Thanks for the help Tony, I believe your theory is right but I cannot understand why my Talkswitch is not handling the refer... See output below... Is it because the refer-to has [EMAIL PROTECTED] instead of [EMAIL PROTECTED] perhaps?
REFER sip:[EMAIL PROTECTED]:5060 SIP/2.0 From: <sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> >;tag=1807241842 To: "Lance Leger" <sip:[EMAIL PROTECTED]<[EMAIL PROTECTED]> >;tag=7713448461976159696 Call-Id: [EMAIL PROTECTED] Cseq: 3 REFER Contact: <sip:[EMAIL PROTECTED]:5100 ;transport=tcp;voicexml=https%3A%2F%2Flocalhost%3A8091%2Fcgi-bin%2Fvoicemail%2Fmediaserver.cgi%3Faction%3Dautoattendant%26name%3Daa_7> Referred-By: <sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]>> Refer-To: < sip:[EMAIL PROTECTED] > Date: Wed, 04 Jun 2008 16:33:26 GMT Max-Forwards: 19 User-Agent: sipXecs/3.10.1 sipXecs/vxml (Linux) Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY Supported: sip-cc, sip-cc-01, replaces, replaces Proxy-Authorization: Digest username="~~id~media", realm="mycompany.com", nonce="9e6bb036c2bccab3d4a34738219c430b4846c3d6", uri=" sip:[EMAIL PROTECTED]:5060", response="17810aa87d3f9c30ea261ea8d9b0a718" Via: SIP/2.0/UDP 192.168.100.107 ;branch=z9hG4bK-sipXecs-006a102f330db57a749e458bb4d9e4e3b4b3 Via: SIP/2.0/UDP 192.168.100.107:5100 ;branch=z9hG4bK-sipXecs-002bfad0799316c65cecfcbfd534ef8fa9e6 Content-Length: 0 On Mon, Jun 2, 2008 at 12:24 PM, Tony Graziano <[EMAIL PROTECTED]> wrote: > The auto attendant 'transfers' calls. > > The Talkswitch needs to be able to handle the refer properly. It's > getting the "refer' but not handling it, and am unafmiliar with > Talkswitch so i can't offer more. > > As I see it, the Talkswitch is sitting in between the registered user > and sipx, so it is a gateway. What kind of gateway it is does matter. > This sort of thing has been well discussed so maybe someone here can > assist you. > > The way sipx handles calls depends on the function. Once the AA answers > and you enter in your desired extension, your gateway needs to be able > to interpret the messages and right now I'm sure it is not. Dialing user > to user may not even pass through sipx, since Talkswitch sits in the > middle, the CDR would tell you more. > > This is very much like the sort of issues I found when trying to find a > gateway to register sip trunks with ITSP's. I finally had to settle on > Ingate, since they could deliver on all the problems we had. We could > call DID numbers, but couldn't route through the AA --OR-- transfer > outside calls from an outside user to another extension with some > gateways. > > Hope this helps. > > Tony > >>> "Lance Leger" <[EMAIL PROTECTED]> 06/02/08 12:58PM >>> > The phone is a Aastra 9133i (TalkSwitch brand) and it does not register > with > sipx directly, instead my Talkswitch device registers the phone to > sipx... > See registration from sipx below. I don't seem to have any problems > accessing this extension if I call it directly, it's just when I try to > call > it from the auto attendant. > > URI > Contact > Expiration [s] > > sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> < > [EMAIL PROTECTED] <[EMAIL PROTECTED]>> > 701011< > sip:[EMAIL PROTECTED]:5060> 1218 > > > On Sun, Jun 1, 2008 at 7:32 PM, Tony Graziano > <[EMAIL PROTECTED]> > wrote: > > > Did you check to see that the phone is actually registered and as a > 6 > > digit extension? > > > > What type of phone is it? Have you tested your DNS and made sure it > is > > working appropriately? Is the phone on the same subnet/network as > the > > sipx server? > > > > >>> "Lance Leger" <[EMAIL PROTECTED]> 06/01/08 20:14 PM >>> > > Thanks for the response Tony, re-activating the voicemail dial plan > > allows > > me to dial 8-xxxxx and access a voice mailbox, however, I'm still > unable > > to > > access 6 digit extensions from auto attendant. I get the > transferring > > message but hear nothing and the remote phone does not ring. Any > > suggestions? Should I undo the change I made to EXTENSION_LENGTH in > > config.defs? Thanks! > > > > > > On Sun, Jun 1, 2008 at 5:56 AM, Tony Graziano > > <[EMAIL PROTECTED]> > > wrote: > > > > > Lance, > > > > > > Editing the files manually is not a good idea, since they will get > > > overwritten by the system. > > > > > > When you modify/add/delete a dialing rule it's important to > "activate" > > > the dialing rule changes, which will restart the services and > write > > the > > > new rule into the dialing plan so the system can use it. > > > > > > Manual edits will be overwritten at this point, so using sipXconfig > to > > > do this is the best approach. > > > > > > I'm curious if you changed the internal length back to "3" if it > works > > > at all. Can one assume it worked before the extension length was > > > changed? > > > > > > > I would assume, and might be wrong, that any edit shows up in your > > > mappingrules.xml (click "show xml" on the activation screen) > before > > you > > > activate it. > > > > > > What type of phone are you dialing from and was it configured via > sipx > > > (the phones have a dialing plan too)? With some ATA adapters there > is > > a > > > "timeout" function or a "#" (terminate/send) character before the > > string > > > is sent. > > > > > > Tony > > > > > > >>> "Lance Leger" <[EMAIL PROTECTED]> 06/01/08 00:10 AM >>> > > > Greetings, are there any special configuration changes that have > to > > be > > > made > > > to use 6 digit extensions? I'm able to define and use the > extensions > > > when > > > making direct calls between phones, however, I can't access the > > > extensions > > > from the auto attendant or via the voicemail prefix. > > > > > > When I dial extension 100 to access the auto attendant, I enter the > 6 > > > digit > > > extension and hear a prompt advising that I'm being transferred > but > > all > > > I > > > hear is silence and the remote phone never rings.... > > > > > > Additionally, when I dial the voicemail prefix which is 8 followed > by > > > the 6 > > > digit extension, I hear a prompt advising that the destination is > > > unknown, I > > > get the same prompt if I dial any extension that is not 3 digits > even > > > though > > > I have changed the extension length for my voicemail dial plan. > > > > > > I've also tried changing the EXTENSION_LENGTH variable in > config.defs > > > with > > > the same result. > > > > > > Any suggestions? Thanks! > > > > > > > > > > >
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