@mjb2000 wrote:

My ISP here in Mexico, Telmex, provides a Fiber Optic Optical Network Terminal (ONT) - Essentially a piece of Consumer Premises Equipment (CPE) that functions as a modem, router, switch, WiFi access point and crucially - as an ATA, connecting to the ISP via SIP and offering a phone line from the telephone company.

I was able to obtain the SIP credentials by connecting a serial console to my modem. I have been able to use these credentials successfully with PhonerLite, Zoiper and MicroSIP (for all of these software packages I can register and make phone calls.

TELMEX seems to use SRV records which I couldn’t get working with MicroSIP, so I substituted the proxy domain name with the IP address that PhonerLite was using and that meant I was able to get MicroSIP to register correctly.

Asterisk info shows “Request sent” for registration, but is never registers.

@Stewart1 was trying to help me on this thread, but suggesting creating a new thread specifically for my problem. One this he mentioned was “copying a register to a file and using sipsak”… This is above my level of understanding and I don’t know how I would go about this. Does anyone else know what might be going on here? How can I take my working MicroSIP config and use it successfully in FreePBX?

My current config is:

Outbound
Trunk Name: Telmex
PEER Details: [BLANK]
Inbound
USER Context: +52xxxxxxxxxx
USER Details:

secret=PASSWORD
username=+520000000000
user=+520000000000
fromuser=+520000000000
realm=ims.telmex.com
domain=ims.telmex.com
authdomain=ims.telmex.com
fromdomain=ims.telmex.com
outboundproxy=189.247.242.147
host=189.247.242.147
fullcontact=+520000000000@ims.telmex.com
authname=+520000000000@ims.telmex.com
type=peer
insecure=very
context=from-sip-external

Register string:
+520000000000@ims.telmex.com:PASSWORD:+520000000000@189.247.242.147

Asterisk SIP log…

*.*.*.* = my external IP address, not the IP of the FreePBX box (which is 10.0.0.32)

[2018-07-09 12:35:41] WARNING[1019] chan_sip.c: Section 'Telmex' lacks type
[2018-07-09 12:35:41] VERBOSE[1019] chan_sip.c: Really destroying SIP dialog '75db2cf62a7ed81e14c45f5e28956512@10.200.0.32' Method: REGISTER
[2018-07-09 12:35:41] NOTICE[1019] chan_sip.c: -- Re-registration for +520000000000@189.247.242.147
[2018-07-09 12:35:41] VERBOSE[1019] chan_sip.c: REGISTER 11 headers, 0 lines
[2018-07-09 12:35:41] VERBOSE[1019] chan_sip.c: Reliably Transmitting (NAT) to 189.247.242.147:5060:
REGISTER sip:ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5160;branch=z9hG4bK4af132fc;rport
Max-Forwards: 70
From: <sip:+520000000000@ims.telmex.com>;tag=as3d7f6af0
To: <sip:+520000000000@ims.telmex.com>
Call-ID: 1cbc86d41b0e907b31bf29842558a31e@10.200.0.32
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-14.0.3.6(14.7.6)
Expires: 120
Contact: <sip:s@*.*.*.*:5160>
Content-Length: 0


---
[2018-07-09 12:35:42] VERBOSE[1019] chan_sip.c: Retransmitting #1 (NAT) to 189.247.242.147:5060:
REGISTER sip:ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5160;branch=z9hG4bK4af132fc;rport
Max-Forwards: 70
From: <sip:+520000000000@ims.telmex.com>;tag=as3d7f6af0
To: <sip:+520000000000@ims.telmex.com>
Call-ID: 1cbc86d41b0e907b31bf29842558a31e@10.200.0.32
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-14.0.3.6(14.7.6)
Expires: 120
Contact: <sip:s@*.*.*.*:5160>
Content-Length: 0


---
[2018-07-09 12:35:43] VERBOSE[1019] chan_sip.c: Retransmitting #2 (NAT) to 189.247.242.147:5060:
REGISTER sip:ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5160;branch=z9hG4bK4af132fc;rport
Max-Forwards: 70
From: <sip:+520000000000@ims.telmex.com>;tag=as3d7f6af0
To: <sip:+520000000000@ims.telmex.com>
Call-ID: 1cbc86d41b0e907b31bf29842558a31e@10.200.0.32
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-14.0.3.6(14.7.6)
Expires: 120
Contact: <sip:s@*.*.*.*:5160>
Content-Length: 0

PhonerLite config (working)

Proxy/Registrar: voipnvcompigl.telmex.net
Domain/Realm: ims.telmex.com
Username: +52xxxxxxxxxx
Password: ############
Authentication name: +52xxxxxxxxxx@ims.telmex.com

MicroSIP config (working)

SIP Server: ims.telmex.com
SIP Proxy: 189.247.242.147
Username: +52xxxxxxxxxx
Domain: ims.telmex.com
Login: +52xxxxxxxxxx@ims.telmex.com
Password: ############

PhonerLite log

From here you can see why I ended up using the proxy IP address

I am not sure what the various forbidden and timeout messages are, but PhonerLite does seem to work correctly (I can make and receive calls).

-------------------------------------------
09:52:17,558: R: DNS lookup for 'voipnvcompigl.telmex.net'
start resolving SRV (UDP)...
-------------------------------------------
09:52:17,561: R: DNS lookup for 'slbcompigl.voip.telmex.net'
189.247.242.147:5060 (TTL=1699)
-------------------------------------------
09:52:17,561: R: open UDP port (SIP): 5060

-------------------------------------------
09:52:17,562: R: open TCP port (TLS listen): 5061

-------------------------------------------
09:52:17,562: R: open TCP port (TCP listen): 5060

09:52:17,611: Listen Confirm: 0E 00 08 00 05 81 9E 02 01 00 00 00 00 00 
09:52:17,611: Listen Confirm
-------------------------------------------
09:52:17,563: R: open UDP port (mDNS): 5353

09:52:17,621: Facility Confirm: 1A 00 08 00 80 81 A0 02 01 00 00 00 00 00 03 00 09 00 00 06 00 00 3D 01 00 00 
09:52:17,621: Facility Confirm (Supplementary Services)
09:52:17,621: Facility Request: 16 00 08 00 80 80 A1 02 01 00 00 00 03 00 07 01 00 04 3D 01 00 00 
09:52:17,621: Facility Request (Listen To Supplementary Services)
09:52:17,621:  Get Supported Services: success
-------------------------------------------
09:52:17,563: T: 189.247.242.147:5060 (UDP)
REGISTER sip:ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.50:5060;branch=z9hG4bK80a6192d2c81e81187bbbb0be53ecaeb;rport
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=1934202406
To: <sip:+52xxxxxxxxxx@ims.telmex.com>
Call-ID: 80A6192D-2C81-E811-87B8-BB0BE53ECAEB@10.0.0.50
CSeq: 326 REGISTER
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>;+sip.instance="<urn:uuid:00D1773F-C27F-E811-97E2-EED72484AD05>"
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 900
Content-Length: 0


-------------------------------------------
09:52:17,563: T: mDNS refresh: sip:+52xxxxxxxxxx@ims.telmex.com = 169.254.106.102:5060, ttl=900
SIPPER for PhonerLite
09:52:17,621: Facility Confirm: 16 00 08 00 80 81 A1 02 01 00 00 00 00 00 03 00 05 01 00 02 00 00 
09:52:17,621: Facility Confirm (Supplementary Services)
09:52:17,621:  Listen: success
-------------------------------------------
09:52:17,621: T: 189.247.242.147:5060 (UDP)
SUBSCRIBE sip:+52xxxxxxxxxx@ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.50:5060;branch=z9hG4bK80a6192d2c81e81187bcbb0be53ecaeb;rport
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=3576163364
To: <sip:+52xxxxxxxxxx@ims.telmex.com>
Call-ID: 80A6192D-2C81-E811-87B9-BB0BE53ECAEB@10.0.0.50
CSeq: 327 SUBSCRIBE
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>;+sip.instance="<urn:uuid:00D1773F-C27F-E811-97E2-EED72484AD05>"
Max-Forwards: 70
User-Agent: SIPPER for PhonerLite
Expires: 1800
Event: message-summary
Accept: application/simple-message-summary
Content-Length: 0


-------------------------------------------
09:52:17,705: R: 189.247.242.147:5060 (UDP)
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.0.50:5060;received=*.*.*.*;branch=z9hG4bK80a6192d2c81e81187bcbb0be53ecaeb;rport=60580
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=3576163364
To: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=aprqngfrt-cpkluc30080e4
Call-ID: 80A6192D-2C81-E811-87B9-BB0BE53ECAEB@10.0.0.50
CSeq: 327 SUBSCRIBE


-------------------------------------------
09:52:17,847: R: 189.247.242.147:5060 (UDP)
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.50:5060;received=*.*.*.*;branch=z9hG4bK80a6192d2c81e81187bbbb0be53ecaeb;rport=60580
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=1934202406
To: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=dqdq0bbq
Call-ID: 80A6192D-2C81-E811-87B8-BB0BE53ECAEB@10.0.0.50
CSeq: 326 REGISTER
WWW-Authenticate: Digest realm="ims.telmex.com", nonce="PBk3fi0dSOVla8Cwre8YRw==",algorithm=MD5
Content-Length: 0


-------------------------------------------
09:52:17,848: T: 189.247.242.147:5060 (UDP)
REGISTER sip:ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.50:5060;branch=z9hG4bK80a6192d2c81e81187bdbb0be53ecaeb;rport
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=1934202406
To: <sip:+52xxxxxxxxxx@ims.telmex.com>
Call-ID: 80A6192D-2C81-E811-87B8-BB0BE53ECAEB@10.0.0.50
CSeq: 328 REGISTER
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>;+sip.instance="<urn:uuid:00D1773F-C27F-E811-97E2-EED72484AD05>"
Authorization: Digest username="+52xxxxxxxxxx@ims.telmex.com", realm="ims.telmex.com", nonce="PBk3fi0dSOVla8Cwre8YRw==", uri="sip:ims.telmex.com", response="bd636de3caf80c6c64ba2910c30555f9", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 900
Content-Length: 0


-------------------------------------------
09:52:18,100: R: 189.247.242.147:5060 (UDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.50:5060;received=*.*.*.*;branch=z9hG4bK80a6192d2c81e81187bdbb0be53ecaeb;rport=60580
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=1934202406
To: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=k2dqaboc
Call-ID: 80A6192D-2C81-E811-87B8-BB0BE53ECAEB@10.0.0.50
CSeq: 328 REGISTER
P-Associated-URI: <sip:+52xxxxxxxxxx@ims.telmex.com;user=phone>
P-Associated-URI: <sip:+52xxxxxxxxxx@ims.telmex.com>
Accept-Resource-Priority: wps.4
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>;expires=30;q=1;+sip.instance="<urn:uuid:00D1773F-C27F-E811-97E2-EED72484AD05>"
Content-Length: 0


-------------------------------------------
09:52:18,123: T: 189.247.242.147:5060 (UDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.247.242.147:5060;branch=z9hG4bKrc0icv204g1ge1u3epu0.1
From: <sip:+52xxxxxxxxxx@ims.telmex.com:5060>;tag=2gdrr79g-CC-20
To: <sip:+52xxxxxxxxxx@ims.telmex.com:60580>;tag=003db22d2c81e81187bdbb0be53ecaeb
Call-ID: 8g323bee3s8749i8a4r9r9iaa2a34s93@19500.0.ATS.ats01.ims.telmex.com.20
CSeq: 1 NOTIFY
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0


-------------------------------------------
09:52:18,293: R: 189.247.242.147:5060 (UDP)
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 10.0.0.50:5060;received=*.*.*.*;branch=z9hG4bK.bbkAMUUuC;rport=60580
From: <sip:52xxxxxxxxxx@voipnvcompigl.telmex.net>;tag=mTMk6w0eA
To: <sip:52xxxxxxxxxx@voipnvcompigl.telmex.net>;tag=uhf99zaa
CSeq: 210 REGISTER
Call-ID: grNrFpLzKe
Warning: 399 P.5.127.ims.telmex.com "SS170001F133L3261S0E0[00001] Hllm query failed"
Content-Length: 0

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@fearx wrote:

I spent the weekend studying and trying to fix this error, but neither Google nor Wiki helped me much. Has anyone experienced this and knows how to fix it?

1. All phones connect
2. they call, it appears that they are “calling”
3. when it answers (and gives the browser permission to get media), the connection ends

[2018-07-10 09:01:38] ERROR[16570][C-0000003b]: res_rtp_asterisk.c:2557 __rtp_recvfrom: DTLS failure occurred on RTP instance ‘0x7f7b600542d0’ due to reason ‘certificate verify failed’, terminating
[2018-07-10 09:01:38] WARNING[16570][C-0000003b]: res_rtp_asterisk.c:5310 ast_rtp_read: RTP Read error: Unspecified. Hanging up.

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@futurcom wrote:

hi,
each mornings all phones ring one time with message in the display : “inactive network”.
the problem is not always at the same time of the morning.

thank’s for your help

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@comtech wrote:

SNG7 FreePBX14/Asterisk 13

Hello,

I am sending faxes, which appears to be working well; however, I would like to limit the number of retries to one.

I see the max retries matrix in the wiki:
https://wiki.freepbx.org/display/ST/Sending+a+Fax#SendingaFax-MaximumRetryAttempts

My question is how can I adjust this so that other only tries once?

Thanks!

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@Matthew99 wrote:

Hi

i’m trying to enable call recording for 1 extension, steps taken so far

under extension set call recording to yes

image.png884x230 20.1 KB

so in theory these should show in /var/spool/asterisk/monitor ?

just made a test call and see no wav’s listed.

what am i missing

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@LuizClaudio wrote:

I have a FreePBX version 14, the calls from outside to inside fall in 32 seconds, regardless if there is voice or not, I researched it, checked all possibilities plus nothing at all, my FreePBX NAT identifies the external IP normally, the RTP ports 10000-20000, SIP 5060 are open, I put in sip.conf the option canreinvite = no, in advanced settings the NAT = no or yes, I changed the touch times, and nothing changed, in SIP settings the timeout RTP = 30s Hold Timeout = 300 Keep Alive = 0, I made the change of seconds of RTP timeout nothing else has changed.
I use CHAN_PJSIP, the firewall is configured to allow the ip and ports that are used by Freepbx, I tested with the firewall disabled, without any other firewall in the network and also continued in the same way.
I do not use STUN and TURN, they are by default. I debugged the asterisk CLI but I could not see anything that could break my call in those 32s. The PINGS test for specific IPs is not lost at all.
The calls from the inside out do not fall.
FreePBX has two network interfaces one with the internal LAN and the other for authentication with VoIP provider Algar Telecom.
Freepbx is authenticated with the Algar Telecom telephony provider, where I do the configuration on a network interface, the IPs they provide to get to the IP of their SIP server, is working normally.
Is there anything else you can do?
Can it be a problem with the VoIP provider?

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@rmatteson wrote:

Here’s what happens when someone calls our reception number normally:

Event Type	CID Name	CID Num	Exten	Context	Channel
CHAN_START	Name Omitted	401640xxxx	Incoming #	from-trunk	SIP/VOIPMS-00000a45
ANSWER	Name Omitted	401640xxxx	s	ivr-1	SIP/VOIPMS-00000a45
CHAN_START	Front Desk	10	s	from-internal	PJSIP/10-00000b6b
ANSWER	Front Desk	10	10	from-internal	PJSIP/10-00000b6b
BRIDGE_ENTER	Front Desk	10		from-internal	PJSIP/10-00000b6b
BRIDGE_ENTER	Name Omitted	401640xxxx	s	macro-dial-one	SIP/VOIPMS-00000a45
BRIDGE_EXIT	Name Omitted	401640xxxx	s	macro-dial-one	SIP/VOIPMS-00000a45
BRIDGE_EXIT	Front Desk	10		from-internal	PJSIP/10-00000b6b
HANGUP	Front Desk	10		from-internal	PJSIP/10-00000b6b
CHAN_END	Front Desk	10		from-internal	PJSIP/10-00000b6b
HANGUP	Name Omitted	401640xxxx	h	ext-local	SIP/VOIPMS-00000a45
CHAN_END	Name Omitted	401640xxxx	h	ext-local	SIP/VOIPMS-00000a45
LINKEDID_END	Name Omitted	401640xxxx	h	ext-local	SIP/VOIPMS-00000a45

Over the weekend this started occuring sometimes:

Event Type	CID Name	CID Num	Exten	Context	Channel
CHAN_START	401640xxxx	401640xxxx	Incoming #	from-trunk	SIP/VOIPMS-00000a46
ANSWER	Name Omitted	401640xxxx	s	ivr-1	SIP/VOIPMS-00000a46
CHAN_START	Front Desk	10	s	from-internal	PJSIP/10-00000b6c
HANGUP	Front Desk	10	10	from-internal	PJSIP/10-00000b6c
CHAN_END	Front Desk	10	10	from-internal	PJSIP/10-00000b6c
CHAN_START			401253xxxx	from-internal	Local/401253xxxx@from-internal-00000042;1
CHAN_START			401253xxxx	from-internal	Local/401253xxxx@from-internal-00000042;2
CHAN_START			s	from-trunk	SIP/VOIPMS-00000a47
ANSWER		401253xxxx	401253xxxx	from-trunk	SIP/VOIPMS-00000a47
ANSWER		401640xxxx	s	macro-dialout-trunk	Local/401253xxxx@from-internal-00000042;2
ANSWER		10	10	from-internal	Local/401253xxxx@from-internal-00000042;1
BRIDGE_ENTER		10		from-internal	Local/401253xxxx@from-internal-00000042;1
BRIDGE_ENTER		401253xxxx		from-trunk	SIP/VOIPMS-00000a47
BRIDGE_ENTER	Name Omitted	401640xxxx	docfu	macro-exten-vm	SIP/VOIPMS-00000a46
BRIDGE_ENTER		401640xxxx	s	macro-dialout-trunk	Local/401253xxxx@from-internal-00000042;2
BRIDGE_EXIT	Name Omitted	401640xxxx	docfu	macro-exten-vm	SIP/VOIPMS-00000a46
BRIDGE_EXIT		10		from-internal	Local/401253xxxx@from-internal-00000042;1
HANGUP		10		from-internal	Local/401253xxxx@from-internal-00000042;1
CHAN_END		10		from-internal	Local/401253xxxx@from-internal-00000042;1
BRIDGE_EXIT		401640xxxx	s	macro-dialout-trunk	Local/401253xxxx@from-internal-00000042;2
BRIDGE_EXIT		401253xxxx		from-trunk	SIP/VOIPMS-00000a47
HANGUP		401253xxxx		from-trunk	SIP/VOIPMS-00000a47
CHAN_END		401253xxxx		from-trunk	SIP/VOIPMS-00000a47
HANGUP		401640xxxx	h	from-internal	Local/401253xxxx@from-internal-00000042;2
CHAN_END		401640xxxx	h	from-internal	Local/401253xxxx@from-internal-00000042;2
HANGUP	Name Omitted	401640xxxx	h	ext-local	SIP/VOIPMS-00000a46
CHAN_END	Name Omitted	401640xxxx	h	ext-local	SIP/VOIPMS-00000a46
LINKEDID_END	Name Omitted	401640xxxx	h	ext-local	SIP/VOIPMS-00000a46

The Front Desk extenstion does not always answer the call and join the bridge, but the strangest part is that a call is then intiated out to an external number. I’ve looked at the Extension, Find Me/Follow Me, the phone, but I cannot seem to figure out:

1. Why an outbound call is being initiated
2. Where the external number that’s being called has been configured

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@kendalldever15 wrote:

Is there a way to copy an IVR vs. having to create from scratch every time?
Thanks in advance.

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@georgedal wrote:

Hey everyone,

I have a vps server in production with asterisk now (Version: 13.18.3) . I have installed a2billing for billing purposes. The database of a2billing (mya2billing) was accidentally deleted and I immediately restored it from backups. After the mysqlchek for repairing the database everything was back in place and calls can be made successfully.

While using the GUI of freepbx if I make any changes and hit the apply button I get the following error:

exit: 255
Authentication service cannot retrieve authentication info
You (asterisk) are not allowed to access to (/usr/bin/crontab) because of pam configuration.
Unable to continue. Cron line added didn’t remain in crontab on final check in /var/www/html/admin/libraries/BMO/Cron.class.php on line 113
#0 /var/www/html/admin/libraries/BMO/Cron.class.php(180): FreePBX\Cron->addLine(’* * * * * [ -x …’)
#1 /var/www/html/admin/modules/timeconditions/Timeconditions.class.php(167): FreePBX\Cron->add(’* * * * * [ -x …’)
#2 /var/www/html/admin/modules/timeconditions/functions.inc.php(193): FreePBX\modules\Timeconditions->updateCron()
#3 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(95): timeconditions_get_config(‘asterisk’)
#4 /var/lib/asterisk/bin/retrieve_conf(864): FreePBX\DialplanHooks->processHooks(‘asterisk’, Array)
#5 {main}

While I’m using putty to ssh to the vps with the root account I can run amportal reload with no problems.

Something that I can’t explain is that If my use my provider’s VNC viewer for the vps I cannot login with the root account (saying password incorrect) but using the same credentials I am able to access it via putty.

Any help would be much appreciated

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@charlietoner wrote:

After the latest update, the UCP Server node will not start. FreePBX 14.0.3.6, UCP 14.0.2.7, Asterisk: 14.7.5.

This is in the ucperror.log:

2018-07-11 14:48 -04:00: Error: Cannot find module ‘ini’
at Function.Module._resolveFilename (module.js:547:15)
at Function.Module._load (module.js:474:25)
at Module.require (module.js:596:17)
at require (internal/module.js:11:18)
at Object. (/var/www/html/admin/modules/ucp/node/lib/freepbx.js:9:9)
at Module._compile (module.js:652:30)
at Object.Module._extensions…js (module.js:663:10)
at Module.load (module.js:565:32)
at tryModuleLoad (module.js:505:12)
at Function.Module._load (module.js:497:3)

Thoughts?

Charlie

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@PCS wrote:

We have a client who is complaining that our system answers TOO quickly. The IVR message plays before they hear any ringing. Is there a simple way of generating a ring or two before the IVR?

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@thanhdang wrote:

My company has FreePBX 5.211.65-21 and we just have NAS Sysnology DS218. I would like to install Cloud Station Drive (client software like OneDrive) to synchronize call recordings from FreePBX to our NAS. The Cloud Station Drive supports Windows, Mac, Fedora (rpm) and Ubuntu (deb). Could I setup the Cloud Station Drive on FreePBX 5.211.65-21?

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@slashroot wrote:

Hi,

I’m trying to upgrade the FreePBX distro to SNG7 by following the official guide(https://wiki.freepbx.org/display/PPS/Upgrading+from+FreePBX+10.13.66+to+SNG7).

Installing:
yum -y install http://package1.sangoma.net/distro-upgrade-1804-1.sng7.noarch.rpm

Then “distro-upgrade” says everything is fine and takes the server down for the first reboot.

The following is added to grub.conf(automatically):
default=0
timeout=5
hiddenmenu
title System Upgrade (redhat-upgrade-tool)
root (hd0,0)
kernel /boot/vmlinuz-redhat-upgrade-tool ro root=/dev/xvda1 console=ttyS0 xen_blkfront.sda_is_xvda=1 rd_NO_LUKS LANG=en_US.UTF-8 rd_NO_MD KEYTABLE=us crashkernel=auto SYSFONT=latarcyrheb-sun16 upgrade init=/usr/libexec/upgrade-init selinux=0 rd.plymouth=0 plymouth.enable=0 net.ifnames=0 consoleblank=0
initrd /boot/initramfs-redhat-upgrade-tool.img
title SHMZ (2.6.32-642.6.2.el6.x86_64)
root (hd0,0)
kernel /boot/vmlinuz-2.6.32-642.6.2.el6.x86_64 ro root=/dev/xvda1 console=ttyS0 xen_blkfront.sda_is_xvda=1 rd_NO_LUKS LANG=en_US.UTF-8 rd_NO_MD KEYTABLE=us crashkernel=auto SYSFONT=latarcyrheb-sun16
initrd /boot/initramfs-2.6.32-642.6.2.el6.x86_64.img
title CentOS 6.5 bashton1
root (hd0,0)
kernel /boot/vmlinuz-2.6.32-431.el6.x86_64 ro root=LABEL=/ console=ttyS0 xen_blkfront.sda_is_xvda=1 rd_NO_LUKS LANG=en_US.UTF-8 rd_NO_MD KEYTABLE=us
initrd /boot/initramfs-2.6.32-431.el6.x86_64.img

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@lgraham wrote:

FreePBX 13.0.195.1 with Polycom IP331 phones. My receptionist tells me this is NEW behavior. My SIP vendor tells me nothing has changed on their side (of course).

I have extensions that do not ring to endpoints, but have Follow Me enabled to call a cell phone. Voicemail is not enabled on the extensions, so that the cell phone’s voicemail will handle messages if the call is not answered.

Follow Me sends the call to the cell phone. If the call is answered, no problem.

However if the call is UNANSWERED (the cell phone’s voicemail picks up), our PBX loses the call. Fast-busy, hangup.

Initial Ring Time: 0
Ring Strategy: ringall2-prim
Ring Time:20
Destination/No Answer: Follow Me, Normal Extension Behavior

Is there something I am missing?

Thank you in advance.

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@sentinelace wrote:

Are there any third party modules that allow a voicemail to be transcribed to txt? We get the wav file but we have to listen to them, and log them into our system. Would be nice like on an Iphone where it’s typed out. Anyone done this?

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@brianbkis wrote:

I have 4 lines in a hunt group with my provider (Cox) and four trunks setup with Obi 110s as FXO Gateways (https://wiki.freepbx.org/pages/viewpage.action?pageId=4161594). Two of the trunks are working just fine…two aren’t. If the first two trunks are in use and a call is pushed to Trunk 3 we get the following.

• Inbound calls: The inbound caller just hears constant ringing. Handsets in the office will ring but when a user picks up they get a busy signal. When the user hangs up the call continues to ring…and on and on.
• Outbound calls: The outbound caller gets a busy signal when trying to place a call on the third or fourth trunk.

I believe I have all settings identical between the four trunks, aside from those that should be unique.

Here’s an example from the Asterisk Logfiles for an inbound call on Trunk 3:

[2018-07-12 10:29:43] VERBOSE[23136][C-0000066c] bridge_channel.c: Channel SIP/OBITRUNK3-00001ae7 left ‘simple_bridge’ basic-bridge <341f261e-b588-4af2-a3f4-ea5b76d9bb55>
[2018-07-12 10:29:43] VERBOSE[23136][C-0000066c] app_macro.c: Spawn extension (macro-dial, s, 22) exited non-zero on ‘SIP/OBITRUNK3-00001ae7’ in macro ‘dial’
[2018-07-12 10:29:43] VERBOSE[23136][C-0000066c] pbx.c: Spawn extension (ext-group, 500, 18) exited non-zero on ‘SIP/OBITRUNK3-00001ae7’
[2018-07-12 10:29:43] VERBOSE[23136][C-0000066c] pbx.c: Executing [h@ext-group:1] Macro(“SIP/OBITRUNK3-00001ae7”, “hangupcall,”) in new stack
[2018-07-12 10:29:43] VERBOSE[23136][C-0000066c] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/OBITRUNK3-00001ae7”, “1?theend”) in new stack
[2018-07-12 10:29:43] VERBOSE[23136][C-0000066c] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-07-12 10:29:43] VERBOSE[23199][C-0000066c] bridge_channel.c: Channel SIP/301-00001ae8 left ‘simple_bridge’ basic-bridge <341f261e-b588-4af2-a3f4-ea5b76d9bb55>
[2018-07-12 10:29:43] VERBOSE[23199][C-0000066c] app_stack.c: SIP/301-00001ae8 Internal Gosub(crm-hangup,s,1) start
[2018-07-12 10:29:43] VERBOSE[23199][C-0000066c] pbx.c: Executing [s@crm-hangup:1] NoOp(“SIP/301-00001ae8”, “Sending Hangup to CRM”) in new stack
[2018-07-12 10:29:43] VERBOSE[23199][C-0000066c] pbx.c: Executing [s@crm-hangup:2] NoOp(“SIP/301-00001ae8”, “HANGUP CAUSE: 16”) in new stack
[2018-07-12 10:29:43] VERBOSE[23199][C-0000066c] pbx.c: Executing [s@crm-hangup:3] ExecIf(“SIP/301-00001ae8”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2018-07-12 10:29:43] VERBOSE[23199][C-0000066c] pbx.c: Executing [s@crm-hangup:4] NoOp(“SIP/301-00001ae8”, “MASTER CHANNEL: 1531416575.7529 = 1531416575.7528”) in new stack
[2018-07-12 10:29:43] VERBOSE[23199][C-0000066c] pbx.c: Executing [s@crm-hangup:5] GotoIf(“SIP/301-00001ae8”, “1?return”) in new stack
[2018-07-12 10:29:43] VERBOSE[23199][C-0000066c] pbx_builtins.c: Goto (crm-hangup,s,8)
[2018-07-12 10:29:43] VERBOSE[23199][C-0000066c] pbx.c: Executing [s@crm-hangup:8] Return(“SIP/301-00001ae8”, “”) in new stack
[2018-07-12 10:29:43] VERBOSE[23199][C-0000066c] app_stack.c: Spawn extension (macro-dial, s, 1) exited non-zero on ‘SIP/301-00001ae8’
[2018-07-12 10:29:43] VERBOSE[23199][C-0000066c] app_stack.c: SIP/301-00001ae8 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

In that example, the User at 301 picked up and received the busy signal.

Any help would be greatly appreciated. Thanks!

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@WQGJ587 wrote:

I cant seem to keep XMPP Daemon running. Every time I go into Dashboard its down.

When I tried to search for info. I find some entries from several years ago. and the Repeated solution is.

pm2 v13.0.3.3

Well, not sure what kind of instructions that is. I thought it may be one of the other module’s but not sure which one. and since the solution is from 3 to 4 years ago, i figure all updates are done past that version anyway.

FreePBX 13.0.195.4

I am afraid to try the upgrade to 14 since I tried to install 14 some time ago when it was first being posted. It just will not run on my system. Cant seem to find a list of system requirements anywhere and find it to be a touchy, unanswered subject if I ask.
So this time, I wont ask. I’ll just try to see if anyone can help with current XMPP issue?

Tom

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@fearx wrote:

I’m using Zoiper, and it works with SIP. But on my webphone, the extension is PJSIP. It works pretty fine, but I got this doubt: Should I use PJSIP to Zoiper too?

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@yannickg wrote:

Hi, I’m currently testing out a small FreePBX install at home, it’s going rather well but it’s unclear to me what options I have for phone provisioning.

EPM is the commercial module at 149$
OSS EPM is free but seems not maintained anymore

Is there any other options, if so, is there any guide or tutorial I could read?

Thanks

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@codec wrote:

Hello everybody,
my asterisk logfile has lots of entries like this:
[2018-07-12 11:40:27] WARNING[19102] pbx_config.c: The use of ‘_.’ for an extension is strongly discouraged and can have unexpected behavior. Please use ‘_X.’ instead at line 5806 of /etc/asterisk/extensions_additional.conf

I checked and found a lot of these entries in file extensions_additional.conf like these:

[sub-record-hh-check]
include => sub-record-hh-check-custom
exten => _.,1,Noop(Callee: ${MIXMONITOR_FILENAME})
exten => _.,n(exit),Return()

As this is an autocreated file, I cannot change these entries within the file. What should I do to get rid of these warnings???

Thanks
Codec

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