Cisco Support Community
cancel
Showing results for 
Search instead for 
Did you mean: 
Announcements

Welcome to Cisco Support Community. We would love to have your feedback.

For an introduction to the new site, click here. And see here for current known issues.

New Member

SIP trunk debugging

I've got the following setup with CME 3.2 on a 2811 router. Outgoing / Incoming calls dont work. I'm not 100% sure if the problem is with my configuration or if my SIP account isnt setup correctly.

In "show sip-ua connections udp detail" I have:

Remote-Agent:87.x.x.101, Connections-Count:1

Remote-Port Conn-Id Conn-State WriteQ-Size

=========== ======= =========== ===========

5060 1 Established 0

Which leads me to believe the sip trunk is fine.

The following is my config:

voice-card 0

no dspfarm

!

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

h323

sip

header-passing

registrar server expires max 3600 min 3600

!

voice class codec 1

codec preference 1 g711ulaw

!

voice translation-rule 9

rule 1 /^9\(.*\)/ /\1/

rule 2 /^911$/ /911/

!

voice translation-profile PSTN_Outgoing

translate called 9

!

control-plane

!

dial-peer voice 100 voip

description Outgoing Call to SIP

translation-profile outgoing PSTN_Outgoing

destination-pattern 9T

voice-class codec 1

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

!

dial-peer voice 101 voip

description Incoming call from SIP

voice-class codec 1

session protocol sipv2

session target sip-server

incoming called-number .%

dtmf-relay rtp-nte

no vad

!

sip-ua

authentication username <username> password <password>

no remote-party-id

retry invite 2

retry register 10

timers connect 100

mwi-server dns:proxy.entacall.com expires 3600 port 5060 transport udp unsolicited

registrar dns:proxy.entacall.com expires 3600

sip-server dns:proxy.entacall.com

!

telephony-service

load 7910 P00403020214

load 7960-7940 P00305000301

max-ephones 24

max-dn 48

ip source-address 192.168.10.10 port 2000

auto assign 1 to 1

system message Comtek

network-locale GB

create cnf-files version-stamp 7960 Aug 16 2006 13:05:02

max-conferences 8

call-forward pattern .T

dn-webedit

transfer-system full-consult

transfer-pattern 9.T

secondary-dialtone 9

!

ephone-dn 1

number 2000

description Common

name Common

!

ephone-dn 5 dual-line

number 2004

label Phone One

description CIPC Phone One

name Phone One

call-forward busy 2006

call-forward noan 2006 timeout 15

no huntstop

!

!

ephone-dn 6

number 2005

label Phone Two

description CIPC Phone Two

name Phone Two

call-forward busy 2006

call-forward noan 2006 timeout 15

no huntstop

hold-alert 30 originator

!

ephone 4

mac-address 0030.4883.CA01

type CIPC

button 1:5 2:1 3:4

!

ephone 5

mac-address 0000.E254.B03C

type CIPC

button 1:6 8:19

!

ephone 6

mac-address 0003.0D03.1128

type CIPC

button 1:7 8:20

!

Is there anything obviously wrong with that? Or should I be contacting my SIP provider.

19 REPLIES
New Member

Re: SIP trunk debugging

You don't need the "allow-connections" commands, they are only used in IP2IP gateways (i.e., using the router to convert a SIP trunk to H323 or vice-versa).

Try some debugs like "debug ccsip errors" or "debug ccsip events"

New Member

Re: SIP trunk debugging

With them commands i get the following when trying to place a call.

.Aug 17 14:51:53 BST: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event

from SIP SPI : SIPSPI_EV_CC_CALL_SETUP

.Aug 17 14:51:53 BST: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event

from SIP SPI : SIPSPI_EV_DNS_RESOLVE

.Aug 17 14:51:57 BST: //47/5EF05A508058/SIP/Error/act_sentinvite_wait_100: Out o

f retries

.Aug 17 14:51:57 BST: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event

from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT

And with "debug ccsip calls"

.Aug 17 15:06:42 BST: //61/6E5E50C48076/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x457DB14C

State of The Call : STATE_DEAD

TCP Sockets Used : NO

Calling Number : 2006

Called Number : 01642

Source IP Address (Sig ): 192.168.10.10

Destn SIP Req Addr:Port : 87.x.x.101:5060

Destn SIP Resp Addr:Port : 87.x.240.101:5060

Destination Name : proxy.entacall.com

.Aug 17 15:06:42 BST: //61/6E5E50C48076/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream : 1

Negotiated Codec : No Codec

Negotiated Codec Bytes : 0

Negotiated Dtmf-relay : 0

Dtmf-relay Payload : 0

Source IP Address (Media): 192.168.10.10

Source IP Port (Media): 18672

Destn IP Address (Media): 0.0.0.0

Destn IP Port (Media): 0

Orig Destn IP Address:Port (Media): 0.0.0.0:0

.Aug 17 15:06:42 BST: //61/6E5E50C48076/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC) : 102

Disconnect Cause (SIP) : 200

With the debug ccsip calls command it looks like its working to me. but i've not got much experience with call manager express so i'm not sure if the error is with my setting up of the SIP trunk or if its the other end.

Thanks for the help so far Matt.

New Member

Re: SIP trunk debugging

So i've moved onto trying to connect a SIP softphone (X-Lite) to my SIP provider. That doesnt work either, so my problem doesnt appear to be with my CME setup.

What I would like to check is what ports are needed to be open / forwarded?

Thanks.

David

New Member

Re: SIP trunk debugging

You will need to open the SIP control port (5060, or 5061 for TLS) and ports 10000 to 10999 for RTP voice bearing traffic.

I must apologize for not seeing your earlier post, I apparently missed the reply notification email.

New Member

Re: SIP trunk debugging

Another note, the RTP ports may differ based on your provider. My provider happens to use the above ports, but Cisco SIP endpoints use the same as SCCP, 16384 to 32768.

New Member

Re: SIP trunk debugging

Perhaps I should have figured all of this out beforehand. I've found several different situations, one person had to open 5060 to 5090 TCP/UDP and 10000-20000 UDP. RTP ports will depend on your provider, but I thought the 31 ports for SIP control was quite interesting.

New Member

Re: SIP trunk debugging

Thanks Matt.

I really need to get a response from my SIP provider about the specific ports. I've fired them an email and they're dragging their heels getting back to me.

Does the actual CME configuration to use the trunk look ok though? That's all i really need to nail down at the moment.

Many thanks.

David

New Member

Re: SIP trunk debugging

The config looks good, however I would remove the voice service commands for allow connections sip / h323 etc. Those are only necessary for IP2IP gateway conversions.

New Member

Re: SIP trunk debugging

sorry. double post.

New Member

Re: SIP trunk debugging

Thanks Matt.

I believe I've we've nearly cracked it. There's one thing specifically that I need to fix which is the Calling Number of my sip calls. Here is a part of my debug logs:

Call Control Block (CCB) : 0x4838BFD4

State of The Call : STATE_DEAD

TCP Sockets Used : NO

Calling Number : 2000

Called Number : 01833xxxxxx

Source IP Address (Sig ): 83.x.x.x

Destn SIP Req Addr:Port : 87.x.x.x:5060

Destn SIP Resp Addr:Port : 87.x.x.x:5060

Destination Name : proxy.sip.com

The calling number "2000" is that of the ephone-dn that places the outgoing call. I need to change this to my SIP account name, which is a telephone number "441833xxxxxx".

Would that be achieved with Translation Rules?

New Member

Re: SIP trunk debugging

Finally I can make outgoing calls.

Just to complete the thread off the translation rules i used were.

translation-rule 1

Rule 0 ^90 0

!

!

translation-rule 2

Rule 0 ^2000 441833xxxxxx

With this in the dial-peer

dial-peer voice 2000 voip

description Outgoing Call via SIP

translate-outgoing calling 2

translate-outgoing called 1

thanks once again.

New Member

Re: SIP trunk debugging

Hello, I have a cisco 2651 with ccme 3.3 and I want to route my external calls PSTN through my SIP provider.

parameters form provider are:

sip.provider.com

login and password port 5060

I have configured this in the sip-ua but I have problems.

All of the IP phones I have behind the router try to authenticate.

I only want 1 authentication from the router

Then i need to tell the internal extensions when they have to go to the PSTN via the SIP provider.

Please can you help me? Is there any example for connecting tipical sip providers?

New Member

Re: SIP trunk debugging

You will need to create an outgoing translation pattern to convert your internal numbers to your SIP number. That way when any outbound call is going to the SIP provider, it will replace the phone extension with the SIP phone number.

Check the attached configuration using Viatalk as the SIP provider.

New Member

Re: SIP trunk debugging

I am trying it but i'm very upset because I cannot get it.

I have configured the sip-ua but the only things that are trying to register in the SIP proxy are the IP phones.

I don't want that I want only register one time with the account I have configured in the authentication and then route calls with pattern 9........ to the sip provider

Please I need help

This is my configuration:

dial-peer voice 100 voip

destination-pattern 9........

session protocol sipv2

session target sip-server

codec g711ulaw

!

sip-ua

authentication username xxx password xxx

retry register 10

registrar ipv4:213.162.201.146 expires 60

sip-server ipv4:213.162.201.146

!

!

gatekeeper

shutdown

!

!

telephony-service

max-ephones 2

max-dn 2

ip source-address 192.168.3.1 port 2000

auto assign 1 to 2

user-locale ES

network-locale ES

create cnf-files version-stamp Jan 01 2002 00:00:00

max-conferences 4 gain -6

transfer-system full-consult

!

!

ephone-dn 1 dual-line

number 1 no-reg both

!

!

ephone-dn 2 dual-line

number 2 no-reg both

!

!

ephone 1

mac-address 0013.60C3.CE02

type 7902

button 1:1

!

!

!

ephone 2

mac-address 0012.431E.BF3E

type 7902

button 1:2

!

The logs I receive in the SIP proxy when I try to call to an external number like 964525351 are:

Admission Request BB00000149 : 1@192.168.2.200 => 964525351

$I=BB00000149 exec ifone..sp_gk_start_call '192.168.2.200','','1','964525351','BB00000149','1','Oct 07 2006 19:10:34','',1,'20','','','','DE813E7E-411611D6-80B2C5E4-BE2FDD31','213.162.201.146','2.5.0000@Cisco-SIPGateway/IOS-12.x'

go

BB00000149|STOP|404|>>APPL: AUTH FAILURE/UNKNOWN USER[CODE:2 ACCTID:-1]

Admission Reject BB00000149 : 1@192.168.2.200 => 964525351

I am very frustrated so your help will be great.

New Member

Re: SIP trunk debugging

"Admission Reject BB00000149 : 1@192.168.2.200"

That to me looks like your SIP provider is regecting you because your details to them are wrong.

I had a similar problem when setting up my sip trunk. When i first tried to make external calls i was trying to setup using a SIP address like "2000@81.130.170.76" when it should have been "username@81.138.170.76" the 2000 in that was my phone extension and the ip is my routers IP.

To fix that what i did was an outgoing translation rule which replaces 2000 with my SIP username.

This is what i have for my outgoing dial-peer:

dial-peer voice 2000 voip

description Outgoing Call via SIP

translation-profile outgoing SIPout

destination-pattern 9T

voice-class codec 1

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

with the following translation profile applied to it:

voice translation-profile SIPout

translate calling 2

translate called 1

voice translation-rule 1

rule 1 /^90/ /0/

rule 2 /^80/ /0044/

!

voice translation-rule 2

rule 1 /^2.../ /441833600000/

If you do:

voice translation-rule 1

rule 1 /1/ /964812530/

voice translation-profile SIPout

translate calling 1

dial-peer voice 100 voip

translation-profile outgoing SIPout

That might let you dial out on ephone 1.

But that SIP log is still showing your private IP address in the SIP address. I'm not sure why that's there.

Apply what i have suggested, then capture the "debug ccsip all" logs while attempting a call and post them back here. I'll see if i can help out at all.

New Member

Re: SIP trunk debugging

Ok I will test it and will tell you the results. But I have a doubt. As you can see in the config file before I have configured the sip-ua

authentication username xxx password xxx

retry register 10

registrar ipv4:213.162.201.146 expires 60

sip-server ipv4:213.162.201.146

First of all, the account configured in the router doesn't try to authenticate in the sip proxy.

Normally all the endpoints register in this way: This is the log of registered users in the sip proxy:

??

Registered:

User: ID00000015 - 964812507@195.53.203.2:1027; Dynamic

User: ID00000894 - 964812520@195.53.203.2:5060; Dynamic

User: ID00000738 - 964812524@201.114.222.102:5060; Dynamic

User: ID00000006 - 964812509@81.203.4.91:32768; Dynamic

So I thought that first of all the Cisco cme had to be authenticated in the sip proxy.

I only can see trying authentications from internal extensions.

And every telephone try to authenticate separately ( for example if I have 15 telephones ) I have 15 extensions trying to autheticate.

Is this normal? It'd be better only one authentication from the router, in my case

964812530@195.53.203.5

New Member

Re: SIP trunk debugging

I have done some test and I have got to translate all the extensions for outgoing calls to the username account for authentication. So I can do external calls.

I have created this rules:

voice translation-rule 1

rule 1 /2../ /964812530/

!

!

voice translation-profile SIPout

translate calling 1

dial-peer voice 100 voip

translation-profile outgoing SIPout

destination-pattern 9........

session protocol sipv2

session target sip-server

codec g711ulaw

!

sip-ua

authentication username 964812530 password 03256C2A5E5E6B1B16584146

retry register 10

registrar ipv4:213.162.201.146 expires 60

sip-server ipv4:213.162.201.146

!

!

gatekeeper

shutdown

!

!

telephony-service

max-ephones 2

max-dn 2

p source-address 192.168.3.1 port 2000

auto assign 1 to 2

user-locale ES

network-locale ES

create cnf-files version-stamp Jan 01 2002 00:00:00

max-conferences 4 gain -6

transfer-system full-consult

!

!

ephone-dn 1 dual-line

number 200

!

!

ephone-dn 2 dual-line

number 201

But I have a doubt. As you can see in the config file before I have configured the sip-ua

authentication username xxx password xxx

retry register 10

registrar ipv4:213.162.201.146 expires 60

sip-server ipv4:213.162.201.146

First of all, the account configured in the router (username 964812530 pass *****)doesn't try to authenticate in the sip proxy.

Normally all the endpoints register in this way: This is the log of registered users in the sip proxy:

??

Registered:

User: ID00000015 - 964812507@195.53.203.2:1027; Dynamic

User: ID00000894 - 964812520@195.53.203.2:5060; Dynamic

User: ID00000738 - 964812524@201.114.222.102:5060; Dynamic

User: ID00000006 - 964812509@81.203.4.91:32768; Dynamic

So I thought that first of all the Cisco cme had to be authenticated in the sip proxy.

I only can see trying authentications from internal extensions.

And every telephone try to authenticate separately ( for example if I have 15 telephones ) I have 15 extensions trying to authenticate.

Is this normal? It'd be better only one authentication from the router, in my case

964812530@195.53.203.5

New Member

Re: SIP trunk debugging

Try adding "no-reg" on the end of your "number 2xx" command in the ephone-dn section:

ephone-dn 1 dual-line

number 200 no-reg

!

ephone-dn 2 dual-line

number 201 no-reg

New Member

Re: SIP trunk debugging

Ok I have tried it and I have fixed it.

How can I do for having a second dial tone for routing the external calls?

For example press 0 for doing the external calls.

1400
Views
9
Helpful
19
Replies
CreatePlease login to create content