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? 

[Amjad khan]

These are actual rules in CME router.
voice translation-rule 2
rule 1 /^9\(.*\)/ /\1/
voice translation-rule 4
rule 1 /^69\(...\)/ /+9714XXX1\1/
voice translation-profile 4
voice translation-profile Strip_9
translate calling 4
translate called 2
translation-profile outgoing Strip_9

The Call goes out but i hear Arabic Message the number is wrong

[TONY SMITH]

Can you post an example of your actual outgoing Invite to the service provider.  Better still one for each call type, local international etc.   If there's nothing obviously amiss then it's going to be a matter of asking the carrier what's wrong with your presentation.

[Amjad khan]

They are asking me to change SIP Invite? They will send me format. But any earlier suggestions for what to change sip invite in sip profiles?

[Amjad khan]

I changed Session target Server from "Proxy Server Address" to "Domain " and outgoing started working. Now Incoming calls are not coming. Any suggestions?

[TONY SMITH]

We'd need to see debugs from when incoming calls were working, and from now, so we can see the different.   Can you also share an outbound Invite from before these changes, the instructions from the service provider, and the configuration changes that you made.

[Amjad khan]

Hi,
Thanks for the Support. Outgoing Issue is resolved. Incoming calls are coming but on random extensions. I suspect one issue. My sip extensions are also getting registered with Service provider somehow as seen below.
Router#show sip-ua register st

Tenant: 1
--------------------- Registrar-Index 1 ---------------------
Line peer expires(sec) reg survival P-Associ-URI
================================ ========= ============ === ======== ============
+97145XXXXXX -1 10 yes normal +97145XXXXXX
+97145XXXX!*
+97145XXXXXX
+97145XXXX!*

Tenant: 1
--------------------- Registrar-Index 1 ---------------------
Line peer expires(sec) reg survival P-Associ-URI
================================ ========= ============ === ======== ============
69201 40010 10 yes normal +97145XXXXXX
+97145XXXX!*
+97145XXXXXX
+97145XXXX!*

Tenant: 1
--------------------- Registrar-Index 1 ---------------------
Line peer expires(sec) reg survival P-Associ-URI
================================ ========= ============ === ======== ============
69203 40005 12 yes normal +97145XXXXXX
+97145XXXX!*
+97145XXXXXX
+97145XXXX!*

Tenant: 1
--------------------- Registrar-Index 1 ---------------------
Line peer expires(sec) reg survival P-Associ-URI
================================ ========= ============ === ======== ============
69204 40005 1848 yes normal +97145XXXXXX
+97145XXXX!*
+97145XXXXXX
+97145XXXX!*

Tenant: 1
--------------------- Registrar-Index 1 ---------------------
Line peer expires(sec) reg survival P-Associ-URI
================================ ========= ============ === ======== ============
69205 40003 13 yes normal +97145XXXXXX
+97145XXXX!*
+97145XXXXXX
+97145XXXX!*

The dial-peer i configured for incoming calls never gets hit. The extension dial peers like 40003 40005 40007 get the calls and randomly routing the calls. Any suggestion for this?

[Roger Kallberg]

Have a look at this post. https://community.cisco.com/t5/ip-telephony-and-phones/pots-dial-peer-trying-to-register-with-sip-registrar/td-p/1764632
It should have what you need to stop the dial peers to register with your SIP service provider.

[Amjad khan]

This was helpful.

[TONY SMITH]

Not sure if the incoming dial peer issue could be related to registration issue.  If it persists once that's fixed, we'd need to see your incoming dial peer configuration, and anything it relies on.

[Amjad khan]

on my incoming calls, i am getting SIP 404 continuously now. Can you suggest anything?

[TONY SMITH]

---

@Amjad khan wrote:
on my incoming calls, i am getting SIP 404 continuously now. Can you suggest anything?

---

404 is "Not Found" suggesting the called number in the incoming Invite isn't being matched.  Something up with dial peers and/or translation rules.  We'd need to see the configuration, and the incoming Invite to be more specific.

[Amjad khan]

I created an ACL and Now no Phone is registering with Service Provider Sip trunk. Only SIP UA is getting registered.

[Roger Kallberg]

---

@Amjad khan wrote:
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?

---

For this the recommendation would be for you to look at using voice class tenant configuration to have the SIP trunk registration in that instead of under sip-ua that is global.

voice class tenant 2000
 credentials username <XXXX> password <secret> realm <SP realm>
 authentication username >XXXX> password <secret> realm <SP realm>
 registrar dns:<SP registrar dns name> expires 3600
 sip-server dns:<SP SIP server dns name>
 bind control source-interface GigabitEthernet0/0/1
 bind media source-interface GigabitEthernet0/0/1
 outbound-proxy dns:<SP dns name for proxy> reuse
 session transport tcp
 audio forced
 connection-reuse

You then use this on the inbound and outbound dial-peers to/from the ITSP.

dial-peer voice 100 voip
 description Inbound calls from PSTN
 translation-profile incoming PSTN-IN
 session protocol sipv2
 incoming uri via PSTN
 voice-class codec 2000  
 voice-class sip tenant 2000
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 110 voip
 description Outbound calls to PSTN
 translation-profile outgoing PSTN-OUT
 session protocol sipv2
 session server-group 2000
 destination e164-pattern-map 2000
 voice-class codec 2000  
 voice-class sip profiles 10
 voice-class sip tenant 2000
 voice-class sip options-keepalive profile 2000
 dtmf-relay rtp-nte
 no vad

[Rizwan Haider]

Hi, are you able to get it resolved?

401 - unauthorized from Telco. error. can you please share running configuration?