[sipx-users] pickup service problem with audiocodes fxs

Hello, I have a problem with pickup service on release 4.0.1. It works fine with polycom phones , but it doesn't work with Analog phones registered through MP114 fxs. When the analog phone 503 is ringing, from polycom I dial *78503 but I get not found from SIPXecs And I don't see why

[sipx-users] Audiocodes MP118 FXO One-stage dialing with SIPX

Hi, Anyone here have any instruction of setting one-stage dialing of Audiocodes MP118 with SIPX ? Regards Jun ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe:

[sipx-users] Fw: sipx an PhonerLite

Just for info: The beta version of phonerlite site is fixed now, it registers. Looks like a good free softphone (free, not Open source). Best regards / Mit freundlichen Grüßen / Sincères salutations Paul Scheepens - Forwarded by Paul Scheepens/EPO on 01-07-2009 14:58 - Paul

Re: [sipx-users] Scalability figures

Hi Gabor, Back in February I conducted an experiment that aimed at finding how many call attempts per second (CAPS) could be achieved on a single sipXecs. The setup involved a series a SIPp scripts simulating load and was executed on a single Dell PowerEdge R300 server equipped with 2.50GHz

Re: [sipx-users] pickup service problem with audiocodes fxs

On Wed, 2009-07-01 at 13:05 +0200, heros wrote: I have a problem with pickup service on release 4.0.1. It works fine with polycom phones , but it doesn’t work with Analog phones registered through MP114 fxs. When the analog phone 503 is ringing, from polycom I dial *78503 but I get “not

[sipx-users] Freeswitch and sipXproxy services

Hi, We installed sipxecs 4.0.1 (stable) and services freeswitch and sipxproxy take 99% of the processor which is causing us problems with internal and external calls, so far I have not managed to find the reason of this problem. Any help with this problem is welcome.

[sipx-users] HA DNS SRV entries

Hi, I try to understand the High Availability DNS configuration and I am stuck with these entries. ; sipx1.example.com routing for registry/redirect service _sip._tcp.rr.sipx1.example.com. IN SRV 1 0 5070 sipx1.example.com. _sip._udp.rr.sipx1.example.com. IN SRV 3 0 5070

Re: [sipx-users] Freeswitch and sipXproxy services

Spiralling SIP requests due to an erroneous dial plan? (proxy sends back the SIP requests to itself). Check sipXproxy.log in /var/log/sipxpbx. Regards, Gabor -Original Message- From: Bernardo Ortega [mailto:jbort...@fschad.com] Sent: 01 July 2009 16:13 To: SIPxecs Support Subject:

Re: [sipx-users] HA DNS SRV entries

On Wed, 2009-07-01 at 16:26 +0100, Gabor Paller wrote: ; sipx1.example.com routing for registry/redirect service _sip._tcp.rr.sipx1.example.com. IN SRV 1 0 5070 sipx1.example.com. _sip._udp.rr.sipx1.example.com. IN SRV 3 0 5070 sipx1.example.com. _sip._tcp.rr.sipx1.example.com. IN SRV 2

Re: [sipx-users] Freeswitch and sipXproxy services

On Wed, 2009-07-01 at 11:12 -0400, Bernardo Ortega wrote: We installed sipxecs 4.0.1 (stable) and services freeswitch and sipxproxy take 99% of the processor which is causing us problems with internal and external calls, so far I have not managed to find the reason of this problem. Any help

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

I built a SIP trunk between a CUCM6 and SIPXECS server in our lab. Everything in CUCM6 regarding SIP trunk is set to non-secure standard. I can make phone calls from the phones registered to the sipxecs server to the phones registered to CUCM6 server. However, when I try the the other way

Re: [sipx-users] Phones behind Nat/Pat firewall do not work

Hey Guys thanks for all the valuable information I will reply back /i have tested them as you have directed me to... thanks, emery On Tue, Jun 30, 2009 at 6:45 PM, Picher, Michael mpic...@cmctechgroup.comwrote: Well, he said he tested behind his ASA and it didn't work... the reason it didn't

Re: [sipx-users] Problems with inbound call transfers

Ok, I finally got myself enough time to try and resolve this problem, and have made some progress. I have attached the ngrep output, but from my understanding of it it seems the topex box is not handling the SIP refer (required to transfer a call) being sent to it from sipx server .. although the

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

Hey Mike, I am able to make calls one way from phones on sipXecs server to phones on call manager but cannot make calls in the other direction, no calls from phones on call manager to phones on sipXecs server. I made call manager as the unmanaged Gateway and gave the dial plan it works only one

Re: [sipx-users] Problems with inbound call transfers

On Wed, 2009-07-01 at 19:45 +0100, Carl Anthony wrote: Ok, I finally got myself enough time to try and resolve this problem, and have made some progress. I have attached the ngrep output, but from my understanding of it it seems the topex box is not handling the SIP refer (required to

[sipx-users] Eyebeam Registers but can't make calls

Hi. I'm new to sipx. I have successfully installed 4.01 and have configured 3 phones, two GXP-2000 phones and one Eyebeam phone. The GXP's can call one another and can call the operator (100) and voice mail (101) lines. The eyebeam, however, can receive calls but cannot place them. I'm using a

Re: [sipx-users] Gateway Suggestion for testing

I just wanted to chime in here: You guys may find the configuration spreadsheet available from Patton a huge resource. It's written for the Astra 800, but the same principals apply. http://www.patton.com/resources/files/snconfig_aastra800.xls Also, look at the configuration library on Patton's

Re: [sipx-users] Eyebeam Registers but can't make calls

On Wed, 2009-07-01 at 18:14 -0600, Rob Hicks wrote: I have successfully installed 4.01 and have configured 3 phones, two GXP-2000 phones and one Eyebeam phone. The GXP's can call one another and can call the operator (100) and voice mail (101) lines. The eyebeam, however, can receive calls