cosymed

Here is the link: Inbound calls There are perhaps wrong examples all over the page.

Problem solved. I had to add another "!" to the Send call To Extension option. The correct regular expression that worked for me is "!([0-9]{2}$)!\1!t!40!". A lot of the examples from the wiki, if not all of them, didn't work for me and are perhaps also missing the last "!". Thanks.

Inbound calling works with Send call to extension set to "!([0-9]{2}$)!\1!t!41". But it sends the call always to the extension 41 even if the called number is 939642. Does someone know what is really used from "To" (log: To is <sip:939642@192.168.4.210>) in the Send call to extension regular expression? "!([0-9]{2}$)!t!41" doesn't work btw.

I already tried that, but nevertheless changed it again. # Trunk 1 in domain pbx.cosymed.de Name: berofix Type: gateway To: sip RegPass: ******** Direction: Disabled: false Global: true Display: RegAccount: RegRegistrar: 192.168.4.230 RegKeep: RegUser: Icid: Require: OutboundProxy: 192.168.4.230 Ani: DialExtension: !([0-9]{2}$)!t!41 Prefix: Trusted: true AcceptRedirect: false RfcRtp: false Analog: false SendEmail: UseUuid: false Ring180: false Failover: never HeaderRequestUri: {request-uri} HeaderFrom: {ext} HeaderTo: {request-uri} HeaderPai: {ext} HeaderPpi: {ext} HeaderRpi: {ext} HeaderPrivacy: id HeaderRpiCharging: icid-value={icid-value};icid-generated-at={ip-address};orig-ioi={domain} BlockCidPrefix: Glob: RequestTimeout: Codecs: 8 CodecLock: true DtmfMode: Expires: 3600 FromUser: Tel: true TranscodeDtmf: true AssociatedAddresses: 192.168.4.230 InterOffice: false DialPlan: UseEpid: false CidUpdate: Ignore18xSDP: false UserHdr: Diversion: Colines: DialogPermission: I still get the same error message. Edit: Forgot to change the Associated Adresses and the error message is now Trunk berofix@pbx.cosymed.de has country code 49, area code 8407. So i think it's one step in the right direction.

Yes i configured a trunk and i already added the configuration of the trunk in my first post. I can already place outbound calls and i set the "Send call to extension" option in the trunk to "!([0-9]{2}$)!\1!t!41", which should mitigate adding an alias name to an extension, if i understood that correctly. The /1 gets stripped in the snom ONE web interface if it should show the trunk in text form.

I'm new to all this voip stuff, so please excuse my missing knowledge. I can successfully place outbound calls with our berofix and snomONE/4.5.0.1090 Epsilon Geminids. But receiving calls doesn't work and i'm quite lost here. I followed the description on the beronet wiki Berofix with Snom One to set the system up but i'm always getting the error message "Received incoming call without trunk information and user has not been found". I've "played" with the General Settings of the trunk to no avail. Here is the sip log: [8] 2012/10/17 11:51:46: Last message repeated 2 times [5] 2012/10/17 11:51:46: SIP Rx udp:192.168.4.230:5060: INVITE sip:939642@192.168.4.210 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.230;rport;branch=z9hG4bKFSr4807ve0N7m Max-Forwards: 70 From: "0840793960" <sip:0840793960@192.168.4.230>;tag=Dv5766vK8pNUg To: "" <sip:939642@192.168.4.210> Call-ID: dbfe9e45-92f3-1230-7a93-899e4e38b674 CSeq: 34901287 INVITE Contact: <sip:192.168.4.230;transport=udp> User-Agent: Berofix VOIP Gateway (2.2) Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO, REGISTER Supported: timer, 100rel, replaces Min-SE: 120 Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 220 P-Asserted-Identity: "" <sip:0840793960@192.168.4.230;user=phone> P-Preferred-Identity: "" <sip:0840793960@192.168.4.230;user=phone> Remote-Party-ID: "" <sip:0840793960@192.168.4.230;user=phone>;party=calling;privacy=off;screen=no v=0 o=- 513107643738542725 5066403480193230594 IN IP4 192.168.4.230 s=- c=IN IP4 192.168.4.230 t=0 0 m=audio 5004 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 [8] 2012/10/17 11:51:46: Allocating for call port 49, SIP call id dbfe9e45-92f3-1230-7a93-899e4e38b674 [5] 2012/10/17 11:51:46: SIP Tx udp:192.168.4.230:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.4.230;rport=5060;branch=z9hG4bKFSr4807ve0N7m From: "0840793960" <sip:0840793960@192.168.4.230>;tag=Dv5766vK8pNUg To: <sip:939642@192.168.4.210>;tag=0efc4008a0 Call-ID: dbfe9e45-92f3-1230-7a93-899e4e38b674 CSeq: 34901287 INVITE Content-Length: 0 [7] 2012/10/17 11:51:46: Set packet length to 20 [6] 2012/10/17 11:51:46: Call-leg 49: Sending RTP for dbfe9e45-92f3-1230-7a93-899e4e38b674 to 192.168.4.230:5004, codec not set yet [8] 2012/10/17 11:51:46: Incoming call: Request URI sip:939642@192.168.4.210, To is <sip:939642@192.168.4.210> [5] 2012/10/17 11:51:46: Received incoming call without trunk information and user has not been found [8] 2012/10/17 11:51:46: call port 49: state code from 0 to 404 [7] 2012/10/17 11:51:46: Set packet length to 20 [5] 2012/10/17 11:51:46: SIP Tx udp:192.168.4.230:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.4.230;rport=5060;branch=z9hG4bKFSr4807ve0N7m From: "0840793960" <sip:0840793960@192.168.4.230>;tag=Dv5766vK8pNUg To: <sip:939642@192.168.4.210>;tag=0efc4008a0 Call-ID: dbfe9e45-92f3-1230-7a93-899e4e38b674 CSeq: 34901287 INVITE Contact: <sip:939642@192.168.4.210:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1090 Epsilon Geminids Content-Length: 0 [5] 2012/10/17 11:51:46: SIP Rx udp:192.168.4.230:5060: ACK sip:939642@192.168.4.210 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.230;rport;branch=z9hG4bKFSr4807ve0N7m Max-Forwards: 70 From: "0840793960" <sip:0840793960@192.168.4.230>;tag=Dv5766vK8pNUg To: "" <sip:939642@192.168.4.210>;tag=0efc4008a0 Call-ID: dbfe9e45-92f3-1230-7a93-899e4e38b674 CSeq: 34901287 ACK Content-Length: 0 [8] 2012/10/17 11:51:46: Clearing call port 49, SIP call id dbfe9e45-92f3-1230-7a93-899e4e38b674 [8] 2012/10/17 11:51:46: Hangup: Call 49 not found and here the configuration of the trunk: # Trunk 1 in domain pbx.cosymed.de Name: berofix Type: proxy To: sip RegPass: ******** Direction: Disabled: false Global: false Display: RegAccount: snomone RegRegistrar: RegKeep: RegUser: snomone Icid: Require: OutboundProxy: 192.168.4.230 Ani: DialExtension: !([0-9]{2}$)!!t!41 Prefix: Trusted: true AcceptRedirect: false RfcRtp: false Analog: false SendEmail: UseUuid: false Ring180: false Failover: never HeaderRequestUri: {request-uri} HeaderFrom: {ext} HeaderTo: {request-uri} HeaderPai: {ext} HeaderPpi: {ext} HeaderRpi: {ext} HeaderPrivacy: id HeaderRpiCharging: icid-value={icid-value};icid-generated-at={ip-address};orig-ioi={domain} BlockCidPrefix: Glob: RequestTimeout: Codecs: 8 CodecLock: true DtmfMode: Expires: 3600 FromUser: Tel: true TranscodeDtmf: true AssociatedAddresses: 1.2.3.4 InterOffice: false DialPlan: UseEpid: false CidUpdate: Ignore18xSDP: false UserHdr: Diversion: Colines: DialogPermission: Thanks.