similar to: OT: Which SIP method to use for this specific behaviour ?

Hi All, I realise this is off topic, but its likely the best place to ask! I sent an email to snom support a few days ago but have yet to recieve a response.. Perhaps some one has found a solution to this problem already? I've searched the mailing lists and google and found nothing useful. I've also read Snom's mass deployment documentation but thats no real help in this case.

Hi, I've seen some hardphones or Softswitchs now support this sip.instance feature : http://www.softarmor.com/wgdb/docs/draft-jennings-sipping-instance-id-01.txt I don't really see any convincing use of this draft but I would be curious to share thoughts on it. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

ok, that's really all I need to know. Of course, if anyone else wants to throw in their two cents, don't let me stop you :) -Thufir On Wed, Jul 6, 2016 at 1:36 AM, Frank Vanoni <mailinglist at linuxista.com> wrote: > I'm currently using Asterisk 11.7.0 on a Raspberry Pi 2 Model B with > Ubuntu Server 14.04. > > Works fine! :-) > > Frank > > On Wed,

Does anyone have any recent news on the Virbiage FT201 IAX Hardphone? I'd *really really* like to deploy these phones instead of SIP hardphones, and I can't help but wonder if I'm going to shoot myself in the foot (or another sensitive area) by deploying a ton of SIP phones just to find the IAX Hardphones were released a week later... Thanks, Brian D'Arcy

Hello all. Now I aleady installed the Asterisk. I could make communication between 2 XLite client through Asterisk. I tryed to test the voicemail function as follow. 1, I make a call to 1001 from 1002 2, Start ringing 3, Wait untill time out for ringing If no problem, 1001 go to voicemail and unavailable message will be played. But 1001 receive a 403 forbidden massage and connection go

Hi, I've been asked sometimes to tailor call history features embeded in SIP hardphones. For example, a cutomer wanted internal call to be taken out. Another wanted calls to sorted according specific criteria. 1. Have you identified a phone offering the possibility to display as Call History, an XML list produced on a distant web server ? With this feature, you would simply have to tell the

Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to

I have a client that want to try the softphone with USB handsets route to see if hardphones will even be needed. I always push for hardphones (Polycom) so I am not sure about softphones or USB handsets. This is going to be for a 300+ seat call center onsite and many offsite, I plan on using OpenVPN for the offsite machines. Any advice on softphones, handsets, or practical experience with

Hi, This question is both technical and business related. I've got a prospective customer in France which belongs to Hotel industry. He has a lot of visitors coming from the Nederlands. I'm studying the opportunity to offer phone services to those visitors. The service I'm thinking about is plain local call termination : hotel guests cost effectively call their relatives in their

Hi, Let's say you have an Asterisk server running. Which parameters would you check to improve service continuity ? I was thinking of : - telco lines status (make sure every is up) - registered hardphones - config files backup (compare live and saved configuration files, if files differ, notifies the administration team) - systems variables (disk and CPU) - log files (trigger an alarm for

Hello, I'm new to LDAP. I've read Device class exists (oid 2.5.6.14) in rfc2256. I've heard a Pluggable Device sub-class (a device with a MAC address) also exists though I can't find its OID at the moment. 1. Does any standard class specifically defines IP Phones or SIP hardphones or ATAs or Trunk lines ? What's their OID ? 2. How does one can find by himself if such classes

Hi, I have a requirement to limit the calls to our agents via a queue to 5 minutes. I had posted this to a previous thread by name "Maximum talktime in a queue?" One work around that was suggested was to use the S(x) in the dial command to the agents, so that all calls to that extension would be terminated after x seconds. So I modified the dial command to the agent as: exten =>

06.10.2015 1:22, Joshua Colp ?????: > On 15-10-05 05:58 PM, Dmitriy Serov wrote: >> 05.10.2015 23:24, Joshua Colp ?????: >>> On 15-10-05 05:22 PM, Dmitriy Serov wrote: >>>> Hello. Do I understand correctly that the current implementation >>>> res_pjsip does not support ZRTP? >>>>

Hello Since SIP/RTP is a pain to use with road warriors who need to connect from any location over the Internet, I'd like to get them some IAX phones instead. For those of you using this protocol instead of SIP, what would you recommend as IAX hardphones and Windows (and ideally Mac) softphones? Thank you.

Hello All, I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The softphone can call the other one but I can' t hear any voice. My configuration files are below: [root at localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend

Hi, Is anyone aware of a wired sip hardphone supporting 802.1x authentication ? I've been told some Avaya and Alcatel ip phones supported 802.1x. As 802.1x is widely used with wireless hardphones, I'm wondering whether or not, 802.1x could also be valuable for wired environments. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone, I have a .csv file with the following format: uniqueID SubjectID Distance_miles Tag 1 1001 5.5 3 2 1001 7 1 3 1001 6.5 1 4 1001 5 1 5 1002

Hi, May be this helps: set.seed(25) dat1<- data.frame(ID=c("1001#01","1001#02","1001#03","1002#01","1002#02"),val=rnorm(5),stringsAsFactors=FALSE) ?dat1$ID<-as.numeric(gsub("#.*","",dat1$ID)) ?dat1 #??? ID??????? val #1 1001 -0.2118336 #2 1001 -1.0415911 #3 1001 -1.1533076 #4 1002? 0.3215315 #5 1002 -1.5001299 A.K.

I have a new wellgate 3802 unit. I have not gotten it to register with asterisk 1.2.6. My proxy setting is the correct IP in the 3802. My security config is 1001/1001 and 1002/1002 on the wellgate (simple at this time). My sip.conf has: [wellgate3802L1] type=friend dtmfmode=inband username=1001 secret=1001 host=dynamic canreinvite=yes nat=no context=wellgate [wellgate3802L2] type=friend

Dear Friends, We have implemented "Asterisk" in our organization. There are 150 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff. Can anybody suggest me that what is the best hardphone for Asterisk with low-cost? Thank you. Regards, Chandra. --------------------------------- Ring'em or ping'em. Make PC-to-phone