[Solved] Incomming Calls will not be processed

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#[email protected]\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

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.

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

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.

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

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.

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:

Hi Daniel

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

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.

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 [[email protected]:1] GotoIf("SIP/SipCall-00000003", "0?initialized") in new stack
-- Executing [[email protected]:2] Set("SIP/SipCall-00000003", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [[email protected]:3] Set("SIP/SipCall-00000003", "NOW=1447230022") in new stack
-- Executing [[email protected]:4] Set("SIP/SipCall-00000003", "__DAY=11") in new stack
-- Executing [[email protected]:5] Set("SIP/SipCall-00000003", "__MONTH=11") in new stack
-- Executing [[email protected]:6] Set("SIP/SipCall-00000003", "__YEAR=2015") in new stack
-- Executing [[email protected]:7] Set("SIP/SipCall-00000003", "__TIMESTR=20151111-092022") in new stack
-- Executing [[email protected]:8] Set("SIP/SipCall-00000003", "__FROMEXTEN=unknown") in new stack
-- Executing [[email protected]:9] Set("SIP/SipCall-00000003", "__MON_FMT=wav") in new stack
-- Executing [[email protected]:10] NoOp("SIP/SipCall-00000003", "Recordings initialized") in new stack
-- Executing [[email protected]:11] ExecIf("SIP/SipCall-00000003", "0?Set(ARG3=dontcare)") in new stack
-- Executing [[email protected]:12] Set("SIP/SipCall-00000003", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [[email protected]:13] ExecIf("SIP/SipCall-00000003", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [[email protected]:14] GotoIf("SIP/SipCall-00000003", "2?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [[email protected]:17] GotoIf("SIP/SipCall-00000003", "1?sub-record-check,in,1") in new stack
-- Goto (sub-record-check,in,1)
-- Executing [[email protected]:1] NoOp("SIP/SipCall-00000003", "Inbound Recording Check to 4141511****") in new stack
-- Executing [[email protected]:2] Set("SIP/SipCall-00000003", "FROMEXTEN=unknown") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/SipCall-00000003", "10?Set(FROMEXTEN=0415102121)") in new stack
-- Executing [[email protected]:4] Gosub("SIP/SipCall-00000003", "recordcheck,1(dontcare,in,4141511****)") in new stack
-- Executing [[email protected]:1] NoOp("SIP/SipCall-00000003", "Starting recording check against dontcare") in new stack
-- Executing [[email protected]:2] Goto("SIP/SipCall-00000003", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [[email protected]:3] Return("SIP/SipCall-00000003", "") in new stack
-- Executing [[email protected]: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 [[email protected]:1] GotoIf("SIP/SipCall-00000003", "0?blacklisted") in new stack
-- Executing [[email protected]:2] Set("SIP/SipCall-00000003", "CALLED_BLACKLIST=1") in new stack
-- Executing [[email protected]: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 [[email protected]:1] GotoIf("SIP/SipCall-00000003", "0?begin") in new stack
-- Executing [[email protected]:2] Answer("SIP/SipCall-00000003", "") in new stack
-- Executing [[email protected]:3] Wait("SIP/SipCall-00000003", "1") in new stack > 0x7f53547862b0 -- Probation passed - setting RTP source address to 212.117.203.45:59382
-- Executing [[email protected]:4] NoOp("SIP/SipCall-00000003", "Playing announcement tt-monkeys") in new stack
-- Executing [[email protected]:5] Playback("SIP/SipCall-00000003", "tt-monkeys,noanswer") in new stack
-- <SIP/SipCall-00000003> Playing 'tt-monkeys.slin' (language 'de')
-- Executing [[email protected]:6] Goto("SIP/SipCall-00000003", "") in new stack
1 Like

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.

Hi Daniel

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

Have a nice day
Remo

Hi Remo,

I am living in Geneva.

Thank you,

Daniel Friedman
Trixton LTD.

1 Like