Sip UA Registration Failure

[Amjad khan]

Hi,

I have Cisco ISR4331 with Version (Version 03.16.04b.S )((X86_64_LINUX_IOSD-UNIVERSALK9-M), Version 15.5(3)S4b)
 . I am trying to register sip phones on this and have succeeded. I used bind address command so that sip phones do get registered. but i am not able to register siptrunk with Service provider. 

For sip-ua following are my configs

sip-ua
credentials username +971xxxxxxxx@ims.etisalat.ae password 7 1234 realm ims.etisalat.ae
authentication username +971xxxxxxxx@ims.etisalat.ae password 7 1234 realm ims.etisalat.ae
retry invite 2
retry bye 1
retry register 10
timers expires 60000
timers connect 100
registrar dns:vims-siptrunk.etisalat.ae:5060 expires 3600
sip-server dns:vims-siptrunk.etisalat.ae
no transport tcp
connection-reuse
host-registrar
permit hostname dns:vims-siptrunk.etisalat.ae

X-lite Softphone works fine with the settings but cisco cme doesnot gets registered. Can anybody assist what is wrong? 

[Roger Kallberg]

That’s a quite old version of IOS your using, I would recommend you to upgrade to something more current, for example 16.9.5. For this please also first verify that the ROMMON is on an appropriate level prior to the upgrade of IOS. If not first upgrade ROMMON.

After this please check again if you are able to get it to register.

[Amjad khan]

I already tried to the lastest firmware. Results are same. The reason for downgrade was i tried to add sip authentication password but it gave me error of minimum length of 10 digits. It is a very difficult process for this ISP to change sip authentication password so i downgraded to the factory firmware. 

[George Sotiropoulos]

Dear Amjad,

Your configuration looks fine however, it seems that your IOS is rather old and you better proceed with upgrade prior of testing the registration.

George

[Amjad khan]

Please see above reply

[TONY SMITH]

I don't know how much help this is but I had a quick look at a gateway we configured on Etisalat in Abu Dhabi.  In the sip-ua we have a "credentials number" entry where the number is the billing number in National format as well as the username.  Actual numbers and IP address redacted as this a production system.  That gateway has a dedicated connection for SIP using RFC private addressing, rather than connecting over the Internet.

sip-ua
 credentials number 2xxxxxxx username 2xxxxxxx.etisalat password 7 yyyyyyyyyy realm etisalat.com
 authentication username 2xxxxxxxx.etisalat password 7 yyyyyyyyyy realm etisalat.com
 retry invite 2
 timers trying 300
 registrar dns:xxxxxxxx.etisalat expires 3600
 sip-server ipv4:z.z.z.z
 connection-reuse

[Amjad khan]

Did you bind any interface? Were you using SIP Extensions?

[TONY SMITH]

On that gateway I have bind commands explicitly set on each of the dial peers.  The site uses a mix of SIP and SCCP handsets.

[Amjad khan]

Can you share those configs? i am struggling a little

[Amjad khan]

Can i use same CME router to terminate SIP Trunk as i am using bind source command under voice service voip for sip phones registration?

[TONY SMITH]

Bind configuration on the dial peers will over-ride anything set globally.   What I'm not sure of is how the sip-ua registration binds.   Anyway here are my dial peers with personal information redacted ...

dial-peer voice 9 voip
 corlist incoming CALL-CUCM
 corlist outgoing CALL-PSTN
 description *** Etisalat SIP Trunk In/Out (02xxxxXX) ***
 translation-profile incoming SIP-IN
 translation-profile outgoing SIP-OUT
 preference 1
 destination-pattern 9T
 b2bua
 session protocol sipv2
 session target ipv4:y.y.y.y
 incoming uri via ITSP
 voice-class codec 9
 voice-class sip options-keepalive up-interval 120 retry 1
 voice-class sip bind control source-interface GigabitEthernet0/0/2
 voice-class sip bind media source-interface GigabitEthernet0/0/2
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 20 voip
 corlist incoming CALL-PSTN
 corlist outgoing CALL-CUCM
 description *** SIP Inbound/Outbound CUCM  ***
 destination-pattern 4xxxx..
 session protocol sipv2
 session server-group 20
 incoming uri via CUCM
 voice-class codec 20
 no voice-class sip early-offer forced
 voice-class sip profiles 20
 voice-class sip options-keepalive profile 20
 voice-class sip bind control source-interface Loopback0
 voice-class sip bind media source-interface Loopback0
 dtmf-relay rtp-nte sip-kpml
 no vad

I've just noticed something else.  We have a sip profile configured as default, under voice service, which fixes up the To and From headers in Register requests.  Where 2xxxxxxx is their specific trunk pilot number.

voice service voip
 <skip irrelevant lines> ip address trusted list
 sip
  sip-profiles 10

voice class sip-profiles 10
 request REGISTER sip-header To modify "<sip:.*>" "\"02xxxxxxx\"<sip:02xxxxxxx@2xxxxxxx.etisalat>"
 request REGISTER sip-header From modify "<sip:.*>" "\"02xxxxxxx\"<sip:02xxxxxxx@2xxxxxxx.etisalat>"
!

[Amjad khan]

I tried but still the same. Following is the result.
Router#show sip-ua register status
Line peer expires(sec) reg survival P-Associ-URI
================================ ========== ============ === ======== ============
+9714xxxxxxx -1 163 no normal
69208 40001 0 no normal

[TONY SMITH]

Did you get any sort of documentation or specification from the provider?   Looking at my notes we got the information drip fed to us, initially just the number range and connection IP addresses.  Later on they provided the following ...

Please find below standard SIP trunk configuration by Etisalat.

  SIP trunk Transport layer - UDP
  Codecs Supported          - G711A,G.729 ( Preferred Codec is G.711 A)
  Fax Codec                 - T.38 , G.711 A
  DTMF                      - RFC 2833, In Band raw 
  DTMF.Packetisation time   - 20 ms.
  Public IP                 192.168.1.x ( x= 8 to 62)
  No of calls depends on customer package if 30chanells then 30calls and so on 
  Account number is username
  Registration is required
  Domestic dialing          02xxxyyyy
  International dialing     00971xyyyzzzz

I can also see from the notes that we initially had problems with registration, we sent the provider a debug of our register message, and they responded with the fields they didn't like and the changes they wanted to see, which are the changes in that SIP profile I posted earlier.

We may be able to learn something if you post the SIP debug from the registration attempts.  You may have to include "debug ccsip non-call" as well as "debug ccsip mess".  I think that's platform or version dependent.

[Amjad khan]

Thanks Tony, Your such a great guy. Adding Sip profiles, and a separate Tenant as suggested by Roger Kallberg solved the issue. Now Sip Trunk is Registered. But a small issue still exist. When i dial outside, i receive Service Provider Message that number dialed is in correct, i am pasting my configs here for outbound dial-peer. can you assist if any mistake in my config? The internal Extension series is from 69201 to 69215. DID is +9714XXXX200-99.
voice translation-rule 4
rule 1 /^2\(..\)/ /+9714xxxx2\1/
!
voice translation-rule 2
rule 1 /^9\(.*\)/ /\1/

voice translation-profile Strip_9
translate calling 4
translate called 2

dial-peer voice 3 voip
description *** National Land Line ***
translation-profile outgoing Strip_9
destination-pattern 90[2346789].......

Thanks for The Support

[TONY SMITH]

"My" Etisalat install is in Abu Dhabi rather than Dubai, but as I understand it numbering is consistent across the Emirates.  We definitely place outbound calls in National format, starting with  "0" as your configuration shows.  I've just had a quick look at call records to confirm.

One thing is that your voice translation rule 4 will not match an extension number in the form you show, it's looking for a number beginning with "2".   Could you paste up an outgoing Invite as an example?