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@MemberConnector-00001c4d;1 is ringing
-- SIP/voip-trunk-00003a10 is ringing
-- SIP/voip-trunk-00003a10 answered
Local/SIP-5002@MemberConnector-00001c4d;2
-- Local/SIP-5002@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@MemberConnector-00001c4d;1 joined
'simple_bridge' basic-bridge <e7f93c94-5928-4454-9cdf-5ef7811d7eb5>
-- Channel Local/SIP-5002@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@MemberConnector-00001c4d;2'
-- Stopped music on hold on Local/SIP-5002@MemberConnector-00001c4d;2
-- Channel Local/SIP-5002@MemberConnector-00001c4d;2 left
'simple_bridge' basic-bridge <69a9abf3-3f1a-4efc-a069-3531622b6294>
-- Executing [5009@from-voip:1]
Set("Local/SIP-5002@MemberConnector-00001c4d;2", "DID=5009") in new stack
-- Executing [5009@from-voip:2]
Goto("Local/SIP-5002@MemberConnector-00001c4d;2", "s,1") in new stack
-- Goto (from-voip,s,1)
-- Executing [s@from-voip:1]
NoOp("Local/SIP-5002@MemberConnector-00001c4d;2", ""from-voip: 2169"")
in new stack
-- Executing [s@from-voip:2]
Set("Local/SIP-5002@MemberConnector-00001c4d;2", "CHANNEL(language)=it")
in new stack
-- Executing [s@from-voip:3]
Hangup("Local/SIP-5002@MemberConnector-00001c4d;2", "") in new stack
== Spawn extension (from-voip, s, 3) exited non-zero on
'Local/SIP-5002@MemberConnector-00001c4d;2'
-- Channel SIP/voip-trunk-00003a10 left 'simple_bridge' basic-bridge
<69a9abf3-3f1a-4efc-a069-3531622b6294>
-- Channel Local/SIP-5002@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:[email protected]
/usr/sbin/opensips[27360]: d59a893ecb225520 - Route RELAY ACK To: 5000,
From: 2169, RURI: sip:[email protected]:5060
/usr/sbin/opensips[27362]: User net group is 1
[voip.unisi.it/sip:[email protected]/voip.unisi.it/sip:[email protected]/172.20.1.5]
/usr/sbin/opensips[27362]: Context for 5002 is voip
/usr/sbin/opensips[27362]:
[email protected] - Route RELAY INVITE To:
5002, From: 2169, RURI: sip:[email protected]:37496
/usr/sbin/opensips[27362]:
[email protected] - Route NEW BRANCH To:
5002, From: 2169, RURI: sip:[email protected]:37496
/usr/sbin/opensips[27360]:
[email protected] - Route RELAY ACK To:
5002, From: 2169, RURI: sip:[email protected]:37496
/usr/sbin/opensips[27359]: User net group is 1
[voip.unisi.it/sip:[email protected]/172.20.1.5/sip:[email protected]:5060/172.20.1.47]
/usr/sbin/opensips[27359]:
[email protected] - Route RELAY INVITE To:
2169, From: 5002, RURI: sip:[email protected]:5060
/usr/sbin/opensips[27359]:
[email protected] - Route NEW BRANCH To:
2169, From: 5002, RURI: sip:[email protected]:5060
/usr/sbin/opensips[27359]:
[email protected] - Route RELAY ACK To:
2169, From: 5002, RURI: sip:[email protected]:5060
/usr/sbin/opensips[27361]:
[email protected] - Route RELAY ACK To:
2169, From: 5002, RURI: sip:[email protected]:5060
/usr/sbin/opensips[27362]:
[email protected] - Route RELAY REFER To:
2169, From: 5002, RURI: sip:[email protected]:5060
/usr/sbin/opensips[27359]:
[email protected] - Route RELAY NOTIFY To:
5002, From: 2169, RURI: sip:[email protected]:37496
/usr/sbin/opensips[27361]: d59a893ecb225520 - Route RELAY BYE To: 2169,
From: 5000, RURI: sip:[email protected]:5060
/usr/sbin/opensips[27362]:
[email protected] - Route RELAY NOTIFY To:
5002, From: 2169, RURI: sip:[email protected]:37496
/usr/sbin/opensips[27361]: 3134353333303331343237333731-iqp91rwhq7h6 -
Route RELAY NOTIFY To: 5009, From: 5002, RURI: sip:[email protected]:32768
/usr/sbin/opensips[27361]:
[email protected] - Route RELAY BYE To:
2169, From: 5002, RURI: sip:[email protected]:5060
Thanks for any help or suggestion !
Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - [email protected]
Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di
Ateneo, http://www.faq.unisi.it
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