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OUTbound Caller ID lookup HTTP query

July 29, 2015, 11:55 pm
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@ambiorixg12:

I just wanted to thank you for the excellent information! cURL was exactly what I was looking for and I'm so glad that function existed. It was even built into CentOS, which was awesome.

I didn't specify it, but I don't need the cURL function to retrieve any information, I just need it to load a URL and then ignore the result. (Because simply loading the URL will add the information to my CRM system). But the other issue I'm not sure of is if I'll be able to insert a variable for "the phone number being dialed".

My questions:
What do I use to reference the variable of the number we are dialing out?
If I'm running Elastix 2.4 on CentOS, would you recommend your first or second option?
Which file do I edit to modify the master outbound dial plan?
Why did you specify *764 as the extension / feature code?

Statistics : Posted by pinellascompute • on Wed Jul 29, 2015 11:55 pm • Replies 18 • Views 3757
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OUTbound Caller ID lookup HTTP query

July 29, 2015, 11:55 pm
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exten => *764,1,Verbose(2, Run CURL to get IP address from ifconfig.me)
same => n,Answer()
same => n,Set(MyIPAddressIs=${CURL(http://ifconfig.me/)})
same => n,SayAlpha(${MyIPAddressIs})
same => n,Hangup()



Also you can use the Linux curl command


exten =>_X.,1,System(curl https://www.edcmeals.com/voip/confirmation.tpl -G -d"ordernumber=${order}&response=4")

Statistics : Posted by pinellascompute • on Wed Jul 29, 2015 11:55 pm • Replies 18 • Views 3757
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OUTbound Caller ID lookup HTTP query

July 29, 2015, 11:55 pm
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Thanks! I'll look into that function. Do you have an examples of having used this?

Statistics : Posted by pinellascompute • on Wed Jul 29, 2015 11:55 pm • Replies 18 • Views 3757
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OUTbound Caller ID lookup HTTP query

July 29, 2015, 11:55 pm
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Use curl() function in your outbound dialplan before dial()

Statistics : Posted by pinellascompute • on Wed Jul 29, 2015 11:55 pm • Replies 18 • Views 3757
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OUTbound Caller ID lookup HTTP query

July 29, 2015, 11:55 pm
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Hi there everyone,

This is only my second post on here; last time you guys were super helpful. Long story short is I'm trying to get our Asterisk System to integrate with our web app CRM system for outbound calls we make. I'm not sure if what I'm looking for exists, but I figured that someone could at least point me in the right direction. Here's the details of what I'm trying to accomplish:

Our web app CRM system currently provides an HTTP caller ID look up string. Our PBX seamlessly uses this for a caller ID look up source; which pulls the "customer name" from our CRM database, to provide a caller ID for incoming phone calls. Additionally, when that caller ID look up source is queried, it makes a note in the customers account that they called in (it doesn't detect the call, it just assumes they called because the web query was accessed).

What I'm trying to accomplish is exactly the opposite. I'd like to run the same type of web query when we making outbound phone call. I'd like our PBX to run a simple HTTP command or parse a URL upon making outbound phone call. We would then be able to use a similar caller ID web query to detect when we make outbound phone calls to customers; therein automatically noting in our CRM software that we called.

This must be something that either Asterisk or a third-party plug-in can do, right? Any insight or direction would be greatly appreciated. Cheers!

Statistics : Posted by pinellascompute • on Wed Jul 29, 2015 11:55 pm • Replies 18 • Views 3757
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Incoming call does not work Analog Trunk

January 18, 2016, 1:53 pm
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Hello All,

This is my work...
i am running Aterisk 11.20. on Ubuntu server,

i 've installed a card for outside connectivity such as PSTN line, all users can make outgoing call by prefixed the number with 9..

;General options
usecallerid = yes
hidecallerid = no
callwaiting = yes
threewaycaing = yes
transfer = yes
echocancel = yes
echocancelwhenbridged = yes
rxgain = 0.0
txgain = 0.0
FXO Modules
group = 1
echocancel = yes
signalling = fxs_ks
context = voip
channel = 1

and extensions.conf : users are able to make outgoing call with this following line>>

[voip]
exten => _9.,1,Dial(DAHDI/g1/www${EXTEN : 1})
exten => _9.,2,Congestion

This one is for internal call between users >>

[voip]
exten => _1XX,1,Dial(SIP/${EXTEN},20)
exten => _1XX,2,Hangup()

So now i passed this following command line to route incoming call to extension 100

[voip]
exten => s,1,Answer
exten => s,2,Dial(DAHDI/g1/20,rt)
exten => s,3,Dial(SIP/100)

i got this following erro :
Channel 'DAHDI/1-1' sent to invalid extension but no invalid handler :

please help me to fix this,

Thanks !

Statistics : Posted by tella • on Mon Jan 18, 2016 3:53 pm • Replies 3 • Views 95
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DTMF logging while transferring incoming call to outbound

January 18, 2016, 8:09 am
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UPDATE: Just tested and found that the digits that IVR accepts are not logged via features and other digits are logged fine Image

Here is typical setup... Caller calls our number (111 for example) provided by voip provider and we redirect it to IVR server (222 for example).

Image
http://prntscr.com/9s0ehi

Suppose IVR has 3 menu options ( press 1 for blah, press 2 for blah, press 3 for blah) so when we press either 1,2 or 3 then dtmf are passed to IVR and IVR moves forward but these dtmf are not detected by features application map which I setup. All other digits will be captured by features fine.

I think Local channel scenario will also have this problem???
I think I need sometype of event logger/sniffer setup like chanspy. Any idea how to implement it ?

Statistics : Posted by nasirjavaid • on Mon Jan 18, 2016 10:09 am • Replies 5 • Views 140
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DTMF logging while transferring incoming call to outbound

January 18, 2016, 8:09 am
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You are right but as its needed by the client so I have to opt it... Anyways I have managed to test it using dynamic features and calling AGI script which logs digit in text file.

But I see some notices whenever feature is activated as below

-- Feature Found: log8 exten: log8
-- Launched AGI Script /var/lib/asterisk/agi-bin/logdtmf.agi
[Jan 19 15:19:56] NOTICE[28239]: mtp.c:1993 mtp_thread_main: Full dahdi input buffer detected, incoming packets may have been lost on link 'l1' (count=64.
[Jan 19 15:19:56] NOTICE[28239]: mtp.c:1993 mtp_thread_main: Full dahdi input buffer detected, incoming packets may have been lost on link 'l3' (count=64.
[Jan 19 15:19:56] NOTICE[28239]: mtp.c:2063 mtp_thread_main: Empty Dahdi output buffer detected, outgoing packets may have been lost on link 'l1'.
[Jan 19 15:19:56] NOTICE[28239]: mtp.c:2063 mtp_thread_main: Empty Dahdi output buffer detected, outgoing packets may have been lost on link 'l3'.
-- <SIP/smart2-0000001c>AGI Script logdtmf.agi completed, returning 0


I think I can also accomplish it using asterisk function FILE so can you suggest which approach is better? I think FILE is fast as its native?

I am using asterisk-1.8.25.0 and dahdi 2.9.0

the destination number is running IVR so is it possible that these digits will also reach to the ivr or only are limited to features?

Cheers!

Statistics : Posted by nasirjavaid • on Mon Jan 18, 2016 10:09 am • Replies 5 • Views 140
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DTMF logging while transferring incoming call to outbound

January 18, 2016, 8:09 am
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It's an abuse of the logging mechanism, which is there to help debug Asterisk's own handling of DTMF, not to record DTMF in transit, so I doubt that many, if any, other people have tried this.

Statistics : Posted by nasirjavaid • on Mon Jan 18, 2016 10:09 am • Replies 5 • Views 140
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DTMF logging while transferring incoming call to outbound

January 18, 2016, 8:09 am
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Thanks a lot dave for giving timely clue. I am going to test this scenario now. If there are any examples you can share then I will be obliged as I have to finish this by tonight Image

Cheers!

Statistics : Posted by nasirjavaid • on Mon Jan 18, 2016 10:09 am • Replies 5 • Views 140
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DTMF logging while transferring incoming call to outbound

January 18, 2016, 8:09 am
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Introduce a non-optimising local channel, to break the native bridge, or you could try defining a feature code with a, b, c. or d digits, and enable that feature in the dial application

Statistics : Posted by nasirjavaid • on Mon Jan 18, 2016 10:09 am • Replies 5 • Views 140
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DTMF logging while transferring incoming call to outbound

January 18, 2016, 8:09 am
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Hi,

On normal asterisk server I am able to log the dtmf dialled by caller successfully by enabling dtmf=>dtmf in logger.conf

Now scenario is that I have configured a voip trunk in my server and I am simply redirecting incoming call to another number where an IVR is running. I want to capture the dtmf digits pressed by callers.

Here is sip.conf

[general]
register => xxxxx:xxxxxxxx@sip.voipprovider.com/xxxxxxx

[voipprovider]
type=friend
secret=xxxxxx
username=xxxxx
fromuser=xxxxx
fromdomain=sip.voipprovider.com
host=sip.voipprovider.com
insecure=invite
directmedia=no
dtmfmode= rfc2833
context=fromvoipprovider

in extensions.conf

[fromvoipprovider]
exten => _.,1,Answer
exten => _.,n,Wait(1)
exten => _.,n,Dial(SIP/voipprovider/${EXTEN})

in logger.conf

console => notice,warning,error,debug,dtmf
messages => notice,warning,error,debug,dtmf
full => notice,warning,error,debug,verbose,dtmf,fax

so please help how can I capture the digits pressed by caller..

Cheers!

Statistics : Posted by nasirjavaid • on Mon Jan 18, 2016 10:09 am • Replies 5 • Views 140
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Voicemail Trash?

January 13, 2016, 2:34 pm
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If you are using a database (ODBC) to store your voicemail messages, you can create a sql trigger to copy all messages in a another table. That's what we did. Users can retrieve their voicemail messages up to 30 days via a web interface.

Statistics : Posted by tim007 • on Wed Jan 13, 2016 4:34 pm • Replies 2 • Views 138
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Voicemail Trash?

January 13, 2016, 2:34 pm
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Howdy,

A trash folder probably isn't a bad idea.

You may or may not want to approach that by modifying app_voicemail - it's pretty gnarly. It might be just as easy to build a voicemail replacement that has your desired feature. We did an example of voicemail in node here: https://github.com/asterisk/node-voicemail

It may be that's an easier jumping off point.

Cheers

Statistics : Posted by tim007 • on Wed Jan 13, 2016 4:34 pm • Replies 2 • Views 138
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Voicemail Trash?

January 13, 2016, 2:34 pm
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Are there any thoughts / plans on implementing a "trash" folder for voicemails? I have users who would like to be able to retrieve recently deleted voicemails. My thought would be that it would work like many email systems, where it would keep deleted messages in the trash for some period of time before automatically deleting them completely.

Statistics : Posted by tim007 • on Wed Jan 13, 2016 4:34 pm • Replies 2 • Views 138
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no ipv6 dns name resolution?

December 22, 2014, 10:59 am
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no ipv6 dns name resolution?

December 22, 2014, 10:59 am
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Yes, I did, went through that already...

Here are the (relevant) entries in the configuration files:

pjsip.conf
Code: [udp-ipv6]
type=transport
protocol=udp
bind=[2a02:8070:86c0:2ca0:ba27:ebff:feda:bbb6]:5060

...

[endpoint-ipv6](!)
type=endpoint
context=ExternalSets
transport=udp-ipv6
rtp_ipv6=yes
disallow=all
;allow=g722
allow=alaw
allow=gsm
allow=g726
allow=ulaw

[registration-ipv6](!)
type=registration
transport=udp-ipv6
;auth_rejection_permanent=no
retry_interval=60

[easybell](registration-ipv6)
outbound_auth=easybell
server_uri=sip:sip1.easybell.de:5060
client_uri=sip:<userid>@sip1.easybell.de:5060

[easybell](endpoint-ipv6)
;context=ExternalSets
outbound_auth=easybell
aors=easybell


The own IPV6 address is recognized:

Code: [Dec 22 19:24:14] DEBUG[25222] pjsip:     udpv60x998200 SIP UDP IPv6 transport started, published address is [2a02:8070:86c0:2ca0:ba27:ebff:feda:bbb6]:5060

...

[Dec 22 19:24:20] VERBOSE[25222] res_pjsip_multihomed.c: Local IPv4 address determined to be: 192.168.178.99
[Dec 22 19:24:20] VERBOSE[25222] res_pjsip_multihomed.c: Local IPv6 address determined to be: [2a02:8070:86c0:2ca0:ba27:ebff:feda:bbb6]


The IP address of the partner is recognized:
Code: [Dec 22 19:24:20] VERBOSE[25222] config.c: Parsing '/etc/asterisk/pjsip.conf': Found
[Dec 22 19:24:20] DEBUG[25222] netsock2.c: Splitting 'sip1.easybell.de' into...
[Dec 22 19:24:20] DEBUG[25222] netsock2.c: ...host 'sip1.easybell.de' and port ''.
[Dec 22 19:24:20] DEBUG[25222] netsock2.c: Splitting '2001:4090:4008::124' into...
[Dec 22 19:24:20] DEBUG[25222] netsock2.c: ...host '2001:4090:4008::124' and port ''.
[Dec 22 19:24:20] DEBUG[25222] acl.c: [2001:4090:4008::124]:0/[ffff:ffff:ffff:ffff:ffff:ffff:ffff:ffff]:0 sense 0 appended to ACL


The problem is when Asterisk tries to register with the other host.

Statistics : Posted by rbasche • on Mon Dec 22, 2014 12:59 pm • Replies 5 • Views 903
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no ipv6 dns name resolution?

December 22, 2014, 10:59 am
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Did you compile pjproject w/ IPv6 support? pjproject does not enable IPv6 by default.

https://wiki.asterisk.org/wiki/display/ ... +pjproject

Quote:IPv6 support in pjproject is, by default, disabled. To enable it, add CFLAGS='-DPJ_HAS_IPV6=1' to your http://forums.asterisk.org/configure command.


Statistics : Posted by rbasche • on Mon Dec 22, 2014 12:59 pm • Replies 5 • Views 903
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no ipv6 dns name resolution?

December 22, 2014, 10:59 am
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When trying to setup asterisk 13.1 with PJSIP to connect to my IPV6 capable SIP provider the registration fails.

Code: [Dec 22 19:24:24] DEBUG[25247] pjsip:      tsx0x110736c .Transaction created for Request msg REGISTER/cseq=36181 (tdta0x721d90)
[Dec 22 19:24:24] DEBUG[25247] pjsip:      tsx0x110736c Sending Request msg REGISTER/cseq=36181 (tdta0x721d90) in state Null
[Dec 22 19:24:24] DEBUG[25247] pjsip:     sip_resolve.c .Starting async DNS A query: target=sip1.easybell.de, transport=Unspecified, port=5060
[Dec 22 19:24:24] DEBUG[25247] pjsip:        resolver.c .Transmitting 34 bytes to NS 0 (192.168.178.1:53): DNS A query for sip1.easybell.de: Success
[Dec 22 19:24:24] DEBUG[25247] pjsip:      tsx0x110736c .State changed from Null to Calling, event=TX_MSG
[Dec 22 19:24:24] DEBUG[25252] pjsip:        resolver.c Received 50 bytes DNS response from 192.168.178.1:53
[Dec 22 19:24:24] DEBUG[25252] pjsip:        resolver.c Nameserver 192.168.178.1:53 state changed Active --> Active
[Dec 22 19:24:24] WARNING[25252] pjsip:      tsx0x110736c Failed to send Request msg REGISTER/cseq=36181 (tdta0x721d90)! err=171064 (Unsuitable transport selected (PJSIP_ETPNOTSUITABLE))
[Dec 22 19:24:24] DEBUG[25252] pjsip:      tsx0x110736c State changed from Calling to Terminated, event=TRANSPORT_ERROR
[Dec 22 19:24:24] DEBUG[25252] pjsip:      tsx0x110736c Timeout timer event
[Dec 22 19:24:24] WARNING[25247] res_pjsip_outbound_registration.c: No response received from 'sip:sip1.easybell.de:5060' on registration attempt to 'sip:004970718604945@sip1.easybell.de:5060', retrying in '60'
[Dec 22 19:24:24] DEBUG[25252] pjsip:      tsx0x110736c .State changed from Terminated to Destroyed, event=TIMER


For my naive eyes it looks as if there is no ipv6 name resolution (a DNS AAAA query) performed.

What is missing? Please help.

Statistics : Posted by rbasche • on Mon Dec 22, 2014 12:59 pm • Replies 5 • Views 903
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PJSIP Outbound Proxy

January 19, 2016, 7:57 am
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If there's a wiki page you came across which would have helped please leave a comment there with some information about how it could have and we'll incorporate something.

Statistics : Posted by asackheim • on Tue Jan 19, 2016 9:57 am • Replies 3 • Views 125
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