Connecting the UC520 with a Sip trunk from an ITSP (Internet Telephony Service Provider) NexVortex
After seeing a few posts on the site regarding ITSPs I decided to bite the bullet and try one out. I chose nexVoterx at random and must say I was impressed. They had a very automated setup that assisgned me my new phone numbers in minutes. I got three numbers and they were in my local area code, though I could have chosen a number from anywhere in the US.
I used the Cisco configuration assistant to configure the SIP trunk and it did a great Job with one noteable exception.
All Calls to my phones work fine, but and calls directed to the auto attendant or transferred to voicemail fail. I found there were three main settings to change to give me the functionallity I needed.
Step 1
Type the following to allow hairpinnig of calls to the Unity Express module.
config t
voice service voip
allow-connections sip to h323
Step 2
Change the DTMF Relay type to rtp-nte under the autoattendant's dial peer.
dial-peer voice 2001 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 498
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
Step 3
Allow SIP through the firewall to the Unity Express Module
The Cisco Configuration Assistant does an OK job of letting SIP Traffic in (from the internet) to phones, but the firewall must be manually modified to allow SIP traffic from the internet to the auto attendant. If you have a private SIP connection on an internal interface you should not have to modify the firewall.
I found this Guide at Cisco's site for configuring SIP on the Router based firewall.
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Comments
Internet Telephony Service is
Internet Telephony Service is superb but there are some demerits of Internet telephony service as well. We can manage more calls as like we can do in telephone through
hosted PBX. This is the reason why the internet telephony is not super seating the real telephone.
SIP with Key System
I want to use my UC520 as a Key System with SIP Lines. Is that possible?
I'm using Callcentric.com It
I'm using Callcentric.com
It works fine as a generic SIP Trunk provider programmed straight from the Configuration Assistant, as long as you check that your ACLs are open on TCP/UDP ports 5060-5080 (config assist doesnt do that).
I did go back and adjust the codec order to match what they prefer, but I dont think its required.
Workaround for voicemail getting cut off on the UC520
We have had callers calling to our customers, and our customers have complained about voicemail messages getting cut off after approximately 25 seconds. We had to add voice-class source interface Loopback0 to the voicemail and auto attendant dial peers.
Here is one of the dial peers, after making that change:
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
destination-pattern 299
b2bua
voice-class source interface Loopback0
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
Adding voice-class source interface Loopback0 to the Auto Attendant and Voicemail dial peers does solve the problem of voicemail messages getting cut off after approximately 25 seconds.
What is Loopback 0
We are not using Loopback 0 in our CME. What is your loopback 0 required to resolve this issue.
Gig 0/0 is External IP
Gig 0/1.2 is Voice VLAN
CUE is on Service-module 1/0 and runs unnumbered from Gig 0/1.2
So, what do I use for Loopback 0?
Loopback 0 is a virtual
Loopback 0 is a virtual interface that you can create at will when in config mode with the command: interface loopback0. After that you can configure it as any other interface.
Normally it is used to give a router a fixed interface , fixed ip-address, that does not go down, and is mostly used to manage the router.
let's try that again
I am quite comfortable with what a loopback interface is, but a refresh never hurts. Thanks.
Our problem is that ALL inbound calls that connect to our CUE voicemail system drop after 30 seconds. From my analysis, it appears we are NOT sending any RTCP packets to the SIP provider during the time the voicemail is being recorded, so the SIP provider drops the call.
From my understanding, expected behaviour would be for the CUE to source RTCP packets every 5 seconds (report interval).
Our topology is:
Gig0/0 <- Internet Service Provider aka NAT outside
Gig 0/1.1 <- Inside Data. (NAT inside).
Gig 0/1.2 <- Inside Voice (NAT Inside)
ip address 172.30.1.1 255.255.255.0
Service Engine 1/0 (CUE)
IP Unnumbered gig 0/1.2
There is a B2BUA configured on the dial peer to connect the inside leg to CUE and the outside leg to the SIP provider.
dial-peer voice 2 voip
destination-pattern 3..
b2bua
session protocol sipv2
session target ipv4:172.30.1.2
dtmf-relay sip-notify
codec g711ulaw
no vad
So, how does "voice-class source interface loopback" help solve the problem? Am I to move the service-engine to run unnumbered off a new subnet?
e.g
int loopback 1
ip address 172.30.2.1
int service-engine1.0
ip unnumbered loopback 1
etc..
dial-peer voice 2 voip
session target ipv4:172.30.2.2
voice-class source interface loopback 1
Why should we expect this to act any differently?
What is causing the loss of the RTCP packets?
Is CUE just simply NOT SENDING RTCP?
Routers are suppose to be deterministic. What is the expected cause of RTCP packet loss that adding "voice-class source interface loopback 1" will resolve and how?
Is there a debug or show command that we can use to validate why the packets are being lost? or, Is CUE just simply NOT SENDING RTCP?
I have also seen a suggestion that this may be resolved by changing the "Inspect" configuration on the router, but the instructions were poorly assembled and did not work.
Any tips on this one would be greatly appreciated.
Cisco Unity Express 7.0.2
CUE Running on NM
voice class source interface
Voice class source interface is binding the SIP call control and I believe RTP and RTCP traffic when using SIP to the loopback. The router when acting as a voice gateway will try to use all gateway/router interfaces for this traffic. This can be problematic especialy in troubleshooting. By binding your traffic to your loopback (an interface which cannot go down) the system should be more reliable. It is best to use a loopback if you have multiple routes to the gateway (for redundancy and reachability). We also use a similar command to typically bind h.323 traffic to the loopback h323-gateway voip bind srcaddr.
Additional SIP Trunking Options for UC520
- Cisco Service Node
- Cbeyond
- AT&T
SIP TRUNK
Hi, is there a way to configure sip truk without translations, and how i can use 2-3 or more phone numbers and sip providers.
ip-ua
correct command is sip-ua. Might want to correct.
Problem with forwarded calls
HI there,
I'm having the same issue as described with call's being dropped when being forwarded to CUE.
setup is system only has a sip trunk (no PSTN trunks).
Inbound/Outbound calls work fine.
Example of main line call flow is
Main number is tied as a secondary number on ephone-dn
This then forwards call's to the pilot of a ephone-hunt sequential hunt group.
this then run's thru all extensions in hunt-group with final being voicemail.
At the beginning of the call the appropriate codec is negotiated (I've tried both g711alaw&g711ulaw) and extensions tied to the hunt group ring.
final is voicemail
ccsip debug's show that no media codec is being provided to the dial-peer for voicemail.
i've double-triple checked that the dtmf-relay & codec settings are the same on both the inbound SIP dial-peer and the voicemail dial-peer
Firstly I've tried the above (sip aware ip inspect voip sip) on both the external and voicemail interface
I've also allowed the on the in access-list sip proxy and external wan gateway address on the voicemail interface as well
I've tried the transcoding steps as well with no luck.
does call-flow go through something else when going to voicemail off a huntgroup ?
appreciate any help with this
rule of thum
hi there, we use sip trunks with voip.co.uk
as a rule of thumb...
use the latest ios/firmware throughout, cisco voip stuff is getting better all the time and adds lots of new features e.g. extension mobility (hotdesking)....
uc500 comes with lots of transcoders as its fixed config - this is good (g729)!
use g729 if using dsl (30 concurrent calls with 1mb)
you need to transcode (quite complex) unless using native g711 (the voicemail/cue) will only use g711.
ensure you are using the uc500 gateway as a firewall (without an additional firewall layer) as otherwise you will need a sip-aware firewall.
depending on the provider you may need to transcode anyway to get dtmf tones transcoded to the voicemail.
version 3.x of cue uses rtp-nte version 2.x uses sip-notify
use translation rules (voip-incoming translation-profile ... ) and on dial-peers to make things easier
in "voice service voip" allow sip to sip (ie trunk to voicemail) (sample below:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
h323
h225 signal overlap
sip
transcoding hint (use show sccp to verify)
sccp local vlan1
sccp ccm [phonevlanip] identifier 1
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register transcodemeup
!
dspfarm profile 1 transcode
codec g729br8
codec g729r8
codec ilbc
codec g723r63
codec g723r53
codec gsmamr-nb
codec g729ar8
codec g729abr8
codec g711ulaw
codec g711alaw
maximum sessions 20
associate application SCCP
sdspfarm units 1
sdspfarm transcode sessions 128
sdspfarm tag 1 transcodemeup
nexvortx - cli for trunk
We are adding a SIP trunk to the UC520 already in production. The last thing I want to do is turn up the CCA and have that screw up a perfectly good configuration. Does anyone have the CLI commands to turn up the nexvortex service as a SIP trunk using the CLI? If so, could you post a link, hint or example?
Final Connect Corp
www.finalconnect.com
We just got some new servers
We just got some new servers in our call center haven't installed them yet but expanding our answering service capability with new agents and extensions.
Here is a SIP config that CCA did that works
dial-peer voice 1000 voip
description ** Incoming call from SIP trunk **
translation-profile incoming CUE_Incoming
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1001 voip
description ** Outgoing call to SIP trunk (Generic SIP Trunk Provider) **
translation-profile outgoing PSTN_Outgoing
destination-pattern 9T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1002 voip
description ** Emergency outgoing call to SIP trunk **
translation-profile outgoing PSTN_Outgoing
destination-pattern 911
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1004 voip
corlist outgoing call-local
description ** star code to SIP trunk **
destination-pattern *..
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
ip-ua
authentication username USERNAME password 7 PASSWORD
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar ipv4:NEXVORTEX IP HERE expires 3600
sip-server ipv4:NEXVORTEX IP HERE
host-registrar
Problem with forwarded calls
I tried to set up a SIP trunk with NexVortex and CME, and direct calls work fine in and out. Call forwarding, however, does not. For example, I can send the call to the AutoAttendant on Unity Express, but when I dial the extension I want, Unity says "transferring to extension xxxx", and then the call drops. I know that DTMF is ok, since it understood the dialed digits, but I cannot figure out why the call is dropping. Any hints welcomed.
maybe try this . . .
add the lines
allow-connections sip to h323
allow-connections h323 to sip
this will allow hairpinning betweeen the two protocals.
Vonage SIP
Would anyone know if this will work with Vonage and a Linksys WRT31P2 BroadBand router?
Do you know of other SIP
Do you know of other SIP trunk providers besides nexVoterx ?
Cisco Certified SIP Trunk Providers for the UC500 in the UK
Hi there just responding to this post. There is only one Cisco Certified Internet Telephony Service Provider (ITSP) for the UC500 in the UK. That company is Breeze8 (www.breeze8.co.uk) and the telephone number is 01635 246030. Breeze8 offer a certified migration plan onto there SIP trunk service for UC500. Breeze8 are also the only certified Cisco Hosted IP Telephony Provider in the UK. If anyone wants to confirm these details and that they have a Cisco verified process for migrating UC500's to SIP. Please feel free to email Philip Smith the head of Cisco UK Service Providers on philismi@cisco.comI am James Harries the Technical Director of Breeze8 and we have over 100 UC500's connected to our service as well as hundreds of Cisco end-points connected via our Cisco Hosted IP Telephony Service. If anyone wants a copy of a migration plan or needs some straight forward advice on their migration plan. Or even to speak to any of our customer that have migrated and how seemless it was. Please email me at james.harries@breeze8.co.uk and I will happily help.