Re: [sipx-users] V.O.P tested

Their presence works very well. Actually, don't put the string in the presence configuration. Just enable presence on the individual users in your list, and it will work fine. I've found a few issues in the past with this application, and they were incredible, having them fixed the next day.

[sipx-users] V.O.P tested

Tested soft operator console called V.O.P and it works really well and they also support BLF but I have not tested this yet. www.voiceoperatorpanel.com Regards, Hiral Patel ___ sipx-users mailing list sipx-u

[sipx-users] 5.4 firmware for Audiocodes MP 114

Does anyone have firmware 5.4 for an Audiocodes MP 114 that they would like to share? I'm assuming it isn't illegal to share it. I've found 5.6 for download, but not 5.4. Anyone know of a location it can be downloaded from or feel like sharing a copy with me? If not, can anyone at least tell me

Re: [sipx-users] spa3102 for outbound calls

Hmm. The delay isn't there when I call my home phone. Just my cell phone. I got it down to about 8 seconds (calling my cell) so I guess that is ok considering this is only for an emergency. Thank you for all the tips. On 2/18/2010 7:31 PM, Eric Varsanyi wrote: > When you connected a phone to th

Re: [sipx-users] Call cancel after transfer isn't recognized

On Mon, Feb 15, 2010 at 11:15 AM, Gabor Paller wrote: > Hi, Ranga, > > I confirm that we see a similar transfer problem with sipxbridge. Our > version is 4.1.6-017914. > > I send the snapshot in a personal mail. Hi all, I have committed a fix to this problem in r18038 in SVN main. Regards,

Re: [sipx-users] spa3102 for outbound calls

Recent blog story about how much easier it was to trouble shoot with the old buttinski on our hip. http://pbx2sip.com/There is also a story there about how much market share Open Source has today (or at least January 2009). And, we forget about those simple tasks of just listening to the audi

Re: [sipx-users] Enabling IM for a user wipes out user's name

And, it wiped out the user's Unified Messaging email address setting too! On Feb 18, 2010, at 5:44 PM, Pizza Napoletana wrote: > Testing 4.1.6 IM feature. > I enabled instant messaging for a user from Users->->Instant > Messaging menu option in the GUI. > After that, I found that this action cha

[sipx-users] Patton configuration [was Re: spa3102 for outbound calls]]

If you can directly use one the configurations from Jim Canfield (and he's got a bunch of them on the wiki) its pretty easy to configure w/o understanding, especially if your'e familiar with any Cisco IOS like configuration paradigm. He's got configurations posted for the straightforward tasks t

[sipx-users] Wrong log message by sipxivr

Testing 4.1.6 I set a Primary and an Additional email address to be notified in Users->user id->Unified Messaging screen. The mails go out fine to the two separate email addresses. But if you check sipxivr.log, it reports that both emails were sent to the primary address. The log message is ina

[sipx-users] Enabling IM for a user wipes out user's name

Testing 4.1.6 IM feature. I enabled instant messaging for a user from Users->->Instant Messaging menu option in the GUI. After that, I found that this action changed the first name and the last name of that user to "unknown". ___ sipx-users mailing li

Re: [sipx-users] spa3102 for outbound calls

Gotcha. I do find it interesting that it behave the exact same way when I dialed. I will put a splitter on the port tomorrow and attach a butt set to listen in. I got your PDF. I will take a look at it. On 2/18/2010 7:31 PM, Eric Varsanyi wrote: > When you connected a phone to the 'telephone' ja

Re: [sipx-users] spa3102 for outbound calls

On Thu, 2010-02-18 at 19:18 -0600, mkitchin.pub...@gmail.com wrote: > Has there ever been an effort to configure sipx so it can configure > the Patton devices? Not that I know of. That would certainly be a welcome contribution. ___ sipx-users mailing

Re: [sipx-users] spa3102 for outbound calls

Now that I fully understand how Audiocodes devices work it only takes about 5 minutes of config to get one going for me because all I have to do is slightly modify the config file and set the IP settings, username, and password and I'm done. True the factory doesn't offer direct support but I'v

Re: [sipx-users] spa3102 for outbound calls

When you connected a phone to the 'telephone' jack on the SPA that was the FXS side, that side cares about dial plans (its the thing that gives dialtone and 'sends' the call on the SIP side when it thinks it has the whole number). For debugging I was suggesting you attach a phone in parallel to

Re: [sipx-users] spa3102 for outbound calls

Has there ever been an effort to configure sipx so it can configure the Patton devices? I'm sure they are easy to configure once you have done it once, but I was told on here the learning curve can be steep. That is why I chose the audiocodes for my first gateway i had to do. I had it done in 5

Re: [sipx-users] spa3102 for outbound calls

This thing has so many 'sides' I have probably said something incorrect at some point. I'm trying to use it as an outbound gateway only. I'm connecting a POTS line to it and want to be able to make outbound calls from sipx over a POTS line. I believe that all falls under FXO. given the differen

Re: [sipx-users] spa3102 for outbound calls

Apologies, I thought you were talking about the FXO side. 2 stage dialing for FXO is (as I understand it from the "docs") where the first portion of the dialing comes in via SIP then the user gets a dialtone from the next hop and dials manually. I didn't pay a lot of attention to that section bu

[sipx-users] SPA3102 echo cancellation [was Re: sipx-users Digest, Vol 72, Issue 146]

http://www-europe.cisco.com/en/US/products/ps10024/products_qanda_item09186a0080a359cc.shtml -Eric On Feb 18, 2010, at 6:35 PM, warner steven wrote: > > Do you possibly have a pointer to that doc? > > s > > On Feb 18, 2010, at 12:16 PM, Eric Varsanyi wrote: > >> I had echo issues with the de

Re: [sipx-users] spa3102 for outbound calls

Nope. I like to have something that senses disconnects properly. Is factory supported. That is highly configurable. Patton's that ticket for me. Imagine a text config file where you simple replace one or two pieces of information everytime you add one, and paste the config in. It's a wonderful lif

Re: [sipx-users] spa3102 for outbound calls

It is possible. They have a million options, I know that much. Pressing pound does force the call to dial immediately. I know. It isn't a great device. For the price, and for what I need it to do, it is worth a try. I have 2 audiocodes MP 114s on the way too, but I know you aren't a big fan of a

Re: [sipx-users] spa3102 for outbound calls

Do you mean to 2 different places to define the dialing plan? If not, I'm not sure what 2 stage refers to. I didn't see a PDF you sent. Did I miss something? I can't find anything. I plugged a handset into the phone port on the SPA, and it behaves the same way when I dial form a handset. On 2/18

Re: [sipx-users] Up a creek....

The problem as he originally explained it stated that AT&T was sending the signalling to 5080, if this is so, it would not. Agreed it would "IF" that was not the case. It's a little unclear from the original thread, in which AT&T stated they had set that up. We install Ingates and are very happy

Re: [sipx-users] spa3102 for outbound calls

I have a lot of clients who ask me to come in and explain those to folks, auditors too. Add to that my kids say my favorite word is "No.". I shoot so straight people put on a a Bad Company album when I walk in the room. Move over Lord Vader. On Thu, Feb 18, 2010 at 6:14 PM, Eric Varsanyi wrote:

Re: [sipx-users] Up a creek....

I would think that you absolutely could solve the problem with an Ingate... Mike From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Thursday, February 18, 2010 3:12 PM To: Andrew Cotter Cc: Sipx-users list Subject: R

Re: [sipx-users] spa3102 for outbound calls

Yeah, I wish Patton had a 1 port FXO at twice the price of an SPA3102. A Patton 1FXO+1FXS with failover relay would remove any desire to suffer at the hands of Linksys. What really kills me about Cisco/Linksys is the hardware isn't bad and the firmware would actually be OK (if a little unintuit

Re: [sipx-users] spa3102 for outbound calls

When you're the man - you're the man...I certainly wouldn't argue with his sipx advice/guidance. Even if it is: "you're an idiot - buy good equipment and set your system up right the first time" (paraphrased of course) I think I'll setup an inbound sip trunk that forwards calls to Tony - the i

Re: [sipx-users] spa3102 for outbound calls

Tony has beaten around the bush so much over the years, there is nothing to beat on. All that is left on those bushes is the cold hard facts..which is gladly shares with us. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org

Re: [sipx-users] spa3102 for outbound calls

One thing I'll give Tony - he doesn't beat around the bush :) -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Thursday, February 18, 2010 5:44 PM To: sip...@eljv.com; mkitchin.pub...@gmail.com

Re: [sipx-users] spa3102 for outbound calls

Methinks these devices suck rottens eggs. In any case, what happens if you end the dialed number with its dial string termination character (#?)? Does that speed it up? When I need a cheap device I cringe, because the time ain't worth the troubles. Tony Graziano, Mana

Re: [sipx-users] spa3102 for outbound calls

Maybe something related to the 2 stage dialing config? I didn't notice any delays like this using the config I sent in that PDF but I was just thrilled it could make calls at all and might just not have noticed the delay. Maybe plug in a butt-set or a parallel phone and listen for where the dela

Re: [sipx-users] spa3102 for outbound calls

I have everything working except what I assume is a dialing rule problem. As soon as I hit send on the Ploycom, I do see the call transferred to the IP of the SPA. If I have dialed a 11 digit number (1 + area code + xxx-) the call rings immediately. If I dial a 10 digit number (area code + xx

Re: [sipx-users] A proposed change in directed call pickup

Rolled out in production in 6 departments: http://wiki.sipfoundry.org/display/~dead_again/Home Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 2/18/2010 3:52 PM, Nathaniel Watkins wrote: > Josh - has your county actually rolled sipx out - or is it just used

Re: [sipx-users] A proposed change in directed call pickup

Via BLF, meaning when a users phone is ringing the monitoring user sees the blinking green light and presses the button to answer the ringing extensinon. Via speed dial, meaning I set up a speed dial for them that dials *78 - picks up a hunt group or extension. (For BLF, whatever the light

Re: [sipx-users] A proposed change in directed call pickup

On Thu, 2010-02-18 at 15:10 -0600, Josh Patten wrote: > The entirety of my directed call pickup scenario is done via either BLF > or speed dial. None of my users actually dial codes because I would, no > doubt, be receiving many calls from users asking what the codes were > (even though they got

Re: [sipx-users] A proposed change in directed call pickup

The entirety of my directed call pickup scenario is done via either BLF or speed dial. None of my users actually dial codes because I would, no doubt, be receiving many calls from users asking what the codes were (even though they got a documentation packet with all the info in it). I'm not sur

Re: [sipx-users] A proposed change in directed call pickup

The entirety of my directed call pickup scenario is done via either BLF or speed dial. None of my users actually dial codes because I would, no doubt, be receiving many calls from users asking what the codes were (even though they got a documentation packet with all the info in it). I'm not sur

[sipx-users] ATA Cannot Call Phone

Hello all, We have a sipx installation (4.0.4-017289 2009-11-19T18:15:29 ecs-centos5) running with multiple managed polycom stations (all working correctly). I have added a new extension and registered a line on a Sipura SPA-2002 ATA to connect to our sipx installation. When I try to call a statio

Re: [sipx-users] Up a creek....

On Thu, Feb 18, 2010 at 3:00 PM, Andrew Cotter wrote: > Anyone have a paddle? > > Thank you everyone for the suggestions of the past few weeks.  My saga with > AT&T may be coming to an end, unfortunately in a negative way.  AT&T is > starting to pull at straws since they have basically denied my r

Re: [sipx-users] Up a creek....

On Thu, Feb 18, 2010 at 3:00 PM, Andrew Cotter wrote: > Anyone have a paddle? > > Thank you everyone for the suggestions of the past few weeks.  My saga with > AT&T may be coming to an end, unfortunately in a negative way.  AT&T is > starting to pull at straws since they have basically denied my r

Re: [sipx-users] Up a creek....

On Thu, Feb 18, 2010 at 3:12 PM, Tony Graziano wrote: > > > On Thu, Feb 18, 2010 at 3:00 PM, Andrew Cotter > wrote: >> >> Anyone have a paddle? >> >> Thank you everyone for the suggestions of the past few weeks.  My saga >> with >> AT&T may be coming to an end, unfortunately in a negative way.  A

Re: [sipx-users] did inbound to conference

Well, I: NEVER create a user with anything other than the line number (Internal). It's VERY typical jobs or roles are reassigned, and DID numbers can be easily moved without a phone change or a user retrain. NEVER login as an alias. If people use their line number, they'll always know their intern

Re: [sipx-users] did inbound to conference

Agreed. The advice I got was to create a user for each DID (I named mine DID001, DID002, etc.) For each user that represents a DID number, add the number to the "Aliases". I added the number three times in the alias field, once with 7 digits (no area code), once as 10 digits (area code +

Re: [sipx-users] Up a creek....

On Thu, Feb 18, 2010 at 3:00 PM, Andrew Cotter < andrew.cot...@somersetcapital.com> wrote: > Anyone have a paddle? > > Thank you everyone for the suggestions of the past few weeks. My saga with > AT&T may be coming to an end, unfortunately in a negative way. AT&T is > starting to pull at straws

[sipx-users] Up a creek....

Anyone have a paddle? Thank you everyone for the suggestions of the past few weeks. My saga with AT&T may be coming to an end, unfortunately in a negative way. AT&T is starting to pull at straws since they have basically denied my requests for both a port change for signaling (5080 for sipXbrid

Re: [sipx-users] did inbound to conference

On Thu, Feb 18, 2010 at 2:41 PM, Michael Scheidell wrote: > > > On 2/18/10 2:36 PM, Scott Lawrence wrote: > > > See > http://wiki.sipfoundry.org/display/xecsuserV4r0/How+to+configure+Direct+Inbound+Dialing+(DID)#HowtoconfigureDirectInboundDialing(DID)-DefiningaDIDprefixDialingRule > > for how to s

Re: [sipx-users] IMAP questions, logging :: 4.1.6.017914

On Thu, Feb 18, 2010 at 2:29 PM, Tony Graziano wrote: > OK, here is a sipxivr.log (debug) file for the one voicemail and it's > deletion. I just don't see anywhere it is doing anything IMAP in here. I > also ran another one with pcap looking for port 143, there were no packets > captured (my IMAP

Re: [sipx-users] did inbound to conference

On 2/18/10 2:36 PM, Scott Lawrence wrote: See http://wiki.sipfoundry.org/display/xecsuserV4r0/How+to+configure+Direct+Inbound+Dialing+(DID)#HowtoconfigureDirectInboundDialing(DID)-DefiningaDIDprefixDialingRule for how to strip numbers from an inbound DID. yuck, dial plan rules. I did

Re: [sipx-users] did inbound to conference

On Thu, 2010-02-18 at 14:31 -0500, Michael Scheidell wrote: > > so, the inbound DID is 1561xxx > > I put that number as the extension for the conference, and any call to > that (externally) goes to that conference. > > (internally, 1561 tries to dial extension 1561) > > so, if I could strip

[sipx-users] did inbound to conference

Someone earlier asked about inbound dedicated to a conference bridge in sipx. It was suggested to create a user, have the inbound go to that user, and FWD the user to the conference extension. depending on your setup.. you might not need to do that. I created a conference with the 'extension

Re: [sipx-users] A proposed change in directed call pickup

On Thu, 2010-02-18 at 12:04 -0500, Burden, Mike wrote: > Could that be a configurable setting, so that it can be configured by > the Administrator? Configuration is the enemy of both administrative simplicity and robustness. If I could cut the number of configuration settings in the system in ha

Re: [sipx-users] IMAP questions, logging :: 4.1.6.017914

I could turn on an old imap server I have mothballed, but I can also create a pcap file on my firewall. I also think that with so many companies using email hosting, as long at their IMAP server supports IDLE, then this is a good exwrcise to go through. Tony Graziano,

Re: [sipx-users] IMAP questions, logging :: 4.1.6.017914

I have the IMAP sync working correctly in my preliminary tests (except when deleting via the web GUI), and the only messages related to IMAP that I see in the sipxivr.log file are the ones you mentioned - i.e., the smtp message being sent with imap formatting, as follows: sipxivr.log:"2010-02-1

Re: [sipx-users] A proposed change in directed call pickup

On Thu, 2010-02-18 at 12:04 -0500, Burden, Mike wrote: > Could that be a configurable setting, so that it can be configured by > the Administrator? > > That sounds like something were either way you set it, someone will want > the opposite... Technically it would be straightforward to implement,

Re: [sipx-users] IMAP questions, logging :: 4.1.6.017914

In this case I am testing against a yahoo hosted account (IMAP, port 143 is reachable, etc.). I've left two messages and deleted on via email and the other via the User portal, in either case I did not see a message from sipxivr.log that anything was syncing or changing. I would assume when the XM

Re: [sipx-users] spa3102 for outbound calls

Just my luck... Good luck I guess. I bought the audicodes device, and now I got the SPA3102 to work. Not sure what I changed that was so significant, but something did it. On 2/18/2010 12:03 PM, mkitchin.pub...@gmail.com wrote: > Thanks. I got an MP 118 working with no problems, and the 114 look

Re: [sipx-users] sipx-users Digest, Vol 72, Issue 146

I have gotten the SPA3102 running for In and Out. I can help with settings if anyone really wants it. But this device has horrible echo cancellation problems. I finally gave up and with with AudioCodes, which is has much higher audio quality. s >> >>> I'm trying to configure

Re: [sipx-users] spa3102 for outbound calls

Thanks. I got an MP 118 working with no problems, and the 114 looks about the same. I never really touched any of the settings on the Audiocodes interface, so I guess I may keep it that way. I'll keep playing with the SPA3102, but I probably should get something a little more standard and known

Re: [sipx-users] spa3102 for outbound calls

I started with an Audiocodes gateway back in October, it was the one model (FXO+FXS) that sipxecs wouldn't configure and the sipxecs configuration stuff for FXO required it to be treated as a homogenous group of ports. Two things led me to return it: 1) The documentation and manual configura

Re: [sipx-users] spa3102 for outbound calls

This ebay auction is starting to look tempting :) http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=350265531020&rvr_id=&crlp=1_263602_263622&UA=%3F*I7&GUID=d74220781250a0e203017dc1fbe784da&itemid=350265531020&ff4=263602_263622 Audio Codes MP-114 FXO VOIP Gateway - NEW US $249.99 On 2/18/2010 1

Re: [sipx-users] A proposed change in directed call pickup

I agree with Mike. I am fine with the way it currently works but I don't have a whole lot of forwarding going on. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 2/18/2010 11:04 AM, Burden, Mike wrote: > Could that be a configurable setting, so that it can b

Re: [sipx-users] spa3102 for outbound calls

For debugging if you set it up to send syslog messages and turn the level all the way up it sometimes produces semi-useful output. You don't have to have a syslog server set up to catch it if you can run tcpdump or socat. If you can capture traffic to/from the device with tcpdump that's probably

Re: [sipx-users] spa3102 for outbound calls

I have no doubt it is a PEBKAC, I just don't where to look. I changed it to 5061 (I now see that setting in the PSTN Line tab on the spa3102). The logs look about the same to me. I don't see anything that even tells me it is making it to the spa3102. On 2/18/2010 10:54 AM, Eric Varsanyi wrote:

Re: [sipx-users] A proposed change in directed call pickup

Could that be a configurable setting, so that it can be configured by the Administrator? That sounds like something were either way you set it, someone will want the opposite... Mike Burden Lynk Systems, Inc e-mail: m...@lynk.com Phone: 616-532-4985 -Original Message- From: sipx-users

[sipx-users] A proposed change in directed call pickup

We are considering changing the behavior of directed call pickup. Suppose a user, say 111, has (immediate) call forwarding set to another extension, say 222. When a call comes in for 111, the call will be sent to 222 and will start ringing there. If someone on another phone dials *78-111, the sy

Re: [sipx-users] spa3102 for outbound calls

When I set mine up late last year the only issue I had making outbound calls (that wasn't PEBKAC) was the thing didn't think there was a line attached and returned something like 'resource not avaiable' to the invite. I had to change the line voltage threshold down in the international settings

[sipx-users] spa3102 for outbound calls

I'm trying to configure a spa3102 for outbound calls only. I know people pull their hair out over these devices, but I wanted to give it a shot. My only gateways I've worked with so far are sipxbridge and an audiocodes configred from within sipx, so I haven't really done too much manual FXO con

[sipx-users] IMAP questions, logging :: 4.1.6.017914

In trialling the IMAP functionality, I am able to ONLY see message emailed to the user. I do not see any IMAP syncs occurring. I'd like to verify this, and am looking at the sipxivr.log in /var/log/sipxpbx. Is this the proper log to look for these mailer messages? The only messages relating to mai

Re: [sipx-users] calls not disconnecting

On Wed, Feb 17, 2010 at 2:27 PM, Francis Tinio wrote: > sipXconfig (4.0.4-017289 2009-11-19T18:15:29 ecs-centos5) > > > - Original Message - > From: "Robert Joly" > To: "Francis Tinio" , "sipx-users" < > sipx-users@list.sipfoundry.org> > Sent: Wednesday, February 17, 2010 2:23:07 PM GMT