I will look at the wiki for guidance. At my current knowledge level your last response “does not compute”.
I do really appreciate your assistance with this. But I do need to educate myself more to comprehend and take action on what you advise.
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Dial Plan Help
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Upgrade to FreePBX 15 worked flawlessly
This topic was automatically closed 14 days after the last reply. New replies are no longer allowed.
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HT813 With FreePBX - can't make outbound calls
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Vega-100G Fax Pro
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Help with custom extensions for FreePBX / Asterisk
Hello!
I’m trying to find a robust way to use the PITCH_SHIFT feature of Asterisk, in combination with FreePBX. I’ve have a setup of Asterisk and FreePBX fully running, and would like to be able to use the PITCH_SHIFT feature on demand.
Right now, I have the following line in /etc/asterisk/extensions_custom.conf:
exten => _[*#0-9]!,1,Set(PITCH_SHIFT(rx)=2)
This works like a charm, for any number, by using the _[*#0-9]! . However, I would like to find a way to trigger the PITCH_SHIFT on demand, not in a static way.
For example: prepending a phone number with *123*, and then dial a normal number, which will activate the PITCH_SHIFT.
Is something like this possible by maybe using a dial pattern which catches this and strips it off and goes ahead and dials a number? I could then change the extension line to something like *123*! .
Also, I have all recording options on Forced, for both the extension I’m using, but also the routes. However, when the above PITCH_SHIFT function is active, calls aren’t being recorded anymore. Is there a solution to this?
BTW: I’m using freepbx/Asterisk as a hobby project. So fiddling around with settings like these are fully possible for me.
Thanks in advance!
KR,
Fairlynuts
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Using ftp client is not possible
I’ve got SNG7 Distro installed and I had several issues of the Distro not picking up the IP address and getting myself booted out of network several times. Another issue was getting Freepbx activated until I bypassed DNS setting. I got it.
It seemed everything worked.
Now I had an issue of uploading and downloading files to my server, even though I can connect to it using ssh on port 22 but ftp client can connect but as I try to upload a file, it disconnects. I thought this is the case of Ipv6 being enabled. So I changed /etc/sysctl.conf (net.ipv6.conf.all.disable_ipv6 = 1 net.ipv6.conf.default.disable_ipv6 = 1) and restarted the network, but no luck.
Help me… gezzee. another challenge for tonight.
cf. my server is miles away from where I am, in an IDC center. So doing anything physically to the server is not an option.
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Wcaxx : Quad-FXO at base 0 failed initialization
Hello FreePBX Community members,
I have an issue on a FreePBX14 server where incoming and outgoing calls via DAHDI channels are not going through even my dialplan is OK - nothing changed from months because it was working 24/7 without problems until yesterday, it seems like Digium card (A8B) problems - some ports not working - maybe from any voltage/electric discharge from telco lines or something else.
Anyway does anybody here faced the same problem on past and which could be the solution maybe if you had one?
The problem message on this case is : wcaxx : Quad-FXO at base 0 failed initialization
When I look at /etc/dahdi/system.conf i see this config/message
Autogenerated by /usr/sbin/dahdi_genconf on Wed Jun 19 12:12:51 2019
If you edit this file and execute /usr/sbin/dahdi_genconf again,
your manual changes will be LOST.
Dahdi Configuration File
This file is parsed by the Dahdi Configurator, dahdi_cfg
Span 1: WCTDM/0 "Wildcard A8B" (MASTER)
fxsks=1
echocanceller=oslec,1
channel 2, WCTDM/0/1, no module.
channel 3, WCTDM/0/2, no module.
channel 4, WCTDM/0/3, no module.
fxsks=5
echocanceller=oslec,5
fxsks=6
echocanceller=oslec,6
fxsks=7
echocanceller=oslec,7
fxsks=8
echocanceller=oslec,8
Global data
loadzone = us
defaultzone = us
So do you think the card is broken-burned or it’s a workaround way to make it usable as card again?
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Intel Server System R1304SPOSHBNR intsallation
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Sangoma S70x phones and Walkie Talkie Interference
These phones are Wifi capable, I initially thought that you are using them on Wifi, and this is why it’s happening.
I am shocked that this is happening without actually using Wifi…
I think the only way to go, is to contact support, and have them look into it.
We have several S500/5s in a facility that uses walkie talkies as well, and no issues.
What type/brand of walkie talkies?
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Dial Plan Help
Your PBX is very old. Consider updating.
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Advanced route for incoming calls?
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Help with custom extensions for FreePBX / Asterisk
You can define a feature code prefix with dialplan of this format:
[from-internal-custom]
exten => _**22XX!,1,NoOp(Entering custom defined context from-internal-custom in extensions_custom.conf)
exten => _**22XX!,n,Set(PITCH_SHIFT(rx)=2)
exten => _**22XX!,n,Goto(from-internal,${EXTEN:4},1) ; strip prefix and proceed to from-internal↧
Sangoma S70x phones and Walkie Talkie Interference
Uhm, turn off the wifi on the phones?
If the radios are on then they can pick up things. RF is RF.
Apples to oranges. The S500/5’s don’t have wifi radios.
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Phones unreachable from outside
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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FreePBX to FreePBX SIP URI calling
I don’t know about your definition of “manual”, but Trunks are the way to go and setting them up manually is the right way.
Now, if you have nothing but dedicated IP addresses, you can set up a single PJ-SIP trunk that accepts traffic from all of your “client” PBXes and controls their access by their destination IP address. Your client machines will also need to set up trunks to your central system, so this is going to be a pain no matter how you do it.
Rest assured though, that setting up SIP URI for all of your systems will be an even more enormous pain.
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Using ftp client is not possible
If you can establish an SSH connection to the server from your local machine, use SFTP to transfer the files. Most SSH clients have a SFTP client either adjacent or built in.
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Sangoma S70x phones and Walkie Talkie Interference
The ARRL website has this to say about interference: The FCC rules require the equipment manufacturer or importer to design and test his products to ensure that they do not exceed the absolute maximum limits. In addition, the FCC requires that Part 15 devices be operated in such a way that they not cause harmful interference .
If the radios do not exceed the allowable interference levels, the receiving device “should absorb” as much as possible.
Also, LMRs (Land Mobile Radios) in a commercial environment MUST be licensed. You can’t use consumer/hobby class LMRs in any commercial environment.
I’d check the following:
- Make sure the LMRs being used are licensed for commercial use.
- Check to make sure that your equipment is properly grounded. This includes any radio base stations or repeaters, as well as network devices.
- If you can’t knock it down that way, the first party that needs to react is the radio manufacturer. If they are within limits, check with your other equipment providers to make sure they have done the minimum for harmful interference rejection.
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Codecs.conf. not working as should
Hello,
I’m struggling with setting a personal opus configuration codecs.conf, would very appreciate if someone could help…
cat /etc/schmooze/pbx-version
12.7.5-1807-1.sng7
Asterisk 15.5.0
Here is my codecs.conf file:
[opus16]
type=opus
fec=yes
cbr=yes
dtx=yes
packet_loss=30
max_playback_rate=16000
bitrate=16000
max_bandwidth=medium
When I start asterisk, it loads it correctly:
[2019-06-20 10:27:22] VERBOSE[4456] loader.c: Loading codec_opus.so.
[2019-06-20 10:27:22] VERBOSE[4456] format_cache.c: Updated cached format with name ‘opus’
[2019-06-20 10:27:22] VERBOSE[4456] format_cache.c: Created cached format with name ‘opus16’
[2019-06-20 10:27:22] VERBOSE[4456] translate.c: Registered translator ‘lintoopus’ from codec slin to opus, table cost, 600000, computational cost 999999
[2019-06-20 10:27:22] VERBOSE[4456] translate.c: Registered translator ‘opustolin’ from codec opus to slin, table cost, 900000, computational cost 999999
[2019-06-20 10:27:22] VERBOSE[4456] loader.c: codec_opus.so => (OPUS Coder/Decoder)
[2019-06-20 10:27:22] VERBOSE[4456] loader.c: Loading format_ogg_opus.so.
[2019-06-20 10:27:22] VERBOSE[4456] file.c: Registered file format ogg_opus, extension(s) opus
[2019-06-20 10:27:22] VERBOSE[4456] loader.c: format_ogg_opus.so => (OGG/Opus audio)
So I set the extension to
disallow=all
allow=opus16
But when I make the call I get this:
Rx-Frame Retry[ No] – OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 00003ms SCall: 00009 DCall: 00000 189.60.29.241:4569
VERSION : 2
CALLING NUMBER :
CALLING NAME :
CALLING PRESNTN : 1
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
FORMAT : 0
FORMAT2 : opus
CAPABILITY : 0
CAPABILITY2 : opus
USERNAME : 5899
CALLED NUMBER : 984158843
DNID : 984158843
ADSICPE : 0
CALLTOKEN : 51 bytes
FW BLOCK DATA : 18 bytesTx-Frame Retry[-01] – OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 00003ms SCall: 13887 DCall: 00009 189.60.29.241:4569
Tx-Frame Retry[000] – OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ
Timestamp: 00012ms SCall: 13887 DCall: 00009 189.60.29.241:4569
AUTHMETHODS : 2
CHALLENGE : \x35\x36\x33\x37\x39\x38\x39\x37\x38
USERNAME : 5899Rx-Frame Retry[ No] – OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP
Timestamp: 00041ms SCall: 00009 DCall: 13887 189.60.29.241:4569
MD5 RESULT : b383564445946a5cd0c3222514250f75Tx-Frame Retry[-01] – OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK
Timestamp: 00041ms SCall: 13887 DCall: 00009 189.60.29.241:4569
Tx-Frame Retry[000] – OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT
Timestamp: 00051ms SCall: 13887 DCall: 00009 189.60.29.241:4569
CAUSE : Unable to negotiate codec
CAUSE CODE : 58
If I choose allow=opus on the the extension it works correctly though.
All opus options are enabled on my zoiper softphone and opus is globally enabled to SIP and IAX.
What am I doing wrong please?
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Custom message while on hold
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
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FreePBX to FreePBX SIP URI calling
As @lgaetz already pointed out, each Asterisk system needs to allow the other side to send it calls and accept them. This means a peer (chan_sip) or an endpoint (pjsip) needs to be involved. You cannot get around this without opening your PBX to accept requests from anywhere.
It doesn’t matter how the Dial() is done the end result is always sip:user@domain:port <-- That is a SIP URI. All calls leave the PBX in a full SIP URI format. Just like all requests from your IP phone leave the phone in a full SIP URI format.
The only thing SIP URI Dialing does on the phone is lets you enter a full SIP URI vs the phone using what is in the host/proxy settings.
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