Release 5.0/BSR222/IP Phones/SIP trunk incompatibility 1

[RLSbutton]

I have a home Nortel BCM 5.0 strictly for a hobby at my house and of course to make and receive phone calls. I personally do not like the clicks and pops of analog trunks when you answer and hang up. I also do not like the delayed CO trunk that doesn't disconnect immediately because of CO features and the delayed caller ID of Analog trunks. A PRI or BRI is ridiculously expensive for an 800 square foot Chicago loft, so I looked into SIP trunking since I have upgraded my system to a 5.0. This is all for fun for me, I LOVE working with Nortel equipment, yes I am a geek, but I am sure I am not alone!

My system is configured with a BCM50 5.0, BSR222 Router, 2 Analog trunks (1 AT&T/Ameritech POTS line, and 1 Vonage VOIP line, and 2 DID SIP trunks from Nexvortex.com, 1 Nortel BES50, and 5 IP phones (3 1140E's and 2 i2004), and 2 digital phones T7316.

Problem, I cannot access the SIP trunking from the IP phones but I CAN access them from the digital T series sets. I've set up a SIP trunk routing code of 9 to reach an outside line and I can make and receive calls on the digital sets via the SIP trunking with no problem. AGAIN, I HAVE NO PROBLEM MAKING SIP TRUNK CALLS ON THE DIGITAL PHONES.

The problem is that I have these expensive beautiful 1140/i2004 IP phones, which are IP based, but cannot access the IP trunks, it seems so backwards. Some people have suggested that I should just use the digital phones and get rid of the IP phones, but I just won't accept that as an answer. If Avaya wants to stay competitive in this market, they should make the BSR222 router be compatible with their own 1100E series telephones and the wave of the future SIP trunking!!!!!

Also, when I call into an 1140 E phone via the SIP trunks, the line connects and immediately disconnects. Apparently my system keeps sending out a "reinvite" message to the VOIP providers server. If I park a call from the digital phones on the SIP trunk and I pick it up on an 1140E phone the line immediately disconnects. Strangely enough, when I pick up the parked call on the i2004 I get one way conversation on the receiving end, but not on the i2004 handset. I can send the same call to the other digital phone and it works fine.

Another really strange thing, VERY STRANGE, is that I have 2 DID's on the SIP trunks. So say I make a call on a T7316 to another T7316 via the SIP trunks, the BCM answers this call as an internal intercom call, even though it's external. Call Pilot answers the call as though it's coming from within the intercom system and not an Outside call as well. I get one way conversation or partial 2 way faded conversation

I've discussed this with someone at Avaya and they told me I just need to wait until Release 6.0 and just use my digital sets until then. This seems like an easy answer, but I assume maybe some of you have figured this out.

My goal is to eliminate the POTS analog trunks totally, and use the Nexvortex.com SIP trunking for my outgoing trunks on my system. (I will still keep the "Ma Bell" analog trunk with a stripped down line for $15 bucks a month for emergency purposes) Is that so hard to ask? I want the crispness of full packet switched technology and Non analog technology. Apparently my request is something that even the manufacturer is having a hard time answering.

THIS IS FRUSTRATING AND ANNOYING, but kind of fun actually. Hopefully someone can help!

Sincerely,

Joe
Nortel Chicago Guy

 

[telcodog]

Check the payload sizes of your codecs. Default is 30ms but not all voip providers accept that, Whatever it is, make sure that both the trunks and the sets are set to the correct value

 

[RLSbutton]

Hey Telcodog,

Thank you for your reply. I will check this out. Can I do this under the Element Manager or do I do it on the actual 1140E phone itself?

Again, the Digital phones work without any issues and have crisp clear voice quality, just like a PRI.

I did run some traces and had a Nortel guy look at the traces and he told me that the BSR222 is accepting the SIP calls from Nexvortex, but that a NAT error is keeping that call from moving beyond the BSR222 Router to the IP phones. We tried Port Triggering and all of that stuff, but to no success.

Apparently RLS6.0 or buying a large Enterprise $1000K router (cant afford that) will fix this. I don't want to wait until then if there is another way to fix this. Plus it's for a hobby, it's hard for me to drop $400 everytime a new RLS comes out!

Any other advice?

Thank you,

Joe

 

[telcodog]

Wow, if you can get an R6 upgrade for $400, you would be wise to take it.

I have 2 sites currently running R6 beta software and this stuff works fantastic. Especially the remote teleworker stuff. The best part about it is, you don't need the built-in routers or those POS BSR222s. Just buy the appropriate keycodes, forward the appropriate ports in your router and put in the public address of your router into the S1 and S2 fields of the set and voila. You won't believe how well this works and is worth the upgrade alone. But hey, that's just my opinion.

In any case, you check the payload size under the Resources tab>Telephony Resources>IP Sets>IP Terminal Global Options. There you will see the payload size for all the different codecs. Just make sure they match what your telco expects. If you don't know, ask first but you could try the different options and see if you can stumble on it but given that you can make calls with digital sets indicates that whatever you have set on the trunk side is the correct one. Just make sure the sets match because if they are set higher, it will override the trunk setting and cause the call to be rejected by the carrier.

That's one part but NAT does wreak havoc on IP phones (especially if you're connecting via vpn). I will assume that your voip carrier has installed a router somewhere in there to get the traffick to you. whatever the case is, make sure that NAT traversal is enabled everywhere as well.

 

[RLSbutton]

Hey Telcodog,

Thank you for your helpful information. I didn't get a chance to check my system last night, but I will try it tonight.

I was able to get RLS5, sorry not 6.0. for about $420 from a distributor in Bellevue, WA, I was assuming RLS6 would be about the same price, maybe a bit more?

Do you know of a way to get the Beta software.....being an enthusiast, I have found many bugs even in 5.0. How does Beta software work? Have you heard when RLS6 may be available for purchase? Would you be willing to walk me through the process once I do get the RLS6.0 for my system? I am very good at setting up on the telco side, but the Routers and Networking side can be a bit challenging.

I can't wait to get rid of the analog lines....

Also, I am going to post another question for you on here about the ring tones vs. BCM and CS1000 and why they don't match, as in the larger/smaller Cisco phones do.

Have a good day!

Joe

 

[telcodog]

Sorry, but beta trials are now closed but in the future, you need to have a distributor to make application on your behalf, then Nortel/Avaya decides if your system is suitable.

Last I heard, R6 should be available around Sept/Oct timeframe but that will depend on what they find during these trials and how long it takes them to fix the bugs.

As for the upgrades, they are the same as any other. Just get the license, start up the upgrade disk (or downloaded file and sit back and wait. There will be a proper procedure document released for it so just follow the instructions.

 

[Frideo]

I've seen similar problems before.

When a digital phone is on a phone call over the SIP trunk, then there's a voice path from the phone to the BCM (over digital) and another voice path from the BCM to the VOIP provider (over IP).

When an IP phone is on a call over the SIP trunk, there may be only one voice path: from the IP phone directly to/from the VOIP provider.
Because of this, if you don't have firewall rules set up correctly, or if you don't have the default gateway correctly set up on the phone, or if NAT is causing a problem, then you may get one-way-audio or no-way-audio problems.

Does the BCM have a public IP address assigned to it for trunking?
Or else is it using a private IP address and connecting through the BSR router? If so, what firewall and/or port-forwarding rules have you set up?
Do the IP phones have private IP addresses? Are they on the same subnet as the BCM? Is the default gateweay correctly set on the phones?

I don't know for sure that I can solve your problem, but I may spot something.

 

[RLSbutton]

Telcodog,

I also meant, would you be able to help me through the whole s1/s2 setup once I do get RLS 6.0 installed?

Any idea what date this product will come out on? A company in Washington named Comtech phones is very reasonably priced in Nortel/Avaya products. FYI.

Thank you,



Joseph Sus Jr. Nortel Enthusiast

 

[RLSbutton]

Frideo,

A few answers to your questions,

The BCM sends out random public IP addresses everytime it calls the SIP trunks, so I assume it connecting via a private IP address through the BSR router. I have not been able to figure out what I should apply towards the firewall port forwarding rules....that is what kills me. The IP phones have Private Ip addresses. I will double check the subnets and default gateway. Should they be exactly the same subnet as the BCM? Please help!

Joe

Joseph Sus Jr. Nortel Enthusiast

 

[telcodog]

RLSButton,

As of today, GA release of 6.0 is slated for Sept 15 2010 barring any show stoppers that come out of the beta trials so it's not much longer.

As for the teleworkers, you will need to buy a teleworker keycode and enough IP client licenses to support the total number of sets you want. You will, of course, need to connect the BCM to your local network and assign it the usual ip info. Then in your network router you will need to forward UDP ports 7000-7002 and TCP ports 30000-30099 (for BCM50) or 30000-30999 (for BCM450) to the BCM's ip address. These ports can be changed if you want in the BCM.

Don't forget to enable NAT Traversal on your routers as well.

Once you have your teleworker licence, you will find a check box under the Resources>Telephony Resources>IP Sets. If you don't have a license, this box will be greyed out. Just check it and you're good to go.

As for the sets, just set everything at the default value except your S1 and S2 addresses. In there, you simply put in the public address of the router that serves the BCM.

 

[RLSbutton]

Hey Telcodog,

Thanks for your help! Just to clarify, I do have 5 IP sets already enabled in my system already and they operate with all functions except when I try to use the SIP trunks on them. I can make and receive analog trunk calls on the IP phones, just not SIP calls.

My BCM is currently connected to the LAN via the 192.128.1.2 address. I am a bit inexperienced with setting the UDP ports on both the BCM and BSR222 Router. Are you saying that I should make these changes once 6.0 is installed or should I try them now with 5.0? Here is what my contact at Avaya responded back to me with:

"I did get a response back from my internal contact, and they have given me the unfortunate news and conclusion that BCM50 + BSR222 SIP ALG + port forwarding/triggering has been very problematic, and is not recommended. His recommendation is to wait until BCM Release 6.0, which includes a few SIP enhancements that should address your issues."

So I guess this means I need to wait for 6.0?



Joseph Sus Jr. Nortel Enthusiast

 

[telcodog]

What I described was all for R6 to support remote IP sets.

The problem you have seems to be with connecting to the SIP trunks through your IP sets. You will have the same issue with R6.

If you can connect to them with your digital sets, tell me what the payload size for each codec is on the trunks side (in the BCM). What codecs are you allowing? The payload size for the codecs for the sets will need to be the same as what is set for trunks.

What is the SIP error code you are getting back from the carrier? In one of your previous notes, you said you are getting a re-invite message. Are you getting anything that says "call unacceptable" before the re-invite?

I do have to ask though......you are allowing access to the SIP trunk pool bloc for those extensions right?

 

[RLSbutton]

Telcodog,

Here is an attached trace that I received from the SIP provider a month or so ago, see if there is anything helpful in here:

call from 13122615594 to 17738002600



we send an INVITE to your network

U 2010/05/10 03:16:52.294516 66.23.129.253:5060 -> 75.3.124.10:10052
INVITE sip:17738002600@75.3.124.10:10052;transport=UDP;transport=udp
SIP/2.0..Record-Route: [sip:17738002600@66.23.129.253:5060;
nat=yes;ftag=a9d5ed0-13c 4-4be77aa4-edda6cc-58c0dfbf;lr=on]..
From: [sip:13122615594@208.94.157.10:5060];tag=a9d5ed0-13c4-4be77a
a4-edda6cc-58c0dfbf..To: [sip:17738002600@66.23.129.253:5060]..
Call-ID: CXC-484-658498b0-a9d5ed0-13c4-4be77aa4-edda6cc-
127d6d19@208.94.157.10..CSeq: 1 INVITE..Via: SIP/2.0/UDP 66.23.129.253:5060;
branch=z9hG4bK1c7b.80d4b786.0..Via: SIP/2.0/UDP 208.94.157.10:5060;
branch=z9hG4bK-3fb92-4be77aa4-edda6cc-1b0 5a68c..
Max-Forwards: 16..Supported: 100rel..Remote-Party-ID:
[sip:13122615594@cxc.dashcs.com:5060];id-type=subscriber;pri vacy=off;screen=no..
Content-Disposition: session;handling=optional..Contact:
[sip:13122615594@208.94.157.10:5060;transport=udp]..
Min-SE: 900..Session-Expires: 1800..Content-Type: application/sdp..
Content-Length: 238....v=0..o=Acme_UAS 0 1 IN IP4 208.94.157.10..
s=SIP Media Capabilities..c=IN IP4 208.94.157.10..t=0 0..m=audio
21098 RTP/AVP 0 18 101..a=rtpmap:0 PCMU/8000..


U 2010/05/10 03:16:54.130022 75.3.124.10:10052 -> 66.23.129.253:5060
SIP/2.0 200 OK..From: [sip:13122615594@208.94.157.10:5060];
tag=a9d5ed0-13c4-4be77a a4-edda6cc-58c0dfbf..To: [sip:17738002600@66.23.129.253:5060];
tag=10902448-c0a80102-1 3c4-4be77ac0-40024148-4be77ac0..
Call-ID: CXC-484-658498b0-a9d5ed0-13c4-4be77aa4-edda6cc-127d6d19@208. 94.157.10..
CSeq: 1 INVITE..Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bK1c7b.80d4b786.0..
Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-3fb92-4be77aa4-edda6cc-1b0 5a68c..
Supported: 100rel,sipvc,x-nortel-sipvc,replaces..User-Agent: Nortel Networks BCM
VoIP Gateway release_45 version_45.25.0.33..x-nt-corr-id: CXC-484-658498b0-a9d5ed0-
13c4-4be77aa4-edda6cc-127d6d19@208. 94.157.10..Contact:
[sip:17738002600@75.3.124.10:10052;transport=udp;user=phone] ..
Record-Route: [sip:17738002600@66.23.129.253:5060;lr;nat=yes;ftag=a9d5ed0-
13c4-4be77aa4-edda6cc-58c0dfbf]..Allow: INVITE,UPDATE,INFO,ACK,OPTIONS,
CANCEL,BYE,NOTIFY,PRACK,REFER ..
Content-Type: application/SDP..Content-Length: 257....v=0..
o=- 1273461442 1273461442 IN IP4 75.3.124.10..s=-..
c=IN IP4 75.3.124.10..t=0 0..a=sqn:0.

your system sends a 200 OK

U 2010/05/10 03:16:54.222290 192.168.1.50:5060 -> 75.3.124.10:10052
ACK sip:17738002600@75.3.124.10:10052;transport=udp;
user=phone SIP/2.0..Record-Route: [sip:17738002600@66.23.129.253:5060;
nat=yes;ftag=a9d5ed0-13c 4-4be77aa4-edda6cc-58c0dfbf;lr=on]..From: [sip:13122615594@208.94.157.10:5060];tag=a9d5ed0-13c4-4be77a a4-edda6cc-58c0dfbf..
To: [sip:17738002600@66.23.129.253:5060];tag=10902448-c0a80102-1
3c4-4be77ac0-40024148-4be77ac0..Call-ID: CXC-484-658498b0-a9d5ed0-
13c4-4be77aa4-edda6cc-127d6d19@208. 94.157.10..CSeq:1 ACK..
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=0..Via: SIP/2.0/UDP 208.94.157.10:5060;
branch=z9hG4bK-3fb9a-4be77aa6-eddae6e-25f 14ca7..Max-Forwards: 16..
Contact: [sip:13122615594@208.94.157.10:5060;transport=udp]..
Content- Length: 0....

we send an ACK that we got the 200 OK


it sends another INVITE but to 13122615594 the caller number and the carrier disconnects the call

U 2010/05/10 03:17:06.565116 75.3.124.10:10052 -> 66.23.129.253:5060
INVITE sip:13122615594@208.94.157.10:5060;transport=udp SIP/2.0..
From: [sip:17738002600@66.23.129.253:5060];tag=10902448-c0a80102-1
3c4-4be77ac0-40024148-4be77ac0..To: [sip:13122615594@208.94.157.10:5060];
tag=a9d5ed0-13c4-4be77a a4-edda6cc-58c0dfbf..Call-ID: CXC-484-658498b0-a9d5ed0-
13c4-4be77aa4-edda6cc-127d6d19@208. 94.157.10..CSeq: 2 INVITE..
Via: SIP/2.0/UDP 75.3.124.10:10052;branch=z9hG4bK-4be77ace-8037b803-40ef967..
Max-Forwards: 70..Supported: 100rel,sipvc,x-nortel-sipvc,replaces..
User-Agent: Nortel Networks BCM VoIP Gateway release_45 version_45.25.0.33..
x-nt-corr-id: CXC-484-658498b0-a9d5ed0-13c4-4be77aa4-edda6cc-127d6d19@208.94.157.10..
Contact: [sip:17738002600@75.3.124.10:10052;transport=udp;user=phone] ..
Route: [sip:17738002600@66.23.129.253:5060;lr;nat=yes;
ftag=a9d5ed0- 13c4-4be77aa4-edda6cc-58c0dfbf]..Allow:
INVITE,UPDATE,INFO,ACK,OPTIONS,CANCEL,BYE,NOTIFY,PRACK,REFER ..
Content-Type: application/SDP..Content-Length: 298....v=0..o=-
1273461442 1273461444 IN IP4 75.3.124.10..s=-..c=IN IP4 75.3.124.1051..t=0 0..
a=sqn:0..a=cdsc:1 im


What's the purpose of reINVITE to the caller 13122615594 , why would it attempt an outbound call to 13122615594 ?

What is the BCM trying to accomplish by sending a reINVITE to the caller number?

Joseph Sus Jr. Nortel Enthusiast

 

[RLSbutton]

Here are some translations of those errors by a Nortel guy I know.

I took a look at your traces, here is my analysis.

- 1140E basic outbound: You're seeing a fast busy because nV is sending back a 400 Bad Request response when your BCM tries to initiate a call to them. I don't know why nV is sending that when you initiate an outbound call from the i2004 set. My hunch would be that the nV endpoint that the call is ultimately sent to is not liking it (it's possible that they route calls to different PSTN gateways for each call placed). You would have to check with nV, as they are sending the error response.

- Parked and transferred calls to 1140E: I'm seeing a 500 Internal Server Error response from nV when you try to pick up on the 1140E set. It's another one of those calls where nV needs to see what they don't like.

- i2004 calls with one-way media: You're seeing this because of the BSR port issue. The BSR is sending audio packets still to the BCM (192.168.1.2) when it should be going to the i2004 set (192.168.1.8) - this would be a limitation and feature gap on the SIP ALG on the router.


You may have to use your digital sets for the time being, until R6 comes out.


Joseph Sus Jr. Nortel Enthusiast

 

[leftbase2009]

Great stuff. I have a home BCM50 as well. I have a copy of R5 and am unsure how hard it is to install. I would also love to get R6 when it comes out. I have a couple i2004 at remote locations over VPN on cisco 871 routers.

 

[RLSbutton]

Hey Leftbase,

Glad to hear you have a BCM50 as well. R5 was very easy to install, just a matter of inserting the CD's and waiting. The system booted up great, but I was dissapointed I still wasn't able to get the SIP trunks to work on my 1140E phones. I do have a couple of i2004 phones, and strangely enough, they give me one way SIP trunk voice paths, while the 1140E's don't have any voice path. Avaya tells me I need to wait for RLS6 and it should solve everything....but seeing that you have some expertise in the router end, maybe you along with the others on here, will be able to help me get this working on RLS5.

Up for a challenge?

Joseph Sus Jr. Nortel Enthusiast

 

[RLSbutton]

Telcodog or Frideo

Did you get anything out of the traces I listed on here? Anything there help you out?

Thank you,

Joe

Joseph Sus Jr. Nortel Enthusiast

 

[telcodog]

Did you check the payload sizes of the codecs for both the trunks ans the sets?

 

[RLSbutton]

Yes, they are both matching at G.711. I tried making both match at G.729 and still nothing. I also uploaded the latest smartupdate patch from July 15th that addresses some of these issues and still nothing. I think I need a person familiar with the networking/router side to take a look via webx and make sure everything is entered properly.

Joseph Sus Jr. Nortel Enthusiast

 

[RLSbutton]

Telcodog,

And as in expert, I mean you....you seem to know the most about this stuff. Are there some things I should copy and paste from element manager into here for everyone to see to try and figure this out?

Joe

Joseph Sus Jr. Nortel Enthusiast