Call queue transfer issue

[Gianmarco]

Hi all,
I installed a SipXcom version 15.06 using Polycom SoundpointIP 650 phones running firmware 4.0.4.
I try to configure a call queue with a transfer when no agent is logged in. Everything works, if the transfer is an internal number, but if I transfer the call from the queue to an external number the call hang up .

Below the log of the call taken from freeswitch.log, 203 is the caller, 001131xxxxx is the external number transferred from the queue

 [NOTICE] switch_channel.c:1055 New Channel sofia/sipx.mydomain.it/2...@sipx.mydomain.it [88de80fa-2967-11e5-8559-8306f26faa14]
 [NOTICE] sofia_media.c:92 Pre-Answer sofia/sipx.mydomain.it/2...@sipx.mydomain.it!
 [NOTICE] mod_dptools.c:1258 Channel [sofia/sipx.mydomain.it/2...@sipx.mydomain.it] has been answered
 [NOTICE] switch_channel.c:1055 New Channel sofia/sipx.mydomain.it/00113...@sipx.mydomain.it [9502a08c-2967-11e5-8564-8306f26faa14]
 [NOTICE] sofia.c:7530 Hangup sofia/sipx.mydomain.it/00113...@sipx.mydomain.it [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
 [NOTICE] switch_channel.c:4724 Hangup sofia/sipx.mydomain.it/2...@sipx.mydomain.it [CS_EXECUTE] [NO_ROUTE_DESTINATION]
 [NOTICE] switch_core_session.c:1633 Session 968 (sofia/sipx.mydomain.it/00113...@sipx.mydomain.it) Ended
 [NOTICE] switch_core_session.c:1637 Close Channel sofia/sipx.mydomain.it/00113...@sipx.mydomain.it [CS_DESTROY]
 [NOTICE] switch_core_session.c:1633 Session 967 (sofia/sipx.mydomain.it/2...@sipx.mydomain.it) Ended
 [NOTICE] switch_core_session.c:1637 Close Channel sofia/sipx.mydomain.it/2...@sipx.mydomain.it [CS_DESTROY]

Any ideas?

[George Niculae]

Can you try directly calling that number and see if working? Could be dial plan related...

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[Nathaniel Watkins]

You may need to setup a phantom user.  Have the phantom setup to do call forwarding to the external number.  Then have the call queue send unanswered calls to the phantom.  I don't think you can send calls directly to an outside number (I've had issues on an AA, etc.).

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[Gianmarco]

Hi George,
of course, I can call directly the number, I don't think it's a problem related to dial plan.

[Mihai Costache]

Hi

Do you have any permissions set up on that route to  001131xxxxx. If so you need to do what Nathaniel told you.

Since freeswitch is not a real user it does not have any permissions to access the matching rule from dial plan
 

Best Regards,

Mihai Costache

"No problem can withstand the assault of sustained thinking. "

                                -  Voltaire

To view this discussion on the web visit https://groups.google.com/d/msgid/sipxcom-users/82bc07eb-782a-4a0c-8b5b-c41f6c2410b4%40googlegroups.com.

[Gianmarco]

Hi Nathaniel,
I still tried to setup a phantom user with a call forwarding to the external number but it doesn't work. The call hang up silently...

Below the log of the call taken from freeswitch.log, 203 is the caller, 218 is the phantom with the call forwarding. In this case it seems there's nothing strange inside freeswitch logs.
The same works if i call forward to internal number.

 [NOTICE] switch_channel.c:1055 New Channel sofia/mydomain.it/2...@mydomain.it [4847902e-29f6-11e5-859b-8306f26faa14]
 [NOTICE] sofia_media.c:92 Pre-Answer sofia/mydomain.it/2...@mydomain.it!
 [NOTICE] mod_dptools.c:1258 Channel [sofia/mydomain.it/2...@mydomain.it] has been answered
 [NOTICE] switch_channel.c:1055 New Channel sofia/mydomain.it/2...@mydomain.it [54b9a874-29f6-11e5-85a6-8306f26faa14]
 [NOTICE] switch_channel.c:1055 New Channel sofia/mydomain.it/2...@x.x.x.x [54d6e312-29f6-11e5-85aa-8306f26faa14]
 [NOTICE] switch_core_state_machine.c:315 sofia/mydomain.it/2...@x.x.x.x has executed the last dialplan instruction, hanging up.
 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/mydomain.it/2...@x.x.x.x [CS_EXECUTE] [NORMAL_CLEARING]
 [NOTICE] switch_core_session.c:1633 Session 978 (sofia/mydomain.it/2...@x.x.x.x) Ended
 [NOTICE] switch_core_session.c:1637 Close Channel sofia/mydomain.it/2...@x.x.x.x [CS_DESTROY]
 [NOTICE] sofia.c:7530 Hangup sofia/mydomain.it/2...@mydomain.it [CS_CONSUME_MEDIA] [NORMAL_CLEARING]
 [NOTICE] switch_channel.c:4724 Hangup sofia/mydomain.it/2...@mydomain.it [CS_EXECUTE] [NORMAL_CLEARING]
 [NOTICE] switch_core_session.c:1633 Session 977 (sofia/mydomain.it/2...@mydomain.it) Ended
 [NOTICE] switch_core_session.c:1637 Close Channel sofia/mydomain.it/2...@mydomain.it [CS_DESTROY]
 [NOTICE] switch_core_session.c:1633 Session 976 (sofia/mydomain.it/2...@mydomain.it) Ended
 [NOTICE] switch_core_session.c:1637 Close Channel sofia/mydomain.it/2...@mydomain.it [CS_DESTROY]

[Mihai Costache]

Proxy and Registrar logs could help here

Best Regards,

Mihai Costache

"No problem can withstand the assault of sustained thinking. "

                                -  Voltaire

To view this discussion on the web visit https://groups.google.com/d/msgid/sipxcom-users/1c7b3987-3450-4abd-9d15-c3e823b33ebc%40googlegroups.com.

[Mihai Costache]

Hi Gianmarco,

This is the problem. looks like audiocodes could not find a route to destination. if we could see audiocodes logs we can find more :)

"2015-07-14T10:39:08.948750Z":8356:INCOMING:INFO:vm-sipx01.mydomain.it:SipClientUdp-11:7f056a6d0700:sipxproxy:"Read SIP message:
----Local Host:10.102.227.45---- Port: 5060----
----Remote Host:158.102.16.12---- Port: 5060----
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.102.227.45;branch=z9hG4bK-XX-2170_iMN376`M7vcs7eH7ZPUdA;sipxecs-lineid=1
Via: SIP/2.0/UDP 10.102.227.45;branch=z9hG4bK-XX-216dKHMCIgoygN4f9Rh8bkgJjg~8VCaOBYKVRzO1trYTELurg;id=16572-114
Via: SIP/2.0/UDP 10.102.227.45;branch=z9hG4bK-XX-2168GwddEw_jrl1rIYih2YJfYg~bU9l1aZzkph3XMcMoU2KhA
Via: SIP/2.0/UDP 10.102.227.45:15060;rport=15060;branch=z9hG4bK7UFUK8482ve8e
From: \"User1\" <sip:2...@10.102.227.45>;tag=K46Z0657mjQyF
To: <sip:2...@mydomain.it>;tag=1c970278437
Call-ID: 6623def0-a4b7-1233-efb9-0050569f0e2f
CSeq: 78099111 INVITE
Record-Route: <sip:10.102.227.45:5060;lr;sipXecs-CallDest=AL%2CLD;sipXecs-rs=%2Aauth%7E.%2Afrom%7ESzQ2WjA2NTdtalF5Rg%60%60.900_ntap%2Aid%7EMTY1NzItMTE0%21641010792b6e0e5c4beb8b4e576a0dd5>
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000 - MSBG/v.6.40A.019.008
Reason: Q.850 ;cause=3 ;text=\"local\"
Content-Length: 0

Is that 158.102.16.12 the correct IPaddress where calls should be sent?

Noticed this too: -- like your gateway is not correctly resolved by dns srv's

"2015-07-14T10:39:08.925456Z":8192:SIP:DEBUG:vm-sipx01.mydomain.it:SipSrvLookupThread-17:7f0563701700:sipxproxy:"SipSrvLookup::res_query_and_parse out_name = '_sip._tcp.gw.csi.it', out_response = (nil)"
"2015-07-14T10:39:08.925790Z":8193:SIP:DEBUG:vm-sipx01.mydomain.it:SipSrvLookupThread-19:7f05634ff700:sipxproxy:"DNS query for name 'gw.csi.it', type = 1 (A): returned error"
"2015-07-14T10:39:08.925824Z":8194:SIP:DEBUG:vm-sipx01.mydomain.it:SipSrvLookupThread-19:7f05634ff700:sipxproxy:"SipSrvLookup::res_query_and_parse out_name = 'gw.csi.it', out_response = (nil)"
"2015-07-14T10:39:08.926342Z":8195:SIP:DEBUG:vm-sipx01.mydomain.it:SipSrvLookupThread-16:7f0563802700:sipxproxy:"SipSrvLookup::res_query_and_parse out_name = '_sip._udp.gw.csi.it', out_response = 0x2750bd0"
"2015-07-14T10:39:08.926378Z":8196:SIP:DEBUG:vm-sipx01.mydomain.it:SipSrvLookupThread-16:7f0563802700:sipxproxy:"SipSrvLookup::res_query_and_parse in_name = 'gwsip-csi02.gw.csi.it', type = 1 (A)"
"2015-07-14T10:39:08.926395Z":8197:SIP:DEBUG:vm-sipx01.mydomain.it:SipSrvLookupThread-16:7f0563802700:sipxproxy:"SipSrvLookup::res_query_and_parse out_name = 'gwsip-csi02.gw.csi.it', out_response = 0x2750bd0"
"2015-07-14T10:39:08.926411Z":8198:SIP:DEBUG:vm-sipx01.mydomain.it:SipSrvLookupThread-16:7f0563802700:sipxproxy:"SipSrvLookup::res_query_and_parse in_name = 'gwsip-csi03.gw.csi.it', type = 1 (A)"
"2015-07-14T10:39:08.926435Z":8199:SIP:DEBUG:vm-sipx01.mydomain.it:SipSrvLookupThread-16:7f0563802700:sipxproxy:"SipSrvLookup::res_query_and_parse out_name = 'gwsip-csi03.gw.csi.it', out_response = 0x2750bd0"
"2015-07-14T10:39:08.926459Z":8200:SIP:DEBUG:vm-sipx01.mydomain.it:SipSrvLookupThread-16:7f0563802700:sipxproxy:"SipSrvLookup::res_query_and_parse in_name = 'gwsip-csi01.gw.csi.it', type = 1 (A)"
"2015-07-14T10:39:08.926482Z":8201:SIP:DEBUG:vm-sipx01.mydomain.it:SipSrvLookupThread-16:7f0563802700:sipxproxy:"SipSrvLookup::res_query_and_parse out_name = 'gwsip-csi01.gw.csi.it', out_response = 0x2750bd0"
"2015-07-14T10:39:08.926599Z":8202:SIP:DEBUG:vm-sipx01.mydomain.it::7f0569214700:sipxproxy:"SipTransaction::recurseDnsSrvChildrenTimer C transaction 0x31d0000 setting timeout 300 secs."
"2015-07-14T10:39:08.926666Z":8203:SIP:DEBUG:vm-sipx01.mydomain.it::7f0569214700:sipxproxy:"SipTransaction::recurseDnsSrvChildrenExpire transaction 0x31d0000 setting timeout 7200 secs."
"2015-07-14T10:39:08.926930Z":8204:SIP:INFO:vm-sipx01.mydomain.it::7f0569214700:sipxproxy:"SipProtocolServerBase::findExistingClientForDestination found good flow SipClientUdp-1842 for target 158.102.16.12:5060"
 

Best Regards,

Mihai Costache

"No problem can withstand the assault of sustained thinking. "

                                -  Voltaire

On Tue, Jul 14, 2015 at 2:25 PM, Gianmarco <gianmarco...@gmail.com> wrote:

Hi Mihai,
Notice and INFO log level doesn't give me relevant result so, I logged with Debug level. The queue (204) transfer the call to ghost user (218), 218 has a transfer to an external number(00113xxxxx) but the call goes to 218 Voicemail even if the external number is reacheable.
The logs are in attached.

Best Regards

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[Gianmarco]

Hi Mihai,
Notice and INFO log level doesn't give me relevant result so, I logged with Debug level. The queue (204) transfer the call to ghost user (218), 218 has a transfer to an external number(00113xxxxx) but the call goes to 218 Voicemail even if the external number is reacheable.
The logs are in attached.

Best Regards

Il giorno martedì 14 luglio 2015 09:32:33 UTC+2, Mihai Costache ha scritto:

[Nathaniel Watkins]

Go to System-> Media Services -> click on server -> Make sure 'Allow Blind Transfer' is checked.

To view this discussion on the web visit https://groups.google.com/d/msgid/sipxcom-users/7a0f24b3-5cd6-4a2a-8863-c21a878509b7%40googlegroups.com.

[Gianmarco]

Hi Nathaniel,
I checked Allow Blind Transfer but nothing appens.
The strange is that if I call directly the ghost number the tranfer works, it doesn't work if the transfer come from the queue.

[George Niculae]

You may want to uncheck Simplify call transfer option in System-> Media Services -> click on server and try again

[Gianmarco]

Hi George,
it doesn't works... the system has the same previous behaviour.
Almost always the call is transfered to ghost number voicemail but sometimes hung up... but i can't reproduce this behaviour.
I attached the freeswitch log, one file is for the hang up, the other is referred to tranfer to voicemail.

[Gianmarco]

Hi Mihai,
the audiocodes works correctly and i can't see anything wrong on logs... i can call to external number  and external number can call to internal, i also tried tranfer ato internal / external and evrerything works well...

Il giorno martedì 14 luglio 2015 13:53:39 UTC+2, Mihai Costache ha scritto:

Hi Gianmarco,

This is the problem. looks like audiocodes could not find a route to destination. if we could see audiocodes logs we can find more :)

"2015-07-14T10:39:08.948750Z":8356:INCOMING:INFO:vm-sipx01.mydomain.it:SipClientUdp-11:7f056a6d0700:sipxproxy:"Read SIP message:
----Local Host:10.102.227.45---- Port: 5060----
----Remote Host:158.102.16.12---- Port: 5060----
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.102.227.45;branch=z9hG4bK-XX-2170_iMN376`M7vcs7eH7ZPUdA;sipxecs-lineid=1
Via: SIP/2.0/UDP 10.102.227.45;branch=z9hG4bK-XX-216dKHMCIgoygN4f9Rh8bkgJjg~8VCaOBYKVRzO1trYTELurg;id=16572-114
Via: SIP/2.0/UDP 10.102.227.45;branch=z9hG4bK-XX-2168GwddEw_jrl1rIYih2YJfYg~bU9l1aZzkph3XMcMoU2KhA
Via: SIP/2.0/UDP 10.102.227.45:15060;rport=15060;branch=z9hG4bK7UFUK8482ve8e

From: \"User1\" <sip...@10.102.227.45>;tag=K46Z0657mjQyF
To: <sip...@mydomain.it>;tag=1c970278437

[Mihai Costache]

Can you please send that audiocodes logs (edited)

Did you see this message in audicodes ?

SIP/2.0 404 Not Found
......
.........
..........
..............

Server: Audiocodes-Sip-Gateway-Mediant 1000 - MSBG/v.6.40A.019.008
Reason: Q.850 ;cause=3 ;text=\"local\"

From here

http://cnes.com/causecodes.html

cause 3 = Cause No. 3 - No route to destination [Q.850]
This cause indicates that the called party cannot be reached because the network through which the call has been routed does not serve the destination desired. This cause is supported on a network dependent basis.

Best Regards,

Mihai Costache

"No problem can withstand the assault of sustained thinking. "

                                -  Voltaire

To view this discussion on the web visit https://groups.google.com/d/msgid/sipxcom-users/6675e132-1e17-41e4-856b-04a39c326512%40googlegroups.com.

[Gianmarco]

Hi Mihai,
I think, I've found the problem. Actually the problem is the configuration of my Audiocodes, you're right! :-)
My Audiocodes is setup to permit only to packets from mydomain.it to exit on public network.
I configured them in Routing -> IP to Trunk Group Routing-> Source Host Prefix, Audiocodes check it on the SIP "From uri" field.

When Freeswitch try to contact the gateway replaces mydomain.it with the IP address of the sipx server in the From field.

f./:.......INVITE sip:2...@sipx.mydomain.it SIP/2.0
Via: SIP/2.0/UDP 10.102.227.45:15060;rport;branch=z9hG4bK8Fca6pZ5656Fr
Max-Forwards: 15
From: "User1" <sip:2...@10.102.227.45>;tag=vyU2FaHZt8FFg
To: <sip:2...@sipx.mydomain.it>
Call-ID: 0f38d867-a59e-1233-efb9-0050569f0e2f
CSeq: 78148644 INVITE
Contact: <sip:mod_...@10.102.227.45:15060>
User-Agent: FreeSWITCH-mod_sofia/1.4.15~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 329
X-Sipx-Authidentity: <sip:51...@sipx.mydomain.it;signature=55A669B3%3Ade46c60d16bf584c7bc629a066898951>
X-Sipx-Handled: X10.102.227.45-10.102.227.45
X-FS-Support: update_display,send_info
Remote-Party-ID: "User1" <sip:2...@10.102.227.45>;party=calling;screen=yes;privacy=off

Is it possible to change the behavior of freeswitch?

In attachment the tcpdump of the call catched from sipx server and the error given from Audiocodes.

[Mihai Costache]

Let's wait George's answer on this one

"Is it possible to change the behavior of freeswitch?"

Best Regards,

Mihai Costache

"No problem can withstand the assault of sustained thinking. "

                                -  Voltaire

To view this discussion on the web visit https://groups.google.com/d/msgid/sipxcom-users/0a7ae42d-d6ab-4371-917c-544711a68069%40googlegroups.com.

[George Niculae]

I found it hard to believe FS changing it, how do you register phones, using SIP domain or IP? Please attach a pcap (I cannot open the one you attached with wireshark) and explain the entire call flow (who called who)

George

[George Niculae]

Paste also output for:

fs_cli

sofia status

George