[Solved] Incomming Calls will not be processed

[remosti]

Hello Everybody

I need your help. I am setting up a FreePBX-Installation and it works fine, but i dont get the incomming calls to work. I tried every combiination, i could find in the internet or i had in mind, but without success.

What i found ou till now is the following:

1. I can see the incomming call on the Firewall
2. The Call gets forwarded to the FreePBX, so i can see it on the PBX-VM
3. I can see the call come in in the Asterisk CLI in Debug-Mode, but there is no prompt, that the call will be processed by any Dialplan Rule

In the following, there are the TCPDump output from the Firewall and the PBX-VM and also the Debug-Output from Asterisk.
TCPDump-File Firewall:

INVITE sip:4141511****@85.14.240.152:5060;transport=UD\000\000\021\000\000\000\330w\034\010xq\034\010\000\000\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000xq\034\010`z!\010files\000\000\000\000\000\000\000\021\000\000\000\006\000\000\000`\000\000\000\000\000\000\000\021\000\000\000u\312e\267,tS\267\000\000\000\000\021\000\000\000\210z\034\010Xz\034\010rpc\000\031\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000!\000\000\000/lib/libnss_files.so.2\000\000\000\000\000\000\021\000\000\000py\034\010\360{\034\010\230z!\010\201\002\000\000\000j\027\312\000q\342\312\000L\367\313\000S\302\314\220\353#\025\220\334\023\026\220\315\003\027\220\276\363\027\220\257\343\030\220\240\323\031\220\221\303\032\020\275\274\033\020\256\254\034\020\237\234\035\020\220\214\036\020\201|\037\020rl \020c\!\020TL"\020E<#\0206,$\020'\034%\020\030\014&\220C\005'\2204\365'\220%\345(\220\026\325)\220\007\305*\220\370\264+\220\351\244,\220\332\224-\220\313\204.\220\274t/\220\255d0\020\331]1\020\264r2\020\273=3\020\226R4\020\235\0355\020x26\020\177\3756\220\224\0338\020a\3358\220v\3739\020C\275:\220X\333;\220_\246<\220:\273=\220A\206>\220\034\233?\220#f@\0209\204A\220\005FB\020\033dC\220\347%D\020\375CE\220\311\005F\020\337#G\020\346\356G\020\301\003I\020\310\316I\020\243\343J\020\252\256K\220\277\314L\020\214\216M\220\241\254N\020nnO\220\203\214P\220\212WQ\220elR\220l7S\220GLT\220N\027U\220),V\2200\367V\020F\025X\220\022\327X\020(\365Y\220\364\266Z\020\012\325[\020\021\240\\020\354\264]\020\363\177^\020\316\224_\020\325_`\220\352}a\020\267?b\220\314]c\020\231\037d\220\256=e\220\265\010f\220\220\035g\220\227\350g\220r\375h\220y\310i\220T\335j\220[\250k\020q\306l\220=\210m\020S\246n\220\037ho\0205\206p\020<Qq\020\027fr\020\0361s\020\371Et\020\000\021u\220\025/v\020\342\360v\220\367\016x\020\304\320x\220\331\356y\020\246\260z\220\273\316{\220\302\231|\220\235\256}\220\244y~\220\177\216\177\000\001\000\001\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\002\003\000\000 \034\000\000\001\000\000\000\020\016\000\000\000\005\000\000 \034\000\000\001\000\001\001\020\016\000\000\000\005\001\001CEST\000CET\000\000\000\000\351\001\000\000\003\000\000\000\003\000\000\000\000\000\000\000`\000\000\000\001\000\000\000\000\000\000\000\002\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\267\021\000\000\000\000\000\000\000\000\000\000\001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000Hp\034\010\000\000\000\000\350\003\000\000\000\000\000\000\000\000\000\000\004\000\000\000\001\000\000\000\000\000\000\000`\000\000\000hp\034\010\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000p,\011\010\000/\011\010\340A\011\010\2600\011\010\000\000\000\000\340N\011\0100N\011\010\260/\011\010\0005\011\010\001\000\000\000Hv!\010\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000
ACK sip:4141511****@85.14.240.152:5060;transport=UDP S\000\000\021\000\000\000\330w\034\010xq\034\010\000\000\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000xq\034\010`z!\010files\000\000\000\000\000\000\000\021\000\000\000\006\000\000\000`\000\000\000\000\000\000\000\021\000\000\000u\312e\267,tS\267\000\000\000\000\021\000\000\000\210z\034\010Xz\034\010rpc\000\031\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000!\000\000\000/lib/libnss_files.so.2\000\000\000\000\000\000\021\000\000\000py\034\010\360{\034\010\230z!\010\201\002\000\000\000j\027\312\000q\342\312\000L\367\313\000S\302\314\220\353#\025\220\334\023\026\220\315\003\027\220\276\363\027\220\257\343\030\220\240\323\031\220\221\303\032\020\275\274\033\020\256\254\034\020\237\234\035\020\220\214\036\020\201|\037\020rl \020c\!\020TL"\020E<#\0206,$\020'\034%\020\030\014&\220C\005'\2204\365'\220%\345(\220\026\325)\220\007\305*\220\370\264+\220\351\244,\220\332\224-\220\313\204.\220\274t/\220\255d0\020\331]1\020\264r2\020\273=3\020\226R4\020\235\0355\020x
Via: SIP/2.0/UDP 85.14.240.152:5060;br\000\000\021\000\000\000\330w\034\010xq\034\010\000\000\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000xq\034\010`z!\010files\000\000\000\000\000\000\000\021\000\000\000\006\000\000\000`\000\000\000\000\000\000\000\021\000\000\000u\312e\267,tS\267\000\000\000\000\021\000\000\000\210z\034\010Xz\034\010rpc\000\031\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000!\000\000\000/lib/libnss_files.so.2\000\000\000\000\000\000\021\000\000\000py\034\010\360{\034\010\230z!\010\201\002\000\000\000j\027\312\000q\342\312\000L\367\313\000S\302\314\220\353#\025\220\334\023\026\220\315\003\027\220\276\363\027\220\257\343\030\220\240\323\031\220\221\303\032\020\275\274\033\020\256\254\034\020\237\234\035\020\220\214\036\020\201|\037\020rl \020c\!\020TL"\020E<#\0206,$\020'\034%\020\030\014&\220C\005'\2204\365'\220%\345(\220\026\325)\220\007\305*\220\370\264+\220\351\244,\220\332\224-\220\313\204.\220\274t/\220\255d0\020\331]1\020\264r2\020\273=3\020\226R4\020\235\0355\020x26\020\177\3756\220\224\0338\020a\3358\220

PBX-VM:

Via: SIP/2.0/UDP 10.1.1.2:5060;branch=z9hG4bK0d4789df;rport
Contact: <sip:4141511****@10.1.1.2:5060>
Call-ID: [email protected]:5060
Via: SIP/2.0/UDP 10.1.1.2:5060;branch=z9hG4bK0d4789df;rport=5060
From: "Unknown" <sip:4141511****@10.1.1.2:5060>;tag=as7e695de9
Call-ID: [email protected]:5060
INVITE sip:4141511****@10.1.1.2:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 212.117.203.43:5060;branch=z9hG4bK-524287-1---f61e3c432916d409;rport
Record-Route: <sip:212.117.203.43:5060;transport=UDP;lr>
Via: SIP/2.0/UDP 212.117.203.43:5060;branch=z9hG4bK-524287-1---f61e3c432916d409;received=212.117.203.43;rport=5060
ACK sip:4141511****@10.1.1.2:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 212.117.203.43:5060;branch=z9hG4bK-524287-1---f61e3c432916d409;rport
Via: SIP/2.0/UDP 10.1.1.2:5060;branch=z9hG4bK3ae56652;rport
Contact: <sip:4141511****@10.1.1.2:5060>
Via: SIP/2.0/UDP 10.1.1.2:5060;branch=z9hG4bK3ae56652;rport=5060
Via: SIP/2.0/UDP 10.1.1.2:5060;branch=z9hG4bK0409184f;rport
Contact: <sip:4141511****@10.1.1.2:5060>

Debug Asterisk:

[2015-11-10 08:43:37] DEBUG[17274]: chan_sip.c:9178 __find_call: = Looking for  Call ID: [email protected]~1o (Checking From) --From tag kfitw5kzkponolzp.o --To-tag
[2015-11-10 08:43:37] DEBUG[17274]: acl.c:946 ast_ouraddrfor: For destination '212.117.203.43', our source address is '10.1.1.2'.
[2015-11-10 08:43:37] DEBUG[17274]: chan_sip.c:3866 ast_sip_ouraddrfor: Setting AST_TRANSPORT_UDP with address 10.1.1.2:5060
[2015-11-10 08:43:37] DEBUG[17274]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '212.117.203.43:5060' into...
[2015-11-10 08:43:37] DEBUG[17274]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '212.117.203.43' and port '5060'.
[2015-11-10 08:43:37] DEBUG[17274]: chan_sip.c:8765 __sip_alloc: Allocating new SIP dialog for [email protected]~1o - INVITE (No RTP)
[2015-11-10 08:43:37] DEBUG[17274][C-00000027]: chan_sip.c:28034 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
[2015-11-10 08:43:37] DEBUG[17274][C-00000027]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '212.117.203.43:5060' into...
[2015-11-10 08:43:37] DEBUG[17274][C-00000027]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '212.117.203.43' and port '5060'.
[2015-11-10 08:43:37] DEBUG[17274][C-00000027]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '212.117.203.43' into...
[2015-11-10 08:43:37] DEBUG[17274][C-00000027]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '212.117.203.43' and port ''.
[2015-11-10 08:43:37] DEBUG[17274][C-00000027]: chan_sip.c:3709 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 212.117.203.43:5060
[2015-11-10 08:43:37] DEBUG[17274]: chan_sip.c:9178 __find_call: = Looking for  Call ID: [email protected]~1o (Checking From) --From tag kfitw5kzkponolzp.o --To-tag as73caef88
[2015-11-10 08:43:37] DEBUG[17274][C-00000027]: chan_sip.c:28034 handle_incoming: **** Received ACK (6) - Command in SIP ACK
[2015-11-10 08:43:37] DEBUG[17274][C-00000027]: chan_sip.c:4406 __sip_ack: Stopping retransmission on '[email protected]~1o' of Response 866: Match Found
[2015-11-10 08:43:44] DEBUG[17274]: chan_sip.c:4262 __sip_autodestruct: Auto destroying SIP dialog '[email protected]~1o'
[2015-11-10 08:43:44] DEBUG[17274]: chan_sip.c:6707 sip_destroy: Destroying SIP dialog [email protected]~1o

About the System:
I am running FreePBX 13.0.11 on a Ubuntu 14.04.3 LTS with Asterisk 13.6.0. I also tried it with FreePBX 12.x.xx without success.

Does anybody has a similary problem or any idea, what the problem could be? im searching now for almost 2 Weeks, and i dont get it.

Thank you for your help

Remo Stirnimann

[danielf]

Hello,

Are you sure you have setup your NAT rules correctly? you need to configure port forwarding of UDP ports 5060 and UDP 10000-20000 to the PBX.
After that make sure that you setup correctly your SIP settings (localnet and external IP). Furthermore, make sure that you have configured a default gateway in your server.

If after all that, you still do not get any response please attach a log of the sip debug from the Asterisk server (sip set debug on).

Thank you,

Daniel Friedman
Trixton LTD.

[remosti]

Hello Daniel

Thank you for your Answer.
What i can see, is the Portforwarding correct. Otherwise i could not see the Call with TCPDump on the PBX-Server. The Default-Gateway is set up and it works (tested while installing the PBX).
About the NAT: No matter what i configure there, it is not working. The External IP is set on the “General SIP Settings” and at “Chan SIP Settings” i have set NAT to yes. IP Configuration is set to “Public IP”

For the Debug-Log, please see the previous Post around the second half of the post.

Thank you for your help
Remo

[danielf]

Hi Remo,

You do not have a publc ip but a static ip, so you should change to that and fill in your external ip and your local networks.
Please provide the log that i have requested and not the tcp dump.

Thank you,

Daniel Friedman
Trixton LTD.

[remosti]

Hi Daniel

Thank you for your Answer. I checked it with the Static IP, but without success.
Please see below the Debug-Log from Asterisk. This log represents one incomming call.

[2015-11-10 17:36:07] DEBUG[17274]: chan_sip.c:9178 __find_call: = Looking for  Call ID: [email protected]~1o (Checking From) --From tag alzprq5e525wwy2n.o --To-tag
[2015-11-10 17:36:07] DEBUG[17274]: acl.c:946 ast_ouraddrfor: For destination '212.117.203.43', our source address is '10.1.1.2'.
[2015-11-10 17:36:07] DEBUG[17274]: chan_sip.c:3833 ast_sip_ouraddrfor: Target address 212.117.203.43:5060 is not local, substituting externaddr
[2015-11-10 17:36:07] DEBUG[17274]: chan_sip.c:3866 ast_sip_ouraddrfor: Setting AST_TRANSPORT_UDP with address 85.14.240.152:5060
[2015-11-10 17:36:07] DEBUG[17274]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '212.117.203.43:5060' into...
[2015-11-10 17:36:07] DEBUG[17274]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '212.117.203.43' and port '5060'.
[2015-11-10 17:36:07] DEBUG[17274]: chan_sip.c:8765 __sip_alloc: Allocating new SIP dialog for [email protected]~1o - INVITE (No RTP)
[2015-11-10 17:36:07] DEBUG[17274][C-0000002d]: chan_sip.c:28034 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
[2015-11-10 17:36:07] DEBUG[17274][C-0000002d]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '212.117.203.43:5060' into...
[2015-11-10 17:36:07] DEBUG[17274][C-0000002d]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '212.117.203.43' and port '5060'.
[2015-11-10 17:36:07] DEBUG[17274][C-0000002d]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '212.117.203.43' into...
[2015-11-10 17:36:07] DEBUG[17274][C-0000002d]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '212.117.203.43' and port ''.
[2015-11-10 17:36:07] DEBUG[17274][C-0000002d]: chan_sip.c:3709 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 212.117.203.43:5060
[2015-11-10 17:36:07] DEBUG[17274]: chan_sip.c:9178 __find_call: = Looking for  Call ID: [email protected]~1o (Checking From) --From tag alzprq5e525wwy2n.o --To-tag as378c03e9
[2015-11-10 17:36:07] DEBUG[17274][C-0000002d]: chan_sip.c:28034 handle_incoming: **** Received ACK (6) - Command in SIP ACK
[2015-11-10 17:36:07] DEBUG[17274][C-0000002d]: chan_sip.c:4406 __sip_ack: Stopping retransmission on '[email protected]~1o' of Response 349: Match Found
[2015-11-10 17:36:13] DEBUG[17289]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead
[2015-11-10 17:36:13] DEBUG[17289]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 7 instead
[2015-11-10 17:36:13] DEBUG[17289]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 6 instead
[2015-11-10 17:36:13] DEBUG[17289]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead
[2015-11-10 17:36:14] DEBUG[17274]: chan_sip.c:4262 __sip_autodestruct: Auto destroying SIP dialog '[email protected]~1o'
[2015-11-10 17:36:14] DEBUG[17274]: chan_sip.c:6707 sip_destroy: Destroying SIP dialog [email protected]~1o

[danielf]

Hi,

Make sure that you are working only with chan_sip and not chan_pjsip.
Please provide your sip trunk settings in here.

Please add verbosity to the console (core set verbose 4) and then send an inbound call.

Thank you,

Daniel Friedman
Trixton LTD.

[remosti]

Hi Daniel

As far as i can see it (see Picture in the bottum), the pjsip is disabled. There is no Bind-Adress where he can listen on. Are there some other settings, to deactivate it? Just to be sure.

Below the Log with Verbose-Level 5 and Debug-Level 5:

RSSTPBX01*CLI> core set debug 5
Core debug is still 5.
RSSTPBX01*CLI> core set verbose 5
Console verbose is still 5.
[2015-11-11 08:46:11] DEBUG[17274]: chan_sip.c:9178 __find_call: = Looking for  Call ID: [email protected]~1o (Checking From) --From tag tebckgqd2nzvkpds.o --To-tag
[2015-11-11 08:46:11] DEBUG[17274]: acl.c:946 ast_ouraddrfor: For destination '212.117.203.43', our source address is '10.1.1.2'.
[2015-11-11 08:46:11] DEBUG[17274]: chan_sip.c:3833 ast_sip_ouraddrfor: Target address 212.117.203.43:5060 is not local, substituting externaddr
[2015-11-11 08:46:11] DEBUG[17274]: chan_sip.c:3866 ast_sip_ouraddrfor: Setting AST_TRANSPORT_UDP with address 85.14.240.152:5060
[2015-11-11 08:46:11] DEBUG[17274]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '212.117.203.43:5060' into...
[2015-11-11 08:46:11] DEBUG[17274]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '212.117.203.43' and port '5060'.
[2015-11-11 08:46:11] DEBUG[17274]: chan_sip.c:8765 __sip_alloc: Allocating new SIP dialog for [email protected]~1o - INVITE (No RTP)
[2015-11-11 08:46:11] DEBUG[17274][C-00000033]: chan_sip.c:28034 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
[2015-11-11 08:46:11] DEBUG[17274][C-00000033]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '212.117.203.43:5060' into...
[2015-11-11 08:46:11] DEBUG[17274][C-00000033]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '212.117.203.43' and port '5060'.
[2015-11-11 08:46:11] DEBUG[17274][C-00000033]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '212.117.203.43' into...
[2015-11-11 08:46:11] DEBUG[17274][C-00000033]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '212.117.203.43' and port ''.
[2015-11-11 08:46:11] DEBUG[17274][C-00000033]: chan_sip.c:3709 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 212.117.203.43:5060
[2015-11-11 08:46:11] DEBUG[17274]: chan_sip.c:9178 __find_call: = Looking for  Call ID: [email protected]~1o (Checking From) --From tag tebckgqd2nzvkpds.o --To-tag as2c23370b
[2015-11-11 08:46:11] DEBUG[17274][C-00000033]: chan_sip.c:28034 handle_incoming: **** Received ACK (6) - Command in SIP ACK
[2015-11-11 08:46:11] DEBUG[17274][C-00000033]: chan_sip.c:4406 __sip_ack: Stopping retransmission on '[email protected]~1o' of Response 889: Match Found
[2015-11-11 08:46:17] DEBUG[17274]: chan_sip.c:4262 __sip_autodestruct: Auto destroying SIP dialog '[email protected]~1o'
[2015-11-11 08:46:17] DEBUG[17274]: chan_sip.c:6707 sip_destroy: Destroying SIP dialog [email protected]~1o

Sip-Trunk Config:

[remosti]

Hi Daniel

I have just seen, that the picture is kind a dificulty to see. So here it is a little bit bigger.

[danielf]

Hi,

I am working with Sipcall/Backbone for a while so I know how to configure it. Your trunk configuration is wrong. You should configure only the peer settings like that:

username=4141511****
type=friend
secret=**********
qualify=yes
nat=yes
insecure=invite,port
host=business2.voipgateway.org
canreinvite=no
disallow=all
allow=alaw&ulaw
dtmfmode=rfc2833

Please delete the user section. Make sure that you have an inbound route for this number.
After configuring this please disable all sip debugging and general debugging like this:

core set verbose 4
core set debug 0
sip set debug off

Please report back with the Asterisk’s console log.

Thank you,

Daniel Friedman
Trixton LTD.

[remosti]

Hi Daniel

YOU ARE AWSOME!!!
It works now. After 2 Weeks of searching in the Web and Trying everything i could find about similar problems, its only a wrong Trunk-Setting.

Thank you so much for your help. You saved me a lot more work

See you
Remo Stirnimann

PS:
Following, the Asterisk-Log:

-- Executing [4141511****@from-trunk-sip-SipCall:1] Set("SIP/SipCall-00000003", "GROUP()=OUT_2") in new stack
-- Executing [4141511****@from-trunk-sip-SipCall:2] Goto("SIP/SipCall-00000003", "from-trunk,4141511****,1") in new stack
-- Goto (from-trunk,4141511****,1)
-- Executing [4141511****@from-trunk:1] Set("SIP/SipCall-00000003", "__FROM_DID=4141511****") in new stack
-- Executing [4141511****@from-trunk:2] Gosub("SIP/SipCall-00000003", "sub-record-check,s,1(in,4141511****,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/SipCall-00000003", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("SIP/SipCall-00000003", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("SIP/SipCall-00000003", "NOW=1447230022") in new stack
-- Executing [s@sub-record-check:4] Set("SIP/SipCall-00000003", "__DAY=11") in new stack
-- Executing [s@sub-record-check:5] Set("SIP/SipCall-00000003", "__MONTH=11") in new stack
-- Executing [s@sub-record-check:6] Set("SIP/SipCall-00000003", "__YEAR=2015") in new stack
-- Executing [s@sub-record-check:7] Set("SIP/SipCall-00000003", "__TIMESTR=20151111-092022") in new stack
-- Executing [s@sub-record-check:8] Set("SIP/SipCall-00000003", "__FROMEXTEN=unknown") in new stack
-- Executing [s@sub-record-check:9] Set("SIP/SipCall-00000003", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("SIP/SipCall-00000003", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("SIP/SipCall-00000003", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("SIP/SipCall-00000003", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("SIP/SipCall-00000003", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("SIP/SipCall-00000003", "2?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("SIP/SipCall-00000003", "1?sub-record-check,in,1") in new stack
-- Goto (sub-record-check,in,1)
-- Executing [in@sub-record-check:1] NoOp("SIP/SipCall-00000003", "Inbound Recording Check to 4141511****") in new stack
-- Executing [in@sub-record-check:2] Set("SIP/SipCall-00000003", "FROMEXTEN=unknown") in new stack
-- Executing [in@sub-record-check:3] ExecIf("SIP/SipCall-00000003", "10?Set(FROMEXTEN=0415102121)") in new stack
-- Executing [in@sub-record-check:4] Gosub("SIP/SipCall-00000003", "recordcheck,1(dontcare,in,4141511****)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/SipCall-00000003", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/SipCall-00000003", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("SIP/SipCall-00000003", "") in new stack
-- Executing [in@sub-record-check:5] Return("SIP/SipCall-00000003", "") in new stack
-- Executing [4141511****@from-trunk:3] Gosub("SIP/SipCall-00000003", "app-blacklist-check,s,1()") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/SipCall-00000003", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/SipCall-00000003", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/SipCall-00000003", "") in new stack
-- Executing [4141511****@from-trunk:4] Set("SIP/SipCall-00000003", "CDR(did)=4141511****") in new stack
-- Executing [4141511****@from-trunk:5] ExecIf("SIP/SipCall-00000003", "0 ?Set(CALLERID(name)=0415102121)") in new stack
-- Executing [4141511****@from-trunk:6] Set("SIP/SipCall-00000003", "CHANNEL(musicclass)=default") in new stack
-- Executing [4141511****@from-trunk:7] Set("SIP/SipCall-00000003", "__MOHCLASS=default") in new stack
-- Executing [4141511****@from-trunk:8] Set("SIP/SipCall-00000003", "__REVERSAL_REJECT=FALSE") in new stack
-- Executing [4141511****@from-trunk:9] GotoIf("SIP/SipCall-00000003", "1?post-reverse-charge") in new stack
-- Goto (from-trunk,4141511****,11)
-- Executing [4141511****@from-trunk:11] NoOp("SIP/SipCall-00000003", "") in new stack
-- Executing [4141511****@from-trunk:12] Set("SIP/SipCall-00000003", "__CALLINGNAMEPRES_SV=allowed_not_screened") in new stack
-- Executing [4141511****@from-trunk:13] Set("SIP/SipCall-00000003", "__CALLINGNUMPRES_SV=allowed_not_screened") in new stack
-- Executing [4141511****@from-trunk:14] Set("SIP/SipCall-00000003", "CALLERID(name-pres)=allowed_not_screened") in new stack
-- Executing [4141511****@from-trunk:15] Set("SIP/SipCall-00000003", "CALLERID(num-pres)=allowed_not_screened") in new stack
-- Executing [4141511****@from-trunk:16] Goto("SIP/SipCall-00000003", "app-announcement-1,s,1") in new stack
-- Goto (app-announcement-1,s,1)
-- Executing [s@app-announcement-1:1] GotoIf("SIP/SipCall-00000003", "0?begin") in new stack
-- Executing [s@app-announcement-1:2] Answer("SIP/SipCall-00000003", "") in new stack
-- Executing [s@app-announcement-1:3] Wait("SIP/SipCall-00000003", "1") in new stack > 0x7f53547862b0 -- Probation passed - setting RTP source address to 212.117.203.45:59382
-- Executing [s@app-announcement-1:4] NoOp("SIP/SipCall-00000003", "Playing announcement tt-monkeys") in new stack
-- Executing [s@app-announcement-1:5] Playback("SIP/SipCall-00000003", "tt-monkeys,noanswer") in new stack
-- <SIP/SipCall-00000003> Playing 'tt-monkeys.slin' (language 'de')
-- Executing [s@app-announcement-1:6] Goto("SIP/SipCall-00000003", "") in new stack

[danielf]

Hi Remo,

Thank you for your compliments. I am glad that it is working for you. I am living now in Switzerland so you can contact me if you will need any help.

Thank you,

Daniel Friedman
Trixton LTD.

[remosti]

Hi Daniel

Thats nice from you. Thank you.
From whitch region are you? German-, French- or Italien-Switzerland?

Have a nice day
Remo

[danielf]

Hi Remo,

I am living in Geneva.

Thank you,

Daniel Friedman
Trixton LTD.