similar to: Problem with Cisco Phones

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.

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

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

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

> 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/ 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?

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? -

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

> 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

Hello, I am playing around with events in asterisk via asterisk manager - i've noticed it doesnt seem to be emitting events to my connected client. Is there something that I need to do to receive events? Also output from 'manager show events' voip*CLI> manager show events Events: -------------------- -------------------- -------------------- OriginateResponse

> -----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

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

Hello, I'm customizing squirrelmail and I want to print a line from the message in front of the subject in mailbox listing (like gmail). the problem is, when I use 'FETCH {SOME_ID} BODY[]' the message gets marked as seen. How can I fetch without setting the senn flag ? Thanks Behrooz

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 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

I found some postings from Google (notably from Mark Spencer) about successful integration of a legacy Toshiba Strata system and Asterisk. I am also facing that current dilemma. The general legacy solutions that I can come up with is very easy -- either making Asterisk a "proxy" (or frontdoor) to the Toshiba system, or have it operate behind the Toshiba via regular extensions. I'm

Dear List I am trying to draw a heatmap using ggplot2. In this heatmap I have faceted my data by 'infection' of which I have four. These four infections break down into two types and I would like to reorder the 'infection' column of my data to reflect this. Toy example below: library(ggplot2) # test data for ggplot reordering genes <- (rep (c(rep('a',4),