Follow this link , It’s for a 8800. But you will get an idea. You may have a problem with Dial Plan.
https://voipnotes.ru/instrukcii/nastroiyka-cisco-spa-8800/
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No incoming calls
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No incoming calls
In Inbound Routes your… Description (blank)…
DID Number(User ID from spa)…CallerID Number(Blank for any)…Set Destination(anywere you want toe call to go to within freepbx)
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What would you like to see added in FreePBX 15?
I second this! I did a 50 user setup the other day…and enabled Tour Mode, nobody used it so I had to configure 50 UCPs myself and now everybody is happy
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Redirecting to cell phone in ring group if extension is offline (follow-me?)
Hi Thomas,
I came across your posting as I was trying to achieve the same thing and initially had trouble.
I have an office phone and also SIP softphone on my cell phone. When someone calls me I want both to ring, BUT if the softphone is unreachable due to my cell being in a poor data coverage area or maybe the softphone client is not running, then I would like my cell to ring + my office phone.
I believe this is what you are trying to achieve.
I successfully achieved this by using a personal ring group within a primary ring group.
Similar to what you’re doing but without Misc destinations.
11…extension for dedicated SIP phone (Office Phone)
12…extension for a SIP softphone
13…Secondary Ring group (agents = 12, cell phone) (ring strategy “hunt”)
10…Primary Ring group for calling extension 11 and ring group 13 together (ring strategy “ringall”)
The trick was to;
- in the primary ring group 10, when specifying ring group 13 in extension list, you must suffix it with a # (this is the only way to have a ring group within a ring group).
- in Ring group 13, I found I needed to suffix my cell phone number with a #
- In Ring group 13, set the Ring Strategy to “hunt”. The PBX will try to call you soft phone first, then if it can’t, will call your cell #. Also if you “reject” the call on your soft phone, the call will come to you cell phone. This is handy if you want to take the call on your cell phone if you know you’re not in a good data area.
I also found the confirm calls option to work well in Ring group 13. Because I suffixed my cell phone with a #, and enabled confirm calls, the PBX would ask me to confirm the call when I answer my cell phone, but not when I answer from the soft phone or office phone.
This worked so well that I extended the config out to several others using the PBX and it is now a default config. I have created personal hunt groups for all users of the PBX containing their softphone and their cell so now when I want to add someone to a hunt group, I add their personal ring group instead of their extensions.
Hope this helps you and anyone else searching on how to achieve this.
Greg
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What would you like to see added in FreePBX 15?
A stripped-down version of Inbound Routes on a Per DID basis, that could be integrated into UCP (Admin) that would work like this:
DID Number XXXXXXXX
Normal Routing >>>> Yes (Radio Button) - If “No” then…
Change Routing>>>>>Input Field (User inputs Destination Number)
Save and send Email Confirmation to designated Email.
Why? Primarily for Hosted Systems so customers can modify where their numbers go in an emergency without calling in to my office
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What would you like to see added in FreePBX 15?
If you typed in ‘sip client for android phones’ you would get a lot of responses, zoiper and linphone are between them multiprotocol, multi account, both license free and unencumberred and and both integrate seemlessly into your android’s native dialing,
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Chan_dongle
Did you read the wiki yet? Inbound routes are kind of FreePBX 101.
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What would you like to see added in FreePBX 15?
The issue with that is it requires a FreePBX reload to change that and everything we do in UCP we do with no reloads to Asterisk. You never want to let end users just modify and reload diaplan.
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I'm putting together a report and wouldn't mind some help
I’m creating a report on all the important locations that freepbx/asterisk keeps data. So far I know about the cdr records, auth logs, asterisk logs. Off the top of my head I cant think of any others but I know for sure theres much more I haven’t listed.
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What would you like to see added in FreePBX 15?
I’ve tested linphone and it crashed multiple times on my Droid Turbo so I gave up on it. Could be hardware-specific I don’t know. Zoiper is not free for commercial use and my own use is commercial so I naturally started by testing the free stuff first. Since csipsimple worked for me I stopped testing. If it hadn’t then I would have tested Zoiper.
I’ve read the Sangoma webpage on Zulu and I and none of my customers would use 90% of the features in it, otherwise I’m sure it’s great, but I am not in that market.
Much of the “phone system” market out there seems aimed at midsized to large, or to the kinds of companies who are currently wasting my time calling my cell phone trying to scam me with junk calls. But, I don’t sell to those customers because I am not marketing to that size business. You see once a business gets past maybe 40 head, IT and phone system companies come out of the woodwork every time a new system initiative is done. Those companies muddy the water and I would rather let 3 different “phone system” companies battle it out with each other over the fish in that pool.
I market to the under 15 head companies who are more price sensitive. With something like FreePBX I can go into one of those and lay out a 20 extension asterisk-based system with a trade-in credit on their old system. They end up with a modern VoIP system (modern compared to the 20 year old digital phone system they were using) that has a nice voicemail on it and a bit of capability to run an IVR and I take their old system and Ebay it off to pay for the “credit” and what I have just successfully done there was turned the phone system into an “IT” thing, instead of a “phone guy” thing in their minds. This is basically protecting my IT turf because if I let them go to “a phone guy” well a lot of times those phone guys want to do IT too. I don’t want a competitor in my customer. I cannot prevent it if they initiate contact to a competitor but I’m sure not going to refer them to one for a phone system that I could supply with FreePBX.
As for the Cisco debate it seems like every time the “C” word comes up in networking people either love their stuff or hate it. Well I’ve used their gear for years and I’ll tell anyone that the only way to really know with their product is to get it and try it. I didn’t test with any of their phones because I got Polycom stuff quicker and cheaper and it works great. Now I wish I had gotten some sample instruments and tested them. But in any case just to clarify with Cisco, if you go to their website and look at it you will see that all the SPA phones are gone, the older enterprise 7900 series are being de-emphasized, and they are moving to a single one-size-fits-all line of instruments, the 6800, 7800 and 8800 devices sold either with a multiplatform firmware load that will either speak SIP or SPCP depending on how it’s booted up and is specifically aimed at 3rd party softPBXes, or with the UCaaS/enterprise/skinny firmware. The multiprotocol devices (which can be distinguished by the 3PCC in their part number) were just released less than 9 months ago and specifically list asterisk compatibility, look at their datasheets. I just didn’t test one because I didn’t feel like throwing $250 at a phone. But I very much doubt they are as bad as a SIP-loaded 7900 series from years ago or an antique Cisco SPA 900 series.
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Can not get Cisco Phones to Register
I am getting the same error message on my 7970 but it has no impact on the operation.
I think it would be worth following @Stewart1’s line of thought - to figure out what the firewall is doing here. Is it possible to turn off the firewall completely just to eliminate that possibility?
@John45, did you specify the ntp server as “nist.gov” or did you provide the IP address? From memory, I think I had to provide the actual IP address. If you are not seeing any ntp packets and if you have specified the dns name then one possibility is that the phone cannot resolve the name at boot time and hence not sending out ntp requests.
@John45, Here a resource pertaining to the exact model you have (be mindful that on post provides config settings on 7640 handset which is NOT applicable in your case as you have a 7641). Also noted that one of the issues highlighted there is that dhcp was not providing dns server details so you may want to check your dhcp settings to make sure dns server details are sent to the phone at boot time.
Of particular interest for you would be the simplified example config file given by ‘newboy’ towards the end of the post
https://forums.whirlpool.net.au/archive/1895964
Either way, I think it would be best to disable the firewall before you try this out.
Hope this helps.
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Can not get Cisco Phones to Register
@John45, and here is another good one…which is supposed to have been solved.
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I'm putting together a report and wouldn't mind some help
The log files record historical events.
Asterisk ONLY uses the files in/etc/asterisk/*.conf and /etc/asterisk/keys/* (for ssl) to define asterisks behavior, /etc/asterisk/voicemail.conf is an anomalous flat form ascii text file that maintains comedian mail’s (the voicemail system) structure and /var/lib/asterisk/astdb.sqlite3 which is an sqlite3 database that has the current “state” of the running asterisk,
FreePBX uses a mysql (type) backend that is used to regenerate the above as needed, there are two databases maintained, asterisk ( the asterisk machine itself) and asteriskcdrdb (the cdr records) the tables within each database further define the data. And a web site rooted at /var/www/html/admin that has helper functions to display and set all the above. /var/lib/asterisk/* /var/spool/asterisk/* and possiblly /home/asterisk/* also have pertinent “custom” data
Peripherally, the backup system maintains copies of all the above in various locations and with various levels of detail as defined by the user.
I am pretty sure that those are all you will need.
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Rhino analog card driver installation
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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DOException (42S22) SQLSTATE[42S22]: Column not found: 1054 Unknown column 'time_mode' in 'field list'
hi support , can we have other way to fix without reinstall ?
thank you in advance
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Chan_dongle
where can I find it? I have look through the pages
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Firewall Not working?
Recently I noticed my PBX’s are coming up even though the IP is not in the firewall. It was working before fine. The firewall is enabled. Did a module update change?
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Hangup on connect
Yes - trunks pjsip as well.
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Help converting sip trunk to pjsip
Hi,
can you please help me convert the follow sip trunk config to pjsip? Thanks!
PEER DETAILS:
username=+301234567890
type=peer
secret=XXXXXXXX
host=ims.otenet.gr
fromuser=+301234567890
qualify=yes
fromdomain=ims.otenet.gr
insecure=invite
USER DETAILS:
type=user
secret=XXXXXXXXX
context=from-trunk
host=ims.otenet.gr
Registration string:
+301234567890:XXXXXXXXXX:+301234567890@ims.otenet.gr@ims.otenet.gr:5060/+301234567890
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Fax error "FAXSTATUS="FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries"
Asterisk 13.91.1
FreePBX 14.0.2.10
SIP Settings: T.38 Passthrough: No
Inbound Route:
Detect Faxes: Yes
Fax Detection Type: SIP
Fax Ring: No
Fax Detection Time: 5
Destination: Fax Recipient --> Ext 950 (Fax Line)
User Management (Username 950)
Fax (enabled)
Email Address: populated
Email sending capabilities tested
Issue: Some attempts for inbound faxes (specifically from an online test fax service called FaxZero, result in a disconnection, however the 2nd attempt is successful. The error which stands out is "FAXSTATUS=“FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries”.
[2018-04-02 09:04:39] VERBOSE[29728][C-00001a6a] res_fax.c: Redirecting 'SIP/flowroute-00000208' to fax extension due to CNG detection
[2018-04-02 09:04:39] VERBOSE[29728][C-00001a6a] pbx.c: Spawn extension (from-trunk, fax, 1) exited non-zero on 'SIP/flowroute-00000208'
[2018-04-02 09:04:39] VERBOSE[29728][C-00001a6a] pbx.c: Executing [fax@from-trunk:1] Set("SIP/flowroute-00000208", "__DIRECTION=INBOUND") in new stack
[2018-04-02 09:04:39] VERBOSE[29728][C-00001a6a] pbx.c: Executing [fax@from-trunk:2] Goto("SIP/flowroute-00000208", "ext-fax,3,1") in new stack
[2018-04-02 09:04:39] VERBOSE[29728][C-00001a6a] pbx_builtins.c: Goto (ext-fax,3,1)
[2018-04-02 09:04:39] VERBOSE[29728][C-00001a6a] pbx.c: Executing [3@ext-fax:1] Set("SIP/flowroute-00000208", "FAX_FOR=Fax Line (3)") in new stack
[2018-04-02 09:04:39] VERBOSE[29728][C-00001a6a] pbx.c: Executing [3@ext-fax:2] NoOp("SIP/flowroute-00000208", "Receiving Fax for: Fax Line (3), From: "BLANK" <+17708240780>") in new stack
[2018-04-02 09:04:39] VERBOSE[29728][C-00001a6a] pbx.c: Executing [3@ext-fax:3] Set("SIP/flowroute-00000208", "FAX_RX_USER=3") in new stack
[2018-04-02 09:04:39] VERBOSE[29728][C-00001a6a] pbx.c: Executing [3@ext-fax:4] Set("SIP/flowroute-00000208", "FAX_RX_EMAIL_LEN=17") in new stack
[2018-04-02 09:04:39] VERBOSE[29728][C-00001a6a] pbx.c: Executing [3@ext-fax:5] Goto("SIP/flowroute-00000208", "s,receivefax") in new stack
[2018-04-02 09:04:39] VERBOSE[29728][C-00001a6a] pbx_builtins.c: Goto (ext-fax,s,3)
[2018-04-02 09:04:39] VERBOSE[29728][C-00001a6a] pbx.c: Executing [s@ext-fax:3] StopPlayTones("SIP/flowroute-00000208", "") in new stack
[2018-04-02 09:04:39] VERBOSE[29728][C-00001a6a] pbx.c: Executing [s@ext-fax:4] ReceiveFAX("SIP/flowroute-00000208", "/var/spool/asterisk/fax/1522674278.778.tif,f") in new stack
[2018-04-02 09:04:39] VERBOSE[29728][C-00001a6a] res_fax.c: Channel 'SIP/flowroute-00000208' receiving FAX '/var/spool/asterisk/fax/1522674278.778.tif'
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] pbx.c: Executing [s@ext-fax:5] ExecIf("SIP/flowroute-00000208", "1?Set(FAXSTATUS="FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries")") in new stack
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] pbx.c: Executing [s@ext-fax:6] Hangup("SIP/flowroute-00000208", "") in new stack
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] pbx.c: Spawn extension (ext-fax, s, 6) exited non-zero on 'SIP/flowroute-00000208'
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] pbx.c: Executing [h@ext-fax:1] GotoIf("SIP/flowroute-00000208", "1?failed") in new stack
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] pbx_builtins.c: Goto (ext-fax,h,104)
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] pbx.c: Executing [h@ext-fax:104] NoOp("SIP/flowroute-00000208", "FAX "FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries" for: Fax Line (3) , From: "BLANK" <+17708240780>") in new stack
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] pbx.c: Executing [h@ext-fax:105] Macro("SIP/flowroute-00000208", "hangupcall,") in new stack
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/flowroute-00000208", "1?theend") in new stack
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/flowroute-00000208", "0?Set(CDR(recordingfile)=)") in new stack
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] pbx.c: Executing [s@macro-hangupcall:4] NoOp("SIP/flowroute-00000208", " monior file= ") in new stack
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] pbx.c: Executing [s@macro-hangupcall:5] AGI("SIP/flowroute-00000208", "attendedtransfer-rec-restart.php,,") in new stack
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] res_agi.c: <SIP/flowroute-00000208>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] pbx.c: Executing [s@macro-hangupcall:6] Hangup("SIP/flowroute-00000208", "") in new stack
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'SIP/flowroute-00000208' in macro 'hangupcall'
[2018-04-02 09:05:01] VERBOSE[29728][C-00001a6a] pbx.c: Spawn extension (ext-fax, h, 105) exited non-zero on 'SIP/flowroute-00000208'↧