Re: [sipx-users] Cluster - Number of Servers?

Eloquently stated. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Scott Lawrence Sent: Thursday, March 18, 2010 3:37 PM To: Picher, Michael Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Cluster - N

Re: [sipx-users] Polycom dialing/call-back without pressing 9...

> > > On Thu, Mar 18, 2010 at 12:23 PM, Robert B wrote: > >> Tony, >> >> I'm half-way there on following you. >> >> So, a call comes into the Polycom. I miss the call. The "Missed call" list >> shows a ten digit number (NPA-NXX and subscriber ID). If I select one of >> those numbers and press the

Re: [sipx-users] how to notify a polycom phone about the messageswaiting for another phone in the system?!?!

Thanks Mike, I am trying to see what my options are with possible deployments. One last thing I am hoping you can give me your opinion onLet's say I have 2 independent sipx servers (server A and B) in the same network. Do you think it is possible to use server B as the external voicemail fo

Re: [sipx-users] Snom 320 & 870

On Thu, Mar 18, 2010 at 9:45 PM, Scott Howell wrote: > > Content-Type: text/plain; > charset="utf-8" > Content-Transfer-Encoding: 8bit > Organization: SipXecs Forum > In-Reply-To: > X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <43541> > Message-ID: > > > > Thanks for the great info and links.

Re: [sipx-users] Snom 320 & 870

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <43542> Message-ID: One more thing! For those that are curious it appears that the Snom 370 template works fine for the 870. I ha

Re: [sipx-users] Snom 320 & 870

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <43541> Message-ID: Thanks for the great info and links. I don't know why I didn't find the info for this earlier. At any rate I

Re: [sipx-users] Polycom dialing/call-back without pressing 9...

On Thu, Mar 18, 2010 at 12:23 PM, Robert B wrote: > Tony, > > I'm half-way there on following you. > > So, a call comes into the Polycom. I miss the call. The "Missed call" list > shows a ten digit number (NPA-NXX and subscriber ID). If I select one of > those numbers and press the Dial softkey,

Re: [sipx-users] how to notify a polycom phone about the messageswaiting for another phone in the system?!?!

Voicemail must be on config server. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of arda savran Sent: Thursday, March 18, 2010 8:43 PM To: WORLEY, DALE R (DALE); sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] how to notify a po

Re: [sipx-users] how to notify a polycom phone about the messages waiting for another phone in the system?!?!

Thanks for the help guys, I am working on something that might fix our voicemail and MWI issues with SIPX. Let me ask you something; please let me know if you heard about this issue before. We have a cluster of 4 SIPX servers. The 4th one in the cluster is the dedicated voicemail server. Manage

Re: [sipx-users] Latest Version?

Cool, thanks :) On Thu, 18 Mar 2010 19:34:42 -0400, Tony Graziano wrote: > Correct amundo. >  >  > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 >  > Email: tgrazi...@myitdepartment.net >  > LAN/Telephony/Security and Control Systems Helpdesk: > T

Re: [sipx-users] Latest Version?

Correct amundo. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.

[sipx-users] Latest Version?

I had not done an update in some time and am not seeing any. Can anyone confirm that I am at the latest stable? No software updates available The package information was last updated Mar 18, 2010 6:28 PM. The system is currently running sipxcommons 4.0.4-017289 _

Re: [sipx-users] Cluster - Number of Servers?

On Thu, 2010-03-18 at 11:41 -0400, Picher, Michael wrote: > At one point it was slated for 4.2 (at least on the road map)... The 'road map' is a fluid thing, and frankly we don't always get around to updating the various web pages that purport to describe it. The tracker is the best (if still not

Re: [sipx-users] INVITE "From" information

That did it! Thanks! Mike Burden Lynk Systems, Inc e-mail: m...@lynk.com Phone: 616-532-4985 -Original Message- From: M. Ranganathan [mailto:mra...@gmail.com] Sent: Thursday, March 18, 2010 5:03 PM To: Burden, Mike Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] INV

Re: [sipx-users] INVITE "From" information

On Thu, Mar 18, 2010 at 4:39 PM, Burden, Mike wrote: > Our ITSP created an account for us on a “test” server.   Calls placed > through the “test” server fail with 404. > > > > The ITSP says that it’s because in the From field of the INVITE, we are > sending extens...@ouritsp.com (1...@ouritsp.com)

[sipx-users] INVITE "From" information

Our ITSP created an account for us on a "test" server. Calls placed through the "test" server fail with 404. The ITSP says that it's because in the From field of the INVITE, we are sending extens...@ouritsp.com (1...@ouritsp.com) instead of use...@ouritsp.com (ly...@ouritsp.com) I thought

Re: [sipx-users] grandstream gxp2010 does not work in remote NAT

Grandstream + sipXecs is just asking for problems. I fought them for a year before giving up and buying Polycom phones. -Original Message- From: JOLY, ROBERT (ROBERT) Sent: Thursday, March 18, 2010 1:45 PM To: Wang Zhiping ; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] gran

Re: [sipx-users] grandstream gxp2010 does not work in remote NAT

> > Hi, > > I think I followed all the 7 steps mentioned in > http://wiki.sipfoundry.org/display/xecsuserV4r0/Remote+User+NA > T+Traversal. > > I met the problem #3 - Remote Worker (gxp2010) registers and > can make calls but media is blocked in both directions. > > But the remote worker (x-

Re: [sipx-users] Call failures via UDP

___ From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen [thod...@verizon.net] 17/03 12:10:10.231 DEBUG1 UA [75:2xxx273807] <-- INVITE SDP 6835264022320204725 1 10.110.2.10:30002 PCMU/G729/DTMF sendre

Re: [sipx-users] Call failures via UDP

H, good question. I'll have to go out and do a packet capture to look at the packets. All I have today is a debug from the console. Which btw, is a great feature I'd love to see in more devices - like your favorite Patton gateway. -Original Message- From: Tony Graziano [mailto:tgraz

Re: [sipx-users] Call failures via UDP

The udp packet is probably too big. What size is the failed one? Methinks the vop adds more to it and they (vop) need to make available a settable threshold to it. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN

[sipx-users] grandstream gxp2010 does not work in remote NAT

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <43521> Message-ID: Hi, I think I followed all the 7 steps mentioned in http://wiki.sipfoundry.org/display/xecsuserV4r0/Remote+User+NAT+Traversa

[sipx-users] Call failures via UDP

I have a scenario where one specific company calling into a customer's console has calls that drop when answered. The thing that we find unique for these calls is that they are UDP, where TCP calls don't fail. Here is the setup - sipXecs 4.0.4. Console is a Voice Operator Panel, with a Polyco

Re: [sipx-users] Call Routing

VPN would allow you to do that also. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http:

Re: [sipx-users] Call Routing

Ok, thanks for the link. We'll work on that configuration in our lab. Ken -Original Message- From: Scott Lawrence [mailto:xmlsc...@gmail.com] Sent: Thursday, March 18, 2010 11:51 AM To: Ken Fulmer Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Call Routing On Thu, 2010-03

Re: [sipx-users] how to notify a polycom phone about the messages waiting for another phone in the system?!?!

___ From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] On Behalf Of arda savran [ardasav...@yahoo.com] I built the configuration of polycom phone B on sipx. I put the extension of phone A (4003) under the messaging section o

Re: [sipx-users] Low Voicemail Message Volume

Even with a lot of traffic on a packet capture, using the telecom feature isolates just your sip calls into a separate list that is easy to use. It's an excellent feature for this type of troubleshooting. From the graphic, you see each event for that one particular call, and when you click on it,

Re: [sipx-users] Polycom dialing/call-back without pressing 9...

In your dial plan, have you made the dial 9 mandatory? Do you in fact need that 9 dialed because of route concerns, or is it unnecessary. You can make the dialing of that prefix optional if you want in your gateway options. The 1 for long distance is optional as well if you chose. -Original

Re: [sipx-users] Cluster - Number of Servers?

I seem to remember it was in 5.0. What was in 4.2 was a feature for Branch selection of Trunk, and I think that is in there if I'm not mistaken. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Picher, Michael Se

Re: [sipx-users] Call Routing

On Thu, 2010-03-18 at 11:01 -0500, Ken Fulmer wrote: > Is there a way to route calls from system to system to the PSTN > without the servers being in a cluster? In other words, if I have a > phone attached to one server (PBX) at a corporate site, can I route > calls to the PSTN through a standalone

Re: [sipx-users] Low Voicemail Message Volume

It does feel like its root cause appears at the audiocodes, but I cannot replicate why its only affecting a single user. I'm going to try to replace the extension and give the new extension as an alias to give it a try. I'm also going to try to switch out the phone. Hopefully I can mirror some

Re: [sipx-users] Low Voicemail Message Volume

It appears you have determined that it is related to the audio codes, since calls from a SIP trunk did not experience the same issue. Are there configurations specific to that extension in your audio codes that is different than your control extension? Have you done a stare and compare of the

Re: [sipx-users] Polycom dialing/call-back without pressing 9...

For example my itsp sends me calls as +1xxxyyy, and when I send calls I have to send the same, so I add all calls at the gateway (with a trunk) to add +1. In the dialplan entry I add a custom rule to strip the +1 and send the 10 digits to the gateway so it does nit do that twice. ==

Re: [sipx-users] Polycom dialing/call-back without pressing 9...

No. Whatever the callerid shows on the missed phone is what the dialplan entry needs to have. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8

Re: [sipx-users] Polycom dialing/call-back without pressing 9...

Tony, I'm half-way there on following you. So, a call comes into the Polycom. I miss the call. The "Missed call" list shows a ten digit number (NPA-NXX and subscriber ID). If I select one of those numbers and press the Dial softkey, I get rapid busy because I have no matching dialplan on the P

Re: [sipx-users] Polycom dialing/call-back without pressing 9...

Custom dialplan in sipconfig. Prefix 9 + 11 digits Resulting action send "matched suffix". Etc. Or 91 + 10 digits Send matched suffix Be sure to place it in the appripriate place in your dialplan, activate. Phones no changes. Tony Graziano, Manager Telephone: 434.984

Re: [sipx-users] Polycom dialing/call-back without pressing 9...

Tony, Great -- please explain, because I think that's what I am asking about... :) -- Robert On 3/18/2010 10:56 AM, Tony Graziano wrote: > I think it might a lot easier to fix it with a dialplan entry to strip > that... > > Tony Graziano, Manager > Telephone: 434.98

[sipx-users] Call Routing

Is there a way to route calls from system to system to the PSTN without the servers being in a cluster? In other words, if I have a phone attached to one server (PBX) at a corporate site, can I route calls to the PSTN through a standalone server at a remote site (with no phones attached)? Than

Re: [sipx-users] Polycom dialing/call-back without pressing 9...

I think it might a lot easier to fix it with a dialplan entry to strip that... Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.98

[sipx-users] Polycom dialing/call-back without pressing 9...

Can anyone shed some light on how I can enable returning calls from the missed call list on my Polycom 550's? The CID info in the call list obviously doesn't include the "9", so when I press the dial softkey there's no matching dialplan. I suppose I could create a dialplan, but I'd rather see

Re: [sipx-users] Low Voicemail Message Volume

On Thu, Mar 18, 2010 at 11:44 AM, Picher, Michael wrote: > Wow… last ditch, delete the user, change out their phone, wipe out that > extension, create a new account with a new extension and add her old one as > an alias. > > > > If that doesn’t solve it… > > > Swap out the AC for a Patton and se

Re: [sipx-users] Low Voicemail Message Volume

Wow... last ditch, delete the user, change out their phone, wipe out that extension, create a new account with a new extension and add her old one as an alias. If that doesn't solve it... From: Lara Johnson [mailto:lcr...@ciscorp.biz] Sent: Thursday, March 18, 2010 8:51 AM To: Tony Grazia

Re: [sipx-users] Cluster - Number of Servers?

At one point it was slated for 4.2 (at least on the road map)... > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users- > boun...@list.sipfoundry.org] On Behalf Of Scott Lawrence > Sent: Thursday, March 18, 2010 7:38 AM > To: Josh Patten > Cc: sipx-users@li

Re: [sipx-users] Call Center - ACD

http://track.sipfoundry.org/secure/IssueNavigator.jspa? Plus the biggie... it is a no-no to transfer calls out of queue. > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users- > boun...@list.sipfoundry.org] On Behalf Of Francis Tinio > Sent: Thursday, Mar

Re: [sipx-users] Snom 320 & 870

Hi! Most instructions i found on the net did not work for me. dhcpd.conf below works perfectly for me, Note that you have to provide all localization unless you are using english, >From what i could find Snome 320 only has room for one language if provisioned from sipx. I added some mac mangling t

Re: [sipx-users] how to notify a polycom phone about the messages waiting for another phone in the system?!?!

There's likely a better way, but what I do with several 650's is to just put the user's line with the voice on the phone in question. On the 'watching' phones I just set them not to ring on that line. I also have these as shared appearances but that doesn't appear to work right yet (you can't pu

[sipx-users] how to notify a polycom phone about the messages waiting for another phone in the system?!?!

I am trying to subscribe a polycom soundpoint phone (phone B: extension 2002) for MWI service for another extension (phone A) on sipx. My intensions are to notify phone B as well when there is a new message waiting for phone A. I built the configuration of polycom phone B on sipx. I put the exte

Re: [sipx-users] Snom 320 & 870

Hi Scott, Snom 320 are going to get the provisioning path with DHCP options 66/67. Please read the following for further info: http://sipx-wiki.calivia.com/index.php/HowTo_configure_SNOM_SIP_phone_with_sipX Unfortunately I don't own a Snom 870. I can't tell how different is the config file. I wou

Re: [sipx-users] Error in sipproxy

2010/3/18 Nitin Mirchandani > Hello > > Sipproxy is failing giving this error - > > Standard error > >- *** stack smashing detected ***: /usr/bin/sipXproxy terminated > > Return to > services

[sipx-users] Snom 320 & 870

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <43493> Message-ID: I am just setting up my first system and doing some testing but have an issue with two Snom phones. BTW, the system passes a

[sipx-users] Error in sipproxy

Hello Sipproxy is failing giving this error - Standard error *** stack smashing detected ***: /usr/bin/sipXproxy terminated Return to services What exactly is to be done? sipXconfig (4.1.6-018058 2010-02-20T17:33:02 ecs-centos5) ___

Re: [sipx-users] Call Center - ACD

all users are currently logged in. what issues exactly does acd have? i have build 4.0.4 if that matters. On Mar 18, 2010, at 9:39 AM, Tony Graziano wrote: > If you don't have any acd users logged in it won't ring the phones. Its been > noted before the acd has some issues too. > ==

Re: [sipx-users] Call Center - ACD

If you don't have any acd users logged in it won't ring the phones. Its been noted before the acd has some issues too. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpd

[sipx-users] Call Center - ACD

Anyone using Call Center - ACD features? I can't seem to get it to work. I'm trying to implement ACD instead of call hunting in our deployment. But I can't get the calls to go through. I have anabled ACD, created a QUEUE, and assigned users to the queue. Then I created a phantom user to forw

Re: [sipx-users] Low Voicemail Message Volume

Ok, here's what I did yesterday. Sorry for not keeping completely up to the moment. I called in on every line on the AC (there's six) and left two voicemails, one on the problem user and one on a "control" extension that did not have the issues. Every call exhibited the same thing. I did call

Re: [sipx-users] How to configure for different incoming numbers?

ahh... ok i see it now. I had the assumption before that the dialplan in system are only for outgoing calls. thanks On Mar 18, 2010, at 7:53 AM, Tony Graziano wrote: > On Thu, Mar 18, 2010 at 7:46 AM, Francis Tinio wrote: > Tony, > > I'm trying to set after hours attendant. I have successf

Re: [sipx-users] How to configure for different incoming numbers?

On Thu, Mar 18, 2010 at 7:46 AM, Francis Tinio wrote: > Tony, > > I'm trying to set after hours attendant. I have successfully configured > the hunt group (ext 5010) and created a phantom user (ext 5007) with alias > on my incoming number. > I also created a schedule for weekdays to ring the hun

Re: [sipx-users] How to configure for different incoming numbers?

Tony, I'm trying to set after hours attendant. I have successfully configured the hunt group (ext 5010) and created a phantom user (ext 5007) with alias on my incoming number. I also created a schedule for weekdays to ring the hunt group. However, how do I point the after hours schedule to an

Re: [sipx-users] Cluster - Number of Servers?

On Wed, 2010-03-17 at 20:36 -0500, Josh Patten wrote: > Just as a curiosity, will the new branch features help with this > limitation? I am a little unclear what features of branching will be > working in 4.2. I remember there being talk of each site having their > own system with the ability t

Re: [sipx-users] MoH instead of ringing

I already try that but with no sucess. I have a queue with 4 people and when the 4 are unavailable the caller will get MoH however when when one (or more) agent is available and their phone starts ringing the MoH stop on the caller. So if agents are available the caller will always get ringto