Re: [sipx-users] CDR problems

On 02/15/2012 06:48 PM, Tony Graziano wrote: > Restart call resolver. > I did a "/etc/init.d/sipxecs restart", but the problem was still there. Restarting only CDR solved the problem. > Plan to upgradeto4.4 ASAP. :-) Thank you. Massimo -- Massimo Vignone UniMORE -

[sipx-users] CDR problems

3:18.665072Z":DEBUG:getActiveCalls 6 "2012-02-15T16:43:48.66Z":DEBUG:getActiveCalls 6 "2012-02-15T16:44:18.657144Z":DEBUG:getActiveCalls 6 "2012-02-15T16:44:48.653167Z":DEBUG:getActiveCalls 6 "2012-02-15T16:45:18.647663Z":DEBUG:getActiveCalls 6

[sipx-users] Polycom Soundpoint with PPS

ate on the display. Do you have experienced some problems like this? Is it a firmware bug? Thank for your help. Massimo -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unimore.it IS

Re: [sipx-users] Recommended Firmware for Polycom use with SipX 4.4

Quoting from 3.2.5 Release Notes: "Although it is not a requirement, it is recommended that BootROM 4.2.3 be used in conjunction with SIP 3.2.5." HTH, Max -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:ma

Re: [sipx-users] connecting to a legacy pbx

ns Hicom, and we have plans to dismiss all the Siemens switches in the next months... I think that this can be replicated with every pbx with an available E1/T1 port. HTH, Massimo -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034

Re: [sipx-users] connecting to a legacy pbx

Hi, Actually I have 2 Siemens Hipath and 3 Siemens Hicom pbxs connected to SipXecs. On each pbx I've used an E1 port connected to a Patton Smartnode 4960, with Q.SIG license. Patton gateways are configured on SipXecs as unmanaged gateways. HTH, Massimo -- Massimo Vignone Un

Re: [sipx-users] SPA941's on 4.2.1

On 12/03/2010 05:25 PM, Tony Graziano wrote: > Anyone seeing issues with incoming calls on these? > I have some spa942 and some spa922, running 6.1.5a firmware, with no issues. What's the problem? -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2

Re: [sipx-users] ENUM not working

using any SBC, I have a public address on my proxy. To make it work do I need to enable the SIP trunking role? -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unimore.it ISN:80

[sipx-users] ENUM not working

17:B62FAB90:SipRegistrar:"[160-ENUM] SipRedirectorENUM::LookUp NAPTR regexp '^\\+390592058(.*)$' does not match for ENUM translation of 'sip:*2390592058...@unimore.it;user=phone' - no contact generated" Any suggestuion? Massimo -- Massimo Vignone UniMORE - Servi

Re: [sipx-users] Conference costs

uot;Invite partecipants" feature in the conference web page are logged in the SIPXCDR database as transferred calls. At the moment I can't find how to get those calls from the database. Do you know how to get back those calls? Massimo -- Massimo Vignone UniMORE - Servizi Informati

[sipx-users] Conference costs

Hi, I need to calculate costs generated by a conference call where most of partecipants were invited by the conference owner. Any suggestion? Massimo -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign

Re: [sipx-users] 1 Ring Diversions to Voicemail

not tagging correct DSCP value to some packets (Trying,Ringing and OK). HTH, Massimo -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unimore.it ISN:8032*881 ___ sipx-u

Re: [sipx-users] wireless SIP phones - opinions

I'm using Kirk WS6000 and I'm quite satisfied: it's simple to configure, very good audi oquality and radio coverage.and But as I wrote in a previous mail, it has an issue about call transfer. Massimo -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone:

Re: [sipx-users] SIP Door Phone or Entry Phone

On 10/23/2010 02:55 AM, Tony Graziano wrote: > (That new fisheye door cam is awesome). Just trying Mobotix T24: at the moment it seems that DNS SRV records are not supported. Massimo -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032

Re: [sipx-users] Critical: delayed TRYING with Polycom Soundpoint IP

etween them... The problem is that I can't reproduce the timeout. I'm wondering if using NAPTR and "source" validation for INVITEs could cause a slow answer from the phone. I will try to setup phones to use a static cache. Massimo -- Massimo Vignone UniMORE - Servizi In

Re: [sipx-users] Critical: delayed TRYING with Polycom Soundpoint IP

On 10/05/2010 01:09 PM, Michael Picher wrote: > Are the phones and sipXecs server on the same LAN segment / VLAN with no > firewall / nat between them? > > Mike No, they are in different subnets and VLANS, packets go through routers, without any NAT or firewall between them.

Re: [sipx-users] High availability troubles

nes (3.1.3revC). On these phones BLF is not used. -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unimore.it ISN:8032*881 ___ sipx-users mailing list sipx-users@list.

Re: [sipx-users] High availability troubles

On 10/01/2010 12:14 AM, Worley, Dale R (Dale) wrote: > > From: sipx-users-boun...@list.sipfoundry.org > [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Massimo VIGNONE > [massimo.vign...@unimore.it] > > In a HA scenario, with t

Re: [sipx-users] High availability troubles

On 09/24/2010 03:25 PM, Tony Graziano wrote: > what version of polycom firmware? > 3.1.3 RevC (3.1.3.0439) -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unimore.it ISN:80

Re: [sipx-users] High availability troubles

On 09/23/2010 10:10 PM, Massimo VIGNONE wrote: > Hi, > > In a HA scenario, with two servers running Sipxecs 4.2.1-018932, users > registered on one server can't call users on the other servers: they get > back a "408 Request timeout message". > > The problem ha

Re: [sipx-users] High availability troubles

> On 9/23/10 4:10 PM, Massimo VIGNONE wrote: > Sounds like DNS issues (if you have multiple a records with internal, > then external ip's?) We use only public IPs (we have a class B network addressing) My DNS setup: mydomain.net. IN NAPTR 2 0 "s" "SIP+D

[sipx-users] High availability troubles

Hi, In a HA scenario, with two servers running Sipxecs 4.2.1-018932, users registered on one server can't call users on the other servers: they get back a "408 Request timeout message". The problem happens randomly, so I can't reproduce it and tracing calls with Sipviewer doesn't help. So I have

[sipx-users] Another bug in upgrading to 4.2.1?

will not use the absoluthe path, but will follow the symbolic link /usr/share/www/doc/stdprompts/conf to copy the files. Should I open a JIRA issue? Massimo -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo

Re: [sipx-users] Upgrading from 4.2.0 to 4.2.1

I've solved the problem reverting to the package stunnel-4.26-1.i386.rpm and editing the file /etc/sipxpbx/sipxcallresolver-agent-config removing the line fips=no Thanks for your help. Max -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032

Re: [sipx-users] Upgrading from 4.2.0 to 4.2.1

I resend the mail, because I've replied to the wrong thread. Sorry!!! --- In the previous mail I was wrong... I've tried using stunnel 4.26, 4.28 and 4.33, but the CDR HA Tunnel never started. So now I'm wondering about what to do... Is the 4.2.1 release compatible with CentOS 5.2 or should I

Re: [sipx-users] Caller ID settings ignored since upgrade to 4.2.1.

In the previous mail I was wrong... I've tried using stunnel 4.26, 4.28 and 4.33, but the CDR HA Tunnel never started. So now I'm wondering about what to do... Is the 4.2.1 release compatible with CentOS 5.2 or should I have to upgrade the OS to a more recent version? If so, which version? I've

Re: [sipx-users] Upgrading from 4.2.0 to 4.2.1

the error message goes away. Hope this is enough to get the system working. Max -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unimore.it ISN:8032*881 ___ sipx-use

Re: [sipx-users] Upgrading from 4.2.0 to 4.2.1

s not properly created. Massimo -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unimore.it ISN:8032*881 ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Arc

[sipx-users] Upgrading from 4.2.0 to 4.2.1

16:07:18 LOG3[20031:3086472912]: SSL_CTX_set_cipher_list: 1410D0B9: error:1410D0B9:SSL routines:SSL_CTX_set_cipher_list:no cipher match Any idea? Massimo -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unim

[sipx-users] Updating localization package

Hi, I want to upgrade my localization package in my production system. I'm guessing if uploading it will reset the dial plan. If so, is there a smart upgrading procedure that will not reset the dial plan to defaults? Max -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Si

[sipx-users] Location based routing

ng to multiple group. Can you help me? thanks, Massimo -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unimore.it ISN:8032*881 ___ sipx-users mai

[sipx-users] Cannot restart Linksys phones

s that they don't accept the check-sync event. I can restart them only using the sipsend command, using the "reboot" event (or using the "reboot-cisco" event). The Auth_Resync-Reboot_1_ parameter is set to no, phones use 5.2.5 and 6.1.5a firmware. Any idea? Massimo

[sipx-users] Too many hops

Hi, I'm moving from a 3.8 based large installation to a 4.0.4 sipxecs based pbx. I have several patton smartnode gateways. All of them worked correctly with the 3.8 release. All of them are configured using the following example: ... gateway sip GW-MYDOMAIN bind interface LAN router servi

Re: [sipx-users] T38modem and Sipx

ely feasible to wrap that up in a tidy VM of some sort. If > youdo get it working though, please zip it up and share it somewhere! > > > Tony Thanks, Max -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign..

Re: [sipx-users] T38modem and Sipx

can do call setup in SIP and h.323. I'd like to do a fax server based on hylafax, using t38modem, and integrated in a sipx based installation. It is not a problem about t.38, but I'd like to know more about how to configure t38modem to interact with Sipx. Thanks, Max -- Massimo

[sipx-users] T38modem and Sipx

Hi everybody, Anyone has got some experience using t38modem (and hylafax) with a sipx HA installation? Thanks, Max -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unimore.it ISN:8032*881

Re: [sipx-users] HA sanity check

their own HA sip solutions and run > over our WAN? > How remote sites are connected to the headquarter? > Thanks, > > Kenny Mitchell > INEOS Refinery > Grangemouth > > > > > ___ > sipx-users mailing list > sipx-users@l

Re: [sipx-users] Routing calls from another PBX to the PSTN

Scott Lawrence wrote: > On Fri, 2009-04-03 at 09:57 +0200, Massimo Vignone wrote: >> Ok, this is the expected behaviour, so I configured an user on the >> gateway and in sipx with the right permission for long distance calls. >> But debugging on the gateway, it seems that t

Re: [sipx-users] Routing calls from another PBX to the PSTN

cy pbx were challenged again for authentication. That's why I wrote to the list: it seems that calls from gateway are always challenged for authentication with the user in the from header, and I'm guessing how I can change this behaviuor (if this was possible...) Today I'll do othe

Re: [sipx-users] Routing calls from another PBX to the PSTN

at most, even for hundreds of users, > and you get a granular level of control and monitoring out of it. i > think the ROI is ok there =) > Legacy pbxs can manage cdrs on their own so this is not a big problem. The goal is to maintain gateway configuration as simple as possible. Max -

Re: [sipx-users] Routing calls from another PBX to the PSTN

al plan are set with no particular permission, and if there is a way to bypass/disable authentication. Thanks, Max -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unimore.it ISN:8032*881 ___

Re: [sipx-users] Routing calls from another PBX to the PSTN

xml file containing new rules and modify the /etc/sipxpbx/sipxconfig.properties.in file to inject the rules in Sipx. Am I right? Where I can find some examples? Max -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign.

[sipx-users] Routing calls from another PBX to the PSTN

Hi everybody, Take a look at this scenario. Sipx with two gateways: gateway A is connected to another (traditional) PBX and gateway B is connected to PSTN. The traditinal PBX sends outgoing calls from extension 200 through the gateway A, then Sipx routes calls according its dial plan. In the di

Re: [sipx-users] Smartnode Gateways ACL troubles

the Outbound Proxy Server field. Sorry! Max -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unimore.it ISN:8032*881 ___ sipx-users mailing list sipx-users@list.

Re: [sipx-users] Smartnode Gateways ACL troubles

e *78+ext method on the same phone, directed call-pickup works well. It seemed to me that the former method bypasses the proxies, but I can't figure out why. Max -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:ma

[sipx-users] Smartnode Gateways ACL troubles

me why? Thanks, Max -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unimore.it ISN:8032*881 ___ sipx-users mailing list sipx-users@list.sipfoundry.org Lis

Re: [sipx-users] SipServerBroker error

dleMessage >> >> transport error" > > That last is probably important... it looks like the proxy is unable to > open the connection to the CDR database on that server. Check to see > that postgres is running. > > > > -- Massimo Vigno

Re: [sipx-users] SipServerBroker error

in.netForkingProxyCseObserver-10:B77A0B90:SipProxy:"ForkingProxyCseObserver::handleMessage transport error" Any idea? Massimo Scott Lawrence wrote: > On Fri, 2009-03-20 at 13:36 +0100, Massimo Vignone wrote: >> Hi, >> >> In my HA installation, on the distribu

[sipx-users] SipServerBroker error

n UNINITIALIZED or TERMINATED state (1)" Can you tell what does it mean? Thanks Massimo -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unimore.it ISN:8032*881 _

[sipx-users] How to lock/unlock a phone?

Hi everybody, I have some phones in a public area. Phones can call only internal extensions. I want to enable some users to unlock those phones and make them able to place outgoing (pstn) calls. How can i get that? Is it possible to lock/unlock a phone? TIA, Max

Re: [sipx-users] Gateways

Actually I have six Patton Smartnode 4960 with Q.SIG license connected to six Siemens Hicom/Hipath PBXs through E1 connections, plus a Patton Smartnode 4960 connected to the PSTN via an ISDN PRI. Everything works well. Hope this helps, Max -- Massimo Vignone UniMORE - Servizi Informatici

Re: [sipx-users] Integration with door access control systems

e same intercom interface connected to a Linksys PAP2. I can open the door answering the call and using two tones (pressing "#" and "2"). HTH, Max -- Massimo Vignone CeSIA - Università di Modena e Reggio Emilia Phone: +39.

[sipx-users] HA setup guide and sipxha-distrib.sh

27;t figure out how to finish the sipx installation. Thanks, Massimo -- Massimo Vignone CeSIA - Università di Modena e Reggio Emilia Phone: +39.059.2058032 Fax: +39.059.2058034 sip:massimo.vign...@unimore.it ISN:8032*881 ___ sipx-users mailing

Re: [sipx-users] Missing sipxha-distrib.sh in 3.10.3?

Massimo Vignone wrote: > Hi, > > I'm installing two Centos 5.2 boxes, with Sipxecs 3.10.3. > > I'm trying to configure them as master and distributed servers. > > Trying to enable SSL connections on the distributed server, I was not > able to find th

[sipx-users] Missing sipxha-distrib.sh in 3.10.3?

Hi, I'm installing two Centos 5.2 boxes, with Sipxecs 3.10.3. I'm trying to configure them as master and distributed servers. Trying to enable SSL connections on the distributed server, I was not able to find the sipxhadistrib.sh script? Any suggestion? Max -- Massimo Vig

[sipx-users] Polycom directed pickup

. I've checked the "directed call pickup" feature on SipXconfig, and the config file is correctly generated. I think there's some settings on the Polycom phone that I've not configured properly. Anyone can help me? Thanks, Max -- Massimo Vignone CeSIA - Uni

Re: [sipx-users] Siemens OptiPoint 420 problem - call drops at 20 min

mber all the steps I took to resolve it. Which firmware version do you have? Point your browser to the following link, and see if there's a release note related to a firmware version that addresses your problem. http://wiki.siemens-enterprise.com/index.php/optiPoint_410/420_S_Release_Notes By

Re: [sipx-users] GSM Gateway not working

mented UDP packets are dropped. I've submitted this issue to the manufacturer, I hope they could fix it. Thanks a lot, Massimo -- Massimo Vignone CeSIA - Università di Modena e Reggio Emilia Phone: +39.059.2058032 Fax: +39.059.2058034 sip:[EMAIL PROTECTED] ISN:8032*881 ___

Re: [sipx-users] GSM Gateway not working

Gateway: 155.185.3.247 CallId: D33B0B93CCC340659F7408F38CDB4BC00x9bb90303 Thanks, Max -- Massimo Vignone CeSIA - Università di Modena e Reggio Emilia Phone: +39.059.2058032 Fax: +39.059.2058034 sip:[EMAIL PROTECTED] ISN:8032*881 test-gsm-2-gw.xml.gz Description: GNU Zip compressed data

Re: [sipx-users] GSM Gateway not working

Thanks for your reply. Attached you will find a sipx-trace generated file. Call-id is dd42d8f69c661ea5 Sipx (v3.10.2) is 155.185.3.99 Grandstream phone is 155.185.3.121 GSM gateway is 155.185.3.247 The Grandstream phone calls a cellular phone, but gsm gateway doesn't respond to INVITE sent by Si

[sipx-users] GSM Gateway not working

t;algorithm=MD5" string. Attached to this mail you can find two scenario traces, both using Grandstream phones. Can you help me to understand where is the problem (and how to solve it?). Thanks in advance, Massimo -- Massimo Vignone CeSIA - Università di Modena e Reggio Emilia Phone: