Re: [asterisk-users] Asterisk - can't hear other side. Or other side does not hear us
Antony, Ok, I see what you are saying. Yes, than NAT occuring on our router. Asterisk server is on internal IP (192.168..) # Now that I read what you say I think there might be 2 issues. "Randomness" is one, but I am not even sure we have it (randomness). All recent complains were from specific callers and we can replicate those. #What am I looking for? Should it be "tshark" or there is means within Asterisk application/configs to write those logs?I haven't used "tshark" before. And more specifically, what am I looking for? Thank you! On Thursday 28 February 2019 at 17:40:28, Ivan Demkovitch wrote: > Noone connects to Asterisk box/server from outside. Callcentric SIP trunk> > configured and Asterisk maintains connection to it itself. Okay, I didn't actually mean "does anyone connect *inbound* to your Asterisk server" - I was more asking about the connectivity between the Asterisk box and (anything on) the Internet (eg: Callcentric), and whether there is NAT involved. > No special ports opened, nothing. Connection happens from us> to Callcentric > and all calls routed in from CallcentricI don't know> exactly how it's doing > it by it works. Does your Asterisk box have an RFC1918 address (ie: 10.0.0.0/8. 172.16.0.0/12 or 192.168.0.0/16), or does it have a public IP address on its own interface? If the address on your Asterisk server falls in the RFC1918 range, then you have NAT occurring on your router, and this is known to cause one-way (or sometimes no-way) audio from time to time. > Debugging with "tshark" should be done on Asterisk machine I asume? Yes, or else on any router between the Asterisk machine and the other end of the affected calls (ie: Callcentric). It's very probably simplest to do it on the Asterisk server, though. Antony. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - can't hear other side. Or other side does not hear us
Antony, It is correct. Noone connects to Asterisk box/server from outside.Callcentric SIP trunk configured and Asterisk maintains connection to it itself. No special ports opened, nothing. Connection happens from us to Callcentric and all calls routed in from CallcentricI don't know exactly how it's doing it by it works. Again, keep in mind it is working for many years for most / 90+% of calls #1. I don't think it will be possible to know :( Been 3 years. #2-3. All callers call public phone number and they all come in to asterisk from Callcentric context.When we call out - it goes out through Callcentric SIP trunk. When we dial internal each others extensions there is no NAT, trunk or anything else and all works just fine... Debugging with "tshark" should be done on Asterisk machine I asume? Thank you! On Thursday 28 February 2019 at 00:26:17, Ivan Demkovitch wrote: > Asterisk is NOT exposed to internet, noone connects to Asterisk> from > internet. We use Callcentric for VOIP trunk. That's the point where you lost me. Callcentric is out on the Internet. How does it connect to your Asterisk server? > External callers get in via Callcentric. Right... > 1. Outside caller calls us but can't hear us. I beleive they talked to their> > phone provider and it works now? It would be good to know what got changed to make that work. > 2. We have one caller where EVERY time they call - they can't hear us. They> > just say "ok, call us back". We call back and it works :) So, they connect in via Callcentric too? Just the same as Caller 1? > 3. We have one caller where when we call them - they cannot> hear us, but we > can hear them. They called back - all works. What is the difference between callers 1, 2 and 3, in terms of how they connect to your Asterisk server, or how you connect to them? > I feel like we need to trace SIP protocol. How do I do that? I may get on> of > those callers to work with us on testing. I would start with something like: # tshark -i any -f "port 5060" -w "sip.debug.pcap" and then afterwards look at the pcap file with tshark (tshark -r "sip.debug.pcap -V") or some SIP tool such as sngrep. Antony. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - can't hear other side. Or other side does not hear us
Antony, thanks for response! It wasn't technical, now it's getting there :) 1. It's asterisk 13.1-cert12. Network. I actually tried multiple things initially but now it's plain vanilla. No NAT. I have Asterisk on our network. All of our phones is IN our network as well, same subnet. All internal. Asterisk is NOT exposed to internet, noone connects to Asterisk from internet. We use Callcentric for VOIP trunk. External callers get in via Callcentric. We have Mikrotik router and SIP handlers were disabled from beginning I gathered more info now about 3 issues we seen1. Outside caller calls us but can't hear us. I beleive they talked to their phone provider and it works now?2. We have one caller where EVERY time they call - they can't hear us. They just say "ok, call us back". We call back and it works :)3. We have one caller where when we call them - they cannot hear us, but we can hear them. They called back - all works. So, as you see we don't have NAT stuff [general] dtmfmode=rfc2833 context=unauthenticated allowguest=no srvlookup=no udpbindaddr=0.0.0.0 tcpenable=no callcounter=yes ; This is to enable device state for queues nat=no session-timers=refuse type=peer context=internal host=dynamic disallow=all allow=ulaw allow=alaw I feel like we need to trace SIP protocol. How do I do that? I may get on of those callers to work with us on testing. Thanks! > Hello,> This is not technical post, Hm, no? > just looking for suggestions on what to check.I have asterisk for long time, Which version? > no updates, just maintain OS updates. I use SPA504G phones. Tell us about your network - where is Asterisk (inside your network, externally hosted on public IP address, other), where are your phones (inside one network (maybe the same as Asterisk is on), randomly distributed around the Internet, other), how do external callers manage to contact you? > Very rarely and randomly when we pickup a phone - other side does not hear> > us. Call them back and all works. Now I have couple people I'm talking to> > and it seems like very call like this. Someone can't hear someone. Don't> > know where to start to troubleshoot and what to look for. Short answer: NAT Longer answer: Check the type of firewall / router / NAT device you have between Asterisk and the phones (most likely at the telephone end) and see whether it offers "SIP ALG" (Application Layer Gateway) - if it does, turn it *off*. Also, check the sip.conf definitions you have for the phones which are affected by this, and make sure you have NAT set to one of yes, force_rport or comedia (you may hav eto experiment to see which works best in your environment). Check https://www.voip-info.org/asterisk-sip-nat/ for some guidance. https://www.voip-info.org/asterisk-sip-nat-solutions/ may also give you some further clues. On the other hand, note that https://www.voip-info.org/asterisk-config-sipconf/ is woefully outdated (at least as far as NAT is concerned). If none of that helps, I suggest doing a SIP packet trace at the server (and at the phone end if you can) and see what addresses are being passed between the two for RTP. That should tell you why one end can't contact the other. Regards, Antony. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - can't hear other side. Or other side does not hear us
Hello, This is not technical post, just looking for suggestions on what to check.I have asterisk for long time, no updates, just maintain OS updates. I use SPA504G phones Very rarely and randomly when we pickup a phone - other side does not hear us. Call them back and all works. Now I have couple people I'm talking to and it seems like very call like this. Someone can't hear someone. Don't know where to start to troubleshoot and what to look for. Thanks!-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue not dialing out to cell phone for some reason
Got it working! Thanks a lot again. As a bonus, is there a background on why
SIP/ did not work with a sip trunk provider? :)
From: John Kiniston
To: Ivan Demkovitch
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, November 16, 2018 3:08 PM
Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some
reason
So, LOCAL in this context is a 'Technology' or 'Channel Driver' , Instead of
PJSIP, SIP, IAX, it's sending a call to a dialplan target.
Your entry in queues.conf with LOCAL/105@internal would send the call to the
context 'internal' extension '105' and execute whatever that dialplan does.
The parameters I gave are actually part of the Queue member definition,
>From the example queues.conf:
Each member of this call queue is listed on a separate line in
; the form technology/dialstring. "member" means a normal member of a
; queue. An optional penalty may be specified after a comma, such that
; entries with higher penalties are considered last. An optional member
; name may also be specified after a second comma, which is used in log
; messages as a "friendly name". Multiple interfaces may share a single
; member name. An optional state interface may be specified after a third
; comma. This interface will be the one for which app_queue receives device
; state notifications, even though the first interface specified is the one
; that is actually called.
;
; A hint can also be used in place of the state interface using the format
; hint:@. If no context is specified then 'default' will
; be used.
So 0 is the Penalty for the user
Then 'eric' is the Member name
and the state interface is using the hint defined for the user.
On Fri, Nov 16, 2018 at 1:58 PM Ivan Demkovitch wrote:
John,
Thanks for reply! I use 13.1-cert1, plain vanilla Asterisk. Installed and
configured as per book..
So, from what I understand - LOCAL means I want local extension to be a member
of a queue.
For example, I have this:
[internal]
;Eric on extension 105
exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
same => n,VoiceMail(105@default,u)
Do I understand correctly that I should just put this in queues? That would
replace 2 members I had (office and cell)
member => LOCAL/105@internal,0,Eric,hint:105@internal
Can you direct me to specification of parameters under LOCAL (tried to search
but don't see any)what is 0? What is "Eric"? hint? Wonder what all of them do.
Also, my queues.conf setup like this:
timeout=30
retry=1
Which means if I send it to "Eric" - it will go to his voicemail after 30
seconds. Should I change timings?
Thank you!
From: John Kiniston
To: Ivan Demkovitch ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, November 16, 2018 2:43 PM
Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some
reason
My settings for the queue.log are in the [general] section of logger.conf
I'm running 13, I didn't see what version you said you were running.
If I wanted to add a LOCAL channel to my queue I'd do it as
member => LOCAL/7124@kiniston-intern,0,John,hint:7124@kiniston-intern
On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch wrote:
John,
FF1565AABB2D-SLS is probably invalid because it's not registered/lost
registration. This client is connected via VPN to our network, it usually works
when it's "warm". Not concerned about it too much.
155@callcentric OTOH is an actual cell phone that should be dialed out
via callcentric trunk. Maybe I'm smoking something thinking it was working
before. I know it works from
extensions.conf -[globals]
ERIC_CELL=SIP/155@callcentric...
exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
same => n,VoiceMail(105@default,u)
---
but in queues.conf I can't use same globals so I just put it in like that.What
do you mean by using LOCAL channel? Can you be more specific? I'm not very good
at this :)
This is logger.conf. Where(which section) should I place logging configuration?
[general]
dateformat=%F %T
[logfiles]
console => notice,warning,error,dtmf
messages => security,notice,warning,error,fax
verbose => verbose
Thank you!
From: John Kiniston
To: idemkovi...@yahoo.com
Sent: Thursday, November 15, 2018 3:17 PM
Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some
reason
OK.
So it looks like asterisk can't ring FF1565AABB2D-SLS because it's invalid.
is the user at '155' actually able the answer calls? I wouldn't expect
that agent to work configured that way, I'd use a LOCAL channel to direct the
call to a context that sets the call up before dialing out.
You configure queue logging in logger.conf , Look at the settings
queue_log = yes
queue_log_to_file = yes
queue_log_name = queue_log
On Thu, Nov 15,
Re: [asterisk-users] asterisk-users Digest, Vol 171, Issue 9
Sebastian,
Well, this can't be problem with trunk because:1. Call coming from outside, so
trunk works2. sip show registry shows it registered.
Trunk allows for 2 channels which is not a problem here either
It's just weird that out of 4 queue member only 2 being called and log doesn't
show anything else.
From: "asterisk-users-requ...@lists.digium.com"
To: asterisk-users@lists.digium.com
Sent: Thursday, November 15, 2018 11:20 AM
Subject: asterisk-users Digest, Vol 171, Issue 9
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Today's Topics:
1. Queue not dialing out to cell phone for some reason
(Ivan Demkovitch)
2. Re: Queue not dialing out to cell phone for some reason
(Sebastian Nielsen)
3. Re: Queue not dialing out to cell phone for some reason
(Ivan Demkovitch)
4. Re: Queue not dialing out to cell phone for some reason
(Sebastian Nielsen)
--
Message: 1
Date: Thu, 15 Nov 2018 16:53:38 +0000 (UTC)
From: Ivan Demkovitch
To: "asterisk-users@lists.digium.com"
Subject: [asterisk-users] Queue not dialing out to cell phone for some
reason
Message-ID: <897612684.1161831.1542300818...@mail.yahoo.com>
Content-Type: text/plain; charset="utf-8"
Hello,
I have queues.conf setup with a group like so:
[Sales](StandardQueue)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/1314555@callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink
So, my idea here that it should ring all 4 phones at the same time. And it does
work but randomly.I did trace a call and this is what I see. Only 2 phones
(internal) called. External SIP@callcentric is not being called.
Any idea why it's not being called?
-- Executing [1@automated_attendant_normal:1]
Verbose("SIP/callcentric15-0435", "1, Caller "DEMKOVITCH,IVAN"
<13144880983> has entered the sales queue") in new stack
Caller "aa" <155> has entered the sales queue
-- Executing [1@automated_attendant_normal:2]
Goto("SIP/callcentric15-0435", "queues,7001,1") in new stack
-- Goto (queues,7001,1)
-- Executing [7001@queues:1] Verbose("SIP/callcentric15-0435", "2,"aa"
<155> entering sales queue") in new stack
== "aa" <155> entering sales queue
-- Executing [7001@queues:2] BackGround("SIP/callcentric15-0435",
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
-- Playing
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')
-- Executing [7001@queues:3] Queue("SIP/callcentric15-0435",
"sales85") in new stack
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-0435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0437 is ringing
-- SIP/FF9EF375CCFC-SLS-0436 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
-- Playing 'queue-periodic-announce.ulaw'
(language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-0435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0439 is ringing
-- SIP/FF9EF375CCFC-SLS-0438 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
-- Playing 'queue-periodic-announce.ulaw'
(language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-0435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-043b is ringing
-- SIP/FF9EF375CCFC-SLS-043a is ringing
-- Stopped music on hold on SIP/callcentric15-0435
== Spawn extension (queues, 7001, 3) exited non-zero on
'SIP/callcentric15-0435'
-- next part --
An HTML attachment was scru
Re: [asterisk-users] Queue not dialing out to cell phone for some reason
John,
Thanks for reply! I use 13.1-cert1, plain vanilla Asterisk. Installed and
configured as per book..
So, from what I understand - LOCAL means I want local extension to be a member
of a queue.
For example, I have this:
[internal]
;Eric on extension 105
exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
same => n,VoiceMail(105@default,u)
Do I understand correctly that I should just put this in queues? That would
replace 2 members I had (office and cell)
member => LOCAL/105@internal,0,Eric,hint:105@internal
Can you direct me to specification of parameters under LOCAL (tried to search
but don't see any)what is 0? What is "Eric"? hint? Wonder what all of them do.
Also, my queues.conf setup like this:
timeout=30
retry=1
Which means if I send it to "Eric" - it will go to his voicemail after 30
seconds. Should I change timings?
Thank you!
From: John Kiniston
To: Ivan Demkovitch ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, November 16, 2018 2:43 PM
Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some
reason
My settings for the queue.log are in the [general] section of logger.conf
I'm running 13, I didn't see what version you said you were running.
If I wanted to add a LOCAL channel to my queue I'd do it as
member => LOCAL/7124@kiniston-intern,0,John,hint:7124@kiniston-intern
On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch wrote:
John,
FF1565AABB2D-SLS is probably invalid because it's not registered/lost
registration. This client is connected via VPN to our network, it usually works
when it's "warm". Not concerned about it too much.
155@callcentric OTOH is an actual cell phone that should be dialed out
via callcentric trunk. Maybe I'm smoking something thinking it was working
before. I know it works from
extensions.conf -[globals]
ERIC_CELL=SIP/155@callcentric...
exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
same => n,VoiceMail(105@default,u)
---
but in queues.conf I can't use same globals so I just put it in like that.What
do you mean by using LOCAL channel? Can you be more specific? I'm not very good
at this :)
This is logger.conf. Where(which section) should I place logging configuration?
[general]
dateformat=%F %T
[logfiles]
console => notice,warning,error,dtmf
messages => security,notice,warning,error,fax
verbose => verbose
Thank you!
From: John Kiniston
To: idemkovi...@yahoo.com
Sent: Thursday, November 15, 2018 3:17 PM
Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some
reason
OK.
So it looks like asterisk can't ring FF1565AABB2D-SLS because it's invalid.
is the user at '155' actually able the answer calls? I wouldn't expect
that agent to work configured that way, I'd use a LOCAL channel to direct the
call to a context that sets the call up before dialing out.
You configure queue logging in logger.conf , Look at the settings
queue_log = yes
queue_log_to_file = yes
queue_log_name = queue_log
On Thu, Nov 15, 2018 at 2:08 PM Ivan Demkovitch wrote:
John,
This is output of command below. How do I enable and log queue events?The
1555@callcentric is the one I'm curious about. I just tried calling into
"sales" again and it didn't change this "last was 1219067" output
Sales has 0 calls (max unlimited) in 'ringall' strategy (9s holdtime, 156s
talktime), W:0, C:4, A:6, SL:0.0% within 0s
Members:
SIP/155@callcentric (ringinuse disabled) (Not in use) has taken 4
calls (last was 1219067 secs ago)
SIP/FF4C119EEBF8-SLS (ringinuse disabled) (Not in use) has taken no calls
yet
SIP/FF1565AABB2D-SLS (ringinuse disabled) (Invalid) has taken no calls yet
SIP/FF9EF375CCFC-SLS (ringinuse disabled) (Not in use) has taken no calls
yet
No Callers
[Sales](StandardQueue)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/1314555@callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Astricon is coming up October 9-11! Signup is available at:
https://www.asterisk.org/community/astricon-user-conference
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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A human being should be able to change a diaper, plan an invasion, butcher a
hog, conn a ship, design a building, write a sonnet, balance accounts, build a
wall, set a bone, comfort the dying, take orders,
Re: [asterisk-users] Queue not dialing out to cell phone for some reason
John,
FF1565AABB2D-SLS is probably invalid because it's not registered/lost
registration. This client is connected via VPN to our network, it usually works
when it's "warm". Not concerned about it too much.
155@callcentric OTOH is an actual cell phone that should be dialed out
via callcentric trunk. Maybe I'm smoking something thinking it was working
before. I know it works from
extensions.conf -[globals]
ERIC_CELL=SIP/155@callcentric...
exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
same => n,VoiceMail(105@default,u)
---
but in queues.conf I can't use same globals so I just put it in like that.What
do you mean by using LOCAL channel? Can you be more specific? I'm not very good
at this :)
This is logger.conf. Where(which section) should I place logging configuration?
[general]
dateformat=%F %T
[logfiles]
console => notice,warning,error,dtmf
messages => security,notice,warning,error,fax
verbose => verbose
Thank you!
From: John Kiniston
To: idemkovi...@yahoo.com
Sent: Thursday, November 15, 2018 3:17 PM
Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some
reason
OK.
So it looks like asterisk can't ring FF1565AABB2D-SLS because it's invalid.
is the user at '155' actually able the answer calls? I wouldn't expect
that agent to work configured that way, I'd use a LOCAL channel to direct the
call to a context that sets the call up before dialing out.
You configure queue logging in logger.conf , Look at the settings
queue_log = yes
queue_log_to_file = yes
queue_log_name = queue_log
On Thu, Nov 15, 2018 at 2:08 PM Ivan Demkovitch wrote:
John,
This is output of command below. How do I enable and log queue events?The
1555@callcentric is the one I'm curious about. I just tried calling into
"sales" again and it didn't change this "last was 1219067" output
Sales has 0 calls (max unlimited) in 'ringall' strategy (9s holdtime, 156s
talktime), W:0, C:4, A:6, SL:0.0% within 0s
Members:
SIP/155@callcentric (ringinuse disabled) (Not in use) has taken 4
calls (last was 1219067 secs ago)
SIP/FF4C119EEBF8-SLS (ringinuse disabled) (Not in use) has taken no calls
yet
SIP/FF1565AABB2D-SLS (ringinuse disabled) (Invalid) has taken no calls yet
SIP/FF9EF375CCFC-SLS (ringinuse disabled) (Not in use) has taken no calls
yet
No Callers
[Sales](StandardQueue)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/1314555@callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Astricon is coming up October 9-11! Signup is available at:
https://www.asterisk.org/community/astricon-user-conference
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Queue not dialing out to cell phone for some reason
From: John Kiniston To: idemkovi...@yahoo.com Sent: Thursday, November 15, 2018 3:17 PM Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some reason OK. So it looks like asterisk can't ring FF1565AABB2D-SLS because it's invalid. is the user at '155' actually able the answer calls? I wouldn't expect that agent to work configured that way, I'd use a LOCAL channel to direct the call to a context that sets the call up before dialing out. You configure queue logging in logger.conf , Look at the settings queue_log = yes queue_log_to_file = yes queue_log_name = queue_log On Thu, Nov 15, 2018 at 2:08 PM Ivan Demkovitch wrote: John, This is output of command below. How do I enable and log queue events?The 1555@callcentric is the one I'm curious about. I just tried calling into "sales" again and it didn't change this "last was 1219067" output Sales has 0 calls (max unlimited) in 'ringall' strategy (9s holdtime, 156s talktime), W:0, C:4, A:6, SL:0.0% within 0s Members: SIP/155@callcentric (ringinuse disabled) (Not in use) has taken 4 calls (last was 1219067 secs ago) SIP/FF4C119EEBF8-SLS (ringinuse disabled) (Not in use) has taken no calls yet SIP/FF1565AABB2D-SLS (ringinuse disabled) (Invalid) has taken no calls yet SIP/FF9EF375CCFC-SLS (ringinuse disabled) (Not in use) has taken no calls yet No Callers [Sales](StandardQueue) announce = first member => SIP/FF4C119EEBF8-SLS member => SIP/FF9EF375CCFC-SLS member => SIP/1314555@callcentric ;Eric's cell member => SIP/FF1565AABB2D-SLS ;Eric's Yealink -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue not dialing out to cell phone for some reason
John,
This is output of command below. How do I enable and log queue events?The
1555@callcentric is the one I'm curious about. I just tried calling into
"sales" again and it didn't change this "last was 1219067" output
Sales has 0 calls (max unlimited) in 'ringall' strategy (9s holdtime, 156s
talktime), W:0, C:4, A:6, SL:0.0% within 0s
Members:
SIP/155@callcentric (ringinuse disabled) (Not in use) has taken 4
calls (last was 1219067 secs ago)
SIP/FF4C119EEBF8-SLS (ringinuse disabled) (Not in use) has taken no calls
yet
SIP/FF1565AABB2D-SLS (ringinuse disabled) (Invalid) has taken no calls yet
SIP/FF9EF375CCFC-SLS (ringinuse disabled) (Not in use) has taken no calls
yet
No Callers
From: John Kiniston
To: idemkovi...@yahoo.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Sent: Thursday, November 15, 2018 2:21 PM
Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some
reason
what does the output of 'queue show sales' show?
Do you have queue logging enabled? Have you looked in the queue log to see what
events are firing?
On Thu, Nov 15, 2018 at 9:55 AM Ivan Demkovitch wrote:
Hello,
I have queues.conf setup with a group like so:
[Sales](StandardQueue)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/1314555@callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink
So, my idea here that it should ring all 4 phones at the same time. And it does
work but randomly.I did trace a call and this is what I see. Only 2 phones
(internal) called. External SIP@callcentric is not being called.
Any idea why it's not being called?
-- Executing [1@automated_attendant_normal:1]
Verbose("SIP/callcentric15-0435", "1, Caller "DEMKOVITCH,IVAN"
<13144880983> has entered the sales queue") in new stack
Caller "aa" <155> has entered the sales queue
-- Executing [1@automated_attendant_normal:2]
Goto("SIP/callcentric15-0435", "queues,7001,1") in new stack
-- Goto (queues,7001,1)
-- Executing [7001@queues:1] Verbose("SIP/callcentric15-0435", "2,"aa"
<155> entering sales queue") in new stack
== "aa" <155> entering sales queue
-- Executing [7001@queues:2] BackGround("SIP/callcentric15-0435",
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
-- Playing
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')
-- Executing [7001@queues:3] Queue("SIP/callcentric15-0435",
"sales85") in new stack
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-0435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0437 is ringing
-- SIP/FF9EF375CCFC-SLS-0436 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
-- Playing 'queue-periodic-announce.ulaw'
(language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-0435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0439 is ringing
-- SIP/FF9EF375CCFC-SLS-0438 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
--
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Re: [asterisk-users] Queue not dialing out to cell phone for some reason
Sebastian,
I don't think it has to do anything with registration. It is dialing through
the SIP trunk, so it goes out as normal cell phone call.Also, why I don't see
anything in a log? I see only first 2 members being dialed.
From: Sebastian Nielsen
To: 'Ivan Demkovitch' ; 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Thursday, November 15, 2018 10:58 AM
Subject: SV: [asterisk-users] Queue not dialing out to cell phone for some
reason
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div.yiv7898733751WordSection1 {}#yiv7898733751 I would suspect that the cell
phone does use battery saving causing the SIP application to lose registration
with the server. Would also suggest using TCP with a fairly short keepalive to
prevent the cellular network from tearing down the connection to the asterisk
server.You need to go into android settings and make sure the SIP client is
whitelisted in battery management. Från: asterisk-users
För Ivan Demkovitch
Skickat: den 15 november 2018 17:55
Till: asterisk-users@lists.digium.com
Ämne: [asterisk-users] Queue not dialing out to cell phone for some reason
Hello, I have queues.conf setup with a group like so: [Sales](StandardQueue)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/1314555@callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink So, my idea here that it should
ring all 4 phones at the same time. And it does work but randomly.I did trace a
call and this is what I see. Only 2 phones (internal) called. External
SIP@callcentric is not being called. Any idea why it's not being called?
-- Executing [1@automated_attendant_normal:1]
Verbose("SIP/callcentric15-0435", "1, Caller "DEMKOVITCH,IVAN"
<13144880983> has entered the sales queue") in new stack
Caller "aa" <155> has entered the sales queue
-- Executing [1@automated_attendant_normal:2]
Goto("SIP/callcentric15-0435", "queues,7001,1") in new stack
-- Goto (queues,7001,1)
-- Executing [7001@queues:1] Verbose("SIP/callcentric15-0435", "2,"aa"
<155> entering sales queue") in new stack
== "aa" <155> entering sales queue
-- Executing [7001@queues:2] BackGround("SIP/callcentric15-0435",
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
-- Playing
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')
-- Executing [7001@queues:3] Queue("SIP/callcentric15-0435",
"sales85") in new stack
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-0435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0437 is ringing
-- SIP/FF9EF375CCFC-SLS-0436 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
-- Playing 'queue-periodic-announce.ulaw'
(language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-0435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0439 is ringing
-- SIP/FF9EF375CCFC-SLS-0438 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
-- Playing 'queue-periodic-announce.ulaw'
(language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-0435'
== Using S
[asterisk-users] Queue not dialing out to cell phone for some reason
Hello,
I have queues.conf setup with a group like so:
[Sales](StandardQueue)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/1314555@callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink
So, my idea here that it should ring all 4 phones at the same time. And it does
work but randomly.I did trace a call and this is what I see. Only 2 phones
(internal) called. External SIP@callcentric is not being called.
Any idea why it's not being called?
-- Executing [1@automated_attendant_normal:1]
Verbose("SIP/callcentric15-0435", "1, Caller "DEMKOVITCH,IVAN"
<13144880983> has entered the sales queue") in new stack
Caller "aa" <155> has entered the sales queue
-- Executing [1@automated_attendant_normal:2]
Goto("SIP/callcentric15-0435", "queues,7001,1") in new stack
-- Goto (queues,7001,1)
-- Executing [7001@queues:1] Verbose("SIP/callcentric15-0435", "2,"aa"
<155> entering sales queue") in new stack
== "aa" <155> entering sales queue
-- Executing [7001@queues:2] BackGround("SIP/callcentric15-0435",
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
-- Playing
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')
-- Executing [7001@queues:3] Queue("SIP/callcentric15-0435",
"sales85") in new stack
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-0435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0437 is ringing
-- SIP/FF9EF375CCFC-SLS-0436 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
-- Playing 'queue-periodic-announce.ulaw'
(language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-0435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0439 is ringing
-- SIP/FF9EF375CCFC-SLS-0438 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
-- Playing 'queue-periodic-announce.ulaw'
(language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-0435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-043b is ringing
-- SIP/FF9EF375CCFC-SLS-043a is ringing
-- Stopped music on hold on SIP/callcentric15-0435
== Spawn extension (queues, 7001, 3) exited non-zero on
'SIP/callcentric15-0435'
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Check out the new Asterisk community forum at: https://community.asterisk.org/
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https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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Re: [asterisk-users] Is there any way to pass caller id to
Sorry guys! Here is what Callcentric tech support provided. They asked me to
add 2 settings to SIP.conf:sendrpid=pai
trustrpid=no
And modify incoming context like so:
[from-pstn-toheader-inreplyto]
exten => s,1,Noop(Trying to add ${SIPCALLID} to the In-Reply-To Header)
exten => s,2,SIPAddHeader(In-Reply-To: ${SIPCALLID})
exten => s,3,Goto(automated_attendant,s,1)
Basically it's all about adding header they support on their end. I do not know
if they had to make any config changes on their side or not but seems like it's
a supported and legit feature.Now I need to figure out how to figure out when
it's a call from office :)))
Thank you,Ivan
Message: 2
Date: Mon, 15 Oct 2018 23:39:31 +0200
From: Daniel Tryba
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Is there any way to pass caller id to
cell phone?
Message-ID: <20181015213930.2a4uulq2z6xbfjcb@bogus>
Content-Type: text/plain; charset=us-ascii
On Thu, Oct 11, 2018 at 05:18:24PM +, Ivan Demkovitch wrote:
> Where problem comes in - if person not at the desk - his cell phone shows
> call from OFFICE number and there is no way to tell who is really calling.
> We use Callcentric as a trunk if it makes any difference.
> I'd like to add info about caller when passing to cell phone if possible. Is
> there any way to do that?
Maybe you should ask them how to do this! Maybe you should add a
Diversion header, maybe they don't allow this kind of spoofing at all.
This is a common request from users of SIP trunks and your use case is
legit. If Callcentric does checks on callerid validity and there is a
call to a customer with callerid X, they should be able to use this
callerid X when forwarding to an external device/number.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Astricon is coming up October 9-11! Signup is available at:
https://www.asterisk.org/community/astricon-user-conference
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Is there any way to pass caller id to
Thanks all, I did contact Callcentric about it and their tech support helped meget those headers established. They even helped to troubleshoot Asterisk dialplan. A the end all works as it should. Thank you,Ivan Message: 2 Date: Mon, 15 Oct 2018 23:39:31 +0200 From: Daniel Tryba To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is there any way to pass caller id to cell phone? Message-ID: <20181015213930.2a4uulq2z6xbfjcb@bogus> Content-Type: text/plain; charset=us-ascii On Thu, Oct 11, 2018 at 05:18:24PM +, Ivan Demkovitch wrote: > Where problem comes in - if person not at the desk - his cell phone shows > call from OFFICE number and there is no way to tell who is really calling. > We use Callcentric as a trunk if it makes any difference. > I'd like to add info about caller when passing to cell phone if possible. Is > there any way to do that? Maybe you should ask them how to do this! Maybe you should add a Diversion header, maybe they don't allow this kind of spoofing at all. This is a common request from users of SIP trunks and your use case is legit. If Callcentric does checks on callerid validity and there is a call to a customer with callerid X, they should be able to use this callerid X when forwarding to an external device/number. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any way to pass caller id to cell phone?
Abdul,
Added code like you proposed, I see it in logs but still don't see caller ID
coming in:
-- Goto (internal,101,1)
-- Executing [101@internal:1] NoOp("SIP/callcentric13-06d1", "Call ID:
"DEMKOVITCH,IVAN" <155>") in new stack
-- Executing [101@internal:2] Dial("SIP/callcentric13-06d1",
"SIP/649EF375,30") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/649EF375
-- SIP/649EF375-06d2 is ringing
-- Nobody picked up in 3 ms
-- Executing [101@internal:3] Dial("SIP/callcentric13-06d1",
"SIP/166@callcentric,20") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/166@callcentric
-- SIP/callcentric-06d3 is making progress passing it to
SIP/callcentric13-06d1
> 0x7f3b3801c800 -- Probation passed - setting RTP source address to
222.11.192.162:56680
-- SIP/callcentric-06d3 is ringing
-- SIP/callcentric-06d3 is making progress passing it to
SIP/callcentric13-06d1
-- SIP/callcentric-06d3 is ringing
-- SIP/callcentric-06d3 is making progress passing it to
SIP/callcentric13-06d1
-- SIP/callcentric-06d3 is ringing
-- SIP/callcentric-06d3 is making progress passing it to
SIP/callcentric13-06d1
-- SIP/callcentric-06d3 is ringing
-- SIP/callcentric-06d3 is making progress passing it to
SIP/callcentric13-06d1
== Spawn extension (internal, 101, 3) exited non-zero on
'SIP/callcentric13-06d1'
From: Abdul Basit
To: idemkovi...@yahoo.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Sent: Thursday, October 11, 2018 12:42 PM
Subject: Re: [asterisk-users] Is there any way to pass caller id to cell phone?
Hi Ivan,
Check whats CallerID you are getting before initiating dial command.
;Eric on extension 105
exten => 105,1,NoOp( Call ID: ${CALLERID(all)} )
exten => 105,n,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
same => n,VoiceMail(105@default,u)
Also what Caller ID is set on outgoing trunk? Is that enforced in trunk
configuration?
--
regards,
abdul basit
On Thu, 11 Oct 2018 at 22:19, Ivan Demkovitch wrote:
We have following problem. On some of the extentions I call cell phone after 10
seconds or so.Or, like this one below- we call cell and office phone at the
same time
;Eric on extension 105
exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
same => n,VoiceMail(105@default,u)
Where problem comes in - if person not at the desk - his cell phone shows call
from OFFICE number and there is no way to tell who is really calling.
We use Callcentric as a trunk if it makes any difference.
I'd like to add info about caller when passing to cell phone if possible. Is
there any way to do that?
Thank you,Ivan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Astricon is coming up October 9-11! Signup is available at:
https://www.asterisk.org/community/astricon-user-conference
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Astricon is coming up October 9-11! Signup is available at:
https://www.asterisk.org/community/astricon-user-conference
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any way to pass caller id to cell phone?
Abdul,
What is the point of chacking caller id? Does it "make" Asterisk forward it
later on?
Here is outgoing trunk sip.config. Don't know much about it, pretty much
copy/paste from their instructions.
register => dsfsadfa...@callcentric.com[callcentric]
type=peer
context=from-callcentric
host=callcentric.com
fromdomain=callcentric.com
defaultuser=1777555
fromuser=1777555
secret=abcd1234
insecure=port,invite
disallowed_methods=UPDATE
directmedia=no
videosupport=no
disallow=all
allow=ulaw[callcentric1](callcentric);
host=alpha1.callcentric.com[callcentric2](callcentric);
host=alpha2.callcentric.com[callcentric3](callcentric);
host=alpha3.callcentric.com[callcentric4](callcentric);
host=alpha4.callcentric.com[callcentric5](callcentric);
host=alpha5.callcentric.com
Thank you,Ivan
From: Abdul Basit
To: idemkovi...@yahoo.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Sent: Thursday, October 11, 2018 12:42 PM
Subject: Re: [asterisk-users] Is there any way to pass caller id to cell phone?
Hi Ivan,
Check whats CallerID you are getting before initiating dial command.
;Eric on extension 105
exten => 105,1,NoOp( Call ID: ${CALLERID(all)} )
exten => 105,n,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
same => n,VoiceMail(105@default,u)
Also what Caller ID is set on outgoing trunk? Is that enforced in trunk
configuration?
--
regards,
abdul basit
On Thu, 11 Oct 2018 at 22:19, Ivan Demkovitch wrote:
We have following problem. On some of the extentions I call cell phone after 10
seconds or so.Or, like this one below- we call cell and office phone at the
same time
;Eric on extension 105
exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
same => n,VoiceMail(105@default,u)
Where problem comes in - if person not at the desk - his cell phone shows call
from OFFICE number and there is no way to tell who is really calling.
We use Callcentric as a trunk if it makes any difference.
I'd like to add info about caller when passing to cell phone if possible. Is
there any way to do that?
Thank you,Ivan
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[asterisk-users] Is there any way to pass caller id to cell phone?
We have following problem. On some of the extentions I call cell phone after 10
seconds or so.Or, like this one below- we call cell and office phone at the
same time
;Eric on extension 105
exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
same => n,VoiceMail(105@default,u)
Where problem comes in - if person not at the desk - his cell phone shows call
from OFFICE number and there is no way to tell who is really calling.
We use Callcentric as a trunk if it makes any difference.
I'd like to add info about caller when passing to cell phone if possible. Is
there any way to do that?
Thank you,Ivan
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[asterisk-users] Asterisk inside network. What phone works well?
Hello list, I have Asterisk running well inside our network. I did some experiments exposing it to internet but had some issues: 1. NAT issues (voice one way, etc). From what I understand double-NAT users will always have something like this 2. Immediately I see people trying to hack into. I did configure Fail2Ban and it works somewhat, but not 100%. Erroneous logs, etc So.. I ended up closing network. Currently most users inside network. My home router have GRE tunnel to office so phone works just fine. Another user uses VPN and soft phone and it works good too. Now I need to setup some users with actual phone devices and none of those solutions will work. So, I did some research and found that some phones have VPN capability built in. Right now I use Cisco SPA504G phones. We have auto-provisioning for them, works well. But I don’t think they have VPN capability. What I found it that Cisco 525g2 has AnyConnect functionality (SSL VPN) but not sure if this is what I need. We have Mikrotik router. Can I setup VPN on router and have this Cisco phone auto-dial VPN and then connect to Asterisk? I’m asking to see if this will work before I go in and buy that phone. Or maybe there is other devices/solutions you suggest? I’d like to stay with Cisco because I’m somewhat familiar with provisioning those.. Thank you Ivan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting better Caller ID
Hello, We use Asterisk and as per book we use MAC addresses as user names. So, when call coming in from outside (SIP trunk) - caller id is good. But when users calling each other on extensions - they see MAC addresses. How would I make it so we see actual names instead of MAC addresses? Without changing users.. Thank you, Ivan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] First call - one way audio
Hello folks, I hope this is simple issue because it seems like something with registration expiration, etc. We use Asterisk (plain setup) with Cisco SPA phones (also nothing changed on settings, just proxy/UN/Password Everything on same LAN So, what we observe is when call coming in - we can’t hear opponent but they can hear us. And this happens only when phone idle for some time. Second call coming in - all works fine. Any idea what might be causing this? Hopefully this is common issue. Thank you, Ivan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Please help me understand lines and extensions little better
Hello, Probably obvious to most of you but I’m still learning VOIP concept(and telephony inside organization in general) I have bunch of unrelated questions on my mind :) - I picked SPA504G phones as “phone to go” for our small company. They are cheap and available. Any pros/cons you can come up with against this device? I setup direct extensions in Asterisk, they work great. But under conference settings - I have to set which SIP devices participate. It just sends calls to those devices and they come in as main extension. It’s not super-convenient. Main problem is - I want to know if user came from SALES/SUPPORT or DIRECT line. -Now I’m getting more into phone setup little more and I think I’m not doing it right. Current on a phone I setup only 1 extensions with SIP ID/Password and then I assign this extension 1 to all 4 “Line keys” I understand this is how I can switch between calls. Let’s say one call comes in - I can be on call and next call comes in - I can press “Line 2” button and pickup. Correct? So, now I got idea about Line key 3 & 4. If my main SIP ID XXX I can set 2 more id’s: XXX_SLS XXX_SPT Then I can register them under ext 3/4 on a phone and set line keys to ext 3/4 Of course, I will add XXX_SLS and XXX_SPT under appropriate queues.conf This way I will setup more extensions but users will have much better visibility of who’s calling. Is this correct/common approach? Just trying to understand what is the best way to setup phones/extensions.. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help with my dial plan - general logic flaws
Hello group,
I’m developer myself but creating dial plans is little bit different I guess :)
I created very simple automated attendand (with a help of book), below is code.
But logic is simple:
Depending on time - I want:
If during business hours - give them menu and handle extensions
If after hours - give them message and take to voicemail.
This dial plan accomplishes just that.
What I don’t like is side behavior of this plan
When after hours - I can still press 1/2/3 and go into Sales/Support
queue or to operator, etc.
It’s probably OK to allow dialing of extensions of actual users (101, 102, etc)
But I don’t want them to dial operator (0) or 1,2,3 and such.
How do I fix it?
Thank you!
Ivan
[automated_attendant]
exten => fax,1,Goto(fax_incoming,fax,1)
exten => s,1,Verbose(1, Caller ${CALLERID(all)} has entered the auto attendant)
same => n,Answer()
same => n,Set(TIMEOUT(digit)=2) ;this sets the inter-digit timer
same => n,Wait(1) ;wait 1 sec to establish audio
same =>
n(menustart),GotoIfTime(9:00-17:00,mon-fri,*,*?daygreeting:afterhoursgreeting)
;depending if it's work time or not go to labels
same =>
n(afterhoursgreeting),Background(/etc/asterisk/automated-attendant-prompts/ditat_afterhour_greeting)
; after hours greeting
same => n,VoiceMail(99@default,u)
same => n,Hangup()
same =>
n(daygreeting),Background(/etc/asterisk/automated-attendant-prompts/ditat_main_greeting)
; day greeting
same => n,WaitExten(4) ; wait 4
sec max before give up
same => n,Goto(0,1) ; treat
as caller pressed '0'
exten => 1,1,Verbose(1, Caller ${CALLERID(all)} has entered the sales queue)
same => n,Goto(Queues,7001,1) ; Sales
Queue
exten => 2,1,Verbose(1, Caller ${CALLERID(all)} has entered the support queue)
same => n,Goto(Queues,7002,1) ;
Support Queue
exten => 3,1,Verbose(1, Caller ${CALLERID(all)} has entered the support queue
via Other extension)
same => n,Goto(Queues,7002,1) ;
Support Queue
exten => 0,1,Verbose(1, Caller ${CALLERID(all)} wants an operator)
same => n,Goto(internal,101,1)
exten => i,1,Verbose(1, Caller ${CALLERID(all)} has entered an invalid
selection)
same => n,Playback(invalid)
same => n,Goto(s,menustart)
exten => t,1,Verbose(1, Caller ${CALLERID(all)} has timed out)
same => n,Goto(0,1)
; -- CALLING EXTENSIONS DIRECTLY -
exten => _1XX,1,Verbose(1,Call to an extension starting with '1')
same => n,Goto(internal,${EXTEN},1)
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[asterisk-users] Receiving faxes with spandsp question
Hello!
I’m trying to add fax functionality to my asterisk installation. Right now I’m
focusing on receiving faxes. This is not explained in a book, but I assume that
I can use same context, add “fax” extension and if someone calls to send fax -
it will autodetect. Right?
Per book, I made following setup additions:
1. In sip.conf [general] I added:
;FAX stuff
faxdetect=yes
t38pt_udptl=yes
2. In extensions.conf I hade something like this:
[from-callcentric]
exten = s,1,Goto(automated_attendant,s,1)
; FAX handling stuff AS IN BOOK
exten = fax,1,Verbose(3,Incoming Fax)
same = n,Set(FAXDEST=/tmp) ; folder where faxes will be stored
same = n,Set(tempfax=${STRFTIME(,,%C%y%m%d%H%M)})
same = n,ReceiveFax(${FAXDEST}/${tempfax}.tif)
same = n,Verbose(3, - Fax receipt completed with status: ${FAXSTATUS})
Well, that didn’t work. Trying to send fax - it was going to my autoattendant
and never triggered fax. So, I made a change like so:
3. Changed extensions.conf
[from-callcentric]
; FAX handling stuff AS IN BOOK
exten = s,1,Verbose(3,Incoming Fax)
same = n,Set(FAXDEST=/tmp) ; folder where faxes will be stored
same = n,Set(tempfax=${STRFTIME(,,%C%y%m%d%H%M)})
same = n,ReceiveFax(${FAXDEST}/${tempfax}.tif)
same = n,Verbose(3, - Fax receipt completed with status: ${FAXSTATUS})
I just made it fax handling context, and I got FAX :) But, while fax was
received I was getting following:
[2015-06-24 23:40:28] WARNING[47369][C-0005]: res_fax_spandsp.c:438
spandsp_log: WARNING T.30 ECM carrier not found
QUESTIONS:
1. Should I do something about this warning?
2. How do I receive fax and have main entry to auto attendant in a same
context? Can I have it on same puplic phone number?
Thanks!
Ivan
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Re: [asterisk-users] Calling multiple phones at once
Hi again!
Also, given my setup below, how do I send caller id to my cell?
SIP/83@callcentric is my cell, when I get incoming call when someone dials into
Asterisk - I just see public calcentric’s DID number.
I want to send a number of who CALLED IN into the Asterisk and possibly add
couple numbers upfront or something like this to signal me that this call comes
through the PBX and not directly to my cell?
Thank you!
On Jun 19, 2015, at 2:30 PM, Ivan Demkovitch idemkovi...@yahoo.com wrote:
Hello All!
I asked week a so ago about how to call multiple phones alltogether (home,
office, cell)
Dial app looks simple, this is kind of what I have now:
-
[globals]
IVAN_HOME_OFFICE=SIP/BF8
IVAN_OFFICE=SIP/CFC
IVAN_CELL=SIP/83@callcentric
[internal]
exten = 101,1,Dial(${IVAN_HOME_OFFICE}${IVAN_OFFICE}${IVAN_CELL},60)
same = n,VoiceMail(101@default,u)
——
Now, I have basic automated attendant and I have Queues setup
———
[general]
autofill=yes
shared_lastcall=yes
[StandardQueue](!)
musicclass=default
joinempty=yes
leavewhenempty=yes
ringinuse=no
[Sales](StandardQueue)
member = SIP/BF8
member = SIP/CFC
[Support](StandardQueue)
member = SIP/BF8
member = SIP/CFC
——
Now this setup becomes less than ideal. I’d like to create “bundles” of
extensions/numbers and just use them everywhere throughout the system.
How do I do that?
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[asterisk-users] Calling multiple phones at ones
Hello group! I’m new to Asterisk but got one running finally :) Now I’m trying to solve following problem. I have company Automated Attendant and each employee have SIP phone at home, SIP phone in office, cell phone. I want all those 3 phones to be “one”. So, if someone calls our company number and dials my extension - I’d like 3 phones to ring at the same time. What is this feature and where should I look for samples, etc? I’m going by “Asterisk: The definite guide” book and pretty confident with those concepts described but not sure how to achieve what I described above. Thank you, Ivan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to connect to remote asterisk
Hello list! I’m working on a fresh Asterisk install over CentOS7 base. I’m using “Asterisk. The Definite guide” book as a reference. I connect and work using SSH Problem I have - I can’t connect to asterisk from remote. Getting error: $ sudo asterisk -rvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Yes, it exist, and service runs: [asteriskpbx@localhost asterisk]$ service --status-all asterisk.service - LSB: Asterisk PBX Loaded: loaded (/etc/rc.d/init.d/asterisk) Active: active (running) since Sun 2015-03-22 21:04:22 CDT; 23min ago Process: 1615 ExecStart=/etc/rc.d/init.d/asterisk start (code=exited, status=0/SUCCESS) Main PID: 2931 CGroup: /system.slice/asterisk.service ├─ 2924 /bin/sh /usr/sbin/safe_asterisk └─88248 sleep 4 If I start using -c - there is some warnings, but I don’t think it’s a problem? Running as user 'asteriskpbx' Running under group 'asteriskpbx' [ Initializing Custom Configuration Options ] XSLT support not found. XML documentation may be incomplete. CDR simple logging enabled. 178 modules will be loaded. No configured users for ARI Error loading module 'res_ari_mailboxes.so': /usr/lib64/asterisk/modules/res_ari_mailboxes.so: undefined symbol: stasis_app_mailbox_to_json Module 'res_ari_mailboxes.so' could not be loaded. !!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the global setting can make !!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users !!! will be sent to a different port than replies for an existing peer/user. If at all possible, !!! use the global 'nat' setting and do not set 'nat' per peer/user. !!! (config category='0001' global force_rport='No' peer/user force_rport='Yes') !!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the global setting can make !!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users !!! will be sent to a different port than replies for an existing peer/user. If at all possible, !!! use the global 'nat' setting and do not set 'nat' per peer/user. !!! (config category='0002' global force_rport='No' peer/user force_rport='Yes') SIP channel loading... No IAX provisioning configuration found, IAX provisioning disabled. Adding default_menu menu to app_confbridge Failed to load configuration file. Module not activated. No mappings found in cel_custom.conf. Not logging CEL to custom CSVs. Failed to load configuration file. *CLI Asterisk Ready. Another thing, everywhere in doc’s and references I see that I can use “exit” on CLI. If I try to type “exit” - it says there is no such command. Please pardon me, I’m new to both Linux and Asterisk, trying to learn from beginning.. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to connect to remote asterisk
Please disregard this post! I found issue, wonder if there is more elegant solution. I had to disable SELinux. Is it good idea? On Mar 22, 2015, at 9:56 PM, Ivan Demkovitch idemkovi...@yahoo.com wrote: Hello list! I’m working on a fresh Asterisk install over CentOS7 base. I’m using “Asterisk. The Definite guide” book as a reference. I connect and work using SSH Problem I have - I can’t connect to asterisk from remote. Getting error: $ sudo asterisk -rvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Yes, it exist, and service runs: [asteriskpbx@localhost asterisk]$ service --status-all asterisk.service - LSB: Asterisk PBX Loaded: loaded (/etc/rc.d/init.d/asterisk) Active: active (running) since Sun 2015-03-22 21:04:22 CDT; 23min ago Process: 1615 ExecStart=/etc/rc.d/init.d/asterisk start (code=exited, status=0/SUCCESS) Main PID: 2931 CGroup: /system.slice/asterisk.service ├─ 2924 /bin/sh /usr/sbin/safe_asterisk └─88248 sleep 4 If I start using -c - there is some warnings, but I don’t think it’s a problem? Running as user 'asteriskpbx' Running under group 'asteriskpbx' [ Initializing Custom Configuration Options ] XSLT support not found. XML documentation may be incomplete. CDR simple logging enabled. 178 modules will be loaded. No configured users for ARI Error loading module 'res_ari_mailboxes.so': /usr/lib64/asterisk/modules/res_ari_mailboxes.so: undefined symbol: stasis_app_mailbox_to_json Module 'res_ari_mailboxes.so' could not be loaded. !!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the global setting can make !!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users !!! will be sent to a different port than replies for an existing peer/user. If at all possible, !!! use the global 'nat' setting and do not set 'nat' per peer/user. !!! (config category='0001' global force_rport='No' peer/user force_rport='Yes') !!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the global setting can make !!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users !!! will be sent to a different port than replies for an existing peer/user. If at all possible, !!! use the global 'nat' setting and do not set 'nat' per peer/user. !!! (config category='0002' global force_rport='No' peer/user force_rport='Yes') SIP channel loading... No IAX provisioning configuration found, IAX provisioning disabled. Adding default_menu menu to app_confbridge Failed to load configuration file. Module not activated. No mappings found in cel_custom.conf. Not logging CEL to custom CSVs. Failed to load configuration file. *CLI Asterisk Ready. Another thing, everywhere in doc’s and references I see that I can use “exit” on CLI. If I try to type “exit” - it says there is no such command. Please pardon me, I’m new to both Linux and Asterisk, trying to learn from beginning.. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
