Re: [asterisk-users] Deleting voicemail by program

2023-10-09 Thread Michael Bradeen
Hi Mike, New AMI actions were recently added to app_voicemail to let you remotely manipulate a mailbox: https://github.com/asterisk/asterisk/issues/181 Hope this helps. BR, -Mike On Mon, Oct 9, 2023 at 1:06 PM Mike Diehl wrote: > Hi all, > > I need to be able to delete a voicemail message us

Re: [asterisk-users] Asterisk Release 20.3.1

2023-07-08 Thread Michael Maier
At the moment, I can't see any differences here. sha512sum is identical. Regards Michael On 08.07.23 at 01:50 Jean-Denis Girard wrote: Le 07/07/2023 à 12:49, Joshua C. Colp a écrit : On Fri, Jul 7, 2023 at 6:40 PM Jean-Denis Girard <mailto:jd.gir...@sysnux.pf>> wrote:     There

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

2023-07-06 Thread Michael Ulitskiy
FYI: i've created a feature request to add SIP_CODEC_INBOUND equivalent functionality to chan_pjsip: https://github.com/asterisk/asterisk-feature-requests/issues/9 Let's see where it goes *Michael Ulitskiy* Ace Innovative Networks, Inc. Main/SMS: 212-868-2366 Direct/SMS: 212-812-

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

2023-07-06 Thread Michael Ulitskiy
Oh, that's great. It wasn't clear from that page, at least not for me. :-( Having it clearly stated on the document would save me (and probably others) lots of time. Thanks for clarifying it. Any idea on the timeframe of implementation? *Michael Ulitskiy* Ace Innovative Networks,

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

2023-07-06 Thread Michael Ulitskiy
on in the above scenario should be ulaw in both call legs, thus avoiding transcoding, but actual asterisk behavior differs. Am I missing something. What are your thoughts? Thanks, *Michael Ulitskiy* Ace Innovative Networks, Inc. Main/SMS: 212-868-2366 Direct/SMS: 212-812-1203 https://www.

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

2023-07-05 Thread Michael Ulitskiy
Well, I'm trying to migrate to chan_pjsip so that I don't have to do that. It's so surprising that the issue so seemingly obvious and trivial hasn't been addressed yet that I wanted to query the collective wisdom of this list to verify my observations. Thanks for github p

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

2023-07-05 Thread Michael Ulitskiy
Hi Michael, Thanks for the reply. I was referring to the scenario you named as 'outbound broken'. I didn't get to look at inbound call behavior yet, as I got stuck with inability to avoid transcoding on outbound calls. To be more specific the scenario is as follows: 1. a ph

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

2023-07-05 Thread Michael Maier
Hello Michael, you are referring to the following behavior - did I get it correctly?: outbound broken: asterisk offers g722 / g711 to provider (callee), callee answers g711. Asterisk now transcodes between caller and callee (g722 <-> g711). inbound works: call from provider

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

2023-07-05 Thread Michael Ulitskiy
t up using ulaw end-to-end Can somebody please advise how to achieve the same with chan_pjsip? Thanks, Michael On 6/30/23 09:30, Michael Ulitskiy wrote: Hello, I finally got to look at chan_sip to chan_pjsip migration again. This time I’m having problems with influencing codec selection on o

[asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

2023-06-30 Thread Michael Ulitskiy
${SIP_CODEC_INBOUND}? Or, in other words, what are we supposed to do to influence /calling/ channel codec selection from dialplan? I’m working with asterisk 20.3.0. Thank you, Michael -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] TLS and NAT

2023-04-10 Thread Michael Maier
On 09.04.23 at 19:55 Steve Matzura wrote: Thanks, Michael. A few questions: Is [transport_name] a reserved word, or am I supposed to replace it with a name of my own, like '[did-transport]'? Yes. You are free. Some of the keywords I haven't seen before. Is ca_list_file su

Re: [asterisk-users] TLS and NAT

2023-04-08 Thread Michael Maier
external_media_address=your.ext.host.name ; hostname pointing to your ext. IP external_signaling_address=your.ext.host.name ; hostname pointing to your ext. IP local_net=192.168.0.0/24 # your local net Regards Michael On 07.04.23 at 17:25 Steve Matzura wrote: I want to configure communication

Re: [asterisk-users] extensions.conf asterisk 18.8.0 question

2022-01-11 Thread Michael Englehorn
delete files. I use `audit2why` and `audit2allow` in policycoreutils-devel (on CentOS) to generate SELinux policy modules. -Michael Englehorn ‐‐‐ Original Message ‐‐‐ On Monday, January 10th, 2022 at 1:03 PM, Jerry Geis wrote: > I am trying to run this command: > exten => _

Re: [asterisk-users] voicemail message not deleted

2021-07-26 Thread Michael Keuter
quot;delete=yes" as option per mailbox account. 100 => 1234,Test,,,delete=yes The global setting is only an example. Michael http://www.mksolutions.info -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] problems with natted phones

2021-07-08 Thread Michael L. Young
ki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT -- Michael L. Young (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community

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

2021-07-01 Thread Michael Maier
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?

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

2021-06-30 Thread Michael Maier
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 pjsip and NAT just doesn't work

2021-05-03 Thread Michael Maier
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

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

2021-05-02 Thread Michael Maier
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 ju

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

2021-05-02 Thread Michael Maier
rking. Could somebody maybe give me a reference configuration for a working NAT configuration? Thanks Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community

Re: [asterisk-users] pjsip transport and dynamic WAN IP address (not: NAT)

2021-05-01 Thread Michael Maier
On 01.05.21 at 07:13 Michael Maier wrote: Hello all! I'm actually wondering about how to achieve fast fail handling for the pjsip transport if underlying WAN IP address changes. Forgot to mention: - I'm using TLS! - pjsip tries every 92s to send the Registration until the timeou

[asterisk-users] pjsip transport and dynamic WAN IP address (not: NAT)

2021-05-01 Thread Michael Maier
n to reduce the timeout until restart of the individual transport? Maybe in pjsip (connection timeout detection)? Another idea would be to restart the relevant transports based on dnsmgr detecting new external IP. Thanks Michael --

[asterisk-users] Channel Variable inheritance

2021-02-23 Thread Michael Munger
ption that the variable would be inherited into subsequent channels, but that does not work either. What am I missing? Asterisk: 13.14.1~dfsg-2+deb9u4 OS: Debian 9.13 (Stretch). -- Michael Munger, dCAP, MCPS, MCNPS, MBSS *Microsoft Certified Professional* *Microsoft Certified Small Busin

Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

2021-02-19 Thread Michael Maier
On 18.02.21 at 20:01 Luca Bertoncello wrote: Am 18.02.2021 um 18:59 schrieb Michael Maier: On 17.02.21 at 21:46 Luca Bertoncello wrote: Am 16.02.2021 um 22:32 schrieb Michael Maier: Hi Michael Maybe could you send me an abstract of your configuration? Take a look here [1] So, maybe I

Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

2021-02-18 Thread Michael Maier
On 17.02.21 at 21:46 Luca Bertoncello wrote: > Am 16.02.2021 um 22:32 schrieb Michael Maier: > > Hi Michael > >>> Maybe could you send me an abstract of your configuration? >> >> Take a look here [1] > > So, maybe I got it... > I tested the configurati

Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

2021-02-16 Thread Michael Maier
On 16.02.21 at 20:33 Luca Bertoncello wrote: Am 16.02.2021 um 19:56 schrieb Michael Maier: Hi Michael, Do I use pjsip? pjsip show registrations gw*CLI> pjsip show registrations No objects found. So I don't use pjsip... :( Yes. Maybe could you send me an abstract

Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

2021-02-16 Thread Michael Maier
Hi Luca, On 15.02.21 at 21:48 Luca Bertoncello wrote: > Am 15.02.2021 um 21:40 schrieb Michael Maier: > > Hi Michael, > >> They're switching to DNS NAPTR / SRV[1]. If you are using Asterisk / >> pjsip and hostnames (tel.t-online.de e.g. for the AllIP service), yo

Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

2021-02-15 Thread Michael Maier
nline.de. 3600 IN SRV 10 0 5060 s-epp-110.edns.t-ipnet.de. _sip._tcp.tel.t-online.de. 3600 IN SRV 20 0 5060 h2-epp-100.edns.t-ipnet.de. Asterisk now must use always the same server for all activities to Telekom - like register, invite, options - but that's not yet support

Re: [asterisk-users] CentOS 7 yum-update and Gotoif has stoppped to workink

2021-02-05 Thread Michael L. Young
- On Feb 5, 2021, at 11:18 AM, Michael L. Young wrote: > - On Feb 4, 2021, at 4:26 PM, Social Boh wrote: >> The problem is with this CentOS 7 glibc version: >> 2.17-317.el7 >> After the library update and system reboog, >> gotoif Asterisk applicati

Re: [asterisk-users] CentOS 7 yum-update and Gotoif has stoppped to workink

2021-02-05 Thread Michael L. Young
a 'yum history' and note the transaction ID of the update. Then try running 'yum history undo [transaction id]'. That should roll you back to the previous glibc. Looks like Red Hat is already working on it: https://access.redhat.com/solutions/57

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

2021-01-30 Thread Michael Maier
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

2021-01-29 Thread Michael Maier
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 asteris

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

2021-01-29 Thread Michael Maier
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"

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

2021-01-27 Thread Michael Maier
! Take care, that the file access rights of the file and the complete path are ok. Do a strace to verify, if the file is really loaded at all. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

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

2021-01-24 Thread Saint Michael
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

Re: [asterisk-users] Get a SHAKEN Identity Token (Alexander Perkins)

2021-01-24 Thread Saint Michael
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 service. On Sun, Jan 24, 2021 at 1:00 PM wrote: > Send asterisk-users mailing list submissions to >

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

2021-01-08 Thread Saint Michael
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 > asteris

Re: [asterisk-users] Asterisk 18.0.0 Now Available

2020-10-21 Thread Michael Maier
On 21.10.20 at 12:49 Joshua C. Colp wrote: > On Wed, Oct 21, 2020 at 7:46 AM Michael Maier wrote: > >> Hello! >> >> On 20.10.20 at 14:00 Asterisk Development Team wrote: >>> The Asterisk Development Team would like to announce the release of >> Asterisk 18

Re: [asterisk-users] Asterisk 18.0.0 Now Available

2020-10-21 Thread Michael Maier
d you please add the correct configuration you expect to get the expected result alaw? Thanks Kind regards Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk commu

Re: [asterisk-users] linphone calls not missed due to cause not 487

2020-10-16 Thread Michael Maier
On 16.10.20 at 11:07 sergio wrote: > On 16/10/2020 10:11, Michael Maier wrote: >>> Sometimes, linphone shows missed calls as missed. >> You could try to reproduce it > > I can't reproduce it, it happens less than once a month. Then you should enable the tracing as

Re: [asterisk-users] linphone calls not missed due to cause not 487

2020-10-16 Thread Michael Maier
this? You could try to reproduce it while activating pcap traces and analyze it afterwards - or you could enable pcap traces on asterisk[1] itself and just wait for the different issues and compare them. Regards Michael [1] activating pcap traces on asterisk / pjsip pjsip set logger on pjsip

Re: [asterisk-users] BLF support in Asterisk and early/confirmed/terminated/proceeding NOTIFY states.

2020-10-03 Thread Michael Keuter
advised “hey I > am not available” and actually show a RED dot. > thanks > > -- Hi Jobst, there is a setting for the Yealink phones so that the BLF keys are "off" instead of "green" when idle: BLF LED Mode = 1 in t

Re: [asterisk-users] asterisk-users Digest, Vol 193, Issue 15

2020-09-26 Thread Saint Michael
memory vs disk cache > This is an issue that has plagued Asterisk since day one. Basically there >> is no solution available because there is no way to set aside memory to be >> kept from a growing disk cache. I did some research and this looks like a >> bad design from the Kernel people. Meanwhil

[asterisk-users] 1. memory issues (hw)

2020-09-26 Thread Saint Michael
> > This is an issue that has plagued Asterisk since day one. Basically there > is no solution available because there is no way to set aside memory to be > kept from a growing disk cache. I did some research and this looks like a > bad design from the Kernel people. Meanwhile all you can do us eve

Re: [asterisk-users] func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts

2020-09-06 Thread Michael Maier
with the bundled pjsip library. Doing it this way ensures that pjsip and asterisk match for sure (and some additional patches are applied to pjsip on top regarding usage of pjsip in asterisk). Greetings Michael [1] https://downloads.asterisk.or

[asterisk-users] Stir Shaken

2020-07-14 Thread Saint Michael
I need to point out the this is factually misleading and materially false: "I think this, being the basis of your whole argument, is the fallacy. S/S is forcing people to take responsibility, for sure, but carriers won't just let their customers leave because they don't want to sign calls. It will

[asterisk-users] Stir Shaken

2020-07-13 Thread Saint Michael
> > There is a big confusion here about Stir Shaken. It is NOT a provider > issue. Un fact, all providers are whasing their hands and modifying their > swihtches to pass-through the Signature. They cannot sign the call because > then the become the responsible party for the call before the FCC, and

Re: [asterisk-users] Stir Shaken is upon us

2020-07-13 Thread Michael Maier
On 13.07.20 at 10:54 Joshua C. Colp wrote: > On Sun, Jul 12, 2020 at 11:37 PM Michael Maier wrote: >> One more question, >> what about the pjsip pcap support? Will it be backported to Asterisk 16, >> too? Would be absolutely cool! Debugging encrypted SIP is really a pain

Re: [asterisk-users] Stir Shaken is upon us

2020-07-12 Thread Michael Maier
oo? Would be absolutely cool! Debugging encrypted SIP is really a pain. BTW: what about the (encrypted) RTP packets? Will they be dumped, too? Thanks Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-di

[asterisk-users] Stir Shaken is upon us

2020-07-12 Thread Saint Michael
WORLDWIDE EMERGENCY The code below needs to be executed before any SIP or PJSIP call destined to the US network, or soon no call will terminate. This is called Stir-Shaken, a new law from the FCC. If this is not working the whole Asterisk industry will crash, vanish, be gone. I am assuming that the

Re: [asterisk-users] Voice broken during calls (again...)

2020-07-07 Thread Michael Maier
/ configurations to get a proper and reliable rtp stream - even over hours. Regards Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: http

Re: [asterisk-users] Voice broken during calls (again...)

2020-06-24 Thread Michael Maier
Am 24.06.20 um 08:10 schrieb Luca Bertoncello: Am 24.06.2020 05:05, schrieb Michael Maier: Hi Your basic architecture looks good to me - now you have to start the Nice to hear it... analysis of the problem with pcapsipdump and wireshark as I wrote before to get an idea what actually

Re: [asterisk-users] Voice broken during calls (again...)

2020-06-23 Thread Michael Maier
On 23.06.20 at 21:10 Luca Bertoncello wrote: > Am 23.06.2020 um 21:08 schrieb Michael Maier: >> On 23.06.20 at 08:05 Luca Bertoncello wrote: >>> Am 23.06.2020 07:27, schrieb Luca Bertoncello: >>> >>> I again >>> >>>>> Do not change MTU. P

Re: [asterisk-users] Voice broken during calls (again...)

2020-06-23 Thread Michael Maier
-packets never reach this size. Size is about 214 bytes. Regards Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org

Re: [asterisk-users] Voice broken during calls (again...)

2020-06-22 Thread Michael Maier
ause they usually provide optimal sip servers for you (regarding distance). You're RTT of ping (18 ms) is pretty bad. I'm having here 5 ms to the primary server (Telekom provides 3). See dig +noall +answer _sip._udp.tel.t-online.de SRV e.g. (don't know the hos

Re: [asterisk-users] Voice broken during calls (again...)

2020-06-22 Thread Michael Keuter
network (using the "internal number") the quality is excellent. >> >> If I call my wife using the "external number", the quality is very bad... >> >> Thanks >> Luca Bertoncello >> (lucab...@lucabert.de) Michael http://www.mksolutions.info

Re: [asterisk-users] Voice "broken" during calls

2020-06-14 Thread Michael Keuter
law for a test. > > I really do *not* expect any change in the situation... I think, the > problem should be somewhere by Deutsche Telekom... > > What is your opinion? > > Btw: I did all tests with my father in law, since he had time for me > today, but the problem exist

Re: [asterisk-users] Voice "broken" during calls

2020-06-13 Thread Michael Keuter
So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello : > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer " for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177

Re: [asterisk-users] Voice "broken" during calls

2020-06-13 Thread Michael Keuter
a phone. Then "sip show channels" during an existing call. And "sip show channel " for more info. Michael http://www.mksolutions.info -- _ -- Bandwidth and Colocation Provided by http:

Re: [asterisk-users] CLI color prompt

2020-06-01 Thread Michael Maier
output like core set debug 5 Maybe some link time functionality not enabled? Don't know ... . Or some other additional asterisk switch needed? Thanks Regards Michael -- _ -- Bandwidth and Colocation Provided by http://www

[asterisk-users] Extracting a SIP Header from a 302 Response

2020-05-30 Thread Saint Michael
I got the response below from a provider. How do I extract the Identity header and apply it to the next INVITE? Is it possible at all with PJSIP? SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 172.16.7.254:52169 ;rport=52169;received=XX.205.172.89;branch=z9hG4bK-524287-1---129f4244aaba9f04 Call-ID:

[asterisk-users] Stir-Shaken clarified

2020-05-29 Thread Saint Michael
https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN The Wiki above is misleading in what Stir-Shaken means and how it works. End users cannot get a certificate, they cannot self-certify their calls. Somebody completely misunderstood the model. I am afraid the moment will come and thousands o

[asterisk-users] STIR-Shaken

2020-05-28 Thread Saint Michael
> > My company is one if the six service providers approved. We are not ready > yet, probbably next week, since the certificate needs to be issued by the > Certification Authority. As I said, we are the ONLY provider that you may > use with Asterisk remotely, via UnixODBC. The rest of the other pr

[asterisk-users] Stir-Shaken for asterisk

2020-05-27 Thread Saint Michael
In a few weeks, no SIP call is going to terminate unless they are signed properly, as mandated by law. We are in the business of Stir-Shaken, signing calls, as an FCC-approved provider. A big differentiator between our service and the rest: we are the only ones who don't need to receive the calls

Re: [asterisk-users] PJSIP sending RTP to private address

2020-05-17 Thread Saint Michael
About this case: the old SIP channel behaves correctly. On Sun, May 17, 2020 at 2:44 AM Saint Michael wrote: > My phone is located behind a NAT, 172.16.0.0/21. > Asterisk 16 is on a public IP. > PJSIP has the config below: > force_rport=yes > direct_media=yes > disable_direct_

Re: [asterisk-users] Meaning of RTT in channelstats

2020-05-17 Thread Michael Maier
On 17.05.20 at 01:28 Joshua C. Colp wrote: > On Sat, May 16, 2020 at 10:58 AM Michael Maier wrote: > >> => How are the RTT values exactly calculated? Which values are actually >> used for? >> > > The value is calculated according to the logic in the RFC[

[asterisk-users] Help missing

2020-05-16 Thread Saint Michael
I want to see the help when I type core show application , and it's not available. This is asterisk 16 from sources. I have libxml2-dev installed. Ubuntu 19 What am I missing? Philip -- _ -- Bandwidth and Colocation Provided

[asterisk-users] PJSIP does not stop sending invites after call is canceled

2020-05-16 Thread Saint Michael
Endpoint sends an INVITE Asterisk send an INVITE to the Carrier Carrier is down, does not even sends ACK PJSIP sends several INVITES End point sends <--- Received SIP request (397 bytes) from UDP ::50187 ---> CANCEL sip:xxx@xxx SIP/2.0 Via: SIP/2.0/UDP xxx :50187;branch=z9hG4bK-524

Re: [asterisk-users] Meaning of RTT in channelstats

2020-05-16 Thread Michael Maier
exact measuring points located? => How are the RTT values exactly calculated? Which values are actually used for? => What about the processing time between the inbound leg and the outbound leg (transcoding, ...)? Thanks Michael --

[asterisk-users] Meaning of RTT in channelstats

2020-05-15 Thread Michael Maier
xxx-0 03:22:42 alaw 608K 00 0.000 608K 00 0.000 0.023 Thanks Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum a

Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?

2020-05-14 Thread Michael L. Young
owse/ASTERISK-26143 | https://issues.asterisk.org/jira/browse/ASTERISK-26143 ] Not sure if this is the answer to your problem but thought that I would throw that out there. Michael L. Young (elguero) -- _ -- Bandwidth and

[asterisk-users] New RTP engine

2020-05-11 Thread Saint Michael
> > Asterisk needs urgently to push the RTP engine to the Kernel, away from > userland, like professional and commercial softwares do. I measured the > cost of passing call from a public IP to a private IP, like typically a > Session Border Controller may do. In Asterisk, ulaw, no transcoding, it >

[asterisk-users] PJSIP crashes

2020-02-26 Thread Saint Michael
> > I have no control over the SIP calls I receive. PJSIP should log a warting > and continue. It is causing the CPU usage to spike dramatically. > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check

[asterisk-users] PJSIP crashes

2020-02-25 Thread Saint Michael
PJISP cannot handle the From field when it does not contain a number. Can this be fixed? [Feb 25 12:35:43] ERROR[7143]: pjproject: : sip_transport.c Error processing 400 bytes packet from UDP 8.38.43.67:5060 : PJSIP syntax error exception when parsing 'From' header on line 4 col 40: CANCE

[asterisk-users] avoiding any media proxy with PJSIP

2020-02-13 Thread Saint Michael
Is there a guide on how to use PJSIP and never have the media travel inside Asterisk? No matter what I do, I cannot make this work. Philip Orleans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check o

Re: [asterisk-users] permission woes on systemd

2020-01-22 Thread Michael L. Young
as user asterisk work, but fail using > systemd ? > Have you checked SELinux? After creating the configuration files, did you run 'restorecon' on the appropriate asterisk directories? If not, the files are not labeled correctly and SELinux might

[asterisk-users] How to: send dtmf back to the calling channel from post-answer subroutine executed on outbound channel

2019-12-17 Thread Saint Michael
I have a customer who wants me to send a DTMF on the calling channel if the called channel says any word. So I am using [my_gosub_routine] exten => s,1,NoOp(ARG1=${ARG1} ARG2=${ARG2}) same => n,Playback(hello) same => n,Return() [default] exten => _X.,1,NoOp() same => n,Dial(PJSIP/alice,,U(my

[asterisk-users] Own MOH incorrectly kicking in instead of the MOH of the callee

2019-11-01 Thread Michael Maier
of FreePBX? Thanks Michael -- _ -- 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? Start here: https://wi

Re: [asterisk-users] FREEPBX Mailinglist

2019-09-11 Thread Michael Maier
.522 ms 14 * wi-mke1-dc2-c5-47-haf-7578.cyberlynk.net (66.185.29.26) 118.165 ms * 15 199.102.239.92 (199.102.239.92) 122.352 ms 126.759 ms 123.944 ms 16 199.102.239.92 (199.102.239.92) 123.890 ms 124.069 ms 126.670 ms Thanks Michael -- __

[asterisk-users] asterisk 16.5 / pjsip outage because of task processor queue >= 500 tasks and too many open files later on

2019-08-16 Thread Michael Maier
ration problem wasn't a problem of the provider not answering but in fact a problem of asterisk being unable to correctly perform the ReRegistration. The final question: === Is there a problem with taskprocessors probably not being canceled on some co

Re: [asterisk-users] Anyone ever experienced a crash where Asterisk debug output a line with all nulls

2019-08-14 Thread Michael Maier
shutdown (= all file systems have been correctly unmounted and all processes have been stopped before unmounting)? Is the startup slow even if it's done w/o reboot before? Regards Michael -- _ -- Bandwidth and Co

Re: [asterisk-users] Anyone ever experienced a crash where Asterisk debug output a line with all nulls

2019-08-14 Thread Michael Maier
08/12 15:33:03.240] NOTICE[1385] loader.c: 286 modules will be loaded. > [08/12 15:33:23.844] VERBOSE[1385] loader.c: Loading extconfig. > > > Loading the modules is taking 20 seconds after this incident occurred. > Looking at the debug logs, I see th

[asterisk-users] Wanted: professional softphone

2019-07-24 Thread Michael Maier
Hello! Does anybody by chance know of a softphone which additionally has a management suite to fully configure it userID based for Windows clients? Any idea is appreciated! Thanks Michael -- _ -- Bandwidth and Colocation

[asterisk-users] PJSIP / tcp: define local port to use on base of trunk definition

2019-07-08 Thread Michael Maier
ent users). Besides the possibility to use different IP addresses (aliases) - is there a generic way to define on trunk base which local port to use for each user / number? Thanks Michael -- _ -- Bandwidth and Colocation Provi

Re: [asterisk-users] unsolved: Re: solved: how to create a working certificate for using TLS?

2019-07-07 Thread Michael Maier
t supposed to be like this, or should I make a bug report? I think it's supposed to behave like this, because it would mean to disconnect all running calls on sip reload. That's probably not what most of the people expect / want to have. Regards Michael -- __

Re: [asterisk-users] unsolved: Re: solved: how to create a working certificate for using TLS?

2019-07-06 Thread Michael Maier
On 06.07.19 at 12:16 hwilmer wrote: > On 7/6/19 10:40 AM, Michael Maier wrote: >> On 05.07.19 at 22:02 hw wrote: >>> >>> openssl verify -CAfile ca.pem asterisk.pem >>> asterisk.pem: OK >>> >>> >>> When I set tlsdontverifyserver=yes,

Re: [asterisk-users] unsolved: Re: solved: how to create a working certificate for using TLS?

2019-07-06 Thread Michael Maier
ficate to connect via tls to the ISP? To be able to verify the certificate of the ISP, asterisk has to know the local CA database. For CentOS 7, this is /etc/pki/tls/certs/ca-bundle.crt. Regards Michael -- _ -- Bandwidth a

Re: [asterisk-users] High delay and some echo

2019-06-21 Thread Michael Maier
00:00:39 alaw 1299 00 0.000 1296 00 0.000 0.000 Instead of "pjsip show channelstats" you have to use something like sip show [press 2 times tab key] to get the possible commands. Each call generates two entries: one for the call fr

Re: [asterisk-users] asterisk-users Digest, Vol 177, Issue 11

2019-05-25 Thread Saint Michael
Joshua Is there a way in PJSIP to send the audio between the parties always, unless one of the parties is behind a NAT? A session refresh would work. That my only problem with PJSIP. This is routine in the old sip channel. On Sat, May 25, 2019 at 1:03 PM wrote: > Send asterisk-users mailing list

[asterisk-users] pjsip and tls client: How to decrypt Wireshark trace?

2019-05-11 Thread Michael Maier
Hello! I'm just wondering if it's possible to decrypt sips packages in Wireshark while asterisk runs as sips client (connecting to the provider w/ tls 1.2)? I don't use an own certificate. Thanks Michael -- ___

Re: [asterisk-users] Forking AGI or GoSub

2019-04-19 Thread Michael Munger
last time I looked into this with PHP was under PHP 5.6, and tests at that time did not yield the results we wanted. We ultimately moved to Python when we needed multi-threading, which is extremely elegant and reliable for this application. [cid:image001.png@01D4F6B9.8C1499D0] Michael J. Munger,

Re: [asterisk-users] Forking AGI or GoSub

2019-04-19 Thread Michael Munger
continue. Would need testing so it fails gracefully. [cid:image001.png@01D4F6B8.F3CEF8F0] Michael J. Munger, dCAP, MCPS, MCNPS, MBSS Microsoft Certified Professional Microsoft Certified Small Business Specialist Digium Certified Asterisk Professional High Powered Help, Inc. p: 678-905-8569 w

Re: [asterisk-users] Paging systems?

2019-03-21 Thread Michael Munger
Excellent point. This is it: https://www.valcom.com/pdf/v-1109rthf.pdf Get Outlook for Android<https://aka.ms/ghei36> On Thu, Mar 21, 2019 at 7:22 PM -0400, "John Novack" mailto:jnov...@comcast.net>> wrote: Michael Munger wrote: Does anyone have an (overhead) paging

Re: [asterisk-users] Paging systems?

2019-03-21 Thread Michael Munger
It worked on the old system. I am open to suggestions, but don't want (or have the option) to add a TDM card. Michael Munger, dCAP, MCPS, MCNPS, MBSS *Microsoft Certified Professional* *Microsoft Certified Small Business Specialist* *Digium Certified Asterisk Professional*

[asterisk-users] Paging systems?

2019-03-21 Thread Michael Munger
er happens. [cid:image001.png@01D4DFF6.9C1F1AA0] Michael J. Munger, dCAP, MCPS, MCNPS, MBSS Microsoft Certified Professional Microsoft Certified Small Business Specialist Digium Certified Asterisk Professional High Powered Help, Inc. p: 678-905-8569 w: hph.io<https://hph.io> e: m...@hph.io<

[asterisk-users] Cannot change astdatadir?

2019-03-04 Thread Michael Munger
normally compile from source, which uses /var/lib/asterisk by default. First time using a repo package) Asterisk version: 13.14.1~dfsg-2+deb9u4 OS: Debian 9.8 (stretch). Fully updated. [cid:image001.png@01D4D272.0AFF3870] Michael J. Munger, dCAP, MCPS, MCNPS, MBSS Microsoft Certified Pr

Re: [asterisk-users] Outbound call: caller gets no ringback on session progress

2019-01-22 Thread Michael Maier
On 17.12.18 at 11:52 Joshua C. Colp wrote: > On Sun, Dec 16, 2018, at 4:43 AM, Michael Maier wrote: > > > >> >> Another question: is there any use case for 183 Session Progress w/o >> SDP? IOW: Is a 183 Session >> Progress just a bug of the ISP? If so, pro

Re: [asterisk-users] MWI Delayed on Polycom VVX phones

2019-01-15 Thread Michael Keuter
990a42e1317f4aa2bad51a3ef9f17 I am using "mailboxes=##@default" and had the issue as well (before). Michael http://www.mksolutions.info -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

[asterisk-users] Compiling error

2019-01-10 Thread Saint Michael
> > when compiling the latest version, it fails here ./configure LDFLAGS="-z muldefs" --libdir=/usr/lib64 --with-unixodbc=$(odbc_config --include-prefix)/ --disable-asteriskssl -enable-xmldoc NOISY_BUILD=yes > gcc -o res_pjsip/config_transport.o -c res_pjsip/config_transport.c -MD > -MT res_pj

Re: [asterisk-users] Voice mail: MWI problem / pjsip (13.24.0)

2018-12-28 Thread Michael Maier
On 28.12.18 at 13:20 Doug Lytle wrote: >>>> Before I'm opening an issue, I would like to prove my expectations - maybe >>>> it isn't a problem at all or it's a problem of the phone. > > Michael, > > Just a side note. I've had reports of

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