[sipx-users] phantom calls with huntgroup

idea about the nature of the problem? Does anyone already saw it? Thanks, Heros Ing. Heros Deidda Pre-sales VoIP Engineer ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/

Re: [sipx-users] audiocode mediant 1000 receive as answer from sipx a "407 authorization required"

Hi Nickolay, thank you for your answer. This way it works of course...but this is a workaround because a gateway should be not be a user. Heros Ing. Heros Deidda Pre-sales VoIP Engineer - Messaggio originale - Da: "Nikolay Kondratyev" A: "Heros Deidd

[sipx-users] audiocode mediant 1000 receive as answer from sipx a "407 authorization required"

ewhy should SCS ask audiocode for authentication? Any suggestion is appreciated. Thanks, Heros Details: INVITE sip:123456...@20044.it;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.172.3;branch=z9hG4bKac688684571 Max-Forwards: 70 From: ;tag=1c688678648 To: Call-ID: 688677735112000213.

[sipx-users] "Skype like" TAPI client to perform click to dial anywhere

quot; feature. Despite this, I could not find till now, any valid software (commercial or free) to perform this task. Anyone found anything like that? Thank you!! Heros Ing. Heros Deidda Pre-sales VoIP Engineer ___ sipx

Re: [sipx-users] HA DNS server failure on slave when master is down

1800 ; expire (30 minutes) 1800 ; minimum (30 minutes) ) Thank you, Heros Ing. Heros Deidda Pre-sales VoIP Engineer - Messaggio originale - Da: "Dale R Worley (Dal

[sipx-users] HA DNS server failure on slave when master is down

SRV 2 100 5070 scsslave.mysipdomain.it. $ORIGIN mysipdomain.it. scsslaveA 172.16.4.1 $ORIGIN scsslave.mysipdomain.it. _sip._tcp.rrSRV 1 0 5070 scsslave.mysipdomain.it. SRV 2 100 5070 scsmaster.mysipdomain.it. Thank you fo

[sipx-users] Problems performing SCS HA Upgrade from 3.10 to 4.04

day, Heros Ing. Heros Deidda Pre-sales VoIP Engineer ___ 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

Re: [sipx-users] sipxbridge does not send authorization

I confirm the problem was the firewall. For some reasons it rewrites the "via" field in the wrong way. Be careful to ALG. Problem solved. Thank you guys! Heros - Messaggio originale - Da: "Heros Deidda" A: sipx-users@list.sipfoundry.org Inviato: Mercole

[sipx-users] sipxbridge does not send authorization

=z9hG4bK9fcf98f381f708cd921a8bc04cc88761;rport=10342 172.16.172.2:5080:5080 is quite ugly and I suspect the firewall do that. I'll try changing that and provide feedback Heros Ing. Heros Deidda Pre-sales VoIP Engineer - Messaggio originale - Da: "Michael Picher" A: "Heros

[sipx-users] sipxbridge does not send authorization

bindings) Note that after the unauthorized there are 4 seconds before the new request. Inside the third SIP packet I see always REGISTER CSEQ 1 and not CSEQ 2 as I would expect It seems as if SIPXBridge dont listen the Unauthorized Any help would be appreciated Thank you, Heros

[sipx-users] Audiocode FXS 24 ports

Again on audiocodes! I have a MP-124 FXS, 24 port. Telephones are registered on sipX they can both place and receive calls BUT..no call progress tones are heard by the user. I suspect this is connected to CPT tones. Problem is that I don't know How to generate CPT tones for audiocodes.

[sipx-users] R: DECT/Wi-Fi SIP phones

I tried Nokia E65 and it is good with sipX. I suppose all E6X have similar performances. About DECT I would suggest Kirk server v500 as a good DECT system for big deployements, but I tried it with asterisk not with sipX. Heros -Messaggio originale- Da: Keith Gearty [mailto:ke

[sipx-users] R: Audiocodes packet loss when connected to enterprise switches

Hi Tony, Not all possible permutations. I tried Full duplex 100 on both. I’ll try 100 Half duplex too and provide a feedback. Heros Da: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Inviato: martedì 20 ottobre 2009 15.54 A: heros Cc: sipx-users@list.sipfoundry.org

[sipx-users] Audiocodes packet loss when connected to enterprise switches

you all, Heros ___ 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

[sipx-users] R: Sipxbridge stop working

000Z":36709:JAVA:DEBUG:scs500.myvoipdomain.it:Thread -2::CrLfTimerTask:"sending heartbeat to voip.eutelia.it" After that line sipxbridge just tell us that is sending heartbeat to Eutelia.. I cant understand why sipxbridge is not sending an INVITE to EUTELIA...it seems the call

[sipx-users] Sipxbridge stop working

provider? Thank you in advance, Heros ___ 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 si

[sipx-users] GUI Permissions for SipXecs users

voicemail. At least is there a way to permit only voicemail for all users? (without call forwarding) Thank you for your suggestions, Heros ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org

[sipx-users] R: R: 1535 tftp provisioning

/index.html Heros -Messaggio originale- Da: Gmb [mailto:sipxm...@gmail.com] Inviato: lunedì 14 settembre 2009 10.09 A: heros Cc: sipx-users@list.sipfoundry.org Oggetto: Re: R: [sipx-users] 1535 tftp provisioning Hi heros, thanks for the response, i see on Nortel site that last firmware is

[sipx-users] R: 1535 tftp provisioning

SD/MMC" - The firmware update will take about 10 minutes. During this time you'll see the screen changing colours and make strange things. Don’t worry! Heros -Messaggio originale- Da: Gmb [mailto:sipxm...@gmail.com] Inviato: giovedì 10 settembre 2009 15.31 A: sipx-users@li

[sipx-users] R: how to do Fax in SipXecs

>Hi, How to receive fax form PSTN to Mail server in SipXecs? >--Winson In Italy I got good results with evolve appliance + audiocode , but it's commercial: www.empixevolve.com Don't know if there is anyone that successfully did it with

[sipx-users] wish list

Hi all, is there a wish list for SipXecs project where people can "wish" features or vote for them? Thank you, Heros ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/

[sipx-users] R: R: R: Multitenanancy

an half a second before the autoattendant answer. Heros Da: Keith Gearty [mailto:ke...@glensound.co.uk] Inviato: venerdì 10 luglio 2009 13.04 A: heros Cc: 'Paul Scheepens'; sipx-users@list.sipfoundry.org Oggetto: Re: [sipx-users] R: R: Multitenanancy heros wr

[sipx-users] R: R: Multitenanancy

” The same for calls directed to 600 from 500 A bit tricky but working Heros Da: Paul Scheepens [mailto:pscheep...@epo.org] Inviato: venerdì 10 luglio 2009 11.19 A: sipx-users@list.sipfoundry.org Cc: h.dei...@sidin.it Oggetto: Re: R: Multitenanancy And what would

[sipx-users] R: Multitenanancy

r a phone to call a range of other phones. This obviously doesn't work if you use a phone that doesn't support digitmap Apart this limit, I don't see any other limit. Am I wrong? Heros ___ sipx-users maili

[sipx-users] Multitenanancy

Hi all, I was wondering if multitenancy is possible on SCS. -We have source routing -I just don't know how to avoid that a group of users call another group of users registered on SCS Any ideas? Thank in advance, Heros ___ sipx-

[sipx-users] R: Advice on Wi-Fi device

. On asterisk systems they could register and transfer calls I’m interested to know if this works too on SIPXecs Heros Da: Nikolay Kondratyev [mailto:k...@nstel.ru] Inviato: giovedì 9 luglio 2009 8.49 A: 'Kurt Siegfried'; sipx-users@list.sipfoundry.org Oggetto: Re: [sipx-use

[sipx-users] R: R: SIP TAPI no longer working?

Sorry for the question. I disco verde there are binaries for 64 bit too. I'll test it and provide feedback Here on the users list Heros -Messaggio originale- Da: heros [mailto:h.dei...@sidin.it] Inviato: venerdì 3 luglio 2009 17.43 A: 'Klaus Darilion'

[sipx-users] R: SIP TAPI no longer working?

Hi Klaus, thank you for siptapi I tested SIPTAPI 0.2.6 it on SIPX 4.01 and it works nicely for me. Is there a version of siptapi available for windows 64 bit systems? Thank you, Heros -Messaggio originale- Da: Klaus Darilion [mailto:klaus.mailingli...@pernau.at] Inviato: martedì 9

[sipx-users] pickup service problem with audiocodes fxs

don't see why Heros ___ 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/

[sipx-users] R: display caller id on polycom 330

is provided in the From headers. With audiocode I just set "Use Source Number as Display Name": yes Thank you all guys Heros ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/ar

[sipx-users] R: display caller id on polycom 330

tten 011234567. I think "unknown" is something related to the phonebook When language is changed then "unknown" changes according to the language. Heros ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http:

[sipx-users] display caller id on polycom 330

he same/ found a solution? Thank you in advance, Heros ___ 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-use

[sipx-users] R: R: R: Routing incoming PSTN or GSM calls

the "TO:" field should be populated by the gateway...maybe it is something like "TO: unkn...@mysipdomain.it". Then your prefix to manage all calls from that gateway will be "unknown". Heros ___ sipx-users mailin

[sipx-users] R: R: Routing incoming PSTN or GSM calls

in dialplan with prefix 0122345. Then activate the dialplan Heros ___ 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-user

[sipx-users] R: Increasing ring-time for extension??

is sent to voice mail when no answer?? Kind Regards, Dave N. You have to modify “Default Serial Fork Expiration” in system àserversàmyscsserver.mysipdomain àsip proxy Heros <>___ sipx-users mailing list sipx

[sipx-users] R: Routing incoming PSTN or GSM calls

SipX route the calls from a sip PSTN or GSM gateway using the rules in dialplan. If no rule that match incoming DID is defined than SipX rejects the call. Heros -Messaggio originale- Da: ingo...@netvision.an [mailto:ingo...@netvision.an] Inviato: giovedì 25 giugno 2009 15.19 A: sipx

[sipx-users] R: R: Get the busy tone on busy phone preserving voicemail

with call diversion for the Polycom phone. There is an option called "forward on busy". Then the call is forwarded to an autoattendant with a preregistered message with the "busy tone". 3) Create the autoattendant with the "busy tone" Heros -Messaggio origina

[sipx-users] R: R: Get the busy tone on busy phone

ooking for a cleaner solution. Heros -Messaggio originale- Da: Picher, Michael [mailto:mpic...@cmctechgroup.com] Inviato: venerdì 26 giugno 2009 9.48 A: heros; Dale Worley Cc: sipx-users@list.sipfoundry.org Oggetto: RE: [sipx-users] R: Get the busy tone on busy phone Actually, you might b

[sipx-users] R: Get the busy tone on busy phone

the only way I found. Maybe a second solution is forwarding to a "dummy" extension for which voicemail has been disabled. In this second case I have to dedicate a user Heros -Messaggio originale- Da: Dale Worley [mailto:dwor...@nortel.com] Inviato: giovedì 25 g

[sipx-users] Get the busy tone on busy phone

ng on busy" towards a dummy destination. but again I get the voicemail message. Thank you in advance for any suggestion. Heros ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/

[sipx-users] mailing list tool

Hi all, The question should be considered trivial: is there a tool to browse the mailing list and to perform a free text search inside it? Thank you, Heros ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http

[sipx-users] R: R: schedules for time-based rules on dialplan

n Italy) 0 followed by any number of digits --> (all calls permitted) Got It! Last but not least: You give to the superuser permissions for the first rule and to the limited user permissions for the second rule I think the same can be applied to force phones to use differe

[sipx-users] R: R: schedules for time-based rules on dialplan

t now using a SIP gateway or external SBC, you can do what I wrote only using another PBX.. Have a nice day, Heros -Messaggio originale- Da: Scott Lawrence [mailto:scott.lawre...@nortel.com] Inviato: martedì 14 aprile 2009 16.06 A: Picher, Michael Cc: sipx-users@list.sipfoundry.org

[sipx-users] R: schedules for time-based rules on dialplan

: giovedì 9 aprile 2009 16.13 A: heros; sipx-users@list.sipfoundry.org Oggetto: RE: [sipx-users] schedules for time-based rules on dialplan Not a bug... This is how the dial plans work... I'd love for it to change but this has never gotten any traction. The first dial plan entry that you match is

[sipx-users] schedules for time-based rules on dialplan

Hi all, I wrote a dialplan using schedules to provide time-based behaviour. Each rules is assigned to a schedule. Basically Schedule "Day" (9:00-13:00 14:00-18:00) is assigned to "ruleday", match calls from 01122334455 and dial extension 200 Schedule "night" (13:00-14:00 18:00-9:00) is assign