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791
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2
Helpful
3
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Phone call are not forwarded to external numbers(mobile)

san4ez39
Level 1
Level 1

‎04-26-2023 01:17 AM

Hello,

I have a problem that started long time ago and no one been able solve that and now it is me who tasked to solve this problem.

Phone calls are not forwarded to external numbers like mobile. I have collected the logs from test phone call, but i am not really VoIP person..so maybe someone can help here. I attached loges in a file as there are quite a lot of them.

Calls to external numbers and from external are working fine. But if there is forward applied and someone calls from internal to internal(forward applied) - then silence. If call goes to external and on internal phone forward is set - same thing.

 

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san4ez39
Level 1
Level 1

‎04-26-2023 01:24 AM

Just to add router configuration regarding voip. phones are configured on router.

voice-card 0
dsp services dspfarm
!
!
!
voice service voip
ip address trusted list
ipv4 172.22.131.0 255.255.255.128
ipv4 172.22.149.0 255.255.255.0
ipv4 172.22.148.0 255.255.255.0
ipv4 172.22.52.100 255.255.255.255
ipv4 172.22.52.99 255.255.255.255
ipv4 185.44.215.244 255.255.255.255
address-hiding
mode border-element
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
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
rel1xx disable
subscription maximum accept 0
registrar server expires max 1800 min 300
!
!
voice class uri HUBS sip
host ipv4:172.22.149.101
host ipv4:172.22.149.102
!
voice class uri CUCM sip
host ipv4:172.22.52.100
host ipv4:172.22.52.99
!
voice class uri SIPTRUNK sip
host 185\.44\.215\.244
voice class codec 7800
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
!
voice class codec 9999
codec preference 1 transparent
!
voice class codec 9900
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
!
voice class codec 100
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
!
voice class codec 199
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
!
voice class codec 9899
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
!
!
voice class sip-profiles 100
request ANY sip-header P-Asserted-Identity modify "185.128.151.115" "sip2.telezero.net"
response ANY sip-header P-Asserted-Identity modify "185.128.151.115" "sip2.telezero.net"
request ANY sip-header From modify "185.128.151.115" "sip2.telezero.net"
response ANY sip-header From modify "185.128.151.115" "sip2.telezero.net"
response ANY sip-header To modify "185.128.151.115" "sip2.telezero.net"
request ANY sip-header Contact copy "sip:(.*)@" u03
request ANY sip-header Remote-Party-ID modify "sip:(.*)@185.128.151.115" "sip:\u03@sip2.telezero.net"
response ANY sip-header Contact copy "sip:(.*)@" u04
response ANY sip-header Remote-Party-ID modify "sip:(.*)@185.128.151.115" "sip:\u04@sip2.telezero.net"
request REINVITE sdp-header Audio-Attribute modify "a=sendonly" "a=sendrecv"
request INVITE sdp-header Audio-Attribute modify "a=sendonly" "a=sendrecv"
request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"
!
voice class sip-profiles 199
request ANY sip-header P-Asserted-Identity modify "185.128.151.115" "sip2.telezero.net"
response ANY sip-header P-Asserted-Identity modify "185.128.151.115" "sip2.telezero.net"
request ANY sip-header From modify "185.128.151.115" "sip2.telezero.net"
response ANY sip-header From modify "185.128.151.115" "sip2.telezero.net"
response ANY sip-header To modify "185.128.151.115" "sip2.telezero.net"
request ANY sip-header Contact copy "sip:(.*)@" u03
request ANY sip-header Remote-Party-ID modify "sip:(.*)@185.128.151.115" "sip:\u03@sip2.telezero.net"
response ANY sip-header Contact copy "sip:(.*)@" u04
response ANY sip-header Remote-Party-ID modify "sip:(.*)@185.128.151.115" "sip:\u04@sip2.telezero.net"
request REINVITE sdp-header Audio-Attribute modify "a=sendonly" "a=sendrecv"
request INVITE sdp-header Audio-Attribute modify "a=sendonly" "a=sendrecv"
!
!
voice class e164-pattern-map 9900
description TO-HUBS
e164 62[0-4].T
e164 83...$
e164 6[0-1,3-9]..T
e164 [1-5,7-9]...T
!
!
voice class e164-pattern-map 100
description ToITSP
e164 00039..........T
e164 01..T
e164 +T
e164 0.T
e164 1..T
e164 ......T
!
!
voice class server-group 1000
ipv4 172.22.52.100
ipv4 172.22.52.99
description CUCMs
hunt-scheme round-robin
!
voice class sip-options-keepalive 9900
description SIP-OPTIONS-PING-TO-HUBS
down-interval 15
up-interval 15
retry 2
transport udp
!
!
voice iec syslog
voice register global
mode cme
source-address 172.22.148.160 port 5060
timeouts interdigit 2
max-dn 100
max-pool 50
load 7821 sip78xx.10-3-1-12
load 7861 sip78xx.10-3-1-12
authenticate register
authenticate realm retalindustries.eu
olsontimezone Europe/London version 2015a
time-format 24
date-format D/M/Y
service https
url directory http://172.22.148.160/localdirectory
tftp-path flash:
create profile sync 1200211631044132
conference hardware

voice register dialplan 1
type 7940-7960-others
pattern 1 6...
pattern 2 7...
pattern 3 5...
pattern 4 4...
pattern 5 00039..........
pattern 6 1...
pattern 7 04..... timeout 2
pattern 8 06..... timeout 2
pattern 9 02..... timeout 2
pattern 10 03...........
pattern 11 32... timeout 4
pattern 12 34... timeout 4
pattern 13 3...

voice statistics type iec
voice statistics time-range since-reset
voice statistics time-range periodic 60minutes start 00:00 end 00:00 days-of-week daily
voice statistics max-storage-duration day 2
!
voice translation-rule 10000
rule 4 /\(^00\)/ /+/
rule 5 /\(^\+\)/ /+/
rule 6 // /0/

voice translation-rule 19901
rule 1 /^\+/ /00/
rule 8 /\(^1..$\)/ /\1/
rule 50 /^0/ //
!
!
voice translation-profile FROM_SIPTRUNK
translate calling 10000
translate called 10001
!
voice translation-profile TO_SIPTRUNK
translate calling 19900
translate called 19901

interface Loopback0
description Internal VoIP
ip address 172.22.148.160 255.255.255.255
ip nat inside
ip virtual-reassembly in
!
interface Loopback1
description External VoIP
ip address 172.22.149.160 255.255.255.255

mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
sccp local Loopback0
sccp ccm 172.22.148.160 identifier 9999 version 7.0
sccp
!
sccp ccm group 9999
associate ccm 9999 priority 1
associate profile 9901 register DSPCFB9901
associate profile 9902 register DSPXC9902
!
!
!
dspfarm profile 9902 transcode
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 4
associate application SCCP
!

dspfarm profile 9901 conference
codec g729r8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 2
associate application SCCP
!
dial-peer voice 9999 voip
description INBOUND-FROM-HUBS
session protocol sipv2
incoming uri via HUBS
voice-class codec 9999
voice-class sip bind control source-interface Loopback1
voice-class sip bind media source-interface Loopback1
dtmf-relay rtp-nte
fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 2 fallback pass-through g711alaw
no vad
!
dial-peer voice 9901 voip
description OUTBOUND-TO-HUBS
session protocol sipv2
session target ipv4:172.22.149.101
destination e164-pattern-map 9900
voice-class codec 9900
voice-class sip options-keepalive profile 9900
voice-class sip bind control source-interface Loopback1
voice-class sip bind media source-interface Loopback1
dtmf-relay rtp-nte
fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 2 fallback pass-through g711alaw
no vad
!
dial-peer voice 9902 voip
description OUTBOUND-TO-HUBS
session protocol sipv2
session target ipv4:172.22.149.102
destination e164-pattern-map 9900
voice-class codec 9900
voice-class sip options-keepalive profile 9900
voice-class sip bind control source-interface Loopback1
voice-class sip bind media source-interface Loopback1
dtmf-relay rtp-nte
fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 2 fallback pass-through g711alaw
no vad
!

dial-peer voice 100 voip
description OUTBOUND-TO-SIPTRUNK
translation-profile outgoing TO_SIPTRUNK
session protocol sipv2
session target dns:sip2.telezero.net
session transport udp
destination e164-pattern-map 100
voice-class codec 100
voice-class sip early-offer forced
voice-class sip profiles 100
voice-class sip options-keepalive up-interval 15 down-interval 15 retry 2
voice-class sip bind control source-interface GigabitEthernet0/0.501
voice-class sip bind media source-interface GigabitEthernet0/0.501
dtmf-relay rtp-nte
fax-relay ecm disable
fax protocol pass-through g711alaw
no vad
!
dial-peer voice 199 voip
description INBOUND-FROM-SIPTRUNK
translation-profile incoming FROM_SIPTRUNK
session protocol sipv2
incoming uri via SIPTRUNK
voice-class codec 199
voice-class sip early-offer forced
voice-class sip profiles 199
voice-class sip bind control source-interface GigabitEthernet0/0.501
voice-class sip bind media source-interface GigabitEthernet0/0.501
dtmf-relay rtp-nte
fax-relay ecm disable
fax protocol pass-through g711alaw
no vad
!
dial-peer voice 9899 voip
description INBOUND-FROM-CUCM
translation-profile incoming FROM_CUCM
session protocol sipv2
session server-group 1000
incoming uri via CUCM
voice-class codec 9899
voice-class sip bind control source-interface Loopback1
voice-class sip bind media source-interface Loopback1
voice-class sip midcall-signaling passthru media-change
dtmf-relay rtp-nte sip-notify
fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 2 fallback pass-through g711alaw
no vad
!
!
gateway
timer receive-rtp 1200

sip-ua
credentials username 22751 password 7 104A0217071C075B1412 realm telezero.net
authentication username 22751 password 7 045F00080D2A591E110F realm telezero.net
retry invite 3
retry register 10
timers connection aging 15
registrar 1 dns:sip2.telezero.net expires 60 auth-realm telezero.net
sip-server dns:sip2.telezero.net:5060
connection-reuse via-port
permit hostname dns:sip2.telezero.net

gatekeeper
shutdown
!
!
ephone-type 7925G
device-id 485
device-name WiFi phone
num-buttons 1
max-presentation 1

telephony-service
sdspfarm units 2
sdspfarm transcode sessions 4
sdspfarm tag 1 DSPCFB9901
sdspfarm tag 2 DSPXC9902
conference hardware
max-ephones 6
max-dn 6
ip source-address 172.22.148.160 port 2000
service phone webAccess 0
service dnis dir-lookup
timeouts interdigit 2
url directories http://172.22.148.160/localdirectory
load 7925G CP7925G-1.4.8.4.LOADS
time-zone 22
time-format 24
date-format dd-mm-yy
max-conferences 2 gain -6
moh enable-g711 "flash:/cme/music-on-hold.au"
time-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Mar 30 2023 12:01:33

0 Helpful

Mohammed-A.R.A
Level 1
Level 1

‎01-17-2024 03:05 PM - edited ‎01-17-2024 03:06 PM

Where is the forwarding applied?

Maren Mahoney
VIP
VIP

‎01-18-2024 10:02 AM

Call Forward All requires that a CSS is applied to a directory number. That CSS is two-fold: One part is the "Calling Search Space Activation Policy" which is set in the CallManager Service parameters. The second is a CSS applied on the directory number.

So, a policy must be applied and a CSS applied, and those CSSes must include access to a Route Pattern that will allow the call to the external number to complete. Additionally, the number entered in CallForwardAll must be in a format that matches a Route Pattern.

Can you provide a screenshot of the Call Forward section of your DNs configuration, and what the Calling Search Space Activation Policy is set to?

Maren

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