Thank you Josh But what I can't understand is, for example, if we use sipXecs as our phone system, and one of our customer uses aster...@home with the same version what I am using now, if the customer can't dial our number, I can't tell him to set up a sip trunk to point to our server or something, there should be a way to detect the repetition and correct it, what do you think?
Regards Phinux On Tue, Dec 1, 2009 at 3:19 PM, Josh Patten <[email protected]> wrote: > As I said before, you HAVE to have a SIP trunk set up on Asterisk pointing > to sipX and vice-versa to properly communicate between the two, there is NO > other way. What you are seeing is common with using the wrong type of DTMF > mode (digit repetition) and you can specify what type of DTMF to use if you > specify a trunk. Try what I said to try and report the results back. > > Phinux Zhang wrote: > > 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. >> > > > ------------------------------ > > _______________________________________________ > sipx-users mailing list [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/ > > >
_______________________________________________ sipx-users mailing list [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/
