I think we had a misunderstand here, I didn't not try to integrate Asterisk with sipXecs using SIP trunk or something, they are just separate system with different gateways, I just dialed sipXecs extension from Asterisk, and Asterisk is using IAX trunk from our provider, and sipXecs using SIP trunk from the same provider.
I found this in the sipxivr.log, you can see attendant collected digites=11221177, what I dialed is 1217, and for 111011, what I dialed is 1101, for 44, what I dialed is 4. "2009-12-01T03:19:24.874000Z":222:sipXivr:INFO:*:Thread-17:00000000:sipxivr:"Attendant::attendant Collected digits=111011" "2009-12-01T03:19:24.874000Z":223:sipXivr:INFO:*:Thread-17:00000000:sipxivr:"Attendant::attendant Extension 111011 is not valid" "2009-12-01T03:19:49.075000Z":224:sipXivr:INFO:*:Thread-17:00000000:sipxivr:"Attendant::attendant Collected digits=11221177" "2009-12-01T03:19:49.075000Z":225:sipXivr:INFO:*:Thread-17:00000000:sipxivr:"Attendant::attendant Extension 11221177 is not valid" "2009-12-01T03:20:06.934000Z":226:sipXivr:INFO:*:Thread-17:00000000:sipxivr:"Attendant::attendant Collected digits=44" "2009-12-01T03:20:06.935000Z":227:sipXivr:INFO:*:Thread-17:00000000:sipxivr:"Attendant::attendant Extension 44 is not valid" Regards Phinux On Tue, Dec 1, 2009 at 1:51 AM, Josh Patten <[email protected]> wrote: > Do not set it on the IAX trunk, set it on the SIP trunk to your sipX > installation. It doesn't appear you have one set up, and I'm wondering how > you've got it working without a SIP trunk set up to that server. Create a > new SIP trunk and send all the numbers to sipX out that SIP trunk in your > outbound routing. Your SIP trunk should have the following settings: > > host=IP.OF.SIPX.INST > port=5080 > type=friend > insecure=invite,port > context=from-internal (check this one, I'm not sure what > AAH uses, or if you even need it) > disallow=all > allow=ulaw > dtmfmode=auto > > Remember to set your inbound/outbound routing rules to send the desired > numbers to sipX. > > Josh Patten > Assistant Network Administrator > Brazos County IT Dept. > (979) 361-4676 > > > > Phinux Zhang wrote: > >> Hi Josh >> >> I tried these four options (rfc2833, info, inband, auto) on our IAX trunk, >> but didn't work for me. We are using aster...@home 2.7 with Asterisk >> version 1.2.5. >> >> I am not sure if I was modifying on the right place, please see >> screenshot, is it right? >> >> Regards >> >> Phinux >> >> On Mon, Nov 30, 2009 at 5:02 PM, Josh Patten <[email protected]<mailto: >> [email protected]>> wrote: >> >> Try all three different modes until you find one that works for >> you. Usually dtmfmode=rfc2833 will work, with dtmfmode=info being >> used if dtmfmode=rfc2833 doesn't work, and dtmfmode=inband as a >> last choice. Only modify this for the SIP trunk you use for sipX. >> Leave everything else alone otherwise you may inadvertently cause >> things to go haywire on your production system. >> >> >> Phinux Zhang wrote: >> >>> Hi Josh >>> >>> Thank you for the info. I read the sipXbridge page, yes, as you >>> guess, "dead air" problem has been fixed. >>> >>> And about AA problem, actually, we have PSTN gateway and IAX >>> gateway configured on Asterisk, if I use PSTN gateway to dial >>> sipXecs AA, I can dial exntensions, but if I use IAX gateway, I >>> can't. Just for sure, you mean if I add dtmfmode=inband on IAX >>> configuation page will solve this problem, is that right? The >>> aster...@home currently is in production mode, I wouldn't touch >>> it unless I am sure. >>> >>> Thanks again. >>> >>> Regards >>> >>> Phinux >>> >>> On Mon, Nov 30, 2009 at 3:22 PM, Josh Patten >>> <[email protected] <mailto:[email protected]>> wrote: >>> >>> It would appear to me that something is not running through >>> sipXbridge or there is a misconfiguration with one of your >>> gateways >>> >>> what type of PSTN gateway are you using, or does it run >>> through Asterisk? >>> >>> Did you set Asterisk to point to port 5080 on your sipX box >>> after configuring sipXbridge? sipXbridge runs on port 5080. >>> >>> http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_and_Configuration >>> for more info on using sipXbridge. This should fix the "dead >>> air" after the "Please hold while I transfer your call" message >>> >>> I am 99% sure that the AA problems are related to DTMF. try >>> setting your DTMF settings to one of the options here: >>> http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode >>> >>> If you are using an old version of Asterisk (1.2 or so) then >>> you cannot directly connect the two systems, you have to use >>> an "intermediary" like sipXbridge due to the inadequacies of >>> Asterisk's SIP stack. >>> >>> If you need more Asterisk configuration information I will >>> try to oblige, as I know a bit about Asterisk. >>> >>> Phinux Zhang wrote: >>> >>>> Hello All >>>> >>>> We are working on deployment of sipXecs 4.0.4 in our >>>> company, but we have the following two problems related with >>>> sip trunk, could you please help me to take a look and give >>>> me some suggestions? Thanks in advance for any advices. >>>> >>>> 1. We used aster...@home as our production phone system, >>>> when I dial from Asterisk to sipXecs, the AutoAttends(will >>>> use AA for short) works fine, but I can't dial any extension >>>> from AA, AA keeps saying the extension is invalid, but I am >>>> sure the extension I dialed is valid, and AA even can't >>>> recognize the number specified on AA configuration page. >>>> >>>> 2. I can dial extension from AA when using PSTN line, but >>>> after I dialed one extension, AA said "Please wait....", >>>> after that just silence, I can't hear dial tone, but the >>>> phone I dialed ding ring, it's very strange. >>>> >>>> I hope I can fix this today, or we have to switch to other >>>> solution like Trixbox. I am waiting for you, I really hope >>>> you professional guys can help to figure it out. Thanks in >>>> advance, and if you need any information, just let me know, >>>> I'll do what I can. >>>> >>>> Thank you very much. >>>> >>>> Regards >>>> >>>> Phinux >>>> >>>> ------------------------------------------------------------------------ >>>> _______________________________________________ sipx-users >>>> mailing list [email protected] >>>> <mailto:[email protected]> List Archive: >>>> >>>> http://list.sipfoundry.org/archive/sipx-users Unsubscribe: >>>> http://list.sipfoundry.org/mailman/listinfo/sipx-users >>>> sipXecs IP PBX -- http://www.sipfoundry.org/ >>>> >>> >>> >> >> ------------------------------------------------------------------------ >> >> -- Phinux Zhang Network & System Administrator Nanjing Trixan Information Technology p: +86 25 8482 9559 ext.1512 f: +86 25 8482 2653 e: [email protected] w: www.trixan.com This electronic message, including its attachments, is confidential and may be privileged or otherwise protected. 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