[Home]

Summary:ASTERISK-19312: No DTMF decoding on outbound call via SS7 E1 channel
Reporter:Igor Nikolaev (microlana)Labels:
Date Opened:2012-02-08 07:10:35.000-0600Date Closed:2012-03-05 15:37:27.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_dahdi/SS7
Versions:10.1.1 Frequency of
Occurrence
Constant
Related
Issues:
Environment:Attachments:( 0) ss7_open_media.patch
Description:Call scenario:

SIP endpoint (DTMF via any outband class, ex. INFO) --> asterisk --> E1 SS7 --> PSTN

Call originated from SIP endpoint and terminated on PSTN. SIP device use any outband type DTMF handling (example INFO).

Abonent from PSTN unable to send DTMF to the SIP endpoint, because DSP not setup on outbound call where call is established. Therefore, DTMF is sent via inband mode, while SIP endpoint accept only outband.


Comments:By: Matt Jordan (mjordan) 2012-02-09 16:42:09.193-0600

We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



By: Igor Nikolaev (microlana) 2012-02-10 00:26:51.913-0600

Hardware equipment:
Digium, Inc. Wildcard TE220 dual-span T1/E1/J1 card 3.3V (PCI-Express) (5th gen) (rev 02)
(E1 mode)

2xE1 wire to PSTN SS7 equipment

Software low level drivers:

dahdi-linux-complete version 2.5.0.2+2.5.0.2

/etc/dahdi/system.conf
span=1,1,0,ccs,hdb3,yellow
bchan=1-15,17-31
mtp2=16
span=2,2,0,ccs,hdb3,yellow
bchan=32-46,48-62
mtp2=47

Software high level:

libss7 1.0.2

Asterisk 10.1.0 built by root @ ss7 on a x86_64 running Linux on 2012-02-07 08:29:27 UTC

/etc/asterisk/chan_dahdi.conf
signalling=ss7
group = 1
context = pstn
linkset = 1
pointcode = XXX
adjpointcode = YYY
defaultdpc = YYY
cicbeginswith = 1
sigchan = 16
channel = 1-15
cicbeginswith = 17
channel = 17-31
cicbeginswith = 33
sigchan = 47
channel = 32-46
cicbeginswith = 49
channel = 48-62

/etc/asterisk/sip.conf
[voip_device]
context=voip
type=friend
host=xxx.xxx.xxx.xxx
qualify=yes
directmedia=no
dtmfmode=rfc2833
t38pt_udptl=yes
qualify=yes
insecure=PORT,INVITE
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.xxx/255.255.255.255
diallow=all
allow=alaw

/etc/asterisk/extensions.conf
[voip]
exten => _X.,1,NoOp(VoIP --> SS7)
exten => _X.,n,Set(CALLERID(num-pres)=allowed)
exten => _X.,n,Set(FAXOPT(gateway)=yes)
exten => _X.,n,Dial(DAHDI/g1/${EXTEN})
[pstn]
exten => _X.,1,NoOp(SS7 --> VoIP)
exten => _X.,n,Set(CALLERID(num-pres)=allowed)
exten => _X.,n,Set(FAXOPT(gateway)=yes)
exten => _X.,n,Dial(SIP/voip_device/${EXTEN})

Call originated from SIP/voip_device, bridged with DAHDI/g1/* channel.

Example call tracing:
 == Using SIP RTP CoS mark 5
   -- Executing [111111@voip:1] NoOp("SIP/voip -0000047d", "VoIP --> SS7") in new stack
   -- Executing [111111@voip:2] Set("SIP/voip -0000047d", "CALLERID(num-pres)=allowed") in new stack
   -- Executing [111111@voip:3] Set("SIP/voip -0000047d", "FAXOPT(gateway)=yes") in new stack
   -- Executing [111111@voip:4] Dial("SIP/voip -0000047d", "DAHDI/g1/111111") in new stack
   -- Called DAHDI/g1/111111    
   -- DAHDI/3-1 is proceeding passing it to SIP/voip-0000047d
   -- DAHDI/3-1 is ringing
   -- DAHDI/3-1 answered SIP/voip-0000047d

DTMF from remote PSTN side sent to the SIP/voip_device with inband, instead of rfc2833.



By: Matt Jordan (mjordan) 2012-02-10 08:29:21.849-0600

While this information is useful, this is not a complete DEBUG log.  Please follow the instructions linked in the previous comment and provide a DEBUG log demonstrating this issue.

By: Igor Nikolaev (microlana) 2012-02-10 08:44:01.221-0600

Patch: Initialize media channel when SS7 call  is created.


By: Igor Nikolaev (microlana) 2012-02-10 08:50:28.649-0600

I can't collect original debug information because this issue was fixed by me and now my asterisk run at business project. I can't stop it now. I haven't other Digium card for simulate original issue.

Also, as i can see, this issue may be fixed by SS7-27, patch
SS7-27_dahdi_v2.diff



By: Matt Jordan (mjordan) 2012-02-10 10:58:39.587-0600

Ah - gotchya.  The patch should be sufficient - thanks!