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From where can I download FreePBX 15 announced on the 31st of October?

November 11, 2019, 7:13 am
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Sip Trunk Deutsche Telekom

November 11, 2019, 7:26 am
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my Server URI in the trunks PJSIP settings - advanced now looks as follows:
sip:xxxxxxxxxxx@reg.sip-trunk.telekom.de:5060“;transport=tcp”

I still get an
[2019-11-11 16:23:45] WARNING[14701]: pjsip:0 <?>: tsx0x7fde84e78 Failed to send Request msg REGISTER/cseq=6890 (tdta0x7fde84003d70)! err=320053 (DNS “Name Error” (PJLIB_UTIL_EDNS_NXDOMAIN))
[2019-11-11 16:23:45] WARNING[28479]: res_pjsip_outbound_registration.c:751 schedule_retry: No response received from ‘sip:xxxxxxxxxxxx@reg.sip-trunk.telekom.de:5060’ on registration attempt to ‘sip:+4923190220@sip-trunk.telekom.de’, retrying in ‘60’
[2019-11-11 16:24:01] WARNING[28479]: pjsip:0 <?>: tsx0x7fde84e78 .Failed to send Request msg OPTIONS/cseq=50514 (tdta0x7fde84003d70)! err=320047 (No answer record in the DNS response (PJLIB_UTIL_EDNSNOANSWERREC))
[2019-11-11 16:24:01] ERROR[28479]: res_pjsip.c:3167 endpt_send_request: Error 320047 ‘No answer record in the DNS response (PJLIB_UTIL_EDNSNOANSWERREC)’ sending OPTIONS request to endpoint

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Audio problem between two sip trunk interfaces and wan / lan

November 11, 2019, 7:27 am
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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

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Sip Trunk Deutsche Telekom

November 11, 2019, 7:27 am
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Discourse apparently modified what I said. The correct format to add is:

\;transport=tcp
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Sip Trunk Deutsche Telekom

November 11, 2019, 7:30 am
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ok, thanks, now it looks like this
sip:551134788732@reg.sip-trunk.telekom.de:5060;transport=tcp

CLI shows:
[2019-11-11 16:28:25] WARNING[28479]: pjsip:0 <?>: tsx0x7fde84e78 .Failed to send Request msg OPTIONS/cseq=35129 (tdta0x7fde84003d70)! err=320047 (No answer record in the DNS response (PJLIB_UTIL_EDNSNOANSWERREC))
[2019-11-11 16:28:25] ERROR[28479]: res_pjsip.c:3167 endpt_send_request: Error 320047 ‘No answer record in the DNS response (PJLIB_UTIL_EDNSNOANSWERREC)’ sending OPTIONS request to endpoint
[2019-11-11 16:28:26] WARNING[28479]: pjsip:0 <?>: tsx0x7fde84e78 .Failed to send Request msg REGISTER/cseq=7017 (tdta0x7fde84003d70)! err=320053 (DNS “Name Error” (PJLIB_UTIL_EDNS_NXDOMAIN))
[2019-11-11 16:28:26] WARNING[28479]: res_pjsip_outbound_registration.c:751 schedule_retry: No response received from ‘sip:xxxxxxxxxxxx@reg.sip-trunk.telekom.de:5060;transport=tcp’ on registration attempt to ‘sip:+49zzzzzzzz@sip-trunk.telekom.de’, retrying in ‘60’

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Sip Trunk Deutsche Telekom

November 11, 2019, 7:31 am
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Nothing stands out, so it may be a limitation of the PJSIP SRV support in Asterisk 13 holding it back.

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Sip Trunk Deutsche Telekom

November 11, 2019, 7:38 am
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Thanks a lot Joshua! I still got the wrong DNS query

_sip._udp.reg.sip-trunk.telekom.de: type SRV, class IN

I guess we have to stick with the chan_sip trunk for now and work on a pjsip solution. And this seems to require Asterisk 16. So I have to setup a new freePBX system and prepare a migration.

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Sip Trunk Deutsche Telekom

November 11, 2019, 7:38 am
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Using the following:

server_uri=sip:reg.sip-trunk.telekom.de\;transport=tcp

With no “transport” specified worked as expected.

|[Nov 11 15:38:16] DEBUG[6809]: pjproject: <?>: |         sip_resolve.c .Starting async DNS SRV query: target=_sip._tcp.reg.sip-trunk.telekom.de, transport=TCP, port=0|

|—|---|
|[Nov 11 15:38:16] DEBUG[6809]: pjproject: <?>: |_sip._tcp.reg.sip-trun .Starting async DNS SRV query_job: target=_sip._tcp.reg.sip-trunk.telekom.de:5060|
|[Nov 11 15:38:16] DEBUG[6809]: pjproject: <?>: | resolver.c .Nameserver 172.16.1.1:53 state changed Active --> Probing|
|[Nov 11 15:38:16] DEBUG[6809]: pjproject: <?>: | resolver.c .Transmitting 52 bytes to NS 0 (172.16.1.1:53): DNS SRV query for _sip._tcp.reg.sip-trunk.telekom.de: Success|
|[Nov 11 15:38:16] DEBUG[6809]: pjproject: <?>: | tsx0x7f5370012288 .State changed from Null to Calling, event=TX_MSG|
|[Nov 11 15:38:17] DEBUG[6808]: pjproject: <?>: | resolver.c Received 280 bytes DNS response from 172.16.1.1:53|
|[Nov 11 15:38:17] DEBUG[6808]: pjproject: <?>: | resolver.c Nameserver 172.16.1.1:53 state changed Probing --> Active|
|[Nov 11 15:38:17] DEBUG[6808]: pjproject: <?>: |_sip._tcp.reg.sip-trun SRV query_job for _sip._tcp.reg.sip-trunk.telekom.de completed, 3 of 3 total entries selected:|
|[Nov 11 15:38:17] DEBUG[6808]: pjproject: <?>: |_sip._tcp.reg.sip-trun 0: SRV 0 5 5060 n-ipr-a01.sip-trunk.telekom.de (217.0.26.67)|
|[Nov 11 15:38:17] DEBUG[6808]: pjproject: <?>: |_sip._tcp.reg.sip-trun 1: SRV 1 5 5060 n-ipr-a02.sip-trunk.telekom.de (217.0.26.69)|
|[Nov 11 15:38:17] DEBUG[6808]: pjproject: <?>: |_sip._tcp.reg.sip-trun 2: SRV 10 5 5060 d-ipr-a01.sip-trunk.telekom.de (217.0.26.131)|
|[Nov 11 15:38:17] DEBUG[6808]: pjproject: <?>: |_sip._tcp.reg.sip-trun Server resolution complete, 3 server entry(s) found|
|[Nov 11 15:38:17] DEBUG[6808]: pjproject: <?>: | tcpc0x7f52d400db58 TCP client transport created|
|[Nov 11 15:38:17] DEBUG[6808]: pjproject: <?>: | tcpc0x7f52d400db58 TCP transport 172.16.10.11:40903 is connecting to 217.0.26.67:5060…|

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Sip Trunk Deutsche Telekom

November 11, 2019, 7:48 am
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That actually gives me some hope. Unfortunately I don’t know how to unset the transport.

pjsip settings general:
Transport 0.0.0.0-tcp
I can’t deselect or unset this in the GUI

Update:
I just tried it by disabling the tcp transport under SIP Settings - Chan PJSIP Settings
CLI and tcpdump shows the same

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Porque no muestra la opción de copias de seguridad

November 11, 2019, 7:57 am
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Sip Trunk Deutsche Telekom

November 11, 2019, 8:04 am
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Did you also remove the port? What’s the actual entry in the PJSIP configuration file?

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Porque no muestra la opción de copias de seguridad

November 11, 2019, 8:19 am
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Que tal.
Probablemente no este instalado el modulo.

Lo puedes hacer mediante el comando: fwconsole ma downloadinstall backup
Despues ejecutas : fwconsole reload

Saludos.

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Sip Trunk Deutsche Telekom

November 11, 2019, 8:20 am
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Yes, I also tried removing the port. Didn’t help.

Going through the generated pjsip config files I found so far:

/etc/asterisk/pjsip.transports.conf includes pjsip.transports_custom.conf, which is empty for now

/etc/asterisk/pjsip.registration.conf lists the trunk
[telekom-pjsip]
type=registration
transport=
outbound_auth=telekom-pjsip
retry_interval=60
max_retries=10
expiration=3600
auth_rejection_permanent=yes
contact_user=+49zzzzzzzz
server_uri=sip:xxxxxxxxxxxx@reg.sip-trunk.telekom.de;transport=tcp
client_uri=sip:+49zzzzzzzz@sip-trunk.telekom.de
outbound_proxy=sip:reg.sip-trunk.telekom.de

/etc/asterisk/pjsip.identify.conf
[telekom-pjsip]
type=identify
endpoint=telekom-pjsip
match=217.0.0.0/13

/etc/asterisk/pjsip.endpoint.conf (and there we have the problem, transport=udp)
[telekom-pjsip]
type=endpoint
transport=udp
context=from-pstn-toheader-de-do
disallow=all
allow=g722,g726,alaw,ulaw,gsm
aors=telekom-pjsip
language=en
outbound_proxy=sip:reg.sip-trunk.telekom.de
outbound_auth=telekom-pjsip
from_domain=sip-trunk.telekom.de
t38_udptl=no
t38_udptl_ec=none
fax_detect=no
t38_udptl_nat=no
dtmf_mode=rfc4733

[dpma_endpoint]
type=endpoint
context=dpma-invalid

auth looks good, so I continue with
/etc/asterisk/pjsip.aor.conf (seems to use the server port from the trunks pjsip settings general)
[telekom-pjsip]
type=aor
qualify_frequency=60
contact=sip:xxxxxxxxxxxxx@reg.sip-trunk.telekom.de:5060
outbound_proxy=sip:reg.sip-trunk.telekom.de

/etc/asterisk/pjsip.conf
…
[global]
type=global
user_agent=FPBX-13.0.190.19(13.7.1)
default_outbound_endpoint=dpma_endpoint
#include pjsip_custom_post.conf

How to solve the wrong transport setting in pjsip.endpoint.conf? I’d put “transport=tcp” in pjsip.endpoint_custom.conf like this
[telekom-pjsip]
transport=tcp
but I expect this not to work, since it will be overwritten after the inclusion by “transport=udp” in pjsip.endpoint.conf

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Sip Trunk Deutsche Telekom

November 11, 2019, 8:25 am
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You’d need to set it on the outbound_proxy instead. I did not know you had one set.

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Hash/Pound Key Not recognized in IVR anymore

November 11, 2019, 8:28 am
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Hi all, this is my first post.
we have an IVR running on 14.0.9.4. Seemingly, a couple of days ago, when someone presses the Hash “#” or Pound key as one of the options to the directory, the Announcement states “we have not received a valid response”

the Pcap registers a # in the RTPEVENT, but in the log i see this.

[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [1@timeconditions:35] GotoIfTime(“SIP/P1HANC_TRUNK-0000c98c”, “00:00-23:59,*,10,jun?truestate”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [1@timeconditions:36] GotoIf(“SIP/P1HANC_TRUNK-0000c98c”, “0?truegoto”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [1@timeconditions:37] ExecIf(“SIP/P1HANC_TRUNK-0000c98c”, “0?Set(DB(TC/1)=)”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [1@timeconditions:38] Set(“SIP/P1HANC_TRUNK-0000c98c”, “DEVICE_STATE(Custom:TC1)=INUSE”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [1@timeconditions:39] ExecIf(“SIP/P1HANC_TRUNK-0000c98c”, “0?Set(DEVICE_STATE(Custom:TCSTICKY)=INUSE)”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [1@timeconditions:40] GotoIf(“SIP/P1HANC_TRUNK-0000c98c”, “1?ivr-4,s,1”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx_builtins.c: Goto (ivr-4,s,1)
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [s@ivr-4:1] Set(“SIP/P1HANC_TRUNK-0000c98c”, “INVALID_LOOPCOUNT=0”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [s@ivr-4:2] Set(“SIP/P1HANC_TRUNK-0000c98c”, “_IVR_CONTEXT_ivr-4=”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [s@ivr-4:3] Set(“SIP/P1HANC_TRUNK-0000c98c”, “_IVR_CONTEXT=ivr-4”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [s@ivr-4:4] Set(“SIP/P1HANC_TRUNK-0000c98c”, “__IVR_RETVM=”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [s@ivr-4:5] GotoIf(“SIP/P1HANC_TRUNK-0000c98c”, “0?skip”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [s@ivr-4:6] Answer(“SIP/P1HANC_TRUNK-0000c98c”, “”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [s@ivr-4:7] Set(“SIP/P1HANC_TRUNK-0000c98c”, “IVR_MSG=custom/100_HANCElementaryMainAA_updated”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [s@ivr-4:8] Set(“SIP/P1HANC_TRUNK-0000c98c”, “TIMEOUT(digit)=5”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] func_timeout.c: Digit timeout set to 5.000
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] pbx.c: Executing [s@ivr-4:9] Read(“SIP/P1HANC_TRUNK-0000c98c”, “IVREXT,custom/100_HANCElementaryMainAA_updated,0,5”) in new stack
[2019-11-11 11:09:20] VERBOSE[25062][C-0000937b] file.c: <SIP/P1HANC_TRUNK-0000c98c> Playing ‘custom/100_HANCElementaryMainAA_updated.slin’ (language ‘en’)
**

> [2019-11-11 11:09:22] VERBOSE[25062][C-0000937b] app_read.c: User entered nothing.

**
[2019-11-11 11:09:22] VERBOSE[25062][C-0000937b] pbx.c: Executing [s@ivr-4:10] GotoIf(“SIP/P1HANC_TRUNK-0000c98c”, “0?t,1”) in new stack
[2019-11-11 11:09:22] VERBOSE[25062][C-0000937b] pbx.c: Executing [s@ivr-4:11] ExecIf(“SIP/P1HANC_TRUNK-0000c98c”, “0?Set(LOCALEXT=1)”) in new stack
[2019-11-11 11:09:22] VERBOSE[25062][C-0000937b] pbx.c: Executing [s@ivr-4:12] GotoIf(“SIP/P1HANC_TRUNK-0000c98c”, “1?i,1”) in new stack
[2019-11-11 11:09:22] VERBOSE[25062][C-0000937b] pbx_builtins.c: Goto (ivr-4,i,1)
[2019-11-11 11:09:22] VERBOSE[25062][C-0000937b] pbx.c: Executing [i@ivr-4:1] Set(“SIP/P1HANC_TRUNK-0000c98c”, “INVALID_LOOPCOUNT=1”) in new stack
[2019-11-11 11:09:22] VERBOSE[25062][C-0000937b] pbx.c: Executing [i@ivr-4:2] GotoIf(“SIP/P1HANC_TRUNK-0000c98c”, “0?final”) in new stack
[2019-11-11 11:09:22] VERBOSE[25062][C-0000937b] pbx.c: Executing [i@ivr-4:3] Set(“SIP/P1HANC_TRUNK-0000c98c”, “IVR_MSG=no-valid-responce-pls-try-again&custom/100_HANCElementaryMainAA_updated”) in new stack
[2019-11-11 11:09:22] VERBOSE[25062][C-0000937b] pbx.c: Executing [i@ivr-4:4] Goto(“SIP/P1HANC_TRUNK-0000c98c”, “s,start”) in new stack
[2019-11-11 11:09:22] VERBOSE[25062][C-0000937b] pbx_builtins.c: Goto (ivr-4,s,8)
[2019-11-11 11:09:22] VERBOSE[25062][C-0000937b] pbx.c: Executing [s@ivr-4:8] Set(“SIP/P1HANC_TRUNK-0000c98c”, “TIMEOUT(digit)=5”) in new stack
[2019-11-11 11:09:22] VERBOSE[25062][C-0000937b] func_timeout.c: Digit timeout set to 5.000
[2019-11-11 11:09:22] VERBOSE[25062][C-0000937b] pbx.c: Executing [s@ivr-4:9] Read(“SIP/P1HANC_TRUNK-0000c98c”, “IVREXT,no-valid-responce-pls-try-again&custom/100_HANCElementaryMainAA_updated,0,5”) in new stack
[2019-11-11 11:09:22] VERBOSE[25062][C-0000937b] file.c: <SIP/P1HANC_TRUNK-0000c98c> Playing ‘no-valid-responce-pls-try-again.slin’ (language ‘en’)
[2019-11-11 11:09:23] VERBOSE[25062][C-0000937b] file.c: <SIP/P1HANC_TRUNK-0000c98c> Playing ‘custom/100_HANCElementaryMainAA_updated.slin’ (language ‘en’)
[2019-11-11 11:09:23] VERBOSE[25062][C-0000937b] app_read.c: User disconnected

as you can see this states that i haven’t entered any digits. this just inexplicably stopped working.

i believe this is the latest ivr verision
thanks

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Sip Trunk Deutsche Telekom

November 11, 2019, 8:37 am
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I could only get rid of “transport=udp” in /etc/asterisk/pjsip.endpoint.conf by setting transport tcp in general sip settings pjsip settings within the GUI, resulting in

[user@pbx]# grep -ri “transport=” /etc/asterisk/pjsip*
/etc/asterisk/pjsip.aor.conf:outbound_proxy=sip:reg.sip-trunk.telekom.de;transport=tcp
/etc/asterisk/pjsip.endpoint.conf:transport=0.0.0.0-tcp
/etc/asterisk/pjsip.endpoint.conf:outbound_proxy=sip:reg.sip-trunk.telekom.de;transport=tcp
/etc/asterisk/pjsip.registration.conf:transport=0.0.0.0-tcp
/etc/asterisk/pjsip.registration.conf:server_uri=sip:xxxxxxxxxxxxx@reg.sip-trunk.telekom.de:5060;transport=tcp
/etc/asterisk/pjsip.registration.conf:outbound_proxy=sip:reg.sip-trunk.telekom.de;transport=tcp

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Calendar module

November 11, 2019, 8:43 am
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All calendar modes use the iCal parser. The bug was recently fixed in the parser this last weekend so now it just needs to be applied to the calendar module.

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No ACK to Reinvite

November 11, 2019, 9:48 am
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Running FreePBX 14.0.13.6

We are seeing our SIP trunk provider is sending us re-invites at 30-minute intervals but FreePBX isn’t sending back an ACK so the SIP trunk provider is terminating the call. Essentially the call drops right at 30-minutes.

What should I be looking for in this instance to resolve this?

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Filestore , S3

November 11, 2019, 10:09 am
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Recordings would have to be done via script of some sort This would have to be patched in.

Essentially asterisk would record to a drive, then you would move that file to a cloud location. Then anything that references that recording would need to know the file location to pull it down as needed.

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Filestore , S3

November 11, 2019, 10:11 am
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