Is there anybody experienced similar problems? Why Auto-Attendant collects twice for the extension user dialed? Thank you all
Regards Phinux On Tue, Dec 1, 2009 at 2:29 PM, Phinux Zhang <[email protected]>wrote: > 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. The information is > solely for the intended recipient. If you are not the intended > recipient, this message was sent to you in error and you are hereby > advised that any review, disclosure, copying, distribution or use > of this message or any of the information included in this message > by you is unauthorized and strictly prohibited. If you have received > this electronic transmission in error, please immediately and > permanently delete this message and notify the sender by collect > telephone call to +86 25 8482 9559 or by reply to this e-mail > message. Thank you. >
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