06-06-2018 03:29 PM - edited 03-20-2019 10:22 PM
Dear Members!
I have an issue with the following setup:
ISDN BRI line -> CME -> SIP softphone
I can make outgoing calls over the ISDN BRI line using the softphone, no problem at all except a little latency...
But when I terminate an incoming call from ISDN BRI port to a SIP extension then the sip phone rings (so all the translation rules are fine) but as soon as I pick up the call on the sip softphone the line disconnects, on the ISDN q931 I see the following:
000263: Jun 6 22:24:00.446: ISDN BR0/1/0 Q931: TX -> CALL_PROC pd = 8 callref = 0x81
Channel ID i = 0x89
Exclusive, B1
000264: Jun 6 22:24:01.446: ISDN BR0/1/0 Q931: TX -> CALL_PROC pd = 8 callref = 0x81
Channel ID i = 0x89
Exclusive, B1
000265: Jun 6 22:24:01.462: ISDN BR0/1/0 Q931: TX -> ALERTING pd = 8 callref = 0x81
000266: Jun 6 22:24:07.018: ISDN BR0/1/0 Q931: TX -> CONNECT pd = 8 callref = 0x81
Channel ID i = 0x89
Exclusive, B1
000267: Jun 6 22:24:07.122: ISDN BR0/1/0 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x01
000268: Jun 6 22:24:07.126: ISDN BR0/1/0 Q931: TX -> DISCONNECT pd = 8 callref = 0x81
Cause i = 0x8090 - Normal call clearing
000269: Jun 6 22:24:07.134: ISDN BR0/1/0 Q931: RX <- RELEASE pd = 8 callref = 0x01
Cause i = 0x829A - Non-selected user clearing
000270: Jun 6 22:24:07.142: ISDN BR0/1/0 Q931: RX <- RELEASE pd = 8 callref = 0x01
Cause i = 0x829A - Non-selected user clearing
000271: Jun 6 22:24:07.254: ISDN BR0/1/0 Q931: RX <- RELEASE pd = 8 callref = 0x01
000272: Jun 6 22:24:07.258: ISDN BR0/1/0 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x81
The call disconnection says:
Cause i = 0x8090 - Normal call clearing
So I'm quite sure that the CME blocks the call somehow... on the sip softphone side I receive a timer expired event with the BYE message so I already added:
voice service voip
ip address trusted list
ipv4 192.168.0.0 255.255.255.0
ipv4 0.0.0.0 0.0.0.0
allow-connections sip to sip
sip
min-se 1200 session-expires 1200
registrar server expires max 3600 min 3600
no update-callerid
early-offer forced
no call service stop
Is there also a way to have an option like allow-connection isdn to sip ? :))))
06-07-2018 06:33 AM
06-07-2018 01:42 PM
Hello,
please find the attached log to this comment.
So what I did:
I made a call from my mobile (3670XXXX) to the ISDN incoming number 827370.
it is clearly seen that after the sip client (sip extension num,ber: 1010) rings, then the call is accepted (ACK message received), the ISDN leg disconnects....
Can you tell me what happens on the CCAPI level ??
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