CallManager Express to BroadVoice or other SIP Provider

[justincohen]

has anyone been able to connect a CME or a Cisco IOS gateway to an external VoIP Provider such as Broadvoice.

Such services normally can be connected to using a SIP device like a 7960 W sip firmware.

Is there a configuration example?

[gruiz]

I haven't try configuring the CME for Broadvoice but the link below shows how to configure a router with a SIP proxy.

http://www.cisco.com/en/US/products/sw/voicesw/ps2157/products_configuration_guide_chapter09186a00803e5d12.html

[justincohen]

Anyone? Any thoughts?

I've tried a few things, but I need to somehow register with the remote provider.

[sticano]

After much aggravation, I was able to connect to Broadvoice using Asterisk as my middleman. So I have a fully functioning Callmanager 4.1 environment with traditional PRIs, but I also have a sip trunk to my asterisk server which then forwards all calls to Broadvoice. Not the prettiest, but Callmanager does not have the authentication parameters for SIP yet.

[gruiz]

I've seen in the IOS documentation (the link below) support on SIP registration to a proxy. This could mean that the Router could actually register to Broadvoice. The soilution could work by using H323 from the Call Manager to the Gateway and SIP from the Gateway to Broadvoice.

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_configuration_guide_chapter09186a00803e5d12.html#wp1182326

[chmccane1]

Yesterday I configured CME 3.2 to use a SIP trunk to BroadVoice. At this point it is working for inbound calls only. Outbound calls are getting a "404 Not Found" error. Here's the pertinent section of the config at this point:

dial-peer voice 10 voip

preference 1

answer-address 21actualnumberdeleted

destination-pattern 1T

session protocol sipv2

session target dns:proxy-dca.broadvoice.com

dtmf-relay sip-notify

codec g711ulaw

!

sip-ua

authentication username xxxx password xxxx

calling-info sip-to-pstn name set 21deleted

calling-info sip-to-pstn number set 21deleted

calling-info pstn-to-sip from name set 21deleted

registrar dns:sip.broadvoice.com expires 3600

This config works extremely well for inbound calls. The voice quality is outstanding. It's clear that the CME 3.2 router (2620XM) is registering (and authenticating) my PSTN phone number at BroadVoice. When I call my BroadVoice PSTN number from my cell phone, it rings the 7960 (running SCCP) on my desk. The outbound part needs some work. Any ideas would be welcomed.

Thanks,

C. McCane, CCIE#5163

[mfernandez]

Hi.

Perhaps it depends on SIP provider specific authentication issues. I get to work with some voip providers, but not with others.

SIP Digest authentication at sip-ua level to provide sip proxy server globally and registrar. But I used it also at dial-peer level to provide several numbers registering.

sip-ua

authentication username xxxxxx password yyyyyyy

retry register 3

registrar dns:voiptalk.org expires 3600

sip-server dns:voiptalk.org

For outgoing. Allowed once you are authenticated with the calling num you are using.Important dtmf rtp-nte for SIP servers such as SIP Express or Asterisk:

dial-peer voice 7004 voip

description Conexion con VoIPTalk

translation-profile outgoing ITSP1

destination-pattern 1111.T

session protocol sipv2

session target sip-server

dtmf-relay sip-notify rtp-nte

For incomming calls. DID to an ext in translation profile:

dial-peer voice 7005 voip

translation-profile incoming VoiptalkDID

incoming called-number xxxxxxx

dtmf-relay rtp-nte

codec g729br8

I discovered that the only way to register with the account number (tel num xxxxxx) is to provide a "dummy" pots dial-peer with digest authentication. Take care of the rest of pots and ephone-dn, use "no-reg". Some providers don't like a lot of numbers registering and failing. You can see with "sh sip-ua register status"

dial-peer voice 7007 pots

destination-pattern xxxxxx

port 2/0

authentication username xxxxxx password yyyyyyy

It works great, even call transfer, or conference. Only TCL AA is not possible.

One important problem I see is that you can only register with one SIP provider, and that is a pitty. Any solutions for that?

Regards.

Miguel.

[justincohen]

This is all great info, i'll start playing myself.

I'm glad i've started this thread, perhaps some cisco folks can join in, i'll send off some queries.

[chmccane1]

Further research and lab testing has led me to believe that BroadVoice is denying SIP calls from CME because CME generates too much traffic by trying to register every configured extension with the provider SIP server. I have 10 extensions configured on CME. I only need for one specific extension to register with the SIP provider but I see continuous registration requests and failures for all 10 extensions. Probably too much overhead for a public SIP gateway provider to deal with so they discourage using Cisco CME by not supporting it and by denying access. I resorted to compiling Asterisk on Red Hat 9 and configuring it as a tandem switch. Asterisk only registers the specific extension it is told to register.

[ugianfra]

Hi,

try the command no-reg in number ephone-dn configuration.

[Steven Juras]

Did you ever find the answer to your question?

We are looking into the same thing as several potential customers want to use Internet VoIP - we've explained the danger with no control of QoS, but the buyers "see" VoIP as a cost savings.

[liamkeegan]

I've tried several SIP providers with the intention of using my CCME system to connect to the PSTN. My end goal was to port my home phone away from Qwest and use just my high-speed internet connection for phone service. My end goal was to connect the CCME to a SIP provider without any analog interfaces (I didn’t want to go from an ATA to a VIC card on the router).

The providers that I've tried have advertised a BYOD plan, but not all of them have been compatible with CCME. I’ve been running 12.3(11)T5 IP Plus on a Cisco 3640 router. I have a few 7960 IP phones that are attached as standard ephones.

Here's what I've found so far:

Broadvoice: Incoming DID works, but I never was able to make outgoing calls. I used the configuration from this thread, and saw the same issues with placing calls.

VoicePulse Connect: I couldn't make incoming or outgoing calls on their BYOD plan. Something about the SIP authentication never worked correctly.

SipPhone.com: Incoming and outgoing work just fine, but they have an (annoying) IVR that tells you your account balance every time you originate an outgoing call. They also don't support LNP.

Viatalk: The Viatalk service works like a champ with CCME. They support only G.711 calls, and have very good latency from Comcast in Denver. I signed up and received a temporary number (before I decided to port), and was able to get all the main features working: call waiting, three way calling, MWI, voicemail, etc. The port of my number from Qwest to Viatalk took 15 days.

As for call quality, I have QoS enabled on both interfaces of my 3640. For the most part, I have heard nothing but good reports, however I have run into a few calls where people say that the sound has been somewhat garbled. I’ve never heard any of these issues on the receiving end. For what I’m paying, I’m willing to put up with it. I signed up for the 500 minute plan, prepaid for 2 years and used a $25 coupon (CQCRNCBQRQ) – it works out to $9/month.

One note:

12.3(11)T6 and some 12.3(14)T trains modify the E.164 registered number to add a $ at the end. This breaks the SIP registration. For instance, if you have an ephone-dn with a number of 2001, you’ll see that the router is trying to register “2001$” instead of “2001” – shown in a ‘sho sip-ua registrar status”. Anyone know how to get around this?

[liamkeegan]

Here’s my configuration that I’m using for Viatalk with CCME

(tailored for Denver 10-digit calling):

voice translation-rule 1

rule 1 /^2.../ /13035551212/

!

voice translation-rule 2

rule 1 /^91(..........$)/ /11/

rule 2 /^9(..........$)/ /11/

rule 3 /^9011($)/ /0111/

rule 4 /^9(.11)/ /1/

!

voice translation-profile SIP

translate calling 1

translate called 2

!

!

!

dial-peer voice 10 voip

translation-profile outgoing SIP

destination-pattern 9[49]11

session protocol sipv2

session target dns:555.303.1.switch.vtnoc.net

dtmf-relay rtp-nte

codec g711ulaw

!

dial-peer voice 11 voip

translation-profile outgoing SIP

destination-pattern 9303.......

session protocol sipv2

session target dns:555.303.1.switch.vtnoc.net

dtmf-relay rtp-nte

codec g711ulaw

!

dial-peer voice 12 voip

translation-profile outgoing SIP

destination-pattern 9720.......

session protocol sipv2

session target dns:555.303.1.switch.vtnoc.net

dtmf-relay rtp-nte

codec g711ulaw

!

dial-peer voice 13 voip

translation-profile outgoing SIP

destination-pattern 91[2-9].........

session protocol sipv2

session target dns:555.303.1.switch.vtnoc.net

dtmf-relay rtp-nte

codec g711ulaw

!

dial-peer voice 14 voip

translation-profile outgoing SIP

destination-pattern 9011T

session protocol sipv2

session target dns:555.303.1.switch.vtnoc.net

dtmf-relay rtp-nte

codec g711ulaw

dial-peer voice 15 voip

translation-profile outgoing SIP

destination-pattern *123

session protocol sipv2

session target dns:555.303.1.switch.vtnoc.net

dtmf-relay rtp-nte

codec g711ulaw

sip-ua

authentication username xxxx password YourPassword

no remote-party-id

registrar dns:555.303.1.switch.vtnoc.net expires 3600

sip-server dns:555.303.1.switch.vtnoc.net

telephony-service

load 7960-7940 P00303020214

max-ephones 10

max-dn 20

ip source-address 10.1.1.1 port 2000

time-zone 6

create cnf-files version-stamp 7960 Jul 27 2005 23:39:25

voicemail *123

mwi sip-server 67.15.74.73 unsolicited

max-conferences 4

call-forward pattern .T

moh flash:music-on-hold.au

transfer-system full-consult

transfer-pattern .T

secondary-dialtone 9

after-hours block pattern 1 9411 7-24

after-hours block pattern 2 91900 7-24

after-hours block pattern 3 91010 7-24

after-hours block pattern 4 9303976 7-24

after-hours block pattern 5 9720976 7-24

after-hours block pattern 6 91809 7-24

after-hours block pattern 7 91758 7-24

after-hours block pattern 8 91664 7-24

after-hours block pattern 9 91284 7-24

after-hours block pattern 10 91876 7-24

!

ephone-dn 1 dual-line

number 2001 no-reg primary

name Phone1

!

ephone-dn 2 dual-line

number 2002 no-reg primary

name Phone2

!

ephone-dn 10 dual-line

number 13035551212

label Incoming Calls

!

ephone 1

mac-address 0012.D9BB.26DE

button 1:1 2:10

!

ephone 2

mac-address 0030.94C3.E550

button 1:2 2:10

!

[Steven Juras]

I see that you're running the SCCP images on the phones...so just to confirm, you can run the Skinny protocol on the handsets (79XXs), you're just translating to SIP when handing off to ViaTalk.

Am I thinking that right?