Re: [sipx-users] Question about using Remote Worker Nat Traversal with a Freeswitch SBC

The short answer is, since FreeSWITCH is authoritative (A registrar by itself), It will not pass through the registrations to sipx nor would it passthrough challenge responses (401 and 407). This would limit you to use FreeSWITCH only as a sip trunk (unmanaged) gateway. On 01/10/2013 07:01 A

Re: [sipx-users] Polycom Auto Answer for Call Control Web Service

Configure the polycom as an intercom and dial *76(number) http://wiki.sipfoundry.org/display/sipXecs/Intercom+Calling On 01/09/2013 10:21 AM, Jeff Ferrara wrote: Hello, We have implemented a click to dial feature on our CRM using the sipX Call Control Web Service. A user clicks on the number

Re: [sipx-users] Incoming calls to SNOM disconnect after 30 seconds or so.

You should also take a look as to why proxy thinks its external port is 55060. sipx only supports synchronized firewall ports. -- INVITE sip:11404@192.168.1.110:2048;line=5nrfoelc SIP/2.0 Record-Route: Call-Id: bee104006623-50e810fd-729c9139-3b5e9008-8c3a119@127.0.0.1-0 Cseq: 210

Re: [sipx-users] Internal RLS traffic

I think 404 is the desired behavior and no change is needed on the config side. A non existing resource list because there is no current member of that list is the logical consequence. However, if openfire subscribe client is not checking for the existence of the resource list and blindly se

Re: [sipx-users] Intercom in 4.6

es menu, >> yes it is selected. >> >> On Wed, Jan 2, 2013 at 5:40 AM, George Niculae wrote: >>> Is intercom feature enabled in telephony tab? >>> >>> George >>> >>> >>> On Wednesday, January 2, 2013, Joegen Baclor wrote:

Re: [sipx-users] Intercom in 4.6

Compring the line below to a working test i did, there is one difference. A working intercom registrar log looks like this: "2013-01-02T05:27:00.231989Z":765:SIP:DEBUG:devat1.ossapp.com:SipRedirectServer-9:7fd1f167c700:SipRegistrar:"[130-MAPPING] SipRedirectorMapping::lookUp got 0 UrlMapping P

Re: [sipx-users] Intercom in 4.6

-0 repeatedEOFs-0" > "2013-01-02T03:50:10.321021Z":365:ODBC:DEBUG:sip.domain.com:SipRedirectServer-10:b41f3b70:SipRegistrar:"*76...@domain.com > is NOT present in namespace imdb.entity" > "2013-01-02T03:50:10.321118Z":366:ODBC:INFO:

Re: [sipx-users] Intercom in 4.6

Registrar log should tell. put it in debug and post. On 01/02/2013 11:34 AM, Roman Gelfand wrote: > I am using snom 370. The phone is set to auto answer. When dialing > *76{ext}, I get not found message. > > Thanks in advance > ___ > sipx-users maili

Re: [sipx-users] Latest Stage (4.4) outbound dialing issues with dialplan

Highly important that you confirm that this works for you pre dec-5 update. On 12/12/2012 12:01 AM, Tony Graziano wrote: Is it just me? I have a dialplan rule when I use "xx" plus 10 digits to strip the "xx" and send the 10 digits to a specified gateway. When I do this with the latest stage (

Re: [sipx-users] Intercom speed dials and presence

It's not removed in 4.6. It is a regression that is getting fixed at this very moment. It's one of those quirks trying to improve performance of RLS an you break something along the way. You will get this feature back in the next update. On 12/06/2012 11:22 AM, Kyle Haefner wrote: Hi all,

Re: [sipx-users] Can the Call Control Web API be used to transfer calls? (sipXecs 4.6)

On 12/05/2012 12:24 AM, Alan Worstell wrote: > On 12/3/12 5:17 AM, Joegen Baclor wrote: >> By existing call, you mean calls initiated by call controller or >> existing call as in any call that is active in sipX? For call >> initiated by call controller, you can simply se

Re: [sipx-users] Can the Call Control Web API be used to transfer calls? (sipXecs 4.6)

On 11/30/2012 06:43 AM, Alan Worstell wrote: > Hello, > I was able to use the Call Control Web API through the REST api to place > calls in sipXecs 4.6 (Thanks again, Mircea!) Now I am curious if the > same can be done to transfer existing calls? Say, User1 is on a call > with an outside person, I

Re: [sipx-users] Upgraded to v4.4.0 and now no MWI

look for SIP_REGISTRAR.MWI.UA.CISCO : ^(CSCO|Cisco) in /etc/sipxpbx/sipxregistrar/registrar-config.vm. Change it to SIP_REGISTRAR.MWI.UA.CISCO : ^(DISABLED). Resend profiles and see if you still get the error. I'm not sure if Cisco now supports solicited notify. This config change disables

Re: [sipx-users] Upgarded to v4.4.0 and now no MWI

Without the logs, it's any one's guess. Debug elvel log of from sip status would be nice. On 11/28/2012 05:51 PM, Paul Sander wrote: > > I am upgrading our voicemail to v4.4 and everything is ok > apart from no MWI. I see when debugging and error message > within sipstatus that says '481 Subscr

Re: [sipx-users] sipx 4.6 inbound ivr-->extension always not available

;m a bit hesitant to blame Bria. I will tomorrow also send Bria a log, and ask for their input. But my gut tells me its something else than their software. I have a spare Polycom at work -- I'll also get that provisioned and working. Nicholas --------

Re: [sipx-users] sipx 4.6 inbound ivr-->extension always not available

its something else than their software. I have a spare Polycom at work -- I'll also get that provisioned and working. Nicholas -------- *From:* Joegen Baclor [jbac...@ezuce.com] *Sent:* November 27, 2012 7:26 PM *To:* Discuss

Re: [sipx-users] sipx 4.6 inbound ivr-->extension always not available

Check the registration status of the phone in the admin UI. Is it registered? If so, is it registered as NATed (sipXnonat tag not present in contact)? If it is NATED, can you confirm if OPTIONS keep-alive is received by the phone occasionally if you sniff the packets from the phones networ

Re: [sipx-users] something weird with proxy(sipXtackLib) [high cpu]

:28 AM, Joegen Baclor wrote: Domenico, I committed to 4.6. Would you mind sending a patch for release-4.4 as well? On 11/26/2012 05:25 PM, Domenico Chierico wrote: well I've rewritten the patch against 4.6 this one should apply clearly thanks Domenico Chierico On Mon, Nov 26, 2012 at 8:

Re: [sipx-users] something weird with proxy(sipXtackLib) [high cpu]

Domenico, I committed to 4.6. Would you mind sending a patch for release-4.4 as well? On 11/26/2012 05:25 PM, Domenico Chierico wrote: > well I've rewritten the patch against 4.6 this one should apply clearly > > thanks > Domenico Chierico > > On Mon, Nov 26, 2012 at

Re: [sipx-users] something weird with proxy(sipXtackLib) [high cpu]

Domenico, I've reviewed your patch and I am accepting it for commit. However: [joegen@sipdevel sipxecs-master]$ git apply --check ~/Desktop/fix_sipxclient.patch error: patch failed: sipXtackLib/src/net/SipClient.cpp:834 error: sipXtackLib/src/net/SipClient.cpp: patch does not apply Can you c

Re: [sipx-users] Subscribe forwarding 4.4.0-update #22

; Sent: Thursday, November 15, 2012 14:21 > To: Discussion list for users of sipXecs software > Subject: Re: [sipx-users] Subscribe forwarding 4.4.0-update #22 > > Joegen, can you please also provide a 32bits RPM for testing? > > - > MM > > > On Thu, Nov 15, 2012 at 9:

Re: [sipx-users] Subscribe forwarding 4.4.0-update #22

t; which kind of approach are you following? > My basic idea was to use SUBSCRIBE plugin to filter out wrong contacts > and then start to improve the plugin implementation with a per > Request behaviour. > > On Thu, Nov 15, 2012 at 8:24 AM, Joegen Baclor wrote: >> Domenico, >

Re: [sipx-users] Subscribe forwarding 4.4.0-update #22

Domenico, Thanks for the patch. Yes this will work but it will break external voicemails like exchange. I'm working on a patch that would tackle both aliases and fallback rules and will be sending an rpm to test shortly. Joegen On 11/15/2012 12:21 AM, Domenico Chierico wrote: This patch

Re: [sipx-users] something weird with proxy(sipXtackLib) [high cpu]

sults, what I really like to know is your opinion about the validity of the approach, basically I think that check if socket is broken before read or write on it seems to be more safe way of manage. Do you agree ? On Tue, Nov 13, 2012 a

Re: [sipx-users] something weird with proxy(sipXtackLib) [high cpu]

Domenico, Thanks for the patch. Just clarifying, this patch is for the behavior you specified in the August 3 post? If I'm correct, All I need to do to reproduce is send an INVITE using TCP, on receipt of 183, close the socket. -j On 11/13/2012 10:53 PM, Domenico Chierico wrote: Just to s

Re: [sipx-users] Music on Transfer

There is a mix up. The fix is with 4.6 and it is for calls from a trunk provider to sipX traversing sipXbridge. In your case, the call is from internal phone towards the ITSP and you are asking whether it is possible to play MoH instead of hearing actual ring, then the answer to that is no.

Re: [sipx-users] Karoo Configs

Send me a tarball of /etc/karoo.conf.d and /var/log/karoo/sbc.log. Make sure it is in debug level . In the command line, just key in: #/usr/bin/karoo --reset-log-level debug You can send it to me directly if you don't want to expose your route info. Joegen Note: Two interfaces is not requ

Re: [sipx-users] Subscribe forwarding 4.4.0-update #22

My plate is a bit full currently but it should not take until mid November for a fix to be out. On 11/08/2012 06:25 PM, Elwin Formsma wrote: Hi George, Correct, stumbled uppon this issue on update #16. Decided to try #22 before posting to you guys. Any clue on when this might be fixed? Kin

Re: [sipx-users] High CPU sipXproxy (update #22)

Two things can cause this. 1. You have a port scanner scanning your TCP ports. 2. You have a remote connection attempting to connect as using an unsupported version of SSL (TLSv1, SSLV2, SSLV3). I have checked sipXportLib ssl implementation and we are configured to support all three so this

Re: [sipx-users] FreeSWITCH for SIP Trunking

I'd be more than willing to work with someone who is interested in contributing an admin to Karoo Bridge. Starting version 1.6 Karoo Bridge will be fully open source with the exception of the media engine which OSS Software Solutions would license commercially to support the Karoo Bridge deve

Re: [sipx-users] New 4.6 Beta Installation

List, This thread has been hijacked twice. It was originally [sipx-users] Exchange UM SipXecs 4.6 Voicemail Not Working hijacked as [sipx-users] Wiki seems to be down Then finally hijacked as [sipx-users] New 4.6 Beta Installation Please refrain from replying to existing thread then give

Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)

Transferring ITSP originated calls requires that your ITSP supports INVITE without SDP. Before barking on something on the system, check first if your ITSP supports this. If not, there is no way your ITSP will work with sipx initiated transfers. On 10/19/2012 01:19 PM, Tony Graziano wrote: >

Re: [sipx-users] 4.6 Cluster

need to be able to use TLS and clustering, and from what i see here i won't have comunication between my sites if i use anything except UDP. *From:*Joegen Baclor [mailto:jbac...@ezuce.com] *Sent:* Thursday, October 04, 2012 4:39 PM *To:* Discussion list for users of sipXecs softwar

Re: [sipx-users] 4.6 Cluster

Finally I got the trace I need to provide you with an analysis ./Server1/sipregistrar.log-20121004:"2012-10-03T11:28:33.730099Z":367:INCOMING:INFO:sipx1.callidus.local:SipClientTcp-31:7f1f8acf5700:SipRegistrar:"Read SIP message: Local Host:192.168.0.46 Port: 5060 Remote Host:192

Re: [sipx-users] 4.6 Cluster

Here's what happened to the bad call. Unforutenately, there is not enough information in the proxy what happened to the call prior to hitting IVR. On 10/04/2012 04:53 PM, George Niculae wrote: On Thu, Oct 4, 2012 at 11:47 AM, darthzejdr wrote: "201" 2472 "200" 3407 This are the curren

Re: [sipx-users] Outgoing calls - sipgate

In your dialing rule, get rid of 5060 in the host and try again. On 10/03/2012 06:14 PM, Mladen Gulan wrote: Hi, We installed new sipx on public ip without nat, outgoing calls still don`t works. In the attach there is snapshoot and homer logs. -Original Message- From: sipx-users-bou

Re: [sipx-users] ] SipXecs not transferring calls

Luis, Thanks for the logs. I'm from Manila too. Good to hear your company is using sipXecs. Hi George, I am referring this case to you. Perhaps you can shed light as to why the proxy is unable to find the credentials for ~~id~media. "2012-10-03T04:49:11.672255Z":10857:SIP:DEBUG:si

Re: [sipx-users] ] SipXecs not transferring calls

Luis, Unfortunately, the snapshot does not contain a single INVITE in the logs. Please do a log rotate or simply delete sipXproxy.log and do the test again. Send the proxy log instead of the whole snapshot. It should give us the information we need to shed some light on your issue. Are y

Re: [sipx-users] 4.6 Cluster

a/sipx-snapshot-sipx1.callidus.local.tar.gz https://www.dropbox.com/s/xwjq7q2px45b4sn/sipx-snapshot-sipx2.callidus.local.tar.gz calls are: 201 -> 200 works 200 -> 201 doesn't work 200 -> 201@192.168.0.46 <mailto:201@192.168.0.46> works *From:*Joegen Baclor [mailto:jbac...@ezu

Re: [sipx-users] 4.6 Cluster

George, the log below is for determining caller-id and has nothing to do with user or domain alias. For the meantime, can you make something out of this log. Registrar is unable to route to the user mailbox. "2012-10-02T11:51:25.801279Z":1352:SIP:DEBUG:sipx2.callidus.local:SipClientTcp-31:7f01a

Re: [sipx-users] 4.6 Cluster

Please send snapshots with both proxy and registrar in debug level. On 10/02/2012 07:08 PM, darthzejdr wrote: I've done a few more tests(extensions registered with normal domain(callidus.local) but using proxy field with ip adress) but again the same problem. II can make calls from primary se

Re: [sipx-users] Incorrect SDP from sipX after xfer completion

XX-10464>. - Jeff On Tue, Sep 25, 2012 at 12:57 PM, Joegen Baclor <mailto:jbac...@ezuce.com>> wrote: Jeff, I might be missing something but the SDP you pasted o=- 1348230370 1348230371 IN IP4 172.21.201.60 s=Polycom IP Phone c=IN IP4 172.21.201.60 t=0 0

Re: [sipx-users] Incorrect SDP from sipX after xfer completion

g behind a firewall". Is that relevant to the NAT traversal issue you mentioned? - Jeff On Tue, Sep 25, 2012 at 12:42 PM, Joegen Baclor <mailto:jbac...@ezuce.com>> wrote: >> In my configuration there is no NAT whatsoever. Is there a way to disable NAT travers

Re: [sipx-users] Incorrect SDP from sipX after xfer completion

no SDP. I'll open a tracker shortly. In my configuration there is no NAT whatsoever. Is there a way to disable NAT traversal completely, thereby working around this issue for the time being? - Jeff On Tue, Sep 25, 2012 at 12:09 PM, Joegen Baclor <mailto:jbac...@ezuce.com>> wro

Re: [sipx-users] Incorrect SDP from sipX after xfer completion

Jeff, good bug reporting! By late negotiation, do you mean INVITE with no SDP? There is a known issue with NAT traversal plugin not being able to handle this properly. If you don't mind, please open a tracker in jira and attach packet captures. On 09/25/2012 11:57 PM, Jeff Pyle wrote: Hell

Re: [sipx-users] MWI Subscription trouble with Polycom VVX500.

Remove the ** 5060 ** port in the URI sent by VVX500 SUBSCRIBE sip:0183820...@pros0x.sip.prosodie:5060 SIP/2.0 On 09/25/2012 10:12 PM, Marand Remi wrote: Hello, I can not obtain the MWI indication with a Polycom VVX500 Phone. SipXecs version : 4.4.0, Yum update is ok today. VVX 500 firmware

Re: [sipx-users] new patch for XX-10177

/25/2012 1:44 AM, Joegen Baclor wrote: Andrew, any update on this? Update #19 fixed many issues we were having with offsite registration. With #18 we had turned off 5061 and that seemed to fix most clients. We have not turned on 5061 with #19 yet. I would like a day or two of stability before

Re: [sipx-users] new patch for XX-10177

Andrew, any update on this? On 09/20/2012 04:58 AM, andrewpit...@comcast.net wrote: Hi Joegen, I just put it on my test system. We'll see... Thanks! Andy *From: *"Joegen Baclor" *To: *andrewpit...@

Re: [sipx-users] Difference

http://tools.ietf.org/pdf/draft-marjou-sipping-b2bua-01.pdf Freeswitch. sipXbridge sipXsbc (openuc only) On 09/24/2012 05:38 PM, Kumaran wrote: Hi All, Whats the difference between b2bua and proxy? Also in our server which service or feature act as b2bua ? Regards, Kumaran T _

Re: [sipx-users] Polycom template reject busy on dnd

George and myself tested this scenario right now in 4.6. I set my status to DND on Polycom and the call rolled over the VM. If you can replicate this consistently, please provide debug level proxy logs. On 09/22/2012 02:02 AM, Tony Graziano wrote: I'm see I g on 4.4 the call gets dropped an

Re: [sipx-users] Call forward fails to external number

This is a long standing issue and is all about 302 redirects not able to grant permissions of the called number to the caller. On top of this, branches will also be enforced if you have set one. The caller will not inherent the branch where the callee is located. On 09/20/2012 10:04 AM, Jeff

Re: [sipx-users] new patch for XX-10177

other places we could look for the root cause of this problem? Thanks, Andy *From: *"Joegen Baclor" *To: *andrewpit...@comcast.net *Cc: *"Discussion list for users of sipXecs software" *Sent: *Wednesday

Re: [sipx-users] linphone and jaK message

On Sep 3, 2012 6:58 PM, "Joegen Baclor" <mailto:jbac...@ezuce.com>> wrote: ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/

Re: [sipx-users] linphone and jaK message

Domenico, Thanks for the patch. But no offence, having to check jaK string in the stack layer is a bit off. How about this. If (buffer_len <= 4 && buffer != "CRLFCRLF") scram!; Gerneric and still wintin specs for keep-alive. Joegen On 09/03/2012 11:30 PM, Domenico Chierico wrote: > here

Re: [sipx-users] linphone and jaK message

On 09/01/2012 12:20 AM, Joegen Baclor wrote: > On 08/31/2012 08:01 PM, Domenico Chierico wrote: >> Seems that linphone sends to port 5060 some messages reporting "jaK" >> inside for keepalive purpose. >> Seems that this messages give troubles to the proxy. >&g

Re: [sipx-users] linphone and jaK message

On 08/31/2012 08:01 PM, Domenico Chierico wrote: > Seems that linphone sends to port 5060 some messages reporting "jaK" > inside for keepalive purpose. > Seems that this messages give troubles to the proxy. > I'm trying to fix this avoiding to parse this as SIP stuff > > thanks > Domenico Chierico

Re: [sipx-users] Is sipxProxy using wrong IP address? Causing one-way audio.

The answer is staring us right on the face Tony. The INVITE is towards sip:69143662@176.34.141.24 and not the domain. The proxy thinks it's routing out so it uses the external IP. Configure the calling UA to use domain. On 08/28/2012 08:14 PM, Tony Graz

Re: [sipx-users] 4.4 ACD

From the looks of it, sipxbridge somehow is sending the external IP of the server that is if 75.144.86.61:5080 is the IP of your firewall. If so, please send pertinent logs from sipXbridge in debug level. I've attached the filtered pcap in this mail. On 08/23/2012 06:56 AM, Gerald Drouillar

Re: [sipx-users] new patch for XX-10177

e any messages matching the PUBLISH method. I can send you these off list if you'd like. -Andy *From: *"Joegen Baclor" *To: *"Discussion list for users of sipXecs software" *Cc: *andrewpit.

Re: [sipx-users] new patch for XX-10177

en happening to some of our customers more than once within a 24 hour period, in order to keep them from cancelling with us we've had to resort to a watchdog script which sends OPTIONS messages to the servers periodically and restarts sipXproxy if it fails to respond.

Re: [sipx-users] new patch for XX-10177

sed to > consider any combination of CR's and LF's (as long as the > buffer contains only these) as a keepalive? > > Joegen Baclor wrote on Thu, 02 August 2012 22:28 >>>> Then it is not the issue. However, that perl script >> just >>>> became a

Re: [sipx-users] Cisco Hold/Resume

2012 03:43 AM, Ly Tran wrote: > Hi Joegen, > > Here's a trace of two external calls to the Cisco phone I made. One was put > on hold and the next one a transfer. > > Ly Tran > > -Original Message- > From: Joegen Baclor [mailto:jbac...@ezuce.com] > Sent: Tuesday,

Re: [sipx-users] sipx really likes 1.5GB ram

On 08/16/2012 12:42 AM, Michael Scheidell wrote: > you hear see the engineers at DOS, DISA, NASA, United Defense, laugh > when you talk about 'real time java' Really not a place to ridicule a tool where such tool shined. (coming from a C++ perspective)

Re: [sipx-users] Cisco Hold/Resume

You will usually get better results reporting your issues when there is something the developers could look at. If it's an easy fix, there is no reason why it can't be fixed. Send a trace/tcp dump. On 08/15/2012 05:22 AM, Ly Tran wrote: > The firmware running on the Cisco phones is 8-12. Work

Re: [sipx-users] SipXecs 4.6 latest build dependency error

I will fix this. for now, manually install libev yum install libev On 08/13/2012 03:20 PM, Mark Dutton wrote: > > Not sure if this should go in the tracker, but when I run a > yum update I get this at the end of the dependency list. > > --> Processing Dependency: libev.so.4()(64bit) for package

Re: [sipx-users] MWI and Aastra

Excerp from RFC 3261: For two URIs to be equal, the user, password, host, and port components must match. A URI omitting the user component will not match a URI that includes one. A URI omitting the password component will not match a URI that includes one.

Re: [sipx-users] new patch for XX-10177

re based on that. I'll continue my digging there, but I wanted to let you know. Would you mind having a look and let me know what you think? I'll forward along some logs from a recent hang, as well to make sure we're still on the right track.

Re: [sipx-users] Cisco 7940 Consultative Transfer

This is the record route issue he is facing and this is a known bug on cisco phones and will remain broken as long as sipx uses record routes to preserve states. I suggest you put cisco behind an SBC. See http://www.ossapp.com for a free one. On 08/02/2012 09:09 AM, Tony Graziano wrote: Is

Re: [sipx-users] good work sipx 4.40 and faxes, need help though.

I am only aware of blind transfer and ha support for cisco getting fixed by these params. Josh has better insight on this. Lets see if he chimes in. On 07/30/2012 07:11 PM, Michael Scheidell wrote: On 7/30/12 5:04 AM, Joegen Baclor wrote: This should give you back cisco blind transfers

Re: [sipx-users] Homer timezone and missing lines

Thanks. I took hold of the ticket. Make sure you also make a separate jira for the timezone issue. On 07/30/2012 06:14 PM, Mark Dutton wrote: > > Done. XX-10333 ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfo

Re: [sipx-users] good work sipx 4.40 and faxes, need help though.

Michael, Try the latest RPM from the repo and set the these values in the proxy config SIPX_TRAN_HOOK_LIBRARY.100_sipheadercheck : /usr/lib64/transactionplugins/libSipHeaderCheck.so SIPX_TRAN.100_sipheadercheck.REMOVE_ACCEPT_LANGUAGE : true SIPX_TRAN.100_sipheadercheck.REMOVE_REMOTE_PARTY_ID

Re: [sipx-users] Homer timezone and missing lines

Mark, Please create a jira for the missing packets. I'll spend some time looking into it. On 07/30/2012 03:12 PM, Mark Dutton wrote: > > Homer is now working (mostly) in 4.6. > > However, I see two issues. One is a frustration, the other a > bit more serious. > > The first issue is that althoug

Re: [sipx-users] Karoo Bridge Core Library is now open source!

, Jul 26, 2012 at 12:17 AM, Joegen Baclor wrote: >> OSS Software Solutions released the OSS Core library for Karoo Bridge >> under the L-GPL license today. This library is the B2BUA engine for >> Karoo Bridge and can give way for other projects such as sipXecs to link >>

Re: [sipx-users] Increase RTP port range

way audio/dead air issues. Dave. *From:*Joegen Baclor [mailto:jbac...@ezuce.com] *Sent:* Thursday, July 26, 2012 12:34 PM *To:* Discussion list for users of sipXecs software *Cc:* Dave May *Subject:* Re: [sipx-users] Increase RTP port range You left out the most vital piece of information. How m

Re: [sipx-users] Increase RTP port range

You left out the most vital piece of information. How many calls do you have physically active in space time? Is it about a 1000? On 07/27/2012 12:19 AM, Dave May wrote: We ran out of available sessions on our sipXecs OpenACD server this morning, so we increased the max-sessions from 1000

Re: [sipx-users] 4.6.0 install - homer dependencies

Yeah. There are still some issues related to SQL not saving the correct port information. On 07/26/2012 02:07 PM, Bryan Anderson wrote: I now see packets getting captured, just can't search. :) -Bryan Anderson On Wed, Jul 25, 2012 at 5:12 PM, Douglas Hubler > wrot

[sipx-users] Karoo Bridge Core Library is now open source!

OSS Software Solutions released the OSS Core library for Karoo Bridge under the L-GPL license today. This library is the B2BUA engine for Karoo Bridge and can give way for other projects such as sipXecs to link with a well tested and robust library for B2BUA and SBC purposes. On top of this,

Re: [sipx-users] Version 4.4 how many users support?

If you are using Karoo, then the proxy should have categorized the call as not natted. If you are running Karoo within a DMZ, make sure you are using setInterfaceAddress() function in your route.js to set the correct interface used to connect to sipx. Feel free to send in traces. On 07/24/20

Re: [sipx-users] [sipx-dev] SIP INFO Digit and sipx trunk

Probably. http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#liberal-dtmf On 07/20/2012 05:08 PM, Michele Mallano wrote: Hi there, I have a big problem with digits sent via SIP INFO instead of RTP way. The scenario is one Sipxecs connected to an ITSP provider via sip trunk. Everything

Re: [sipx-users] Bug fix release update: sipXecs 4.4.0 update #17

r. Thanks! Kyle On Wed, Jul 11, 2012 at 2:46 AM, Joegen Baclor wrote: Unfortunately not. The outbound processor only allows you to modify SIP headers but not retarget nor to drop certain messages. On 07/11/2012 02:48 PM, Claas Hilbrecht wrote: Hello, just a quick question. As I mentioned h

Re: [sipx-users] Snom 320 phones not pulling configs

I got a snom pulling config from 4.4 here. I simply followed what's on the wiki. On 07/11/2012 05:33 PM, Douglas Hubler wrote: > On Tue, Jul 10, 2012 at 9:55 PM, Kurt Albershardt wrote: >> Plugged in three Snom 320 phones after configuring MAC addresses and >> extensions in sipx 4.6 >> >> sipx

Re: [sipx-users] Bug fix release update: sipXecs 4.4.0 update #17

On 07/11/2012 06:06 PM, Claas Hilbrecht wrote: > Hello Joegen, > > thanks for the quick replay. > >> Unfortunately not. The outbound processor only allows you to modify SIP >> headers but not retarget nor to drop certain messages. > What a pitty. I thought this could be a solution to solve the ban

Re: [sipx-users] Bug fix release update: sipXecs 4.4.0 update #17

Unfortunately not. The outbound processor only allows you to modify SIP headers but not retarget nor to drop certain messages. On 07/11/2012 02:48 PM, Claas Hilbrecht wrote: > Hello, > > just a quick question. As I mentioned here I have some trouble to get speed > dials BLF working on a VPN setu

Re: [sipx-users] Source code repository

Hi Ranga, Good to hear about this. I'll be interested in contributing to this project if you decide to proceed with the Idea. I am also involved on a similar (closed source) project called Karoo Bridge and previously with the defunct OpenSBC project. Joegen On 07/08/2012 06:40 AM, M. Ranga

Re: [sipx-users] Softphone sending ACK to wrong address

On 07/05/2012 01:06 AM, Sven Evensen wrote: Yes to all your questions. And the call in the capture is from ext 11403 (Name Sven 403) to 499 which is VM. There is no 499 calling 499. This worked fine yesterday and suddenly today the Contact field in the 200 OK is wrong. The contact filed is

Re: [sipx-users] Cannot get transfer to work on unmanaged GW

sipX sends REFER to the "CME VoIP PBX Gateway" with the following Refer-To: Which should have resulted to an INVITE coming from "CME VoIP PBX Gateway" to sipX as follows INVITE sip:69143...@ec2-176-34-141-24.eu-west-1.compute.amazonaws.com SIP/2.0 X-sipX-Authidentity: s

Re: [sipx-users] new patch for XX-10177

On 06/21/2012 10:34 PM, andrewpitman wrote: Okay, I ran two versions of the Perl script against my test server for a few hours last night, both of which loop and send the crlf keepalive messages as fast as they could be sent. One was sending UDP and the other TCP. I didn't manage to get proxy

Re: [sipx-users] new patch for XX-10177

On 06/21/2012 10:34 PM, andrewpitman wrote: > > Okay, I ran two versions of the Perl script against my test > server for a few hours last night, both of which loop and > send the crlf keepalive messages as fast as they could be > sent. One was sending UDP and the other TCP. I didn't > manage to g

Re: [sipx-users] new patch for XX-10177

t and ran it continuously from a while loop > in the shell. Still no dice. > > I could make it tighter still by looping in the Perl script with no sleeps. ;) > > Andy > -- > Sent from my iPhone appendage > > On Jun 19, 2012, at 20:57, Joegen Baclor wrote: > &

Re: [sipx-users] new patch for XX-10177

, I took the 2 > second sleep out of the Perl script and ran it continuously from a while loop > in the shell. Still no dice. > > I could make it tighter still by looping in the Perl script with no sleeps. ;) > > Andy > -- > Sent from my iPhone appendage > > On Jun 19, 20

Re: [sipx-users] new patch for XX-10177

Yes that is it. From the logs that was previously posted, this was the most evident thing that was going crazy before the server was hung. So two things, 1. We are totally mistaken that this was the cause. 2. 1 packet every 5 seconds is not enough to cause havoc in the system. I am hoping

Re: [sipx-users] new patch for XX-10177

Hi Andrew, Did you check the logs if we are able to trigger the message corruption in your unpatched server? On 06/18/2012 11:20 PM, andrewpitman wrote: > > George, Joegen, > > I've been running this Perl script against an unpatched test > server for over 3 days now, and I still haven't been ab

Re: [sipx-users] Frustrating Question: OpenSBC HTTP password reset issue

Todd, I wrote OpenSBC so I guess the OP is just chasing me here. That opensipstack mailing list is no longer active. I am really surprised people are still using it. But yes, OpenSBC is indeed a bit off topic here. Joegen On 06/16/2012 01:19 PM, Todd Hodgen wrote: I believe you have po

Re: [sipx-users] Frustrating Question: OpenSBC HTTP password reset issue

It's been a lng, looong time since I last ran OpenSBC in windows. I am surprised those setup files still work at this time. Try searching for OpenSBC.ini file (or something like this) in your windows machine and delete it. On 06/16/2012 12:32 PM, Gerry Hull wrote: Hi, I've got a stra

Re: [sipx-users] new patch for XX-10177

Change PeerAddr => '127.0.0.1:5060' to the address of sipx and cron this perl script every 5 seconds. On 06/13/2012 06:34 AM, andrewpitman wrote: We've been running the patched version on our test servers since yesterday, and so far so good. Once everything checks out, we can apply this to

Re: [sipx-users] snom m3/adtran ta904

Karoo runs perfectly well on a VM. On 06/12/2012 11:02 PM, milosz wrote: >> BTW, you can use Karoo bridge to hide these funky record routes from the >> uncompliant phones. > i'll try it, an sbc seems like a useful thing to have for other > reasons, too. have people had success running it in a vm

Re: [sipx-users] snom m3/adtran ta904

BTW, you can use Karoo bridge to hide these funky record routes from the uncompliant phones. On 06/12/2012 09:33 PM, Joegen Baclor wrote: Indeed. The route header is blank and this gives it 0% chance of ever working with sipx. sipXecs record routes are divine. Thou shalt not mess with it

Re: [sipx-users] snom m3/adtran ta904

Indeed. The route header is blank and this gives it 0% chance of ever working with sipx. sipXecs record routes are divine. Thou shalt not mess with it. On 06/12/2012 02:44 PM, milosz wrote: It is probably not getting or missing an ack to indicate the call established. This would indicate

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 +

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