[sipx-users] Setting of LBA in Polycom Soundpoint 430

Hi, I am trying to verify the feature of LBA in my Polycom 430 upon sipx 4.0.2-16166. According to the wiki http://sipx-wiki.calivia.com/index.php/Bridged_Line_Appearance, I should mark a user as "Shared" on the main User Identification screen of sipx, whereas, I cannot find any item box related t

Re: [sipx-users] good starting page for sipXecs(http://sipxecs.sipfoundry.org)

Personally, I would be against any "Official" forum that does not resolve to the SipFoundry domain or a sub domain. Having forums scattered about will only fracture the excellent support we get today from developers, and take precious development time away from them. -Original Message- Fr

Re: [sipx-users] Seeking: Unmanaged PRI gateway, under $1K

Though you have to use sipXbridge to handle REFER with it, you can buy an Adtran Total Access 904 for under $1k and it has full SIP call routing capabilities including DID routing, number manipulation, etc. and one DSX-0 PRI port. Firmware updates are free. I use the Total Access 908e with a PR

Re: [sipx-users] good starting page for sipXecs(http://sipxecs.sipfoundry.org)

I agree that an official sipX forum would make thing much easier. Searching thorough old mailing list archives is worse than pulling teeth. Plus everyone uses different email clients which creates a lot of inconsistency and if you get the digest like I do (because I don't want 1,000 new emails

Re: [sipx-users] Router for SIPX

describe inexpensive its not so much the requirements for the application, yet thats a heafty pipe, anything over that size pipe is going to come with some price tag. i mean inexpensive to one might be 1k, to you for this project it might be 5-10k, so whats the budget... On Mon, Aug 10, 2009 a

Re: [sipx-users] Router for SIPX

A used or refurbished Cisco 2621XM would probably fit the budget and work nicely. milosz wrote: what does "prioritizes voip over the internet" mean?  the qos you can do on streams where you don't control the network end to end is pretty limited.  if you don't understand this already then you a

Re: [sipx-users] good starting page for sipXecs(http://sipxecs.sipfoundry.org)

Again, happy to host a web site with forums for the sipx community, at no cost, if anyone wants to take me up on it. Mike On Wed, 12 Aug 2009 11:42:49 -0700, Jonathan Petersen wrote: > I found the wiki and list very easily when getting started with sipXecs.  I >  > first arrived at the www.sipf

[sipx-users] Seeking: Unmanaged PRI gateway, under $1K

Anyone aware of or have a used unmanaged gateway which works perfectly well with sipx? I need a minimum of 1 PRI to start, expandable would be great, able to send calls to different servers based on incoming DID and, flexible configuration, good feature set, software can be upgraded at little

Re: [sipx-users] good starting page for sipXecs(http://sipxecs.sipfoundry.org)

I found the wiki and list very easily when getting started with sipXecs. I first arrived at the www.sipfoundry.org website and immediately started looking for the word documentation... because I didn't find that I looked at the menu options and discovered wiki... seems close enough so I checked it

Re: [sipx-users] Router for SIPX

what does "prioritizes voip over the internet" mean? the qos you can do on streams where you don't control the network end to end is pretty limited. if you don't understand this already then you are in trouble. you should get a piece of gear that can shape inbound as well as outbound traffic. i

Re: [sipx-users] SipXbridge and Bandtel

On Wed, Aug 12, 2009 at 10:07 AM, Matt White wrote: >>Regardless of which ITSPs support it, it is a requirement of the >>offer-answer model that re-invite with no SDP attribute return an >>offer. This offer can be the same as the original offer but it is a >>protocol error to return OK with no SDP

Re: [sipx-users] SipXbridge and Bandtel

On Wed, 2009-08-12 at 08:28 -0400, Matt White wrote: > Official answer from Bandtel > > "we do not support reinvite without an SDP, this is a grey area in the > RFC it is not a rule that is a MUST. Nobody on the IETF SIP mailing lists would agree with that. For example, in RFC 3261, section 14.1

Re: [sipx-users] SipXbridge and Bandtel

>Regardless of which ITSPs support it, it is a requirement of the >offer-answer model that re-invite with no SDP attribute return an >offer. This offer can be the same as the original offer but it is a >protocol error to return OK with no SDP body. > >Note (thanks to Dale for digging this out for m

[sipx-users] How to change the IP Address

I need to change the IP address of the sipx server, and the gateway, where can that be done and if its unix how? Dan White Citadel IT Group | Network Solutions 2500 Quantum Lakes Drive Suite 203 Boynton Beach FL 33426 561.547.1207 dwh...@citadelitg.com www.CitadelITG.com

Re: [sipx-users] SipXbridge and Bandtel

On Wed, Aug 12, 2009 at 8:28 AM, Matt White wrote: >>>It appears Bandtel does not apparently handle re-INVITE with no SDP >>>correctly, >>> >>>Please report the problem to them. >>> >>>Ranga > > Official answer from Bandtel > > "we do not support reinvite without an SDP, this is a grey area in th

Re: [sipx-users] SipXbridge and Bandtel

>>I had this issue with bandtel. I was using an ingate with them. The issue I >>recall was the ORDER in which the SDP was sent, >>and i finally left them and >>went to Bandwidth.com where I had no issues with that. I didn't have a rosy >>experience with them >>(200 support emails in under 12 mo

Re: [sipx-users] good starting page for sipXecs (http://sipxecs.sipfoundry.org)

I'm new to the list but totally agree that a forum would be a better means of communications. I would be happy to offer the hosting/bandwidth for such a forum at no cost if anyone is interested. Mike On Wed, 12 Aug 2009 09:59:26 -0700, Todd Hodgen wrote: >  >  > -Original Message- > Fr

Re: [sipx-users] good starting page for sipXecs (http://sipxecs.sipfoundry.org)

-Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Damian Krzeminski Sent: Wednesday, August 12, 2009 6:29 AM To: sipx-users@list.sipfoundry.org Cc: peterja...@yahoo.com Subject: [sipx-users] good starting page for s

Re: [sipx-users] SipXbridge and Bandtel

>>> On 8/12/2009 at 7:15 AM, in message >>> <4a826c1402f1b...@mail.thesummit-grp.com>, "Matt White" >>> wrote: On Wed, Aug 12, 2009 at 6:12 AM, Matt White wrote: > I'm setting up a new connection to Bandtel with SipXbrdige. > > Calls are disconnected during a transfer or hold. For ex

Re: [sipx-users] SipXbridge and Bandtel

>>It appears Bandtel does not apparently handle re-INVITE with no SDP correctly, >> >>Please report the problem to them. >> >>Ranga Official answer from Bandtel "we do not support reinvite without an SDP, this is a grey area in the RFC it is not a rule that is a MUST. on transfer or on hold the

Re: [sipx-users] SipXbridge and Bandtel

>>I assume you have already applied >>http://track.sipfoundry.org/secure/attachment/20698/patch7.zip >> >>It appears Bandtel does not apparently handle re-INVITE with no SDP correctly, >> >>Please report the problem to them. >> >>Ranga Is this behavior configurable in sipXbrdige...ie to send the S

Re: [sipx-users] SipXbridge and Bandtel

Although bandtel worked flawlessly on my trixbox, I had a lot of issues with bandtel in regards to sipx. I moved to bandwidth.com and they all went away. I don't think bandtel is supporting Re-invite, so check with them about that first. On Aug 12, 2009, at 6:37 AM, Matt White wrote: >> I

Re: [sipx-users] SipXbridge and Bandtel

>I did not have the patch applied. I have now applied it but I now get an even >stranger behavior. Now a call comes >in, when it goes to transfer, the call >does not drop nor does it ring the phone. > >The call trace shows the last seq of events is Refer to SipXproxy and then a >407 Proxy Auth

Re: [sipx-users] Dealing with combined voice/faxing

The answer might be interesting to sipx users who really want a UC environment is this could be made to work. Basically, I think what is needed is an 'asterisk bridge gateway' with t38 handler. In our testing, we found that we could call voice or send a fax to the same DID so long as the asteri

Re: [sipx-users] SipXbridge and Bandtel

On Wed, Aug 12, 2009 at 6:12 AM, Matt White wrote: > I'm setting up a new connection to Bandtel with SipXbrdige. > > Calls are disconnected during a transfer or hold. For example, the call > comes into the autoattendant, caller dials an extension, you hear the > "please hold while try that exten

Re: [sipx-users] Dealing with combined voice/faxing

> I think Todd indicated the AudioCodes Mediant 1000 can handle this for > you in the gateway. I don't know that product so I won't comment about > it. Strange, I don't have that email in my thread but I do know about the audiocodes, the others mentioned in the book/documentation, and you and Ton

Re: [sipx-users] SipXbridge and Bandtel

On Wed, Aug 12, 2009 at 6:12 AM, Matt White wrote: > I'm setting up a new connection to Bandtel with SipXbrdige. > > Calls are disconnected during a transfer or hold.  For example, the call > comes into the autoattendant, caller dials an extension, you hear the > "please hold while try that extensi

[sipx-users] Direwall Ports

Which firewall ports have to be open to get full functionality of SIPX through the internet? Dan White Citadel IT Group | Network Solutions 2500 Quantum Lakes Drive Suite 203 Boynton Beach FL 33426 561.547.1207 dwh...@citadelitg.com www.CitadelITG.com

[sipx-users] SipXbridge and Bandtel

I'm setting up a new connection to Bandtel with SipXbrdige. Calls are disconnected during a transfer or hold. For example, the call comes into the autoattendant, caller dials an extension, you hear the "please hold while try that extension" and then the call drops. A call trace shows sipxbr

[sipx-users] good starting page for sipXecs (http://sipxecs.sipfoundry.org)

I found this comment from one of the users in JIRA: http://track.sipfoundry.org/browse/XX-6270 "I am at my wits end with this system. I cannot find a How to anywhere for this. Asteriske, Trixbox etc, are all over the place with videos, forums etc, but this, now where. Am I looking at the right pl

[sipx-users] DTMF not working with sipxbridge, inconsistently with unmanaged gateway

Has anyone experienced an issue with sipxbridge not allowing DTMF from les.net or bandtel.com? I was not able to get DTMF working with an AA when it originates from either of these providers. However, when I create an unmanaged gateway or bridge it will work inconsistently but requires a restart of

Re: [sipx-users] Phones not Restarting

If this is the initial profile send to the phone they will not reboot because the phone is not yet registered to the PBX. Restart the phones manually on the first profile send. Mike -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry

Re: [sipx-users] Dealing with combined voice/faxing

Mike, I think Todd indicated the AudioCodes Mediant 1000 can handle this for you in the gateway. I don't know that product so I won't comment about it. The SmartNode gateways that I like can detect fax tones and can route T.38 signaling but can not route to a different destination based on whethe