Re: [sipx-users] Register Phones from the world

OK, I figured it out. Apparently, if the Sip Proxy fails a test, it's not actually started. I fixed the problem, Sip Proxy started, and now everything works…. ~Noah On Jun 8, 2012, at 1:57 AM, Noah Mehl wrote: I installed wireshark and watched that interface: [root@sipx1 ~]# tshark -i eth0

Re: [sipx-users] Register Phones from the world

I installed wireshark and watched that interface: [root@sipx1 ~]# tshark -i eth0 -f "udp port 5060" Running as user "root" and group "root". This could be dangerous. Capturing on eth0 0.00 71.67.121.189 -> 172.18.16.230 SIP Request: REGISTER sip:inno-360.com 0.003951 71.67.121.189 -> 172.1

Re: [sipx-users] Register Phones from the world

I've followed that article. But it's apparent to me that the sipxecs is ignoring my register packets. And when I say no firewall, I mean that I am not firewalling any port on the sipxecs machine. The remote site is behind a firewall/nat, but that doesn't explain why I don't see any register e

Re: [sipx-users] Register Phones from the world

A google search of "sipxecs remote worker wiki" brings this page up in the search - - http://wiki.sipfoundry.org/display/sipXecs/Remote+User+NAT+Traversal BTW - you indicate there is no firewall - are you saying that at the remote worker location, and at the sipXecs location you don't have a fi

Re: [sipx-users] Register Phones from the world

Terribly sorry, but I'm not following at all. Where is the "support remote workers" option live? Could you please point me to a place in the wiki describing it? Thanks. ~Noah On Jun 7, 2012, at 11:05 PM, Tony Graziano wrote: If you have the option support remote workers enabled. The worker

Re: [sipx-users] Register Phones from the world

If you have the option support remote workers enabled. The workers firewall may be an issue. On Jun 7, 2012 10:20 PM, "Noah Mehl" wrote: > There is no firewall, only static NAT. iptables is off on the sipxecs > machine. So, you're saying that sipxecs should, by default, respond to > register fro

Re: [sipx-users] Backup to restore onto newer versions

On Thu, Jun 7, 2012 at 3:49 PM, m...@grounded.net wrote: > Why would it not be possible to build into sipx, a method by which users > could backup and restore onto newer versions. > > Several times over the course of using sipx and having to upgrade, it has > been a rather painful process to go

Re: [sipx-users] Register Phones from the world

There is no firewall, only static NAT. iptables is off on the sipxecs machine. So, you're saying that sipxecs should, by default, respond to register from anywhere? ~Noah On Jun 7, 2012, at 10:17 PM, "Tony Graziano" mailto:tgrazi...@myitdepartment.net>> wrote: Yes. This would really depend o

Re: [sipx-users] Register Phones from the world

Yes. This would really depend on the configuration of any firewall in between though. On Jun 7, 2012 8:28 PM, "Noah Mehl" wrote: > It seems that sipxecs is ignoring my sip registrations from soft phones > from the world. Is sipxecs configured to not respond to 5060 from the > world? > > ~Noah >

Re: [sipx-users] 4.2.1 sends bad 200 OK Contact IP on incoming Trunk calls

I think Tony has stressed the fact that 4.2.1 is very out of date. Here is a reason to upgrade pasted from the other thread. --- Upgrade to 4.4. This has been fixed in this commit https://github.com/dhubler/sipxecs/commit/606b856cd0ee4e33310ed6ffa4173dd5385add3c These lines in a parti

Re: [sipx-users] Transfer from attendant failing

Upgrade to 4.4. This has been fixed in this commit https://github.com/dhubler/sipxecs/commit/606b856cd0ee4e33310ed6ffa4173dd5385add3c These lines in a particular + 2556 +// JEB: If this is a response for the ITSP set the global IP here 2557 +

[sipx-users] Register Phones from the world

It seems that sipxecs is ignoring my sip registrations from soft phones from the world. Is sipxecs configured to not respond to 5060 from the world? ~Noah Scanned for viruses and content by the Tranet Spam Sentinel service. ___ sipx-users mailing list

Re: [sipx-users] Backup to restore onto newer versions

I've been through the pains but it's been a while since that and searching for replies leads to so many different things. It was easier to just ask again. Thanks. On Thu, 7 Jun 2012 16:46:54 -0400, Tony Graziano wrote: > It does backup the SSL, but when you create a new server you must also > c

Re: [sipx-users] Transfer from attendant failing

Hi Ray My issue is a bit different as sipx converts REFER into INVITE without SDP and my ITSP does not support that. So I have convince them of that. I am sure your issue is HA related and I am of no help there I am afraid. Sven On Thu, Jun 7, 2012 at 9:55 PM, Ray Bobbitt wrote: > > > Sven, d

Re: [sipx-users] Transfer from attendant failing

Sven, did you get a fix? I'm having a similar problem on 4.2.1. with SipXecs replying to Incoming calls from Bandwidth.com with seemingly inconsistent contact info in the 200 OK reply to the Invite. Sometimes the contact contains the public IP and sometimes the Private IP of the primary server.

Re: [sipx-users] Directory and speed dial not loading

You have to send profiles and reboot the phones in order for them to load... On Thu, Jun 7, 2012 at 3:36 PM, Shawn Beard wrote: > I deployed about 25 Polycom 550's this morning running 3.2.6 . Some of > them have directories, some don't. None of them can see speed dials beyond > the 3 on the l

Re: [sipx-users] Backup to restore onto newer versions

It does backup the SSL, but when you create a new server you must also create SSL to get through the install process in order to restore. When restoring, all IP/hostanme/sipdomain may match in order to be successful. This has been asked and answered many times, until 4.6 comes out this is the proc

Re: [sipx-users] 4.2.1 sends bad 200 OK Contact IP on incoming Trunk calls

Anyone? ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/

[sipx-users] Backup to restore onto newer versions

Why would it not be possible to build into sipx, a method by which users could backup and restore onto newer versions. Several times over the course of using sipx and having to upgrade, it has been a rather painful process to go from one server to another. One of the problems is the ssl certifi

[sipx-users] Directory and speed dial not loading

I deployed about 25 Polycom 550's this morning running 3.2.6 . Some of them have directories, some don't. None of them can see speed dials beyond the 3 on the line buttons. If they hit the up arrow they don't see any speed dials. The mac-directory.xml files are fully populated. Any ideas ? Tha

Re: [sipx-users] Transfer from attendant failing

That has been the case that the Invite is sent out without SDP. It is my understanding that the receiver must accept that as a valid invite. It seems that is a way to query the other side to see what they have to offer. One ITSP that I have worked with had issues with that, but after showi

Re: [sipx-users] Regarding JIRA XX-10177.

Hi Andy. I just looked at the snapshot. There are many TCP connections to proxy. Do you use TCP transport for the phones ? Your dns zone file shows TCP and UDP have same weight. If you want to see the error faster, you can prioritize TCP maybe. _lvnd_ (^_^) On Thu, Jun 7, 2012 at 1:07 AM, wr

Re: [sipx-users] Transfer from attendant failing

Sorry for being so vague. The URI of the INVITE is > INVITE sip:194.6.238.85:5061 SIP/2.0 On another system I have, identical scenarion, it looked like this (other IP address of course) > INVITE sip:mod_sofia@194.6.238.85:5061 SIP/2.0 The 200OK coming back had then mod_sofia as in the contact fi

Re: [sipx-users] Transfer from attendant failing

Not sure what you mean by "because of our INVITE". The INVITE we sent is correctly formatted. pasting it again. What do find wrong in the INVITE? > INVITE sip:194.6.238.85:5061 SIP/2.0 > CSeq: 1 INVITE > To: >;tag=rpz2agt27

Re: [sipx-users] Transfer from attendant failing

But I believe the reason they are sending no Contact field is because of our INVITE 10600 21:21:22.433629 10.1.1.15 5080 194.6.238.85 5060 SIP Request: INVITE sip:194.6.238.85:5061, in-dialog Do you know why the URI has no proper address? On Thu, Jun 7, 2012 at 11:51 AM, Joegen Baclor wrote: >

Re: [sipx-users] Transfer from attendant failing

Indeed. 200 OK has no contact as well. So you now have a double whammy. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 176.34.136.92 : > 5080;branch=z9hG4bK158560057fd8d521ed786e9511dbfcf0343836 > To: >;tag=rpz2agt27fobzvvd.o > From:

Re: [sipx-users] Transfer from attendant failing

I totally agree! Just that often the answer is "Find another ITSP". In this case I want to put in some extra effort to resolve the issue. Not sure yet which side the issue originates. Sometimes it i easier to make a fix on our side than to get ITSP to change things. Sven On Thu, Jun 7, 2012 at 11

Re: [sipx-users] Transfer from attendant failing

Sven Evensen wrote on 07-06-2012 12:24:11: > I saw that and I believe we made a patch once to add SDP to the > reInvite. Will have to check that. > > But the BYE from sipx says "sipXbridge;cause=205;text=\"Protocol > Error - 200 OK with no contact\"". I am trying to understand if sipX > or Sip

Re: [sipx-users] Transfer from attendant failing

I saw that and I believe we made a patch once to add SDP to the reInvite. Will have to check that. But the BYE from sipx says "sipXbridge;cause=205;text=\"Protocol Error - 200 OK with no contact\"". I am trying to understand if sipX or Sippy is causing this. The original INVITE from Sippy has "Ano

Re: [sipx-users] Transfer from attendant failing

sipX bridge disconnects because when it sent an INVITE without SDP to Sippy B2BUA, that entity also responded with a 200 OK without SDP. That is uncompliance in Sippy part. INVITE sip:194.6.238.85:5061 SIP/2.0 CSeq: 1 INVITE To: ;tag=rpz2agt27fobzvvd.o From: ;tag=55886164 Call-ID: A916B10F-A

Re: [sipx-users] Transfer from attendant failing

Anyone? On Tue, Jun 5, 2012 at 1:05 PM, Sven Evensen wrote: > I have a new installation with a new SIP trunk provider. It seems > everything is working except transfer. sipx 4.4. > > Incoming external come through SIP trunk to attendant > Hit 1 to transfer to conference bridge > When transfer sta