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SIP to H.323 proxy / H323 To SIP Signalling Proxy - SIP REFER error
« on: April 21, 2015, 11:41:27 AM »
Hi all,
I have setup yate as H323 To SIP Signalling Proxy between Avaya and Asterisk.
The proxy works fine until i try to transfer the call.
I am getting the below error as response from yate to the SIP REFER message of Asterisk .
<sip/5:STUB> initTransfer. Possible incomplete NOTIFY party creation [0x7f20a00230e0]
I have enabled transfer in ysipchan.conf
transfer=enable
and my routing in regexroute.conf is the below (example form official website)
${rtp_forward}possible=;rtp_forward=yes
${formats}^\([^,]*\)=;formats=\1
${module}^sip$=h323/${called}@50.x.x.50:1720
${module}^h323$=sip/sip:${called}@50.x.x.30:5060
.*=-;error=forbidden;reason=Protocol not allowed
I would appreciate some guidelines. Any additional configurations needed for yate in order to handle REFER message?
Below is the log with SIP REFER along with sniffed call.route
------
REFER sip:3xxxxxxxxxxxx4@50.x.x.30:5065 SIP/2.0
Via: SIP/2.0/UDP 50.x.x.30:5060;branch=z9hG4bK6ddcc84b;rport
Max-Forwards: 70
From: <sip:20000@50.x.x.30:5060>;tag=as1ca0dea3
To: "50.x.x.50" <sip:3xxxxxxxxxxxx4@50.x.x.30:5065>;tag=1643256148
Contact: <sip:20000@50.x.x.30:5060>
Call-ID: 829575849@50.x.x.30:5065
CSeq: 102 REFER
User-Agent: Asterisk PBX 1.8.32.2
Refer-To: <sip:3000@50.x.x.30:5065>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Referred-By: <sip:20000@50.x.x.30:5060>
Content-Length: 0
------
2015-04-21_19:24:14.404448 <sip:INFO> 'udp:50.x.x.30:5065' sending code 100 0x7f2a04021120 to 50.x.x.30:5060 [0x12bc560]
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 50.x.x.30:5060;branch=z9hG4bK6ddcc84b;rport=5060;received=50.x.x.30
From: <sip:20000@50.x.x.30:5060>;tag=as1ca0dea3
To: "50.x.x.50" <sip:3xxxxxxxxxxxx4@50.x.x.30:5065>;tag=1643256148
Call-ID: 829575849@50.x.x.30:5065
CSeq: 102 REFER
Server: YATE/5.4.3
Content-Length: 0
------
2015-04-21_19:24:14.404936 <sip/1:STUB> initTransfer. Possible incomplete NOTIFY party creation [0x7f2a04005760]
Sniffed 'call.route' time=1429633454.404720
thread=0x7f2a0402b3d0 'YSIP Transfer'
data=(nil)
retval='(null)'
param['id'] = 'h323/1'
param['billid'] = '1429633433-1'
param['caller'] = '3xxxxxxxxxxxx4'
param['callername'] = '50.x.x.50'
param['called'] = '3000'
param['calledname'] = ''
param['diverter'] = '20000'
param['divertername'] = ''
param['reason'] = 'transfer'
param['sip_contact'] = '<sip:20000@50.x.x.30:5060>'
param['sip_user-agent'] = 'Asterisk PBX 1.8.32.2'
param['sip_refer-to'] = '<sip:3000@50.x.x.30:5065>'
param['sip_allow'] = 'INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE'
param['sip_supported'] = 'replaces, timer'
param['sip_referred-by'] = '<sip:20000@50.x.x.30:5060>'
2015-04-21_19:24:14.406008 <INFO> Routing call to '3000' in context 'default' via '-' in 417 usec
Returned true 'call.route' delay=0.001310
thread=0x7f2a0402b3d0 'YSIP Transfer'
data=(nil)
retval='-'
param['id'] = 'h323/1'
param['billid'] = '1429633433-1'
param['caller'] = '3xxxxxxxxxxxx4'
param['callername'] = '50.x.x.50'
param['called'] = '3000'
param['calledname'] = ''
param['diverter'] = '20000'
param['divertername'] = ''
param['reason'] = 'Protocol not allowed'
param['sip_contact'] = '<sip:20000@50.x.x.30:5060>'
param['sip_user-agent'] = 'Asterisk PBX 1.8.32.2'
param['sip_refer-to'] = '<sip:3000@50.x.x.30:5065>'
param['sip_allow'] = 'INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE'
param['sip_supported'] = 'replaces, timer'
param['sip_referred-by'] = '<sip:20000@50.x.x.30:5060>'
param['handlers'] = 'subscription:100,jingle:100,h323:100,regexroute:100'
param['formats'] = ''
param['error'] = 'forbidden'
2015-04-21_19:24:14.410481 <sip:INFO> 'udp:50.x.x.30:5065' sending code 603 0x7f29f4001c10 to 50.x.x.30:5060 [0x12bc560]
------
Thanks
I have setup yate as H323 To SIP Signalling Proxy between Avaya and Asterisk.
The proxy works fine until i try to transfer the call.
I am getting the below error as response from yate to the SIP REFER message of Asterisk .
<sip/5:STUB> initTransfer. Possible incomplete NOTIFY party creation [0x7f20a00230e0]
I have enabled transfer in ysipchan.conf
transfer=enable
and my routing in regexroute.conf is the below (example form official website)
${rtp_forward}possible=;rtp_forward=yes
${formats}^\([^,]*\)=;formats=\1
${module}^sip$=h323/${called}@50.x.x.50:1720
${module}^h323$=sip/sip:${called}@50.x.x.30:5060
.*=-;error=forbidden;reason=Protocol not allowed
I would appreciate some guidelines. Any additional configurations needed for yate in order to handle REFER message?
Below is the log with SIP REFER along with sniffed call.route
------
REFER sip:3xxxxxxxxxxxx4@50.x.x.30:5065 SIP/2.0
Via: SIP/2.0/UDP 50.x.x.30:5060;branch=z9hG4bK6ddcc84b;rport
Max-Forwards: 70
From: <sip:20000@50.x.x.30:5060>;tag=as1ca0dea3
To: "50.x.x.50" <sip:3xxxxxxxxxxxx4@50.x.x.30:5065>;tag=1643256148
Contact: <sip:20000@50.x.x.30:5060>
Call-ID: 829575849@50.x.x.30:5065
CSeq: 102 REFER
User-Agent: Asterisk PBX 1.8.32.2
Refer-To: <sip:3000@50.x.x.30:5065>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Referred-By: <sip:20000@50.x.x.30:5060>
Content-Length: 0
------
2015-04-21_19:24:14.404448 <sip:INFO> 'udp:50.x.x.30:5065' sending code 100 0x7f2a04021120 to 50.x.x.30:5060 [0x12bc560]
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 50.x.x.30:5060;branch=z9hG4bK6ddcc84b;rport=5060;received=50.x.x.30
From: <sip:20000@50.x.x.30:5060>;tag=as1ca0dea3
To: "50.x.x.50" <sip:3xxxxxxxxxxxx4@50.x.x.30:5065>;tag=1643256148
Call-ID: 829575849@50.x.x.30:5065
CSeq: 102 REFER
Server: YATE/5.4.3
Content-Length: 0
------
2015-04-21_19:24:14.404936 <sip/1:STUB> initTransfer. Possible incomplete NOTIFY party creation [0x7f2a04005760]
Sniffed 'call.route' time=1429633454.404720
thread=0x7f2a0402b3d0 'YSIP Transfer'
data=(nil)
retval='(null)'
param['id'] = 'h323/1'
param['billid'] = '1429633433-1'
param['caller'] = '3xxxxxxxxxxxx4'
param['callername'] = '50.x.x.50'
param['called'] = '3000'
param['calledname'] = ''
param['diverter'] = '20000'
param['divertername'] = ''
param['reason'] = 'transfer'
param['sip_contact'] = '<sip:20000@50.x.x.30:5060>'
param['sip_user-agent'] = 'Asterisk PBX 1.8.32.2'
param['sip_refer-to'] = '<sip:3000@50.x.x.30:5065>'
param['sip_allow'] = 'INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE'
param['sip_supported'] = 'replaces, timer'
param['sip_referred-by'] = '<sip:20000@50.x.x.30:5060>'
2015-04-21_19:24:14.406008 <INFO> Routing call to '3000' in context 'default' via '-' in 417 usec
Returned true 'call.route' delay=0.001310
thread=0x7f2a0402b3d0 'YSIP Transfer'
data=(nil)
retval='-'
param['id'] = 'h323/1'
param['billid'] = '1429633433-1'
param['caller'] = '3xxxxxxxxxxxx4'
param['callername'] = '50.x.x.50'
param['called'] = '3000'
param['calledname'] = ''
param['diverter'] = '20000'
param['divertername'] = ''
param['reason'] = 'Protocol not allowed'
param['sip_contact'] = '<sip:20000@50.x.x.30:5060>'
param['sip_user-agent'] = 'Asterisk PBX 1.8.32.2'
param['sip_refer-to'] = '<sip:3000@50.x.x.30:5065>'
param['sip_allow'] = 'INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE'
param['sip_supported'] = 'replaces, timer'
param['sip_referred-by'] = '<sip:20000@50.x.x.30:5060>'
param['handlers'] = 'subscription:100,jingle:100,h323:100,regexroute:100'
param['formats'] = ''
param['error'] = 'forbidden'
2015-04-21_19:24:14.410481 <sip:INFO> 'udp:50.x.x.30:5065' sending code 603 0x7f29f4001c10 to 50.x.x.30:5060 [0x12bc560]
------
Thanks