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[pete at privateconnect...]

I have two different gateways setup on my server. One with FlowRoute (which uses SIP REGISTER) and one with bandwidth.com (which does not). I can send and receive from both gateways, but the dialplan processing seems to be confused. Calls from bandwidth do not respect the "Extension" param in the gateway configuration and have the Destination-Number = the inbound caller ID, whereas calls coming from flowroute do respect the param, and have a Destination-Number of "PrivateConnect". Setting "From-user" and "from-domain" on the bandwidth gateway does not seems to affect anything either, though these may be for outbound calls only.

One thing I noticed, calls from the FlowRout gateway have a channel of the form of "sofia/external/<some_number>@<some_IP>" where bandwidth.com channels have the form of "sofia/internal/<some_number>@<some_IP>". The "Internal" profile is what's confusing me. as both calls are coming from external gateways (defined in the "sip_profiles/external" directory.

I also noticed that bandwidth uses "internal" non-routeable IP addresses as part of the SIP URI. I thought might be confusing the system.

My goal is:
0) figure out why the bandwidth gateway is being processed as "internal" (this is more of a security thing)
1) have both gateways enter at the same point in the dialplan (this seems to be the purpose of the "Extension" param)
2) be able to identify which gateway the call came in on. I was hoping to set a param in the gateway configuration that would be passed through onto the channel, but have not found one. Worst case, I could have each gateway enter at a different extension in the dialplan, however, that doesn't seem to be working if the channel comes in the "internal" profile.

Thanks for your help. I've provided INFO dumps from both gateways if they help...
-pete

Here is the INFO from a bandwidth.com call:
EXECUTE sofia/internal/+1480xxxxxxx@192.168.47.68 info()
2009-07-21 21:13:47.479471 [INFO] mod_dptools.c:946 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/+1480xxxxxxx@192.168.47.68]
Unique-ID: [0da43b5a-7676-11de-aa26-1dfbbe056c65]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [ringing]
Channel-Read-Codec-Name: [PCMU]
Channel-Read-Codec-Rate: [8000]
Channel-Write-Codec-Name: [PCMU]
Channel-Write-Codec-Rate: [8000]
Caller-Username: [+1480xxxxxxx]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [+1480xxxxxxx]
Caller-Caller-ID-Number: [+1480xxxxxxx]
Caller-Network-Addr: [216.82.224.202]
Caller-Destination-Number: [+14082561963]
Caller-Unique-ID: [0da43b5a-7676-11de-aa26-1dfbbe056c65]
Caller-Source: [mod_sofia]
Caller-Context: [public]
Caller-Channel-Name: [sofia/internal/+1480xxxxxxx@192.168.47.68]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1248236027477846]
Caller-Channel-Created-Time: [1248236027477846]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_sip_received_ip: [216.82.224.202]
variable_sip_received_port: [5060]
variable_sip_via_protocol: [udp]
variable_sip_authorized: [true]
variable_sip_from_user: [+1480xxxxxxx]
variable_sip_from_uri: [+1480xxxxxxx@192.168.47.68]
variable_sip_from_host: [192.168.47.68]
variable_sip_from_user_stripped: [1480xxxxxxx]
variable_sip_from_tag: [gK0b629aa9]
variable_sofia_profile_name: [internal]
variable_sip_req_params: [transport=udp]
variable_sip_req_user: [+14082561963]
variable_sip_req_port: [5060]
variable_sip_req_uri: [+14082561963@70.99.197.36:5060]
variable_sip_req_host: [70.99.197.36]
variable_sip_to_user: [+14082561963]
variable_sip_to_uri: [+14082561963@192.168.4.90]
variable_sip_to_host: [192.168.4.90]
variable_sip_contact_user: [+1480xxxxxxx]
variable_sip_contact_port: [5060]
variable_sip_contact_uri: [+1480xxxxxxx@192.168.47.68:5060]
variable_sip_contact_host: [192.168.47.68]
variable_channel_name: [sofia/internal/+1480xxxxxxx@192.168.47.68]
variable_sip_call_id: [1376511423_73037211@192.168.47.68]
variable_sip_via_host: [216.82.224.202]
variable_max_forwards: [32]
variable_presence_id: [+1480xxxxxxx@192.168.47.68]
variable_sip_nat_detected: [true]
variable_switch_r_sdp: [v=0
o=Sonus_UAC 23552 274 IN IP4 192.168.47.68
s=SIP Media Capabilities
c=IN IP4 67.231.4.98
t=0 0
m=audio 25748 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20
]
variable_remote_media_ip: [67.231.4.98]
variable_remote_media_port: [25748]
variable_read_codec: [PCMU]
variable_read_rate: [8000]
variable_write_codec: [PCMU]
variable_write_rate: [8000]
variable_endpoint_disposition: [RECEIVED]
variable_outside_call: [true]
variable_current_application: [info]

Here is the flowroute INFO:
EXECUTE sofia/external/+1480xxxxxxx@66.53.188.187 info()
2009-07-21 21:17:49.5616 [INFO] mod_dptools.c:946 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/external/+1480xxxxxxx@66.53.188.187]
Unique-ID: [9d9a2b5c-7676-11de-aa26-1dfbbe056c65]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [ringing]
Channel-Read-Codec-Name: [PCMU]
Channel-Read-Codec-Rate: [8000]
Channel-Write-Codec-Name: [PCMU]
Channel-Write-Codec-Rate: [8000]
Caller-Username: [+14802397349]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [+1480xxxxxxx]
Caller-Caller-ID-Number: [+1480xxxxxxx]
Caller-Network-Addr: [70.167.153.130]
Caller-Destination-Number: [PrivateConnect]
Caller-Unique-ID: [9d9a2b5c-7676-11de-aa26-1dfbbe056c65]
Caller-Source: [mod_sofia]
Caller-Context: [public]
Caller-RDNIS: [14805538912]
Caller-Channel-Name: [sofia/external/+1480xxxxxxx@66.53.188.187]
Caller-Profile-Index: [2]
Caller-Profile-Created-Time: [1248236269004608]
Caller-Channel-Created-Time: [1248236269003604]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_sip_received_ip: [70.167.153.130]
variable_sip_received_port: [5060]
variable_sip_via_protocol: [udp]
variable_sip_from_user: [+1480xxxxxxx]
variable_sip_from_uri: [+1480xxxxxxx@66.53.188.187]
variable_sip_from_host: [66.53.188.187]
variable_sip_from_user_stripped: [1480xxxxxxx]
variable_sip_from_tag: [1661604]
variable_sofia_profile_name: [external]
variable_sip_h_P-Asserted-Identity: [+1480xxxxxxx]
variable_sip_cid_type: [pid]
variable_sip_req_params: [transport=udp]
variable_sip_req_user: [14805538912]
variable_sip_req_port: [5080]
variable_sip_req_uri: [14805538912@70.99.197.36:5080]
variable_sip_req_host: [70.99.197.36]
variable_sip_to_user: [+14805538912]
variable_sip_to_port: [5060]
variable_sip_to_uri: [+14805538912@70.167.153.135:5060]
variable_sip_to_host: [70.167.153.135]
variable_sip_contact_params: [transport=udp]
variable_sip_contact_user: [+1480xxxxxxx]
variable_sip_contact_port: [5060]
variable_sip_contact_uri: [+1480xxxxxxx@66.53.188.187:5060]
variable_sip_contact_host: [66.53.188.187]
variable_channel_name: [sofia/external/+1480xxxxxxx@66.53.188.187]
variable_sip_call_id: [1247507800-6012669@LA4_SIP_01]
variable_sip_via_host: [70.167.153.130]
variable_switch_r_sdp: [v=0
o=Flowroute.com 282181633 1248235361 IN IP4 66.53.189.187
s=Flowroute.com
c=IN IP4 66.53.189.187
t=0 0
a=sendrecv
m=audio 10292 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
]
variable_remote_media_ip: [66.53.189.187]
variable_remote_media_port: [10292]
variable_read_codec: [PCMU]
variable_read_rate: [8000]
variable_write_codec: [PCMU]
variable_write_rate: [8000]
variable_endpoint_disposition: [RECEIVED]
variable_max_forwards: [14]
variable_outside_call: [true]
variable_current_application: [info]

[rupa at rupa.com]

On Tue, Jul 21, 2009 at 11:35 PM, Pete Mueller <pete@privateconnect.com (pete@privateconnect.com)> wrote:

[pete at privateconnect...]

0) Rupa, you are absolutely right, I forgot that. ports was never an issue because previous gateways all REGISTERed. I will have to swap my ports around as bandwidth is not flexible.

1) I thought of this, but I have hundreds of DID, (around 600 at the moment) and maintaining that mapping in the dialplan would be a mess. AFTER I know what gateway the call arrived on, I have a database for each gateway that helps me process from there.

2) Yes, separate profiles would work, but does sound gross. I'm going to swap my ports around and see if that clears things up...

-pete

[brian at freeswitch.org]

On Jul 22, 2009, at 5:11 AM, Pete Mueller wrote:

[rupa at rupa.com]

On Wed, Jul 22, 2009 at 5:11 AM, Pete Mueller <pete@privateconnect.com (pete@privateconnect.com)> wrote:

[brian at freeswitch.org]

On Jul 22, 2009, at 8:04 AM, Rupa Schomaker wrote:

[pete at privateconnect...]

0. Probably, gonna have to go get them on the phone.

1. No, but the information I need to process is stored in separate databases by gateway. There is no single table that has all of the DIDs and which gateway they belong to. I could create/maintain that table, but if I can determine the gateway before beginning my logic, I can avoid that.