Re: [asterisk-users] PJSIP T.38 issues
I think the 488 Not acceptable here is occurring because the channel
connecting through is not T.38 capable, that will be the IAX channel
from iaxmomdem.
I've not used PJSIP so cannot offer any advice regarding it however you
may try to make iaxmodem connect through another context using either
SIP or IAX (experiment with both bu most probably IAX) in an attempt to
prevent the rejection of the T.38 establishment forcing the call to
terminate. What I seem to recall when experimenting with SIP as the
trunk, have UDPTL disabled i.e. t38pt_udptl=no, this would also induce
488 Not acceptable here.
Looking at a legacy configuration where I tested iaxmodem
(context=faxgateway-iax) going through Asterisk 1.2 which then forwarded
the request to Asterisk 11 (context=FAX-T30) where it then went out
through the trunk with Fax Gateway enabled.
In short;
Asterisk 1.2
IAX Modem in context faxgateway-iax, could change to faxgateway-sip.
[faxgateway-iax]
; Incoming calls from iaxmodem to Asterisk are directed to
; other Asterisk server.
exten = _XX.,1,Dial(IAX2/faxgw-iax@faxgw-iax/${EXTEN},55,t)
exten = _XX.,n,Wait(1)
exten = _XX.,n,Hangup
;
[faxgateway-sip]
; Incoming calls from iaxmodem to Asterisk are directed to
; other Asterisk server.
exten = _XX.,1,Dial(SIP/${EXTEN}@faxgw-sip,55,t)
exten = _XX.,n,Wait(1)
exten = _XX.,n,Hangup
;
Asterisk 11
IAX user faxgw-iax is in context FAX-T30
extensions.ael on Asterisk 11 contains
context FAX-T30 {
snip
_ = {
// Set(FAXOPT(t38gateway)=yes);
Dial(SIP/${EXTEN}@itsp-fax,55);
Hangup();
};
snip
};
One other note, enable alaw ulaw in iaxmomdem and your iax peer
configuration in Asterisk, just to be sure!
I know this isn't specific to your case but maybe you can make something
from this that helps.
Please note, I don't have the old set up to test so I can't be certain
of the above configurations.
Cheers,
Larry.
On 27/07/2015 11:15 AM, Jean-Denis Girard wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi list,
2 weks ago I asked questions about PJSIP and T.38 but got no replies. I
upgraded Asterisk to git as of yesterday (309dd2a), and I'm still having
the same issues.
In the trace below, I'm sending a fax from Hylafax server through
iaxmodem on Asterisk-13 (tiare) to a second Asterisk (11.18.0 t0gw)
connected to the PSTN via ISDN; the call is to my test fax machine,
connected to the PSTN. chan_pjsip is used on Asterisk-13, and chan_sip
is used on Asterisk-11.
This how endpoint t0gw (Asterisk-11) is configured on tiare (Asterisk-13
):
tiare*CLI pjsip show endpoint t0gw
...
t38_udptl : true
t38_udptl_ec : fec
t38_udptl_ipv6 : false
t38_udptl_maxdatagram : 400
t38_udptl_nat : false
...
Could someone explain why I'm getting Not acceptable below?
-- Accepting AUTHENTICATED call from 127.0.0.1:4570:
-- requested format = slin,
-- requested prefs = (),
-- actual format = slin,
-- host prefs = (slin),
-- priority = mine
-- Executing [40ZZ@fax-sortant:1] NoOp(IAX2/iaxmodem0-7838,
calls 40ZZ (local)) in new stack
-- Executing [40ZZ@fax-sortant:2] Set(IAX2/iaxmodem0-7838,
FAXOPT(gateway)=yes) in new stack
-- Executing [40ZZ@fax-sortant:3] Dial(IAX2/iaxmodem0-7838,
PJSIP/40ZZ@t0gw) in new stack
-- Called PJSIP/40ZZ@t0gw
--- Transmitting SIP request (936 bytes) to UDP:192.168.0.10:5060 ---
INVITE sip:40zzz...@gw.sysnux.pf SIP/2.0
Via: SIP/2.0/UDP
192.168.0.200:5060;rport;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e3
8e5f1
From: SysNux
sip:+68940XX@192.168.0.200;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: sip:40zzz...@gw.sysnux.pf
Contact: sip:63035284-ad7d-484f-8e54-f5ea54f39104@192.168.0.200:5060
Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
CSeq: 31693 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk GPL PBX
Content-Type: application/sdp
Content-Length: 238
v=0
o=- 1710591484 1710591484 IN IP4 192.168.0.200
s=Asterisk
c=IN IP4 192.168.0.200
t=0 0
m=audio 8834 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
--- Received SIP response (585 bytes) from UDP:192.168.0.10:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.200:5060;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e38e5f1;
received=192.168.0.200;rport=5060
From: SysNux
sip:+68940XX@192.168.0.200;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: sip:40zzz...@gw.sysnux.pf
Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
CSeq: 31693 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact:
Re: [asterisk-users] Filters
On Mon, Jul 27, 2015 at 4:51 AM, Stefan Viljoen viljo...@verishare.co.za wrote: Hi list I'm using Asterisk 1.8.11.0 - is there any way to apply (for example) a bandpass filter to Asterisk RTP audio in the realtime audio stream? I'm looking for a way to (for example) filter out a 50Hz AC hum present in some calls I push through my asterisk. Thanks If you're willing to write C, then yes, what you're looking to do is possible. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] No audio on SIP over WebRTC
I'm following this tutorial (
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5) to
deploy WebRTC support but I'm having an issue with RTP when the WebRTC
softphone is behind NAT.
In my scenario, the Asterisk server is running a public IPv4, and the
softphone is behind NAT. I can register and make a call normally, but I
don't get any audio in neither way (Asterisk/softphone and
softphone/Asterisk). Using the very same config files but having the
softphone and Asterisk on the same network it works fine.
Any tips on how to solve this? Here's my relevant files.
*;sip.conf:*
[general]
udpbindaddr=0.0.0.0:5060
realm=10.201.0.106 ;replace with your Asterisk server public IP address or
host
transport=udp,ws,wss
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
[6000]
host=dynamic
secret=mysecret
context=default
type=friend
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
qualify=yes
[6001]
host=dynamic
secret=mysecret
context=default
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
*extensions.conf:*
[default]
exten = _6XXX,1,Dial(SIP/${EXTEN})
*rtp.conf:*
[general]
rtpstart=1
rtpend=2
icesupport=yes
stunaddr=stun.l.google.com:19302
--
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[asterisk-users] Why no CentOS 7 repos?
Any particular reason CentOS 7 repos aren't available? I'm finding integration issues with CentOS 6's ancient versions of MySQL and PHP with third party applications. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- 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 11.19.0-rc1 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.19.0-rc1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.19.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
---
* ASTERISK-25250 - chan_sip - Despite the channel being answered,
caller on a call established via Local channel continues to hear
ringback (Reported by Etienne Lessard)
* ASTERISK-25247 - choppy audio when spying on a g722 channel,
chan_sip or chan_pjsip (Reported by hristo)
* ASTERISK-24853 - Documentation claims chan_sip outbound
registrations support WS or WSS as valid transports (not true)
(Reported by PSDK)
* ASTERISK-25257 - [patch]channels/sig_pri.h - sig_pri_span -
force_restart_unavailable_chans in wrong scope (Reported by
Patric Marschall)
* ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
(Reported by Rusty Newton)
* ASTERISK-22805 - res_rtp_asterisk: Crash when calling
BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
(Reported by Dmitry Burilov)
* ASTERISK-24550 - res_rtp_asterisk: Crash in
ast_rtp_on_ice_complete during DTLS handshake (Reported by
Osaulenko Alexander)
* ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
Badalian Vyacheslav)
* ASTERISK-24832 - [patch]DTLS-crashes within openssl (Reported
by Stefan Engström)
* ASTERISK-25127 - DTLS crashes following Unable to cancel
schedule ID in dtls_srtp_check_pending (Reported by Dade
Brandon)
* ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
INVITE early Replace code (Reported by Walter Doekes)
* ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
(Reported by Walter Doekes)
* ASTERISK-25219 - [patch]Source and destination overlap in memcpy
in rtp_engine.c (Reported by Walter Doekes)
* ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
(Reported by Walter Doekes)
* ASTERISK-19277 - [patch]endlessly repeating error: poll failed:
Bad file descriptor (Reported by Barry Chern)
* ASTERISK-25202 - Hints extension state broken between 13.3.2 and
13.4 (Reported by cervajs)
* ASTERISK-25154 - [patch]fromtag may need to be updated after
successful call dialog match (Reported by Damian Ivereigh)
* ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
(Reported by Gregory Massel)
* ASTERISK-25094 - PBX core: Investigate thread safety issues
(Reported by Corey Farrell)
* ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
but asterisk doesn't detect it. (Reported by ibercom)
* ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
| adpcm | ipc10} (Reported by Badalian Vyacheslav)
* ASTERISK-25100 - asterisk coredump if host has an IPv6 address
that end with ::80 (Reported by Mark Petersen)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0-rc1
Thank you for your continued support of Asterisk!
--
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Re: [asterisk-users] Why no CentOS 7 repos?
On 27/07/2015 1:51 PM, Steve Edwards wrote: Any particular reason CentOS 7 repos aren't available? I'm finding integration issues with CentOS 6's ancient versions of MySQL and PHP with third party applications. You might have o upgrade MySQL and PHP outside of the Centos distribution. I have Centos 6 with MySQL 5.1.73 and PHP 5.3.3 with FreePBX 2.11. Ron -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Why no CentOS 7 repos?
On 27/07/2015 1:51 PM, Steve Edwards wrote: Any particular reason CentOS 7 repos aren't available? I'm finding integration issues with CentOS 6's ancient versions of MySQL and PHP with third party applications. On Mon, 27 Jul 2015, Ron Wheeler wrote: You might have o upgrade MySQL and PHP outside of the Centos distribution. I have Centos 6 with MySQL 5.1.73 and PHP 5.3.3 with FreePBX 2.11. I really prefer to keep to the repos. It's a numbers thing: ) I don't want to spend the time (aka $$$) to track patches to packages. ) I don't want to be 'different.' I want to run the same versions as others so I don't get to discover and resolve incompatibilities all by myself. CentOS 7 was released over a year ago. Seems overdue to me. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why no CentOS 7 repos?
On 27/07/2015 2:38 PM, Steve Edwards wrote: On 27/07/2015 1:51 PM, Steve Edwards wrote: Any particular reason CentOS 7 repos aren't available? I'm finding integration issues with CentOS 6's ancient versions of MySQL and PHP with third party applications. On Mon, 27 Jul 2015, Ron Wheeler wrote: You might have o upgrade MySQL and PHP outside of the Centos distribution. I have Centos 6 with MySQL 5.1.73 and PHP 5.3.3 with FreePBX 2.11. I really prefer to keep to the repos. It's a numbers thing: ) I don't want to spend the time (aka $$$) to track patches to packages. ) I don't want to be 'different.' I want to run the same versions as others so I don't get to discover and resolve incompatibilities all by myself. CentOS 7 was released over a year ago. Seems overdue to me. I absolutely agree with your reasoning but sometimes the realities of using open source means that some of the money that you save on licensing has to be spent on support or worse (pioneering). If you use PHP and MySQL from a repo, Yum will track and install updates. https://dev.mysql.com/downloads/repo/yum/ https://webtatic.com/projects/yum-repository/ I am not sure how you can be sure to match up with others since everyone has their own tolerance to change and will often let a sleeping dog lie until it bites them. Your Centos 6 (or 7) will have much more up to date versions of MySQL and PHP that mine since mine was running in a stable configuration of OS, PHP and database over a year ago. I only discovered the MySQL and PHP repos after I had a stable Asterisk (which has not bit bitten me yet) so I have only used these on Centos 7 for other servers not related to Asterisk. I hope that this helps. Ron -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Filters
Hi list I'm using Asterisk 1.8.11.0 - is there any way to apply (for example) a bandpass filter to Asterisk RTP audio in the realtime audio stream? I'm looking for a way to (for example) filter out a 50Hz AC hum present in some calls I push through my asterisk. Thanks Stefan -- _ -- 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
