@AIC2000 wrote:

Hi,

I have a SIP provider with 20 channels that can be shared between multiple numbers.

Because of the number of businesses and phone numbers, I'd like to keep the FreePBX installs seperate, but pool all incoming and outgoing calls via my own SIP trunk package (with the supplier).

That way, I can pool all the channels, so for example, if I have 100 channels (to keep things simple), and company A only uses 40 but sometimes needs 60 and company B uses 50 but sometimes needs 60 - I don't have to buy 120 channels, I can just buy the 100 channels and get an alert if I ever reached anywhere near total capacity (which I never would, as these aren't call centers)

So what I was thinking was having the main central FreePBX, then having child FreePBX installs per seperate business (so they could have their own internal extensions and call transfers etc yet is trunked through my central FreePBX for outgoing calls.

Is this possible?

I was thinking of having something like this:

                     Main Routing FreePBX Server

---

      Company 1                      Company 2                 Company 3

Etc

I initially thought of setting up extensions on my Main Routing FreePBX server, so that Company 1 Company 2 and Company 3 could connect via Trunks to that server? But it doesn't seem to be working, and before I go too indepth with this I figured I'd check to see what general opinion is on doing this kind of thing?

Thanks

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@vegbrasil wrote:

Hi,

I'm setting a queue loop like this:

Queue 1 failovers to Queue 2
Queue 2 failovers to Queue 1

My idea was, a call comes in and bounces betweens queues until someone picks up.

This setup works well until the third time/bounce, where the caller listens "Forwarding Error" and the system hangs up.

Basically, the call goes to Queue 1 three times and Queue 2 three times, calling the agents as I defined but stops after the third try with said error.

Is there any option that limits the failover attempts? I want to make the call bounce without limits.

I'm running the latest FreePBX distro (17.07-1) with Asterisk 13.

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@mvogel4949 wrote:

Can I have both types of extensions on the system at the same time? Thanks

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@lbocken wrote:

Hi everyone,

I've a question. I'm sorry if this post is not in the right category.
I'll try to explain the problem so good as possible.
I've installed FreePBX in a Hyper-V-machine, and everything works. I can dial internally and externally. Voicemail works, but now the problem comes. When an extension recieves a voicemail-message, i call *97 on that phone. The phone (Siemens Unify Openstage 15) connects to the voicemail, but as soon as the woman says 'You have 1 new message', the connection is gone. But, when an extension has 2 or more messages in it's mailbox, the connection doesn't disconnect. This problem is very, very stange. I've been looking at the internet for hours now, and i think it is better to answer this question on this forum. My Asterisk-version is 13.12.1.

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@spar1grep wrote:

Hello,

I have been checking over the last couple of weeks for an update to FreePBX to address the Asterisk vulnerabilities AST-2017-005, 006 & 007. Last week the asterisk11-core.x86_64 0:11.25.2-1.shmz65.1.122 package was released and everything was OK (no errors in yum update reported). I planned to peform the work today as I have never updated a FreePBX server before, so I imaged the server and started the yum update process. Unfortunately now I get two errors as follows:

Error: Package: asterisk11-core-11.25.2-1.shmz65.1.122.x86_64 (pbx)
Requires: libsrtp.so.0()(64bit)
Removing: libsrtp-1.4.4-4.20101004cvs.el6.x86_64 (@anaconda-PBX-201403180405.x86_64/6.5)
libsrtp.so.0()(64bit)
Updated By: libsrtp-1.5.4-3.el6.x86_64 (epel)
Not found
Error: Package: asterisk11-core-11.25.2-1.shmz65.1.122.x86_64 (pbx)
Requires: libresample.so.1.0()(64bit)
Removing: libresample-0.1.3-11_centos6.x86_64 (@anaconda-PBX-201403180405.x86_64/6.5)
libresample.so.1.0()(64bit)
Updated By: libresample-0.1.3-12.el6.x86_64 (epel)
Not found

I think these errors relate to the server having multiple yum repositories defined, it currently has the FreePBX, Fedora and Dell repositiories defined, having inherited the server I am not too sure why these repositoires have been defined (the Dell one I know it was to install the Dell Linux services so the server can be monitored by Opsview), Fedora may have been defined as the Dell Linux services require something that was not in the FreePBX repository....

I tried to resolve the issue by attempting to update the two packages above first but this fails with a RPM error which I think is because the versions will break the current version of FreePBX (or the update process removes and then installs the newer version which is not being allowed because FreePBX depends on the package), the error is below:

Running rpm_check_debug
ERROR with rpm_check_debug vs depsolve:
libsrtp.so.0()(64bit) is needed by asterisk11-core-11.25.2-1.shmz65.1.122.x86_64
You could try running: rpm -Va --nofiles --nodigest
Your transaction was saved, rerun it with: yum load-transaction /tmp/yum_save_tx-2017-09-16-11-35p64Bj_.yumtx

Not being too good with Linux how do I update FreePBX without breaking the whole product?

Thanks.

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@Gordon0193 wrote:

Are the FreePBX documents a cut and paste from the Sangoma documents? The reason I ask is because I don't see this option in my FreePBX version. Based on the wiki page - "Rob Thomas on Jun 15, 2017". It's appearing that this section/page was touched in 2017 and is making me think it's factual and current. Can someone please help me understand the reason for the delta?

Thanks,

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@jpennell wrote:

Please please please someone take pity on me and help...!

I have a PBX which is behind a router which uses an LTE connection as it's failover.

When the router is using the LTE connection, the trunks fail.

I know that I can get this to work if I can get outbound proxy working correctly but I'm obviously missing something. No matter how I configure chan_sip, the data at my provider's end indicates the IP address of the PBX (which is invalid because, in effect, the LTE network is adding another NAT layer).

On a softphone, if I configure "outbound proxy" to the provider's proxy, the provider's end indicates the IP address of the outbound proxy as the source of the connection - and the softphone functions with two-way audio. If I disable "outbound proxy" on the softphone, the provider's data shows the same information as it does for the PBX - and calls fail.

It seems that FreePBX/Asterisk is ignoring the outboundproxy field on the chan_sip definition. Am I misunderstanding the function of this field? Can I somehow force chan_sip (or PJSip) to use the outbound proxy?

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@raulsalamanca wrote:

Hi,

I created a trunk and the outbound route. If I make a call it doesnt work. That is because Asterisk is sending "56945013154@telvoip" as number. The correct number is "56945013154" (without "@telvoip").

   -- Executing [s@macro-dialout-trunk:30] Dial("PJSIP/1001-0000006f",
"SIP/telvoip/56945013154@telvoip,300,T") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/telvoip/56945013154@telvoip

This:
-- Called SIP/telvoip/56945013154@telvoip
Must be:
-- Called SIP/telvoip/56945013154

If I make a call direct from Asterisk CLI, it works properly.
channel originate SIP/telvoip/56945013154 extension 1001

I guess this is a configuration problem. I've been trying to fix this, but I could not. How can I fix it?

Thanks!

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@mike366 wrote:

In old version 11 of Freepbx, I could easily add both a phone number, and a different fax number, via 2 separate extensions, and have a single user with both a phone line, and a separate fax line. In V13, I am not sure how to do this, since everything seems user oriented, not really extension oriented.

Can someone please point me to any doc on how to set this up in V13, so a single user can support a separate DID for phone, and a different DID for Fax?

Thanks.

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@hack3rx wrote:

I been trying to register my SIP extensions to a Bria or Zoiper softphone. I am able to register the SIP Trunk account(Auth name/Auth PW) to the softphones but am not able to register any of the extensions I have made.

The Extensions are SIP extensions. On the edit SIP extensions page, it says
"This device uses CHAN_SIP technology listening on Port 5160 (UDP - this is a NON STANDARD port)"

I tried to update the port SIP settings on Zoiper to match(5160) this but it still wont register

I also tried to create chan_pjsip extension on UDP 5060 but it still does not want to register.

Extension Credentials

. Login - 30
. PW - Copy Secret from Ext edit screen

Trunk SIP (registers just fine)

. Auth Username
. Auth PW

Can anyone please shed some light on this issue and help guide me to the right direction. Thank you in advance.

This is what the Asterisk Logs say:
[2017-09-17 22:06:21] NOTICE[30597] res_pjsip/pjsip_distributor.c: Request 'INVITE' from '' failed for '192.187.101.82:53293' (callid: 1920343398-1609962747-1454408668) - No matching endpoint found
[2017-09-17 22:06:21] SECURITY[2893] res_security_log.c: SecurityEvent="InvalidAccountID",EventTV="2017-09-17T22:06:21.694-0700",Severity="Error",Service="PJSIP",EventVersion="1",AccountID="75",SessionID="1920343398-1609962747-1454408668",LocalAddress="IPV4/UDP/xx.xxx.xxx.69/5060",RemoteAddress="IPV4/UDP/xx.xxx.xxx.82/53293"
[2017-09-17 22:06:21] NOTICE[30597] res_pjsip/pjsip_distributor.c: Request 'INVITE' from '' failed for 'xx.xxx.xxx.82/53293' (callid: 1920343398-1609962747-1454408668) - Failed to authenticate

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@stegfeld wrote:

hi,

I've installed this plugin
https://wiki.freepbx.org/download/attachments/71270631/ZuluAddInSetup.msi.exe?version=1&modificationDate=1488204159000&api=v2

and changed the name, deleted .exe at the end of filename

I can see the ZULU plugin within outlook 2016 add-ins, but no ZULU add-in is shown on the ribbon

anybody has this plugin running on office 2016?

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@connextechs wrote:

For some reason my intrusion detection blocks my home ip. None of the phones at my house have incorrect credentials in them for fail2ban to block them. It would just block my house because of some SIP registration attempt coming out of my house ip, but I don't know what would it be. I have one Sangoma, one Polycom and one Zoiper phone always connected to the pbx that is on the cloud server. I don't want to make my house IP trusted because I want to figure out why it would block my ip. There is no reason for it to block it since it's ONLY supposed to block when there is an incorrect login attempt. I don't use RF for now.

Do you guys have any thoughts on this one? I would really appreciate it.

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@digiteltlc wrote:

After years of service , an Iax peer has become suddenly UNREACHABLE one way :

Three Freepbx systems are interconnected over a vpn
Site A to site B , as well as site B to site C have their Iax trunks up and running both ways
Site A shows site C iax peer as unreachable, despite site C shows site A iax peer up and running.

All pbx are updated, pingable, there are no firewalls issues.

nmap -sU -p4569 , each other gives :

Host is up (0.071s latency).
PORT STATE SERVICE
4569/udp open|filtered unknown

All machines (routers, firewalls and pbx) have been restarted.

What can I check or try ?
Thanks

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@sentinelace wrote:

I need to setup a blind transfer to voicemail button in endpoint manger. If i hit transfer, then hit * extension it works. I have tried setting up a prefix with no luck. What is the proper way of doing this?

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@Sergey_MGD wrote:

Good afternoon.
I got the configuration of the servers FreePBX and when one of these servers was inserted into the circuit, Yakh2 trunks fell into the loop. Please help, the scheme of the connections is below.

Server #1 looking in internet white IP and listen IAX port 10050
Server #2 is inside the protected perimeter and also listens to the IAX port 10050.
On the network screen (cisco asa), the static NAT is configured to transfer this port to the server 2 and the corresponding acl is created.
Server #2, which is inside is registered on server 1 (the one with a white ip), and server 2 can not.

Server #1 - pure asterisk, settings below:
[general]
bindport=10050
requirecalltoken=no
register => login1:pass1@White IP address cisco ASA:10050
context=internal
autokill=yes

[guest]
type=user
context=denied
callerid="Guest IAX User"

[callcenter]
type=friend
host=dynamic
username=login2
secret=pass2
context=internal
trunk=yes
connectedline=yes
encryption=no
qualify=yes
auth=plaintext
permit=0.0.0.0/0.0.0.0

Server #2 - FreePBX (inside), WEB settings below:
Outgoing:
username=login1
secret=pass1
type=friend
trunk=yes
qualify=yes
host=dynamic
connectedline=yes

Incoming:
login2:pass2@95.213.248.176:10050

Server #1
iax2 show peers
Name/Username Host Mask Port Status Description
login2/trunk white IP (D) 255.255.255.255 10050 (T) OK (11 ms)
iax2 show registry
Host dnsmgr Username Perceived Refresh State
White IP ASA:10050 N login1 White IP ASA:10050 60 Rejected

Server #
iax2 show peers
Name/Username Host Mask Port Status Description
TRUNK/off (null) (D) (null) (null)(T) UNKNOWN
iax2 show registry
Host dnsmgr Username Perceived Refresh State
white IP aster:10050 N login2 white IP aster:10050 60 Registered
1 IAX2 registrations.

Plz help...

Sorry for my broken English.

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@sentinelace wrote:

I need to setup a night mode so all incoming calls get routed to an outside number. *271 for example could be a button she manually pushes. I tried time conditions. It says activated but never actually does anything, I still get the IVR

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@alteredstate wrote:

Hello everyone,

I'm using Asterisk 13.12.1 and FreePBX 13.0.192.18 but having a difficult time getting SIP TLS & SRTP to work on a Cisco SPA525G2 phone. I've followed this guide: TLS & SRTP which is pretty much straight forward but from what I gathered the Cisco SPA525G2 requires a "Mini Certificate" and "SRTP Private Key" as shown by the Cisco SPA525G2 web admin screenshot:

Mini Certificate and SRTP Private Key Fields.png817x37 2.7 KB

I'm not quite sure what the "Mini Certificate" is and if you follow this link and scroll down to: Subscriber Information it just states what I need to do with the "Mini Certificate" but not how to obtain one or what it is. Also, if I copy the default.key value from /etc/asterisk/keys on my FreePBX system I'm not able to paste the entire value into the SRTP Private Key field as a good portion of it is cut off after pasting. So that makes me believe I'm going about this all wrong. At this point, I'm not sure what to do so if anyone has an idea or actual success setting this up then I would really appreciate the assistance.

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@Mitcon wrote:

Ok, this is my first time writing on this forum so if I am doing something wrong please let me know.

Anywho, here is the problem. We are running FreePBX 14 for one of our customers along with Grandstream GXP 2170 phones. Since we couldn't implement Call Forwarding from the local calling features of the phone because of a conflict, we had to do it through the server. Now the phone that we are setting up call forwarding on is inside a Ring Group and no matter what I do, it will not forward calls to another local extension. I have tried doing this from the phone GUI and also by dialing the *72 feature code, but to no avail. Is this by design? Or is it some kind of flaw in the system? Has anyone had problems doing call forwarding from a ring group to another local extension?

[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: Starting New Dialparties.agi
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: Caller ID name is 'Dr. Hale' number is '107'
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: CW Ignore is:
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: CF Ignore is:
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: CW IN_USE/BUSY is: 1
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: Methodology of ring is 'memoryhunt'
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: Added extension 101 to extension map
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: Added extension 102 to extension map
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: Extension 101 has call forward set to 103
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: Extension 102 cf is disabled
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: Extension 102 do not disturb is disabled
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: Discovered PJSIP Endpoint PJSIP/102
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: Ended up with real PJSIP Dial string PJSIP/102/sip:102@192.168.69.69:5060
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: Filtered ARG3: 103-102
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: RVOL_MODE ''
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: RVOL is:
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: dialparties.agi: RVOLPARENT is:
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] res_agi.c: AGI Script dialparties.agi completed, returning 0
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:27] NoOp("PJSIP/107-00000028", "Returned from dialparties with 1 hunt members to dial") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:28] Set("PJSIP/107-00000028", "HuntLoop=0") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:29] ExecIf("PJSIP/107-00000028", "0?Set(HuntMembers=0)") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:30] GotoIf("PJSIP/107-00000028", "1?a30") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx_builtins.c: Goto (macro-dial,s,33)
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:33] Set("PJSIP/107-00000028", "HuntMember=HuntMember0") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:34] GotoIf("PJSIP/107-00000028", "0?a32:a35") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx_builtins.c: Goto (macro-dial,s,40)
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:40] GotoIf("PJSIP/107-00000028", "1?a36:a50") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx_builtins.c: Goto (macro-dial,s,41)
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:41] Set("PJSIP/107-00000028", "CTLoop=0") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:42] GotoIf("PJSIP/107-00000028", "0?huntstart") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:43] Set("PJSIP/107-00000028", "CT_EXTEN=103") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:44] Set("PJSIP/107-00000028", "EXTTOCALL=103") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:45] Set("PJSIP/107-00000028", "DB(CALLTRACE/103)=107") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:46] Set("PJSIP/107-00000028", "CTLoop=1") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:47] Goto("PJSIP/107-00000028", "s,a37") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx_builtins.c: Goto (macro-dial,s,42)
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:42] GotoIf("PJSIP/107-00000028", "1?huntstart") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx_builtins.c: Goto (macro-dial,s,48)
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:48] NoOp("PJSIP/107-00000028", "Hunt Dial Start") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:49] ExecIf("PJSIP/107-00000028", "0?Set(HASH(__SIPHEADERS,Alert-Info)=Normal;volume=)") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:50] ExecIf("PJSIP/107-00000028", "0?Set(HASH(__SIPHEADERS,Alert-Info)=Normal;volume=)") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:51] Macro("PJSIP/107-00000028", "dial-hunt-predial-hook,") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial-hunt-predial-hook:1] MacroExit("PJSIP/107-00000028", "") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-dial:52] Dial("PJSIP/107-00000028", "PJSIP/102/sip:102@192.168.69.69:5060,20,TtrM(auto-blkvm)b(func-apply-sipheaders^s^1),") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] app_stack.c: PJSIP/102-00000029 Internal Gosub(func-apply-sipheaders,s,1) start
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf("PJSIP/102-00000029", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@func-apply-sipheaders:2] NoOp("PJSIP/102-00000029", "Applying SIP Headers to channel") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@func-apply-sipheaders:3] Set("PJSIP/102-00000029", "SIPHEADERKEYS=") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@func-apply-sipheaders:4] While("PJSIP/102-00000029", "0") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] app_while.c: Jumping to priority 8
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] pbx.c: Executing [s@func-apply-sipheaders:9] Return("PJSIP/102-00000029", "") in new stack
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] app_stack.c: Spawn extension (from-internal, 600, 1) exited non-zero on 'PJSIP/102-00000029'
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] app_stack.c: PJSIP/102-00000029 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] app_dial.c: Called PJSIP/102/sip:102@192.168.69.69:5060
[2017-09-14 19:49:54] VERBOSE[26496][C-00000015] app_dial.c: PJSIP/102-00000029 is ringing
[2017-09-14 19:50:01] VERBOSE[26496][C-00000015] app_macro.c: Spawn extension (macro-dial, s, 52) exited non-zero on 'PJSIP/107-00000028' in macro 'dial'
[2017-09-14 19:50:01] VERBOSE[26496][C-00000015] pbx.c: Spawn extension (from-internal, 600, 17) exited non-zero on 'PJSIP/107-00000028'
[2017-09-14 19:50:01] VERBOSE[26496][C-00000015] pbx.c: Executing [h@from-internal:1] Macro("PJSIP/107-00000028", "hangupcall") in new stack
[2017-09-14 19:50:01] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("PJSIP/107-00000028", "1?theend") in new stack
[2017-09-14 19:50:01] VERBOSE[26496][C-00000015] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2017-09-14 19:50:01] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("PJSIP/107-00000028", "0?Set(CDR(recordingfile)=)") in new stack
[2017-09-14 19:50:01] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-hangupcall:4] NoOp("PJSIP/107-00000028", "PJSIP/102-00000029 monior file= ") in new stack
[2017-09-14 19:50:01] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-hangupcall:5] AGI("PJSIP/107-00000028", "attendedtransfer-rec-restart.php,PJSIP/102-00000029,") in new stack
[2017-09-14 19:50:01] VERBOSE[26496][C-00000015] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2017-09-14 19:50:01] VERBOSE[26496][C-00000015] res_agi.c: AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2017-09-14 19:50:01] VERBOSE[26496][C-00000015] pbx.c: Executing [s@macro-hangupcall:6] Hangup("PJSIP/107-00000028", "") in new stack
[2017-09-14 19:50:01] VERBOSE[26496][C-00000015] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'PJSIP/107-00000028' in macro 'hangupcall'

This is the console output but frankly I am having some trouble interpreting all of it.....

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@globalgen wrote:

Dear members,
After succesfully setting a BRI DAHDI inbound route (PBXact UC300 & Sangoma Card) and assigning the DID to an extension, I receive the following on incoming call. For privacy reasons XXXXXXXX represents the calling phone number and YYYYYYYY represents the DID number assigned to the BRI service by the carrier.

The issue appears to be assigning an incoming DAHDI to a SIP extension, however can not find sufficiently detailed documentation to assist.

A pointer from anyone is greatly appreciated.

Thanks.

[2017-09-18 19:10:05] VERBOSE[4966][C-00000016] sig_pri.c: Accepting call from 'XXXXXXXX' to 'YYYYYYYY' on channel 0/1, span 1
[2017-09-18 19:10:05] VERBOSE[15731][C-00000016] pbx.c: Executing [YYYYYYYY@from-digital:1] Set("DAHDI/i1/XXXXXXXX-14", "__FROM_DID=YYYYYYYY") in new stack
[2017-09-18 19:10:05] VERBOSE[15731][C-00000016] pbx.c: Executing [YYYYYYYY@from-digital:2] NoOp("DAHDI/i1/XXXXXXXX-14", "Received an unknown call with DID set to YYYYYYYY") in new stack
[2017-09-18 19:10:05] VERBOSE[15731][C-00000016] pbx.c: Executing [YYYYYYYY@from-digital:3] Goto("DAHDI/i1/XXXXXXXX-14", "s,a2") in new stack
[2017-09-18 19:10:05] VERBOSE[15731][C-00000016] pbx_builtins.c: Goto (from-digital,s,2)
[2017-09-18 19:10:05] VERBOSE[15731][C-00000016] pbx.c: Executing [s@from-digital:2] Answer("DAHDI/i1/XXXXXXXX-14", "") in new stack
[2017-09-18 19:10:05] WARNING[15731][C-00000016] chan_sip.c: This function can only be used on SIP channels.
[2017-09-18 19:10:05] VERBOSE[15731][C-00000016] pbx.c: Executing [s@from-digital:3] Log("DAHDI/i1/XXXXXXXX-14", "WARNING,Friendly Scanner from ") in new stack
[2017-09-18 19:10:05] WARNING[15731][C-00000016] Ext. s: Friendly Scanner from
[2017-09-18 19:10:05] VERBOSE[15731][C-00000016] pbx.c: Executing [s@from-digital:4] Wait("DAHDI/i1/XXXXXXXX-14", "2") in new stack
[2017-09-18 19:10:07] VERBOSE[15731][C-00000016] pbx.c: Executing [s@from-digital:5] Playback("DAHDI/i1/XXXXXXXX-14", "ss-noservice") in new stack
[2017-09-18 19:10:07] VERBOSE[15731][C-00000016] file.c: <DAHDI/i1/XXXXXXXX-14> Playing 'ss-noservice.alaw' (language 'en')

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@mvogel4949 wrote:

The default value is hourly but if I attempt to change it to 15min or 30min the changes are not kept and it stays at hourly. Is this not really a variable I can change?

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