Alright. I’ll do that. Thanks so much for your help!
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2x A400 is causing server to be unusable
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Execute AGI Script 15 seconds after call is answered
Your priority 1 answers the call, no prob
nothing is heard by the caller until after 3 seconds of silence you call your agi script, (should that not need to be 15?) but it is unlikely that your agi script is set to play any media on the channel.
What experience do you want your callers experience for the first 3/15 seconds?
post your agi script.
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Updating Asterisk to 15 from 13 broke my install
good call… there was NO manager.conf but there WAS manager.conf.rpmnew and manager.conf.rpmsave… I copied conf.rmpsave to .conf and the GUI has “found” asterisk - but the whole thing is broken – BUT – I have:-
/etc/asterisk/amd.conf.rpmsave
/etc/asterisk/ari.conf.rpmsave
/etc/asterisk/asterisk.conf.rpmsave
/etc/asterisk/ccss.conf.rpmsave
/etc/asterisk/cdr.conf.rpmsave
/etc/asterisk/cdr_adaptive_odbc.conf.rpmsave
/etc/asterisk/cel.conf.rpmsave
/etc/asterisk/cel_odbc.conf.rpmsave
/etc/asterisk/chan_dahdi.conf.rpmsave
/etc/asterisk/confbridge.conf.rpmsave
/etc/asterisk/dnsmgr.conf.rpmsave
/etc/asterisk/enum.conf.rpmsave
/etc/asterisk/extconfig.conf.rpmsave
/etc/asterisk/extensions.conf.rpmsave
/etc/asterisk/features.conf.rpmsave
/etc/asterisk/http.conf.rpmsave
/etc/asterisk/iax.conf.rpmsave
/etc/asterisk/indications.conf.rpmsave
/etc/asterisk/logger.conf.rpmsave
/etc/asterisk/manager.conf.rpmsave
/etc/asterisk/meetme.conf.rpmsave
/etc/asterisk/modules.conf.rpmsave
/etc/asterisk/motif.conf.rpmsave
/etc/asterisk/musiconhold.conf.rpmsave
/etc/asterisk/pjsip.conf.rpmsave
/etc/asterisk/pjsip_notify.conf.rpmsave
/etc/asterisk/queuerules.conf.rpmsave
/etc/asterisk/queues.conf.rpmsave
/etc/asterisk/res_fax.conf.rpmsave
/etc/asterisk/res_odbc.conf.rpmsave
/etc/asterisk/res_parking.conf.rpmsave
/etc/asterisk/rtp.conf.rpmsave
/etc/asterisk/sip.conf.rpmsave
/etc/asterisk/sip_notify.conf.rpmsave
/etc/asterisk/udptl.conf.rpmsave
/etc/asterisk/voicemail.conf.rpmsave
/etc/asterisk/xmpp.conf.rpmsave
/opt/isymphony3/server/conf/administrator.xml.rpmsave
/opt/isymphony3/server/conf/main.xml.rpmsave
/opt/isymphony3/server/conf/administrator/config.ini.rpmsave
/opt/isymphony3/server/conf/administrator/web.xml.rpmsave
/opt/isymphony3/server/conf/agent/config.ini.rpmsave
/opt/isymphony3/server/conf/agent/recording.xml.rpmsave
/opt/isymphony3/server/conf/agent/voicemail.xml.rpmsave
/opt/isymphony3/server/conf/agent/web.xml.rpmsave
/opt/isymphony3/server/conf/client/config.ini.rpmsave
/opt/isymphony3/server/conf/client/web.xml.rpmsave
/opt/isymphony3/server/conf/communication_manager/config.ini.rpmsave
/opt/isymphony3/server/conf/communication_manager/web.xml.rpmsave
/opt/isymphony3/server/conf/servers/default/config.ini.rpmsave
/opt/isymphony3/server/conf/servers/default/web.xml.rpmsave
/usr/local/ncpa/etc/ncpa.cfg.rpmsave
which I presume asterisk-version-switch did to me… or my fiddling… i will rename them all and see what that brings…
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Dynamic outbound call routing
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Raid storage issues - storing to wrong partition?
a 2gb /var partition might be a problem unless you carefully massage your log rotation.
But initially you said
“I have a system set up with Raid that seems to be storing all our info to the wrong partition”
can you be more explicit?. Your /home would likely just have a small asterisk directory so not a huge problem. Anything about LVM in your case is a red-herring, dont worry about it.
your / partition is indeed quite full, lets see the issue of
du -hx --max-depth=1 /|sort -h↧
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Extension not register over WAN
I captured the following message with the command “asterish -vvr”
Dialing out from the extension (x10)… to my cell phone
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Spawn extension (macro-dialout-trunk, s, 24) exited non-zero on ‘SIP/x10-0000003d’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 416xxxxxxx, 7) exited non-zero on ‘SIP/x10-0000003d’
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘SIP/x10-0000003d’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/x10-0000003d’
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Spawn extension (from-trunk, 416xxxxxxx, 1) exited non-zero on ‘SIP/AllOutBound-0000003f’
[2019-05-31 17:34:46] WARNING[12319]: chan_sip.c:4068 retrans_pkt: Retransmission timeout reached on transmission 5a9e364c-efe2a831-371b2e52@192.168.xx.xx for seqno 2 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2019-05-31 17:34:46] WARNING[12319]: chan_sip.c:4092 retrans_pkt: Hanging up call 5a9e364c-efe2a831-371b2e52@192.168.xx.xx - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Spawn extension (macro-dialout-trunk, s, 32) exited non-zero on ‘SIP/x10-0000003e’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 416xxxxxx9, 7) exited non-zero on ‘SIP/x10-0000003e’
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘SIP/x10-0000003e’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/x10-0000003e’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/x10-0000003e’
I really don’t what the message is saying. The thing is when I pick up the call on my cell phone the server hangs up the call. Can anyone explain what is going on with server?
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Execute AGI Script 15 seconds after call is answered
Hey @dicko (can’t help but laugh while typing that),
So preferably what I would like to happen is someone on our end in our office answers the phone (via an extension or ring group) and talks to the potential customer and after 15 seconds of talking with the customer my AGI script would execute and make a call to a URL via HTTP. The test.py AGI script right now is just a test script which writes to a file for testing purposes. I was just using to make sure my AGI script fires. Below is the AGI python script. So here is a little more about our use case depending on the amount of time someone stays on the phone with us we can sell a lead to a potential buyer. That is why we need the URL call to be made 15 seconds into the call as that is a requirement from one of our buyers. I hope this makes a little more sense and if not please let me know. Thanks again.
#!/usr/bin/python
from datetime import datetime
now = datetime.now()
f = open(’/tmp/test.txt’,‘a’)
f.write(str(now) + ‘\n’)
f.close()
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Extension not register over WAN
From the asterisk CLI
sip set debug ip 192.168.xx.xx
(FYI It is unnecessary to obfuscate private ip addresses)
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Execute AGI Script 15 seconds after call is answered
Using custom destinations or dialplan hooks you can easily do things before the dial happens or do something after hangup but it is excruciatingly difficult to do something on answer using only dialplan. The only way to accomplish this with dialplan is to call the macro/sub at time of dialing to the extension, and the confirm calls macro is already used for this purpose. So you could hack the confirm calls macro put your own dialplan there, but I think far better to use AMI (or even ARI) to track the call progress independently from dialplan. This is not a beginner project.
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Phones unreachable from outside
Okay. It won’t be until Monday, but I’ll try and get into the actual computer running this. I know it’s some Linux, but other than that I know nothing about it. Can I putty into it, or do I need to actually be at the machine? Or is that dependent on the OS itself?
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Execute AGI Script 15 seconds after call is answered
For the caller to speak to a person it needs to be answered by an endpoint that connects the call to a person, at that point the call is bridged ( asterisk is a B2BUA) and the dialplan will not continue until the answer-er hangs up or transfers the call to a context that handles your agi script.
As @lgaetz said, unless you are really good, you need to rethink your plan.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_DeadAGI
might be an option.
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Updating Asterisk to 15 from 13 broke my install
Well, that brought it back to life. I will take a backup to be sure and then think about updating to Asterisk 15 again… should “/sbin/asterisk-version-switch” work with this system?
FreePBX 15.0.16.2 and Asterisk 13?
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Setting auth_rejection_permanent to no
Hello. I have an issue where trunks stop trying to re-register after receiving a fatal response 480. In the pjsip.registration.conf file, the following is set: auth_rejection_permanent=yes. I assume changing this to “no” may resolve my issue. Are there any drawbacks to setting this to “no” that I may not be aware of?
Thanks for your help!
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Execute AGI Script 15 seconds after call is answered
So I like to think of myself as a pretty capable when it comes to coding etc. Let me ask if this might work.
I believe I can fork off an AGI execution and let the dialplan continue in which case the call would be answered and bridged to an extension via a ring group. Then I could have the AGI execution I forked off monitor the status of the call via AMI and finally after 15 seconds execute another AGI execution which would make the HTTP request? I looks like in Asterisk 1.6 the ability to execute an AGI execution via AMI is now supported. Let me know if that might work? I am hoping so.
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Execute AGI Script 15 seconds after call is answered
Maybe but you are in a world of howtos and whydoesntitwork
https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573
(asterisk currently supported versions (13 , 15 or 16) is many years ahead of 1.6)
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Execute AGI Script 15 seconds after call is answered
Good point and damn those pesky .'s! I am actually on 16.
core show version
Asterisk 16.1.1 built by root @ ip-10-0-0-161 on a x86_64 running Linux on 2019-01-30 19:03:31 UTC
With that said I just read the ARI documentation and I saw the words Resftul and JSON and immediately got a big smile on my face. I am going to dig deeper into this API and see if it can serve my purposes. You guys rock and I thank you very much. If any ideas on how to support my use case pop up in your mind please let me know. Thanks again.
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Execute AGI Script 15 seconds after call is answered
I would do this in a separate script, independent of Asterisk.
Do you really need to ‘present the lead’ while the prospect is still on the line? If not, I’d just read the CDRs e.g. once an hour, find calls that lasted more than 15 seconds, were answered by an agent in the appropriate group and were from a ‘new’ number. You can configure FreePBX to write CDRs into a simple CSV file.
If you need real-time, use CEL (call event logging) instead. Set FreePBX to create a CSV, have your script follow the growing file, start a 15-second timer when a candidate call is answered, cancel the timer if the call ends, post the lead if the timer expires.
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Execute AGI Script 15 seconds after call is answered
That is an entirely different approach but it might be a much easier one! I like it! Thank you so much.
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Execute AGI Script 15 seconds after call is answered
Nice, extend it with a well constructed trigger:-
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Installed Let's Encrypt but get "NET::ERR_CERT_AUTHORITY_INVALID" for UCP
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
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