I did it with a clean install. It worked.
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OSS Endpoint Manager no longer loads packages after upgrade to 13.0.2
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POE SIP phone w/o lcd
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POE SIP phone w/o lcd
that has a minimum order of 50.... did give me the thought to google "hotel phone" which lead to the http://www.voipsupply.com/grandstream-gxp1105
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"RELease" with a Cause-User Busy
Currently we are using FreePBX 2.11.043 in a series of Radio Stations. Our contest Ques are periodically full and our local "Telus Mobility" people are telling us that our PBX is not sending a proper busy code to their system.
Here is what our current configuration is sending:
Event 10 15/10/23 11:25:47 TEI 0
08 Q.931 Call Control Message
Call Reference Value Dest 239
M 01 Message Type ALERTING
I 1E Progress Indicator Length= 2
81 Coding Standard CCITT
Location Local Private Network
88 Value In-band Information/Pattern Avail
Raw Hex Data
08 02 80 EF 01 1E 02 81 88
Event 11 15/10/23 11:25:47 TEI 0
08 Q.931 Call Control Message
Call Reference Value Dest 239
M 45 Message Type DISCONNECT
I 08 Cause Length = 2
81 Coding Standard CCITT
Location Local Private Network
91 Class Normal Event
Value #17, User Busy
and this is what they are expecting (as shown from another system):
==> 00:06:55:13.41 (CM Time: 11:06:32:18.95).
==> Q931: SETUP: to S[7040] L[1,474,0] E[37,473,0] SPA[----]
CR: 0,15 F5
BC: speech
64 kbit/s
circuit mode
mu-law speech
CID: 18
LENGTH: 03
Channel Selection Info: As Indicated in Following Octets
D-Channel Indicator: D-Channel NOT indicated
Preferred/Exclusive: Exclusive
Interface type = primary rate
Interface Identifier: IID Implicitly Identified
Channel Type: B - Channel Units (3).
Number Map: Channel is indicated by the number following.
Coding Standard: CCITT
Channel Number = 1
FAC:
Protocol Profile: Networking Extensions
8B 01 00 A1 0F 02 01 5D 06 07 2A 86 48 CE 15 00
04 0A 01 00
Interpretation Component
Interpretation Value: Discard
Invoke Component
Invoke Identifier Tag: 2
Invoke identifier: 93
Operation Component
Operation value: informationFollowing
PI: public_network_serving_local_user
origination_address_is_non_ISDN
CGN: e164
national_number
network_provided
presentation_allowed
6042770650
CDN: e164
national_number
604xxxxxxx
<== 00:06:55:13.46 (CM Time: 11:06:32:19.00).
<== Q931: REL COM: from S[7040] L[1,474,0] E[37,473,0] SPA[----]
CR: 1,15 F5
CSE: user
user_busy
Can anyone give me a hand on how to rectify my problem?
I do have a sleazy work around for this but it is not a good one... it involves a failover for the que to a "busy message and hangup".... not polite....
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Different CallerIDs for different outbound routes
I have been trying for about a week to get outbound routes working with different caller IDs, but they all still show my trunk outbound CID regarless of which route I use.
All routes are using the same trunk.
I am using the Extension Routing commercial module to limit certain extensions to certain outbound routes, and each outbound route has caller ID set with a route CID, override extension => yes, and is not set as an emergency route.
Calling out of all routes still presents my trunk CID. The trunk is outbound CallerID is set to "Block Foreign CIDs", not to "Force Trunk CID". The trunk "Hide CallerID" is set to "No".
How in the world does one make this work?
This is similar to this question:
http://community.freepbx.org/t/the-free-extension-routes-module-is-installed-but-not-registered/25779/10 (at the bottom) and...
I'm using Asterisk 13.0.59 and FreePBX Distro 10.13.66-1.
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CHAN_PJSIP disabled
The default 11 that comes with the RasPBX distro.
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CHAN_PJSIP disabled
Asterisk 11 doesn't support PJSIP so that's why it's getting disabled.
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CHAN_PJSIP disabled
I read that 12 should be avoided. Is there an argument for moving to 13 or should I stay with 11?
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Dial by speech, possible?
One of those wild ideas came to me today.
The Cisco phones we're using have a directory function, I've got it listing all the extensions so people can easily dial without memorising a list of extensions etc, it's functional but basically its garbage. Slow and cumbersome to use, not FreePBX's fault.
I want to do it by voice.
I want to press a softkey, say the name of the person and hear a ringing tone. Fast, simple.
As all the names exist already in the system, they would be 'known entities' making the speech recognition easier, we're not transcribing medical reports here!
This must also work for a transfer. Be on a call, press transfer, get dial tone, press the special key, say the name, hang up to wait for answer for attended transfer. The important part here is it mustn't transfer the call to the speech recogniser!
So. Standard questions: Does it exist and is someone already working on it / completed it before?
I've no problem purchasing a recogniser, as long as the price is sane. looking for feedback and suggestions.
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Dial by speech, possible?
possible YES....
These often need proprietary and "not free (as in beer or freedom) libraries.
https://wiki.asterisk.org/wiki/display/AST/Speech+Recognition+API
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Dynamic IP behind NAT not working for me, so audio issues
Hi there, I've been reading several posts about the problem I have and I finally found the problem. I'm behind a NAT and I had 2 way audio issues after following all the official tutorials -having and configuring a dynamic IP-. I tried setting my IP up as a 'Static IP' and it worked like a charm, so my problem is with DNS resolving.
I tried 3 different DNS providers with no luck, 2 way audio issues all the time. I even tried changing the refresh rate from 120 to 60s, just in case.
So, I'm stuck!! Any ideas? You are all welcome to say your piece.
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Inbound call routing on a SIP trunk for PJSIP?
Thanks for the insight.
I took a hint, and went back to chan_sip for the trunk. The endpoints seem to be doing fine on PJSIP for the moment.
Cheers,
Thomas
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IVR Star or Pound Key denied
I'm running on the latest version of FreePBX, and while setting it up i bumped into this issue.
When i try to add a Star or Pound key to the IVR Menu, after pressing Submit a error pops up and says: "Please enter a valid value for Digits Pressed"
Tried rebooting the PBX, any idea how to solve this?
Thanks
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Dial by speech, possible?
Oooooh, that looks easy!
LumenVox mentioned in their sample, looks like $3,500 purchase and $1000 per year, possibly more, and of course they seem not to have a trial version that I can find.
I'd certainly prefer it to cost less! I like the idea though, if it worked perfectly (ha!) then it sure would improve the user acceptance levels.
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Provision and update Grandstream GXP2130
What do you think of the Sangoma phones? (Can we have honest conversations about them here?)
Our standard phone was the Yealink SIP-T26P. It was a good price point, and we loved the 10 extra buttons on the side. They're no longer in production but we're playing with the Grandstream GXP2160s as our potential replacement. Again, price is good, and they have 24 programmable buttons which is fantastic. But there's just a few things that are bothersome, like the fact you can't remotely reboot them and that you can't go backward with the firmware if you need to.
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OSS Endpoint Manager no longer loads packages after upgrade to 13.0.2
Edit: Holy smokes, it was all just a browser issue. Using Firefox now and no issues.
I'm having serious problem like @cyberco, although mine seem a bit worse.
Updated to 13.0.6.6, and now can't even view extensions. Receiving errorCan't Load Local Endpoint Manager Directory!
Tried uninstalling and removing OSS Endpoint completely, and can then view extensions.
Reinstalled, same problem comes back (and now don't have lost my configuration, but can rebuild that I guess).
Tried importing packages and extensions csv manually, but new version cannot upload these files or set any other settings in fact as has been mentioned previously in this thread, so doesn't seem working for me.
Now on beta version, but same results as above.
Next up I've tried a new install like @tm1000 suggested, but same problems. I installed OSS EPM first, and checked (same issue). Then upgraded other modules and still the same problem.
One other thing that may be of note, when trying to uninstall it on my production instance, I receive the following (although does appear to be uninstalled).Error(s) removing endpointman: Failed to run un-installation scripts
Is there a way of perhaps installing the old version manually from before this started happening?
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Fax to filesystem
hi,
I just started using freePBX 13 after having asterisk successfully running on a small system for a couüle of trunks and a straight forward, simple configuration including fax.
With freePBX i only see the capability of using fax2mail and mail2fax, but for now I urgently need faxes to be simply stored in the filesystem, because I want a DMS to manage those documents. With asterisk it is for receiving faxes quite easy (sending I didn't implement yet), but how should I configure it in freePBX? I guess it is not doable by GUI, so I have to configure it in some files. Can I simply add this inbound route to that extensions_forgotthesuffix.conf and remove everything else for this trunk from freePBX?
regards,
andre
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Provision and update Grandstream GXP2130
I love the idea of the zero touch provisioning on the Sangoma. All of my workers are remote, so when new hires come in, I can just drop ship and not have to lay hands on it, or do a remote desktop session to set it up. I wish there was bluetooth on the Sangoma, but not a dealbreaker. I understand that it may be coming on later versions. The tight integration with FreePBX will make it a go to for me provided there is something the client needs that is not there.
LIke you, I hate the Grandstream firmware that you can't roll back. It seems they are getting more sloppy with buggy firmware, in my opinion. The 1.0.4.17 bricked several of my phones before I was able to stop the updates. That and the fact that GS is not very receptive to getting issues solved. They will tell you to open tickets, but thats as far as the process goes. I have talked with many on their forums that have the same bugs, and GS will tell you they can't recreate it, and be done with it. In hindsight, I would have much rather spent $30 or so per handset more, and not have to put as many hours in troubleshooting them. I bought into the low price point for the client, but as they say "you get what you pay for." The Sangoma's really have a great price point for the feature set, so I'm really excited to get my hands on one. I preordered from E4.
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Custom Destination Based on Call Path
Hey guys, I've got an interesting request and I quite frankly don't know if it's even possible. I just recently changed our Cisco phones to run on Skinny firmware with the chan-sccp-b module and it changed the way our reject button works. It used to be that if I pushed reject (or iDivert as Cisco calls it), the phone would simply send a busy signal back to Asterisk and the call would handle as normal. With the new firmware, the phone actually transfers the call to an extension that I can define in its configuration.
What I'd like to do is make a custom feature code of sorts that sends the call to one of two destinations based on where it came from. Right now we have a ring group for our main inbound number which rings all the phones in our office, and I also have my own inbound number that goes directly to my phone. I'd like to make it so that if I press reject on my phone on the main line then it goes to the main voicemail, and if it's to my direct line I'd like it to go to my personal voicemail. Is there any way to do this? Thanks in advance!
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Digium Addon module missing new install 10.13.66-9
building a new server from the distro 10.13.66-9. ran all updates in the from module admin
the Digium Addon module missing from the admin pull down and from module admin.
the wiki states to "yum install -y php-digium_register"
the CLI gets the response nothing to do.
Any help appreciated.
Chris
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