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Messages - dj_ndc

Pages: [1] 2
1
Yate users hangout place / Re: cisco dss1 with mgcpca.conf
« on: May 16, 2016, 02:44:41 PM »
Hello

Thank You for information, because I have found information like: https://twitter.com/yate_voip/status/18256952436,
and I have looked into source code and have found that there are: isdn-iua types in ysigchan.cpp. But debug doesn't show me anything during start. Now I can stop my experimenting.

and:

http://old.yate.ro/pmwiki/index.php?n=Main.Yate3

Yate 3.0.0

This is the first released version of Yate 3. In addition of what version 2.2 provided the following new features are available:
•Linux kernel SCTP support for SIGTRAN
•SIGTRAN M2PA, M2UA and IUA implementation

 

2
Yate users hangout place / Re: cisco dss1 with mgcpca.conf
« on: May 13, 2016, 06:11:33 AM »
Changing to m3ua, makes AS and ASP ACTIVE, but still has no idea, how to use isdn-iua/isdn-iua-gateway/isdn-iua-client:

[linkset-1]
type=ss7-m3ua
link=link0

[link0]
type=ss7-m2ua
sig=dss-iua

[dss-iua]
local=10.10.10.12:2400
remote=10.10.10.20:2400
type=sctp
stream=no
endpoint=yes

Makes:

2016-05-13_14:08:18.934292 <link0:STUB> Please handle ASP message 0 class MGMT
2016-05-13_14:08:18.942106 <link0:NOTE> ASP traffic is now active [0x1f5a5e0]

CISCO:
IUA: AS mg2600 old-state AS-Inactive new-state AS-Active

Name of AS : mg2600
        Total num of ASPs configured : 1
                yate
        Current state :   ACTIVE
        Active ASP : yate
        Number of ASPs up : 1
        Fail-Over time :  4000 milliseconds
        Local address list :  10.10.10.20
        Local port: 2400
        Interface IDs registered with this AS
                 Interface ID
                 256 (Serial1/0:15)



3
Yate users hangout place / Re: cisco dss1 with mgcpca.conf
« on: May 13, 2016, 02:19:08 AM »
Has someone any experience with this kind of configuration ?

My SCTP from Cisco works only when I place type=ss7-m2pa,
I have tried isdn-iua, isdn-iua-client,  isdn-iua-gateway, isdn-q921 without success, logs doesn't show me anything, like loading configuration with types above are ignored.

I have:

sigtransport.conf

[dss-iua]
local=10.10.10.11:2400
remote=10.10.10.20:2400
type=sctp
stream=no
endpoint=yes

Try to configiure ysigchan.conf:

[link-1]
enable=yes
type=isdn-q931
voice=mg2600

[linkset-1]
type=isdn-q931
link=link0

[link0]
type=isdn-iua
sig=dss-iua
autostart=yes

CISCO:

!
iua
  AS mg2600 10.10.10.20 2400
  AS mg2600 sctp-t1init 1000
   ASP yate AS mg2600 10.10.10.11 2400
!
controller E1 1/0
 pri-group timeslots 1-31 service mgcp
!
interface Serial1/0:15
 no ip address
 no logging event link-status
 isdn switch-type primary-net5
 isdn incoming-voice voice
 isdn bind-l3 iua-backhaul mg2600
 isdn sending-complete
 no cdp enable
!
mgcp
mgcp request timeout 10000
mgcp call-agent 10.10.10.11 2727 service-type mgcp version 0.1

Could You help ? Or tell me if it is possible ?

When I change:

[link0]
type=ss7-m2pa
[linkset-1]
type=ss7-mtp3
link=link0

Sigtran start to work, but expecting SS7,which I don't want to use from IAU:

mg2600# show ip sctp association list

** SCTP Association List **

AssocID: 0,  Instance ID: 0
Current state: ESTABLISHED
Local port: 2400, Addrs: 10.10.10.20
Remote port: 2400, Addrs: 10.10.10.11


2016-05-12_12:04:49.957648 <link0:INFO> Interface is up [0x23d3e90]
2016-05-12_12:04:50.938418 <link0:WARN> Received non M2PA message class 0
2016-05-12_12:04:50.938648 <link0:INFO> SS7M2PA Received:
-----
  Version: 1    Message class: 0    Message type: Unknown
  Stream: 0
  FSN : 7       BSN: 786448
  Data: 01 00 00 01 00 00 00 08

4
Yate users hangout place / Re: cisco dss1 with mgcpca.conf
« on: May 12, 2016, 02:13:40 AM »
I have established SCTP, but I works only when I set "type=ss7-m2pa", nothing happens when I try to set type=isdn-iua or type=isdn-iua-gateway or type=isdn-iua-client.

mg2600#show ip sctp association statistics 0

** SCTP Association Statistics **

AssocID/InstanceID: 0/0
Current State: ESTABLISHED
Control Chunks
  Sent: 4  Rcvd: 1
Data Chunks Sent
  Total: 1  Retransmitted: 0
  Ordered: 1  Unordered: 0
  Avg bundled: 1  Total Bytes: 24
Data Chunks Rcvd
  Total: 4  Discarded: 1
  Ordered: 3  Unordered: 0
  Avg bundled: 1  Total Bytes: 68
  Out of Seq TSN: 0
ULP Dgrams
  Sent: 1  Ready: 3  Rcvd: 3

And with ss7-m2pa I have (what is true, because I have PRI ISDN):

2016-05-12_10:10:20.080969 <link0:INFO> Interface is up [0x1df1650]
2016-05-12_10:10:21.074139 <link0:WARN> Received non M2PA message class 0
2016-05-12_10:10:21.074170 <link0:INFO> SS7M2PA Received:
-----
  Version: 1    Message class: 0    Message type: Unknown
  Stream: 0
  FSN : 7       BSN: 786448
  Data: 01 00 00 01 00 00 00 08
-----
2016-05-12_10:10:25.083986 <link0:NOTE> Aborting alignment: Out of service timeout

5
Yate users hangout place / Re: cisco dss1 with mgcpca.conf
« on: May 11, 2016, 04:49:46 AM »
Hello

I think I should do that by "isdn-iua" but I don't know how to configure ysigchan.conf for it. I have configured cisco iua, as AS and ASP, I have established session by SCTP, but don't know how to configure parameters for "type=isdn-iua" in ysigchan ?

[dss1]
enable=yes
type=isdn-iua
voice=mg2600   ????

[sigrandss1]
local=10.10.10.10:2400
remote=10.10.10.20:2400
type=sctp
stream=no
endpoint=yes


CISCO:

ISDN Serial1/0:15 interface
        dsl 0, interface ISDN Switchtype = primary-net5
        L2 Protocol = Q.921  L3 Protocol(s) = IUA BACKHAUL

Name of ASP : yate
Current State of ASP: ASP-Down
Current state of underlying SCTP Association IUA_ASSOC_ESTAB , association id 0





6
Yate users hangout place / cisco dss1 with mgcpca.conf
« on: May 06, 2016, 05:48:26 AM »
Hello

Is there a way to configure Yate to work with cisco configured with DSS1 ?
Like:

pri-group timeslots 1-31 service mgcp

and mgcpca ?

And:

module=sig,trunk=dss1-1,type=isdn-pri-net;circuits=30,status=Layer 2 missing,calls=0,available=30,resetting=0,locked=0,idle=30

How to configure sig parameter for "type=isdn-pri-net" ?

To do sometning like ciscosm, but for DSS1 ?

Greetings
Andrzej Ciupek

7
Yate users hangout place / Re: fork wave/play then callto.3=sip
« on: May 02, 2016, 11:23:03 AM »
Hello

I have solved it by second Yate that works only as wave/player.
So I use the same fork, but I have changed:

callto.1=wave/play//greeting.au;autoprogress=yes

to:

callto.1=sip/sip:${called}@10.11.20.55;rtp_forward=yes;formats=alaw,g729

where in default I have:

^123456789$=wave/play//greeting.au;autoprogress=yes

Then after progress playback, fork return to callto.3, and forward RTP from Gateway, not try to use own rtpproxy.

Greetings
Andrzej

8
Yate users hangout place / Re: fork wave/play then callto.3=sip
« on: April 30, 2016, 06:53:24 AM »
I would line to do something in Asterisk style:

exten => _X.,1,Answer()
exten => _X.,2,Playback(greetings)
exten => _X.,3,Dial(SIP/${EXTEN}@10.11.20.15,,)
exten => _X.,4,Hangup()

but I would like to do it by regexroute, not ivr scripting.

9
Yate users hangout place / fork wave/play then callto.3=sip
« on: April 29, 2016, 04:04:50 PM »
Hello

I need to play progress wave file, before I place a call to destination PBX.
I use fork for this, but after playing wav, when call goes to PBX DTMF doesn't works, they are not forwarded to target PBX, but collected by yrtpwrapper:

<yrtp:INFO> YRTPWrapper::gotDTMF('9') [0x1596290]

I am using:

.*$=fork;callto.1=wave/play//greeting.au;autoprogress=yes;callto.2=|;callto.3=sip/sip:${called}@10.11.20.15;rtp_forward=yes;formats=alaw,g729;stoperror=busy;maxcall=20000;callto.1.fork.calltype=persistent;callto.1.fork.autoring=true;callto.1.fork.automessage=call.progress

is it possible to do ? When I place call only by:

.*$=sip/sip:${called}@10.11.20.15;rtp_forward=yes;formats=alaw,g729

dtmf in rfc2833 is forwarded to destination pbx.


10
Yate bugs / Re: TCP SIP Trunking - Invalid address
« on: March 25, 2016, 07:14:28 AM »
Hello

I think I have found the solution.
I have add to ysipchan.conf:

[listener tcpconn]
type=tcp
addr=10.22.22.9
port=5060

then:

in regexroute:

.*=;oconnection_id=tcpconn;oip_transport=tcp;osdp_forward=no
^.*$=sip/sip:${called}@10.10.10.18:5060

11
Yate bugs / TCP SIP Trunking - Invalid address
« on: March 25, 2016, 06:51:12 AM »
Hello

When I have changed signalling from UDP to TCP, and make connection by regexroute expression, does regexroute for SIP in TCP different ?

^.*$=sip/sip:${called}@10.10.10.18:5060;ip_transport=tcp

I get:

2016-03-25_09:38:13.434008 <sip:ALL> YateSIPEndPoint::buildParty(0x1adbba0,'(null)',0,(nil))
2016-03-25_09:38:13.434088 <sip/14:WARN> Could not create party for 'sip:322000@10.10.10.18' [0x1a84ef0]

SIP/2.0 503 Invalid address: sip:322000@10.10.10.18

error='noconn' code=503 reason='Invalid address: sip:322000@10.10.10.18:5060'

There is no problem when I move back to UDP.


12
Yate users hangout place / Re: Jitter Buffer - SIP to E1 PSTN
« on: December 21, 2015, 02:51:45 AM »
Hello

I have figured out, it was my mistake. I have used old regexroute plan for Cisco SLT with rtp forwarding to Cisco Gateways, so I had:

${rtp_forward}possible=;rtp_forward=yes

I have changed it to rtp_forward=no, and now it works fine, and live changes in yrtpchan.conf works after reload.

Greetings

13
Yate users hangout place / Jitter Buffer - SIP to E1 PSTN
« on: December 18, 2015, 01:15:19 AM »
Hello

I am using Yate with Sangoma as SIP to PSTN Gateway, how to veryfy if jitterbuffer is enabled in configuration ?
I have problem with fax transmission when source transmission came to Yate without jitterbuffering, so I need to enable/determine jitter buffer on Yate, I have set:

yrtpchan.conf

tos=cs0
buffer=200
minjitter=120
maxjitter=200

Greetings

14
Yate bugs / Re: No Prgress Indicator in q931 PROGRESS
« on: October 12, 2015, 03:28:02 AM »
Can it be caused by earlyacm=yes ? Here is ISUP port definition I use, and isdn-pri that call goest to it. I Call from link2 to isup1, PROGRESS goes from isup1 to link2 without Progress Indicator in PROGRESS message:

[isup1]
enable=yes
type=ss7-isup
pointcodetype=ITU
;
;TEST
pointcode=1
defaultpointcode=1
remotepointcode=2
;
netindicator=national
router=ss7router
voice=wanpipe1
lockgroup=yes
earlyacm=yes
ringback=yes
continuity=loopback
sls=cic
numplan=isdn
numtype=national
presentation=allowed
screening=network-provided
print-messages=yes
extended-debug=yes
confirm_ccr=yes
drop_unknown=no
needmedia=no
autostart=yes

[link2]
type=isdn-pri-net
enable=yes
sig=wanpipe2
voice=wanpipe2
switchtype=euro-isdn-e1
strategy=increment
format=alaw
screening=network-provided
numplan=isdn
numtype=national
presentation=allowed
ringback=yes
earlyacm=yes

Greetings
Andrzej

15
Yate bugs / Re: No Prgress Indicator in q931 PROGRESS
« on: September 22, 2015, 01:11:11 AM »
Hello

I think, that Progress Indicator isn't transferred from SETUP to next messages in call proceeding.

Greetings


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