thr3ads.net - asterisk users - [asterisk-users] Call hangup on transfer when originated from a Queue [Feb 2016]

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Michele Pinassi
2016-Feb-04 08:26 UTC
[asterisk-users] Call hangup on transfer when originated from a Queue

Hi all,

i'm writing because going crazy on this issue i'm unable to solve. My
VoIP system is based on OpenSIPS router that forward calls to an
Asterisk BOX to have IVR and Queue services.

If a call was directed to a queue and operator answer, on transfer to
another ext. the call hangup.

On Asterisk console (debug=100, verbose=100. Called ext was 5002) i got:

-- Called SIP/voip-trunk/5002
    -- SIP/voip-trunk-00003a10 is ringing
    -- Local/SIP-5002 at MemberConnector-00001c4d;1 is ringing
    -- SIP/voip-trunk-00003a10 is ringing
    -- SIP/voip-trunk-00003a10 answered
Local/SIP-5002 at MemberConnector-00001c4d;2
    -- Local/SIP-5002 at MemberConnector-00001c4d;1 answered
SIP/voip-trunk-00003a0f
    -- Stopped music on hold on SIP/voip-trunk-00003a0f
    -- Channel SIP/voip-trunk-00003a10 joined 'simple_bridge'
basic-bridge <69a9abf3-3f1a-4efc-a069-3531622b6294>
    -- Channel Local/SIP-5002 at MemberConnector-00001c4d;1 joined
'simple_bridge' basic-bridge
<e7f93c94-5928-4454-9cdf-5ef7811d7eb5>
    -- Channel Local/SIP-5002 at MemberConnector-00001c4d;2 joined
'simple_bridge' basic-bridge
<69a9abf3-3f1a-4efc-a069-3531622b6294>
    -- Channel SIP/voip-trunk-00003a0f joined 'simple_bridge'
basic-bridge <e7f93c94-5928-4454-9cdf-5ef7811d7eb5>
       > 0x97f2cf8 -- Probation passed - setting RTP source address to
172.20.1.47:62070
    -- Started music on hold, class 'default', on channel
'Local/SIP-5002 at MemberConnector-00001c4d;2'
    -- Stopped music on hold on Local/SIP-5002 at MemberConnector-00001c4d;2
    -- Channel Local/SIP-5002 at MemberConnector-00001c4d;2 left
'simple_bridge' basic-bridge
<69a9abf3-3f1a-4efc-a069-3531622b6294>
    -- Executing [5009 at from-voip:1]
Set("Local/SIP-5002 at MemberConnector-00001c4d;2",
"DID=5009") in new stack
    -- Executing [5009 at from-voip:2]
Goto("Local/SIP-5002 at MemberConnector-00001c4d;2", "s,1")
in new stack
    -- Goto (from-voip,s,1)
    -- Executing [s at from-voip:1]
NoOp("Local/SIP-5002 at MemberConnector-00001c4d;2",
""from-voip: 2169"")
in new stack
    -- Executing [s at from-voip:2]
Set("Local/SIP-5002 at MemberConnector-00001c4d;2",
"CHANNEL(language)=it")
in new stack
    -- Executing [s at from-voip:3]
Hangup("Local/SIP-5002 at MemberConnector-00001c4d;2", "")
in new stack
  == Spawn extension (from-voip, s, 3) exited non-zero on
'Local/SIP-5002 at MemberConnector-00001c4d;2'
    -- Channel SIP/voip-trunk-00003a10 left 'simple_bridge' basic-bridge
<69a9abf3-3f1a-4efc-a069-3531622b6294>
    -- Channel Local/SIP-5002 at MemberConnector-00001c4d;1 left
'simple_bridge' basic-bridge
<e7f93c94-5928-4454-9cdf-5ef7811d7eb5>
    -- Channel SIP/voip-trunk-00003a0f left 'simple_bridge' basic-bridge
<e7f93c94-5928-4454-9cdf-5ef7811d7eb5>
  == Spawn extension (macro-service-phone-operator, s, 4) exited
non-zero on 'SIP/voip-trunk-00003a0f' in macro
'service-phone-operator'
  == Spawn extension (ivr-services-phone, 9, 1) exited non-zero on
'SIP/voip-trunk-00003a0f'

and the call simply hangup. Just to clarify call flow, i'm calling from
2169 to Queue "service-phone-operator" that ring on 5002. Later i try
to
transfer call to ext. 5009.

Asterisk box have IP 172.20.1.5 and OpenSIPS router is voip.unisi.it.

On OpenSIPS side i got:

/usr/sbin/opensips[27359]: Forwarding call to IVR_5000
/usr/sbin/opensips[27359]: d59a893ecb225520 - Route MEDIABOX To: 5000,
From: 2169, RURI: sip:IVR_5000 at voip.unisi.it
/usr/sbin/opensips[27360]: d59a893ecb225520 - Route RELAY ACK To: 5000,
From: 2169, RURI: sip:IVR_5000 at 172.20.1.5:5060
/usr/sbin/opensips[27362]: User net group is 1
[voip.unisi.it/sip:2169 at voip.unisi.it/voip.unisi.it/sip:5002 at
voip.unisi.it/172.20.1.5]
/usr/sbin/opensips[27362]: Context for 5002 is voip
/usr/sbin/opensips[27362]:
4796fc0e1fca050c0367076b49ec17bb at voip.unisi.it - Route RELAY INVITE To:
5002, From: 2169, RURI: sip:5002 at 172.20.1.47:37496
/usr/sbin/opensips[27362]:
4796fc0e1fca050c0367076b49ec17bb at voip.unisi.it - Route NEW BRANCH To:
5002, From: 2169, RURI: sip:5002 at 172.20.1.47:37496
/usr/sbin/opensips[27360]:
4796fc0e1fca050c0367076b49ec17bb at voip.unisi.it - Route RELAY ACK To:
5002, From: 2169, RURI: sip:5002 at 172.20.1.47:37496
/usr/sbin/opensips[27359]: User net group is 1
[voip.unisi.it/sip:5002 at voip.unisi.it/172.20.1.5/sip:2169 at
172.20.1.5:5060/172.20.1.47]
/usr/sbin/opensips[27359]:
4796fc0e1fca050c0367076b49ec17bb at voip.unisi.it - Route RELAY INVITE To:
2169, From: 5002, RURI: sip:2169 at 172.20.1.5:5060
/usr/sbin/opensips[27359]:
4796fc0e1fca050c0367076b49ec17bb at voip.unisi.it - Route NEW BRANCH To:
2169, From: 5002, RURI: sip:2169 at 172.20.1.5:5060
/usr/sbin/opensips[27359]:
4796fc0e1fca050c0367076b49ec17bb at voip.unisi.it - Route RELAY ACK To:
2169, From: 5002, RURI: sip:2169 at 172.20.1.5:5060
/usr/sbin/opensips[27361]:
4796fc0e1fca050c0367076b49ec17bb at voip.unisi.it - Route RELAY ACK To:
2169, From: 5002, RURI: sip:2169 at 172.20.1.5:5060
/usr/sbin/opensips[27362]:
4796fc0e1fca050c0367076b49ec17bb at voip.unisi.it - Route RELAY REFER To:
2169, From: 5002, RURI: sip:2169 at 172.20.1.5:5060
/usr/sbin/opensips[27359]:
4796fc0e1fca050c0367076b49ec17bb at voip.unisi.it - Route RELAY NOTIFY To:
5002, From: 2169, RURI: sip:5002 at 172.20.1.47:37496
/usr/sbin/opensips[27361]: d59a893ecb225520 - Route RELAY BYE To: 2169,
From: 5000, RURI: sip:2169 at 172.20.1.4:5060
/usr/sbin/opensips[27362]:
4796fc0e1fca050c0367076b49ec17bb at voip.unisi.it - Route RELAY NOTIFY To:
5002, From: 2169, RURI: sip:5002 at 172.20.1.47:37496
/usr/sbin/opensips[27361]: 3134353333303331343237333731-iqp91rwhq7h6 -
Route RELAY NOTIFY To: 5009, From: 5002, RURI: sip:5009 at 172.20.1.215:32768
/usr/sbin/opensips[27361]:
4796fc0e1fca050c0367076b49ec17bb at voip.unisi.it - Route RELAY BYE To:
2169, From: 5002, RURI: sip:2169 at 172.20.1.5:5060

Thanks for any help or suggestion !

Michele

-- 
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Universit? degli Studi di Siena
tel: 0577.(23)5000 - centralino at unisi.it

Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di
Ateneo, http://www.faq.unisi.it
asterisk users - Feb 2016 - Call hangup on transfer when originated from a Queue

[asterisk-users] Call hangup on transfer when originated from a Queue
