12-09-2012 12:10 PM - edited 03-16-2019 02:37 PM
Hello All,
Apologies in advance for the long winded post. helping with the config woudl be awesome, but I'm not expecitng that and some concise links would be aprreciated.
I've been asked to setup a SIP trunk on the office UC540. Using CCA I thought it would not be a problem, but suffice it to say that when I applied the settings through CCA it broke a lot of stuff that one of the other guys had been doing through CLI - He has left so need to work out on my own.
In summary we have a SIP account with a provider and I've been doing a lot of reading but confusion has set in. Can anyone point me to a good CLI example of the following:
- Setup of SIP trunk
- Dial Peers. In summary I need to be able to press say 8....... to get a Line through SIP and 9...... to get a line through FXO
- An example of how to Map SIP DDI's to internal Extensions.
This is what I have from the Provider:
Number of Channels: 4
IP Address: 178.AA.42.156
SIP signalling gateway address: 88.BB.61.195
- UDP port 5060 egress/ingress to 88.BB.61.195 (IPDC SIP signalling gateways).
- All UDP ports between 6000 - 40000 egress/ingress to 88.BB.61.196 (IPDC Media gateway) Omission of this setting will result in one-way speech
DDI 02034478900- 02034478910
From what I can gather they haven't provided a Username/password and are authenticating my use of the service via my source IP so I assume something like as follows:
Internal Extensions are 2000-2100
voice translation-profile PROVIDER
translate calling 2010
translate called 2011
translate redirect-called 2011
dial-peer voice 888 voip
description SIP to Provider
translation-profile outgoing PROVIDER
preference 1
destination-pattern [8]...
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4:88.BB.61.195
session transport udp
voice-class codec 2
dtmf-relay rtp-nte
fax-relay ecm disable
no vad
I assume I have to create a copy of above for every destination pattern that is present for the FXO to go out the SIP when preceeding 8 is pressed.
No idea on how to map the DDI's to an internal extension.
Am I missing anything?
I should also mention there is another SIP trunk to another UC520 over a VPN. This is what gets messed up when I use the CCA.
Thanks in advance
Dave
Solved! Go to Solution.
02-04-2013 11:34 AM
David,
Configur this on your gateway
Router(config)#voice service voip
Router(conf-voi-serv)#no supplementary-service sip moved-temporarily
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
02-04-2013 11:51 AM
Thanks for helping Aokanlawan
When I apply no supplementatary-service sip moved-temporarily I cant see any translation of the DDI let alone the iversion trying to kick in (see attached)
When I go be to the config below I see the CCSIP message trying to divert to 2810
voice service voip
ip address trusted list
ipv4 10.2.0.0 255.255.255.0
ipv4 88.215.61.0 255.255.255.0
ipv4 10.1.100.0 255.255.255.0
ipv4 10.1.37.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
fax protocol none
no fax-relay sg3-to-g3
sip
registrar server
no update-callerid
early-offer forced
midcall-signaling passthru
CCSIP Message trying to divert
158501: 1w4d: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:2810@10.44.1.254;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <>2810@88.215.61.195:5060;user=phone>>
From: <07825372765>;tag=3568995155-52300307825372765>
P-Asserted-Identity: <07825372765>07825372765>
Call-ID: 711350-3568995155-523000@MSX28.gammatelecom.com
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK3b6aa7d01afb2845ff14f2f6c6435a22
Contact: <07825372765>07825372765>
Diversion: <2805>;reason=unconditional;counter=12805>
Content-Type: application/sdp
Accept: application/sdp
02-04-2013 12:24 PM
David,
Send your config again and enable the supplementary services moved temp.
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
02-04-2013 12:35 PM
02-04-2013 05:24 PM
David the first thing I see from your config is that the you are forwarding from SIP phone 2810 to SCCP phone 2805.
Can you try and do a call forward from a sip phone to another sip phone and see if that works.. If it doesnt send me the debug ccsip all.
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
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