Hello, I've been tinkering with Asterisk today for the fun of it, trying to set up my own domain. I've got pretty much everything working, including a DID number that connects to my extension.
However, I'm having a problem receiving calls from a particular peer, specifically my office's PBX, which allows dialing directly via SIP. (So, just to clarify, the DID number is not involved in this problematic scenario.) The symptom is that my office PBX demands rfc2833 support, and will immediately disconnect a call if the callee doesn't support it. I've tried resolving this a number of ways and the only way that actually works is to create a peer definition in sip.conf for my office's external SIP IP address. Then calls from my office to my Asterisk service work with no problems. However, the peer section is bare: [corp] type=peer host=corp.example.ip port=5060 context=default That's it. That solves the problem. Copying *every setting* in [global] has no effect. In other words, how I configure the peer doesn't appear to matter very much, only that it has a specific configuration at all. This makes no sense to me. I've attached sanitized SIP debug output, with the following replacements: * corp.example.com and corp.example.ip refer to the SIP PBX at my company. (corp.example.com may refer to either the IP or the domain name, due to an oversight while I was sanitizing. This should not be significant. corp.example.ip always refers to the IP.) * asterisk.tld refers to the domain name for my Asterisk install. * asterisk.ip refers to the IP address my Asterisk install binds to (public and static). * NUMBER is my phone number at the company. * phone.nated.ip refers to the public, NATed IP address of the softphone that is registered with Asterisk for my account. * phone.private.ip refers to the private IP address of the softphone. Note that I have nat=yes and directmedia=no for this device in sip.conf, and calls from my DID number work as well as calls to sip:[email protected]. Looking at the SDP negotiation, and at the telephone-event capability in particular, here is what I see happening: 1. Asterisk advertises telephone-event to my softphone. 2. My softphone advertises telephone-event to Asterisk. 3. Asterisk does not offer telephone-event to the company PBX. 4. The company PBX offers telephone-event to Asterisk. After the ACK represented by step 4, the company PBX immediately issues a BYE to Asterisk. If the company PBX has a peer defined in my Asterisk sip.conf file, Asterisk does offer telephone-event to the company PBX in step 3. I've been up and down this issue for a few hours and I cannot for the life of me determine why simply defining a peer causes Asterisk to offer telephone-event. I have tried specifying dtmfmode=rfc2833 or dtmfmode=auto in [global] and neither change has any effect. As I said above, I've copied every configuration directive in [global] into the peer definition for the company PBX, and calls still work. So I'm at a loss to explain this. The problem does not seem to stem from my configuration, but I'm not entirely sure what else could be the problem... an Asterisk bug perhaps? I don't want to jump to that conclusion since this is my first day tinkering with the software. Perhaps someone more knowledgeable can steer me in the right direction? Thanks, -- Chris Howie http://www.chrishowie.com http://en.wikipedia.org/wiki/User:Crazycomputers If you correspond with me on a regular basis, please read this document: http://www.chrishowie.com/email-preferences/ PGP fingerprint: 2B7A B280 8B12 21CC 260A DF65 6FCE 505A CF83 38F5 ------------------------------------------------------------------------ IMPORTANT INFORMATION/DISCLAIMER This document should be read only by those persons to whom it is addressed. If you have received this message it was obviously addressed to you and therefore you can read it. Additionally, by sending an email to ANY of my addresses or to ANY mailing lists to which I am subscribed, whether intentionally or accidentally, you are agreeing that I am "the intended recipient," and that I may do whatever I wish with the contents of any message received from you, unless a pre-existing agreement prohibits me from so doing. This overrides any disclaimer or statement of confidentiality that may be included on your message.
<--- SIP read from UDP:corp.example.com:5060 ---> INVITE sip:[email protected]:5060 SIP/2.0 To: <sip:[email protected]:5060> From: "Chris Howie" <sip:[email protected]>;tag=2958469 Via: SIP/2.0/UDP corp.example.com:5060;branch=z9hG4bKba6f9d45e934097b67461af5f Call-ID: [email protected] CSeq: 1 INVITE Contact: <sip:[email protected]:5060> Max-Forwards: 70 x-inin-crn: 2088742465;loc=%3cRegionDefaultLocation%3e Supported: join, replaces Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, SUBSCRIBE Accept: application/sdp Accept-Encoding: identity Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to corp.example.com : 5060 (NAT) Using INVITE request as basis request - [email protected] No matching peer for 'NUMBER' from 'corp.example.com:5060' Looking for 1000 in default (domain asterisk.tld) list_route: hop: <sip:[email protected]:5060> <--- Transmitting (NAT) to corp.example.com:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP corp.example.com:5060;branch=z9hG4bKba6f9d45e934097b67461af5f;received=corp.example.com From: "Chris Howie" <sip:[email protected]>;tag=2958469 To: <sip:[email protected]:5060> Call-ID: [email protected] CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.9-2+squeeze4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:[email protected]> Content-Length: 0 <------------> Audio is at asterisk.ip port 50646 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to phone.nated.ip:38284: INVITE sip:[email protected]:38284;ob SIP/2.0 Via: SIP/2.0/UDP asterisk.ip:5060;branch=z9hG4bK0c2f6bb1;rport Max-Forwards: 70 From: "Chris Howie" <sip:[email protected]>;tag=as75559e6d To: <sip:[email protected]:38284;ob> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.9-2+squeeze4 Date: Tue, 10 Jan 2012 22:01:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 292 v=0 o=root 110726824 110726824 IN IP4 asterisk.ip s=Asterisk PBX 1.6.2.9-2+squeeze4 c=IN IP4 asterisk.ip t=0 0 m=audio 50646 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:phone.nated.ip:38284 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk.ip:5060;rport=5060;received=asterisk.ip;branch=z9hG4bK0c2f6bb1 Call-ID: [email protected] From: "Chris Howie" <sip:[email protected]>;tag=as75559e6d To: <sip:[email protected];ob> CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:phone.nated.ip:38284 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP asterisk.ip:5060;rport=5060;received=asterisk.ip;branch=z9hG4bK0c2f6bb1 Call-ID: [email protected] From: "Chris Howie" <sip:[email protected]>;tag=as75559e6d To: <sip:[email protected];ob>;tag=9316088d3bcd446e8c6155fc44dd60b0 CSeq: 102 INVITE Contact: <sip:phone.nated.ip:38284> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- Transmitting (NAT) to corp.example.com:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP corp.example.com:5060;branch=z9hG4bKba6f9d45e934097b67461af5f;received=corp.example.com From: "Chris Howie" <sip:[email protected]>;tag=2958469 To: <sip:[email protected]:5060>;tag=as49c0593e Call-ID: [email protected] CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.9-2+squeeze4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:[email protected]> Content-Length: 0 <------------> <--- SIP read from UDP:phone.nated.ip:38284 ---> <-------------> <--- SIP read from UDP:phone.nated.ip:38284 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk.ip:5060;rport=5060;received=asterisk.ip;branch=z9hG4bK0c2f6bb1 Call-ID: [email protected] From: "Chris Howie" <sip:[email protected]>;tag=as75559e6d To: <sip:[email protected];ob>;tag=9316088d3bcd446e8c6155fc44dd60b0 CSeq: 102 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: <sip:phone.nated.ip:38284> Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 272 v=0 o=- 3535203672 3535203673 IN IP4 phone.nated.ip s=pjmedia c=IN IP4 phone.nated.ip t=0 0 a=X-nat:0 m=audio 39692 RTP/AVP 0 96 c=IN IP4 phone.nated.ip a=rtcp:40013 IN IP4 phone.private.ip a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 <-------------> --- (11 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 96 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 96 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port phone.nated.ip:39692 list_route: hop: <sip:phone.nated.ip:38284> set_destination: Parsing <sip:phone.nated.ip:38284> for address/port to send to set_destination: set destination to phone.nated.ip, port 38284 Transmitting (NAT) to phone.nated.ip:38284: ACK sip:phone.nated.ip:38284 SIP/2.0 Via: SIP/2.0/UDP asterisk.ip:5060;branch=z9hG4bK3197ed3b;rport Max-Forwards: 70 From: "Chris Howie" <sip:[email protected]>;tag=as75559e6d To: <sip:[email protected]:38284;ob>;tag=9316088d3bcd446e8c6155fc44dd60b0 Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.9-2+squeeze4 Content-Length: 0 --- Audio is at asterisk.ip port 50330 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP <--- Reliably Transmitting (NAT) to corp.example.com:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP corp.example.com:5060;branch=z9hG4bKba6f9d45e934097b67461af5f;received=corp.example.com From: "Chris Howie" <sip:[email protected]>;tag=2958469 To: <sip:[email protected]:5060>;tag=as49c0593e Call-ID: [email protected] CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.9-2+squeeze4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 236 v=0 o=root 332043567 332043567 IN IP4 asterisk.ip s=Asterisk PBX 1.6.2.9-2+squeeze4 c=IN IP4 asterisk.ip t=0 0 m=audio 50330 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:corp.example.com:5060 ---> ACK sip:[email protected] SIP/2.0 To: <sip:[email protected]:5060>;tag=as49c0593e From: "Chris Howie" <sip:[email protected]>;tag=2958469 Call-ID: [email protected] CSeq: 1 ACK Via: SIP/2.0/UDP corp.example.com:5060;branch=z9hG4bKdc9d109e3af4ad6761bf0b599 Max-Forwards: 70 x-inin-crn: 2088742465 Supported: join, replaces Content-Type: application/sdp Content-Length: 194 v=0 o=ININ 1112266235 1112266237 IN IP4 corp.example.ip s=Interaction c=IN IP4 0.0.0.0 t=0 0 m=audio 0 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer doesn't provide audio <--- SIP read from UDP:corp.example.com:5060 ---> BYE sip:[email protected] SIP/2.0 To: <sip:[email protected]:5060>;tag=as49c0593e From: "Chris Howie" <sip:[email protected]>;tag=2958469 Call-ID: [email protected] CSeq: 2 BYE Via: SIP/2.0/UDP corp.example.com:5060;branch=z9hG4bK1d54c94e4fd318c2ef376d9cf Max-Forwards: 70 x-inin-crn: 2088742465 Supported: join, replaces Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to corp.example.com : 5060 (NAT) <--- Transmitting (NAT) to corp.example.com:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP corp.example.com:5060;branch=z9hG4bK1d54c94e4fd318c2ef376d9cf;received=corp.example.com From: "Chris Howie" <sip:[email protected]>;tag=2958469 To: <sip:[email protected]:5060>;tag=as49c0593e Call-ID: [email protected] CSeq: 2 BYE Server: Asterisk PBX 1.6.2.9-2+squeeze4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE) set_destination: Parsing <sip:phone.nated.ip:38284> for address/port to send to set_destination: set destination to phone.nated.ip, port 38284 Reliably Transmitting (NAT) to phone.nated.ip:38284: BYE sip:phone.nated.ip:38284 SIP/2.0 Via: SIP/2.0/UDP asterisk.ip:5060;branch=z9hG4bK4b209864;rport Max-Forwards: 70 From: "Chris Howie" <sip:[email protected]>;tag=as75559e6d To: <sip:[email protected]:38284;ob>;tag=9316088d3bcd446e8c6155fc44dd60b0 Call-ID: [email protected] CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.2.9-2+squeeze4 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:phone.nated.ip:38284 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk.ip:5060;rport=5060;received=asterisk.ip;branch=z9hG4bK4b209864 Call-ID: [email protected] From: "Chris Howie" <sip:[email protected]>;tag=as75559e6d To: <sip:[email protected];ob>;tag=9316088d3bcd446e8c6155fc44dd60b0 CSeq: 103 BYE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '[email protected]' Method: BYE Really destroying SIP dialog '[email protected]' Method: INVITE
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