IPCOMMS [Outbound from REGISTER ext.] SOLVED
Moderator: Brekeke Support Team
IPCOMMS [Outbound from REGISTER ext.] SOLVED
1. Brekeke Product Name and Version:
Brekeke Sip Server V_2.4.8.6
2. Java version:
jre1.6
3. OS type and the version:
Windows XP \ CentOS 5.8
4. UA (phone), gateway or other hardware/software involved:
Cisco IP Phone 7960
5. Your problem:
I recently signed up for http://2way.ipcomms.net [IPCOMMS FREE DID OFFER] which gives you 2way [inbound/outbound] DID.
I have setup the inbound to work as expected to EXT 0001 to my Cisco IP Phone 7960G. I had to re-direct my DID to my brekeke sip server, then route it to REGISTER EXT 0001.
Now I would like to route over the same phone using EXT 0001 to use the DID service for outbound, but I keep getting 404 not found no matter what I have setup. Can anyone lend me their understanding of what I am doing incorrect?
Any examples,etc [This could be for a similar service like callwithus.com, or callcentric.com]
Thank you,
Kent C.
Brekeke Sip Server V_2.4.8.6
2. Java version:
jre1.6
3. OS type and the version:
Windows XP \ CentOS 5.8
4. UA (phone), gateway or other hardware/software involved:
Cisco IP Phone 7960
5. Your problem:
I recently signed up for http://2way.ipcomms.net [IPCOMMS FREE DID OFFER] which gives you 2way [inbound/outbound] DID.
I have setup the inbound to work as expected to EXT 0001 to my Cisco IP Phone 7960G. I had to re-direct my DID to my brekeke sip server, then route it to REGISTER EXT 0001.
Now I would like to route over the same phone using EXT 0001 to use the DID service for outbound, but I keep getting 404 not found no matter what I have setup. Can anyone lend me their understanding of what I am doing incorrect?
Any examples,etc [This could be for a similar service like callwithus.com, or callcentric.com]
Thank you,
Kent C.
Last edited by KentC on Mon Oct 13, 2014 4:37 am, edited 1 time in total.
Hi!
Here is a new capture where I was able to get something:
My Dial-Plan :
It looks like the invite isn't being received. I used my softphone to make it easier to test.
Q: How can I route my ext 0001 into them via Brekeke , as a "reverse DID" to the number I've been assigned?
Thanks for checking in,
Kent C.
Here is a new capture where I was able to get something:
Code: Select all
INVITE sip:210*******@64.154.41.15 SIP/2.0
Via: SIP/2.0/UDP 123.13*.***:37702;branch=z9hG4bK-d8754z-0021d306e92b3c1d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:0001@123.13*.***:37702>
To: <sip:210*******@64.154.41.15>
From: <sip:0001@123.13*.***>;tag=0970bb52
Call-ID: NjQ4ZWU0MWVhZjkwODk3MWY2OTQxMzc0MmI1OGNiMzQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 495
v=0
o=- 8 2 IN IP4 123.13*.***
s=CounterPath Bria Professional
c=IN IP4 123.13*.***
t=0 0
m=audio 21394 RTP/AVP 107 119 100 106 0 98 8 18 101
a=alt:1 1 : SEWMYfD3 UF5+4SKg 123.13*.*** 21394
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:1BA0BC5E98534CFA84638039FD53B443
INVITE sip:210*******@64.154.41.15 SIP/2.0
Via: SIP/2.0/UDP 123.13*.***:37702;branch=z9hG4bK-d8754z-0021d306e92b3c1d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:0001@123.13*.***:37702>
To: <sip:210*******@64.154.41.15>
From: <sip:0001@123.13*.***>;tag=0970bb52
Call-ID: NjQ4ZWU0MWVhZjkwODk3MWY2OTQxMzc0MmI1OGNiMzQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 495
v=0
o=- 8 2 IN IP4 123.13*.***
s=CounterPath Bria Professional
c=IN IP4 123.13*.***
t=0 0
m=audio 21394 RTP/AVP 107 119 100 106 0 98 8 18 101
a=alt:1 1 : SEWMYfD3 UF5+4SKg 123.13*.*** 21394
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:1BA0BC5E98534CFA84638039FD53B443
INVITE sip:210*******@64.154.41.15 SIP/2.0
Via: SIP/2.0/UDP 123.13*.***:37702;branch=z9hG4bK-d8754z-0021d306e92b3c1d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:0001@123.13*.***:37702>
To: <sip:210*******@64.154.41.15>
From: <sip:0001@123.13*.***>;tag=0970bb52
Call-ID: NjQ4ZWU0MWVhZjkwODk3MWY2OTQxMzc0MmI1OGNiMzQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 495
v=0
o=- 8 2 IN IP4 123.13*.***
s=CounterPath Bria Professional
c=IN IP4 123.13*.***
t=0 0
m=audio 21394 RTP/AVP 107 119 100 106 0 98 8 18 101
a=alt:1 1 : SEWMYfD3 UF5+4SKg 123.13*.*** 21394
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:1BA0BC5E98534CFA84638039FD53B443
INVITE sip:210*******@64.154.41.15 SIP/2.0
Via: SIP/2.0/UDP 123.13*.***:37702;branch=z9hG4bK-d8754z-0021d306e92b3c1d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:0001@123.13*.***:37702>
To: <sip:210*******@64.154.41.15>
From: <sip:0001@123.13*.***>;tag=0970bb52
Call-ID: NjQ4ZWU0MWVhZjkwODk3MWY2OTQxMzc0MmI1OGNiMzQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 495
v=0
o=- 8 2 IN IP4 123.13*.***
s=CounterPath Bria Professional
c=IN IP4 123.13*.***
t=0 0
m=audio 21394 RTP/AVP 107 119 100 106 0 98 8 18 101
a=alt:1 1 : SEWMYfD3 UF5+4SKg 123.13*.*** 21394
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:1BA0BC5E98534CFA84638039FD53B443
INVITE sip:210*******@64.154.41.15 SIP/2.0
Via: SIP/2.0/UDP 123.13*.***:37702;branch=z9hG4bK-d8754z-0021d306e92b3c1d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:0001@123.13*.***:37702>
To: <sip:210*******@64.154.41.15>
From: <sip:0001@123.13*.***>;tag=0970bb52
Call-ID: NjQ4ZWU0MWVhZjkwODk3MWY2OTQxMzc0MmI1OGNiMzQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 495
v=0
o=- 8 2 IN IP4 123.13*.***
s=CounterPath Bria Professional
c=IN IP4 123.13*.***
t=0 0
m=audio 21394 RTP/AVP 107 119 100 106 0 98 8 18 101
a=alt:1 1 : SEWMYfD3 UF5+4SKg 123.13*.*** 21394
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:1BA0BC5E98534CFA84638039FD53B443
INVITE sip:210*******@64.154.41.15 SIP/2.0
Via: SIP/2.0/UDP 123.13*.***:37702;branch=z9hG4bK-d8754z-0021d306e92b3c1d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:0001@123.13*.***:37702>
To: <sip:210*******@64.154.41.15>
From: <sip:0001@123.13*.***>;tag=0970bb52
Call-ID: NjQ4ZWU0MWVhZjkwODk3MWY2OTQxMzc0MmI1OGNiMzQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 495
v=0
o=- 8 2 IN IP4 123.13*.***
s=CounterPath Bria Professional
c=IN IP4 123.13*.***
t=0 0
m=audio 21394 RTP/AVP 107 119 100 106 0 98 8 18 101
a=alt:1 1 : SEWMYfD3 UF5+4SKg 123.13*.*** 21394
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:1BA0BC5E98534CFA84638039FD53B443
INVITE sip:210*******@64.154.41.15 SIP/2.0
Via: SIP/2.0/UDP 123.13*.***:37702;branch=z9hG4bK-d8754z-0021d306e92b3c1d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:0001@123.13*.***:37702>
To: <sip:210*******@64.154.41.15>
From: <sip:0001@123.13*.***>;tag=0970bb52
Call-ID: NjQ4ZWU0MWVhZjkwODk3MWY2OTQxMzc0MmI1OGNiMzQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 495
v=0
o=- 8 2 IN IP4 123.13*.***
s=CounterPath Bria Professional
c=IN IP4 123.13*.***
t=0 0
m=audio 21394 RTP/AVP 107 119 100 106 0 98 8 18 101
a=alt:1 1 : SEWMYfD3 UF5+4SKg 123.13*.*** 21394
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:1BA0BC5E98534CFA84638039FD53B443
Code: Select all
KENTC IPCOMMS DID2
Matching Patterns
$request=^INVITE
$outbound=true
From=sip:(0001)@
To=sip:210*******@
Deploy Patterns
$target=64.154.41.15
From=sip:%1@123.13*.***
To=sip:210*******@64.154.41.15
Q: How can I route my ext 0001 into them via Brekeke , as a "reverse DID" to the number I've been assigned?
Thanks for checking in,
Kent C.
Hello.
I haven't be able to route any DID that has 2way so far with any type of phone but have been able to route into Brekeke.
Normal Call Flow as I've been setting it up:
Service Provider > Brekeke SBC > Brekeke LAN > Soft/Hard IP Phone
REAL DID # > DID To EXT > EXT To Phone > THE CALL
The reverse how I would like to deliver it back outbound:
Soft/Hard IP Phone > Brekeke LAN > Service Provider [SBC if needed]
CALL PLACED > EXT To DID ROUTE > Cloud > Some Phone
I haven't be able to route any DID that has 2way so far with any type of phone but have been able to route into Brekeke.
Normal Call Flow as I've been setting it up:
Service Provider > Brekeke SBC > Brekeke LAN > Soft/Hard IP Phone
REAL DID # > DID To EXT > EXT To Phone > THE CALL
The reverse how I would like to deliver it back outbound:
Soft/Hard IP Phone > Brekeke LAN > Service Provider [SBC if needed]
CALL PLACED > EXT To DID ROUTE > Cloud > Some Phone
No. Once I register to Brekeke, I can call other EXT's but nothing from outside the local Brekeke Sip Server.Tata wrote:Do you know how the EXT-0001 (Cisco IP Phone 7960G) redirects a call?
Is it a 30x message?
I haven't gotten that far, I've completed routing directly Brekeke mostly. I've only seen 404 on my soft/hard phones.
Can you make a call from the registered EXT-0001 (It seems Bria) to the DID service without the redirection?
I'll keep trying stuff to see what i can tinker/find I maybe missing.
Thanks,
Kent C.
Guys,
Here is the dial-plan I used:
***I changed the ext to 0003 on Bria 2.4 for testing***
I see the call reach ipcomm's IP, I'm not sure if that rule changes my ext to actually be my DID, but it would appear I have to register to IPCOMMS IP.
Someone asked if I can REGISTER to their gateway, and after trying all this for hours.. I decided to just try to build the routing from within my Counterpath Bria Pro Softphone:
1st Dial Plan REGISTER to 2way.ipcomms.net
2nd Dial Plan REGISTER to Brekeke Sip Server
With the following I am able to still get routing INTO the brekeke sip server for my DID still pointed to my EXT. If that call fails, the 2way.ipcomms.net dial plan on softphone then trys.
For the outbound, I have to use the 2way.ipcomms.net minus the Brekeke Sip Server.
Anyone got a general tutorial or any advice when trying to route an EXT# to a DID? You can be generic in context with dial-plan building.
I'm not sure if this will work with my Cisco IP 7960G but with the Bria phone, it does.
I've inquired about routing with ipcomms support to see if such a thing can be routed from Brekeke Sip Server to Ipcomms gateway.
I'll post back my results and reply with what they recommend back.
As you can see the call auto-connects for some reason. Doesn't ring first so I think it's a register issue since I see it connecting to server:
Again, please share your thoughts. but for now, I have configured at least my Bria 2.4 Pro to now do both DID inbound via Brekeke and Outbound directly to IPCOMMS - WORKING!
Thanks,
Kent C.
Here is the dial-plan I used:
Code: Select all
Matching Patterns
$request=^INVITE
$registered=false
From=sip:0003@
To=sip:(.+)@
Deploy Patterns
$target=64.154.41.158
$rtp=true
To=sip:%1@64.154.41.158
I see the call reach ipcomm's IP, I'm not sure if that rule changes my ext to actually be my DID, but it would appear I have to register to IPCOMMS IP.
Someone asked if I can REGISTER to their gateway, and after trying all this for hours.. I decided to just try to build the routing from within my Counterpath Bria Pro Softphone:
1st Dial Plan REGISTER to 2way.ipcomms.net
2nd Dial Plan REGISTER to Brekeke Sip Server
With the following I am able to still get routing INTO the brekeke sip server for my DID still pointed to my EXT. If that call fails, the 2way.ipcomms.net dial plan on softphone then trys.
For the outbound, I have to use the 2way.ipcomms.net minus the Brekeke Sip Server.
Anyone got a general tutorial or any advice when trying to route an EXT# to a DID? You can be generic in context with dial-plan building.
I'm not sure if this will work with my Cisco IP 7960G but with the Bria phone, it does.
I've inquired about routing with ipcomms support to see if such a thing can be routed from Brekeke Sip Server to Ipcomms gateway.
I'll post back my results and reply with what they recommend back.
Code: Select all
CDR:
sip:0003@MY.IP.ADDR sip:1210*******@64.154.41.158 00:00:12 Thu Oct 02 15:13:06 CDT 2014 Thu Oct 02 15:13:06 CDT 2014 Thu Oct 02 15:13:18 CDT 2014 Success -1
Code: Select all
Active Sessions:
776 sip:0003@MYIPADDR
(***.**.**.**:42772) sip:1210*******@64.154.41.158
(64.154.41.158) 2014-10-02 15:57:38.875 Talking
Thanks,
Kent C.
Let me ask you guys,
If I register the brekeke sip server to IPCOMMS, like so:
I would like to know if this can be used to setup a dial plan that can be routed using Brekeke as the device?
589####### = my user @ ipcomms.net
(Right now for my Counterpath Bria 2.4 to work, I'm back to using 2 Dial Plan Rules inside of that app.
Each rule in Bria only takes one domain/proxy setting per account. Reason for making 2 rules.)
Thanks,
Kent C.
If I register the brekeke sip server to IPCOMMS, like so:
Code: Select all
User
589#######
Contact URI
sip:589#######@2way.ipcomms.net
Detail
Expires : 3600000
Priority : 1000
User Agent : Brekeke Admintool rev.286.3
Requester : IPADDR.**.**.***:3628
Time Update : Fri Oct 03 11:01:11 CDT 2014
589####### = my user @ ipcomms.net
(Right now for my Counterpath Bria 2.4 to work, I'm back to using 2 Dial Plan Rules inside of that app.
Each rule in Bria only takes one domain/proxy setting per account. Reason for making 2 rules.)
Thanks,
Kent C.
Added that to DP and no effect so far.Tata wrote:Do you see any 401 or 407 response from the DID provider?
If so, let you disable auth at Brekeke SIP Server because a call is asked auth two times.
For example,add "$auth=false" in the DeployPatterns.
I did see a 401 in my capture. I don't get a 180 ringing trying to route away from Brekeke > IPCOMMS.NET. Bria connects instantly and sends rtp packets without an accepted connection made first then finally gives back a BYE from IPCOMMS.
They are asking if I can REGISTER Brekeke and then do routing, please advise.
Thanks,
Kent C.
> I would like to know if this can be used to setup a dial plan that can
> be routed using Brekeke as the device?
Answer: Depending on settings.
> Contact URI
> sip:589#######@2way.ipcomms.net
Do you want to forward call to sip:589#######@2way.ipcomms.net if the SIP Server receives a call to 589#######?
I don't recommend you use a manual registration because you can do the same thing with DialPlan (and also there are some other reasons..)
> be routed using Brekeke as the device?
Answer: Depending on settings.
> Contact URI
> sip:589#######@2way.ipcomms.net
Do you want to forward call to sip:589#######@2way.ipcomms.net if the SIP Server receives a call to 589#######?
I don't recommend you use a manual registration because you can do the same thing with DialPlan (and also there are some other reasons..)
james wrote:> I would like to know if this can be used to setup a dial plan that can
> be routed using Brekeke as the device?
Answer: Depending on settings.
Yes!! Please share with Brekeke Sip Server registered to make outbound calls with Brekeke basically the registered softphone. That would work along with the inbound DID rule I have working now. I had to forward to my Brekeke and get it routed
> Contact URI
> sip:589#######@2way.ipcomms.net
Do you want to forward call to sip:589#######@2way.ipcomms.net if the SIP Server receives a call to 589#######?
Yes. Or any number.
I don't recommend you use a manual registration because you can do the same thing with DialPlan (and also there are some other reasons..)
I want to use a dial plan but haven't had much luck.
Yes. I do want to forward the call in exactly that method. So whatever is called is forwarded to the 2way.ipcomms.net contact:uri provided from their service while Brekeke is registered and uses Brekeke REGISTER to allow the call
CAP:
8 73.367392 64.154.41.158 **.***.***.** SIP 612 Status: 401 Unauthorized
9 73.368328 **.***.***.** 64.154.41.158 SIP 795 Request: SUBSCRIBE sip:589#######@**.***.***.**
10 73.399246 64.154.41.158 **.***.***.** SIP 546 Status: 404 Not found (no mailbox)
That's the call flow for the last call I made. It usually connects instantly and there is RTP being sent already without an ACK.
Thanks for your help! Looking forward to the next attempt.
Kent C.
Hello again,
I studied the wiki more and want to try this:
http://wiki.brekeke.com/wiki/upper-or-thru-registration
Going off the model's on that link above, I want to know how to setup the following:
Upper Registration
Clients d, e, and f register themselves to Brekeke SIP server B. Server B registers those clients to SIP server A. Server A will handle the REGISTER requests as if they came directly from the clients, rather than from server B.Using this Upper Registration function, servers A and B can work together without changing server A's configuration.
Can anyone give advice on this kind of setup? Thank you!
Kent C.
I studied the wiki more and want to try this:
http://wiki.brekeke.com/wiki/upper-or-thru-registration
Going off the model's on that link above, I want to know how to setup the following:
Upper Registration
Clients d, e, and f register themselves to Brekeke SIP server B. Server B registers those clients to SIP server A. Server A will handle the REGISTER requests as if they came directly from the clients, rather than from server B.Using this Upper Registration function, servers A and B can work together without changing server A's configuration.
Can anyone give advice on this kind of setup? Thank you!
Kent C.
KentC wrote:Hello again,
I studied the wiki more and want to try this:
http://wiki.brekeke.com/wiki/upper-or-thru-registration
Going off the model's on that link above, I want to know how to setup the following:
Upper Registration
Clients d, e, and f register themselves to Brekeke SIP server B. Server B registers those clients to SIP server A. Server A will handle the REGISTER requests as if they came directly from the clients, rather than from server B.Using this Upper Registration function, servers A and B can work together without changing server A's configuration.
Can anyone give advice on this kind of setup? Thank you!
Kent C.
EDIT : I FIGURED OUT A WAY GUYS TO MAKE IT WORK OUTBOUND!!
Dial Plan Rule:
Code: Select all
Matching Patterns
$request=^INVITE
$registered=false
To=sip:(.+)@
Code: Select all
Deploy Patterns
To=sip:%1@64.154.41.158
Brekeke Sip Server Settings:
Configuration > System
Network > Interface address 1 > Company Router IP HERE [The one that showed up IPCOMMS as Public IP in call trace we did together troubleshooting]
UPnP > Enable [Tick]
Default router IP Address = Company Router IP used on Interface address 1
Configuration > SIP
Upper Registration > on [Tick]
Register Server = 64.154.41.158 [USE IPCOMMS IP HERE]
Thru Registration > off [Tick]
Registered Clients
Code: Select all
User
589*******
Contact URI
sip:589*******@2way.ipcomms.net
Detail
Expires : 3600000
Priority : 1000
User Agent : Brekeke Admintool rev.286.3
Requester : MY.SERVER.IP.ADDR:3628
Time Update : Fri Oct 03 11:01:11 CDT 2014
This is now routing out from my LAN Brekeke Sip Server to IPCOMMS via UPPER REGISTRATION
Problem solved! This may be used for anyone wishing to try IPCOMMS out as a guide.
Thanks everyone for your help!
Kent C.
Hi James,james wrote:Glad to know it. I didn't imagine such a configuration.
Does the CISCO phone send REGISTER to the Brekeke SIP Server?
Are there any other SIP clients which send REGISTER to Brekeke SIP Server?
This is because the Upper-Rgister feature may affect them. (but I have an idea to ignore possible issues.)
Yes. The Cisco 7960G does register with Brekeke Sip Server but for inbound only on different routing rules from SBC.
Big THANK YOU to IPCOMMS TECH SUPPORT as well! They did free troubleshooting and I've sent a follow-up email to give them the conclusion that worked. They give 50.00 USD free outbound US. Hey it was worth the fight. Them doing call traces and wireshark captures with me also helped me figure out all this.
Thanks again and please share any other routing ideas you may have so I can try.
Kent C.