gmane.comp.telephony.pbx.sipfoundry.general

Re: Trouble dialing in to SipX from PSTN

Michael LeBlanc — 2008-12-04T03:46:53

Re: [sipX-dev] Resend - CDR Enhancement Proposals -reActive Calls

Scott Lawrence — 2008-12-03T15:25:01

On Wed, 2008-12-03 at 20:20 +0530, Vikas Sharma wrote: There is a utility called 'sipxproc' that can restart processes. Note that it is only intended to be used as a developer tool - the interface and capabilities may change between releases.

Re: [sipX-dev] Resend - CDR Enhancement Proposals - reActive Calls

Vikas Sharma — 2008-12-03T14:50:38

Re: Hunt Group calls disconnect after 2-3 secs

Picher, Michael — 2008-12-03T10:35:33

You are correct that it is the root of the problem... NAT translation is a problem inbound AND outbound. If you take a firewall and map ALL traffic on port 5060 & etc. to the internal IP address of the PBX, what happens when that inbound call is told to talk to internal extension 202< at >sip.domain which happens to be at an internal IP address different than the PBX? The PBX doesn't stay in the middle of the call, it hands the call off and it is supposed to be handled in a peer-to-peer manner. The SBC needs to be able to stay in the middle of calls in and out to handle the NAT translations. Mike

Re: caller ID question

Tony Graziano — 2008-12-03T10:20:59

Hunt Group calls disconnect after 2-3 secs

Chad Leigh - Pengar Enterprises Inc — 2008-12-03T07:27:10

using sipXconfig (3.10.2-013143 2008-07-23T18:09:14 ecs-centos5) I set up a Hunt Group with extension 900. I have it concurrently ring extension 200 and 201. When dialing in from externally through my provider, if I have the dialplan for this external provider go to extension 900, about 2-3 seconds after the extension answers, the call is disconnected. If I dial extension 900 from a different internal extension, say 202, then when an extension answers, the calls are not disconnected. I am not sure I understand how the call routing takes place. sipX is on a machine behind my firewall. The firewall has a "virtual server" set up for port 5060 to go to the sipX machine. I have OpenSBC running on port 5062 and it handles the calls from sipX to the outside world but is not involved with calls from the outside world coming in as that seemed to work fine without involvement of OpenSBC by forwarding port 5060 traffic from the firewall. But I have the feeling that is the root of this p

caller ID question

Chad Leigh - Pengar Enterprises Inc — 2008-12-03T06:56:58

using sipXconfig (3.10.2-013143 2008-07-23T18:09:14 ecs-centos5) I configured a caller id on the gateway entry and when I call out that is used as the caller ID. None of the check boxes are checked in the gatewat caller id section. I want specific users to have their own caller id. I go to the caller id section of the user and add a caller id there. No boxes are checked. This caller id is ignored when I dial out through the gateway. How do I go about configuring it to allow me to set a custom caller id for a given user? Thanks Chad

Re: High Availability call fails

Vikas Sharma — 2008-12-03T05:05:23

Re: sipx-users Digest, Vol 58, Issue 3

Cuneyt M — 2008-12-02T18:14:57

Dear All, We have a new mediant 1000 with 2 E1 and 4 4-port FXO modules. Under sipx 3.8, when i add the gateway Mediant 1000, under Ports I can only see the E1 options but not the additional modules (i.e. FXO modules). I have checked the settings and there seems to be no place to key-in the config for the additional modules. Any guidance would be appreciated. cheers!

Re: [sipX-dev] 3.10.4 Release

Scott Lawrence — 2008-12-02T16:06:51

On Tue, 2008-12-02 at 08:33 -0500, Picher, Michael wrote: Correct... there will be a 3.10.3 sipXecs release... watch the build status page, and I (or someone) will post a note on the users list when it's in the release area.

Re: [sipX-dev] 3.10.4 Release

Marden Marshall — 2008-12-02T15:56:08

3.10.4 Release

Tony Graziano — 2008-12-02T12:46:49

Re: HA - Domain name

Picher, Michael — 2008-12-02T12:33:12

Re: HA - Domain name

Tony Graziano — 2008-12-02T11:23:56

You need to confirm your SRV records for sip. -----Original Message----- From: "Vikas Sharma" gmail.com> To: sipxusers list.sipfoundry.org> Sent: 12/2/2008 1:54:03 AM Subject: [sipx-users] HA - Domain name hi all I have installed sipx from CD ( Cent OS) in High Availability environment. domain - coralsip.com Master - ankur.coralsip.com (192.168.4.44) Slave vikas.coralsip.com (192.168.4.13) $nslookup ankur.coralsip.com resolves and give IP add $nslookup vikas.coralsip.com resolves and give IP add $nslookup coralsip.com doesn't resolve give nothing Do i need to give some additional entry in dns files to resolve domain name or no need for that? What server addr should i give, while registering the phone? If i give ankur.coralsip.com all are register to this master machine If i give vikas .coralsip.com all are register to this slave machine. In both cases HA not be achieved i think. If i give coralsip.com phones dont get

HA - Domain name

Vikas Sharma — 2008-12-02T06:54:03

Re: Trouble dialing in to SipX from PSTN

Dale Worley — 2008-12-01T22:14:10

You can run multiple instances on the same host; I've run 5 sipXecs's simultaneously. In principle it's simple, in practice it's messy. The port numbers are controlled by a configuration file. (The exact method is being changed, but the capability remains -- down in the bowels, sipXecs listening ports can be reconfigured.) The mess is that you have to give each sipXecs instance a different tree of directories to store its stuff in. You can do that by creating a chroot environment, or by rebuilding the code from source, providing "./configure --prefix=/root/of/sipXecs/execution/file/tree". Dale

Re: Trouble dialing in to SipX from PSTN

Scott Lawrence — 2008-12-01T20:24:07

On Mon, 2008-12-01 at 10:54 -0800, Michael LeBlanc wrote: yes, with great care - this really isn't a "supported" thing to do, but with manual configuration it can be made to work. no. What are you trying to do that you need this?

Re: Trouble dialing in to SipX from PSTN

Michael LeBlanc — 2008-12-01T18:54:21

Good point -- OpenSIPS is a piece I'd like to remove. I did get sipXproxy to start on a different port -- but then I had trouble configuring X-Lite to connect to the proxy on anything other than 5060. Can I run multiple instances of sipXproxy on the same host, or have the same instance listen on multiple ports? -----Original Message----- From: Scott Lawrence [mailto:scott.lawrence< at >nortel.com] Sent: December 1, 2008 10:39 AM To: Michael LeBlanc Cc: sipx-users< at >list.sipfoundry.org Subject: Re: [sipx-users] Trouble dialing in to SipX from PSTN On Mon, 2008-12-01 at 10:01 -0800, Michael LeBlanc wrote: > I figured out where I went wrong. I was pointing the gateway directly at > the SIP Registrar, rather than the SIP Proxy (5060). Once I had the > gateway talking to the SIP Proxy, the autoattendant and extensions with > aliases set to DiDs were found by SipX. > > The complicating factor is that our gateway has to talk to SipX on a > non-standard port, say port 6060, rather than 5060. So I had to set u

Re: Trouble dialing in to SipX from PSTN

Scott Lawrence — 2008-12-01T18:38:41

On Mon, 2008-12-01 at 10:01 -0800, Michael LeBlanc wrote: It sounds like you've got workarounds on your workarounds Getting sipXproxy to run on a different port is not a big deal.

SipX Active Directory Sync/SIP Password Storage

Michael LeBlanc — 2008-12-01T18:25:51

I was wondering if anyone has gone the route of AD synchronization for SipX user information. In particular, is it possible to actually synchronize a user's username/Sip-password with their AD username/password? I can think of reasons why this wouldn't be desirable (especially with passwords being stored on devices), but I was wondering if it's a possibility. My understanding is that Active Directory won't let you read a password attribute via LDAP, even if you're binding as a privileged user. Also, does SipX store sip-passwords in the clear? If so, are there any ways to hash it -- or is it generally accepted that the sip-password is a low security token, just by virtue of the way it's used (stored in text files on devices, etc...). Cheers, Mike

Re: Trouble dialing in to SipX from PSTN

Tony Graziano — 2008-12-01T18:05:33