Thank you Josh It's a shame that we probably have to give up sipXecs, I worked on it almost one month. sipXecs does give us so many troubles, like call pickup never worked, hunt group chain doesn't really work .... We fixed most of the problems or just simplify our requirements, but at last ... My manager probably have made his decision, really frustrating.
Regards Phinux On Wed, Dec 2, 2009 at 10:11 AM, Josh Patten <jpat...@co.brazos.tx.us>wrote: > If you dial the number of your China office from Sydney going out the IAX > trunk and get the same issue, then it still points to either your provider > or you AAH box. You weren't clear on the exact call patterns that are > causing the issue. > > I've seen this problem several times and it is exclusively a problem with > Asterisk. Asterisk's DTMF support has always been marginal at best, though I > believe 1.6 fixed a lot of DTMF issues. Try some of the steps listed in the > link I sent you in my last post like turning off the IAX jitterbuffer. > > > > Josh Patten > Assistant Network Administrator > Brazos County IT Dept. > (979) 361-4676 > > > > Phinux Zhang wrote: > > Thank you Josh. > > But the strange story is: > > We have set up sipXecs and put it into production mode in our China Nanjing > office, let me demonstrate the two production servers as follow: > > 1. Sydney Office > ITSP(atpNET actually)<--IAX -> aster...@home > ITSP <--PSTN gateway-->aster...@home > > 2. China office > ITSP(China loal) <--PSTN gateway--> sipXecs > > Our China office is using PSTN as outbound trunk in sipXecs, when I tried > to dial the number of our China office, we got the same issue, I can dial > any extension number when using PSTN line, but I can't when using IAX line. > This should be able to demonstrate that our ITSP atpNet should be all right. > > China office was using Trixbox 2.2 with PSTN line before, at that time, > everything works fine whatever gateway I chose to use, PSTN or IAX. So our > aster...@home should be ok, right? Why sipXecs just doesn't read the dtmf > tone correctly? > > Regards > > Phinux > > On Wed, Dec 2, 2009 at 9:14 AM, Josh Patten <jpat...@co.brazos.tx.us>wrote: > >> OOOOHHHH I see...I thought you were trunking them together. In that case >> it's either an issue with the way you are sending DTMF to the provider or an >> issue with the way the provider is dealing with DTMF when converting it from >> IAX to SIP (since they are the same provider they don't actually hit the >> PSTN when you dial from one box to the other, the just take the call in the >> IAX trunk and send it out the SIP trunk). >> http://www.voip-info.org/wiki/view/Asterisk+DTMF gives some information >> about DTMF over IAX and a couple of things you can do to try to make it >> work. >> >> >> Josh Patten >> Assistant Network Administrator >> Brazos County IT Dept. >> (979) 361-4676 >> >> >> >> Phinux Zhang wrote: >> >> Hi >> >> Sorry for the confusion, actually they are two separated systems, the >> difference is that aster...@home is in production mode, and sipXecs are >> in testing environment, we are planing to switch aster...@home to >> sipXecs. >> >> 1. ITSP(atpNET actually)<--IAX -> aster...@home >> ITSP <--PSTN gateway-->aster...@home >> >> 2. ITSP(atpNET) <--SIP trunk--> sipXecs >> >> When testing, we found that if we dial the SIP trunk number attached with >> sipXecs from one phone registered with Asterisk, I can hear Auto-Attendant, >> but can't transfer to any extensions from AA because of duplicate dtmf tone, >> AND this only happens if we use IAX as outbound trunk in Asterisk, if we use >> PSTN line, everything works fine. The SIP trunk number attached with sipXecs >> is +61 2 8231 5745, feel free to test if you are using IAX line and >> aster...@home, but I probably can't pick it up cause I am in another >> office most of time. Does this make sense? >> >> Thank you very much and if you are not clear on some points, just let me >> know. >> >> Regards >> >> Phinux >> >> On Wed, Dec 2, 2009 at 1:21 AM, Josh Patten <jpat...@co.brazos.tx.us>wrote: >> >>> You've lost me >>> >>> From what I understand, here is your setup: >>> >>> PSTN ITSP <--IAX--> aster...@home <--SIP via sipXbridge--> sipX >>> >>> If the above scenario is true, then what I said previously stands; >>> However, if you plan on cutting over to sipX and taking Asterisk out of the >>> mix completely there shouldn't be any DTMF issues. Asterisk, especially 1.2, >>> has known DTMF problems that have caused me headache before, and the >>> solution was to create a SIP trunk and specify the DTMF mode that way >>> Asterisk knows how to deal with them. >>> >>> Could you clarify this is the case? >>> >>> Josh Patten >>> Assistant Network Administrator >>> Brazos County IT Dept. >>> (979) 361-4676 >>> >>> >>> >>> Phinux Zhang wrote: >>> >>> 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 <jpat...@co.brazos.tx.us>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 >>>> <phinux.zh...@trixan.com>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 >>>>> <jpat...@co.brazos.tx.us>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 < >>>>>>> jpat...@co.brazos.tx.us <mailto:jpat...@co.brazos.tx.us>> 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 >>>>>>>> <jpat...@co.brazos.tx.us <mailto:jpat...@co.brazos.tx.us>> >>>>>>>> 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 sipx-users@list.sipfoundry.org >>>>>>>>> <mailto:sipx-users@list.sipfoundry.org> 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: phinux.zh...@trixan.com >>>>> 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 sipx-users@list.sipfoundry.org >>>> >>>> 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/ >>>> >>>> >>>> >>> >
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