Package Details: asterisk 18.4.0-2

Git Clone URL: (read-only, click to copy)
Package Base: asterisk
Description: A complete PBX solution
Upstream URL:
Licenses: GPL
Submitter: pressh
Maintainer: nkukard (alerque)
Last Packager: nkukard
Votes: 106
Popularity: 0.088235
First Submitted: 2008-01-27 13:42
Last Updated: 2021-06-24 00:48

Pinned Comments

alerque commented on 2020-02-22 06:10

PSA: I've started hosting this and all its dependencies as prebuilt packages (x86_86 only) in my repository for those that want to install them using pacman without messing around with building from the AUR.

Latest Comments

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bberlin commented on 2018-02-14 21:28

@caleb, I was able to get it to compile with your argument after installing libsrtp0 from AUR. Looks like it wants libsrtp version 1.x on there like @albertvaka says and doesn't need the latest 1:2.1.0. I hadn't noticed his comment before. I have pjproject 2.7.1-1 like you and I do not have bcg729 installed, not sure why you wouldn't be able to compile the other way.

alerque commented on 2018-02-14 19:41

@bberlin and @d9jWbb42kC3 just a thought, what versions of pjproject do you have installed when you tried and failed to build the way I have this packaged? Maybe a version needs to be added to that dependency. I've got pjproject 2.7.1-1 out of the current AUR.

alerque commented on 2018-02-14 09:30

@bberlin and @d9jWbb42kC3 I am following your comments but unsure what to do here. I have tried and tried again to compile this package without that flag and can't do it across any of my 4 systems. I get this error:

   [LD] abstract_jb.o acl.o adsi.o alaw.o alertpipe.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o astobj2_container.o astobj2_hash.o astobj2_rbtree.o audiohook.o autochan.o autoservice.o backtrace.o bridge.o bridge_after.o bridge_basic.o bridge_channel.o bridge_roles.o bucket.o callerid.o ccss.o cdr.o cel.o channel.o channel_internal_api.o chanvars.o cli.o codec.o codec_builtin.o config.o config_options.o conversions.o core_local.o core_unreal.o crypt.o datastore.o db.o devicestate.o dial.o dns.o dns_core.o dns_naptr.o dns_query_set.o dns_recurring.o dns_srv.o dns_system_resolver.o dns_test.o dns_tlsa.o dnsmgr.o dsp.o endpoints.o enum.o event.o features.o features_config.o file.o fixedjitterbuf.o format.o format_cache.o format_cap.o format_compatibility.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o iostream.o jitterbuf.o json.o loader.o lock.o logger.o manager.o manager_bridges.o manager_channels.o manager_endpoints.o  manager_mwi.o manager_system.o max_forwards.o md5.o media_cache.o media_index.o message.o mixmonitor.o named_acl.o named_locks.o netsock.o netsock2.o optional_api.o parking.o pbx.o pbx_app.o pbx_builtins.o pbx_functions.o pbx_hangup_handler.o pbx_ignorepat.o pbx_include.o pbx_sw.o pbx_switch.o pbx_timing.o pbx_variables.o pickup.o plc.o poll.o presencestate.o privacy.o rtp_engine.o say.o sched.o sdp.o sdp_options.o sdp_srtp.o sdp_state.o sdp_translator.o security_events.o sem.o sha1.o sip_api.o slinfactory.o smoother.o sorcery.o sounds_index.o srv.o stasis.o stasis_bridges.o stasis_cache.o stasis_cache_pattern.o stasis_channels.o stasis_endpoints.o stasis_message.o stasis_message_router.o stasis_system.o stdtime/localtime.o strcompat.o stream.o stringfields.o strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o threadpool.o threadstorage.o timing.o translate.o udptl.o ulaw.o uri.o utils.o uuid.o version.o xml.o xmldoc.o  -> asterisk
./ undefined reference to `initBcg729EncoderChannel' 
./ undefined reference to `bcg729Decoder'
./ undefined reference to `bcg729Encoder'
./ undefined reference to `initBcg729DecoderChannel'
./ undefined reference to `closeBcg729EncoderChannel'
./ undefined reference to `closeBcg729DecoderChannel'

I have the same version of libsrtp as you do, and I've tried it with and without the AUR packages for bcg729 and pjproject.

Any ideas on what other packages or system configs might be playing into this? Perhaps we should be comparing full outputs from ./configure?

bberlin commented on 2018-02-14 01:25

Same exact situation as @d9jWbb42kC3 with libsrtp 1:2.1.0-1. Also took out --without-pjproject-bundled to get it to compile.

d9jWbb42kC3 commented on 2018-02-14 00:31

   [CC] pjsip/dialplan_functions.c -> pjsip/dialplan_functions.o
   [CC] pjsip/cli_commands.c -> pjsip/cli_commands.o
   [LD] chan_pjsip.o pjsip/dialplan_functions.o pjsip/cli_commands.o ->
/usr/bin/ld: cannot find -lsrtp
collect2: error: ld returned 1 exit status
make[1]: *** [/build/yaourt-tmp-admin/aur-asterisk/src/asterisk-15.2.0/Makefile.rules:178:] Error 1
make: *** [Makefile:377: channels] Error 2
==> ERROR: A failure occurred in build().
==> ERROR: Makepkg was unable to build asterisk.

If I compile asterisk without flag --without-pjproject-bundled, everything is ok. Installed extra/libsrtp is 1:2.1.0-1. Arch: x86_64

alerque commented on 2018-02-13 12:10

I've successfully compiled 15.2 for my systems, but only with something that looks like a regression in packaging. I'm not sure it's actually a bad thing or not, but it's the only way I can get 15.2 up and running. I added pjproject back to the dependency list and compiled with --without-pjproject-bundled.

If anybody has a problem with this arrangement and/or knows why the internal version throws bcg729 errors when compiling then I'm happy to hear it.

alerque commented on 2018-02-11 15:21

@albertvaka Could you contact me with any more details on how you got this to work. I have a production phone system that's offline with a broken asterisk install and am trying to get this fixed up. I also just adopted this package and will try to get it working better for everybody. I updated it to the last version, but I'm getting compile time errors ... the error you link to I get with 15.1.x, but on 15.2.x I'm seeing a different problem related to pj and/or bcg729. I'd be interested in hearing about what worked for you, either here, via email or if you want to collaborate I just mirrored this to Github:

albertvaka commented on 2018-02-06 05:18

The current version only builds with libsrtp version 1.

If you have the latest libsrtp installed, it fails with this:

aztec102 commented on 2017-11-02 16:09

{standard input}: Assembler messages:
{standard input}: Fatal error: can't write 159 bytes to section .debug_macro of sip/route.o because: 'No space left on device'
{standard input}: Fatal error: can't close sip/route.o: No space left on device

xdevla commented on 2017-10-09 11:48

Was on my todo to remove it.
Might be something wrong on my side but I still get the same build error as in 14.x when using internal libedit.