Hi all,
Long story short, I have a 1760-V running CCME 4.1 and recently subscribed to an online VoIP provider, just trying to get the two intergrated so I can make calls via CCME.
At the moment I can't make any calls via SIP, I currently have it setup so I dial "0" to get a POTS line (working), and "9" for a SIP (not working). All internal calls are working as well as calls from POTS to internal.
I have not implimented DID for SIP as yet, as the provider haven't assigned me a DID due to a mistake.
Have confirmed "sh sip-ua register status" that the line is registered.
There is also a seperate IOS firewal in the environment with an "ip inspect sip" rule implimented, tho I don't think outbound call would make any difference.
Below is relevant sections of the config, thanks in advance for any tips / hints. I have a feeling that my dial plans are screwed. Have followed various online guides / forums to no avail 🙁
___CONFIG___
**** INFORMATION CUT ****
!
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
h323
call start slow
sip
registrar server
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
!
voice translation-rule 1
rule 1 /^9\(0.........\)/ /\1/
rule 2 /^9\([2-9].......\)/ /\1/
rule 3 /^9\(13[^0]...\)/ /\1/
rule 4 /^9\(1[3,8]00......\)/ /\1/
rule 5 /^9\(0011.*\)/ /\1/
rule 6 /^9\(000\)/ /\1/
!
voice translation-profile SIP
translate called 1
!
**** INFORMATION CUT ****
**** A BUNCH OF PHONE TFTP LOADS ****
!
voice-port 2/0
connection plar opx 8004
description PSTN Trunk
!
voice-port 2/1
!
dial-peer voice 811 pots
destination-pattern 811
port 2/0
no sip-register
!
dial-peer voice 101 pots
description dial 0 for outside POTS local call
destination-pattern 0T
port 2/0
no sip-register
!
dial-peer voice 901 voip
description dial 9 for outside SIP local call
translation-profile outgoing SIP
destination-pattern 9T
session protocol sipv2
session target dns:XXXXX
dtmf-relay rtp-nte
voice-class codec 1
no vad
!
sip-ua
authentication username XXXXX password XXXXX
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
timers register 250
registrar dns:XXXXX
sip-server dns:XXXXX
!
telephony-service
load 7960-7940 P00308000700
load ATA ata030203sccp051201a
load 7912 cmterm-7912-8.0.4-sccp.cop
max-ephones 30
max-dn 150 no-reg primary
ip source-address 172.16.90.2 port 2000
caller-id block code *10
timeouts interdigit 5
system message CME
url services http://phone-xml.berbee.com/menu.xml
time-zone 48
date-format yy-mm-dd
max-conferences 4 gain -6
moh music-on-hold.au
multicast moh 239.15.10.1 port 2000
web admin system name admin secret 5 $1$4F9s$q0G/B7T9Efd1aUcZf9pVu/
dn-webedit
transfer-system full-consult
secondary-dialtone 0
fac standard
create cnf-files version-stamp 7960 Aug 30 2008 18:59:55
!
**** INFORMATION CUT *****
!
ephone-dn 4
number 8004 no-reg both
trunk 811 transfer-timeout 30 monitor-port 2/0
!
***** INFORMATION CUT *****
!
ephone-dn 6 dual-line
number 8006 no-reg both
description Study
name Study
!
***** INFORMATION CUT *****
!
ephone-dn 11 dual-line
number XXXXX (same as sip-ua username above)
description SIP Line
!
**** INFORMATION CUT *****
!
ephone 5
mac-address XXXXX
paging-dn 7
type 7940
keep-conference
button 1:6 2c4,11
!
***** INFORMATION CUT *****
Long story short, I have a 1760-V running CCME 4.1 and recently subscribed to an online VoIP provider, just trying to get the two intergrated so I can make calls via CCME.
At the moment I can't make any calls via SIP, I currently have it setup so I dial "0" to get a POTS line (working), and "9" for a SIP (not working). All internal calls are working as well as calls from POTS to internal.
I have not implimented DID for SIP as yet, as the provider haven't assigned me a DID due to a mistake.
Have confirmed "sh sip-ua register status" that the line is registered.
There is also a seperate IOS firewal in the environment with an "ip inspect sip" rule implimented, tho I don't think outbound call would make any difference.
Below is relevant sections of the config, thanks in advance for any tips / hints. I have a feeling that my dial plans are screwed. Have followed various online guides / forums to no avail 🙁
___CONFIG___
**** INFORMATION CUT ****
!
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
h323
call start slow
sip
registrar server
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
!
voice translation-rule 1
rule 1 /^9\(0.........\)/ /\1/
rule 2 /^9\([2-9].......\)/ /\1/
rule 3 /^9\(13[^0]...\)/ /\1/
rule 4 /^9\(1[3,8]00......\)/ /\1/
rule 5 /^9\(0011.*\)/ /\1/
rule 6 /^9\(000\)/ /\1/
!
voice translation-profile SIP
translate called 1
!
**** INFORMATION CUT ****
**** A BUNCH OF PHONE TFTP LOADS ****
!
voice-port 2/0
connection plar opx 8004
description PSTN Trunk
!
voice-port 2/1
!
dial-peer voice 811 pots
destination-pattern 811
port 2/0
no sip-register
!
dial-peer voice 101 pots
description dial 0 for outside POTS local call
destination-pattern 0T
port 2/0
no sip-register
!
dial-peer voice 901 voip
description dial 9 for outside SIP local call
translation-profile outgoing SIP
destination-pattern 9T
session protocol sipv2
session target dns:XXXXX
dtmf-relay rtp-nte
voice-class codec 1
no vad
!
sip-ua
authentication username XXXXX password XXXXX
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
timers register 250
registrar dns:XXXXX
sip-server dns:XXXXX
!
telephony-service
load 7960-7940 P00308000700
load ATA ata030203sccp051201a
load 7912 cmterm-7912-8.0.4-sccp.cop
max-ephones 30
max-dn 150 no-reg primary
ip source-address 172.16.90.2 port 2000
caller-id block code *10
timeouts interdigit 5
system message CME
url services http://phone-xml.berbee.com/menu.xml
time-zone 48
date-format yy-mm-dd
max-conferences 4 gain -6
moh music-on-hold.au
multicast moh 239.15.10.1 port 2000
web admin system name admin secret 5 $1$4F9s$q0G/B7T9Efd1aUcZf9pVu/
dn-webedit
transfer-system full-consult
secondary-dialtone 0
fac standard
create cnf-files version-stamp 7960 Aug 30 2008 18:59:55
!
**** INFORMATION CUT *****
!
ephone-dn 4
number 8004 no-reg both
trunk 811 transfer-timeout 30 monitor-port 2/0
!
***** INFORMATION CUT *****
!
ephone-dn 6 dual-line
number 8006 no-reg both
description Study
name Study
!
***** INFORMATION CUT *****
!
ephone-dn 11 dual-line
number XXXXX (same as sip-ua username above)
description SIP Line
!
**** INFORMATION CUT *****
!
ephone 5
mac-address XXXXX
paging-dn 7
type 7940
keep-conference
button 1:6 2c4,11
!
***** INFORMATION CUT *****
