05-11-2017 06:32 PM - edited 03-17-2019 10:19 AM
Hi,
We have a UC520 as PBX. Recently we want to use external SIP service via SIP trunk.
I have set up sip-
But there are two problems that are yet to be solved:
1. DID internal extension via SIP.
e.g. our
I want to achieve:
People from outside ring 87654322 will go to extension 101.
87654323 -> 102
87654324 -> 109
87654321 -> 110
voice translation-rule 100
rule 1 /87654321/ /110/
rule 2 /87654322/ /101/
rule 3 /87654323/ /102/
rule 4 /87654324/ /109/
I use voice translation rules for 'incoming called'. But we only see 87654321 as incoming called number. 86754322 only appears in SIP message 'TO:' and does not get translated to 101. So calling 87654322/3/4 all reach extension 110 instead of their intended extensions.
How can I translate DID number from 'TO:' to internal extensions?
2. When calling from internal to external through SIP, everything seems good.
Our UC520 is behind Cisco 887. I already set
in cisco 887,
Could you please help?
Many thanks
05-12-2017 12:59 PM
Can you provider scrubbed invites with replacement numbers/names/IPs for us to look at? The Request header should be where destination routing is done.
The providers inbound Invite will likely be UDP. The SIP ALG for UDP and the NAT UDP timers would be something to suspect for the no audio portion.
05-12-2017 03:17 PM
05-15-2017 04:26 PM
Hi All,
I am glad to report that I have successfully differentiated called numbers using "incoming
Step 1 translate generic incoming number(87654321) to various extensions:
voice translation-rule 3009
rule 1 /87654321/ /101/
voice translation-rule 3010
rule 1 /87654321/ /102/
voice translation-rule 3011
rule 1 /87654321/ /109/
Step 2 create voice class uri for various DID numbers:
voice class uri 200 sip
pattern 87654322@my_ip
voice class uri 201 sip
pattern 87654323@my_ip
voice class uri 202 sip
pattern 87654324@my_ip
Step 3 translation profiles:
voice translation-profile IN_87654322
translate calling 3100
translate called 3009
!
voice translation-profile IN_87654323
translate calling 3100
translate called 3010
!
voice translation-profile IN_87654324
translate calling 3100
translate called 3011
Step 4. Dial-peers
dial-peer voice 2109 voip
translation-profile incoming IN_87654322
voice-class codec 1
voice-class sip asserted-id pai
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
incoming uri to 200
dtmf-relay rtp-nte sip-notify
no vad
!
dial-peer voice 2110 voip
translation-profile incoming IN_87654323
voice-class codec 1
voice-class sip asserted-id pai
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
incoming uri to 201
dtmf-relay rtp-nte sip-notify
no vad
!
dial-peer voice 2111 voip
translation-profile incoming IN_87654324
voice-class codec 1
voice-class sip asserted-id pai
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
incoming uri to 202
dtmf-relay rtp-nte sip-notify
no vad
!
Now incoming calls can correctly reach different extensions.
Move onto the no-voice in-coming calls problem.....
Thanks
Calvin
05-16-2017 01:09 AM
Calvin,
Thanks for posting back your solution.
Are you saying that your translation rules are now matching? I am curious because normally Cisco gateway will route request based on the request uri. We can use several sip headers like to, from, via etc to match a specific dial-peer but the called number will still be used ( in the RURI).
Out of curiosity can you please send the following
debug voip dialpeer inout
debug voice translation
debug voip ccapi inout
05-16-2017 02:31 AM
Hi Ayodeji,
The matching is by
incoming
My particular IOS 12.4(11r)XW doesn't support voice class sip-profiles. So after matching 'TO', I translate the same incoming called number to different extensions.
Also the no voice problem is caused by codec mismatch. I still don't understand.
I had:
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
Then in the
Incoming
After removing voice-class codec 1 in dial-peers, I can hear
But the sound quality is not great, much worse than calling on the ISDN BRI port. Is there a way to force
Now there is another issue. If I set an extension to forward all, calling that extension via DID gives me
Thanks
Calvin
05-16-2017 03:26 AM
For the call forward we need to see the sip logs to know whats going on..
to please send us a "debug ccsip mess" on the UC520
Also your IOS is too old. Consider upgrading it.
For the second question, its better to ask your provider to send you G711u. There is nothing you can do in these scenario
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