Quantcast
Channel: Asterisk Forums
Viewing all 7460 articles
Browse latest View live
↧

Adding an OVH sip line to asterisk via FreePBX

June 9, 2015, 7:06 am
$
0
0
Hello,

I'm rather new to freepbx and asterisk.

I succesfully registered my first trunk and can receive call from it.
But now, i'm looking for something more specific and online i didn't find anything to help me.

Basically i have a OVH sip line (not a trunk just a classic sip) which i can connect to with a softphone (Xlite) but i was wondering if i could add this line in asterisk so it could re-route the calls from this line to others phones.

Actually we have a Gigaset DX800A, and in it we can register sip line (still not trunk) and create a call routing to our different phones and it works just fine (but limited to 7 sip lines).

I wanted to do the same but with asterisk and freepbx (unilimited lines could be added right ? as soon i have enough sip line).

Thanks for your help !!
Image

Statistics : Posted by blet • on Tue Jun 09, 2015 7:06 am • Replies 1 • Views 84
↧
Search

Asterisk not reponse when use transport TLS with blink?

June 7, 2015, 3:36 am
$
0
0
What does the equivalent log from Asterisk show?

You appear to be registering to a public address but specifying a private contact address. That is unlikely to be right.

Statistics : Posted by ntlongit • on Sun Jun 07, 2015 3:36 am • Replies 1 • Views 95
↧
↧

Asterisk not reponse when use transport TLS with blink?

June 7, 2015, 3:36 am
$
0
0
Hello everyone,
I make Secure call follow turorial:

https://wiki.asterisk.org/wiki/display/ ... g+Tutorial

But when I use blink to connect with Asterisk, it cannot register with server. Blink send package to server but don't receive anything from server. This is log file of blink:

Code: 2015-06-07 16:31:09.374653 [blink 15391]: SENDING: Packet 1, +0:00:02.699408
192.168.0.105:41714 -(SIP over TLS)-> 52.68.6.174:5061
REGISTER sip:52.68.6.174 SIP/2.0
Via: SIP/2.0/TLS 192.168.0.105:41714;rport;branch=z9hG4bKPjso-71UQQqAJxF9vE6jAFy1nSEjhuDrzW;alias
Max-Forwards: 70
From: "TLS" <sip:100@52.68.6.174>;tag=k0Vr4lthAdyNqzI8WCtpbPb23zHbLWBR
To: "TLS" <sip:100@52.68.6.174>
Contact: <sip:04187329@192.168.0.105:5061;transport=tls>;+sip.instance="<urn:uuid:84c3c483-4a27-455b-bd0c-18777738c747>";+sip.ice
Call-ID: RzP9cjLA09rBli4L6LHFcCKQrxRt8wlX
CSeq: 1 REGISTER
Expires: 600
Supported: gruu
User-Agent: Blink 1.3.0 (Linux)
Content-Length:  0


--
2015-06-07 16:31:17.088359 [blink 15391]: SENDING: Packet 2, +0:00:10.413114
192.168.0.105:41714 -(SIP over TLS)-> 52.68.6.174:5061
PUBLISH sip:100@52.68.6.174 SIP/2.0
Via: SIP/2.0/TLS 192.168.0.105:41714;rport;branch=z9hG4bKPjGp.HTVJbDS3sJahS04zW-OtKN0Xpsqy.;alias
Max-Forwards: 70
From: "TLS" <sip:100@52.68.6.174>;tag=wBHAr1h3hNYrdLT3ZDUAF53dRJcIIePx
To: "TLS" <sip:100@52.68.6.174>
Call-ID: HNUPxs805C41dvMD4hF28AkBTrt-Tf-b
CSeq: 1 PUBLISH
Event: presence
Expires: 3600
User-Agent: Blink 1.3.0 (Linux)
Content-Type: application/pidf+xml
Content-Length:   763

<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3A100%4052.68.6.174"><tuple id="SID-38aa10cc6cfb81923405855d66bdc666"><status><basic>closed</basic><agp-pidf:extended>offline</agp-pidf:extended></status><caps:servcaps/><contact>sip%3A100%4052.68.6.174</contact><timestamp>2015-06-07T16:17:28.184219+07:00</timestamp></tuple><dm:person id="PID-38aa10cc6cfb81923405855d66bdc666"><rpid:activities><rpid:other>offline</rpid:other></rpid:activities><dm:timestamp>2015-06-07T16:17:28.184219+07:00</dm:timestamp></dm:person></presence>
--
2015-06-07 16:31:27.587794 [blink 15391]: SENDING: Packet 3, +0:00:20.912549
192.168.0.105:45600 -(SIP over TLS)-> 52.68.6.174:5061
REGISTER sip:52.68.6.174 SIP/2.0
Via: SIP/2.0/TLS 192.168.0.105:45600;rport;branch=z9hG4bKPjV9jyK62bRpxCXLTZpeliKfPcqwS9.yKU;alias
Max-Forwards: 70
From: "TLS" <sip:100@52.68.6.174>;tag=.IIaCAgL3vSe5WOK9BydgD85GMnG0IfI
To: "TLS" <sip:100@52.68.6.174>
Contact: <sip:04187329@192.168.0.105:5061;transport=tls>;+sip.instance="<urn:uuid:84c3c483-4a27-455b-bd0c-18777738c747>";+sip.ice
Call-ID: 2el8S9K3dZ5K.nfE4cmfZiEyiWZAJvFZ
CSeq: 1 REGISTER
Expires: 600
Supported: gruu
User-Agent: Blink 1.3.0 (Linux)
Content-Length:  0


--
2015-06-07 16:31:52.659980 [blink 15391]: SENDING: Packet 4, +0:00:45.984735
192.168.0.105:45600 -(SIP over TLS)-> 52.68.6.174:5061
SUBSCRIBE sip:100@52.68.6.174 SIP/2.0
Via: SIP/2.0/TLS 192.168.0.105:45600;rport;branch=z9hG4bKPjqWIWhVq68jbtOesTurODNdmsjIareUPk;alias
Max-Forwards: 70
From: "TLS" <sip:100@52.68.6.174>;tag=da6IzNUWj4Wmdc3eyBMG15ta-X-eXp.L
To: <sip:100@52.68.6.174>
Contact: <sip:04187329@192.168.0.105:5061;transport=tls>
Call-ID: 1sRN-lO1ik1zG.VqF4AhiXFhHDgu4cqj
CSeq: 20688 SUBSCRIBE
Event: message-summary
Expires: 3600
Supported: 100rel, replaces, norefersub, gruu
Accept: application/simple-message-summary
Allow-Events: conference, message-summary, dialog, presence, presence.winfo, xcap-diff, dialog.winfo, refer
User-Agent: Blink 1.3.0 (Linux)
Content-Length:  0


This is my sip.conf:
Code: tlsenable=yes
tlsbindaddr=0.0.0.0:5061
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
tlsdontverifyserver=yes

...
[100]
type=peer
secret=123
host=dynamic
canreinvite=no
context=default
dtmfmode=rfc2833
qualify=yes
disallow=all
allow=gsm
allow=alaw
allow=ulaw
allow=g722
allow=g726
allow=h261
allow=h263
allow=h263p
allow=h264
transport=tls



How can I fix it?
Thanks you everyone.

Statistics : Posted by ntlongit • on Sun Jun 07, 2015 3:36 am • Replies 1 • Views 95
↧

Problem with incoming traffic

June 7, 2015, 6:26 am
$
0
0
users.conf is deprecated.

Typically you would have a NAT or firewall issue, e.g. you may have forgotten to NAT/pass the RTP port range.

I can't tell whether you need special NAT handling without more information on your network connectivity and I can't tell whether you have told Asterisk how to find its external address and possibly override the RTP address without sip.conf.

Statistics : Posted by ballpin • on Sun Jun 07, 2015 6:26 am • Replies 1 • Views 115
↧

Problem with incoming traffic

June 7, 2015, 6:26 am
$
0
0
Hello,

I have succeeded to create my Switchboard the way i want. But there is one issue i don't seem to find a solution for. When i call other users internally there is no sound. According to the statics its because its not getting any incoming packages.

This is how my user looks like in users.conf
Code: [6001]
fullname=User
registersip=no
host=dynamic
callgroup=1
mailbox=6001
call-limit=100
type=peer
username=6001
transfer=yes
callcounter=yes
context=DLPN_DialPlan1
cid_number=6001
hasvoicemail=no
vmsecret=
email=
threewaycalling=no
hasdirectory=yes
callwaiting=no
hasmanager=yes
hasagent=yes
hassip=yes
hasiax=yes
secret=secretpassword
nat=yes
canreinvite=no
dtmfmode=rfc2833
insecure=port,invite
fromdomain=sip.uvtc.net
pickupgroup=1
macaddress=6001
autoprov=yes
label=6001
linenumber=1
LINEKEYS=1
managerread=system,call,log,verbose,command,agent,user,config,originate
managerwrite=system,call,log,verbose,command,agent,user,config,originate
disallow=all
allow=ulaw,gsm

[general]
fullname=New User
userbase=6000
hasvoicemail=yes
vmsecret=1234
hassip=yes
hasiax=yes
hasmanager=no
callwaiting=yes
threewaycalling=yes
callwaitingcallerid=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
callgroup=1
pickupgroup=1
port=5060



I would be very thankful if anyone could help me resolve this. I really want it to work.

// Ivan

Statistics : Posted by ballpin • on Sun Jun 07, 2015 6:26 am • Replies 1 • Views 115
↧
↧

Subscription to resource list fails

June 9, 2015, 9:02 am
$
0
0
Your device appears to be subscribing to application/dialog-info+xml and not RLS. This is also indicated by the Event header being "dialog". As a result res_pjsip_pubsub chooses the dialog-info+xml implementation and not RLS.

Statistics : Posted by leosori • on Tue Jun 09, 2015 9:02 am • Replies 1 • Views 99
↧

Subscription to resource list fails

June 9, 2015, 9:02 am
$
0
0
Hi,

I'm trying to use the resource list feature of Asterisk, but I simply cannot subscribe to the list.

Quote:res_pjsip_exten_state.c:337 new_subscribe: Extension state subscription failed: Extension from-internal does not exist in context 'rlist' or has no associated hint


Extension and context are mixed up in the error message, but I think this is just a bug in the error message string.

I can see a SIP message 'SUBSCRIBE rlist@asterisk' coming from the phone and Asterisk responds with a 404.

Code: <--- Received SIP request (901 bytes) from UDP:192.168.42.71:5060 --->
SUBSCRIBE sip:rlist@192.168.43.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.42.71:5060;branch=z9hG4bK549715202;rport
From: <sip:65@192.168.43.2>;tag=808239491
To: <sip:rlist@192.168.43.2>
Call-ID: 914519428-5060-482@BJC.BGI.EC.HB
CSeq: 24801 SUBSCRIBE
Contact: <sip:65@192.168.42.71:5060>
Authorization: Digest username="65", realm="asterisk", ...
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2160 1.0.4.23
Expires: 3600
Supported: replaces, path, timer, eventlist
Event: dialog
Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<--- Transmitting SIP response (329 bytes) to UDP:192.168.42.71:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.42.71:5060;rport=5060;received=192.168.42.71;branch=z9hG4bK549715202
Call-ID: 914519428-5060-482@BJC.BGI.EC.HB
From: <sip:65@192.168.43.2>;tag=808239491
To: <sip:rlist@192.168.43.2>;tag=z9hG4bK549715202
CSeq: 24801 SUBSCRIBE
Server: Asterisk PBX 13.3.0
Content-Length:  0


I can subscribe to each individual extension in the resource list below and the phones receive the correct state updates. So the general setup seems to be working. Is there any step I am missing?

Config:
Asterisk 13.3.0

pjsip.conf
Code: [rlist]
type = resource_list
event = presence
list_item = 65
list_item = 86
list_item = 88


Statistics : Posted by leosori • on Tue Jun 09, 2015 9:02 am • Replies 1 • Views 99
↧

transfer on hangup

June 8, 2015, 3:15 am
$
0
0
You never said you were using the Queue application. That was critical information.

Unless an option, similar to g, has been added in later versions, you will need to Dial a local channel, with g, and then have the local channel call Queue.

Under no circumstances must you call Hangup.

Your current dialplan will terminate if the call succeeds and go to voicemail otherwise. Without the goto, it wouldn't get beyond the Hangup.

Statistics : Posted by mathabathe • on Mon Jun 08, 2015 3:15 am • Replies 3 • Views 173
↧
Search

transfer on hangup

June 8, 2015, 3:15 am
$
0
0
thank you david for the reply but I am still lost :
here is my dialplan :how do I use g option here so that when the callee hangs up , the call continue on the dial plan on the next priority ?

[Eng-queues]
exten => 1,n,Set(CALLERID(name)= Eng: ${DB(cidname/${CALLERID(dnid)})})
exten => 1,n,NoOp(----${CALLERID(name)}----)
exten => 1,n,NoOp(----${CALLERID(name)}----)
exten => 1,n,NoOp(----${CALLERID(name)}----)
exten => 1,n,NoOp(----${CALLERID(name)}----)
exten => 1,n,NoOp(----${CALLERID(name)}----)
exten => 1,n,NoOp(----${CALLERID(name)}----)
exten => 1,n,NoOp(----${CALLERID(name)}----)
exten => 1,n,NoOp(----${CALLERID(name)}----)
exten => 1,n,NoOp(----${CALLERID(name)}----)
exten => 1,n,NoOp(----${CALLERID(name)}----)
exten => 1,n,NoOp(----${CALLERID(name)}----)
exten => 1,n,QueueLog(201,${UNIQUEID},NONE,INFO,DID|${FROM_DID})
exten => 1,n,Queue(201,tr,,,45)
exten => 1,n,Goto(ext-local,vmb2385,1)
exten => 1,n,Hangup()
exten => 1,n,Dial(SIP/2000,3) ; this is what needs to happen when the callee hangs up

Statistics : Posted by mathabathe • on Mon Jun 08, 2015 3:15 am • Replies 3 • Views 173
↧
↧

transfer on hangup

June 8, 2015, 3:15 am
$
0
0
g option on Dial and test DIALSTATUS for SUCCESS.

Statistics : Posted by mathabathe • on Mon Jun 08, 2015 3:15 am • Replies 3 • Views 173
↧

transfer on hangup

June 8, 2015, 3:15 am
$
0
0
Hi All,

i am running asterisk 1.8

When someone calls our office, I'd like them to take a real quick survey at the end of the call. We can transfer the call manually, but it takes too long and the caller doesn't want to wait for us to transfer them to the survey extension.

So my question is,how can send the callers to the survey when the agent hangs up?



Thanks!

Statistics : Posted by mathabathe • on Mon Jun 08, 2015 3:15 am • Replies 3 • Views 173
↧

Call Delays, redundant SIP

June 10, 2015, 11:25 am
$
0
0
Obtain SIP traces, with timestamps, as evidence that they have a problem, then talk to the ITSP. If that fails to resolve the problem, change provider.

Note there isn't a standard way, as this is not a normal problem.

Statistics : Posted by dbrooke • on Wed Jun 10, 2015 11:25 am • Replies 3 • Views 130
↧

Call Delays, redundant SIP

June 10, 2015, 11:25 am
$
0
0
good suggestion. Image

Statistics : Posted by dbrooke • on Wed Jun 10, 2015 11:25 am • Replies 3 • Views 130
↧
↧

Call Delays, redundant SIP

June 10, 2015, 11:25 am
$
0
0
So commpeak. What is the suggested / standard way to fix this?

Should I changed the SIP provider? Add a redundant SIP provider?

Thanks!

Statistics : Posted by dbrooke • on Wed Jun 10, 2015 11:25 am • Replies 3 • Views 130
↧

Call Delays, redundant SIP

June 10, 2015, 11:25 am
$
0
0
This is an ITSP issue.

Statistics : Posted by dbrooke • on Wed Jun 10, 2015 11:25 am • Replies 3 • Views 130
↧
Search

Call Delays, redundant SIP

June 10, 2015, 11:25 am
$
0
0
hello, I've installed a new AsteriskNow system for the first time. Calls are being delayed significantly when there is a queue. Is this a SIP trunk issue? My default paid SIP account (through commpeak) says they provide 20 channels.. which I thought was well enough. However, do I need a redundant SIP trunk as well?

Thanks,
Donovan

Statistics : Posted by dbrooke • on Wed Jun 10, 2015 11:25 am • Replies 3 • Views 130
↧

Decrease time-out when dialing number with callingcard ext

June 11, 2015, 6:54 am
$
0
0
Hello,

We are using the CallingCard function in Asterisk, we call the PBX and then we can dial a number from there. When i'm in the menu a voice says "Please enter the telephone number". After dialing in the number with my phone it takes very long before asterisk is establishing the call. I think it is some kind of time-out, but i want the time-out to be shorter.

Is this possible?

Best regards,
Joost Lauwen

Statistics : Posted by joostspike • on Thu Jun 11, 2015 6:54 am • Replies 0 • Views 19
↧
↧

How to use the PJSIP_HEADER for Paging in PJSIP?

June 12, 2015, 5:53 am
$
0
0
Oh, and you'll also want to add a priority to your gosub which does Return

Statistics : Posted by phonefxg • on Fri Jun 12, 2015 5:53 am • Replies 7 • Views 101
↧

How to use the PJSIP_HEADER for Paging in PJSIP?

June 12, 2015, 5:53 am
$
0
0
Ah, your PJSIP_HEADER usage is also incorrect.

If you want "Call-Info: ;Answer-After=0" in the SIP message then:

Code: exten => addheader,1,Set(PJSIP_HEADER(add,Call-Info)=\;Answer-After=0)


If you want "Call-Info: Answer-After=0" in the SIP message then:

Code: exten => addheader,1,Set(PJSIP_HEADER(add,Call-Info)=Answer-After=0)


Statistics : Posted by phonefxg • on Fri Jun 12, 2015 5:53 am • Replies 7 • Views 101
↧

How to use the PJSIP_HEADER for Paging in PJSIP?

June 12, 2015, 5:53 am
$
0
0
Still no luck.

Statistics : Posted by phonefxg • on Fri Jun 12, 2015 5:53 am • Replies 7 • Views 101
↧
Viewing all 7460 articles
Browse latest View live
Search
<script src="https://jsc.adskeeper.com/r/s/rssing.com.1596347.js" async> </script>