Re: [asterisk-users] Asterisk Release 20.3.1

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 > wrote:     There seems to be a

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

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: g711 -> asterisk

Re: [asterisk-users] TLS and NAT

ented out in your instructions? Take a look here: https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample Search for "tos=0" Regards, Michael Otherwise, the rest is quite clear. On 4/8/2023 12:35 PM, Michael Maier wrote: Hello Steve, use the followin

Re: [asterisk-users] TLS and NAT

Hello Steve, use the following configuration for the transport and bind this transport to the trunk: [transport_name] type=transport protocol=tls bind=192.168.13.24 ; your bind IP ca_list_file=/etc/pki/tls/certs/ca-bundle.crt ; method=tlsv1_2 verify_server=yes allow_reload=no ;tos=0xb8 ;cos=3

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?

[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 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

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 ju

[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 t

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

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)

Hello all! I'm actually wondering about how to achieve fast fail handling for the pjsip transport if underlying WAN IP address changes. Following scenario: Asterisk runs on a device holding ppp0, which provides the interface for outbound registration to ISP trunks (transport acts as client)

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

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?

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?

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?

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?

Hi! On 15.02.21 at 08:43 Luca Bertoncello wrote: > Hi list! > > I received a letter from Deutsche Telekom where they say me, that I need > to change "something" on my router until 28.02.2021, otherwise I cannot > phone anymore. > Since I use Asterisk and I don't have a router, I'm not sure what I

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 l

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 asteris

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 str

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 by

Re: [asterisk-users] Asterisk 18.0.0 Now Available

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

Hello! On 20.10.20 at 14:00 Asterisk Development Team wrote: > The Asterisk Development Team would like to announce the release of Asterisk > 18.0.0. > This release is available for immediate download at > https://downloads.asterisk.org/pub/telephony/asterisk I just tested the new codec negotiat

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

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

Hi Sergio On 16.10.20 at 07:54 sergio wrote: > Sometimes, linphone shows missed calls as missed. Look like asterisk > replies with cause=487 that time, but I can't understand why. > > Grandstream always shows calls as missed ones. > > How should I investigate this? You could try to reprod

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

On 05.09.20 at 15:22 sean darcy wrote: > asterisk-16.13.0-rc2. Fedora 32 > > pjsip won't load because of undefined symbols: This means, that your pjsip library doesn't match the asterisk binary. It's best to remove the independent pjsip library and compile asterisk[1] with the bundled pjsip lib

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

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

On 13.07.20 at 00:17 Joshua C. Colp wrote: > On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins wrote: > >> Asterisk 18 will have support based on this asterisk update Matt F did for >> CommCon's sponsor slots >> >> https://youtu.be/eas1csaX-wc >> >> > As well support will go into Asterisk 16 and 17 as

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

On 03.07.20 at 19:57 Luca Bertoncello wrote: [...] >> On the Gateway (Banana PI), where the Asterisk server also runs, the >> load is about 0.50 during calls and it has a Gbps LAN. >> I can't believe, the problem is here... > > So, now I know what was the problem and I solved it... > > The proble

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

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...)

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...)

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. Probably there will be another problem. I expect >>> packet size 1466 would pass and higher will have the same result. It RTP-VoIP-packets never reach this size. Siz

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

Am 22.06.20 um 16:48 schrieb Luca Bertoncello: Hi list! So, now I have a business contract and a technician was here to check the DSL... Nothing found, except that for 50Mbps I need now vectoring. Really nice... A couple of years ago I could get 50Mbps without vectoring. Of course, Deutsche Tele

Re: [asterisk-users] CLI color prompt

On 31.05.20 at 19:26 Jeff LaCoursiere wrote: > Hi, > > I had posted this a few hours ago, but got caught in moderation for size.  I > trimmed down the pic and attached. > > I am on an Ubuntu 16 workstation, in an Ubuntu terminal window, ssh'ed to the > PBX (amazon instance).  You can see my ter

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

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[

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

On 15.05.20 at 14:31 Doug Lytle wrote: > Google says Round Trip Time > > https://www.voip-info.org/asterisk-rtcp/ That doesn't answer my question (I know the abbreviation RTT). Therefore I'm trying again: I'm just wondering what the RTT *exactly* means. Where are the exact measuring points loc

[asterisk-users] Meaning of RTT in channelstats

Hello! I'm just wondering what the RTT exactly means. Where are the exact measuring points located? > pjsip show channelstats ...Receive. .Transmit.. BridgeId ChannelId UpTime.. Codec. CountLost Pct J

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

Hello all! I'm reproducibly getting my *own MOH* if I should get the MOH of the Callee instead. I can see this with asterisk 13 and 16 (and probably before, too). The reason of the wrong MOH is an in dialog reInvite received from trunk, which sends a SDP containing a=sendonly After this reInv

Re: [asterisk-users] FREEPBX Mailinglist

On 11.09.19 at 15:24 Joshua C. Colp wrote: > On Wed, Sep 11, 2019, at 10:18 AM, basti wrote: >> Hallo, >> is there a Freepbx mailinglist? or can this be posted here? > > FreePBX does not have a mailing list. People use the community forum[1] > instead. > > [1] https://community.freepbx.org/ Thi

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

Hello! I encountered an outage of asterisk which showed like that: - 2019-08-10 07:22:21 Asterisk start - 2019-08-15 19:39:33 WARNING taskprocessor.c: The 'pjsip/outreg/ispPJSIP-0060' task processor queue reached 500 scheduled tasks. - 2019-08-15 19:39:34 WARNING res_pjsip_outbound_regist

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

On 14.08.19 at 18:12 Dan Cropp wrote: Maybe because the machine is performing a file system check on some other partitions in parallel and it's slowed down therefore? Wouldn't /var/log/syslog show something like this if it's happening in parallel? Well, it was just speculation. Is it even

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

On 14.08.19 at 16:26 Dan Cropp wrote: > We have a customer where their VM running Asterisk appears to have crashed. > Fortunately, we had some debugging enabled. > The asterisk messages file has this... (in notepad+ the blank line in the > middle is all [NUL][NUL] [NUL][NUL]) > > [08/12 15:

[asterisk-users] Wanted: professional softphone

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 Pro

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

Hello! Following problem: If there are different trunks (-> different numbers and users / passwords) to the same destination, asterisk (16.x) always uses the same local tcp port for each connection. This is a problem with Deutsche Telekom (they want to have different local ports for different u

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

On 06.07.19 at 22:16 hwilmer wrote: Is there an advantage to using pjsip?  What's needed for easybell with pjsip? For easybell, I don't know of any advantage. But that's not very reliable, because I'm using easybell for dedicated requirements only. I'm considering chan_sip legacy. I wouldn't b

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

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?

On 05.07.19 at 22:02 hw wrote: > > openssl verify -CAfile ca.pem asterisk.pem > asterisk.pem: OK > > > When I set tlsdontverifyserver=yes, it works (i. e. asterisk registers > to the SIP provider and there is no error message).  Otherwise I'm > getting the error message and asterisk does not reg

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

On 11.06.19 at 20:32 Luca Bertoncello wrote: > Hi list! > > I use Asterisk 13.14.1 from Debian repository on a DSL from Deutsche > Telekom. > > Asterisk works well, but I have really often an high delay (I understand > it since the other party speak some seconds before he hears my question > and

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

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 -- _ -- Bandwidth and

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

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] Voice mail: MWI problem / pjsip (13.24.0)

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 MWI not turning off after a message > has been listened t

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

On 27.12.18 at 18:14 Joshua C. Colp wrote: > On Thu, Dec 27, 2018, at 1:07 PM, Michael Maier wrote: >> Hi! >> >> I just want to say, that 13.24.1 doesn't fix the problem described in >> the posts above. > > You're going to need to file an issue[1] wi

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

Hi! I just want to say, that 13.24.1 doesn't fix the problem described in the posts above. Regards, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community foru

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

On 26.12.18 at 14:48 Michael Maier wrote: > Hello! > > I'm facing a problem concerning MWI. The problem is: > > The phone switches off the MWI exactly at the moment the second NOTIFY > package for one voice mail > arrives. > > The phone switches off MWI independ

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

Hello! I'm facing a problem concerning MWI. The problem is: The phone switches off the MWI exactly at the moment the second NOTIFY package for one voice mail arrives. The phone switches off MWI independently if the voice mail has been acknowledged or not before. The behavior in detail: - On

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

On 12.12.18 at 19:43 Joshua C. Colp wrote: > On Wed, Dec 12, 2018, at 12:31 PM, Michael Maier wrote: > > > >> >> The problem: The extension doesn't create a ringback locally, because >> it most probably expects it to >> be sent by the callee -

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

On 12.12.18 at 19:43 Joshua C. Colp wrote: > On Wed, Dec 12, 2018, at 12:31 PM, Michael Maier wrote: > > > >> >> The problem: The extension doesn't create a ringback locally, because >> it most probably expects it to >> be sent by the callee -

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

Hello! An extension registered at asterisk 13.23 initiates an external call (pjsip). After the Invite, the callee (-> ISP) sends a 100 Trying 183 Session Progress (*without* SDP) Asterisk now sends to the extension: 183 Session Progress (*with* SDP) 183 Session

Re: [asterisk-users] pjsip: TOS not working any more

On 08/09/2018 at 10:49 PM Michael Maier wrote: > Hello! > > I'm using TOS as shown below with pjsip 13.22.0-rc1 (same with 13.21.1). > Unfortunately, the TOS isn't set in reality any more (it used to work > some time ago). Got the problem. I'm using freepbx and for

[asterisk-users] pjsip: TOS not working any more

Hello! I'm using TOS as shown below with pjsip 13.22.0-rc1 (same with 13.21.1). Unfortunately, the TOS isn't set in reality any more (it used to work some time ago). Transport: == Transport:

Re: [asterisk-users] res_fax_spandsp - information about used protocol t38 or g711?

On 05/21/2018 at 06:46 PM Andre Gronwald wrote: > after completion you find ${FAXMODE} filled with audio or T38, depending > on what has been used. > hope that is what you are looking for. Yes, that's exactly what I've been looking for! Thanks, Michael -- __

[asterisk-users] res_fax_spandsp - information about used protocol t38 or g711?

Hello! I'm working on a fax solution, which reports the result of each sent or received fax to the database. When using option "F" or not using option "f", it's clear which protocol has been used. But if "f" is used, it could be g711 or t38. Is there any variable which contains this information a

[asterisk-users] Search for (multi tenant) fax to mail solution

Hello! I'm just searching for a fax to email / email to fax open source based complete solution which covers the following core features: - high availability - possibly multi tenant - about 40,000 users - about 100 lines parallel - supports G.711 / T.38 - scriptable user management (reading, addi

Re: [asterisk-users] What does pct mean?

d Communication by Commend > > FN 178618z | LG Salzburg > > -Ursprüngliche Nachricht- > Von: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Michael Maier > Gesendet: Montag, 12. Februar 2018 17:46 > An: asterisk-us

Re: [asterisk-users] What does pct mean?

16641K 8809 0.000 0.026 => This doesn't sound reliable to me: there are 188K packets and 16641K of them are lost?! The Pct value is fluctuating between about 6009 and 9009. Thanks, Michael > > > Am 11.02.2018 19:27 schrieb "Michael Maier" : > >> Hello,

[asterisk-users] What does pct mean?

Hello, could somebody please tell me the meaning of "Pct" as seen in asterisk cli: ...Receive. .Transmit.. CountLost Pct Jitter CountLost Pct Jitter RTT Thanks, Michael -- _ --

Re: [asterisk-users] asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision

On 11/01/2017 at 10:14 AM Antony Stone wrote: > On Wednesday 01 November 2017 at 08:10:36, Michael Maier wrote: > >> Hello! >> >> I'm facing the following scenario: >> >> - Initial call opened to asterisk: SDP g722,alaw,ulaw >> >> - Outgoing

[asterisk-users] asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision

Hello! I'm facing the following scenario: - Initial call opened to asterisk: SDP g722,alaw,ulaw - Outgoing call to provider started with Invite / SDP alaw, g726 and g729. - Provider sends 183 Session progress SDP: g729, alaw - Provider sends g729 rtp packages But: there is no license to tr

Re: [asterisk-users] asterisk 13.16. - sigseg during negotiation

On 06/18/2017 at 12:11 PM, Joshua Colp wrote: > On Sun, Jun 18, 2017, at 06:00 AM, Michael Maier wrote: >> Hello! >> >> unchanged asterisk crashes during udptl / t.38 negotiation with telekom >> - they do not support t.38 / udptl. > > All Asterisk issues need

[asterisk-users] asterisk 13.16. - sigseg during negotiation

Hello! unchanged asterisk crashes during udptl / t.38 negotiation with telekom - they do not support t.38 / udptl. In detail: fax client -> asterisk -> telekom -> easybell -> asterisk -> fax server Fax server sends t.38 reinvite via asterisk to easybell. Session Description Protocol Versio

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/17/2017 at 02:18 PM, Michael Maier wrote: > On 06/16/2017 at 04:00 PM, Joshua Colp wrote: >> On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote: >> >> >> >>> >>> t38modem and asterisk are using >>> >>> m=image 35622 udp

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/16/2017 at 04:00 PM, Joshua Colp wrote: > On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote: > > > >> >> t38modem and asterisk are using >> >> m=image 35622 udptl t38 >>^ >> >> Provider uses >> >> m

Re: [asterisk-users] pjsip: asterisk can't decide which codec to use

On 05/13/2017 at 07:21 AM Michael Maier wrote: > On 05/12/2017 at 08:49 PM, Joshua Colp wrote: >> On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote: >> >> >> >>> >>> If I'm doing exactly the same call originated with another extension, >&

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

Am 16.06.2017 um 11:12 schrieb Joshua Colp: On Fri, Jun 16, 2017, at 02:13 AM, Michael Maier wrote: Has anybody any idea why asterisk drops the media stream in the 200 OK? The channel has been T38_ENABLED before! Or is it necessary to add more debug code? Who does the negotiating? Only

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/15/2017 at 08:15 AM Michael Maier wrote: > On 06/14/2017 at 10:17 PM, Joshua Colp wrote: >> On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote: >> >> >> >>> >>> I can now say, that asterisk / pjsip seams to work *mostly* as expected. >&

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/14/2017 at 10:17 PM, Joshua Colp wrote: > On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote: > > > >> >> I can now say, that asterisk / pjsip seams to work *mostly* as expected. >> Just one exception - and that's the package in question, which c

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/14/2017 at 05:53 PM Joshua Colp wrote: > On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote: > > > >> >> I added this patch to see, if really all packages are are freed after >> they have been processed: >> >> --- b/res/res_pjsip/pjsip_d

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/11/2017 at 06:51 PM Joshua Colp wrote: > On Sun, Jun 11, 2017, at 01:47 PM, Joshua Colp wrote: >> The distributor is in res/res_pjsip/pjsip_distributor.c, the distributor >> function being the entry point. That function returning PJ_TRUE >> indicates to PJSIP that it has been handled and no s

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/11/2017 at 11:35 PM Daniel Tryba wrote: > On Sun, Jun 11, 2017 at 01:16:10PM +0200, Michael Maier wrote: >> Let's go into details: >> I'm starting at the point where asterisk / fax client receives the T.38 >> reininvite from the server from the provider 19

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/11/2017 at 04:34 PM Michael Maier wrote: > On 06/11/2017 at 01:29 PM Joshua Colp wrote: >> On Sun, Jun 11, 2017, at 08:16 AM, Michael Maier wrote: >> >> >> >>> I did some further investigations and fixed a local problem. Now, >>> a

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/11/2017 at 04:39 PM Joshua Colp wrote: > On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote: > > > >>> >>> PJSIP uses a dispatch model. The request is queued up, acted on, and >>> then that's it. The act of acting on it removes it from the

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/11/2017 at 01:29 PM Joshua Colp wrote: > On Sun, Jun 11, 2017, at 08:16 AM, Michael Maier wrote: > > > >> I did some further investigations and fixed a local problem. Now, >> asterisk is able to successfully activate T.38 - unfortunately just for >> very

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/05/2017 at 09:32 PM Joshua Colp wrote: > On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote: >> On 06/05/2017 at 06:29 PM, Joshua Colp wrote: >>> On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: >>>> >>>> Do you have any idea where to

Re: [asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers

On 06/09/2017 at 08:44 PM Michael Maier wrote: > On 06/08/2017 at 10:22 PM Michael Maier wrote: >> Hello Joshua, >> >> thank you very much for your extremely quick answer! I really appreciate >> your work and your friendly and your patient support! >> >> &g

Re: [asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers

On 06/08/2017 at 10:22 PM Michael Maier wrote: > Hello Joshua, > > thank you very much for your extremely quick answer! I really appreciate > your work and your friendly and your patient support! > > > On 06/07/2017 at 10:33 PM, Joshua Colp wrote: >> On Wed, Jun 7,

Re: [asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers

Hello Joshua, thank you very much for your extremely quick answer! I really appreciate your work and your friendly and your patient support! On 06/07/2017 at 10:33 PM, Joshua Colp wrote: > On Wed, Jun 7, 2017, at 05:28 PM, Michael Maier wrote: >> Hello! >> >> I've g

[asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers

Hello! I've got a problem to select the correct trunk if there is one provider and different numbers with different configurations for this same provider. Example: trunk-prov1-2345 trunk-prov1-2346 trunk-prov1-2347 Each trunk registers an own number (at the same provider) and provides own confi

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/05/2017 at 06:29 PM, Joshua Colp wrote: > On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: >> >> Do you have any idea where to start to look at? Adding additional output >> in the source code? Which functions could be interesting? I may add own >> debug

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/05/2017 at 06:10 PM, Joshua Colp wrote: > On Mon, Jun 5, 2017, at 12:00 PM, Joshua Colp wrote: >> On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote: >>> On 06/05/2017 at 11:30 AM, Joshua Colp wrote: >>>> On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/05/2017 at 05:00 PM, Joshua Colp wrote: > On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote: >> On 06/05/2017 at 11:30 AM, Joshua Colp wrote: >>> On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >>>> On 06/04/2017 at 01:41 PM Telium Technical Sup

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>> Just a guess (without knowing about your network), but are the two ends >>> points on public networks

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/04/2017 at 01:41 PM Telium Technical Support wrote: > Just a guess (without knowing about your network), but are the two ends > points on public networks and visible to one another? If not the reinvite > may be passing an internal (nat'ed) address to the other and the connection > will fail.

[asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

Hello! I'm still trying to get a working t.38 configuration w/ pjsip. I'm now able to send t.38 faxes to my own extension: hylafax -> t38modem -> extension -> extension -> t38modem -> hylafax. The fax is sent by t38modem. The receiving part of t38modem accepts the call, sends ReInvite for t.3

Re: [asterisk-users] pjsip: asterisk can't decide which codec to use

On 05/12/2017 at 08:49 PM, Joshua Colp wrote: > On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote: > > > >> >> If I'm doing exactly the same call originated with another extension, >> there can't be seen these frequent changes. But the strange

Re: [asterisk-users] pjsip: asterisk can't decide which codec to use

On 05/12/2017 at 07:46 PM, Michael Maier wrote: Forgot to mention: It's actual asterisk 13 branch from today (version before I tested, which has the same problem, was 13.15). Regards, Michael Hello! I'm facing completely choppy sound. The wireshark trace shows, that there ar

[asterisk-users] pjsip: asterisk can't decide which codec to use

Hello! I'm facing completely choppy sound. The wireshark trace shows, that there are a lot of codec changes without any trigger (means no options or reinvite or any other package). Background: The call is initiated by asterisk and is received by the same asterisk conference room via Phone extensi

[asterisk-users] Some questions regarding jitterbuffer in asterisk / pjsip

Hello! I just implemented a jitterbuffer for pjsip in the dialplan in a SBC: [fromtrunk] exten => _[+0-9]!,1,Set(JITTERBUFFER(fixed)=default) This jitterbuffer catches all calls coming from ISP. My understanding is, that the incoming rtp stream in leg1a is now buffered and handed out "jitter-o

Re: [asterisk-users] Outbound T.38 via RTP with pjsip does not work as expected

On 04/06/2017 at 08:33 PM Joshua Colp wrote: > On Thu, Apr 6, 2017, at 03:15 PM, Michael Maier wrote: >> Hello! >> >> I'm trying to send a fax via T.38 to a destination, which should be T.38 >> capable. My provider supports T.38, too. Unfortunately, it doesn'

Re: [asterisk-users] Outbound T.38 via RTP with pjsip does not work as expected

On 04/06/2017 at 08:33 PM, Joshua Colp wrote: > On Thu, Apr 6, 2017, at 03:15 PM, Michael Maier wrote: >> Hello! >> >> I'm trying to send a fax via T.38 to a destination, which should be T.38 >> capable. My provider supports T.38, too. Unfortunately, it doesn'

[asterisk-users] Outbound T.38 via RTP with pjsip does not work as expected

Hello! I'm trying to send a fax via T.38 to a destination, which should be T.38 capable. My provider supports T.38, too. Unfortunately, it doesn't work. This means: Call is started and SDP is negotiated w/ alaw. Callee sends reinvite - for alaw again (and not for T.38)!! After about 30s, callee h

Re: [asterisk-users] Asterisk 13.13.1

On 01/30/2017 at 05:55 PM Motty Cruz wrote: > Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here: > http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz > > > > > I continue to see errors like this: > > [2017-01-30 08:37:17] WARNIN

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