Some updates distroyed our Telephone conference service


we have a conference server running using asterisk hat is connected to a PBX using SIP and after some updates (guess debian dist-upgrade without an restart) we restarted the system and now it is not working anymore.

==> /var/log/asterisk/messages <==
[Jan  4 09:25:12] WARNING[1831][C-00000001] chan_sip.c: Declining non-primary audio stream: audio 50002 RTP/AVP 109 9 8 0 18 101 13
[Jan  4 09:25:12] WARNING[1831][C-00000001] chan_sip.c: Can't provide secure audio requested in SDP offer
[Jan  4 09:25:27] WARNING[2463][C-00000002] app_confbridge.c: ConfBridge requires an argument (conference name[,options])

any idea what that means and how to fix it?

here are some version information bout the current system

/ # lsb_release -a
No LSB modules are available.
Distributor ID: Debian
Description:    Debian GNU/Linux 10 (buster)
Release:        10
Codename:       buster
/ # dpkg -s asterisk
Package: asterisk
Status: install ok installed
Priority: optional
Section: comm
Installed-Size: 7682
Maintainer: Debian VoIP Team <>
Architecture: amd64
Version: 1:16.28.0~dfsg-0+deb10u1
Provides: asterisk-1fb7f5c06d7a2052e38d021b3d8ca151
Depends: adduser, asterisk-config (= 1:16.28.0~dfsg-0+deb10u1) | asterisk-config-custom, asterisk-core-sounds-en, asterisk-modules (= 1:16.28.0~dfsg-0+deb10u1), lsb-base (>= 3.0-6), libc6 (>= 2.27), libcap2 (>= 1:2.10), libedit2 (>= 2.11-20080614-0), libjansson4 (>= 2.11), libpopt0 (>= 1.14), libsqlite3-0 (>= 3.7.15), libssl1.1 (>= 1.1.1), libsystemd0, liburiparser1 (>= 0.6.0), libuuid1 (>= 2.16), libxml2 (>= 2.7.4), libxslt1.1 (>= 1.1.25)
Recommends: asterisk-moh-opsound-gsm, asterisk-voicemail | asterisk-voicemail-storage, sox
Suggests: asterisk-dahdi, asterisk-dev, asterisk-doc, asterisk-ooh323, asterisk-opus, asterisk-vpb
 /etc/default/asterisk e2a443d3b806fd294290654ab64f0faa
 /etc/init.d/asterisk b226b626a6deb5a9a4d8500facd42bb2
 /etc/logrotate.d/asterisk d52ae3c0fd7a6f735900a82c951b03a2
Description: Open Source Private Branch Exchange (PBX)
 Asterisk is an Open Source PBX and telephony toolkit.  It is, in a
 sense, middleware between Internet and telephony channels on the bottom,
 and Internet and telephony applications at the top.
 Asterisk can be used with Voice over IP (SIP, H.323, IAX and more) standards,
 or the Public Switched Telephone Network (PSTN) through supported hardware.
 Supported hardware:
  * All Wildcard (tm) ISDN PRI cards from Digium (
  * HFC-S/HFC-4S-based ISDN BRI cards (Junghanns.NET, beroNet, Digium etc.)
  * All TDM (FXO/FXS) cards from Digium
  * Various clones of Digium cards such as those by OpenVox
  * Xorcom Astribank USB telephony adapter (
  * Voicetronix OpenPCI, OpenLine and OpenSwitch cards
  * CAPI-compatible ISDN cards (using the add-on package chan-capi)
  * Full Duplex Sound Card (ALSA or OSS) supported by Linux
  * Tormenta T1/E1 card (
  * QuickNet Internet PhoneJack and LineJack (
 This is the main package that includes the Asterisk daemon and most channel
 drivers and applications.
/ # dpkg -s asterisk-dahdi
Package: asterisk-dahdi
Status: install ok installed
Priority: optional
Section: comm
Installed-Size: 2040
Maintainer: Debian VoIP Team <>
Architecture: amd64
Source: asterisk
Version: 1:16.28.0~dfsg-0+deb10u1
Replaces: asterisk-modules (<< 1:11.6.0~dfsg-2)
Depends: asterisk (= 1:16.28.0~dfsg-0+deb10u1), dahdi, libc6 (>= 2.16), libopenr2-3 (>= 1.3.2), libpri1.4 (>= 1.4.15), libss7-2.0 (>= 2.0.0), libtonezone2.0 (>= 1:
Breaks: asterisk-modules (<< 1:11.6.0~dfsg-2)
Description: DAHDI devices support for the Asterisk PBX
 Asterisk is an Open Source PBX and telephony toolkit.
 This package includes the DAHDI channel driver ( and a number of
 other Asterisk modules that require DAHDI support. They will not be useful
 without kernel-level DAHDI support.
 For more information about the Asterisk PBX, have a look at the Asterisk
/ # dpkg -s asterisk-opus
Package: asterisk-opus
Status: install ok installed
Priority: optional
Section: comm
Installed-Size: 74
Maintainer: Debian VoIP Team <>
Architecture: amd64
Version: 13.7+20171009-2
Depends: asterisk, asterisk-1fb7f5c06d7a2052e38d021b3d8ca151, libc6 (>= 2.14), libopus0 (>= 1.1), libopusfile0 (>= 0.5)
Description: opus module for Asterisk
 Module for the Asterisk open source PBX which allows you to use the
 Opus audio codec.
 Opus is the default audio codec in WebRTC. WebRTC is available in
 Asterisk via SIP over WebSockets (WSS). Nevertheless, Opus can be used
 for other transports (UDP, TCP, TLS) as well. Opus supersedes previous
 codecs like CELT and SiLK. Furthermore in favor of Opus, other
 open-source audio codecs are no longer developed, like Speex, iSAC,
 iLBC, and Siren. If you use your Asterisk as a back-to-back user agent
 (B2BUA) and you transcode between various audio codecs, one should
 enable Opus for future compatibility.
 Opus is not only supported for pass-through but can be transcoded as
 well. This allows you to translate to/from other audio codecs like
 those for landline telephones (ISDN: G.711; DECT: G.726-32; and HD:
 G.722) or mobile phones (GSM, AMR, AMR-WB, 3GPP EVS).
/ #

thanks a lot for any help in advance,


chan_sip is deprecated, and effectively unsupported. In fact it has now been dropped from the master branch on github, in preparation for its removal in Asterisk 22.

The first warning is because the caller is trying to send more than one audio stream but chan_sip has never supported that.

The second one is an incompatible encryption configuration, but you haven’t provided both the the protocol log (sip set debug on) and your sip.conf, so it is not possible to be more precise.

The third one indicates an error in your dialplan, but you haven’t provided your dialplan, and haven’t set a high enough verbosity level for the logging, so we can’t work out what is in your dialplan.

Starting with the first warning, this could be a change in certificates, assuming you were accepting encrypted audio streams before. You might try renewing the certs and then rebooting again.

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