Re: [asterisk-users] Asterisk 18 won't transcode DTMF to inband

On Wed, 2021-10-20 at 06:44 -0300, Joshua C. Colp wrote: > > Should I download and compile this instead? > > > > http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz > > If you want to be running Asterisk 18 and a known released version, yes. Right OK thanks, I'll do

Re: [asterisk-users] Asterisk 18 won't transcode DTMF to inband

On Wed, Oct 20, 2021 at 5:20 AM Kingsley Tart - Barritel Ltd < kingsley.t...@barritel.com> wrote: > On Tue, 2021-10-19 at 15:02 -0300, Joshua C. Colp wrote: > > # asterisk -V > > > Asterisk GIT-master-cc127a999cM > > > # > > > > That's the master branch from around March or so, not 18. > > Wow,

Re: [asterisk-users] Asterisk 18 won't transcode DTMF to inband

On Tue, 2021-10-19 at 15:02 -0300, Joshua C. Colp wrote: > # asterisk -V > > Asterisk GIT-master-cc127a999cM > > # > > That's the master branch from around March or so, not 18. Wow, all this time I thought I was running 18! What version would it be? How can I tell? Should I download and compile

Re: [asterisk-users] Asterisk 18 won't transcode DTMF to inband

On Tue, 2021-10-19 at 15:02 -0300, Joshua C. Colp wrote: > # asterisk -V > > Asterisk GIT-master-cc127a999cM > > # > > That's the master branch from around March or so, not 18. Wow, all this time I thought I was running 18! What version would it be? How can I tell? Should I download and compile

Re: [asterisk-users] Asterisk 18 won't transcode DTMF to inband

On Tue, Oct 19, 2021 at 2:58 PM Kingsley Tart - Barritel Ltd < kingsley.t...@barritel.com> wrote: > Thanks. > > I tried to find the precise version but I got stuck at this point: > > # asterisk -V > Asterisk GIT-master-cc127a999cM > # > > Within /usr/src/asterisk I tried this > > # grep -riE

Re: [asterisk-users] Asterisk 18 won't transcode DTMF to inband

Thanks. I tried to find the precise version but I got stuck at this point: # asterisk -V Asterisk GIT-master-cc127a999cM # Within /usr/src/asterisk I tried this # grep -riE 'version\b *18' . # but it didn't match any lines. So I'm not quite sure what actual version this is. Any idea how I can

Re: [asterisk-users] Asterisk 18 won't transcode DTMF to inband

On Tue, Oct 19, 2021 at 11:46 AM Kingsley Tart wrote: > I forgot to mention that pjsip.conf for this endpoint (that doesn't > support telephone-event) already has this: > > dtmf_mode=auto > What version of 18? Have you enabled Asterisk debug[1] and looked to see if it showed anything? Does Echo

Re: [asterisk-users] Asterisk 18 won't transcode DTMF to inband

I forgot to mention that pjsip.conf for this endpoint (that doesn't support telephone-event) already has this: dtmf_mode=auto Cheers, Kingsley. On Tue, 2021-10-19 at 15:19 +0100, Kingsley Tart wrote: > Hi, > > I'm using Asterisk 18 to receive a call via SIP, dial a different SIP > destination

[asterisk-users] Asterisk 18 won't transcode DTMF to inband

Hi, I'm using Asterisk 18 to receive a call via SIP, dial a different SIP destination and bridge them together. However, even if the destination indicates that it doesn't support telephone-event, Asterisk is still sending DTMF as events, not transcoding to inband. Asterisk is recognising inband

[asterisk-users] Asterisk 16 Realtime outbound register

Hi. I'm using Asterisk 16 with a MySQL realtime DB, containing both outbound registrations to other PBXs (Asterisk as SIP client) and inbound accounts for clients to register to (Asterisk as SIP server). All in all, working well. However, I just had a requirement to register outbound to a PBX

[asterisk-users] Asterisk 18.7.1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.7.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.7.1 resolves an issue reported by the community and would have not been

[asterisk-users] Asterisk 16.21.1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.21.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.21.1 resolves an issue reported by the community and would have not been

[asterisk-users] Asterisk 18.7.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.7.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.7.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.21.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.21.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.21.0 resolves several issues reported by the community and would have not

Re: [asterisk-users] Asterisk stops loading include files it file is missing

On 9/5/2021 5:10 PM, Dovid Bender wrote: > Hi, > > If I have in extensions.conf includes to files that does not exist > Asterisk stop loading all other files. Say for instance I have: > > #include one.conf ; Exists > #include two.conf ; Does not exist > #include three.conf ; Exists > > If two.conf

Re: [asterisk-users] Asterisk stops loading include files it file is missing

Hi Le 05/09/2021 à 23:10, Dovid Bender a écrit : Hi, If I have in extensions.conf includes to files that does not exist Asterisk stop loading all other files. Say for instance I have: #include one.conf ; Exists #include two.conf ; Does not exist #include three.conf ; Exists If two.conf

[asterisk-users] Asterisk stops loading include files it file is missing

Hi, If I have in extensions.conf includes to files that does not exist Asterisk stop loading all other files. Say for instance I have: #include one.conf ; Exists #include two.conf ; Does not exist #include three.conf ; Exists If two.conf does not exist, even if three.conf exists, asterisk will

Re: [asterisk-users] Asterisk sound file cache expiration

Hi, I found https://markmail.org/message/xh5sbqvsgwywrjje#query:+page:1+mid:vbzl4hup6jawrzup+state:results and it seems to have answered my question. By adding an expires value to the response Asterisk seems to cache it for the time specified. On Sun, Sep 5, 2021 at 6:23 AM Dovid Bender wrote:

[asterisk-users] Asterisk sound file cache expiration

Hi, Is there any way to set the default expiration for the media cache? After looking at the sqlite3 db it seems that asterisk by default sets the expiration to the time that the file was accessed so the file is never cached locally and everytime the file is played, it's downloaded again. --

[asterisk-users] Asterisk PJSIP Presence/Subscription Setup with Cisco and Grandstream Phones

Hi, I'm trying to get presence/subscription working with Asterisk 18.6.0 using PJSIP and having no success so far. My setup is: Asterisk -> Upstream VoIP Provider (this part works fine) Connectivity on the LAN is IPv6, with TCP based signalling. TLS is not enabled. The server is locally

Re: [asterisk-users] Asterisk PJSIP Presence/Subscription Setup with Cisco and Grandstream Phones

On Sat, Aug 14, 2021 at 10:36 AM Reuben Farrelly < reuben-asterisk-us...@reub.net> wrote: > Logs show: > > Aug 14 22:54:41] ERROR[26606] res_pjsip.c: Could not create dialog with > endpoint 1001. Invalid URI (PJSIP_EINVALIDURI) > [Aug 14 22:54:41] WARNING[26606] res_pjsip_pubsub.c: Unable to

[asterisk-users] asterisk 18 logger

Hi I saw the example of asterisk18 log as follows: logger.conf.sample If the verbose level is not specified, it ; will log verbose messages following the current level of the root console. so my asterisk CLI> core show settings is Root console verbosity: 3 my logger.conf change settings

[asterisk-users] Asterisk 18.6.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.6.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.6.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.20.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.20.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.20.0 resolves several issues reported by the community and would have not

[asterisk-users] Asterisk 13.38.3, 16.19.1, 17.9.4, 18.5.1 and 16.8-cert10 Now Available (Security)

The Asterisk Development Team would like to announce security releases for Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and 16.8-cert10. These releases are available for immediate download at

Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

On Sat, Jul 3, 2021 at 3:37 PM Jonas Kellens wrote: > Hello Joshua > > > could it be a bug ? > > I am using asterisk-certified-13.21-cert6 > There's been no other reports of issues, but it could be. The 13 branch, however, only receives bug fixes. Additionally the chan_sip module is community

Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

Hello Joshua could it be a bug ? I am using asterisk-certified-13.21-cert6 Kind regards. J. Op 01-07-21 om 20:20 schreef Joshua C. Colp: On Thu, Jul 1, 2021 at 3:15 PM Jonas Kellens > wrote: Hello Joshua this is the SIP REGISTER at 11:10:45

[asterisk-users] Asterisk behind kamailio proxy - contact acl

Hello, I have asterisk behind kamailio, and i want to enable acl for clients that are connecting through the proxy. Basically when i go to asterisk and type "sip show peers" i seem them connected using the proxy IP. I saw in the headers that in the contact header is the IP of user and want to

Re: [asterisk-users] Asterisk / pjsip: RTP - alaw - a=silenceSupp:off

On 30.06.21 at 23:17 Joshua C. Colp wrote: > On Wed, Jun 30, 2021 at 1:36 PM Michael Maier wrote: > >> >> Hello! >> >> Short question: Is it possible to set >> >> a=silenceSupp:off >> >> in the SDP for alaw / ulaw for fax calls? >> > > No. Thanks for your kindly support! Michael --

Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

On Thu, Jul 1, 2021 at 3:15 PM Jonas Kellens wrote: > Hello Joshua > > this is the SIP REGISTER at 11:10:45 > > REGISTER sip:tstv7.domain.tld SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.18:5060 > ;branch=z9hG4bKcfd4c09d669ce595351ea3b2aba3e245;rport > From: > ;tag=3630891428 > To: > > Call-ID:

Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

Hello Joshua this is the SIP REGISTER at 11:10:45 REGISTER sip:tstv7.domain.tld SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bKcfd4c09d669ce595351ea3b2aba3e245;rport From: ;tag=3630891428 To: Call-ID: 3270725701@192_168_1_18 CSeq: 452 REGISTER Contact: Authorization: Digest

Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

On Thu, Jul 1, 2021 at 12:34 PM Jonas Kellens wrote: > Hello Joshua > > > these are the 2 previous events on the Manager interface : > > 2021-06-30 11:10:45 > Array > ( > [0] => Event: PeerStatus > [1] => Privilege: system,all > [2] => SystemName: tstv7 > [3] => ChannelType: SIP

Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

Hello Joshua these are the 2 previous events on the Manager interface : 2021-06-30 11:10:45 Array (     [0] => Event: PeerStatus     [1] => Privilege: system,all     [2] => SystemName: tstv7     [3] => ChannelType: SIP     [4] => Peer: SIP/testacc7700921     [5] => PeerStatus: Registered    

Re: [asterisk-users] Asterisk / pjsip: RTP - alaw - a=silenceSupp:off

On Wed, Jun 30, 2021 at 1:36 PM Michael Maier wrote: > > Hello! > > Short question: Is it possible to set > > a=silenceSupp:off > > in the SDP for alaw / ulaw for fax calls? > No. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and

[asterisk-users] Asterisk / pjsip: RTP - alaw - a=silenceSupp:off

Hello! Short question: Is it possible to set a=silenceSupp:off in the SDP for alaw / ulaw for fax calls? Thanks Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

On Wed, Jun 30, 2021 at 3:28 PM Jonas Kellens wrote: > Hello > > > I see the following event from the Asterisk Manager : > > 2021-06-30 11:20:55 > Array > ( > [0] => Event: PeerStatus > [1] => Privilege: system,all > [2] => SystemName: tstv7 > [3] => ChannelType: SIP > [4] =>

[asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

Hello I see the following event from the Asterisk Manager : 2021-06-30 11:20:55 Array (     [0] => Event: PeerStatus     [1] => Privilege: system,all     [2] => SystemName: tstv7     [3] => ChannelType: SIP     [4] => Peer: SIP/testacc7700921     [5] => PeerStatus: Unregistered     [6] =>

[asterisk-users] Asterisk 18.5.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.5.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.5.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.19.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.19.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.19.0 resolves several issues reported by the community and would have not

Re: [asterisk-users] asterisk 18 and not download janson

On Wed, May 12, 2021 at 4:53 PM Jerry Geis wrote: > using 18.4.0 > > I did a ./configure --with-pjproject-bundled --with-jansson-bundled > and I still saw a message about downloading . > > I have sights that cannot download - how can I "configure" to just use the > bundled and not download

[asterisk-users] asterisk 18 and not download janson

using 18.4.0 I did a ./configure --with-pjproject-bundled --with-jansson-bundled and I still saw a message about downloading . I have sights that cannot download - how can I "configure" to just use the bundled and not download anything? Jerry --

Re: [asterisk-users] Asterisk 18 in chan_sip mode

On Wed, May 12, 2021 at 3:15 PM Jerry Geis wrote: > So installed Asterisk 18. changed the modules.conf to remove the "noload = > chan_sip" line > and did some testing it seems to run fine in that mode. > > I configured asterisk 18 with --with-pjproject-bundled > --with-jansson-bundled > >

[asterisk-users] Asterisk 18 in chan_sip mode

So installed Asterisk 18. changed the modules.conf to remove the "noload = chan_sip" line and did some testing it seems to run fine in that mode. I configured asterisk 18 with --with-pjproject-bundled --with-jansson-bundled Couple questions: 1) Is the older chan_sip still supported ? Bug

[asterisk-users] Asterisk 18.4.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.4.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.4.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.18.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.18.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.18.0 resolves several issues reported by the community and would have not been

Re: [asterisk-users] Asterisk pjsip and NAT just doesn't work

On 02.05.21 at 17:24 Michael Maier wrote: On 02.05.21 at 10:08 Michael Maier wrote: Hello! I've just playing around some time to get NAT and pjsip running with asterisk 18.3 and 18.4 (w/o any patches added). NAT should be used for connection to the trunk. I wasn't able to get it working,

Re: [asterisk-users] Asterisk pjsip and NAT just doesn't work

On 02.05.21 at 10:08 Michael Maier wrote: Hello! I've just playing around some time to get NAT and pjsip running with asterisk 18.3 and 18.4 (w/o any patches added). NAT should be used for connection to the trunk. I wasn't able to get it working, because SDP address rewriting just doesn't

[asterisk-users] Asterisk pjsip and NAT just doesn't work

Hello! I've just playing around some time to get NAT and pjsip running with asterisk 18.3 and 18.4 (w/o any patches added). NAT should be used for connection to the trunk. I wasn't able to get it working, because SDP address rewriting just doesn't work as it should. The situation is like

Re: [asterisk-users] Asterisk Community Services Coming Back Up

On Thu, Apr 8, 2021 at 7:29 PM Joshua C. Colp wrote: > Greetings, > > I know it's been quite some time that the Asterisk community services have > been down but I'm pleased to say that most services are coming back online. > DNS entries have been updated but they may take time to fall out of the

Re: [asterisk-users] Asterisk Community Services Coming Back Up

quit texting me I got over 200 Texts from a 410 number -Original Message- From: Sent: Thu, 8 Apr 2021 19:29:17 -0300 To: 3375012...@txt.att.net Subject: [asterisk-users] Asterisk Community Services Coming Back Up >Greetings, > >I know it's been quite

[asterisk-users] Asterisk Community Services Coming Back Up

Greetings, I know it's been quite some time that the Asterisk community services have been down but I'm pleased to say that most services are coming back online. DNS entries have been updated but they may take time to fall out of the various caches across the known universe, so if they aren't

[asterisk-users] Asterisk Community Services Move

Greetings, The Asterisk Community Services infrastructure (issues.asterisk.org, wiki.asterisk.org, gerrit.asterisk.org, crowd.asterisk.org, git.asterisk.org, downloads.asterisk.org, downloads.digium.com, signup.asterisk.org) is undergoing a physical move today which will result in them being

[asterisk-users] Asterisk 18.3.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.3.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.3.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.17.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.17.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.17.0 resolves several issues reported by the community and would have not

Re: [asterisk-users] Asterisk not following SDP port change

accept_multiple_sdp_answers=yes fixed it. It now follows SDP a total of 3 times in my tests. I had found this setting before posting. And had toggled it. But it didn't make any difference until defined in system (in addition to the endpoint itself). Thanks for your help! On Wed, Mar 3, 2021 at

[asterisk-users] Asterisk 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7 Now Available (Security)

The Asterisk Development Team would like to announce security releases for Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7. These releases are available for immediate download at

Re: [asterisk-users] Asterisk not following SDP port change

On Wed, Mar 3, 2021 at 5:55 PM Nick Olsen wrote: > > SDP for the first 183 > Session Description Protocol > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): Sansay-VSXi 188 1 IN IP4 > XX.XX.XX.12 > Session Name (s): Session

Re: [asterisk-users] Asterisk not following SDP port change

On Wed, Mar 3, 2021 at 12:51 PM Nick Olsen wrote: > Hello! > > I've got a number of asterisk systems running asterisk 16.12.0 currently. > They're configured with PJSIP. > Some of them are behind NAT, some aren't. > All systems have SIP trunks to a Sansay INX. > I've had one-way audio issues

[asterisk-users] Asterisk not following SDP port change

Hello! I've got a number of asterisk systems running asterisk 16.12.0 currently. They're configured with PJSIP. Some of them are behind NAT, some aren't. All systems have SIP trunks to a Sansay INX. I've had one-way audio issues calling a particular number. After some investigation, It seems that

[asterisk-users] Asterisk 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6 Now Available (Security)

The Asterisk Development Team would like to announce security releases for Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6. These releases are available for immediate download at

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

Thanks Sean for the note. It does look Selinux might have a hand in the pot. I did try with selinux permission set to permissive and it made no difference though. Keeping configuration related stuff under /etc/asterisk seems to help. --Ruisheng On Mon, Feb 1, 2021 at 8:09 AM Sean Bright

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

Michael, There weren't any open or openat actions on the cert files (located under /home/asterisk/certs). The same is true for cert files located under /etc/asterisk/keys: 24138 stat("/etc/asterisk/keys/fullchain.pem", {st_mode=S_IFREG|0640, st_size=34 44, ...}) = 0 24138 geteuid()

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

Hi, On 1/26/2021 3:12 PM, Ruisheng Peng wrote: Transport: transport-tls: cert_file /home/asterisk/certs/asterisk.crt is either missing or not readable This error means that the file either does not exist or that Asterisk is not able to open it for reading. In your case it looks like the file

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

On 29.01.21 at 22:33 Ruisheng Peng wrote: Thanks for the detailed explanation Michael. I stop the current asterisk process (started by systemd), and restart it as asterisk: [asterisk@voip1 ~]$ strace -f -o /home/asterisk/strace.log asterisk -fmq -vvv -C /etc/asterisk/asterisk.conf from the

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

beating around bushes, and finally seem to stomp on something that worked! Simply move the cert file locations from /home/asterisk/certs to /etc/asterisk/keys [root@voip1 asterisk]# ls -l keys total 36 -rw-r-. 1 asterisk asterisk 1212 Jan 29 14:18 asterisk.crt -rw-r-. 1 asterisk

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

Thanks for the detailed explanation Michael. I stop the current asterisk process (started by systemd), and restart it as asterisk: [asterisk@voip1 ~]$ strace -f -o /home/asterisk/strace.log asterisk -fmq -vvv -C /etc/asterisk/asterisk.conf from the log there was no attempt to even open the

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

Thanks Stefan for the pointer. There isn't a /etc/ssl/openssl.cnf on the Centos7 box. There is a /etc/pki/tls/openssl.cnf, but there's no MinProtocol or CipherString defined there. I installed corebot (for Letsencrypt auto renewal) thru snap. The openssl.cnf that comes with snap (under

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

On Tue, Jan 26, 2021 at 10:12:22AM -1000, Ruisheng Peng wrote: > The self-sign asterisk.crt: I saved that file in "x.crt". openssl x509 -in x.crt -noout -text RSA Public-Key: (1024 bit) > and Letsencrypt cert.pem: I saved that file in "y.crt". openssl x509 -in y.crt -noout

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

On 29.01.21 at 06:41 Michael Maier wrote: On 27.01.21 at 22:57 Ruisheng Peng wrote: Thanks Michael for the suggestion!  I've installed strace and assigned one of the endpoints (SOFTPHONE_B) to use transport-tls. Then run strace (as user asterisk): [asterisk@voip1 ~]$ strace asterisk -rx

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

On 27.01.21 at 22:57 Ruisheng Peng wrote: Thanks Michael for the suggestion! I've installed strace and assigned one of the endpoints (SOFTPHONE_B) to use transport-tls. Then run strace (as user asterisk): [asterisk@voip1 ~]$ strace asterisk -rx "module reload res_pjsip.so" You should use

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

Thanks Michael for the suggestion! I've installed strace and assigned one of the endpoints (SOFTPHONE_B) to use transport-tls. Then run strace (as user asterisk): [asterisk@voip1 ~]$ strace asterisk -rx "module reload res_pjsip.so" execve("/usr/sbin/asterisk", ["asterisk", "-rx", "module reload

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

On 26.01.21 at 21:12 Ruisheng Peng wrote: > Hi, > > I'm experimenting with Asterisk-16.14.0 on a CentOS7 box, and run into > problems loading the SSL certificate to establish transport-tls. Tried > self-signed certificate generated with ast_tls_cert under contrib/scripts > and the one issued

[asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

Hi, I'm experimenting with Asterisk-16.14.0 on a CentOS7 box, and run into problems loading the SSL certificate to establish transport-tls. Tried self-signed certificate generated with ast_tls_cert under contrib/scripts and the one issued by Letsencrypt, both would bomb out with a parsing

Re: [asterisk-users] asterisk-users Digest, Vol 197, Issue 17

Re: Get a SHAKEN Identity Token (Alexander Perkins) Saint Michael 1:06 PM (0 minutes ago) to Asterisk Please look at this https://issues.asterisk.org/jira/browse/ASTERISK-28924 I have a solution that works for any version of Asterisk, if interested contact me at venefax at the Google mail

[asterisk-users] Asterisk 18.2.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.2.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.2.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 16.16.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.16.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.16.0 resolves several issues reported by the community and would have not

Re: [asterisk-users] asterisk-users Digest, Vol 197, Issue 7

Stir Shaken Asterisk cannot do that, but my company can give you Stir Shaken for Asterisk, via ODBC, any version. Please contact me via email venefax at the google mail system Philip Orleans On Fri, Jan 8, 2021 at 1:00 PM wrote: > Send asterisk-users mailing list submissions to >

[asterisk-users] asterisk 11 -> 16 ForkCDR

hi, after upgrade from asterisk 11 to 16 i have problem with ForkCDR app (probably) snippet of dialplan EXTEN=800800800 backup_number=666777888 exten => _X.,n(forward800),noop(forward800) exten => _X.,n,Gosub(routing800,s,1(${EXTEN})) exten => _X.,n,goto(pstn,${backup_number},1)

Re: [asterisk-users] asterisk 13 takes over an hour to clear the MWI light

I just tired/upgraded to asterisk-16.13.0 and the problem still persist. Looking at some post on digium.support web-page, they don't have a clear solution or know what causing it. On 12/24/2020 05:03 AM, Julian Beach wrote: > Hello Thelma, > > Thursday, December 24, 2020, 9:26:53 AM,

Re: [asterisk-users] asterisk 13 takes over an hour to clear the MWI light

Hello Thelma, Thursday, December 24, 2020, 9:26:53 AM, the...@sys-concept.com wrote: > In astersik-11 MWI light was cleared as soon as I checked the message. > In asterink-13 it takes about 20min to set the light ON and the light > takes over an hour to clear. I had this problem following an

[asterisk-users] asterisk 13 takes over an hour to clear the MWI light

In astersik-11 MWI light was cleared as soon as I checked the message. In asterink-13 it takes about 20min to set the light ON and the light takes over an hour to clear. What had changed? In sip.cong [400] ... mailbox=400 voicemail.conf [default] 400 => ,user, email --

Re: [asterisk-users] [asterisk-app-dev] Handling transfers with ARI

Thank you all for the hints. I ended up using a mix of dialplan to deal with the Local channels, and ARI to detect the transfer and redirect. It doesn't look like a "clean" solution but I have nothing better for the moment : Dialplan : exten  =

[asterisk-users] asterisk-13 MWI - phone not blinking

On asterisk-11 MWI was working correctly, phone message light was blinking (standard phone). With asterisk-13, this feature is not working. Who to trouble shoot? -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] asterisk Unknown DYNAMIC_FEATURES item 'automon' on channel

On 12/23/2020 09:54 AM, Doug Lytle wrote: > Review your features.conf file in /etc/asterisk > > Doug I found id. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

Re: [asterisk-users] asterisk Unknown DYNAMIC_FEATURES item 'automon' on channel

Review your features.conf file in /etc/asterisk Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk?

[asterisk-users] asterisk Unknown DYNAMIC_FEATURES item 'automon' on channel

I just upgraded to asterisk-13 (from 11) and I get some errors: 1.) Unknown DYNAMIC_FEATURES item 'automon' on channel SIP Unknown DYNAMIC_FEATURES item 'automon' on channel IAX2/voip Does anybody know how to get rid of them? --

Re: [asterisk-users] [asterisk-app-dev] Handling transfers with ARI

Thanks for the answer. Not sure I get the idea : when a SIP phone performs a blind-transfer, I have no control over what Asterisk does with the channels. During my tests, Bob's channel was automatically pulled out of the bridge, and replaced with a Local channel whose peer goes through the

[asterisk-users] Asterisk 13.38.1, 16.15.1, 17.9.1 and 18.1.1 Now Available (Security)

The Asterisk Development Team would like to announce security releases for Asterisk 13, 16, 17 and 18. The available releases are released as versions 13.38.1, 16.15.1, 17.9.1 and 18.1.1. These releases are available for immediate download at

[asterisk-users] [asterisk-app-dev] Handling transfers with ARI

Hello, I'm struggling to find a way to properly handle blind transfers with ARI. This is my use case : - Alice calls Bob through Asterisk - dialing and bridging is done with ARI - when Bob blind-transfers to Charlie, I would like to use the "redirect" ARI operation, or the Transfer

Re: [asterisk-users] Asterisk registrations - state?

On Tuesday 15 December 2020 at 22:35:07, Jeff LaCoursiere wrote: > On 12/15/20 3:19 PM, Antony Stone wrote: > >> There is no functionality present to have Asterisk SUBSCRIBE to upstream > >> servers, receive updates, and locally use them. > > > > Hm, thanks for the clarification, this confirms

Re: [asterisk-users] Asterisk registrations - state?

On 12/15/20 3:19 PM, Antony Stone wrote: There is no functionality present to have Asterisk SUBSCRIBE to upstream servers, receive updates, and locally use them. Hm, thanks for the clarification, this confirms what I suspected. Can anyone suggest an alternative application I could

Re: [asterisk-users] Asterisk registrations - state?

On Monday 14 December 2020 at 13:31:54, Joshua C. Colp wrote: > On Tue, Dec 8, 2020 at 1:07 PM Antony Stone wrote: > > > > I'm using Asterisk 16.2.1 with some registrations (ie: my Asterisk server > > is registering to other PBXs as though it were a telephone). > > > > Is there any way I can

Re: [asterisk-users] Asterisk registrations - state?

On Tue, Dec 8, 2020 at 1:07 PM Antony Stone < antony.st...@asterisk.open.source.it> wrote: > Hi. > > I'm using Asterisk 16.2.1 with some registrations (ie: my Asterisk server > is > registering to other PBXs as though it were a telephone). > > Is there any way I can get presence / state

Re: [asterisk-users] Asterisk registrations - state?

Hi. Does nobody have any ideas / suggestions about this? On Tuesday 08 December 2020 at 18:07:05, Antony Stone wrote: > Hi. > > I'm using Asterisk 16.2.1 with some registrations (ie: my Asterisk server > is registering to other PBXs as though it were a telephone). > > Is there any way I can

[asterisk-users] Asterisk, linphone, and jssip webrtc

Hi All, I am new to VoIP world and trying to set up asterisk, linphone, and jssip webrtc. Settings: - transport_wss (127.0.0.1, apache ws_tunnel) - transport_tls (public ip port 5060) - use_avpf=yes - ice_support=yes - dtls enabled (letsencrypt) - rtcp_mux=yes -

Re: [asterisk-users] Asterisk 16.2.1: P-Asserted-Id set to s when Dialing inside a certain Gosub

May I add that, to me, I would expect Asterisk to use CALLERID vlaues (name and num) to set P-Asserted-Id. Maybe in a couple of days, I'll report my findings here if can find some time to experiment with Asterisk 17 or Asterisk 18 and compare behaviours.. Le mar. 8 déc. 2020 à 16:41, Olivier

Re: [asterisk-users] Asterisk and CentOS 8

10.12.2020 03:25, Patrick Wakano пишет: In case anyone out there is working with CentOS, you might reconsider that decision: https://blog.centos.org/2020/12/future-is-centos-stream/ Oracle? :-) --

Re: [asterisk-users] Asterisk and CentOS 8

In case anyone out there is working with CentOS, you might reconsider that decision: https://blog.centos.org/2020/12/future-is-centos-stream/ On Mon, 11 May 2020 at 23:53, George Joseph wrote: > > > On Sun, May 3, 2020 at 6:07 PM Patrick Wakano wrote: > >> Hello George, >> Hope this finds you

[asterisk-users] Asterisk registrations - state?

Hi. I'm using Asterisk 16.2.1 with some registrations (ie: my Asterisk server is registering to other PBXs as though it were a telephone). Is there any way I can get presence / state information from those PBXs in the same way that a registered telephone can? In other words, can I send a

[asterisk-users] Asterisk 16.2.1: P-Asserted-Id set to s when Dialing inside a certain Gosub

Hello, With Debian Buster's Asterisk 16.2.1, please consider the following dialplan ;Case A ;exten = 29,1,Dial(PJSIP/${EXTEN}) ;Case B ;exten = 29,1,Gosub(foo,${EXTEN},1) ;Case C exten = 29,1,Gosub(bar,s,1(${EXTEN})) [foo] exten = _X.,1,Dial(PJSIP/${EXTEN}) same = n,Return() [bar] exten =

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