Re: [asterisk-users] Odd DTMF problem when receiving calls
Thanks for the reply David, this clarifies a lot for me. On 01/10/2012 08:28 PM, David Backeberg wrote: > Wheee. You don't say anything about what 'company PBX' is, so we just > have to guess based on your description. Based on your description, > your 'company PBX' requires that the endpoints it communicates be > registered before-hand. Having a definition for the sip peer in > asterisk makes asterisk continually register with the 'company PBX'. The company's PBX is ININ (Interactive Intelligence IC). > So it is not necessarily your case that you think it is, that rfc2833 > is required, but rather that for 'company PBX', any sip endpoint must > first be registered. Both asterisk and 'company PBX' probably support > large numbers of possible DTMF or signaling possibilities, and it's > not surprising that you can get away with several possible values. I'm fairly certain that RFC2833 is required, as the client software clearly indicates that the connection was dropped due to lack of RFC2833 support: 17:53:23: Initializing 17:53:23: Sent to user Chris.Howie 17:53:23: Outbound Call: m...@myserver.example.com 17:53:23: Dialing 17:53:26: Disconnected [RFC 2833 Required] It appears that, by registering, Asterisk is able to ascertain what capabilities the peer supports and wants, and so it knows to offer the telephone-event stream to incoming calls. Since the SIP dialog differs in how Asterisk responds to the incoming call (it offers telephone-event only when the peer is defined in sip.conf), this seems like the case to me. Does that seem reasonable, or am I missing a piece of the puzzle? If this is the case, how would I get Asterisk to offer telephone-event on incoming calls without having to register every ININ-based PBX that might call me? (This is more theoretical than practical, as I doubt that any ININ PBX is going to call me over SIP.) > If you do 'sip show peers', with and without the config in your > sip.conf (use ; to comment it out and 'sip reload' to commit your > changes), you should be able to verify that THIS is the real > problem. One of the things I discovered during testing is that if I remove the peer and issue "sip reload" then incoming calls are still accepted. I have to completely restart the Asterisk daemon to get it to forget about the peer definition. > And no, this is not an asterisk bug. I suspected that it wasn't, which is why I asked here first instead of using the bug tracker. :) -- 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. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd DTMF problem when receiving calls
On Tue, Jan 10, 2012 at 6:00 PM, Christopher David Howie wrote: > 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? Wheee. You don't say anything about what 'company PBX' is, so we just have to guess based on your description. Based on your description, your 'company PBX' requires that the endpoints it communicates be registered before-hand. Having a definition for the sip peer in asterisk makes asterisk continually register with the 'company PBX'. So it is not necessarily your case that you think it is, that rfc2833 is required, but rather that for 'company PBX', any sip endpoint must first be registered. Both asterisk and 'company PBX' probably support large numbers of possible DTMF or signaling possibilities, and it's not surprising that you can get away with several possible values. If you do 'sip show peers', with and without the config in your sip.conf (use ; to comment it out and 'sip reload' to commit your changes), you should be able to verify that THIS is the real problem. And no, this is not an asterisk bug. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd DTMF problem when receiving calls
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:e...@iptel.org. 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:1...@asterisk.tld:5060 SIP/2.0 To: From: "Chris Howie" ;tag=2958469 Via: SIP/2.0/UDP corp.example.com:5060;branch=z9hG4bKba6f9d45e934097b67461af5f Call-ID: 1cc376aba0701c847ad96d075c4cc...@corp.example.com CSeq: 1 INVITE Contact: 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
