# WellGate FXO and Asterisk **URL:** \ **Category:** Asterisk Support\ **Created:** [July 7, 2006, 12:57pm UTC](https://community.asterisk.org/t/wellgate-fxo-and-asterisk/7599 "2006-07-07T12:57:10Z")\ **Posts on this page:** 5\ **Page:** 1
**Author:** ![oxg915](https://avatars.discourse-cdn.com/v4/letter/o/dbc845/32.png) [@oxg915](https://community.asterisk.org/u/oxg915)\ **Post date:** [July 7, 2006, 12:57pm UTC](https://community.asterisk.org/t/wellgate-fxo-and-asterisk/7599/1 "2006-07-07T12:57:10Z")
Hi all, I have a asterisk PBX that we use on our network. It is for around 30 people to make VoIP calls via their SIP phones. I have the PBX setup with our VoIP provider and this part of the setup is working well. I have now purchased a WellGate 6 port FXO unit, so that I can: 1. Terminate local calls with the FXO - because it is much more expensive to make calls to local numbers using our VoIP provider. 2. I want incoming calls to be answered by the FXO, a greeting played to the caller who will then dial the 7 digit extension number and get connected to the SIP phone of the person they are calling. I haven’t tried item 1 yet but have been working on item 2. I have contacted WellGate support who sent me a PDF with instructions. I have followed these instructions line by line but I’m not having any luck. I can see the FXO is registering with Asterisk. When I make a call the FXO, it unit answers and within 5 seconds hangs up. When monitoring Asterisk, I cannot see the call being forwarded to the PBX. I have searched on Google for examples of others working with the Wellgate 6 port FXO. I have seen some people saying that they use the FXO in this kind of setup, but I haven’t come across any postings/guides about setting it up. I would be grateful if there is anyone here who has setup the Wellgate FXO with Asterisk and who could help me find what I am doing wrong. Many thanks, David. ---
**Author:** ![IronHelix](https://avatars.discourse-cdn.com/v4/letter/i/da6949/32.png) [@IronHelix](https://community.asterisk.org/u/IronHelix)\ **Post date:** [July 7, 2006, 2:16pm UTC](https://community.asterisk.org/t/wellgate-fxo-and-asterisk/7599/2 "2006-07-07T14:16:14Z")
First, im assuming its a sip gateway. Second, I think the problem is on the \* side. I’m assuming you have the gateway registered in sip.conf and that works okay… now look at the contexts. I think the problem is that the gateway ‘dials’ \* with a number which asterisk isnt finding, so it dumps the call and the GW hangs up. First, I would suggest if it is possible, set the gateway to not answer the ringing line until the resulting SIP call is answered. Second, on asterisk do sip debug peer peername, with peername being the gateway. Now watch what happens when a call comes in, and post that here if it doesnt provide any immediate clues. I am going to gander a guess that the gateway is dialing (something)@asterisk, and that the (something) doesn’t exist in the context that the gateway is in… either that or you have an auth problem. Post the sip debug here and I’ll have a look at it… ---
**Author:** ![oxg915](https://avatars.discourse-cdn.com/v4/letter/o/dbc845/32.png) [@oxg915](https://community.asterisk.org/u/oxg915)\ **Post date:** [August 28, 2006, 9:14am UTC](https://community.asterisk.org/t/wellgate-fxo-and-asterisk/7599/3 "2006-08-28T09:14:51Z")
Hi, I am still having a problem with the wellgate unit. The extensions I have created are being registered: Name/username Host Dyn Nat ACL Mask Port Status 1001/1001 80.78.16.211 D N 255.255.255.255 5060 OK (141 ms) I have setup extension 600 as an echo-test, and it works on a normal SIP phone. However, I cannot see data passing from the FXO to asterisk when I make the call and dial extension 600. The FXO just hangs up. I set debug in asterisk, and even without making a call I am getting the following all the time. 80.78.16.214 is the asterisk pbx, 80.78.16.211 is the wellgate FXO. to 80.78.16.211:5060 Scheduling destruction of call ‘504e10d3-13c4-44f2b02e-5e7d88-4695’ in 15000 ms rmt214-16\*CLI\> Sip read: REGISTER sip:80.78.16.214 SIP/2.0 From: sip:1004@80.78.16.214;tag=504e10d3-13c4-44f2b02e-5e7ddd-12e4 To: sip:1004@80.78.16.214 Call-ID: 504e10d3-13c4-44f2b02e-5e7ddd-5076 CSeq: 2 REGISTER Via: SIP/2.0/UDP 80.78.16.211:5060;branch=z9hG4bK-44f2b02f-5e81bb-4703 Max-Forwards: 70 Supported: replaces User-Agent: FXO\_GW Contact: sip:1004@80.78.16.211:5060;q=0.5 Expires: 60 Authorization: Digest username=“1004”, realm=“asterisk”, nonce=“387a424f”, uri=“sip:80.78.16.214”, response="9f2304b2cd2ba4b81cd07d33e07fe343" Content-Length: 0 13 headers, 0 lines Using latest request as basis request Sending to 80.78.16.211 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 80.78.16.211:5060;branch=z9hG4bK-44f2b02f-5e81bb-4703 From: sip:1004@80.78.16.214;tag=504e10d3-13c4-44f2b02e-5e7ddd-12e4 To: sip:1004@80.78.16.214;tag=as32088fb4 Call-ID: 504e10d3-13c4-44f2b02e-5e7ddd-5076 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 60 Contact: sip:1004@80.78.16.214;expires=60 Content-Length: 0 to 80.78.16.211:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 80.78.16.211:5060;branch=z9hG4bK-44f2b02f-5e81bb-4703 From: sip:1004@80.78.16.214;tag=504e10d3-13c4-44f2b02e-5e7ddd-12e4 To: sip:1004@80.78.16.214;tag=as32088fb4 Call-ID: 504e10d3-13c4-44f2b02e-5e7ddd-5076 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 60 Contact: sip:1004@80.78.16.211:5060;expires=60 Date: Mon, 28 Aug 2006 08:59:31 GMT Content-Length: 0 to 80.78.16.211:5060 Scheduling destruction of call ‘504e10d3-13c4-44f2b02e-5e7ddd-5076’ in 15000 ms rmt214-16\*CLI\> Sip read: REGISTER sip:80.78.16.214 SIP/2.0 From: sip:1005@80.78.16.214;tag=504e10d3-13c4-44f2b02e-5e7e32-3749 To: sip:1005@80.78.16.214 Call-ID: 504e10d3-13c4-44f2b02e-5e7e2d-7313 CSeq: 2 REGISTER Via: SIP/2.0/UDP 80.78.16.211:5060;branch=z9hG4bK-44f2b02f-5e8279-4b9 Max-Forwards: 70 Supported: replaces User-Agent: FXO\_GW Contact: sip:1005@80.78.16.211:5060;q=0.5 Expires: 60 Authorization: Digest username=“1005”, realm=“asterisk”, nonce=“50b8bdc7”, uri=“sip:80.78.16.214”, response="6c6c6c3304089377bd43fad179a8ed5c" Content-Length: 0 13 headers, 0 lines Using latest request as basis request Sending to 80.78.16.211 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 80.78.16.211:5060;branch=z9hG4bK-44f2b02f-5e8279-4b9 From: sip:1005@80.78.16.214;tag=504e10d3-13c4-44f2b02e-5e7e32-3749 To: sip:1005@80.78.16.214;tag=as391af9d7 Call-ID: 504e10d3-13c4-44f2b02e-5e7e2d-7313 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 60 Contact: sip:1005@80.78.16.214;expires=60 Content-Length: 0 to 80.78.16.211:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 80.78.16.211:5060;branch=z9hG4bK-44f2b02f-5e8279-4b9 From: sip:1005@80.78.16.214;tag=504e10d3-13c4-44f2b02e-5e7e32-3749 To: sip:1005@80.78.16.214;tag=as391af9d7 Call-ID: 504e10d3-13c4-44f2b02e-5e7e2d-7313 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 60 Contact: sip:1005@80.78.16.211:5060;expires=60 Date: Mon, 28 Aug 2006 08:59:31 GMT Content-Length: 0 to 80.78.16.211:5060 Scheduling destruction of call ‘504e10d3-13c4-44f2b02e-5e7e2d-7313’ in 15000 ms rmt214-16\*CLI\> Sip read: REGISTER sip:80.78.16.214 SIP/2.0 From: sip:1006@80.78.16.214;tag=504e10d3-13c4-44f2b02e-5e7e82-4b5a To: sip:1006@80.78.16.214 Call-ID: 504e10d3-13c4-44f2b02e-5e7e82-5a7c CSeq: 2 REGISTER Via: SIP/2.0/UDP 80.78.16.211:5060;branch=z9hG4bK-44f2b02f-5e8341-505f Max-Forwards: 70 Supported: replaces User-Agent: FXO\_GW Contact: sip:1006@80.78.16.211:5060;q=0.5 Expires: 60 Authorization: Digest username=“1006”, realm=“asterisk”, nonce=“7f5d27ad”, uri=“sip:80.78.16.214”, response="fcd4c564b120f5ef0e8192a3ce00af72" Content-Length: 0 13 headers, 0 lines Using latest request as basis request Sending to 80.78.16.211 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 80.78.16.211:5060;branch=z9hG4bK-44f2b02f-5e8341-505f From: sip:1006@80.78.16.214;tag=504e10d3-13c4-44f2b02e-5e7e82-4b5a To: sip:1006@80.78.16.214;tag=as043d6f9d Call-ID: 504e10d3-13c4-44f2b02e-5e7e82-5a7c CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 60 Contact: sip:1006@80.78.16.214;expires=60 Content-Length: 0 to 80.78.16.211:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 80.78.16.211:5060;branch=z9hG4bK-44f2b02f-5e8341-505f From: sip:1006@80.78.16.214;tag=504e10d3-13c4-44f2b02e-5e7e82-4b5a To: sip:1006@80.78.16.214;tag=as043d6f9d Call-ID: 504e10d3-13c4-44f2b02e-5e7e82-5a7c CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 60 Contact: sip:1006@80.78.16.211:5060;expires=60 Date: Mon, 28 Aug 2006 08:59:31 GMT Content-Length: 0 to 80.78.16.211:5060 Scheduling destruction of call ‘504e10d3-13c4-44f2b02e-5e7e82-5a7c’ in 15000 ms Destroying call '504e10d3-13c4-44f2b02e-5e7cde-d87’ Destroying call '504e10d3-13c4-44f2b02e-5e7d33-4820’ 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:1006@80.78.16.211:5060 SIP/2.0 Via: SIP/2.0/UDP 80.78.16.214:5060;branch=z9hG4bK1a34d51b From: “asterisk” sip:asterisk@80.78.16.214;tag=as4c510f0f To: sip:1006@80.78.16.211:5060 Contact: sip:asterisk@80.78.16.214 Call-ID: [64e3213826741679770a0be37c99bc6d@80.78.16.214](mailto:64e3213826741679770a0be37c99bc6d@80.78.16.214) CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 28 Aug 2006 08:59:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 80.78.16.211:5060 rmt214-16\*CLI\> Sip read: SIP/2.0 200 OK From: "asterisk"sip:asterisk@80.78.16.214;tag=as4c510f0f To: sip:1006@80.78.16.211:5060;tag=504e10d3-13c4-44f2b03e-5ebe06-3375 Call-ID: [64e3213826741679770a0be37c99bc6d@80.78.16.214](mailto:64e3213826741679770a0be37c99bc6d@80.78.16.214) CSeq: 102 OPTIONS Via: SIP/2.0/UDP 80.78.16.214:5060;branch=z9hG4bK1a34d51b Supported: replaces Content-Length: 0 8 headers, 0 lines Destroying call '64e3213826741679770a0be37c99bc6d@80.78.16.214’ 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:1005@80.78.16.211:5060 SIP/2.0 Via: SIP/2.0/UDP 80.78.16.214:5060;branch=z9hG4bK1b11cd34 From: “asterisk” sip:asterisk@80.78.16.214;tag=as3a12b8ec To: sip:1005@80.78.16.211:5060 Contact: sip:asterisk@80.78.16.214 Call-ID: [7cded46d3bee0e826cead573242785e1@80.78.16.214](mailto:7cded46d3bee0e826cead573242785e1@80.78.16.214) CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 28 Aug 2006 08:59:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 80.78.16.211:5060 Destroying call '504e10d3-13c4-44f2b02e-5e7d88-4695’ rmt214-16\*CLI\> Sip read: SIP/2.0 200 OK From: "asterisk"sip:asterisk@80.78.16.214;tag=as3a12b8ec To: sip:1005@80.78.16.211:5060;tag=504e10d3-13c4-44f2b03e-5ebe56-137b Call-ID: [7cded46d3bee0e826cead573242785e1@80.78.16.214](mailto:7cded46d3bee0e826cead573242785e1@80.78.16.214) CSeq: 102 OPTIONS Via: SIP/2.0/UDP 80.78.16.214:5060;branch=z9hG4bK1b11cd34 Supported: replaces Content-Length: 0 8 headers, 0 lines Destroying call '7cded46d3bee0e826cead573242785e1@80.78.16.214’ 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:1004@80.78.16.211:5060 SIP/2.0 Via: SIP/2.0/UDP 80.78.16.214:5060;branch=z9hG4bK684a3701 From: “asterisk” sip:asterisk@80.78.16.214;tag=as01b543a8 To: sip:1004@80.78.16.211:5060 Contact: sip:asterisk@80.78.16.214 Call-ID: [653d2a58753e27e832759f5b4a204b91@80.78.16.214](mailto:653d2a58753e27e832759f5b4a204b91@80.78.16.214) CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 28 Aug 2006 08:59:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 80.78.16.211:5060 Destroying call '504e10d3-13c4-44f2b02e-5e7ddd-5076’ rmt214-16\*CLI\> I would really appreciate it if anyone could help out with this problem. Kind Regards, David. ---
**Author:** ![IronHelix](https://avatars.discourse-cdn.com/v4/letter/i/da6949/32.png) [@IronHelix](https://community.asterisk.org/u/IronHelix)\ **Post date:** [August 28, 2006, 6:08pm UTC](https://community.asterisk.org/t/wellgate-fxo-and-asterisk/7599/4 "2006-08-28T18:08:00Z")
OK your device registers. Your sip debug however has no mention of a call coming in. The only things there are REGISTER/TRYING/OK (the device renewing its registration), and OPTIONS/404/OK (asterisk’s qualify=yes ‘ping’). Your problem, whatever it is, lies in the device. ---
**Author:** ![cangurcan](https://avatars.discourse-cdn.com/v4/letter/c/a87d85/32.png) [@cangurcan](https://community.asterisk.org/u/cangurcan)\ **Post date:** [January 26, 2008, 12:12am UTC](https://community.asterisk.org/t/wellgate-fxo-and-asterisk/7599/5 "2008-01-26T00:12:10Z")
Same problem I have, but I have tested it with same settings in 3CX (a windows PBX) and it is working. No problem seems at 38xx, I couldn’t try on \* 1.4, onyl tried on 1.2. Tried many config, even tried many firmware on fxo. no result, pls inform me if you can make Wellgate 38xx work with asterisk.