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11
YateBTS / Outbound Text Messages? Is it possible? Any one done or thought about it?
« Last post by kpz on July 11, 2018, 03:44:18 PM »
Yo,
I got the SIP outbound call stuff setup and working as well as a data connection for the phone. I can place out bound calls and get out on the web. I just was wondering if there is a next step available with yate and yateBTS to get text messages going outbound? Is there any SIP like services for text messages?

Thank you

Update:
Here is my console output when trying to send a text message, i feel like the answer is here, just cant see it

2018-07-18_15:12:00.347512 <mbts:NOTE> proc 17398 TRXManager.cpp:127:sendCommandPacket: thread 1960834128: this:0x22c29c command CMD 0 NOHANDOVER 0
2018-07-18_15:12:00.348955 <mbts:NOTE> proc 17398 TRXManager.cpp:152:sendCommandPacket: thread 1960834128: this:0x22c29c response RSP NOHANDOVER 0 0 to command CMD 0 NOHANDOVER 0
2018-07-18_15:12:00.349107 <mbts:NOTE> proc 17398 TRXManager.cpp:127:sendCommandPacket: thread 1960834128: this:0x22c29c command CMD 0 NOHANDOVER 0
2018-07-18_15:12:00.350428 <mbts:NOTE> proc 17398 TRXManager.cpp:152:sendCommandPacket: thread 1960834128: this:0x22c29c response RSP NOHANDOVER 0 0 to command CMD 0 NOHANDOVER 0
12
Yate based projects / Re: BILLING FOR EXISTING YATE SETUP
« Last post by midafricam on July 11, 2018, 09:30:13 AM »
If you use Asterisk as your SIP Server - you could use the Opensource A2Billing as a billing engine
Works well on Asterisk for collecting Call Reports for Post and PrePaid customers
 Great installation guide for A2Billing
13
Yate users hangout place / Re: [ YATE-6.0.1 ] CDRBuild Initialize & Finalize Issue
« Last post by marian on July 11, 2018, 12:39:07 AM »
There is an issue with billid: outgoing channel may change its billid when incoming channel is forked and outgoing channel is connected to another channel before being connected to initial channel.
This will be fixed (I can't say when).

Meanwhile just stop using cdrcombine.

You may implement some logic to track all channels and their billid.
You may send a call.cdr with operation=finalize when necessary.
Keep in mind that this may affect cdr storage!
14
Yate based projects / Re: G72x Codecs
« Last post by Bell on July 10, 2018, 02:44:41 PM »
Is it possible to compile codecs for actual yate version or publish howto ?
Thank you!
15
Yate users hangout place / Autodial Implementation Idea
« Last post by ganapathi on July 10, 2018, 12:47:03 PM »
Hi,

As i am trying to do autodial feature on yate but have some question/difficulties faced as of now to do the same.

  • As i need to search the number from database and needs to dial outbound call
  • And if not answered then need to hangup relevant channel and close the call with proper initialize, finalize & combine method
  • If answered then needs to do keep search for available user and transfer the call and finally need to update cdr with proper outleg records with SIP user instead of utility channels.

Here my question is
  • How to create channel and connect call. Is it possible to do without any utility channel. If not then how to use by using fork/dumb channel
  • How to override leg & outleg as sig & sip channel parameters instead of 3 leg on cdr method.

And also attached the code which is initiated as of now. As of now it's calling customer and play the wave file and closing the cdr properly. But if i calling sip user once answered then how to handle cdr .
16
Windows / YATE client with Cisco Expressway
« Last post by lizh on July 10, 2018, 07:10:49 AM »
I've been testing YATE with Cisco Expressway.  SIP-to-SIP works fine, SIP-to-H.323 works fine but H.323-to-SIP doesn't. The receiving (SIP) YATE client says 'Failed to open Audio, Please Check your Sound Card'.   

I also tested with another SIP client to YATE (SIP) and I get the same message.  I also can't see any SDP messages between the two devices before the call fails so first question, does the YATE client support 'delayed offer' and if so, do I need to tweak something to make this work?  Second question, does the message 'Failed to open Audio, Please Check your Sound Card' really mean there is something wrong with my sound card (which would be strange as the other calls connect)?
17
Features requests / Re: SIP UPDATE support
« Last post by marian on July 09, 2018, 08:10:38 AM »
1xx messages are provisional, not final.

For custom methods (added in methods section, not handled internally) you must handle a sip.method_name message.
For your need you must handle the sip.update message and return true.

A simple regexroute would be:

[extra]
sip.update=50

[sip.update]
.*=return true
18
Hi
No My question is to clear cdrcombine entry where it's still not cleared when call legs are cleared.

A(sig/1) calls B(sip/1) : billid 1
then
B(sip/1) calls C ( sip/2) : billid 2

In here cdrcombine is does the job for billid 1. But sometime is not doing for billid2. Where sip/1 & sip/2& sig/1 channels are disconnected but cdrcombine is not performed for billid2 and i think incoming finalize also not performed here.

Logic : It's Just conference call (  A -> B then B calls C with A on another line ).

Please help me to understand and solve this.

19
Features requests / Re: SIP UPDATE support
« Last post by idelac3 on July 09, 2018, 03:13:25 AM »
thanks, it works.

However, I noticed that Yate replies on SIP UPDATE with 100 Trying:

Code: [Select]
------
<sip:INFO> 'udp:0.0.0.0:5060' received 789 bytes SIP message from 192.168.56.102:5060 [0x2449be0]
------
UPDATE sip:1001@192.168.56.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.102:5060;rport;branch=z9hG4bKPjdd9781d1-d31d-431a-a0b4-83e5ad36b7fd
From: "PhonerLite1" <sip:1002@192.168.56.102>;tag=7a8972c8-c040-4a69-b7d9-cfe0eb80d131
To: <sip:1001@192.168.56.1>;tag=394803623
Contact: <sip:b9804293-2d29-46fe-a58d-11ebe10cb442@192.168.56.102:5060>
Call-ID: f79080c4-937a-4ad6-b6fe-5d3187c4d141
CSeq: 6713 UPDATE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length:   240

v=0
o=- 2038308401 2038308402 IN IP4 192.168.56.102
s=Asterisk
c=IN IP4 192.168.56.101
t=0 0
m=audio 5062 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
------

<INFO> Auto changing RTP address from 192.168.56.102:10384 to 192.168.56.101:5062

and here is 100 Trying:

Code: [Select]
<sip:INFO> 'udp:0.0.0.0:5060' sending code 100 0x7f16f800b6a0 to 192.168.56.102:5060 [0x2449be0]
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.56.102:5060;rport=5060;branch=z9hG4bKPjdd9781d1-d31d-431a-a0b4-83e5ad36b7fd;received=192.168.56.102
From: "PhonerLite1" <sip:1002@192.168.56.102>;tag=7a8972c8-c040-4a69-b7d9-cfe0eb80d131
To: <sip:1001@192.168.56.1>;tag=394803623
Call-ID: f79080c4-937a-4ad6-b6fe-5d3187c4d141
CSeq: 6713 UPDATE
Server: YATE/4.2.0
Content-Length: 0

------

Do you know how to make Yate to replay with 200 OK for SIP UPDATE request ?
20
Features requests / Re: SIP UPDATE support
« Last post by marian on July 09, 2018, 01:40:40 AM »
ysipchan.conf

[methods]
update=BOOLEAN

The value of parameter is a boolean (yes/no) indicating auth is required (this is the default) or not
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