Re: [sipx-users] Gateway Suggestion for testing

Same problem here, can't find support for an Audiocodes MP203B Jose kenny.mitch...@ineos.com wrote in news:of4a8a89ef.857acc56-on802575e4.00498de1-802575e4.004a5...@unauthoris ed.ineos_hubs.com: > > The other issue with Audiocodes is support. There isn't any. Not > worth speaking of, anyway.

Re: [sipx-users] Book

Hi Scott, >From http://list.sipfoundry.org/archive/sipx-users/msg14927.html "The book is now available in what Packt calls RAW http://www.packtpub.com/article/raw-faq format. The book title is Building Enterprise Ready Telephony Systems with sipXecs 4.0

[sipx-users] Book

Hi, I while back someone mentioned about a book that could be downloaded about SIPX. Does someone have the information on where to get this book. I believe it was written by some of the Nortel Developers. Thanks Scott ___ sipx-users mailing list sipx-

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

No. There is just a default system wide setting at this time. >>> "Dave Nedved" 06/29/09 8:51 PM >>>

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

Ok, Cheers mate... Is there any way to individually set it per user? From: heros [mailto:h.dei...@sidin.it] Sent: Monday, 29 June 2009 7:19 PM To: 'Dave Nedved'; sipx-users@list.sipfoundry.org Subject: R: [sipx-users] Increasing ring-time for extension?? Da: Dave Nedved [mailt

Re: [sipx-users] unable to specify FTP directory

>From the web interface: System -> Backup, then specify FTP. I get a IP/Hostname box, username, password. If I enter just the IP or hostname it'll work, but if I enter ftp.example.com/gohere it fails. I expect to be able to point to a subdirectory, or a subdir of a subdir, etc. On Mon, 2009-06-29

Re: [sipx-users] unable to specify FTP directory

On Mon, 2009-06-29 at 16:26 -0400, Matt Keys wrote: > I tried using / but that didn't work. Is there a > way to specify which directory to upload to or are you forced to use > whatever chroot the server gives you? could you please be more explicit about what you're doing, what you expect, and what

[sipx-users] unable to specify FTP directory

I tried using / but that didn't work. Is there a way to specify which directory to upload to or are you forced to use whatever chroot the server gives you? ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.or

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

On Mon, 2009-06-29 at 17:23 +0200, heros wrote: > Unfortunately in the from field is only written 011234567. I think "unknown" > is something related to the phonebook > When language is changed then "unknown" changes according to the language. It is likely that there is a setting on the 330 to

Re: [sipx-users] sipx an PhonerLite

I downloaded it and installed it and it worked the first time. Configuration - I put in IP address of my server in proxy/registrar, domain name, user name and password. Worked like a charm. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.si

[sipx-users] sipx an PhonerLite

Hello, is anyone using PhonerLite ( http://www.phonerlite.de/index_en.htm ) together with sipx? I can't get is to register to my sipx. Regards Stephan -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 _

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

Hi heros, >2) forward the call 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". If you really don't want the number to be answered then you could try this

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

> Hi all, > > I’m facing a problem on phones Polycom 330. > > When incoming calls comes I see “unknown” written just before the > Caller ID like “unknown-011234567”. > > The problem is that the display is small on Polycom 330 so I cant see > the entire caller id. > > Anyone saw the same/

Re: [sipx-users] HA Registrar problem

Hi Sathya, Thanks for the issue number, I misread Scott's post..."have an issue" can also mean "have a problem"my fault. I am 99% sure I rebooted the server as well while trying to solve the problem. That would have taken care of the named reload. Also the named.conf file still contained t

Re: [sipx-users] Gateway Suggestion for testing

>>> Keith Gearty 06/29/09 10:47 AM >>> Tony Graziano wrote: > > Patton is good and supported worldwide. >The Patton SmartNodes are very powerful and provide a lot of advanced >features, but be warned that they have a VERY steep learning curve, and >can be very daunting for a new user. Basicall

Re: [sipx-users] Gateway Suggestion for testing

>>> Keith Gearty 06/29/09 10:47 AM >>> Tony Graziano wrote: > > Patton is good and supported worldwide. The Patton SmartNodes are very powerful and provide a lot of advanced features, but be warned that they have a VERY steep learning curve, and can be very daunting for a new user. Basically

Re: [sipx-users] Gateway Suggestion for testing

Tony Graziano wrote: > > Patton is good and supported worldwide. The Patton SmartNodes are very powerful and provide a lot of advanced features, but be warned that they have a VERY steep learning curve, and can be very daunting for a new user. Basically they are supplied in a non-working stat

Re: [sipx-users] SIPTAPI works again with sipXecs

On Mon, 2009-06-29 at 15:23 +0200, Klaus Darilion wrote: > > Maybe someone can update the first line of > http://sipx-wiki.calivia.com/index.php/Click-to-Dial_for_Outlook,_CardScan,_ACT!_using_SIP_TAPI > > and replace it with: > > Note: Recently there were some problems reported with that SIPT

Re: [sipx-users] Gateway Suggestion for testing

>>> 06/29/09 9:32 AM >>> In common with many others in this groupI've used an Audiocodes gateway before and found it less than satisfying. I used a Mediant 1000 with 4-port FXO cards, and found the voicequality to be rather poor. Maybe with a decent digital card it wouldbe fine, but FXO wa

Re: [sipx-users] HA Registrar problem

I forgot to mention that /etc/named.conf is basically configured correctly as a forwarding-only DNS. Problem is that the zone file definitions should not be there, that is this part: zone "th.internal.epo.org" IN { type master; file "th.internal.epo.org.zone"; allow-update { non

Re: [sipx-users] HA Registrar problem

Hi, Please refer XX-5723 (DNS server on master is not reloaded after a redundant proxy is added). Thanks, Sathya Scott Lawrence wrote: On Mon, 2009-06-29 at 14:57 +0200, Paul Scheepens wrote: Hi Scott, Thanks for pointing me in the right direction. I did the tests on my desktop

Re: [sipx-users] Gateway Suggestion for testing

In common with many others in this group I've used an Audiocodes gateway before and found it less than satisfying. I used a Mediant 1000 with 4-port FXO cards, and found the voice quality to be rather poor. Maybe with a decent digital card it would be fine, but FXO was troublesome and the qua

Re: [sipx-users] HA Registrar problem

I reconfigured named.conf on the primary to "caching only". And yes, I see the registrations now. BTW: To the user community: Am I the only one running in HA, or is everybody using sipxecs as a DNS, or . Best regards / Mit freundlichen Grüßen / Sincères salutations Paul Scheepens "Scot

Re: [sipx-users] HA Registrar problem

On Mon, 2009-06-29 at 14:57 +0200, Paul Scheepens wrote: > Hi Scott, > > Thanks for pointing me in the right direction. > > I did the tests on my desktop of course, and no it did not work on my > primary server. > I specified during installation that my primary server was NOT a DNS > serve

[sipx-users] SIPTAPI works again with sipXecs

Hi! I just successfully tested sipX 4.0.0-015321-i386 with SIPTAPI 0.2.6. Maybe someone can update the first line of http://sipx-wiki.calivia.com/index.php/Click-to-Dial_for_Outlook,_CardScan,_ACT!_using_SIP_TAPI and replace it with: Note: Recently there were some problems reported with that

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

On Mon, 2009-06-29 at 14:52 +0200, heros wrote: > Hi all, > > I’m facing a problem on phones Polycom 330. > > When incoming calls comes I see “unknown” written just before the > Caller ID like “unknown-011234567”. > > The problem is that the display is small on Polycom 330 so I cant see > the

[sipx-users] Gateway Suggestion for testing

I am going to order a small gateway today for testing and am wondering of there are any undocumented or otherwise experienced based suggestions anyone can share. I was thinking of an Audiocodes MP201B or Linksys SPA3102. Anyone got any suggestions for a single or dual fxo port unit with good audio

Re: [sipx-users] HA Registrar problem

Hi Scott, Thanks for pointing me in the right direction. I did the tests on my desktop of course, and no it did not work on my primary server. I specified during installation that my primary server was NOT a DNS server, but apparently it still is, named is running and the list of DNS server

[sipx-users] display caller id on polycom 330

Hi all, I'm facing a problem on phones Polycom 330. When incoming calls comes I see "unknown" written just before the Caller ID like "unknown-011234567". The problem is that the display is small on Polycom 330 so I cant see the entire caller id. Anyone saw the same/ found a solution? T

Re: [sipx-users] Caller ID Problem

On Mon, 2009-06-29 at 14:43 +0900, 황용석 wrote: > Hi All, > > I have a CID problem when SIP user calls externally. > > Example: > X-Lite -> sipXecs -> Trunk GW -> PSTN > > [sipXecs] > sipXconfig (4.0.1-015823 2009-06-19T07:16:14 ecs-centos5) > hostname: sip.mydomain.com > ip: 192.168.1.1 >

Re: [sipx-users] HA Registrar problem

On Mon, 2009-06-29 at 14:07 +0200, Paul Scheepens wrote: > Hi, > > I tried to upgrade my test system from 3.10.2 to 4.0.1 and failed. > So I did a clean install from CD, both on the primary and the > secondary server (Redundant SIP router). > > Now I don't see any registrations in the GUI, but

Re: [sipx-users] NETGEAR ProSafe VPN Firewall model FVX538v2

Hi Jun, Our experience shows that SIP ALGs on home/SoHo routers almost always break interoperability with sipXecs, therefore we very strongly recommend that you turn it off on your netgear. Some people had similar issues involving Netgear and things started working after the SIP ALG got disabled (

[sipx-users] HA Registrar problem

Hi, I tried to upgrade my test system from 3.10.2 to 4.0.1 and failed. So I did a clean install from CD, both on the primary and the secondary server (Redundant SIP router). Now I don't see any registrations in the GUI, but I can place calls successfully and clients say they are registered (Bri

Re: [sipx-users] sipXecs CD 4.0.1 fresh install http problem

Boy Aidil Sjam wrote: [...] > [org/sipfoundry/sipxconfig/sip/sip.beans.xml]: Cannot resolve reference to > bean 'sipImpl' while setting bean property 'target'; nested exception is > org.springframework.beans.factory.BeanCreationException: Error creating bean > with name 'sipImpl' defined in class

[sipx-users] Which ports do i need to enable in iptables to get working sipxecs properly with firewall?

Hi, i got sipxecs working properly, but firewall was disabled. Can you tell me which ports do i need to enable in iptables, because i want to use sipxecs with iptables. I don't want to leave this server with disabled firewall. Thanks in advanced! ___

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

>You're missing the point. If a call comes in from a PSTN (POTS) line >into a PSTN<-->SIP gateway, the PSTN line MAY provide Calling Party ID, >if that functionality is enabled at the exchange, but it WILL NOT >provide Called Party ID as there is no standard for providing such >information th

Re: [sipx-users] sipXecs CD 4.0.1 fresh install http problem

-Original Message- From: Boy Aidil Sjam [mailto:aidils...@prawedanet.co.id] Sent: Monday, June 29, 2009 12:45 PM To: 'jbort...@fschad.com' Subject: RE: [sipx-users] sipXecs CD 4.0.1 fresh install http problem I Looked in the ssl folder and found only ssl.crt and ssl.key Generated SSL c

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

heros wrote: >>But how can that be, since PSTN calls don't send a Called Party ID? >>Surely you cannot retrieve the DID number from a PSTN call. >> >> > > > >>Keith. >> >> > > >With a sniffer or tcpdump look at the "TO:" field in the SIP INVITE. >Example: TO: 0122...@mysipdomain.it >T

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

>But how can that be, since PSTN calls don't send a Called Party ID? >Surely you cannot retrieve the DID number from a PSTN call. >Keith. With a sniffer or tcpdump look at the "TO:" field in the SIP INVITE. Example: TO: 0122...@mysipdomain.it Then write a rule in dialplan with prefix 012234

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

Da: Dave Nedved [mailto:dned...@flexinet.com.au] Inviato: lunedì 29 giugno 2009 2.50 A: sipx-users@list.sipfoundry.org Oggetto: [sipx-users] Increasing ring-time for extension?? Hi, simple question... Does anyone know how you can increase the ring-time for an extension before a call

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

But how can that be, since PSTN calls don't send a Called Party ID? Surely you cannot retrieve the DID number from a PSTN call. Keith. heros wrote: >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

[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-use

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

Yes Mike, but I want to preserve voicemail for "no answer" case. I definitely think the solution at this time is forwarding to an autoattendant with a preregistered message. The final working solution is: 1) allow one call per line key for the polycom phone 2) forward the call with call divers