Hi Staffan, Please allow me to comment on your answer to my colleague's post.
Our problem become apparent after Cisco Router forwarded an incoming call to SipXecs server. I observed that AA indeed processed the call transfer of the dialed extension: only that the call transfer don't get thru for some reasons. During the period of AA transferring the call to the dialed extension, there's no ring and took a while before the call is dropped. Is it correct to say, according to your post, that SIPX is doing the SIP REFER to transfer the call to the Cisco router? Should it be the other way around? Thank you for your comments and help on this. We're looking for more inputs from you. Regards and have a nice day! > Could it be the case that SipX is trying to use SIP REFER to transfer the > call and the Cisco Router is not supporting this? You can verify this > using Wireshark or the debug ccsip messages command on the router. On Wed, Mar 31, 2010 at 6:21 PM, ronald teng <ronaldt...@gmail.com> wrote: > Already disabled both acd and sip trunking (w/c were on by default). Calls > still not able to go thru to a user extension. As to whether Cisco is able > to handle "sip refer"... i assumed it does as some articles on the sip > foundry site talks about configuring cisco routers as a gateway. I'll try to > check on this further. If anyone here knows whether a cisco 2651xm using a > vic-2fxo (running firmware C2600 ADVENTERPRISEK9-M Version 12.4(15)T11) can > handle 'SIP REFER', i would really appreciate some info. > > update: found an interesting article about this sip refer thing > > http://www.ciscopartner.info/en/US/docs/ios/12_2t/12_2t15/feature/guide/ftsipcal.html#wp1063962 > will check on this further and update ya'll. Again if anyone already knows > the answer, pls do respond. thnx > > -Ron > > > > On Wed, Mar 31, 2010 at 5:36 PM, Tony Graziano < > tgrazi...@myitdepartment.net> wrote: > >> You go to the server, choose configure, and enable only the roles you >> need. >> >> If you have a PSTN only connection, you do not need SIP trunking enabled. >> I >> think this was mentioned to you before. >> >> Realize 4.1.7 is develoipment code and may not be upgradeable to 4.2 >> either. >> >> If the Cisco is not setup or able to handle "SIP REFER", it means it >> cannot >> successfuly transfer any calls routing through it. >> ============================ >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> Fax: 434.984.8431 >> >> Email: tgrazi...@myitdepartment.net >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> Fax: 434.984.8427 >> >> Helpdesk Contract Customers: >> http://www.myitdepartment.net/gethelp/ >> >> ----- Original Message ----- >> From: sipx-users-boun...@list.sipfoundry.org >> <sipx-users-boun...@list.sipfoundry.org> >> To: Staffan Kerker <ietf-li...@kerker.se> >> Cc: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> >> Sent: Wed Mar 31 05:29:16 2010 >> Subject: Re: [sipx-users] calls from PSTN to operator(AA)=ok but >> transferfromoperator to an extension=failed >> >> I tried doing a debug ccsip for when i am using plar 100 ....refer to >> attached file 'plar100'...cant find anything odd here >> >> I also tried debugging while i'm using plar 210 (an actual user extension, >> imaginary sexy receptionist *grin*) and the results are on attached file >> 'plar210'....this one shows 404 error...don't know why. I have adjusted >> the >> dialpeers accordingly. If you need to see my cisco gateway config, i can >> post it. >> >> not sure about this Sip Refer you mentioned (im totally new to sipX and to >> sip itself) >> as for the sipXbridge....how do i use that for my sipX connection to my >> gateway? I'm currently using sipX 4.1.7 and it seems sip trunking is >> running >> by default unlike in 4.0.4 where i have to turn it on manually. >> >> >> >> On Wed, Mar 31, 2010 at 3:22 PM, Staffan Kerker <ietf-li...@kerker.se >> >wrote: >> >> > On 31 mar 2010, at 08.56, ronald teng wrote: >> > >> > > I have a problem w/ calls from pstn not being able to go through >> to >> > a user extension. It will get to the operator/AA/ext 100 just fine but >> > after >> > dialling a user extension (as per the AA's instruction) the AA says it >> > will >> > be transferring but nothing happens...it just stays silent for around 10 >> > secs then gives a busy signal. I've attached the log sipX generated for >> > further info on the problem. (thnx to todd for explaining how find the >> > logs). >> > >> > >> > Could it be the case that SipX is trying to use SIP REFER to transfer >> the >> > call and the Cisco Router is not supporting this? You can verify this >> > using Wireshark or the debug ccsip messages command on the router. >> > >> > I haven't tried this, but isn't this one of the features of the >> > SipXBridge, >> > to "translate" REFER to SIP Re-INVITES? Maybe you can use >> > the SipXBridge on your SIP connection to the gateway. >> > >> > Regards >> > /Staffan >> > >> > -- >> > Staffan Kerker >> > mail/sip/xmpp: staf...@kerker.se >> > >> > "Don't get involved in politics man, just play the gig..." /Sgt Floyd, >> > Electric Mayhem Band >> > >> > >> > > > _______________________________________________ > 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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