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After call from GSM to Asterisk server to direct extension have problem

February 13, 2019, 10:55 pm
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@hunterman wrote:

Hello everyone

Am to test to call from gsm mobile to Asterisk server to call direct extension like 6000,
I have ringing between GSM and Astersik but the problem when Am answer from extension the call go to be disconnect automatic why?

when disconnect the ssh show me:

0x7f638891f220 – Strict RTP learning after remote address set to: 192.168.29.5:5008
[2019-02-14 08:57:15] WARNING[3555][C-00000088]: channel.c:5600 set_format: Unable to find a codec translation path: (ulaw) -> (g723)
[2019-02-14 08:57:15] WARNING[3555][C-00000088]: channel.c:5600 set_format: Unable to find a codec translation path: (g723) -> (ulaw)
– Connected line update to PJSIP/anonymous-00000044 prevented.
– SIP/6000-0000006b answered PJSIP/anonymous-00000044
0x7f6400915bb0 – Strict RTP learning after remote address set to: 192.168.33.169:8012
[2019-02-14 08:57:15] WARNING[12472]: channel.c:5600 set_format: Unable to find a codec translation path: (g723) -> (alaw)
[2019-02-14 08:57:15] WARNING[12472]: channel.c:5600 set_format: Unable to find a codec translation path: (ulaw) -> (g723)
– Channel SIP/6000-0000006b joined ‘simple_bridge’ basic-bridge
– Channel PJSIP/anonymous-00000044 joined ‘simple_bridge’ basic-bridge
[2019-02-14 08:57:15] WARNING[12478][C-00000088]: channel.c:6560 ast_channel_make_compatible_helper: No path to translate from PJSIP/anonymous-00000044 to SIP/6000-0000006b
– Channel PJSIP/anonymous-00000044 left ‘simple_bridge’ basic-bridge
– Channel SIP/6000-0000006b left ‘simple_bridge’ basic-bridge
== Spawn extension (macro-dial-one, s, 55) exited non-zero on ‘PJSIP/anonymous-00000044’ in macro ‘dial-one’
– SIP/6000-0000006b Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“SIP/6000-0000006b”, “Sending Hangup to CRM”) in new stack
== Spawn extension (macro-exten-vm, s, 19) exited non-zero on ‘PJSIP/anonymous-00000044’ in macro ‘exten-vm’
– Executing [s@crm-hangup:2] NoOp(“SIP/6000-0000006b”, “HANGUP CAUSE: 16”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“SIP/6000-0000006b”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
== Spawn extension (ext-local, 6000, 2) exited non-zero on ‘PJSIP/anonymous-00000044’
– Executing [h@ext-local:1] Macro(“PJSIP/anonymous-00000044”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/anonymous-00000044”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@crm-hangup:4] NoOp(“SIP/6000-0000006b”, “MASTER CHANNEL: 1550123825.209 = 1550123825.208”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“SIP/6000-0000006b”, “1?return”) in new stack
– Goto (crm-hangup,s,8)
– Executing [s@crm-hangup:8] Return(“SIP/6000-0000006b”, “”) in new stack
== Spawn extension (from-internal, , 1) exited non-zero on ‘SIP/6000-0000006b’
– SIP/6000-0000006b Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
– Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/anonymous-00000044”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“PJSIP/anonymous-00000044”, "SIP/6000-0000006b monior file= ") in new stack
– Executing [s@macro-hangupcall:5] AGI(“PJSIP/anonymous-00000044”, “attendedtransfer-rec-restart.php,SIP/6000-0000006b,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
– <PJSIP/anonymous-00000044>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:6] Hangup(“PJSIP/anonymous-00000044”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/anonymous-00000044’ in macro ‘hangupcall’
== Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/anonymous-00000044’
– PJSIP/anonymous-00000044 Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“PJSIP/anonymous-00000044”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“PJSIP/anonymous-00000044”, “HANGUP CAUSE: 16”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“PJSIP/anonymous-00000044”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“PJSIP/anonymous-00000044”, “MASTER CHANNEL: 1550123825.208 = 1550123825.208”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“PJSIP/anonymous-00000044”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“PJSIP/anonymous-00000044”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“PJSIP/anonymous-00000044”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <PJSIP/anonymous-00000044>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“PJSIP/anonymous-00000044”, “”) in new stack
== Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/anonymous-00000044’
– PJSIP/anonymous-00000044 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

note:
Unable to find a codec translation path: (ulaw) -> (g723),
in my astersik server Am enable the codec (ulaw) -> (g723),but my ip phone is Grandstream GXP2160

any help to solve it?
THANKS

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CDR reports stop working module are "not running"

February 14, 2019, 2:30 am
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@Arsalan wrote:

Hello guys, I got freepbx 13.0.192.16-1.shmz65.1.57 and all of sudden CDR stops working i checked master.csv file its also empty i can see all calls in “calls event logging” but CDR shows no results also when i run “module show like cdr” it shows something like this …

CLI Command

Module Description Use Count Status Support Level
app_cdr.so Tell Asterisk to not maintain a CDR for 0 Running core
app_forkcdr.so Fork The CDR into 2 separate entities 0 Running core
cdr_adaptive_odbc.so Adaptive ODBC CDR backend 0 Running core
cdr_manager.so Asterisk Manager Interface CDR Backend 0 Not Running core
cdr_odbc.so ODBC CDR Backend 0 Not Running extended
cdr_syslog.so Customizable syslog CDR Backend 0 Not Running core
func_cdr.so Call Detail Record (CDR) dialplan functi 0 Running core
7 modules loaded

i think those " not running" module are problem but i dont know how to activate them …

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How to Change Caller Announcement to a custom Recording

February 14, 2019, 7:16 am
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@sakbari wrote:

Dear Cummunity

I like to change the Caller Announcement freepbx caller announcements thank you for passion
for a systeem Recording that i have how can i do that

thanks in advance

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fx0 incoming call delay

February 14, 2019, 10:00 am
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@4ycr wrote:

I have FreePBX 14 with asterisk 13.22. I am using a Sangoma A200 FXO FXS Analogue Card PCI Express with 2 * Sangoma A400-O FXO Modules. The phones are snom DECT phones

I can get the system to call out but with an incoming call, it takes about 5 rings on the caller’s phone before the FreePBX phones ring. Is there a way to shorten this delay?

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GXW4108 PSTN Gateway - Only Outbound calls working

February 14, 2019, 12:34 pm
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@lordtron wrote:

FreePBX: 14.0.5.25
Asterisk: 13.19.1

I have everything up and running. The only issue I have ran into is that I can’t get the GXW4108 to work correctly with FreePBX on Inbound calls. Outbound works great.

TRUNK

Outgoing
Name: GXWT1
PEER:
type=friend
qualify=yes
secret=SECRET
host=IP_ADDR
context=from-trunk
port=5060
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw

Incoming
User Context: from-trunk
User Details:
type=peer
qualify=yes
secret=SECRET
host=IP_ADDR
context=from-trunk
port=5060
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw

Currently getting this error: “Rejecting unknown SIP connection from IP_ADDR”

Any help would be greatly appreciated.

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Moved CDR database and broke CDR report, please help

February 14, 2019, 2:03 pm
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@bksales wrote:

I shut down MySQL, moved everything to a new directory (on the same server), updated the my.cnf file, and restarted MySQL. It worked the first time I tested it but on the second one while you can see that the database is being updated the CDR report cant seem to access it. I made sure the directory and the database had the same permissions.

Any suggestions on where to look? Is there anywhere in the GUI to point the report at the right place?

Its an older version of freepbx, 10.13.66-20

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Can not add phones after update

February 14, 2019, 2:35 pm
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@Gordon0193 wrote:

I was running version - 12.7.3-1708-1.sng7

SSH into FreePBX and ran “yum update”

…
Install 8 Packages (+46 Dependent packages)
Upgrade 374 Packages

Total download size: 459 M

I selected ‘y’ to update.
…

When I got a completed message I rebooted the PBX

I show to be on - 12.7.5-1902-3.sng7

I have an existing extension but not an actual phone setup. I bring in the phone config and apply the password.

The PBX reports - [2019-02-14 17:05:08] NOTICE[10199] res_pjsip/pjsip_distributor.c: Request ‘REGISTER’ from ‘“J Smith” sip:305@192.168.1.10’ failed for ‘192.168.1.52:5060’ (callid: 0_1671914067@192.168.1.52) - Failed to authenticate

I go back to FreePBX 12.7.3-1708-1.sng7 (I’m using VM’s thus I’m using my older VM). Made no changes to the Phone. Start the 12.7.3-1708-1.sng7 version and the phone logs in w/o an issue.

The existing phone/extension was able to log into the updated version. Attaching a existing extension to a new phone fails.

Thoughts?

Thanks,

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Unlisted Extensions

February 14, 2019, 3:14 pm
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@kam_abbott wrote:

Hello all

I’m searching for a way to have some extensions not be listed in the contacts list on our Digium/Sangoma phones.

I feel like I must be missing something obvious but I can’t find a way to do it.

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DB20/D10M Distinctive Ring - Is it supported?

February 15, 2019, 6:09 am
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@skwlilac wrote:

Does anyone out there know if the the Sangoma DB20/D10M combination supports distinctive ringtones. I’m a bit of a newbie. I’ve had a look around but don’t seem to be able to find alert-info values that make a difference - and I’m wondering if the basestation/handsets simply don’t support it or whether I have yet to find the right config/incantation.

Thanks

Stuart

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User defined dialplan varaible

February 15, 2019, 7:05 am
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@comtech wrote:

I have a dial plan variable that I want to have the end user be able to update by changing a complex variable (long text string). I was thinking a webpage or email or something. Any ideas here?

The user would have control over a variable that is used for the body of a text message that gets sent out every time that number is dialed.

Does anyone have any ideas here?

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Change Extension Routing module to default permit instead of deny new route

February 15, 2019, 10:59 am
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@avayax wrote:

Using the Extension Routing module to permit/deny certain extensions to access specific routes.
When I add a new route extensions are denied that route by default. I have to manually enable it for all of them individually.
I want it to be the other way round.
On creating a new route, I want it to be enabled for all extensions.

Is there a way to do that?

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Queue playing again after call is answered

February 15, 2019, 1:29 pm
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@anty506 wrote:

We are having reports of strange things going on with FreePBX today. One being the queue message and hold is being played after called is picked up from queue.

Example, caller calls and selects 1. Caller is sent to queue. Agent answers and hears the queue greeting playing again even though the caller is connected to the agent. The agent tells the caller to press one so that it will stop talking. Caller presses 1 and is put back in the queue and both caller and agent can hear hold music and queue position.

The second report we have today is clients complained that they were in the Queue as one number, then after waiting, were given a HIGHER number in the Queue.

I’m assuming there is something going on with the queue app… Just wanted to check to see if anyone has experienced this before.

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UCP Error on Logging In

February 15, 2019, 4:26 pm
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@MollyMae wrote:

When logging into the UCP, I get “There was an error. See the console log for more details.”

I checked the console log and I see a Syntax error. I took a couple of screencaps and stitched them together below. I’ve done some poking around and couldn’t find another thread with the same console message. I’ve poked around, but I can’t figure out what I’m overlooking.

UCP
UCP.png1647x937 47.3 KB

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Call Recording Location Issues

February 15, 2019, 7:37 pm
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@kwriley87 wrote:

Hi there-

I am running FreePBX 14 Distro on Asterisk 13.

I have call recordings configured via Settings > Advanced Settings to save to a block storage drive: /mnt/blockstorage rather than the default /var/spool/asterisk/monitor directory.

It appears that most of the recordings are being saved to the block storage drive as they should be, but there are several outbound calls every day that are still being saved to the /var/spool/asterisk/monitor directory.

The files in /mnt/blockstorage directory are all named something like: out-9082824963-341-20190215-151512-1550265312.584325.wav

The files in the /var/spool/asterisk/monitor directory are all named something like: out-outbound-1541194698.142991.wav

Would anyone happen to know why this may be happening?

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Voicemail not letting callers record past 20 seconds

February 15, 2019, 8:17 pm
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@nickm19371 wrote:

Hello,
Since a few updates ago, my inbound callers have not been able to record voicemails longer than 20 seconds. I have checked all the settings that I know how to check, but they all seem to be correct. The strange thing is that local calls (extension to extension calls) are fine – the voicemail can be as long as it needs to. Also when we transfer an inbound caller to another extension the same thing happens, they only get about 20 seconds before the PBX disconnects the call. As previously stated, this only started happening as of maybe 1 or 2 core updates ago… so I’m not sure what’s going on.

Has anyone else had this issue? Or does anyone know what the problem could be? Any help is greatly appreciated.

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Announce current time as part of call flow

February 15, 2019, 8:48 pm
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@BluePukeko wrote:

Hi, is there a way of having the system announce / speak the current time as part of the normal incoming call flow.

ie:

Thank you for calling XYZ company the current time is

…next step

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No audio for external unreachable calls

February 16, 2019, 7:05 am
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@microtech wrote:

Hello,
I have a problem, if I call a reachable number everything is ok.
But if I call an unattainable or nonexistent number, I do not receive audio, silent for minutes. In the old version of freepbx this did not happen, which module or configuration blocks audio ringing ?

does anyone have any ideas?
Thanks

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Sangoma S305 "Fails to Authenticate" and "Registration Failed"

February 16, 2019, 12:24 pm
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@dsirota wrote:

I have repeatedly tried to provision my new S305 phone. However, I cannot get it to register to an extension. I’ve tried using the End Point Manager to configure it before plugging it in, manually entering all the details, but nothing is working.

I’ve looked at other forum posts and tried everything, and nothing is making the phone work. How on Earth do I get this phone to connect to my extension?

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Calendar using COUNT dashboard warning

February 17, 2019, 7:48 am
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@ngingras wrote:

Hello,

I’ve started seeing the following error on a PBX which I reckon is due to a linked calendar having events with a specified # of repeats:

09%20AM

This was due to high CPU use caused by infinitely recurring events (FREEPBX-17403). While this bug was being worked on, we advised customers to set specific # of repeats to prevent the high CPU issue.

Is there a way to query the DB to get a list of calendar events that have a set number of repeats so we can advise the customer to change them to infinite?

Also I am not seeing a significant performance impact. Can anyone comment on what specifically slows down?

Cheers,

Nate

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Inexpensive freepbx/asterisk compatible phones

February 17, 2019, 1:32 pm
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@Randee wrote:

Hello,

Thanks for this awesome group and great software.

Looking to set up a new phone system. I have setup Asterisk/FreePBX with basic SIP phones but I was looking at some affordable Cisco phones. I have read a few setup walk throughs on this forum and elsewhere online. Is there a table of compatible phones somewhere? Seems like some CISCO phones are simple and others are a nightmare. Watched a youtube video showing someone connecting a spa525g phone and it looked like a piece of cake, however i have read some forum posts where people have had to give up on their cisco phone. Many of those posts were from 4-5 years ago, so I don’t know if new versions are more capable with those phones and are causing less headaches?

Thanks for any input or any place to look for compatibility.

Lastly, has anyone ever had luck with video calling? Just curious, the organization I am working for now I think would benefit from it.

thanks,

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