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Possibly slightly off topic, has anyone ever had Cisco 79xx Series phones come up with "cannot complete conference" errors when trying to conference two calls together? This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes

Hello, I'm having a problem with a few Polycom SoundStation 6000s. Everything works fine, but they drop registration to asterisk after about maybe 30 minutes - the phone does not re-try to register and if you try to dial out on the phone it says "URI Dialing is Disabled" Has anyone else had this issue? I'm running asterisk 11.7.0. This message may be private and confidential.

> Apparently this is a known problem past v8 firmware: > http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update- > version-9/ I've done some more playing about and what I've noticed is that even when using TCP SIP on the 8.x Firmware conferencing doesn?t work - making it use UDP fixes this. So has anyone managed to get the 9.x firmware working with UDP?

Are the phones exposed to the internet (even using NAT)? If so there is a good chance these calls are not coming through your PBX but are coming in direct from someone, usually scammers. Polycom has a config option to disable accepting calls from unknown devices. No idea if Cisco has something similar. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at

Hi, In my dialplan I have the following line. same => n,Set(CALLERID(name)=Support) I am expecting this to always set the caller id name to 'Support' - however, we are getting calls come in as "Anonymous" with the number as something like "unknown at unknown" We're using Cisco 7945 phones - I possibly wonder if they are displaying this rather than asterisk

We were using G722 - I thought similarly and tried a call with alaw. Same problem occurred, any other ideas? > I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can > only do a single G729 channel, and if you require G729 for the second leg of a > conference, it will fail. This message may be private and confidential. If you have received this message in

I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can only do a single G729 channel, and if you require G729 for the second leg of a conference, it will fail. On Tue, Jan 20, 2015 at 10:03 AM, Jordan Cook - Gyron Networks < jordan.cook at gyron.net> wrote: > Possibly slightly off topic, has anyone ever had Cisco 79xx Series > phones come up with ?cannot

Next step is packet capture to see if there is a clue as to the cause of the failure in the SIP signalling. On Tue, Jan 20, 2015 at 10:41 AM, Jordan Cook - Gyron Networks < jordan.cook at gyron.net> wrote: > We were using G722 - I thought similarly and tried a call with alaw. Same > problem occurred, any other ideas? > > > I'm willing to bet you are forcing the use of

To be sure you could setup a soft phone and see if the caller ID name comes in correctly. > On Mar 10, 2015, at 8:41 AM, Jordan Cook - Gyron Networks <jordan.cook at gyron.net> wrote: > > Hi, > > In my dialplan I have the following line. > > same => n,Set(CALLERID(name)=Support) > > I am expecting this to always set the caller id name to ?Support? -

> We run a variety of 5000, 6000, and 7000 series Soundstations running > Asterisk 11.6.0 and the phones are at 4.0.3.7562. We do not see these > registration issues. Would you be willing to send the configuration from asterisk for this? This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may

> -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of Eric Wieling > Sent: 11 March 2015 17:34 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Caller ID Names > > Are the phones exposed to the internet (even using NAT)? If so there is

> Next step is packet capture to see if there is a clue as to the cause of the > failure in the SIP signalling. Right, I see the following when running SIP Debug. Looks to me like the phones are expecting the server to do the conference mixing, which I guess it would do in CallManager? <--- SIP read from TCP:xxx.xxx.xxx.xxx:50604 ---> REFER sip:xxx.xxx.xxx.xxx SIP/2.0 Via:

Apparently this is a known problem past v8 firmware: http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update-version-9/ On Tue, Jan 20, 2015 at 11:16 AM, Jordan Cook - Gyron Networks < jordan.cook at gyron.net> wrote: > > Next step is packet capture to see if there is a clue as to the cause of > the > > failure in the SIP signalling. > > Right, I

Hi folks, I'm trying to set up shared folders via symlinking and have come across a problem. I created a folder for one user, then symlinked it to another. I figured that one thing that is likely to happen at some point is that user 2 is going to decide they don't want to look at that folder any more, and will delete it, so I tried this. Much to my relief, it didn't delete the actual

I have a Windows server 2003 Domain controller with a share called DC02Data. I can connect to this share successfully from several RedHat boxes, running samba 2.2.7. However, on several other RedHat machines, running samba 3.0.9 I cannot connect properly. In fact, I actually seem able to establish a mount successfully, but cannot then view the files, as shown below: [root@ixapp01 mnt]#

Dear Sirs I would like to make you aware of the following training taking place in Chicago next month. Analysis of experiments using ASReml-R, Chicago, 24/25th April This workshop is aimed at scientist/practitioners that are interested in analyzing complex datasets by fitting linear mixed models, particularly users with experience in breeding programs and design and analysis of experiments.

I'm looking for recommendations for UK-based VoIP-PSTN gateways. They should ideally offer: - IAX connection - Multiple simultaneous calls on a single account - Lower call rates than BT Business - Auto-top up or invoicing in arrears I can find several that offer one of these facilities, but none that offer all. Thanks! -- David Gurr Congruity Ltd. Hemel Hempstead, UK

Has anyone out there used the PipeMedia/PipeCall PSTN gateway? Anything good/bad to say about it? I'm considering using them for a new customer. They seem to have good rates, good provisioning tools and (better still) give commission on usage to dealers. -- David Gurr Congruity Ltd. Fax: 0871 661 1756 Hemel Hempstead UK

Now this is really frustrating. Everything was working fine, and now it isn't ... I don't think I've changed anything that would affect this, but I guess you never can be too sure. My setup is as follows: SIP softphone (SJphone) connected to Asterisk running my Linux NAT firewall box. This is all on the internal network. Asterisk then dialing out through various means - SIP to

I'm curious as to folks experiences in selling asterisk-based solutions. In sales-speak, what are the common "compelling reasons to buy"? I can think of the following potential ones, but I'm keen to find out what seems to work in practise: - Customer wants to cut cost of calls, implements * and signs up to a VoIP/PSTN gateway - Customer wants a new PBX but doesn't want to