Re: [sipx-users] ITSP call goes through but no audio

It was enabled by default. I disabled, restarted the services it asked to, and attempted an inbound call. Again I can hear the rings but I can't hear the audio when it's answered. I also tried the RTP keepalive sending an empty packet. I'll get a good trace Monday morning and send it up to the lis

Re: [sipx-users] Possible Deployment Scenario

I tried an almost identical setup with a 4.0 release and failed miserably. What will happen is your internal clients will attempt to register to the public IP and a loop will be created on the sipx box. Trust me on this one--it just won't work right. You may be able to hack various things to where

Re: [sipx-users] Possible Deployment Scenario

Assuming a private and public address? No, it will not work. Two nic's is not supported yet and there is no way to bind specific services to a specific interface. Sipx binds to all. Describing what you are trying to achieve might prove more input. It is NOT a good ide to put sipx outside with

Re: [sipx-users] ITSP call goes through but no audio

On Thu, Jul 2, 2009 at 5:18 PM, Matt Keys wrote: > > The ITSP told me it was disabled. For S&G's he enabled it for one test call > and then disabled it afterward--it had no effect. This happened before I > forcing G.729 on the phone though. It'll be Monday before I can test with > him again. What d

[sipx-users] Possible Deployment Scenario

Hi. Is it possible to deploy sipXecs with a network configurationwhere sipXecs has both a public IP address and a public IP address. Only the Trunking Service would be connected through the public IP address. Thanks! Rob ___

Re: [sipx-users] Gateway Suggestion for testing

Thanks Mike, Patton also sent me two more 5.x configs for the 4960. One with and one without registration (attached). I tested the later and it seems to be working fine on the SIP side. I'll polish them up a bit and put them on the wiki. -Jim On Thu, Jul 2, 2009 at 2:56 PM, Picher, Michael wro

Re: [sipx-users] ITSP call goes through but no audio

On Thu, 2009-07-02 at 17:18 -0400, Matt Keys wrote: > > The ITSP told me it was disabled. For S&G's he enabled it for one test > call and then disabled it afterward--it had no effect. This happened > before I forcing G.729 on the phone though. It'll be Monday before I > can test with him again. Wh

Re: [sipx-users] ITSP call goes through but no audio

The ITSP told me it was disabled. For S&G's he enabled it for one test call and then disabled it afterward--it had no effect. This happened before I forcing G.729 on the phone though. It'll be Monday before I can test with him again. What do you suggest the next test should be? Thanks, Matt

Re: [sipx-users] ITSP call goes through but no audio

On Thu, 2009-07-02 at 12:59 -0400, Matt Keys wrote: > > I'm testing a SIP trunk to Covista Communications. I'm able to receive > and send calls but there's two catches: > > 1. Placing a call from internal to my cell phone, I do not hear ring > tones but the cell phone will ring and caller ID is s

Re: [sipx-users] ITSP call goes through but no audio

The phone is a Polycom SoundPoint IP 301. I disabled all codecs other than G.729 on the phone's config page under Voice/Codecs--still no audio. The ITSP can't force G.711u. I'm not sure what else I can try. -Original Message- From: Robert Joly [mailto:rj...@nortel.com] Sent: Thu 7/2/200

Re: [sipx-users] Gateway Suggestion for testing

Here is a 5.2 config for a 4960... I've "genericized" it. -Original Message- From: Jim Canfield [mailto:jcanfi...@emstar.com] Sent: Thursday, July 02, 2009 9:32 AM To: Picher, Michael Cc: Keith Gearty; Tony Graziano; jcas...@activenetwerx.com; sipx-users@list.sipfoundry.org Subject: Re:

Re: [sipx-users] Gateway Suggestion for testing

I have one Jim... I'll shoot it to you. Mike -Original Message- From: Jim Canfield [mailto:jcanfi...@emstar.com] Sent: Thursday, July 02, 2009 9:32 AM To: Picher, Michael Cc: Keith Gearty; Tony Graziano; jcas...@activenetwerx.com; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users

Re: [sipx-users] SIP trunk between Cisco Callmanager 6 and SIPXECS

Hey Sen, It is wonderful to know that you have managed to make the sipX and call manager work together, but would you mind sharing how you made it happen?? I am having the same problem of not being able to make calls from the call manager to the sipX server. Thanks, Emery On Thu, Jul 2, 2009

Re: [sipx-users] sipx 4.0.1 broken MOH and transfers on outbound calls

Issue fixed in svn version 015877. Special thanks to Ranga for sending the updated version of sipXbridge.jar that resolved this issue for me. I am told this version will make it into the RPM's soon. Thanks, Brent On Jul 2, 2009, at 9:57 AM, Dale Worley wrote: > On Thu, 2009-07-02 at 01:

Re: [sipx-users] ITSP call goes through but no audio

> I'm testing a SIP trunk to Covista Communications. I'm able > to receive and send calls but there's two catches: > > 1. Placing a call from internal to my cell phone, I do not > hear ring tones but the cell phone will ring and caller ID is > shown properly. If I answer the cell I do not hear

[sipx-users] ITSP call goes through but no audio

I'm testing a SIP trunk to Covista Communications. I'm able to receive and send calls but there's two catches: 1. Placing a call from internal to my cell phone, I do not hear ring tones but the cell phone will ring and caller ID is shown properly. If I answer the cell I do not hear any audio i

Re: [sipx-users] SIP trunk between Cisco Callmanager 6 and SIPXECS

I just built a latest Sipx 4.1 trunking with Cisco CM 6 and 7. Both of them working fine. Sipx can call CCM, CCM can call sipx. Thanks, Sen -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Dale Worley Sent: Thu

Re: [sipx-users] wrong refer-to

On Thu, 2009-07-02 at 08:30 -0400, Matus Martinak wrote: > Hello all, > > recently my sipxpbx ( v3.8 ) start doing strange thing. others have already explained why the transfer was correct - I'm just going to jump in here and suggest that you upgrade your software. You're many releases behind. I

Re: [sipx-users] HA DNS SRV entries

On Thu, 2009-07-02 at 10:02 +0100, Gabor Paller wrote: > "It is used by the proxy on sipx1 when it wants to send a request to > "the > registrar"." > > Is it so that in HA mode, sipxproxy queries for this quasi-domain to > obtain the registrar's address? So the "rr" comes from the way sipxproxy >

Re: [sipx-users] sipx 4.0.1 broken MOH and transfers on outbound calls

On Thu, 2009-07-02 at 01:54 -0500, Backup Admin wrote: > When I dial inbound to my sipx 4.0.1 box via a SIP trunk, my polycom > 501 can transfer calls and hold with MOH, but when I call outbound via > the same SIP trunk, transfers and hold with MOH drop the calls. Both > sipx and phones are

Re: [sipx-users] HA DNS SRV entries

On Thu, 2009-07-02 at 10:02 +0100, Gabor Paller wrote: > "It is used by the proxy on sipx1 when it wants to send a request to > "the > registrar"." > > Is it so that in HA mode, sipxproxy queries for this quasi-domain to > obtain the registrar's address? So the "rr" comes from the way sipxproxy >

Re: [sipx-users] SIP trunk between Cisco Callmanager 6 and SIPXECS

The basic logic is that when the proxy receives the INVITE from Callmanager, it believes that the From URI is identifying a user on the sipXecs system. In order to get that user to present credentials, it returns a 407 response. Apparently Callmanager does not have credentials for the realm, so i

Re: [sipx-users] wrong refer-to

On Thu, 2009-07-02 at 08:30 -0400, Matus Martinak wrote: > recently my sipxpbx ( v3.8 ) start doing strange thing. > when someone call my main line, he is answered by autoattendand, then user > type extension 200 ( direct extension ), call blind transferred to phone. > Since yesterday user is back

Re: [sipx-users] SIP trunk between Cisco Callmanager 6 and SIPXECS

On Thu, 2009-07-02 at 08:45 -0500, arda savran wrote: > I attached the wireshark capture. There is only 4 messages > exchanged.You can use wordpad to open the file. After that I get a > fast busy on the Cisco side. It looks like your attachment got lost. BTW, it would help more if you got a sipXe

Re: [sipx-users] SIP trunk between Cisco Callmanager 6 and SIPXECS

Thanks, I attached the wireshark capture. There is only 4 messages exchanged.You can use wordpad to open the file. After that I get a fast busy on the Cisco side. Again, sipX-to-Cisco calls work but the other way does not work. I could not see any domain names mentioned in the attached cap

Re: [sipx-users] wrong refer-to

> Hello all, > > recently my sipxpbx ( v3.8 ) start doing strange thing. > when someone call my main line, he is answered by > autoattendand, then user type extension 200 ( direct > extension ), call blind transferred to phone. > Since yesterday user is back in autoattendand. I was doing > pack

Re: [sipx-users] Gateway Suggestion for testing

Also, I'm currently working on a 5.x config for the 4960. Once I have a functional version, I'll be sure to create a tab (and update others) on the spreadsheet and add it too the wiki. Would any of you happen to have a working 5.x 4960 config for SipX? I just created a ticket with Patton, perha

[sipx-users] wrong refer-to

Hello all, recently my sipxpbx ( v3.8 ) start doing strange thing. when someone call my main line, he is answered by autoattendand, then user type extension 200 ( direct extension ), call blind transferred to phone. Since yesterday user is back in autoattendand. I was doing packet log and had disc

Re: [sipx-users] SIP trunk between Cisco Callmanager 6 and SIPXECS

Hi, Create a custom dial plan. The number pattern should match the numbers used on CCM, the gateway should point to your trunk and only "Local Dialing" permission should be required. Regards, Gabor From: arda savran [mailto:ardasav...@hotmail.com] Sen

Re: [sipx-users] HA DNS SRV entries

"It is used by the proxy on sipx1 when it wants to send a request to "the registrar"." Is it so that in HA mode, sipxproxy queries for this quasi-domain to obtain the registrar's address? So the "rr" comes from the way sipxproxy implemented? Regards, Gabor ___

Re: [sipx-users] sipx 4.0.1 broken MOH and transfers on outbound calls

If you disable MoH in the sipX trunk does the call stay alive? -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Backup Admin Sent: Thursday, July 02, 2009 2:55 AM To: sipx-users@list.sipfoundry.org Subject: [sipx-u

Re: [sipx-users] How to integrate SIPX 4.0 with Call manager 5.0

I just replied to somebody else on the list with seemingly the same problem... I'll respond here as well just for the sake of this thread. If the traffic is coming from the IP address of the CM server, use the IP address in your gateway definition on the sipXecs side. If the traffic is coming

Re: [sipx-users] Gateway Suggestion for testing

The spreadsheet is slick. The configs in our wiki pages came out of frustration with the config notes pages on Patton's web site. http://sipx-wiki.calivia.com/index.php/HowTo_configure_Patton_SmartNode_SIP_Gateway_with_sipX Thanks, Mike -Original Message- From: Jim Canfield [mailto:

Re: [sipx-users] SIP trunk between Cisco Callmanager 6 and SIPXECS

You may want to try and capture some of that traffic with wireshark/ethereal to see how it is coming to your system from Call Manager. If the traffic is coming from the IP address of the CM server, use the IP address in your gateway definition on the sipXecs side. If the traffic is coming from