05-12-2008 11:49 PM - edited 03-15-2019 10:36 AM
Hi folk,
my topology is like this:
- sip-ua to a SIP ISP
- nokia e65 registered on uc500/cme via SIP
- 7960 registered on uc500/cme via SCCP
problem:
when I try to call a .T number through the sip ISP dial-peer, the call from the phone 7960 works correctly (with the command "calling-info pstn-to-sip from number set" on sip-ua config), but no way to use the Nokia mobile phone. But the mobile phone call the number associated to 7960 (601). I see that the Nokia number is registered on uc500/cme as 651@sip.mydomain.dom. Anyway the most important problem that I see is that when I try to call for example 111222, on my mobile phone I see 111222@sip.mydomain.dom instead of 111222@sip.ispdomain.dom.
Any advice will be appreciated, I hope that's clear enough.
Thank you very much for your support
Regards
Andrea
my config is:
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
sip
registrar server expires max 600 min 60
no update-callerid
!
!
voice register global
mode cme
source-address 192.168.17.65 port 5060
max-dn 56
max-pool 14
date-format D/M/Y
!
voice register dn 1
number 651
name Nokia mobile phone
no-reg
!
!
voice register pool 2
id mac 0019.794A.99DB
number 1 dn 1
voice-class codec 2
username test password test
!
!
dial-peer voice 100 voip
description ISP SIP
destination-pattern .T
voice-class codec 1
session protocol sipv2
session target ipv4:212.97.59.76:5061
session transport udp
incoming called-number .T
dtmf-relay sip-notify rtp-nte
no vad
!
!
sip-ua
keepalive target dns:sip.messagenet.it:5061
authentication username 530xxxx password 7 xxxxxxxxxxxxxx
nat symmetric role passive
nat symmetric check-media-src
calling-info pstn-to-sip from number set 530xxxx
retry invite 4
retry response 4
retry bye 4
retry cancel 4
timers expires 300000
timers register 100
registrar dns:sip.messagenet.it:5061 expires 3600
sip-server dns:sip.messagenet.it:5061
!
!
05-13-2008 05:57 AM
In a word:
how could I say: the CME sip server will hand only 6.. internal numbers (that is 6..@sip.mydomain.dom), and will use the dial-peer voice 100 voip for all other calls (.T@sip.ispdomain.dom) ??
Thanks for your support
Regards
Andrea
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