similar to: Receiving Multiple calls on asterisk at home

Howdy - I've been noticing a problem where I only receive a single call, before other calls go to voicemail. This only happens when the user is on the phone. I have the polycom 501's setup for 2 lines per key and 2 line keys for the first registration, which should allow for multiple calls. Anybody have any ideas what is going on? thanks Rolf Brusletto

All - I've a new system, that since it's been in production, has seen a few issues, that look like they should be fixed by upgrading asterisk @ home to the latest version. I was curious if anybody out there can tell me their experiences with this, and what to expect. Thanks, Rolf Brusletto

I can't tell if it's my system or the software, but I've been running ices-kh35 all day and now my RAM is almost gone. I have 1GB. The same behaivior happened with the ices2 CVS snapshot from three days ago. Here's the output from free -m total used free shared buffers cached Mem: 977 969 8 0 128

All, I'm using Asterisk 1.6 and using Polycom 500's with SIP firmware 2.1.3. I can not seem to get the Message Waiting Indicator to work reliably (and in my opinion correctly) with voicemail. I've got the following in my phone.cfg: <reginfo> <msg msg.bypassInstantMessage="1"> <mwi msg.mwi.1.callBack="*97"

Hi Guys Just a quick question regarding on the 501, has anyone been able to configure the transfer button and messaging buttons to work with asterisk? Can you share a configuration to do this? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060301/df504eef/attachment.htm

Hi all I'm trying to limit the maximum concurrent calls on my Asterisk to try and mitigate another problem I posted about earlier. I've edited /etc/asterisk/asterisk.conf And uncommented this line, and put a value of 60 in there: maxcalls = 60 in an effort to limit my Asterisk to 60 simultaneous calls. I did a core reload in the CLI after doing that.

Hi all - This isn't really directly Asterisk related, but has anyone successfully set up a Polycom phone to register two lines on two different Asterisk boxes? I can get the first line to register, but the second one does not. I can still place calls from that second line, which indicates to me the server, user, and secret are correct. I'm running the newest 2.6 series firmware with the

Hi All - I'm running version SIP version 1.3.4 on various IP300, IP500, and IP600 Polycom phones. I'm having a tough time with MWI. I thought I remembered somebody on the list saying that they had it working, but I can't find it in the archives now. I have all the phones configured for MWI as specified in the WIKI: ipdmid.cfg: up.oneTouchVoiceMail="1"

Hey all - I haven't run into this problem before, so I'm a little confused as to why I'm seeing it now. I've been testing with Polycom 501's in a ringall queue situation, and everything seemed to be fine. Now, if I make a call into the queue, and let it ring a few rings, then pick it up, the other phones setup in the ring-all queue, will keep ringing, even though another

Hi it's possible into Asterisk 1.6.x to limit a call at 120 mn ? after 120mn, hangup and the customer call a new time thanks olivier

Hey all, I'll give my reseller a call for support in the morning, but I usually have better/faster luck on the list. I've got a SoundPoint IP500 that I upgraded to BootROM 2.6.2 and SIP image 1.5.2 on someone elses advice, I forgot to change out the old config for the new when I loaded the image up (I guess the config changed a bunch between 1.5.2 and 1.3.1) I was prompted with an error

Hi ALL, After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk. pollmailboxes=yes -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110608/e1f59a92/attachment.htm>

Commit 501db511397f ("virtio: don't set VIRTIO_NET_HDR_F_DATA_VALID on xmit") in fact disables VIRTIO_HDR_F_DATA_VALID on receiving path too, fixing this by adding a hint (has_data_valid) and set it only on the receiving path. Cc: Rolf Neugebauer <rolf.neugebauer at docker.com> Signed-off-by: Jason Wang <jasowang at redhat.com> --- drivers/net/macvtap.c | 2 +-

Commit 501db511397f ("virtio: don't set VIRTIO_NET_HDR_F_DATA_VALID on xmit") in fact disables VIRTIO_HDR_F_DATA_VALID on receiving path too, fixing this by adding a hint (has_data_valid) and set it only on the receiving path. Cc: Rolf Neugebauer <rolf.neugebauer at docker.com> Signed-off-by: Jason Wang <jasowang at redhat.com> --- drivers/net/macvtap.c | 2 +-

What is required to get the mwi to work? Is it more of a phone subject or *? I have the mailbox= line in sip.conf, but only one extension is named, and in some of the examples, I have seen that there are two... What is that all about and how does it affect the extensions.conf and voicemail.conf? Thanks again..... Just some background as you start seeing my lists, I just started my own business

Hi, I have lookedaround and don't see this anywhere. Is there a way to tell the ip500 to not make the aural MWI blips?

On 2017?01?21? 00:45, Michael S. Tsirkin wrote: > On Fri, Jan 20, 2017 at 02:32:42PM +0800, Jason Wang wrote: >> Commit 501db511397f ("virtio: don't set VIRTIO_NET_HDR_F_DATA_VALID on >> xmit") in fact disables VIRTIO_HDR_F_DATA_VALID on receiving path too, >> fixing this by adding a hint (has_data_valid) and set it only on the >> receiving path. >>

On 2017?01?21? 00:45, Michael S. Tsirkin wrote: > On Fri, Jan 20, 2017 at 02:32:42PM +0800, Jason Wang wrote: >> Commit 501db511397f ("virtio: don't set VIRTIO_NET_HDR_F_DATA_VALID on >> xmit") in fact disables VIRTIO_HDR_F_DATA_VALID on receiving path too, >> fixing this by adding a hint (has_data_valid) and set it only on the >> receiving path. >>

Hi, I'm receiving this : [Aug 28 02:25:16] NOTICE[5895] chan_sip.c: Received SIP subscribe for peer without mailbox: 9163 I've read this : http://lists.digium.com/pipermail/asterisk-users/2008-May/211701.html I typed this: asterisk -rx "reload" asterisk -rx "voicemail show users" ... and got : default 9163 john doe 0 This "default

Hi.. I am about to replace my aging Nortel Venture system with an Asterisk system and 6 Polycom IP 501 phones, and a couple sipura 841's for less used areas. We have 3 phone lines here. One is SBC, one Vonage, and one Voipjet... One hangup is that I can't figure out how to pass through a voicemail waiting indication from SBC. This is important because my wife and her family all