similar to: Useragent List

Hello All! Is there any chance of showing the real useagent in the "listclients.xsl" I like to keep track of what players and version is used. Winamp 2.x or 5.x only shows Winamp not "WinampMPEG/5.0" Now xmms useragents works like it should! Thanks John

Hello, I'm running an Asterisk 1.4.14 on a linux machine. Serving SIP Snom users. I've noticed that each time Asterisk is restarted, for the first 5-10 minutes, the SIP users can dial but cannot be dialed until each phone re-registers itself against the server. So only after the "Saved useragent...for peer 111" line appears on the Asterisk console, then the 111 user can be

Hi, I have an extension which I want to use only for x-lite, and don't want anybody to register IP phones on it. I can see that 'sip show peer 3547' shows softphone's id. Is there a way to restrict registrations on this extension by useragent id? I googled but so far couldn't find any way to do it. -- Zeeshan A Zakaria -------------- next part -------------- An HTML

Is there a way to display registered SIP useragents and sort them from CLI? I.N.

Hi list ! Is it possible to show the used Useragent of a peer that registered with Asterisk? It's being saved obviously because the console says so when a phone is registering but sip show peers doesn't show it? Is there any other way to view it? Thanks!

Hi 1- Anybody implement mgcp useragent in *. 2- Where can i get that. 3- if no then anybody can help me to write it down. Best Regards Ibrar Ahmed Project Manager. Comcept (Pvt) Ltd. Islamabad Pakistan www.com-cept.com ibrarahmed@com-cept.com abrar_@yahoo.com Ph # (Off) +92-51-111784784 Ph # (Res) +92-51-2271283 Ph # (Mob) +92-3009543001 Fax # 92-51-111784785 www.com-cept.com Pick battles that

Greetings list,this question is rather a pain in my side.. i have been trying to figure it out.. it could be simple.i have a customer with a callcenter .. we developed a CRM "Customer Relations Management" with an SIP dialers built in.the question is the following.. is it possible to force the agents (users) to use a certain UserAgent which is the one built-in our system? this way will

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Wednesday, August 22, 2007 10:51 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 37, Issue 88 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com

Hi All, I'm having problems trying to get a TE120P operational in Canada. I keep getting a congestion error when I try to make a call. I'm not sure if my switching, parity, etc is correct. I'm hoping that someone will be able to verify my config. The Telco is SaskTel, with a 10 channel 50 DDI service. Zap show channels show and ztcfg -vv looks ok and the zttool show

All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch,

Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -------------- next part

Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With SIP 3.2.X firmware (available on the Polycom download site) and Asterisk 1.6.1, Polycom phones now support a full featured BLF showing statuses of Ringing, Inuse and Online and one touch directed call pickup. On the asterisk side all that needs to be done is to add a hint to the extension and enable directed pickup.

I think I have my echo problem solved, now i need to tackle the MWI. I can't seem to get it to light up. I'm using Asterisk 1.4.14. Here's a section from my sip.conf for my test phone: [general] context=internal allowguest=no allowoverlap=no allowtransfer=yes notifyhold=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=yes vmexten=9998 at internal ;vmexten=*97

Hi All, I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 I'm not sure if the above is correct.

I hope this doesn't already exist, but I couldn't find anything to help. I am installing a brand new Asterisk server, and want to use the Polycom IP331 phones. Does anyone have any steps on how to configure these? I have softphones working just fine, but for some reason I can't find a clear step by step on provisioning the Polycoms. Any help is greatly appreciated! Mark J.

We must have had the magic version of 1.6.x then, because we increased our buddy watch limit from 8 to 48 in that version. -----Original Message----- From: Jessee J Holmes [mailto:jholmes@atacomm.com] Sent: Wednesday, October 11, 2006 8:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list Limit

I think that limit was increased in 1.6.6 or 1.6.7. -----Original Message----- From: C F [mailto:shmaltz@gmail.com] Sent: Tue 10/10/2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list On 10/10/06, Eric ManxPower Wieling <eric@fnords.org> wrote: > What I

All, I have a Polycom 650 phone, when turned on displays "Checking application". Can any give me some information as to what is wrong? I have copied the CFG files from a 601 phone to work with this 650. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070429/daadda2a/attachment.htm

Hello, I have some reports from users that occasionally DTMF will stop working in voicemail and they will have to exit the system to get it to work again. The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with Ulaw codec. This is all on an internal switched 100mb lan. Has anyone else seen anything like this? Thanks, - Brent

Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David