CALL PARKING is not working

Asterisk Asterisk Support
1

why call parking is not workinng???
( using two physical SIP phones and asterisk 11)
i was written dialplan in extestion.conf as fallow
[incoming]
include => parkedcalls
exten => 3000,1,Dial(SIP/3000,40,Tt)
exten => 3001,1,Dial(SIP/3001,40,Tt)

and in feature.conf
[general]
parkextn => 700
parkpos => 701-703
context => parkedcalls

i am calling from 3000 to 3001,once 3001 is ringing and pick the call,then dial 700 from 3001exten phone,ofter dialing 700,didn’t get any reply from sip phone system like “park 701 or 702 or 703”…please give me reason why call parking is not working???

2

What did you do to start the transfer?

3

just enter parking lot extenstion number 700.didn’t get any reply from phone system

4

You must do a SIP transfer to it, or use one of the features.conf codes, either the direct code for parking, or a transfer code. If you dial 700 without doing one of these, the digits will be forwarded to the other party.

You will need to enable any features.conf codes that you use:

[featuremap] ;blindxfer => #1 ; Blind transfer (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call! ;disconnect => *0 ; Disconnect (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call! ;automon => *1 ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call! ;atxfer => *2 ; Attended transfer -- Make sure to set the T and/or t option in the Dial() or Queue() app call! ;parkcall => #72 ; Park call (one step parking) -- Make sure to set the K and/or k option in the Dial() app call! ;automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!

5

thank you very much.

6

I have the following situation that I’m trying to find a solution for:

Party A calls Party B.
Party A and Party B talk for some time.
Party B generates a DTMF, this establishes a call to Party C and how can i establish conferences Parties A, B and C together(three parties communicate each other).

how to achive this feacture???

7

hi David,

DTMF features listed in feature map(#72 for call parked),only work when two channels have answered and are bridged together.; They can not be used while the remote party is ringing or in progress.
how to achieve bothe:how to solve fallowing condition??

with DTMF sequence #72 in feature map: its working but call established in one way

without DTMF sequence:

these concepts clearly mentioned in feature.conf file in etc/asterisk/, i am trying but not working.

thanks

8

You can also park a call by executing the relevant application. By default extension 600 in ParkedCalls is set up to do this. The channel that wants to be parked must do this (often this is the enquiry leg of an attended transfer. You can also do it as a third party request, using AMI.

However, I’ve never done parking when calls are not up, and I don’t know if that is supported. I’m not really sure why you would want to do that, what it would mean.

Also, we wanted the announcements to be suppressed, so I’m no t familiar with the usual use of the announcements.

9

thank you very much,

as your suggestion,i am using DTMF sequence #72 in feature map context for call parked:its working but call established in one way (one way call)

10

Which way. You will also need to provide debugging traces, including those from the channel driver, e.g. for SIP: enable full log logger.conf, type “core set verbose 5”, “core set debug 5”, and “sip set debug on”, at the Asterisk CLI, and capture /var/log/asterisk/full.log

11

this is my capture /var/log/asterisk/full.log…

From: vinay sip:3000@192.168.0.1:5060;tag=3117428269
To: “701” sip:701@192.168.0.1:5060;tag=as71054209
Call-ID: 130822143610135-153286417364@192.168.0.106
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:192.168.0.106:5060 —>
INVITE sip:701@192.168.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.106:5060;branch=z9hG4bK27039159692229028761;rport
From: vinay sip:3000@192.168.0.1:5060;tag=3117428269
To: “701” sip:701@192.168.0.1:5060
Call-ID: 130822143610135-153286417364@192.168.0.106
CSeq: 2 INVITE
Contact: sip:3000@192.168.0.106:5060
Authorization: Digest username=“3000”, realm=“asterisk”, nonce=“27b11831”, uri=“sip:701@192.168.0.1:5060”, response="97ad1ef630b18b0f858cde44555
Max-Forwards: 70
Supported: replaces, join, path
User-Agent: common
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 185

v=0
o=3000 23847259 28983820 IN IP4 192.168.0.106
s=A conversation
c=IN IP4 192.168.0.106
t=0 0
m=audio 10002 RTP/AVP 9 8
a=rtpmap:9 G722/16000
a=rtpmap:8 PCMA/8000
a=sendrecv
<------------->
— (14 headers 9 lines) —
Sending to 192.168.0.106:5060 (NAT)
Using INVITE request as basis request - 130822143610135-153286417364@192.168.0.106
Found peer ‘3000’ for ‘3000’ from 192.168.0.106:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 8
Found unknown media description format G722 for ID 9
Found audio description format PCMA for ID 8
Capabilities: us - 0x80008 (alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.106:10002
Peer doesn’t provide video
Looking for 701 in from-sip (domain 192.168.0.1:5060)
list_route: hop: sip:3000@192.168.0.106:5060

<— Transmitting (NAT) to 192.168.0.106:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.106:5060;branch=z9hG4bK27039159692229028761;received=192.168.0.106;rport=5060
From: vinay sip:3000@192.168.0.1:5060;tag=3117428269
To: “701” sip:701@192.168.0.1:5060
Call-ID: 130822143610135-153286417364@192.168.0.106
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:701@192.168.0.1:5060
Content-Length: 0

<------------>
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
isx_indicate called for line 0 cond 17
case unhold
isx_indicate called for line 0 cond 22
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250

[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
– Executing [701@from-sip:1] ParkedCall(“SIP/3000-00000001”, “701,default”) in new stack
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250

[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:40:45] ERROR[9105]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1038d284 for 0x1038d250
[Feb 24 20:31:49] VERBOSE[5470] channel.c: == Registered channel type ‘MGCP’ (Media Gateway Control Protocol (MGCP))
[Feb 24 20:31:49] VERBOSE[5470] rtp_engine.c: == Registered RTP glue ‘MGCP’
[Feb 24 20:31:49] VERBOSE[5470] loader.c: chan_mgcp.so => (Media Gateway Control Protocol (MGCP))
[Feb 24 20:31:49] VERBOSE[5470] chan_sip.c: SIP channel loading…
[Feb 24 20:31:49] VERBOSE[5470] config.c: == Parsing ‘/etc/asterisk/sip.conf’: [Feb 24 20:31:49] VERBOSE[5470] config.c: == Found
[Feb 24 20:31:49] VERBOSE[5470] config.c: == Parsing ‘/etc/asterisk/users.conf’: [Feb 24 20:31:49] VERBOSE[5470] config.c: == Found
[Feb 24 20:31:49] ERROR[5470] netsock2.c: getaddrinfo(“Nuevo”, “(null)”, …): Name or service not known
[Feb 24 20:31:49] WARNING[5470] acl.c: Unable to lookup ‘Nuevo’
[Feb 24 20:31:49] WARNING[5470] acl.c: Cannot connect
[Feb 24 20:31:49] VERBOSE[5470] chan_sip.c: == SIP Listening on 0.0.0.0:5060
[Feb 24 20:31:49] VERBOSE[5470] netsock2.c: == Using SIP CoS mark 4
[Feb 24 20:31:49] NOTICE[5470] chan_sip.c: The ‘username’ field for sip peers has been deprecated in favor of the term ‘defaultuser’
[Feb 24 20:31:49] WARNING[5470] chan_sip.c: Unknown dtmf mode ‘2833’ on line 1416, using rfc2833
[Feb 24 20:31:49] WARNING[5470] chan_sip.c: Unknown dtmf mode ‘2833’ on line 1433, using rfc2833
[Feb 24 20:31:49] VERBOSE[5470] dnsmgr.c: > doing dnsmgr_lookup for ‘getonsip.com’
[Feb 24 20:31:49] VERBOSE[5470] srv.c: > ast_get_srv: SRV lookup for ‘_sip._udp.getonsip.com’ mapped to host sip.onsip.com, port 5060
[Feb 24 20:31:49] VERBOSE[5470] config.c: == Parsing ‘/etc/asterisk/sip_notify.conf’: [Feb 24 20:31:49] VERBOSE[5470] config.c: == Found
[Feb 24 20:31:49] VERBOSE[5470] channel.c: == Registered channel type ‘SIP’ (Session Initiation Protocol (SIP))
[Feb 24 20:31:49] VERBOSE[5470] rtp_engine.c: == Registered RTP glue ‘SIP’
[Feb 24 20:31:49] VERBOSE[5470] pbx.c: == Registered application ‘SIPDtmfMode’
[Feb 24 20:31:49] VERBOSE[5470] pbx.c: == Registered application ‘SIPAddHeader’
[Feb 24 20:31:49] VERBOSE[5470] pbx.c: == Registered application ‘SIPRemoveHeader’
[Feb 24 20:31:49] VERBOSE[5470] pbx.c: == Registered custom function ‘SIP_HEADER’
[Feb 24 20:31:49] VERBOSE[5470] pbx.c: == Registered custom function ‘SIPPEER’
[Feb 24 20:31:49] VERBOSE[5470] pbx.c: == Registered custom function ‘SIPCHANINFO’
[Feb 24 20:31:49] VERBOSE[5470] pbx.c: == Registered custom function ‘CHECKSIPDOMAIN’
[Feb 24 20:31:49] VERBOSE[5470] manager.c: == Manager registered action SIPpeers
[Feb 24 20:31:49] VERBOSE[5470] manager.c: == Manager registered action SIPshowpeer
[Feb 24 20:31:49] VERBOSE[5470] manager.c: == Manager registered action SIPqualifypeer
[Feb 24 20:31:49] VERBOSE[5470] manager.c: == Manager registered action SIPshowregistry
[Feb 24 20:31:49] VERBOSE[5470] manager.c: == Manager registered action SIPnotify
[Feb 24 20:31:49] VERBOSE[5470] loader.c: chan_sip.so => (Session Initiation Protocol (SIP))
[Feb 24 20:31:49] VERBOSE[5496] dnsmgr.c: > doing dnsmgr_lookup for ‘getonsip.com’
[Feb 24 20:31:49] VERBOSE[5496] srv.c: > ast_get_srv: SRV lookup for ‘_sip._udp.getonsip.com’ mapped to host sip.onsip.com, port 5060
[Feb 24 20:31:49] WARNING[5496] acl.c: Cannot connect
[Feb 24 20:31:49] WARNING[5496] chan_sip.c: sip_xmit of 0x104a7ae8 (len 364) to 199.7.173.100:5060 returned -2: Network is unreachable
[Feb 24 20:31:49] ERROR[5496] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[Feb 24 20:31:49] VERBOSE[5470] config.c: == Parsing ‘/etc/asterisk/iax.conf’: [Feb 24 20:31:49] VERBOSE[5470] config.c: == Found
[Feb 24 20:31:49] VERBOSE[5470] config.c: == Parsing ‘/etc/asterisk/users.conf’: [Feb 24 20:31:49] VERBOSE[5470] config.c: == Found

thank you

12 

Check that you do not have mixed versions of modules.  Once that is eliminated, raise a  bug report on [issues.asterisk.org/jira](https://issues.asterisk.org/jira)

Check that you do not have mixed versions of modules. Once that is eliminated, raise a bug report on issues.asterisk.org/jira

13

thank you