@AmidouFlorian92 wrote:
Hi
I have installed Freepbx version 14.0.10.3 and I want to know that If asterisk java is installed by default. How can I check this?
Thank you
Regards
Amidou Florian TOURE
Posts: 3
Participants: 2
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Asterisk-java
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Res_pjsip.c:4261 endpt_send_request: Error 171060 'Unsupported transport (PJSIP_EUNSUPTRANSPORT)'
@Wanderer wrote:
What does the message in the subject line mean? I tried to change some extensions from SIP to PJSIP but as soon as I apply the changes, on all PJSIP extensions I get these types of errors:
[2019-11-12 05:05:18] ERROR[3957]: res_pjsip.c:4261 endpt_send_request: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending NOTIFY request to endpoint 1100
[2019-11-12 05:05:18] ERROR[1200]: res_pjsip.c:4261 endpt_send_request: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending NOTIFY request to endpoint 1102
[2019-11-12 05:05:18] ERROR[1201]: res_pjsip.c:4261 endpt_send_request: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending NOTIFY request to endpoint 1104
[2019-11-12 05:05:18] ERROR[3958]: res_pjsip.c:4261 endpt_send_request: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending NOTIFY request to endpoint 1103When this happens those extensions become unavailable until Asterisk is restarted. What does this message mean, and is there some way to stop whatever is causing it? Changing the extensions back to CHAN_SIP fixes the problem, but then I’m not using PJSIP.
Posts: 4
Participants: 2
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Deactivate Trunk or outgoing route after x minutes of outgoing call
@Autourdupc wrote:
Hi all.
I would like to know if there is a possibility to deactivate a trunk or an outgoing route after a counter of time of outgoing call, and enable it again after a specific reccuring date.
I have a SIM card that permits only 2 hours of free outgoing calls, then after this 2 hours, calls are extra invoiced.
I would like to be able to stop using the SIM card (GSM gateway with chan-dongle) after 2 hours of outgoing calls, then enable it again next month at anniversary date.Maybe this can be an external script that stop dongle service and restart it after a period.
Any idea how to get the outgoing time counter on a specific trunk ?Regards,
Laurent.
Posts: 4
Participants: 3
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Music On Hold
@AlohatekATX wrote:
Is there a file size and/or length limit for music-on-hold file uploads? I have successfully uploaded a new seasonal .MP3 file for music-on-hold and applied config. FreePBX converted it to a .WAV file. It was 00:03:56 long and 7.20 MB in size. I deleted the existing file. However, the existing one still plays. I trimmed down the seasonal file to 00:02:17 and 4.18 MB and uploaded it successfully and applied config. Still, the existing one plays. Is it a FreePBX issue? Thanks!
Posts: 2
Participants: 2
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Cisco Spa3000 and Vertical Service Activation Codes
@Whacka wrote:
I have a cisco SPA3000 to make outgoing calls. I am wanting it to dial *67 when pressing 7 before the 10 digit phone number. How ever I’m am not having much luck getting it to do both plain 10 digits and adding the *67 before it. I was able to get it to answer dial *67 with a phone number, but it will no longer answer to plain 10 digit numbers. It just says ‘all circuts are busy now, PTYCAL’
When I add *67 to the dial plan 1 it works, but then I can’t dial normal 10 digits, the SPA3000 will not answer to make the outgoing call
When removed from the dial plan, it will no longer answer to numbers with *67
Here are my dial plans in Freepbx and in my SPA3000, dial plan one is currently (xx.) With this I can make plain 10 digit number phone calls.
with (*67xx.) I can make ONLY calls with *67 attached to 10 digit phone numbers. The SPA will not answer pressing 9 before NXXNXXXXXX, nor dialing NXXNXXXXXXDoes anyone know how I can make it dial both?
![spadial]()
Posts: 2
Participants: 2
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Res_pjsip.c:4261 endpt_send_request: Error 171060 'Unsupported transport (PJSIP_EUNSUPTRANSPORT)'
@Wanderer wrote:
What does the message in the subject line mean? I tried to change some extensions from SIP to PJSIP but as soon as I apply the changes, on all PJSIP extensions I get these types of errors:
[2019-11-12 05:05:18] ERROR[3957]: res_pjsip.c:4261 endpt_send_request: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending NOTIFY request to endpoint 1100
[2019-11-12 05:05:18] ERROR[1200]: res_pjsip.c:4261 endpt_send_request: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending NOTIFY request to endpoint 1102
[2019-11-12 05:05:18] ERROR[1201]: res_pjsip.c:4261 endpt_send_request: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending NOTIFY request to endpoint 1104
[2019-11-12 05:05:18] ERROR[3958]: res_pjsip.c:4261 endpt_send_request: Error 171060 ‘Unsupported transport (PJSIP_EUNSUPTRANSPORT)’ sending NOTIFY request to endpoint 1103When this happens those extensions become unavailable until Asterisk is restarted. What does this message mean, and is there some way to stop whatever is causing it? Changing the extensions back to CHAN_SIP fixes the problem, but then I’m not using PJSIP.
Posts: 4
Participants: 2
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Cisco 6851 with EPM
@ralexander wrote:
Followed all the steps here: Cisco 6851 3PCC (Third Party Call Control) - not working at all. Any ideas on what to do next? That post is all I have as far as a clue.
Posts: 1
Participants: 1
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CID not flowing through pstn. Assistance required please
@splows wrote:
I’ve set what I believe are the various CID areas and looking at the Call Detail Record for a test call I see thats the CallerID is assigned.
On the call event log I also see the test CID flowing through as below, but on the dialed phone I don’t get 12345678900, I get Unknown. I would greatly appreciate some assistance please. Thank you.
Time Event Type UniqueID LinkedID Cid num Extension Context Channel Name Wed, Nov 27, 2019 8:09 AM CHAN_START 1574842151.14 1574842151.14 105 9447342052XXX from-internal PJSIP/105-0000000a Wed, Nov 27, 2019 8:09 AM CHAN_START 1574842151.15 1574842151.14 s from-trunk-sip-didlogic SIP/didlogic-00000000 Wed, Nov 27, 2019 8:09 AM ANSWER 1574842151.15 1574842151.14 9447342052XXX 9447342052XXX from-trunk-sip-didlogic SIP/didlogic-00000000 Wed, Nov 27, 2019 8:09 AM ANSWER 1574842151.14 1574842151.14 12345678900 s macro-dialout-trunk PJSIP/105-0000000a Wed, Nov 27, 2019 8:09 AM BRIDGE_ENTER 1574842151.15 1574842151.14 9447342052XXX from-trunk-sip-didlogic SIP/didlogic-00000000 Wed, Nov 27, 2019 8:09 AM BRIDGE_ENTER 1574842151.14 1574842151.14 12345678900 s macro-dialout-trunk PJSIP/105-0000000a Wed, Nov 27, 2019 8:09 AM BRIDGE_EXIT 1574842151.14 1574842151.14 12345678900 s macro-dialout-trunk PJSIP/105-0000000a Wed, Nov 27, 2019 8:09 AM BRIDGE_EXIT 1574842151.15 1574842151.14 9447342052XXX from-trunk-sip-didlogic SIP/didlogic-00000000 Wed, Nov 27, 2019 8:09 AM HANGUP 1574842151.15 1574842151.14 9447342052XXX from-trunk-sip-didlogic SIP/didlogic-00000000 Wed, Nov 27, 2019 8:09 AM CHAN_END 1574842151.15 1574842151.14 9447342052XXX from-trunk-sip-didlogic SIP/didlogic-00000000
Posts: 1
Participants: 1
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Fwconsole reload error
@dux wrote:
Helo, everyone.
After upgrading asterisk today to version 16.2 I have encountered the following php error while reloading, in Sipsettings.class.php, line 433: json_decode() exprects parameter 1 to be string, array given. Asterisk works, but the upgrade itself did not run smoothly this time, probably due to this error. I had to manually restart asterisk after upgrade.
Posts: 15
Participants: 2
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Queue reports
@Spyder13337 wrote:
Good Afternoon,
I require to know how long a person is on being in a queue there for example if the caller ID of 9545845175 was waiting for 20 seconds before have the phone answer is there a report for this in freepbx somewhere
Posts: 6
Participants: 3
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Bulk SIP Extension password change
@dobrosavljevic wrote:
Is there a way to easily change/set the password of all the extensions in a system at once? We imported several hundred extensions through a CVS file but didn’t realize that the password wasn’t auto generated while doing this just like it would when creating a new extension manually and now are facing the daunting task of having to set a password manually for each extension.
Is there an easy way to have the system auto set those using the same facility that is being used when manually creating the extension?
Posts: 5
Participants: 2
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Users Transferring Calls to Pager
@RealRuler2112 wrote:
Several times since migrating to PBXact, customer calls have ended up being transferred to our pager. The pager is a Valcom unit interfaced through a Cisco SPA3104, which in turn is connected to our network and linked to the PBX via a SIP trunk/outbound route. When this happens, the customer does not know they are on our paging system and the only way to get the customer off our overhead speakers is to go downstairs to physically unplug the SPA - there’s no way for one of our users to pick up the call after it’s been transferred.
Our operator wants me to ‘prevent any call from being transferred to the pager’. I explained to him that a transfer to the phone system is really placing an existing call on hold and placing a new call to the pager, then connecting the two & disconnecting the phone so if I stop calls from being able to be transferred to the pager, it would stop anyone from being able to call it; I don’t think he got it. (Not 100% on this - I pieced it together from how I’ve observed the phone behaving & what’s in the CDRs for transfers.)
One idea I had was to create a password on the outbound route, so everyone would have to push something to confirm they want to page, but he thinks this will create more headaches than it is worth.
This is clearly a user training issue, but they’re looking for a technical solution to it. My ideas, some of which I don’t know if they’re even possible and would like input on, are:
Is there indeed any way to prevent transfers to a specific extension/route/trunk?
Is there any way of setting a maximum number of seconds for calls to a specific extension/route/trunk? (While this would not prevent this from happening, it would mitigate the effect of it because it’d drop the person off the overhead after n seconds.)
Any other ideas at all? Really grasping at straws here…
Posts: 1
Participants: 1
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Freepbx - Asterisk 15.5.0 - cut words in some sentences
@mst wrote:
Experts,
I have see some problems last week and get reported by end users about some cutting. If I count from 1 to 20 some numbers are cut and the other person do not hear that. Is happening on all phones. Well recently I have changed phone from Polycom to Yealinks. Yealink with newest firmware: 66.84.0.90
I dont see VAD enable - its disabled so this is the only thing comes to my mind. I have checked QOS on my switches and firewall - I see pockets are categorised correctly so this is not an issue here. I know it smells like QoS but its not. I dont see any errors in call in asterisk besides this:
WARNING[28216][C-000002d4]: translate.c:405 framein: no samples for ulawtolin
I don’t think this cause the problems with Yealink phones. I tried both ulaw and g729 - same issue. Seems like people started complaining after switching to Yealink phones. I am out of any ideas and only bang my head into a wall.
Posts: 1
Participants: 1
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Location of vpn logs
@Red wrote:
Where are the VPN logs to tell me if my vpn server is running or not. Also a way to tell if my tunnels are active. I can see by my vpn server, under sys admin pro, that none of my clients are connecting, and they were Monday.
Posts: 2
Participants: 1
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DID and dahdi on pri
@reyad wrote:
Helllo, i have a PRI E1 with 30 channels with one numerve that use as main numer, and 1 hundrend DIDs that are contiguos, our telco calls this GNR, not sure this is a common term.
In the past with my very old freepbx i was able to forward calls coming from GNR DIDs directly to extensions.
When i upgraded to version 14 i’m not able anymore to receive dids, i only receive an “s” and all calls are forwared to ivr.
anyone that have an advice?All the Best
Rayad
Posts: 2
Participants: 2
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Location of vpn logs
@Red wrote:
Where are the VPN logs to tell me if my vpn server is running or not. Also a way to tell if my tunnels are active. I can see by my vpn server, under sys admin pro, that none of my clients are connecting, and they were Monday.
Posts: 2
Participants: 1
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Just did my first Production 14-to-15 Upgrade - Nice!
@GSnover wrote:
That wasn’t even exciting - It worked perfectly - Thanks for all the hard work Sangoma Guys/Gals!
Posts: 1
Participants: 1
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Location of vpn logs
@Red wrote:
Where are the VPN logs to tell me if my vpn server is running or not. Also a way to tell if my tunnels are active. I can see by my vpn server, under sys admin pro, that none of my clients are connecting, and they were Monday.
Posts: 2
Participants: 1
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Freepbx outbound call not working via Mitel 3300
@adaminnes wrote:
Hi
I am having an issue making an outbound call from freepbx v14 connected via a trunk to a Mitel 3300 that has the SIP provider terminated to it. This is from the UK
I can make inbound calls fine but outbound is failing. I believe it is down to my dial pattern but have tried a few combinations and made no difference. From what I have read 0044+0. should work as a catch all.
My trunk works and I have an outbound route setup connected to that trunk.
In my dial plan just now I have
0044+0.
I have tried
0XXXXXXXXXX
Any ideas where I am going wrong any help much appreciated.
Regards
Adam
Posts: 1
Participants: 1
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Location of vpn logs
@Red wrote:
Where are the VPN logs to tell me if my vpn server is running or not. Also a way to tell if my tunnels are active. I can see by my vpn server, under sys admin pro, that none of my clients are connecting, and they were Monday.
Posts: 2
Participants: 1
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