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Topics - dj_ndc

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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

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.

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.

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

Yate bugs / No Prgress Indicator in q931 PROGRESS

« on: September 18, 2015, 05:33:30 AM »

Hello

Yate after ISUP :

CPR [cic=3 label=x-x-0:1-y-y:3]
protocol-type='itu-t'
message-type='CPR'
EventInformation='progress'
BackwardCallIndicators='charge,called-ordinary,e2e-none,isup-path,sccp-none'

Doesn't place Progress Indicator Len: x02, Data: x8188 in PROGRESS Message, here is Call Frow, so Site A release this call with "(96): Mandatory information element is missing":

L3->L2 : 18.09.2015 13:27:49
ETSI Message Type DCHAN: x07 CALLREF: x0052 MSG = SETUP
ETSI Information Element Identifier
Bearer Capability Len: x03, Data: x9090A3 3.1 kHz audio
Channel Ident. Len: x03, Data: xA183
Progress Indicator Len: x02, Data: x8183
CallingPartyNumber Len: x0B, Data: x2180 'xxxxx'
Type Of Number: National number
Numbering Plan: ISDN/telephony numbering plan
Presentation Indicator: Presentation allowed
Screening Indicator: User-provided, not screened
CalledPartyNumber Len: x0A, Data: x8136 'yyyyyyy'
Type Of Number: Unknown
Numbering Plan: ISDN/telephony numbering plan

L2->L3 : 18.09.2015 13:27:49 08 02 80 52 02 18 03 A1 83 81
ETSI Message Type DCHAN: x07 CALLREF: x8052 MSG = CALL PROCEEDING
ETSI Information Element Identifier
Channel Ident. Len: x03, Data: xA18381

L2->L3 : 18.09.2015 13:27:51 08 02 80 52 03
ETSI Message Type DCHAN: x07 CALLREF: x8052 MSG = PROGRESS

L3->L2 : 18.09.2015 13:27:51 08 02 00 52 7D 08 03 81 E0 1E 14 01 03
ETSI Message Type DCHAN: x07 CALLREF: x0052 MSG = STATUS
ETSI Information Element Identifier
Cause Len: x03, Data: x81E01E (96): Mandatory information element is missing
x14 Len: x01, Data: x03

L2->L3 : 18.09.2015 13:27:51 08 02 80 52 5A 08 03 0A 80 E5
ETSI Message Type DCHAN: x07 CALLREF: x8052 MSG = RELEASE COMPLETE
ETSI Information Element Identifier
Cause Len: x03, Data: x0A80E5

Yate based projects / Setting CalledPartyNumber.nature in ISUP

« on: September 09, 2015, 01:42:49 AM »

Hello

I'am using Sangoma A104. First port is with SS7 and ISUP, 2nd, 3rd, 4th are configured as DSS1.
When I try to send call from DSS to SS7, and try to set CalledPartyNumber.nature and CallingPartyNumber.nature to international. It doesn't work. Nature is always set by Incomming ISDN setup "Calling Party Number".
I am only able to change this Value for SIP calls comming to SS7, here is my route:

.*=sig/${called}.;link=isup1;sig.CalledPartyNumber.nature=international;sig.CallingPartyNumber.screened=network-provided;sig.CallingPartyNumber.nature=international

I works only for SIP calls, when I try to send CALL from ISDN from 4th port, nature is always set by Incomming Setup from that port. Is there a way to force overwrite this Variable ?

Yate bugs / Sangoma + ISUP + error in the calculation of floating point

« on: September 04, 2015, 02:32:52 AM »

Hello

Has someone solved issue that was posted to mailing group 22 Aug 2013:

http://yate.null.ro/archive/?action=show_msg&actionargs[]=86&actionargs[]=4

I have the same problem with Yate (6967), and WANPIPE Release: 7.0.14. When I disable:

;voicegroup=w1g2

in wpcard.conf MTP3 goes Up and operational:

module=sig,component=linkset1,type=ss7-mtp3;status=operational

and receive:

2015-09-04_10:28:45.781650 <linkset1:MILD> Received SLTM ITU,0-13-7:0-27-6:0 (111:222:0) wrong National NI with 6 bytes
2015-09-04_10:28:45.781722 <linkset1:MILD> Sending SLTA ITU,0-27-6:0-13-7:0 (222:111:0) with 6 bytes

When I enable "voicegroup=w1g2" in wpcard.conf. I get error:

Sep 4 10:29:41 ss7 kernel: [74482.610146] YSIG Trunk[7207] trap divide error ip:7f504ac59ae0 sp:7f5047b46dc0 error:0 in ysigchan.yate[7f504ac46000+29000]
Sep 4 10:29:41 ss7 kernel: [74482.612867] af_wanpipe: MAJOR ERROR, Data lost on sock release !!!

Yate reports:

The error in the calculation of floating point.

Yate users hangout place / SIP User-to-User field to ISUP sig.UserToUserInformation

« on: July 30, 2014, 04:45:38 AM »

Hello

I will need to pass User-to-User data from SIP-T to SS7 network in both directions.
Is there a way to do it ?
I have INVITE from SIP-T to Yate with:

INVITE sip:123456789@192.168.1.2 SIP/2.0
User-to-User: 12345678999

But how to place it in regexroute dialplan ? Is there some sip variable that contain this value ?

I have tried tu put constant value like:

.*=sig/${called}.;link=link-1;sig.UserToUserInformation=12345

But there wasn't any trap of it in sigdump file.

Yate bugs / CIC in changing state true

« on: March 04, 2014, 06:58:43 AM »

Hello

Has someone solve this issue ?

http://yate.null.ro/archive/?action=show_msg&actionargs[]=77&actionargs[]=87

I am using YATE 5.0.0-1 r5675 now, but still has problem when some of CICs goes to changing state.
block/unblock/available, shuting down controller doesn't help.

The is no information about this CICs in ysigdata.conf, our remote PC send calls to Us on CICs that we see as changing. So after that Our PC response:

circuit=456,span=E1-15,status=Idle,lockedlocal=false,lockedremote=false,changing=true,flags=0x15

<link-1/ISUP:NOTE> 'IAM' with cic=456: can't reserve circuit

And call is dropped.

Only restarting of Yate help to clear changing state CICs. Is there a way to clear changing CICs to false by control command or some other way ? Without restarting ?

Greetings
Andrzej

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