Re: Problem with Transfer Calls (Sip running in the CME)

Thiago Cella · ‎03-03-2011

When the someone from PSTN or another VOIP system make incomming calls and goes to internal extension , works fine, when the user transfer this call to other internal user, this call down. The same thing happing when internal user want talk wth another internal user, the iniciator user calls to other user, thats works, but when this second user transfer the call to a thrid user , the call down too.

Funny that, before de call down, we can listen the three BIP tones.

In the CME i belive that is running SIP , because the models of the phones are SPA942 and Cisco 79XX, thats i belive the SCCP is not running on the CME.

Follow the configs:

telephony-service
load 7910 P00403020214
load 7960-7940 P00307020200
load 7914 S00104000000
load 7941 term41.default
load 7961 term61.default
max-ephones 20
max-dn 288
ip source-address 192.168.0.10 port 2000
system message office
time-zone 17
time-format 24
date-format dd-mm-yy
dialplan-pattern 1 5904.... extension-length 4 extension-pattern ....
max-conferences 8 gain -6
call-forward pattern .T
web admin system name admin password XXXXXX
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp Jan 01 2002 00:00:00

dial-peer voice 1000 voip
description OUTIGOING CALL TO SIP TRUNK
translation-profile outgoing SIP-TRUNK
destination-pattern 9T
voice-class codec 100
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
clid network-number XXXXXXXXXXX
no vad
!
dial-peer voice 1 voip
destination-pattern 1..
session target ipv4:192.168.0.10
!
dial-peer voice 2 pots
description INCOMMING CALL
destination-pattern 5904....
direct-inward-dial
port 0/0/0:1

voice service voip
address-hiding
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
sip
registrar server expires max 3600 min 600

Thiago Cella · ‎03-03-2011

Please, any idea?

Thiago Cella · ‎03-06-2011

People i have more details,

In this scenario are two sites, the SITE A has a C2801 with CME,the SITE B has a ASA 5510;

When someone in SITE A transfer the call to another user in SITE A, the call transfer works,no problems!

But if a user in the SITE B transfer the call to another user in SITE B or SITE A , the function call transfer dont works!!!

The telephones are Linksys 942, the transfer are done putting the button XFER.

Follow the debug log, when the user in SITE B transfer the call to someone :

AUD:Play Tone 1
AUD:Stop Tone

Calling:150@192.168.0.10:0
[0:0]AUD ALLOC CALL (port=16410)
[0:0]RTP Rx Up
SDP RTPMAP 101 --> 136

+++ bl on for 21474836 s
[1:0]DEC INIT 130
SDP RTPMAP 101 --> 136
[0:0]AUD ALLOC CALL (port=16402)
[0:0]RTP Rx Up
CC:Ringback

[0:0]RTP Rx Dn
AUD:Play Tone 8
PHN_setAudioPath(3)

CC:Connected

[0:0]ENC INIT 8
[0:0]RTP Tx Up (pt=8->c0a8000a:18534)
[0:0]RTCP Tx Up
SDP RTPMAP 101 --> 136
[0:0]RTP Tx Dn
[0:0]ENC INIT 8
[0:0]RTP Tx Up (pt=8->c0a8000a:17712)
[0:0]RTCP Tx Up
CC:Remote Resume
[0:0]RTCP Tx Up
CC:Connected

AUD:Stop Tone
[0:0]RTP Rx Up
[0:0]RTP Rx 1st PKT @16402(2)
[0:0]RTP Rx 1st PKT @16410(3)
[0:0]DEC INIT 8
[0:0]DEC INIT 8
PHN_setAudioPath(1)

[0:0]RTP Tx Dn
[0:0]RTP Rx Dn
SDP RTPMAP 101 --> 136
[0:0]RTP Tx Dn
CC:Remote Hold
AUD:Play Tone 18
[0:0]RTCP Tx Dn

Calling:116@192.168.0.10:0
[0:1]AUD ALLOC CALL (port=16404)
[0:1]RTP Rx Up
SDP RTPMAP 101 --> 136

[Phone]Exit Screen Saver Mode...
+++ bl on for 21474836 s
[1:0]DEC INIT 130
SDP RTPMAP 101 --> 136
[0:0]AUD ALLOC CALL (port=16392)
[0:0]RTP Rx Up
CC:Ringing

[0:1]RTP Rx Dn
AUD:Play Tone 8
[SH:6061]<<192.168.200.95:54321 (6c8e0673 00000100,8)
[SH:6061]<<192.168.200.95:54321 (6c8e0673 00000100,8)
[SH:6061]<<192.168.200.92:54321 (77795cc8 00000100,8)
[SH:6061]<<192.168.200.92:54321 (77795cc8 00000100,8)
[SH:6061]<<192.168.200.13:54321 (16515853 00000100,8)
[SH:6061]<<192.168.200.13:54321 (16515853 00000100,8)
[SH:6061]<<192.168.200.94:54321 (7800d327 00000100,8)
[SH:6061]<<192.168.200.94:54321 (7800d327 00000100,8)
PHN_setAudioPath(3)

CC:Connected

[0:0]ENC INIT 8
[0:0]RTP Tx Up (pt=8->c0a8000a:16692)
[0:0]RTCP Tx Up
SDP RTPMAP 101 --> 136
[0:1]RTP Tx Dn
[0:1]ENC INIT 8
[0:1]RTP Tx Up (pt=8->c0a8000a:18280)
[0:1]RTCP Tx Up
CC:Remote Resume
[0:1]RTCP Tx Up
CC:Connected

AUD:Stop Tone
[0:1]RTP Rx Up
[0:1]RTP Rx 1st PKT @16404(2)
[0:0]RTP Rx 1st PKT @16392(2)
[0:1]DEC INIT 8
[0:0]DEC INIT 8
[SH:6061]<<192.168.200.11:54321 (c14e9e63 00000100,8)
[SH:6061]<<192.168.200.11:54321 (c14e9e63 00000100,8)
CC:Ended

[0:0]AUD Rel Call
CC:Ended

AUD:Stop Tone
[0:0]AUD Rel Call

[0:1]RTP Tx Dn
[0:1]RTP Rx Dn
SDP RTPMAP 101 --> 136
[0:0]RTP Tx Dn
CC:Remote Hold
AUD:Play Tone 18
[0:0]RTCP Tx Dn
SDP RTPMAP 101 --> 136
CC:Hold

AUD:Play Tone 1
[0:1]RTP Tx Dn
CC:Remote Resume
[0:1]RTCP Tx Up
[SH:6061]<<192.168.200.10:54321 (3ccff8cc 00000100,8)
[SH:6061]<<192.168.200.10:54321 (3ccff8cc 00000100,8)
[SH:6061]<<192.168.200.93:54321 (6987e39a 00000100,8)
[SH:6061]<<192.168.200.93:54321 (6987e39a 00000100,8)
[SH:6061]<<192.168.200.6:54321 (c9eb2121 00000100,8)
[SH:6061]<<192.168.200.6:54321 (c9eb2121 00000100,8)
[SH:6061]<<192.168.200.3:54321 (5a62c031 00000100,8)
[SH:6061]<<192.168.200.3:54321 (5a62c031 00000100,8)
[SH:6061]<<192.168.200.17:54321 (e1b0db81 00000100,8)
[SH:6061]<<192.168.200.17:54321 (e1b0db81 00000100,8)
AUD:Play Tone 7
AUD:Stop Tone
PHN_setAudioPath(0)

[SH:6061]<<192.168.200.9:54321 (4fc2ca03 00000100,8)
[SH:6061]<<192.168.200.9:54321 (4fc2ca03 00000100,8)
[SH:6061]<<192.168.200.7:54321 (ec02e3b4 00000100,8)
[SH:6061]<<192.168.200.7:54321 (ec02e3b4 00000100,8)
[SH:6061]<<192.168.200.16:54321 (ec95ee74 00000100,8)
[SH:6061]<<192.168.200.16:54321 (ec95ee74 00000100,8)

Thiago Cella · ‎03-09-2011

People, follow the logs and topology .

Follow the debug log.

Follow the topolgy and config the router too.

i took out in the config file the DNs parameters, but all follow this default :

SPA phones:

voice register dn 7
number 101
no-reg
label user

voice register pool 1
id mac 000E.08D2.5507
number 1 dn 2
max registrations 96
dtmf-relay rtp-nte sip-notify
voice-class codec 1
username 101 password pass

Cisco phones:

ephone-dn 2
number 100
label Sales

ephone 2
mac-address 0023.3319.F493
type 7911
button 1:2

Any idea?

Tks