https://docs.asterisk.org/latest/Configuration/Interfaces/Asterisk-Manager-Interface-AMI/Asterisk-Manager-Interface-AMI-Changes/
On Tue, Jul 11, 2023 at 11:54 AM, TTT <[li...@telium.io](mailto:On Tue, Jul 11,
2023 at 11:54 AM, TTT < wrote:
> Is there a web page that lists the AMI versions mapped
On 7/1/2023 11:40 AM, TTT wrote:
> I thought it was replaced with CALLERPRES(allowed) but this generated an
> error too in Asterisk 20.
From UPGRADE.txt¹:
The CALLERPRES() dialplan function is deprecated in favor of
CALLERID(num-pres) and CALLERID(name-pres).
Kind regards,
Se
I would never recommend new installs use IAX2, so if you envision this
moving beyond the toy/PoC stage I suggest you giving PJSIP another go.
Kind regards,
Sean
1. https://seanbright.com/voipms.png
2. https://wiki.voip.ms/article/Asterisk_PJSIP
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On 3/28/2023 4:03 AM, Fourhundred Thecat wrote:
> > On 2023-03-04 23:11, Sean Bright wrote:
> >
> > cdr/cdr_sqlite3_custom.c line 311
>
> Hello,
>
> I asked here recently how to change the location where
> "cdr_sqlite3_custom" stores the sqlite databa
c line 311
Kind regards,
Sean--
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https:
On 1/26/2023 5:16 AM, Antony Stone wrote:
> It does not work if it's written in AEL - assigning global variables works,
> but the above does not.
I've created a JIRA issue[1] for this as well as a proposed patch[2]. Assuming
all goes well this should work in future releases.
Ki
move the
> buildup of background
> ; noise from the conference. Highly recommended
> for large conferences
> ; due to its performance enhancements.
I would try adding this to all of your user profiles (type = user) and
see if that improv
ration?
Kind regards,
Sean--
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https://w
your dial string:
Dial(MulticastRTP/basic/224.0.0.3:5060//c(g722))
Kind regards,
Sean
1:
https://github.com/asterisk/asterisk/blob/bd821549af3bccb000c809121094adb5b84fec7f/CHANGES#L2598-L2611
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Sean--
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On 12/23/2021 2:47 PM, Dovid Bender wrote:
> Has anyone gotten SAY_DTMF_INTERRUPT to work?
Issue created[1] and tentative patch submitted for review[2].
[1] https://issues.asterisk.org/jira/browse/ASTERISK-29816
[2] https://gerrit.asterisk.org/c/asterisk/+/17712
Kind regards,
S
ariable, so:
Set(SAY_DTMF_INTERRUPT=true)
Should do it. However I have confirmed locally that it does not work as
documented, so feel free to create an issue in JIRA.
Kind regards,
Sean--
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--without-libedit to your configure flags. This will make
Asterisk use the bundled version of the editline library instead of the
system's:
./configure --without-libedit
Kind regards,
Sean
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On 10/24/2021 12:41 PM, cio-al...@playerschool.edu wrote:
> Really destroying SIP dialog
> '5af3bcf012ac9d574b17f2634e48de54@65.21.137.162:5060' Method: OPTIONS
> \U+26504\U+2650A\U+26565\U+26578\U+26569\U+26574\U+2650A\U+2650A
If this is still Asterisk 11 as you've mentioned in other threads, thi
On 6/30/2021 9:50 AM, Dovid Bender wrote:
> Yes that works. It's an "ugly hack". Would this be classified as a bug
> or feature?
It's an existing bug:
https://issues.asterisk.org/jira/browse/ASTERISK
ocalhost/test.gsm?foo=bar&_unused=test.gsm
Kind regards,
Sean
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New
o
modules.conf:
noload = res_adsi.so
And restart Asterisk.
Kind regards,
Sean
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On 5/12/2021 9:37 AM, Jerry Geis wrote:
> Sorry it - may have worked - my person only used a single / not //
> Thanks!
>
> Does this work on version 13 or just version 18 ?
In terms of supported versions of Asterisk it works in 16+
Kind r
14-chan_rtp(waschan_multicast_rtp)
Kind regards,
Sean--
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On 5/11/2021 4:24 PM, Jerry Geis wrote:
> I was using asterisk 13.36.0 and tried to specify a MulticastRTP TTL with
> Channel: MulticastRTP/basic/239.1.2.3:20480/5
> where 5 is the ttl
Try:
MulticastRTP/basic/239.1.2.3:20480//t(5)
Kind rega
ork, it is not referring to line 24 of the
certificate file.
Kind regards,
Sean
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https://issues.asterisk.org/jira/browse/ASTERISK-28695
On Fri, Sep 25, 2020 at 2:32 PM hw wrote:
>
>
> Hi,
>
>
>
> ever since I have switched my server from Centos 7 to Fedora 32, asterisk
>
> is showing memory issues and no calls are possible. I'm using the asterisk
>
> that comes with Fedora
On 9/24/2020 12:28 PM, hw wrote:
What are the requirements for the URLs that can be used with the
'playlist' option in musiconhold.conf?
HTTP(S)
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16.13.0, pjsip
I'd like to get an alert if a call fails to authenticate:
if "Failed to authenticate" then
mail someone the source ip
endif
As I look at ami or ari, they deal with calls in channels. Is there a
way to get failed invites or reg
On 9/7/20 3:41 PM, Joshua C. Colp wrote:
On Sat, Sep 5, 2020 at 10:23 AM sean darcy <mailto:seandar...@gmail.com>> wrote:
module load res_pjsip
Unable to load module res_pjsip
Command 'module load res_pjsip' failed.
ERROR[141535]: loader.c:281 module_load_e
On 9/6/20 3:43 AM, Michael Maier wrote:
On 05.09.20 at 15:22 sean darcy wrote:
asterisk-16.13.0-rc2. Fedora 32
pjsip won't load because of undefined symbols:
This means, that your pjsip library doesn't match the asterisk binary. It's
best to remove the independent pjsip libr
asterisk-16.13.0-rc2. Fedora 32
pjsip won't load because of undefined symbols:
[Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error
loading module 'func_pjsip_aor.so':
/usr/lib64/asterisk/modules/func_pjsip_aor.so: undefined symbol:
ast_sip_location_retrieve_aor_contacts
[Sep 4
; Set connect timeout to 5 seconds
For app_mysql, setting the connection timeout is shown in the documentation:
*CLI> core show application MYSQL
Kind regards,
Sean
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On 4/22/2020 2:55 PM, Dovid Bender wrote:
All the calls are using ulaw. The files that I am playing are gsm. I
suppose doing a file convert with sox to .ulaw may help
You should absolutely do this.
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n (default is 365 days and
some may expect a different duration) ?
I think that would be fine. If you are willing to contribute that
change, feel free to open an issue in JIRA⁴ and attach a patch, or
submit the patch for review yourself⁵.
Kind regards,
Sean
1.
https://github.com/asteris
Should be ulaw
On Wed, Apr 1, 2020 at 11:02 AM Jerry Geis wrote:
> What is the default multicast codec for multicast in Asterisk 13 ?
>
> G.729 or G.711 or other ?
>
> Jerry
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example one you provided.
Kind regards,
Sean
[1] https://issues.asterisk.org/jira
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s error in your logs?
Kind regards,
Sean
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New to Asterisk? Start he
exchanged.
Kind regards,
Sean
1. https://github.com/asterisk/asterisk/blob/16/CHANGES#L1435-L1448
2.
https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-1
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= 0.0.0.0:5061
external_media_address = 52.91.86.158
external_signaling_address = 52.91.86.158
Hope that helps,
Sean
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the request method, but you can match any message
header. Documentation can be found here:
*CLI> config show help res_pjsip_endpoint_identifier_ip identify
match_header
Kind regards,
Sean
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cert_file = /etc/letsencrypt/live/specialdomain.com/fullchain.pem
priv_key_file = /etc/letsencrypt/live/specialdomain.com/privkey.pem
Kind regards,
Sean
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On 1/22/20 11:51 AM, Michael L. Young wrote:
- Original Message -
From: "sean darcy"
To: "Asterisk Users Mailing List, Non-Commercial Discussion"
Sent: Tuesday, January 21, 2020 9:22:28 PM
Subject: [asterisk-users] permission woes on systemd
[..]
So why wou
/etc/asterisk/asterisk.conf
ExecStop=/usr/sbin/asterisk -rx 'core stop now'
ExecReload=/usr/sbin/asterisk -rx 'core reload'
PrivateTmp=true
[Install]
WantedBy=multi-user.target
So why would starting asterisk as user asterisk work, but fail using
systemd ?
Any help appreciate
generated from the source code[1], so
the best place would be the Asterisk Issues Tracker[2]. Even better
would be to attach a patch to the issue you create which beefs up the
documentation to your liking.
Kind regards,
Sean
[1] https://github.com/asterisk/asterisk/blob/17/res/res_agi.c#L321
the syntax you are looking for is:
GET FULL VARIABLE ${myVar}
Kind regards,
Sean
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is doesn't appear to be an Asterisk bug. You should probably try to
find a newer version of neon for your distribution.
Kind regards,
Sean
[1]
http://lists.manyfish.co.uk/pipermail/neon-commits/2014-Septem
On 12/14/19 11:29 AM, Greg Troxel wrote:
sean darcy writes:
There is also the ObiHai OBi202 with an OBiLine, which provides an FXO
port remoted over SIP. (I am not sure if this is discontinued.)
"FXO port remoted over SIP"?
I have an analog phone system. I can use the obi202
On 12/13/19 9:28 PM, Greg Troxel wrote:
sean darcy writes:
I'm moving asterisk to a laptop, so can't use the dahdi board. Is
there any supported USB dahdi device ? I see the Sangoma USBfxo
device, but the dahdi driver no longer supports it. Anything else ?
There is also the Obi
I'm moving asterisk to a laptop, so can't use the dahdi board. Is there
any supported USB dahdi device ? I see the Sangoma USBfxo device, but
the dahdi driver no longer supports it. Anything else ?
sean
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On 10/31/2019 2:13 PM, Carlos Chavez wrote:
I assume this is something created by Freepbx. If I do a "channel
request hangup" it tells me the channel does not exist. Any ideas
Are you trying to hang up "Message/ast_msg_queu" or are you hitting the
tab key to complete it in the CLI? "Message/a
On 8/14/19 6:00 PM, sean darcy wrote:
dahdi built fine on 5.1.20, but on 5.2.7:
.
CC [M]
/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader/dahdi_vpmadt032_loader.o
SHIPPED
/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader
ile:74: modules] Error 2
error: Bad exit status from /var/tmp/rpm-tmp.F8F4dL (%prep)
Any ideas ?
sean
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On 4/9/19 12:14 PM, George Joseph wrote:
On Tue, Apr 9, 2019 at 9:28 AM sean darcy <mailto:seandar...@gmail.com>> wrote:
On 4/8/19 6:18 AM, Joshua C. Colp wrote:
> On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote:
>> On 4/5/19 10:36 AM, sean darcy wrote:
On 4/8/19 6:18 AM, Joshua C. Colp wrote:
On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote:
On 4/5/19 10:36 AM, sean darcy wrote:
I'm trying to set up pjsip to work with an obi202 and google voice. But
I can't configure the endpoint.
pjsip:
[obi202-auth](!)
type = auth
auth_type
On 4/5/19 10:36 AM, sean darcy wrote:
I'm trying to set up pjsip to work with an obi202 and google voice. But
I can't configure the endpoint.
pjsip:
[obi202-auth](!)
type = auth
auth_type = userpass
password =
[obi202-aor](!)
type = aor
max_contacts = 2
; = endpoints ===
00
a=rtpmap:8 PCMA/8000
a=sendrecv
a=ptime:20
a=xg726bitorder:big-endian
[Apr 3 13:17:12] NOTICE[1762]: res_pjsip/pjsip_distributor.c:672
log_failed_request: Request 'INVITE' from
'@10.10.11.180>' failed for ':5062' (
callid: bb384ee02eab7054@)
s", those calls will match whatever the dialed number is.
On 03/28/2019 08:32 PM, sean darcy wrote:
I'm using "s" extension in my dialplan:
[gv-voice]
exten => s,1,Verbose(callerid is "${CALLERID(all)}" or
"${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)}
context, "s" doesn't match the
extension:
res_pjsip_session.c:2991 new_invite: Call from 'gv-voice'
(UDP:10.10.10.80:5062) to extension '' rejected because
extension not found in context 'gv-voice'.
at the end of a pattern.
I tried
[+\ ][1\ ]1234567890
which didn't work, probably because "\ " means space, not nothing.
Any suggestions?
sean
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On 2/14/19 4:23 AM, Administrator TOOTAI wrote:
Le 14/02/2019 à 00:12, sean darcy a écrit :
I'm using BLACKLIST() to check numbers, which does not like leading +
signs. I want to test if there is a plus sign, and then remove it.
I tried:
; strip leading plus sign
same => n,
0?make-em-wait") in new stack
ExecIf correctly finds the comparison false(the "0"), but still executes
the appiftrue .
What am I missing ?
--
Tried the double equal sign. Same result.
sean
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I'm using BLACKLIST() to check numbers, which does not like leading +
signs. I want to test if there is a plus sign, and then remove it.
I tried:
; strip leading plus sign
same => n, Verbose( callerid 0:1 is ${CALLERID(num):0:1} )
same => n,ExecIf($["${CALLERID(num):0:1}" = "+"]?Set(CALLE
I'm running Fedora 29. asterisk starts with a systemd service at boot.
On any reboot I get a LOT of :
[Jan 15 09:30:26] ERROR[1162]: netsock2.c:541 ast_sockaddr_hash: Unknown
address family '0'.
[Jan 15 09:30:35] ERROR[1161]: netsock2.c:541 ast_sockaddr_hash: Unknown
address family '0'.
[Jan 1
I must have missed the memo, but I was surprised to see a new DAHDI
release in downloads. Was there an announcement ? Is there a Changelog ?
Also, it seems there's no longer a wctdm module. What's the plan for the
analog TDM cards ? Or is there on
ash: Unknown address family '0'.
.
If I restart asterisk, they go away.
I don't think they're any harm other than spamming the console. Is there
any way avoid the error without a full restart ?
13.23.0
sean
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On 10/16/18 1:42 PM, Antony Stone wrote:
On Tuesday 16 October 2018 at 19:04:42, Ivan Demkovitch wrote:
Thanks all,
I did contact Callcentric about it and their tech support helped meget
those headers established. They even helped to troubleshoot Asterisk
dialplan. A the end all works as it sho
On 9/24/18 2:57 PM, John T. Bittner wrote:
Hello all,
I am having some trouble converting this setup from SIP to PJSIP. Any
help is much appreciated.
I used the converter script and get most of it but don’t see a
registration entry.
How do you convert this entry into PJSIP.
This working s
On 9/24/18 5:04 PM, John T. Bittner wrote:
Hello all,
I am having some trouble getting this to work under pjsip. Any help is
much appreciated.
I used the converter script and I see it register but can’t receive or
send to ringcentral.
Anyone get this working with PJSIP?
Works with chan_si
On 9/12/18 1:32 PM, Joshua Colp wrote:
On Wed, Sep 12, 2018, at 2:25 PM, sean darcy wrote:
On 9/12/18 1:22 PM, Joshua Colp wrote:
On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:
I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.
SIP/mycall-x
On 9/12/18 1:25 PM, sean darcy wrote:
On 9/12/18 1:22 PM, Joshua Colp wrote:
On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:
I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.
SIP/mycall-x calls and bridges with DAHDI/1-1.
I send SIP
On 9/12/18 1:22 PM, Joshua Colp wrote:
On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:
I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.
SIP/mycall-x calls and bridges with DAHDI/1-1.
I send SIP/ to listen to a long, very long, file
On 9/12/18 1:22 PM, Joshua Colp wrote:
On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:
I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.
SIP/mycall-x calls and bridges with DAHDI/1-1.
I send SIP/ to listen to a long, very long, file
I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.
SIP/mycall-x calls and bridges with DAHDI/1-1.
I send SIP/ to listen to a long, very long, file.
GoSub(play-long-file,s,1)
[play-long-file]
exten=s,1, ;;; Here I want to hangup DAHDI/1-1, t
erisk/res_stun_monitor.conf:stunaddr = stun.counterpath.net
Probably harmless, but odd.
sean
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Astricon is coming up October 9-11! Signup is available at:
I see a lot of tag lines on posts for the Asterisk Community Forum. Is
that forum supposed to supersede this mailing list ?
sean
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Astricon is coming
when I can see it more
readily.
Thanks.
On Wed, 29 Aug 2018 20:31:15 -0400,
sean darcy wrote:
On 08/29/2018 08:07 PM, John Covici wrote:
I wonder if I could have that patch, maybe I could add it to my
fail2ban regexp and if you have the correct regexp, I would apperciate
that as well.
Thanks.
On
better than nothing.
Digium warns not to use fail2ban / log trolling as a security system:
http://forums.asterisk.org/viewtopic.php?p=159984
-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of sean darcy
Sent: Wednesday, August 29, 2018 6
-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of sean darcy
Sent: Wednesday, August 29, 2018 10:46 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] getting invites to rtp ports ??
On 08/29/2018 09:42 AM, Carlos Rojas wrote:
Hi
Probably
On 08/29/2018 09:42 AM, Carlos Rojas wrote:
Hi
Probably somebody is trying to hack your system, you should block that
ip on your firewall.
Regards
On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <mailto:seandar...@gmail.com>> wrote:
I'm getting invites to very high ports ev
2e748
Call-ID: 1504207870-295758084-609228182
CSeq: 1 INVITE
...
WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on
1504207870-295758084-609228182...
I thought invites had to go to port 5060 or so. I don't understand why
somebody (let's assume a bad guy) is trying ports abov
On 05/17/2018 05:29 PM, sean darcy wrote:
On 05/17/2018 04:47 PM, Daniel Tryba wrote:
On Thu, May 17, 2018 at 12:27:17PM -0400, sean darcy wrote:
WARNING.* .*: fail2ban=''
# Option: ignoreregex
# Notes.: regex to ignore. If this regex matches, the line is ignored.
# Val
On 05/17/2018 04:47 PM, Daniel Tryba wrote:
On Thu, May 17, 2018 at 12:27:17PM -0400, sean darcy wrote:
WARNING.* .*: fail2ban=''
# Option: ignoreregex
# Notes.: regex to ignore. If this regex matches, the line is ignored.
# Values: TEXT
#
ignoreregex =
Th
On 05/17/2018 11:38 AM, Frank Vanoni wrote:
On Thu, 2018-05-17 at 11:18 -0400, sean darcy wrote:
3. How do I set up the server to block these ?
4. Can I stop the retransmitting of the 401 Unauthorized packets ?
I'm happy with Fail2Ban protecting my Asterisk 13. Here is my
configuration
T-mobile.
3. How do I set up the server to block these ?
4. Can I stop the retransmitting of the 401 Unauthorized packets ?
Any help appreciated.
sean
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On 02/06/2018 07:58 AM, Tzafrir Cohen wrote:
On Sun, Feb 04, 2018 at 03:15:02PM -0500, sean darcy wrote:
On 13.9.0
https://github.com/traud/asterisk-opus
The README:
Alternatively, you can use the Makefile of this repository to create just
the shared libraries of the modules. That way, you
On 02/03/2018 07:11 PM, sean darcy wrote:
Confused about xmpp.conf with OAuth. Let's assume I have two voice
accounts. Are all the OAuth entries in each account ? It'd be really
great if only separate refresh_token s were required!
For instance- painful:
[general]
...
On 13.9.0
https://github.com/traud/asterisk-opus
The README:
Alternatively, you can use the Makefile of this repository to create
just the shared libraries of the modules. That way, you do not have to
(re-) make your whole Asterisk.
The Makefile generates:
codecs/codec_opus_open_source.so
f
st the
refresh tokens. I hope I don't need a clientid and secret for each account!
sean
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On 12/30/2017 08:18 PM, Dovid Bender wrote:
Script kiddies trying to find vulnerable systems that they can make
calls on. Lock down the box with iptables and use fail2ban to block
them. The via is probably bogus unless a box at the DoD was comprimised.
On Sat, Dec 30, 2017 at 6:49 PM, sean
On 12/30/2017 08:10 PM, Antony Stone wrote:
On Sunday 31 December 2017 at 00:49:17, sean darcy wrote:
I've been getting a lot of timeouts on non-critical invite transactions.
So how is someone on a Dutch ISP using my server to mess with a US DoD
ip address ?
What's your s
I've been getting a lot of timeouts on non-critical invite transactions.
I turned on sip debug. They were the result of SIP invites like this:
Retransmitting #10 (NAT) to 185.107.94.10:13057:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
215.45.145.211:5060;branch=z9hG4bK-524287-1---zg4cfkl50hpwpv4
On 05/31/2016 11:43 AM, Frank Vanoni wrote:
(CALLERID(all)="Jon Doe" <+123456789>)
So simple. just too obvious.
thanks
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Thanks,
sean
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On 13.9. The cli log has these messages every 15 seconds. The end point
to linphone on android.
[May 12 19:02:59] WARNING[2555]: chan_sip.c:3775 __sip_xmit: sip_xmit of
0x7effe40088b0 (len 608) to 10.10.11.95:37855 returned -2: Success
[May 12 19:03:13] WARNING[2555]: chan_sip.c:37
opus patch for
11 that includes PLR and FEC ?
sean
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not on the opus code itself, the patches claim to allow FEC ,
which otherwise has problems, at least with HD voice :
https://github.com/seanbright/asterisk-opus/issues/9
Or am I missing something ?
sean
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the
main patch.
sean
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asterisk-users maili
o: 002 ISeqno: 002 Type: IAX Subclass: ACK
Timestamp: 00997ms SCall: 03639 DCall: 03026 [63.211.239.14:4569
<http://63.211.239.14:4569>]
I believe teliax, no longer supports IAX, sadly.
sean
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g/2449
Any other feedback? I'd like to get an idea of how many folks have
tried it.
Thanks
george
Built on fedora 23. speexdsp-devel is required. It provides speex_echo.h .
Haven't actually run it, but it built.
Thanks for all the work. This much easier. May
;t tried it with this exact setup though.
On Sun, Mar 27, 2016 at 6:13 AM, sean darcy mailto:seandar...@gmail.com>> wrote:
To connect to google voice with xmpp, I've had to turn on the "less
secure apps" switch.
You recently changed your security settings so th
ss secure
apps" ?
Is this just a way of google messing with us ?
sean
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end calls from my in-state
area codes to the 'RING' label, other calls are answered by voicemail.
On Thu, Feb 25, 2016 at 3:13 PM, sean darcy mailto:seandar...@gmail.com>> wrote:
I'd like to transfer all my pesky telemarketing calls to Jolly Roger .
http://www.nyt
er answers.
But blindtransfer requires an extension after you hear "transfer". And I
don't want the caller to hear "transfer", or hear the dialing sequence.
Any suggestions ?
sean
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On 01/29/2016 03:59 PM, Mc GRATH Ricardo wrote:
Hi Sean Darcy
Question about "the remote party always hears an echo on it's side", strange
because eco suppression circuit is for local side.
Mc GRATH Ricardo
OK. Maybe an echo canceller won't make any difference. But
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