p->auth_type = HMAC_SHA1;
p->auth_key_len = 20;
p->auth_tag_len = 4;
p->sec_serv = sec_serv_conf_and_auth;
return 0;
default:
ast_log(LOG_ERROR, "Invalid crypto suite: %u\n", suite);
> On May 30, 2016, at 11:49 AM, Kevin Long wrote:
>
>
>
> Hi folks,
>
>
> At least several endpoints (soft phone and desk phones) are supporting
> various 256 bit ciphers for SRTP these days. I *believe* libsrtp has been
> updated to allow this, and that only the code in Asterisk has not been been
> updated to allow these stronger ciphers.
>
> Would anyone with the know-how be willing/able to submit a patch ?
>
>
> Thank you,
>
> Kevin Long
> --
> _
> -- 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
>
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[asterisk-users] Trying to record incoming calls that go to queues in Asterisk v11
Hi everyone.
It seems that all the documentation for Asterisk has become obsolete
when it comes to using the Monitor command on a call queue.
To the best of my knowledge, the way to get Asterisk to record a call
that goes into one of your call queues is by doing this in the dialplan:
exten => 1,1,Answer()
same => n,Set(DAY=${STRFTIME(${EPOCH},,%Y-%m-%d)})
same => n,Set(TIME=${STRFTIME(${EPOCH},,%H%M%S)})
same => n,Set(MONITOR_FILENAME="incoming/${DAY}/${TIME}-${EXTEN}")
same => n,Queue(lsqueue-all)
same => n,Macro(handle-hangup)
and then add these lines in the queue context in queues.conf:
[lsqueue-all]
monitor-format=wav
monitor-type=MixMonitor
But when I reload Asterisk, I don't get any new files in
/var/spool/asterisk/monitor/incoming.
We're able to record our outgoing calls without any trouble with the
following dialplan:
exten => call,1,NoOp()
same => n,Set(DAY=${STRFTIME(${EPOCH},,%Y-%m-%d)})
same => n,Set(TIME=${STRFTIME(${EPOCH},,%H%M%S)})
same => n,Set(FILENAME="outgoing/${DAY}/${TIME}-${E}")
same => n,Monitor(wav,${FILENAME},m)
same => n,Dial(SIP/dolphintel/${E})
The file permissions on the "outgoing" and "incoming" are the same, plus
we don't get any errors in the Asterisk console about not being able to
write the files, so I'm pretty sure it's not a problem with actually
writing the files. It just doesn't seem to even try.
Any help will be much appreciated.
--
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[asterisk-users] Asterisk 13 IAX and MoH realtime
Asterisk 13.9.1 seems to be ignoring my realtime IAX configuration. I have carried this configuration over from version 1.8 and it worked until 13.7 at least. The config mapping is done: pbxoficina*CLI> core show config mappings Config Engine: mysql ===> ps_contacts (db=general, table=ps_contacts) ===> ps_endpoint_id_ips (db=general, table=ps_endpoint_id_ips) ===> ps_domain_aliases (db=general, table=ps_domain_aliases) ===> ps_aors (db=general, table=ps_aors) ===> ps_auths (db=general, table=ps_auths) ===> ps_endpoints (db=general, table=ps_endpoints) ===> musiconhold (db=general, table=musiconhold) ===> queues (db=general, table=queues) ===> iaxpeers (db=general, table=iaxfriends) ===> iaxusers (db=general, table=iaxfriends) ===> voicemail (db=general, table=voicemail) ===> chan_dahdi.conf (db=general, table=ast_config) ===> extensions.conf (db=general, table=ast_config) Except for IAX and MoH everything else is working. I can get my dialplan from realtime static, pjsip is loading all endpoints, auths and aors, etc. I do not get any errors while loading Asterisk and even unloading and reloading the module does not indicate any problem: pbxoficina*CLI> module unload chan_iax2.so Unloaded chan_iax2.so Unloading chan_iax2.so == Unregistered custom function IAXPEER == Unregistered custom function IAXVAR == Manager unregistered action IAXpeers == Manager unregistered action IAXpeerlist == Manager unregistered action IAXnetstats == Manager unregistered action IAXregistry == Unregistered application 'IAX2Provision' == Unregistered channel type 'IAX2' pbxoficina*CLI> module load chan_iax2.so Loaded chan_iax2.so == Parsing '/etc/asterisk/iax.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Binding IAX2 to default address 0.0.0.0:4569 == Registered application 'IAX2Provision' == Registered custom function 'IAXPEER' == Registered custom function 'IAXVAR' == Manager registered action IAXpeers == Manager registered action IAXpeerlist == Manager registered action IAXnetstats == Manager registered action IAXregistry == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == 10 helper threads started == IAX Ready and Listening == Parsing '/etc/asterisk/iaxprov.conf': Found -- Loaded provisioning template 'default' Loaded chan_iax2.so => (Inter Asterisk eXchange (Ver 2)) MoH has the same problem, I cannot see any of the classes I set up from the database, only what is in the text file. I remember IAX working in Asterisk 13.7.2 but MoH has been broken since 13 came out. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- 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] Need stronger SRTP ciphers (256 bit)
Hi folks, At least several endpoints (soft phone and desk phones) are supporting various 256 bit ciphers for SRTP these days. I *believe* libsrtp has been updated to allow this, and that only the code in Asterisk has not been been updated to allow these stronger ciphers. Would anyone with the know-how be willing/able to submit a patch ? Thank you, Kevin Long -- _ -- 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 odbc segfaults (SOLVED)
show continues *** glibc detected *** /usr/sbin/asterisk: double free or corruption (!prev): 0x0a11e8a0 *** === Backtrace: = /lib/libc.so.6(+0x70c91)[0xb6e12c91] /lib/libc.so.6(+0x736f1)[0xb6e156f1] /usr/lib/libmyodbc5w.so(my_free+0x52)[0xb6160992] /usr/lib/libmyodbc5w.so(free_result_bind+0x95)[0xb6131045] /usr/lib/libmyodbc5w.so(free_current_result+0x60)[0xb6133150] /usr/lib/libmyodbc5w.so(next_result+0x23)[0xb6133193] /usr/lib/libmyodbc5w.so(my_SQLFreeStmtExtended+0x21b)[0xb6125bbb] /usr/lib/libmyodbc5w.so(my_SQLFreeStmt+0x2e)[0xb6125e8e] /usr/lib/libmyodbc5w.so(SQLFreeHandle+0x70)[0xb6126a70] /usr/lib/libodbc.so.2(+0x1ab39)[0xb654bb39] /usr/lib/libodbc.so.2(SQLFreeHandle+0x26)[0xb654c4c6] /usr/lib/asterisk/modules/res_config_odbc.so(+0x4e8f)[0xb5ed3e8f] /usr/sbin/asterisk(ast_load_realtime_multientry_fields+0x5a)[0x811b7ba] /usr/lib/asterisk/modules/res_sorcery_realtime.so(+0x125e)[0xb770d25e] /usr/sbin/asterisk(ast_sorcery_retrieve_by_fields+0x1ed)[0x81f862d] /usr/lib/asterisk/modules/res_pjsip.so(ast_sip_get_endpoints+0x3b)[0xb593efab] /usr/lib/asterisk/modules/chan_pjsip.so(+0x955e)[0xb3be055e] /usr/sbin/asterisk[0x816b1c3] /usr/sbin/asterisk[0x816e248] /usr/sbin/asterisk[0x816e986] /usr/sbin/asterisk[0x80950c2] /usr/sbin/asterisk[0x8097dae] /lib/libc.so.6(__libc_start_main+0xe6)[0xb6db8d26] /usr/sbin/asterisk[0x80719e1] === Memory map: 08048000-082d9000 r-xp 08:01 31383 /usr/sbin/asterisk 082d9000-082e5000 rw-p 00291000 08:01 31383 /usr/sbin/asterisk 082e5000-08309000 rw-p 00:00 0 0966b000-0a137000 rw-p 00:00 0 [heap] b2e2a000-b2e2b000 rw-p 00:00 0 b2e2b000-b2e2c000 ---p 00:00 0 b2e2c000-b2e67000 rwxp 00:00 0 [stack:3966] b2e67000-b2e68000 ---p 00:00 0 b2e68000-b2ea3000 rwxp 00:00 0 [stack:3965] b2ea3000-b2ea4000 ---p 00:00 0 b2ea4000-b2edf000 rwxp 00:00 0 [stack:3964] b2edf000-b2ee ---p 00:00 0 b2ee-b2f1b000 rwxp 00:00 0 [stack:3963] b2f1b000-b2f1c000 ---p 00:00 0 b2f1c000-b2f57000 rwxp 00:00 0 [stack:3962] b2f57000-b2f58000 ---p 00:00 0 b2f58000-b2f93000 rwxp 00:00 0 [stack:3961] b2f93000-b2f94000 ---p 00:00 0 b2f94000-b2fcf000 rwxp 00:00 0 [stack:3960] b2fcf000-b2fd ---p 00:00 0 b2fd-b300b000 rwxp 00:00 0 [stack:3959] b300b000-b300c000 ---p 00:00 0 b300c000-b3047000 rwxp 00:00 0 [stack:3958] b3047000-b3048000 ---p 00:00 0 b3048000-b3083000 rwxp 00:00 0 [stack:3957] b3083000-b3084000 ---p 00:00 0 b3084000-b30bf000 rwxp 00:00 0 [stack:3956] b30bf000-b30c ---p 00:00 0 b30c-b30fb000 rwxp 00:00 0 [stack:3955] b30fb000-b30fc000 ---p 00:00 0 b30fc000-b3137000 rwxp 00:00 0 [stack:3954] b3137000-b3138000 ---p 00:00 0 b3138000-b3173000 rwxp 00:00 0 [stack:3953] b3173000-b31f5000 rw-p 00:00 0 b31f5000-b31f6000 ---p 00:00 0 b31f6000-b3231000 rwxp 00:00 0 [stack:3952] b3231000-b3232000 ---p 00:00 0 b3232000-b326d000 rwxp 00:00 0 [stack:3951] b326d000-b326e000 ---p 00:00 0 b326e000-b32a9000 rwxp 00:00 0 [stack:3950] b32a9000-b32aa000 ---p 00:00 0 b32aa000-b3aaa000 rwxp 00:00 0 [stack:3949] b3aaa000-b3aab000 ---p 00:00 0 b3aab000-b3ae6000 rwxp 00:00 0 [stack:3948] b3ae6000-b3ae7000 ---p 00:00 0 b3ae7000-b3b22000 rwxp 00:00 0 [stack:3947] b3b22000-b3b23000 ---p 00:00 0 b3b23000-b3b5e000 rwxp 00:00 0 [stack:3946] b3b5e000-b3b5f000 r-xp 08:01 31283 /usr/lib/asterisk/modules/res_curl.so b3b5f000-b3b6 rw-p 08:01 31283 /usr/lib/asterisk/modules/res_curl.so b3b6-b3b66000 r-xp 08:01 31326 /usr/lib/asterisk/modules/res_pjsip_mwi.so b3b66000-b3b67000 rw-p 5000 08:01 31326 /usr/lib/asterisk/modules/res_pjsip_mwi.so b3b67000-b3b72000 r-xp 08:01 31333 /usr/lib/asterisk/modules/res_pjsip_outbound_registration.so b3b72000-b3b73000 rw-p b000 08:01 31333 /usr/lib/asterisk/modules/res_pjsip_outbound_registration.so b3b73000-b3b74000 r-xp 08:01 31245 /usr/lib/asterisk/modules/func_vmcount.so b3b74000-b3b75000 rw-p 08:01 31245 /usr/lib/asterisk/modules/func_vmcount.so b3b75000-b3b77000 r-xp 08:01 31188 /usr/lib/asterisk/modules/format_siren7.so b3b77000-b3b78000 rw-p 1000 08:01 31188 /usr/lib/asterisk/modules/format_siren7.so b3b78000-b3b7a000 r-xp 08:01 31208 /usr/lib/asterisk/modules/func_dialplan.so b3b7a000-b3b7b000 rw-p 1000 08:01 31208 /usr/lib/asterisk/modules/func_dialplan.so b3b7b000-b3b7e000 r-xp 08:01 31147 /usr/lib/asterisk/modules/cel_custo
