http://blog.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general hourly 1 1901-01-01T00:00+00:00 http://gmane.org/img/gmane-25t.png http://gmane.org http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12041 VG 2008-10-07T14:20:49 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12034 Hi, we have SipXecs 3.10.2 behind a Patton gateway and an pure Sip-Connect to the Provider. We also use 45 Polycom 560 with the new 3.1 Software on them. When I dial a phone number internal (let say 602) the phone rings for about 20 seconds after that I will be forwarded to the voice mailbox. But when I call the same number from the outside 0892xxx602 then the phone rings 20 seconds and after that stops ringing but for the caller it is still ringing and I do not get to the voice mailbox. After further 60 seconds the call will rejected but remains for 2 more minutes in the SipXecs GUI stated as active. We need the voicemailbox function urgent, where can the problem exist? Thanks Franz Franz Sonnenberger 2008-10-07T06:09:06 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12032 ssmith75409-spix< at >yahoo.com 2008-10-06T19:19:11 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12028 Hi, When leaving a message on an extensions voicemail, the voicemail attendant lists a couple of options. Press 0 for the operator, etc. Is it possible to remove these options and just have the voicemail system take the message? Thanks Damian Damian Dowling 2008-10-06T15:40:04 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12025 Hi, When leaving a message on an extensions voicemail, the voicemail attendant lists a couple of options. Press 0 for the operator, etc. Is it possible to remove these options and just have the voicemail system take the message? Thanks Damian Damian Dowling 2008-10-06T10:22:50 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12024 Hi, When leaving a message on an extensions voicemail, the voicemail attendant lists a couple of options. Press 0 for the operator, etc. Is it possible to remove these options and just have the voicemail system take the message? Thanks Damian Damian Dowling 2008-10-06T09:17:48 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12023 kavita gupta 2008-10-06T06:05:05 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12018 Mark 2008-10-04T05:51:21 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12016 Hi, i want to ask if it is possible to connect sipxecs 3.10 to mysipswtich. I know that 3.10 version does not support completly sip trunks and i would like to configure sipxecs to connect to mysipswitch. This way i can use 3.10 and manage my sip accounts from mysipswitch. I know that the unstable version supports sip things, but it is not enough stable :( Thanks in advanced! Regards, Ali Nebi! ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program. Ali Nebi 2008-10-03T19:08:15 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12015 Matt White 2008-10-03T18:16:53 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12013 I was wondering if there was a way we can remove the tail end of the SIP URI from caller ID coming in. e.g. Calls comes in from bandwidth.com always show up on the caller ID as +1xxxxxxxxxx< at >4.68.25.x I would love to remove the < at >4.x.x.x from the caller ID. Jermaine Pinder 2008-10-03T15:08:50 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12010 Hi, i installed on one of our servers the latest stable versipn of sipxecs and started it without any problems. But now we have a problem with registration of our users to sipxecs system. in the log we get this message: "2008-10-03T09:06:52.755784Z":21:AUTH:ERR:xxxxxx.com:SipRegistrarServer:F69BBB90:SipRegistrar:"Unable to get credentials for 'mxxxxxx< at >xxxxxxx', I did a researching in google for this probem and read that this happen because failed replications. I have a lot of failed replication jobs. How to solve them? sipxecs is installed on centos 5 (x86_64). Also what should the permissions for /etc/sipxpbx and /var/sipxpbx? I have set the owner and group to be sipxchange. I suppose this is correct, but i will be glad to hear your opinion. I will be happy if somone help me to solve this problem with failed jobs. Thanks in advanced! anebi< at >iguanait.com 2008-10-03T09:53:29 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12004 Hi! I am trying to set up an external number that, when called during working hours, goes to an auto attendant and when called after hours goes directly to an extension. I am using sipXecs 3.10. I tried doing this with a dial plan that is only effective during working hours and another one, lower in the list, that is always effective and dials the extension. I was hoping that it would use the first one during working hours and fall through to the second one at other times, but it always matches the first one. I then tried to do it using two auto attendants, because the attendant dial plan allows for working hours and after hours attendants, but the attendant always required the user to dial an extra digit. I want to go directly to an extension after hours. Is there a way to do what I want? regards, Alan van der Vyver Alan van der Vyver 2008-10-03T00:23:00 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12001 Andrew Radke 2008-10-02T07:27:46 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11996 Hi, we want to deploy the SipXecs solution in our organization (about 1,000 employees). We don't want to use commercial solutions based on SipXecs (like Nortel SCS500) because of users number limitation and because we would like to customize the solution. So we are looking for techincal support and maintenance for SipXecs (opensource solution) in Italy or Europe. Do you know anyone which offers support for SipXecs? Thanks Gmb 2008-10-01T15:44:10 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11988 Vikas Sharma 2008-10-01T10:58:30 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11987 Richard Kolkovich 2008-09-30T21:50:04 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11982 Hi everybody, I'm testing a GSM gateway with Sipx 3.10.2. When I try to setup a call to a cellular phone, the gateway doesn't respond to any SIP INVITE sent by the proxy. This behaviour happens with Grandstream and Linksys phones. The same happens using Zoiper. If I use SJphone, the call setup works well. Trying with a different SIP proxy, all the phones work well! Using tcpdump to analyze the packet flow, it seems to me that the gateway respond to INVITE requests with Proxy-Authorization string removed by the proxy, or with Proxy-Autorization not containing the "algorithm=MD5" string. Attached to this mail you can find two scenario traces, both using Grandstream phones. Can you help me to understand where is the problem (and how to solve it?). Thanks in advance, Massimo Massimo Vignone 2008-09-30T11:31:44 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11978 Vasilis Buklas 2008-09-29T06:32:44 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11975 Dean Hiller 2008-09-26T22:02:57 http://comments.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11971 Dean Hiller 2008-09-26T20:01:26 Search the mailing list at Gmane query http://search.gmane.org/?group=$group=gmane.comp.telephony.pbx.sipfoundry.general