The gateway is a Patton Smartnode 4960. I think it is not the bottleneck. In my test only two of the thirty channels were in use.
I compared the 486 messages in both cases. Aside the call-id, from and to-tag etc. they are exactly the same. The only difference is the 180 ringing before that exists or not. Just now I found the release notes of the 3.2.7 firmware of polycom. We are currently using 3.2.6. The release notes contain something interesting regarding this: http://downloads.polycom.com/voice/voip/relnotes/SIP_Software_Release_Notes_3_2_7.pdf *7**7**0**3**8 *Added support to generate ring back after a SIP 183 message, followed by a SIP 180 message. That’s not exactly the problem I see but maybe it’s worth a try. Is there any reason against 3.2.7? _____________________________ Jan Fricke (B.Sc.) *IANT - APPLIED NGN-TECHNOLOGIES **Turn-Key VoIP/UC Solutions and More... *Fon: +49 (5331) 6794 0 Fax: +49 (5331) 6794 499 Mail: [email protected] Web: www.iant.de <http://www.iant.de/en/startseite> IANT is eZuce <http://www.ezuce.com/> Elite Partner for EMEA IANT is Member of GROUPLINK <http://www.grouplink.de/> *Von:* [email protected] [mailto: [email protected]] *Im Auftrag von *Tony Graziano *Gesendet:* Mittwoch, 24. Oktober 2012 13:38 *An:* Discussion list for users of sipXecs software *Betreff:* Re: [sipx-users] Polycom Busy Problem I understand the call flow better now. What type of gateway it is "i think" has a lot to do with it. There is a difference between the called user is actually busy and when the gateway has all of its channels full and cannot process the call. Will you reveal what type of ISDN gateway you are using? In the non-working scenario is the ISDN "full" (meaning does it have channels available)? Do you have a call trace of a failed scenario? What might be meaningful is any text message in the header as to why the call failee (i.e. sip header is 486 but the reason code is "user busy" which in this case might be coming from the gateway itself"). On Wed, Oct 24, 2012 at 7:14 AM, Jan Fricke <[email protected]> wrote: It’s an outbound call from the polycom to pstn and the other side (e.g. my mobile) returns a busy signal. The ISDN gateway then sometimes sends a 180 Ringing to the polycom and after that a 486 Busy. If it sends 180 Ringing depends on the destination that was called. My mobile provider sends an alerting signal before the busy signal but other providers may not. If I call a provider that doesn’t send alerting before it refuses the call the gateway does not send 180 ringing to the polycom. Working scenario: Polycom sends Invite to Gateway. Gateway returns 180 Ringing and then 486 Busy here. Not working scenario: Polycom sends Invite to Gateway. Gateway directly return 486 Busy here. The user does not see anything on the polycom screen and does not hear a busy signal. He doesn’t know anything about the reason why the call could not be established. _____________________________ Jan Fricke (B.Sc.) *IANT - APPLIED NGN-TECHNOLOGIES **Turn-Key VoIP/UC Solutions and More... * Fon: +49 (5331) 6794 0 Fax: +49 (5331) 6794 499 Mail: [email protected] Web: www.iant.de <http://www.iant.de/en/startseite> IANT is eZuce <http://www.ezuce.com/> Elite Partner for EMEA IANT is Member of GROUPLINK <http://www.grouplink.de/> *Von:* [email protected] [mailto: [email protected]] *Im Auftrag von *Tony Graziano *Gesendet:* Mittwoch, 24. Oktober 2012 12:48 *An:* Discussion list for users of sipXecs software *Betreff:* Re: [sipx-users] Polycom Busy Problem Why would the phone ring busy? Have you artificially lowered the number of calls per line? Is VM enabled on the line or not? # of calls per line and whether there is VM will matter. On Oct 24, 2012 5:11 AM, "Jan Fricke" <[email protected]> wrote: Hi, I’m struggling with a Polycom related problem. This is more Polycom related than SipX but I think here are a lot of people that know Polycom phones very well. In some cases my Polycom does not play the user-busy tone. When calling pstn using ISDN it depends on the other side. The case that works: - ISDN Alerting arrives -> 180 Ringing to phone - ISDN Disconnect user-busy arrives -> 486 Busy here to phone The case that does not work: - ISDN Disconnect user-busy arrives -> 486 Busy here to phone The polycom seems to play the busy-tone only if there was a 180 RINGING before the 486 Busy here. Does anybody know if there is an option that I can set from SipX to manipulate this behavior? Sincerely Jan _____________________________ Jan Fricke (B.Sc.) *IANT - APPLIED NGN-TECHNOLOGIES **Turn-Key VoIP/UC Solutions and More... *IANT GmbH Salzdahlumer Straße 46/48 D-38302 Wolfenbüttel Fon: +49 (5331) 6794 0 Fax: +49 (5331) 6794 499 Mail: [email protected] Web: www.iant.de <http://www.iant.de/en/startseite> IANT is eZuce <http://www.ezuce.com/> Elite Partner for EMEA IANT is Member of GROUPLINK <http://www.grouplink.de/> Banking Connections: IANT GmbH, Volksbank BraWo, Konto-Nr. <12%2095%2098%2050>12 95 98 50 00, BLZ <269%20910%2066>269 910 66, IBAN-No.: DE02 <2699%201066%201295>2699 1066 1295 9850 00, BIC: GENODEF1WOB; Steuer-Nr.: <51%20203%2001203>51 203 01203; VAT: DE264352710; HRB 201710, Amtsgericht Braunschweig; CEO: Dipl.-Ing. Jan Schumacher, Prof. Dr.-Ing. Diederich Wermser This e-mail may contain confidential and/or privileged information. 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