Monday, July 16, 2007

What's my name?

This is going to be a different kind of post. This post might actually be useful for people trying to solve this problem. Just the facts, ma'am.

One of the things that has repeatedly come up in my line of work is CallerID name delivery in PRI (Primary Rate Interface) ISDN (Integrated Services Digital Network) configurations. I learned more about CallerID name today than I ever wanted to know. Just kidding - I love getting into stuff like this!

PRI is great because call setup is fast and CallerID information is available instantly. Or is it? I always knew that Caller ID name is not carried over the PSTN (usually - in some countries it is). The number does (obviously), but the name is usually looked up in CNAM by the terminating switch, not the originating switch. What I didn't know is that sometimes this isn't done when the initial Q.931 Setup message comes down the PRI to signal a new call.

Sometimes this CNAM lookup takes a little while (fractions of a second) and the name is sent later in a separate Q.931 Facility message. This is true. Cisco says so (PDF). A Cisco ISDN-SIP gateway can be configured to do this one of two ways:

1) Wait until you receive the Q.931 Facility message with name and shove it into the SIP INVITE using either PAI (P-Asserted-Identity) or RPID (Remote-Party-ID). Send the INVITE to the SIP proxy (or wherever).

2) Send the INVITE ASAP, and then send a SIP INFO packet when the name shows up in the Q.931 Facility message.

The default is #2, which is screwy. Very cool, but still screwy. It is much harder to design a SIP platform that can accept the initial INVITE, begin to process the call, and then append the PAI or RPID information received in the later INFO.

Thanks to Cisco I now understand more about Q.931 and ISDN. Now I need to get this "thing" to work.

My test setup:

PRI -> Asterisk -> PRI -> AS5350XM -> SIP -> OpenSER -> SIP -> Device

I need to get Caller ID with name delivery through this whole mess, from the first PRI to the last SIP device.

The LEC provided the PRI coming into the Asterisk machine. I provided everything else. I saw several roadblocks:

1) Get the CID Name from the LEC (via PRI)
2) Pass it through Asterisk
3) Get it to the 5350 (via PRI)
4) Get it to OpenSER (via SIP)

Knowing what I now know about Caller ID with name in ISDN I knew just what to do for Asterisk. In zapata.conf, my incoming context is lec-in. Here it is (from extensions.conf):

[lec-in]

exten => NXXNXXXXXX,1,Wait(1)
exten => NXXNXXXXXX,n,DoSomethingElse

Yep, that's right. All you need is to Wait a little to get that second Facility IE. Asterisk doesn't support getting the Facility IE later and it certainly doesn't support sending a subsequent SIP INFO. That's a good thing because as I said the "other" way (SIP INFO) just seems goofy to me.

Now I needed to get the CallerID name to the 5350. It didn't seem to work. I start looking at "pri debug span 3" output to see the Q.931 goodness coming from Asterisk. I fired up "debug isdn q931" on the 5350. No dice. It looked like this bug in libpri was killing me:

http://bugs.digium.com/view.php?id=9651

This was committed to libpri SVN about a month ago. I update libpri from SVN, recompile Asterisk, and install the new chan_zap.so. I give it another shot. It looks like the 5350 is now getting the name over Q. 931. Using ngrep I look at the SIP INVITE coming into OpenSER from the 5350. I have an RPID header, but it looks strange. The name field in the Remote-Party-ID header is "pending". What the heck is that about? "pending" was not what I was seeing in Asterisk!

I opened up ngrep a bit to let my see any SIP INFO messages that might be coming later. Sure enough shortly after the SIP INVITE comes a SIP INFO message with my Caller ID name. Going back to my two configuration choices on the 5350 I knew I preferred option #1 (send everything in one SIP INVITE), even if it meant there was a little delay before the caller got audio. How could I configure the 5350 to wait a little and put it all in one SIP INVITE before the Cisco fired it off to OpenSER?

I dug around on cisco.com for a bit. Nothing - at least nothing obvious. You have to love Cisco configuration and Cisco docs. I decided to look around the internet and see if anyone else had this problem.

I looked on Google and found this:

http://puck.nether.net/pipermail/cisco-voip/2005-June/005485.html

I wondered if Mr. Adam Rothschild ever found the solution to his (my) problem. I open up another tab and write him an e-mail. Three minutes later (literally) he sends me this configuration snippet:

---Begin IOS Configuration---
interface Serial3/0:23
no ip address
load-interval 30
isdn switch-type primary-ni
isdn incoming-voice modem
isdn supp-service name calling
isdn negotiate-bchan
no isdn outgoing display-ie
no cdp enable
exit
gateway
timer receive-rtp 1200
sip-ua
disable-early-media 180
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers buffer-invite 500

---End IOS Configuration---

Let's get away from this technical mumbo-jumbo and talk about people for a minute...

Mr. Adam Rothschild got an e-mail from a random stranger across the internet referencing an obscure technical problem that he had over two years ago. In less than three minutes he dug up the solution and wrote me back. I have a SmartNet support contract on this 5350 but I doubt the techs at Cisco could have helped me any better or faster than a nice guy (Adam) helping a stranger (me).

Wipe away your tears, you sentimental fool. We're getting back to configuration. This blog is hardcore. Couldn't you tell?

I applied Adam's config to my AS5350XM running IOS 12.4(15)T. Here is the SIP INVITE from the 5350 to OpenSER:

U 192.168.0.1:61306 -> 192.168.0.10:5060
INVITE sip:9418675309@192.168.0.10:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.1:5060;x-ds0num="ISDN 3/1:D 3/1:DS1
1:DS0";branch=z9hG4bK901AB9.
Remote-Party-ID: "STAR2STAR COMM"
;party=calling;screen=no;privacy=off.
From: "STAR2STAR COMM" ;tag=971D8C-1203.
To: .
Date: Mon, 16 Jul 2007 21:57:24 GMT.
Call-ID: 5E99C7DC-331E11DC-8126E6C7-399CBB13@192.168.0.1.
Supported: 100rel,timer,resource-priority,replaces.
Min-SE: 1800.
Cisco-Guid: 1586976572-857608668-2150694933-1673067056.
User-Agent: Cisco-SIPGateway/IOS-12.x.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER.
CSeq: 101 INVITE.
Max-Forwards: 70.
Timestamp: 1184623044.
Contact: .
Expires: 300.
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Disposition: session;handling=required.
Content-Length: 288.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 7275 8957 IN IP4 192.168.0.1.
s=SIP Call.
c=IN IP4 192.168.0.1.
t=0 0.
m=audio 20746 RTP/AVP 18 0 101.
c=IN IP4 192.168.0.1.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16

Yeah yeah! Look at that Caller ID name in that Remote-Party-ID header! I feel like that's the best looking SIP INVITE I have ever seen. How does one SIP INVITE look better than any other? If you don't know the answer to that question, you haven't been following along.

I wrote Adam back to let him know how it turned out. He wrote me back again, happy to hear that it worked for me. Wow, just wow.

So many things shine through in this post. In one evening I found (and patched) a bug in libpri. I learned more about Q.931 and Caller ID. I found a guy to help me put it all together. The open source development model worked. The promise of easy access to information via the internet skooled me in ISDN. Social networking proved to be very effective, even while using pre-web 2.0 technology (e-mail). Google worked (a lot).

Now I get to put it all together in this blog post to give back a little. Hopefully the next guy (or girl) trying to get some mixed up mess of SIP and ISDN devices to work together with Caller ID Name delivery will get out of the office just a little bit earlier.

6 comments:

Mark Turpin said...

wow! thank you for this! i have h.323 gateways with pri service I want to convert to sip but was concerned about cnam. now i know its possible, so thank you!!

zicore said...

Thanks this solved a caller id problem I was having from cisco -> asterisk.

Larry Leonard said...

Thanks for the solution and the well-written explanation of it. I had been scratching my head for a long time over this nuisance and your fix works perfectly.

David Walker said...

Thanks for posting the info for a "How to" on a Channelized T1 Pri. I had been working on this for some time. I am using a cisco 1751-v 128/32 with pvdm-256k-20 dsp. I downloaded 12.4.15.T10 and added the commands and it work without doing anythings else. I looked in cisco community, and many voip lists with no success. GREAT JOB in sharing information!!!!!

Travis Hegner said...

Great Post! This solved a tough problem for me in 10 minutes. Thanks for giving back!

Tim said...

The important bits are the following:

interface serial 3/0:23
isdn supp-service name calling

sip-ua
timers buffer-invite 500

everything else isn't directly related to delivering Caller ID.